+++ /dev/null
-2016-11-10 19:34 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.12.2 Released.
-
-2016-11-04 10:57 +0000 [a3614d75f6] Kevin Harwell <kharwell@digium.com>
-
- * Revert "chan_sip: Fix lastrtprx always updated"
-
- This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc.
-
- Unfortunately, the aforementioned commit caused a regression (incoming calls
- would eventually disconnect). Thus it is being removed.
-
- ASTERISK-26523 #close
- ASTERISK-25270
-
- Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d
-
-2016-10-27 18:48 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.12.1 Released.
-
-2016-10-26 07:51 +0000 [9c761b8f45] Joshua Colp <jcolp@digium.com>
-
- * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.
-
- When executing the MailboxExists dialplan application and
- MAILBOX_EXISTS dialplan function the passed in temporary voice
- mailbox was not cleared, causing it to try to free garbage.
-
- ASTERISK-26503 #close
-
- Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3
-
-2016-10-25 19:13 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.12.0 Released.
-
-2016-10-17 19:08 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.12.0-rc1 Released.
-
-2016-10-17 11:39 +0000 [546ec4b038] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Add patch to address SSL crash
-
- Addresses crashes when an attempt is made to operate on an SSL socket
- after the socket has been closed.
-
- ASTERISK-26477 #close
-
- Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002
-
-2016-10-15 04:58 +0000 [f1fd873df0] Michael Kuron <m.kuron@gmx.de>
-
- * chan_sip: Only send video on outgoing channel if incoming channel supports it
-
- Previously, the settings videosupport=always and videosupport=yes behaved
- identically and unconditionally caused a video offer to be sent in the SDP on
- an outgoing call. This was a regression introduced with commit
- 5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1.
-
- This commit restores correct behavior: videosupport=always causes a video offer
- to be sent unconditionally, while videosupport=yes will only offer video on an
- outbound channel if the incoming channel it is bridged to also supports video.
- That way, the device receiving the outgoing call can display the correct user
- interface elements for audio or video and will not unnecessarily show a blank
- video window on an audio-only call.
-
- ASTERISK-17470 #close
-
- Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae
-
-2016-10-14 00:18 +0000 [ce4cfd2eca] Corey Farrell <git@cfware.com>
-
- * Fix issues with bundled pjproject cached download.
-
- Previously when testing I had a preexisting makeopts in ASTTOPDIR. The
- ordering of configure.ac causes --with-externals-cache to be processed
- after third-party configure. In cases where the Asterisk clone is
- cleaned it would cause pjproject to be downloaded to /tmp. This
- moves processing of the externals cache and sounds cache to happen
- before third-party configure.
-
- This also addresses a possible issue with the third-party Makefile. If
- TMPDIR is set by the environment it would override the path given to
- --with-externals-cache.
-
- ASTERISK-26416
-
- Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d
-
-2016-10-12 16:24 +0000 [3c54328c57] Richard Mudgett <rmudgett@digium.com>
-
- * Audit ast_json_pack() calls for needed UTF-8 checks.
-
- Added needed UTF-8 checks before constructing json objects in various
- files for strings obtained outside the system. In this case string values
- from a channel driver's peer and not from the user setting channel
- variables.
-
- * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
- object construction.
-
- ASTERISK-26466
- Reported by: Richard Mudgett
-
- Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
-
-2016-10-12 16:20 +0000 [7f8f125738] Richard Mudgett <rmudgett@digium.com>
-
- * json: Check party id name, number, subaddresses for UTF-8.
-
- * Updated unit test as ast_json_name_number() is now NULL tolerant.
-
- ASTERISK-26466 #close
- Reported by: Richard Mudgett
-
- Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6
-
-2016-10-11 18:14 +0000 [9621c9bcbc] Richard Mudgett <rmudgett@digium.com>
-
- * json: Add UTF-8 check call.
-
- Since the json library does not make the check function public we
- recreate/copy the function in our interface module.
-
- ASTERISK-26466
- Reported by: Richard Mudgett
-
- Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99
-
-2016-10-12 17:42 +0000 [e4bb9f9a37] Richard Mudgett <rmudgett@digium.com>
-
- * aoc.c: Whitespace cleanup
-
- * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null()
- when creating the type json object. The ref is a noop.
-
- Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a
-
-2016-10-12 17:27 +0000 [bcac905bd3] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Fix clearing of pause reason string.
-
- The pause reason is not always cleared when it should be cleared.
-
- * Made set_queue_member_pause() always clear pause reason if not pausing
- with a reason string.
-
- Change-Id: I993dad19626ec017478a230e980989438b778c53
-
-2016-10-12 16:22 +0000 [ee4ae2b648] Richard Mudgett <rmudgett@digium.com>
-
- * app_minivm.c: Fix malformed ast_json_pack() call.
-
- Change-Id: I082b239022fac462666e52a14a44304748908dc0
-
-2016-10-12 16:30 +0000 [90ae4e4337] gtjoseph <gjoseph@digium.com>
-
- * res_config_mysql: Fix several issues related to recent table changes
-
- Unlike any of the other database drivers, res_config_mysql checks that
- the table definition matches the requirements for every insert and
- update statement. Since all requirements are forced to 'char', any
- column that isn't a char, like ps_contacts' expiration_time,
- qualify_timeout, etc., will throw a warning. It's kinda harmless but
- very misleading. Since no other driver does those checks on insert
- or update, they've been removed from res_config_mysql. Also, all
- the logic that actually attempted to ALTER the table to fix the issue
- has been removed. With the move to alembic, the auto-alter
- functionality is not only unnecessary, it's also dangerous.
-
- The other issue is that res_config_mysql calls the mysql_insert_id
- function inside store_mysql. Presumably the intention was to return
- the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE
- IS NON_PORTABLE AND MAY CHANGE. That value is then returned to
- config realtime as the number of rows inserted. Guess what? The value
- changed. It now only returns the number of rows inserted if there's an
- auto increment column on the table, which ps_contacts doesn't have.
- Otherwise it returns 0. So now, the insert worked but we tell config
- realtime and sorcery that no rows were inserted. That call to
- mysql_insert_id was removed and we now always return 1 if the insert
- succeeded. We're only inserting 1 row at a time anyway. If the insert
- fails, we still return -1.
-
- ASTERISK-26362 #close
- Reported-by: Carlos Chavez
-
- Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4
-
-2016-09-29 13:08 +0000 [86c15db6a1] Torrey Searle <torrey@voxbone.com>
-
- * res_fax: Fix a tight race condition causing fax to crash in audio fallback
-
- When T.38 gets rejected and G711 failback occurs there is a period of
- time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set,
- leading to a crash.
-
- Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982
-
-2016-09-30 16:29 +0000 [29b7a5b00f] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>
-
- * Add text of cdr directory into README.md for ast-db-manage
-
- Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636
-
-2016-10-06 09:58 +0000 [f919edc4e2] gtjoseph <gjoseph@digium.com>
-
- * app_dial: Add the "Q" option to set the cause on unanswered channels
-
- The "Q" option will set the cause on the unanswered channels when
- another channel answers. It overrides the default of
- ANSWERED_ELSEWHERE.
-
- NOTE: chan_sip does not support setting the cause on a CANCEL to
- anything other than ANSWERED_ELSEWHERE.
-
- ASTERISK-26446 #close
-
- Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47
-
-2016-10-10 16:59 +0000 [a884b26392] Badalyan Vyacheslav <v.badalyan@open-bs.ru>
-
- * vector: After remove element recheck index
-
- Small fix. It is necessary to double-check
- the index that we just removed because there
- is a new element.
-
- ASTERISK-26453 #close
-
- Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7
-
-2016-09-29 12:52 +0000 [349c34f72a] Torrey Searle <torrey@voxbone.com>
-
- * res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge
-
- If a bridge switched to P2P when a DTMF was in progress it
- was possible for the DTMF to continue being sent indefinitely.
-
- Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29
-
-2016-10-10 10:59 +0000 [9da3489d24] Badalyan Vyacheslav <v.badalyan@open-bs.ru>
-
- * res_pjsip_config_wizard: Memory leak in module_unload
-
- Fixed a memory leak. It removes only the first element.
- Added a useful feature in vector.h to remove all items
- under the CMP through a callback function / macro.
-
- ASTERISK-26453 #close
-
- Change-Id: I84508353463456d2495678f125738e20052da950
-
-2016-10-09 21:53 +0000 [fa2885b3ff] Badalyan Vyacheslav <v.badalyan@open-bs.ru>
-
- * cel_odbc: Fix memory leak on module unload
-
- Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715
-
-2016-10-03 11:30 +0000 [e6b0053d75] gtjoseph <gjoseph@digium.com>
-
- * bundled_pjproject: Add tests for programs used by the Makefile, et al.
-
- Added tests for bzip2, tar, patch, sed and nm to configure.ac.
-
- Set DOWNLOAD_TO_STDOUT to a working command line regardless of
- whether the download program is wget, curl or fetch.
-
- Added a 'configure.m4' file to the third-party directory which takes
- care of calling any third-party project setup. Had to move some
- pjproject_bundled stuff up in configure.ac so it was called before
- the third-party configure macro.
-
- The pjproject tarball is now downloaded to the externals_cache_dir if
- it was specified on the ./configure command line
-
- Removed regeneration of the pjproject aconfigure file. It was only
- needed for an old patch that no longer applies.
-
- Converted the tests for symbols to explicit tests since we know that
- they're now available in the bundled version. Saves a little time
- during configure.
-
- ASTERISK-26416 #close
- Reported-by: Corey Farrell
-
- Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b
-
-2016-10-05 14:53 +0000 [0dc0356e39] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Add MALLOC_DEBUG capability
-
- pjproject_bundled will now use the asterisk memory debugging APIs
- if MALLOC_DEBUG is turned on in menuselect.
-
- Because this required stubs for the executable programs and the python
- bindings, some Makefile reorganization was needed to properly handle
- the dependencies. As a result, the makefile now individually makes
- each of the pjproject libraries separately instead of making them all
- in 1 shot. The only visible change is that there are separate status
- lines printed for each library instead oif 1 for all libs. Also, the
- making of the pjproject dependency files was eliminated. They're not
- needed for building unless you're actively modifying pjproject source
- files and it makes the build process faster. Finally, any issues with
- parallel builds should be resolved again making the build faster.
-
- Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0
-
-2016-10-07 17:32 +0000 [dd873bcada] Corey Farrell <git@cfware.com>
-
- * astobj2: Add backtrace to log_bad_ao2.
-
- * Compile __ast_assert_failed unconditionally.
- * Use __ast_assert_failed to log messages from log_bad_ao2
- * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.
-
- Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751
-
-2016-10-04 16:59 +0000 [86550f9c17] gtjoseph <gjoseph@digium.com>
-
- * alembic: Allow cdr, config and voicemail to exist in the same schema
-
- cdr, config and voicemail are all separate alembic trees. Because
- alembic's default is to use a table named 'alembic_version' to store
- the current tree revision, the 3 trees can't exist in the same schema
- without stepping on each other.
-
- Now each tree uses 'alembic_version_<tree_name>' as the version table.
- Each tree's env.py script now first checks for 'alembic_version'. If
- it finds it AND its revision is in the tree's history, the script
- renames it to 'alembic_version_<tree_name>'. Regardless, the script
- then continues with the migration using 'alembic_version_<tree_name>'
- and creates that table if it's not found. The result is that if an
- existing 'alembic_version' table was found but it didn't belong to this
- tree, it's left alone and 'alembic_version_<tree_name>' is used or
- created.
-
- WARNING: If multiple trees are using the same schema, they MUST NOT
- CRU or D any objects with names that might exist in the other trees.
- An example would be 'yesno_values' type. If two trees perform
- operations on it, one tree could pull it out from under the other.
- Thankfully we currently don't share any names among cdr, config and
- voicemail.
-
- NOTE: Since the env.py scripts in each tree were identical, a common
- env.py has been placed in the ast-db-manage directory and a symlink
- to it has been placed in each tree directory.
-
- ASTERISK-24311 #close
- Reported-by: Dafi Ni
-
- Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898
-
-2016-10-05 04:25 +0000 [f166681c12] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Honor support of Symmetric Response (rport) for SIP requests.
-
- In the SIP channel driver chan_sip, the default is "auto_force_rport". When no
- NAT was detected, for example in case of IPv6, Asterisk uses the IP address
- from the headers within the SIP-REGISTER for subsequent SIP signaling. When
- the remote party specifies support for Symmetric Response (RFC 3581) via the
- parameter "rport", Asterisk should not extract the port from the SIP headers
- but reuse the port of the transport. This did not happen because of a typo.
-
- ASTERISK-26438 #close
-
- Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6
-
-2016-10-04 20:46 +0000 [430f6e5388] Michael Walton <mike@farsouthnet.com>
-
- * audiohooks: Remove redundant codec translations when using audiohooks
-
- The main frame read and write handlers in main/channel.c don't use the
- optimum placement in the processing flow for calling audiohooks
- callbacks, as far as codec translation is concerned. This change places
- the audiohooks callback code:
- * After the channel read translation if the frame is not linear before
- the translation, thereby increasing the chance that the frame is linear
- as required by audiohooks
- * Before the channel write translation if the frame is linear at this
- point
- This prevents the audiohooks code from instantiating additional
- translation paths to/from linear where a linear frame format is already
- available, saving valuable CPU cycles
-
- ASTERISK-26419
-
- Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f
-
-2016-09-29 14:02 +0000 [2449d2877c] Kevin Harwell <kharwell@digium.com>
-
- * Remove "format_ogg_opus: New format"
-
- This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c.
-
- ASTERISK-26426 #close
-
- Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5
-
-2016-09-27 16:10 +0000 [f0a2e628d6] gtjoseph <gjoseph@digium.com>
-
- * download_externals: Fix issue with re-install
-
- Needed to ignore an xmlstarlet return code for optional element.
-
- Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873
- Reported-by: Dan Jenkins
-
-2016-09-22 09:49 +0000 [5258c067ae] gtjoseph <gjoseph@digium.com>
-
- * codec_opus: Add download ability to menuselect
-
- Updated codecs/codecs.xml to add codec_opus to the external
- download list.
-
- ASTERISK-26409
-
- Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
-
-2016-07-23 14:50 +0000 [a5af8709c8] gtjoseph <gjoseph@digium.com>
-
- * codec_opus: Replace res_format_attr_opus with the one from codec_opus
-
- Preparation
-
- ASTERISK-26409
-
- Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3
- (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9)
-
-2016-07-23 15:56 +0000 [44c0c51cf1] gtjoseph <gjoseph@digium.com>
-
- * format_ogg_opus: New format
-
- Add Ogg/Opus playback support.
-
- This uses libopusfile in order to be able to read .opus files and play
- them back.
-
- Writing/recording support is not present at this time.
-
- ASTERISK-26409
-
- Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
-
-2016-09-24 19:05 +0000 [0ab443007b] gtjoseph <gjoseph@digium.com>
-
- * build_tools: Add ability to download variants to download_externals
-
- Some external packages have multiple variants that apply to different
- builds of asterisk. The DPMA for instance has a "bundled" variant that
- needs to be downloaded if asterisk was configured with
- --with-pjproject-bundled.
-
- There are 2 ways to specify variants:
-
- If you need the user to make the decision about which variant to
- download, simply create multiple menuselect "member" entries like so...
-
- <member name="res_digium_phone" displayname="..snipped..">
- <support_level>external</support_level>
- <depend>xmlstarlet</depend>
- <depend>bash</depend>
- <defaultenabled>no</defaultenabled>
- </member>
-
- <member name="res_digium_phone-bundled" displayname="..snipped..">
- <support_level>external</support_level>
- <depend>xmlstarlet</depend>
- <depend>bash</depend>
- <defaultenabled>no</defaultenabled>
- </member>
-
- Note that the second entry has "-<variant>" appended to the name.
- You can then use the existing menuselect facilities to restrict which
- members to enable or disable. Youy probably don't want the user to
- enable multiple at the same time.
-
- If you want to hide the details of the variants, the better way to
- do it is to create 1 member with "variant" elements.
-
- <member name="res_digium_phone" displayname="..snipped..">
- <support_level>external</support_level>
- <depend>xmlstarlet</depend>
- <depend>bash</depend>
- <defaultenabled>no</defaultenabled>
- <member_data>
- <downloader>
- <variants>
- <variant tag="bundled"
- condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
- </variants>
- </downloader>
- </member_data>
- </member>
-
- The condition must be a bash expression suitable for use with an "if"
- statement. Any environment variable can be used plus those available
- in makeopts.
-
- In this case, if asterisk was configured with --with-pjproject-bundled
- the bundled variant will be automatically downloaded. Otherwise the
- normal version will be downloaded.
-
- Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e
-
-2016-09-22 01:40 +0000 [a0a17a8c6f] Aaron An <anjb@ti-net.com.cn>
-
- * channels/chan_pjsip: fix HANGUPCAUSE function bug.
-
- HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered.
- This patch change the call order of ast_queue_control_data
- and ast_queue_control in chan_pjsip_incoming_response.
-
- ASTERISK-26396 #close
- Reported by: AaronAn
- Tested by: AaronAn
-
- Change-Id: Ide2d31723d8d425961e985de7de625694580be61
-
-2016-09-23 09:54 +0000 [0502675e5c] Alessandro Crespi
-
- * chan_sip: Resolve externhost not to IPv6; instead go for IPv4.
-
- For the channel driver chan_sip, you specify externhost=example.com in sip.conf
- when your Asterisk is behind a NAT and your IP address is assigned dynamically.
- Or stated differently: You do not have a static IP address to use "externaddr"
- directly. This NAT support is quite handy but just about IPv4. Previously,
- Asterisk resolved "externhost" to any IP version. When the first DNS answer
- resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and
- connection (c=). This happened in outgoing SIP-REGISTER and while answering
- SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an
- IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost".
-
- ASTERISK-18232 #close
- Reported by: Jacek Kowalski
- Tested by: Alexander Traud
- patches:
- changes.patch submitted by Alessandro Crespi
-
- Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac
-
-2016-09-20 09:42 +0000 [0056bcaebd] gtjoseph <gjoseph@digium.com>
-
- * chan_sip: Address runaway when realtime peers subscribe to mailboxes
-
- Users upgrading from asterisk 13.5 to a later version and who use
- realtime with peers that have mailboxes were experiencing runaway
- situations that manifested as a continuous stream of taskprocessor
- congestion errors, memory leaks and an unresponsive chan_sip.
-
- A related issue was that setting rtcachefriends=no NEVER worked in
- asterisk 13 (since the move to stasis). In 13.5 and earlier, when a
- peer tried to register, all of the stasis threads would block and
- chan_sip would again become unresponsive. After 13.5, the runaway
- would happen.
-
- There were a number of causes...
- * mwi_event_cb was (indirectly) calling build_peer even though calls to
- mwi_event_cb are often caused by build_peer.
- * In an effort to prevent chan_sip from being unloaded while messages
- were still in flight, destroy_mailboxes was calling
- stasis_unsubscribe_and_join but in some cases waited forever for the
- final message.
- * add_peer_mailboxes wasn't properly marking the existing mailboxes
- on a peer as "keep" so build_peer would always delete them all.
- * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
- then just creating them again.
-
- All of this was causing a flood of subscribes and unsubscribes on
- multiple threads all for the same peer and mailbox.
-
- Fixes...
- * add_peer_mailboxes now marks mailboxes correctly and build_peer only
- deletes the ones that really are no longer needed by the peer.
- * add_peer_mwi_subs now only adds subscriptions marked as "new" instead
- of unsubscribing and resubscribing everything. It also adds the peer
- object's address to the mailbox instead of its name to the subscription
- userdata so mwi_event_cb doesn't have to call build_peer.
-
- With these changes, with rtcachefriends=yes (the most common setting),
- there are no leaks, locks, loops or crashes at shutdown.
-
- rtcachefriends=no still causes leaks but at least it doesn't lock, loop
- or crash. Since making rtcachefriends=no work wasnt in scope for this
- issue, further work will have to be deferred to a separate patch.
-
- Side fixes...
- * The ast_lock_track structure had a member named "thread" which gdb
- doesn't like since it conflicts with it's "thread" command. That
- member was renamed to "thread_id".
-
- ASTERISK-25468 #close
-
- Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0
-
-2016-09-21 15:03 +0000 [323aff3a09] Joshua Colp <jcolp@digium.com>
-
- * core: Ensure presencestate subtype and message are NULL.
-
- When retrieving presence state information there is no
- guarantee that the subtype and message passed in are
- set to NULL. This change ensures they are.
-
- ASTERISK-26397 #close
-
- Change-Id: I61f8187972d5d8bbd7d6b7f4daa4f4f7e8237b23
-
-2016-09-21 10:48 +0000 [10c180760c] Joshua Colp <jcolp@digium.com>
-
- * res_odbc: Make pooling option deprecation notice more useful.
-
- This changes the notice for the deprecation of the old
- pooling options to point to the new option for doing
- pooling. This gives a clearer direction as to what to
- look into.
-
- ASTERISK-26389 #close
-
- Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10
-
-2016-09-12 07:37 +0000 [42cc267016] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * cdr_mysql: fix UTC support
-
- * Make 'cdrzone=UTC' work properly.
- * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone
-
- ASTERISK-26359 #close
-
- Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778
-
-2016-09-21 08:46 +0000 [f16ab19292] Joshua Colp <jcolp@digium.com>
-
- * odbc: Remove options that are no longer applicable.
-
- The pooling, shared_connection, limit, and idlecheck options
- are no longer used in res_odbc.
-
- ASTERISK-26389
-
- Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6
-
-2016-09-20 15:17 +0000 [c9ce299b64] Corey Farrell <git@cfware.com>
-
- * core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get.
-
- Move the function outside the conditional block that excludes
- LOW_MEMORY.
-
- ASTERISK-26273 #close
-
- Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4
-
-2016-09-20 10:05 +0000 [610eb4c189] Corey Farrell <git@cfware.com>
-
- * logger: Fix default console settings.
-
- When logger.conf is missing or invalid we should be printing notices,
- warnings and errors to the console. The logmask was incorrectly
- calculated.
-
- Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3
-
-2016-06-27 14:26 +0000 [36092ee3a0] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * sd_notify (systemd status notifications) support
-
- sd_notify() is used to notify systemd of changes to the status of the
- process. This allows the systemd daemon to know when the process
- finished loading (and thus only start another program after Asterisk has
- finished loading).
-
- To use this, use a systemd unit with 'Type=notify' for Asterisk.
-
- This commit also adds the function ast_sd_notify(), a wrapper around
- sd_notify that does nothing if not built with systemd support.
-
- Also adds support for libsystemd detection in the configure script.
-
- Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
- (cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c)
-
-2016-09-19 14:21 +0000 [9372d32100] Walter Doekes <walter+github@wjd.nu>
-
- * asterisk.c: Non-root users also get the astcanary after core restart.
-
- Without this change, a 'core restart' would kill the astcanary forever
- if you're not running as root. Both with and without this patch, the
- scheduling priority was still SCHED_RR after restart.
-
- Additionally, the astcanary is now spawned if you start with high
- priority and Asterisk doesn't get a chance to lower it. For example
- through: `chrt -r 10 sudo -u asterisk asterisk -c`
-
- Also reap killed astcanary processes on core restart.
-
- ASTERISK-26352 #close
-
- Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55
-
-2016-09-19 09:40 +0000 [e96448e991] Walter Doekes <walter+github@wjd.nu>
-
- * asterisk.c: When astcanary dies on linux, reset priority on all threads.
-
- Previously only the canary checking thread itself had its priority set
- to SCHED_OTHER. Now all threads are traversed and adjusted.
-
- ASTERISK-19867 #close
- Reported by: Xavier Hienne
-
- Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39
-
-2016-09-09 06:35 +0000 [01884a7af6] Timo Teräs <timo.teras@iki.fi>
-
- * Fix showing of swap details when sysinfo() is available
-
- If sysinfo() is available, but not sysctl() or swapctl() the
- printing code for swap buffer sizes is incorrectly omitted.
- The above condition happens with musl c-library.
-
- Fix #if rule to consider defined(HAVE_SYSINFO). And also
- remove the redundant || defined(HAVE_SYSCTL) which was
- incorrectly there to start with. Now swap information is
- displayed only if an actual libc function to get it is
- available.
-
- This also fixes warnings previously seen with musl libc:
-
- [CC] asterisk.c -> asterisk.o
- asterisk.c: In function 'handle_show_sysinfo':
- asterisk.c:773:6: warning: variable 'totalswap' set but not used
- [-Wunused-but-set-variable]
- int totalswap = 0;
- ^~~~~~~~~
- asterisk.c:770:11: warning: variable 'freeswap' set but not used
- [-Wunused-but-set-variable]
- uint64_t freeswap = 0;
- ^~~~~~~~
-
- Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca
-
-2016-09-12 18:00 +0000 [cdbad152c7] Richard Mudgett <rmudgett@digium.com>
-
- * res_config_odbc.c: Fix buffer size limitation creating invalid SQL.
-
- Creating ODBC SQL queries resulted in queries too large to fit into the
- supplied buffer. The resulting truncated buffer contained an invalid SQL
- query.
-
- * Made SQL query generation code use a thread storage buffer that can
- increase in size as needed.
-
- * Fixed bad multi-line warning messages.
-
- ASTERISK-26263 #close
- Reported by: Jeppe Ryskov Larsen
-
- Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae
-
-2016-09-14 08:42 +0000 [449719be00] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_multihomed: Change Contact port to listening port.
-
- The res_pjsip_multihomed module determines what interface and transport
- a request is going out on and updates the SIP message accordingly with
- the address information. This currently incorrectly updates the Contact
- header for connectionful protocols to the ephemeral connection port,
- instead of the bound address for the listening socket which can actually
- accept the connection back. If the remote side attempts to connect back on
- the epehemeral port it will fail.
-
- This change makes it so the port is updated to the bound port on
- connectionful protocols and is maintained on UDP (as there can be
- multiple of those).
-
- ASTERISK-26374 #close
-
- Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab
-
-2016-09-07 14:48 +0000 [4d64b176eb] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Prevent SERVFAIL from marking name server bad
-
- A name server that returns "Server Failure" is indicating only that
- the server couldn't process that particular request. We should NOT
- assume that the name server is incapable of serving other requests.
-
- Here's the scenario we've been encountering...
-
- * 2 local name servers configured in resolv.conf.
- * An OPTIONS request causes a request for A and AAAA records to go out
- to both nameservers.
- * The A responses both come back successfully resolved.
- * Because of an issue at some upstream nameserver, the AAAA responses
- for that particular query come back as "SERVFAIL" from both local
- name servers.
- * Both local servers are marked as bad and no further queries can be
- sent until the 60 second ttl expires. Only previously cached results
- can be used.
- * In this case, 60 seconds is just enough time for another OPTIONS
- request to go out to the same host so the cycle repeats.
-
- We could set the bad ttl really low but that also affects REFUSED and
- NOTAUTH which probably DO signal a real server issue. Besides, even
- a really low bad ttl would be an issue on a pbx.
-
- Although we use our own resolver in 14 and master and don't have this
- issue there, Teluu has merged this patch upstream so it's appropriate
- to cherry-pick to 14 and master to keep pjproject consistent.
-
-
- Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0
-
-2016-09-14 07:59 +0000 [1cac856e17] Joshua Colp <jcolp@digium.com>
-
- * rtp: Preserve timestamps on video frames.
-
- Currently when receiving video over RTP we store only
- a calculated samples on the frame. When starting the video
- it can take some time for this calculation to actually yield
- a value as it requires constant changing timestamps. As well
- if a video frame passes over multiple RTP packets this calculation
- will fail as the timestamp is the same as the previous RTP
- packet and the number of samples calculated will be 0.
-
- This change preserves the timestamp on the frame and allows
- it to pass through the core. When sending the video this timestamp
- is used instead of a new one being calculated.
-
- ASTERISK-26367 #close
-
- Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
-
-2016-09-14 09:51 +0000 [9df4056d70] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_transport_management: Convert time in log message to seconds.
-
- ASTERISK-26375 #close
-
- Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc
-
-2016-09-13 05:34 +0000 [98e42cc662] Steve Davies <steve@one47.co.uk>
-
- * chan_sip: Fix session timeout on retransmit of non-UDP packets
-
- Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for
- SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP
- connections, allowing the TCP layer to handle the retransmits. Unfortunately,
- this caused sessions to be terminated with a retransmit timeout becasue it
- stopped at the point of the first retrans call.
-
- This patch waits for the 64*T1 timer to expire instead.
-
- ASTERISK-19968
-
- Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204
-
-2016-09-12 12:25 +0000 [0388882cdb] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue: Fix CLI "queue show" and AMI Queues action output truncation.
-
- The output of CLI "queue show" and AMI Queues action is truncated and
- "failed to extend from 240 to 327" messages are generated if the queue
- member and interface names are lengthy.
-
- * Increase the string buffer size from 240 to 512 in order to accommodate
- for more information fields added to the output since v1.8.
-
- ASTERISK-26360 #close
- Reported by: Richard Mudgett
-
- Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d
-
-2016-09-12 03:28 +0000 [da8ba990d1] Walter Doekes <walter+github@wjd.nu>
-
- * chan_sip: Allow target refresh (Contact update) on re-INVITE.
-
- Previously, the Contact was stored only on initial INVITE and on any
- 18X and 200. That meant that after re-INVITEs from *us* the Contact
- could get updated, but after re-INVITEs from the *peer*, it did not.
-
- This changeset fixes this inconsistency, properly allowing target
- refreshes through re-INVITES (RFC3261, 12.2).
-
- If your strictrtp setting allows it, this change allows you to switch
- the source IP of a connected/calling device mid-call with a simple
- re-INVITE from the new IP.
-
- ASTERISK-26358 #close
-
- Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435
-
-2016-08-31 15:22 +0000 [e9ddab4685] Richard Mudgett <rmudgett@digium.com>
-
- * sip_to_pjsip.py: Map legacy_useroption_parsing.
-
- Map the sip.conf general section legacy_useroption_parsing to the
- new pjsip.conf global ignore_uri_user_options.
-
- ASTERISK-26316
- Reported by: Kevin Harwell
-
- Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc
-
-2016-08-29 18:08 +0000 [30af92e78d] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Add ignore_uri_user_options option.
-
- This implements the chan_sip legacy_useroption_parsing option but with a
- better name.
-
- * Made the caller-id number and redirecting number strings obtained from
- incoming SIP URI user fields always truncated at the first semicolon.
- People don't care about anything after the semicolon showing up on their
- displays even though the RFC allows the semicolon.
-
- ASTERISK-26316 #close
- Reported by: Kevin Harwell
-
- Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
-
-2016-09-09 06:26 +0000 [7ed5dc2c58] Walter Doekes <walter+github@wjd.nu>
-
- * contrib: Let safe_asterisk script continue without /dev/tty9.
-
- If you use the safe_asterisk script, it uses hardcoded defaults before
- running configurable values from /etc/asterisk/startup.d. The hardcoded
- default has TTY=9. Some containerized environments don't have such a
- TTY, and safe_asterisk would stop.
-
- The custom configuration from /etc/asterisk/startup.d/* isn't read until
- after it stopped, so changing TTY in a custom config did not help.
-
- This changeset changes safe_asterisk to continue if the TTY setting was
- untouched and /dev/tty9 and /dev/vc/9 aren't found.
-
- Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc
-
-2016-09-09 05:39 +0000 [7580a736bb] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Only invoke unidentified endpoint logic when unidentified.
-
- The code was incorrectly invoking the unidentified logic when
- an endpoint had actually been identified, causing log messages
- to be output.
-
- ASTERISK-26349 #close
-
- Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f
-
-2016-08-23 06:35 +0000 [efcfc4c1ee] Corey Farrell <git@cfware.com> (license 5909)
-
- * chan_sip: Don't allocate new RTP instances on top of old ones.
-
- In some scenarios dialog_initialize_rtp can be called multiple times on
- the same dialog. This can cause RTP instances to be leaked along with
- multiple file descriptors for each instance.
-
- This change makes it so the existing RTP instances are destroyed and
- not overwritten, stopping the memory leak.
-
- ASTERISK-26272 #close
- patches:
- ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909)
-
- Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73
-
-2016-08-16 15:34 +0000 [f1ffc22933] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Do not crash on ACKs from unknown endpoints.
-
- The endpoint identification PJSIP module is intended to identify which
- endpoint an incoming request is from. If an endpoint is not identified,
- then an artificial endpoint is used in its place when proceeding.
-
- The problem is that the ACK request type is an exception to the rule.
- The artificial endpoint is not used when processing an ACK. This results
- in the possibility of having a NULL endpoint being used further on.
-
- The reason ACK is an exception is an attempt not to spam security logs
- with unidentified requests. Presumably, you've already logged the
- unidentified request on the preceeding INVITE.
-
- Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
- didn't cause an issue. A new change in 13.10 added endpoint ACL checking
- shortly after endpoint identification. Because we are accessing a NULL
- endpoint, this ACL check resulted in a crash.
-
- The fix here is to be sure to retrieve the artificial endpoint for all
- request types. ACKs still do not generate unidentified request security
- events.
-
- ASTERISK-26264 #close
- Reported by nappsoft
-
- AST-2016-006
-
- Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703
-
-2016-09-06 11:46 +0000 [23d6ec7417] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_messaging.c: Misc cleanups and fixes.
-
- * Eliminated RAII_VAR in get_outbound_endpoint().
-
- * Simplify update_to() coding. However, this function can only be a NoOp
- because the To string can only be a URI and not a name-address formatted
- string.
-
- * Simplify update_from() coding. Also fixed a code path modifying the
- from string when the caller could still want to use the original string.
-
- * Fixed msg_data_create() incompletely removing the "pjsip:" to then add
- back the "sip:" string if needed. The code didn't handle the "pjsip:sip:"
- case because it left the colon after pjsip in the string.
-
- Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db
-
-2016-09-07 16:00 +0000 [5f19657710] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Allow global headers to be overridden.
-
- Currently when you add global headers from the dialplan both
- the header in the dialplan and the globally configured header
- are added to the resulting SIP INVITE. This change makes it
- so the headers in the dialplan take precedence and are the
- only ones added.
-
- Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
-
-2016-08-11 12:10 +0000 [206d4f57dc] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * followme: initialize all config items on reload
-
- Some configuration directives were not initialized on reload, and hence
- were not reset to default if they were removed from followme.conf.
-
- ASTERISK-26288 #close
-
- Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150
-
-2016-08-01 20:55 +0000 [117a7741c8] gtjoseph <gjoseph@digium.com>
-
- * build: Add download capability for external packages
-
- The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
- http://downloads.digium.com/pub/telephony/ are now listed in the
- "External" sections of the "Resource Modules" and "Codec Translators"
- pages in menuselect. Any that are selected will automatically be
- downloaded and installed when "make install" is run. Their LICENSE and
- README (if avaialble) files will be installed to
- ASTVARLIBDIR/documentation/thirdparty/<product_name>.
-
- Example use with codecs:
-
- The codecs/codecs.xml file is a menuselect style xml file that lists
- the codecs to be included. Their support levels are 'external', which
- triggers the download and install, and defaultenabled is no. Also
- because codec_g729a is actually in a directory named codec_g729 on the
- download server, the newly added 'member_data' element is used to
- override the default of the directory name being the package name. You
- can use the 'directory_name' attribute to keep default base URL
- (http://downloads.digium.com/pub/telephony/) but use the new directory,
- or you use the 'remote_url' attribute to specify a full URL to the
- download directory. In this case, you must still follow the same
- subdirectory naming conventions as that used for the packages located
- at 'http://downloads.digium.com/pub/telephony'.
-
- A new configure option '--with-externals-cache' was added and like
- '--with-sounds-cache' it allows the installer to cache tarballs so
- they're not downloaded every time.
-
- To assist with the download and install process, each external package
- now has a manifest.xml file that, among other things, contains a package
- version and checksums for each file in the tarball. The manifest is
- saved to both the cache directory and ASTMODDIR and together with the
- manifest.xml on the downloads site, tells the install scripts whether
- a download and/or update is needed.
-
- bash and xmlstarlet are required for downloader operation. If they're
- not installed, the external items in menuselect will be unavailable.
-
- Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
-
-2016-09-06 02:41 +0000 [d04ae7d1d8] Walter Doekes <walter+github@wjd.nu>
-
- * chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.
-
- Certain SNOM phones send so-called "optional crypto" in their SDP body.
- Regular SRTP setup looks like this:
-
- m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101
- a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
-
- SNOM-style "optional crypto" looks like this:
-
- m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101
- a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...
-
- A crypto line is supplied, but the m-line does not have SAVP.
-
- When res_srtp.so is *not* loaded, then chan_sip.so treats the optional
- crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the
- incoming call with the following message:
-
- WARNING: process_sdp: Failed to receive SDP offer/answer with
- required SRTP crypto attributes for audio
-
- For platforms that want to start providing SRTP this presents a
- compatibility problem.
-
- This changeset lets chan_sip handle the SDP as if no crypto-line was
- supplied: i.e. accept the call as regular RTP, just like it did before
- res_srtp was loaded.
-
- Now you'll get this informative warning instead:
-
- WARNING: Ignoring crypto attribute in SDP because RTP transport is
- insecure
-
- ASTERISK-23989 #close
- Reported by: Olle Johansson
-
- Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
-
-2016-09-03 16:04 +0000 [df3d0188e4] Matt Jordan <mjordan@digium.com>
-
- * apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option
-
- In any scenario in which the callee is not connected to the caller, the
- current code in app_dial will crash due to raising a Dial End Stasis
- Message after the callee channel has been hung up. This patch corrects
- the error by simply moving the explicit hangup of the callee (peer)
- channel until after the dial end message.
-
- ASTERISK-25691 #close
-
- Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d
-
-2016-09-03 16:02 +0000 [a64063cc97] Matt Jordan <mjordan@digium.com>
-
- * apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5
-
- If the callee selects option '5' using the Dial application's privacy
- (P) option, the DIALSTATUS is erroneously set to ANSWER. This option
- reflects the callee sending the caller to VoiceMail one time; the call
- is definitely *not* ANSWERed in such a scenario. With this patch, the
- DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that
- is set when the 'send to VoiceMail every time' option is set.
-
- ASTERISK-25691
-
- Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358
-
-2016-08-30 16:40 +0000 [03fc438f6e] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_registrar.c: Reduce stack usage in find_aor_name().
-
- Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09
-
-2016-08-29 18:06 +0000 [b5e753227d] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_configuration.c: Ignore repeated identify by methods.
-
- Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838
-
-2016-08-30 17:26 +0000 [9b7501b6ad] Richard Mudgett <rmudgett@digium.com>
-
- * config_global.c: Comments and a default expression adjustment.
-
- Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3
-
-2016-08-31 15:14 +0000 [3314e1cec2] Richard Mudgett <rmudgett@digium.com>
-
- * sip_to_pjsip.py: Map canreinvite as directmedia alias.
-
- Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2
-
-2016-08-31 15:37 +0000 [6372f40ba0] Richard Mudgett <rmudgett@digium.com>
-
- * sip_to_pjsip.py: Fix typo converting outboundproxy registration.
-
- Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15
-
-2016-08-31 15:13 +0000 [11eb1afd2d] Richard Mudgett <rmudgett@digium.com>
-
- * sip_to_pjsip.py: Fix comment typo and tabs.
-
- Change-Id: If35174614545727817d329c60ba4456c028941b5
-
-2016-08-31 15:56 +0000 [0f9b144c1a] Richard Mudgett <rmudgett@digium.com>
-
- * Sample configs: Eliminate false multiline comment block starts.
-
- Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
-
-2016-09-02 11:36 +0000 [8d1c535bd6] Richard Mudgett <rmudgett@digium.com>
-
- * format_cap.c: Fix CLI "core show channeltype Surrogate" crash.
-
- * Make ast_format_cap_get_names() NULL tolerant.
-
- ASTERISK-26331 #close
- Reported by: CGI.NET
-
- Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3
-
-2016-08-18 14:45 +0000 [9bca895469] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_session: segfault on already disconnected session
-
- On heavy loaded system the TCP/TLS incoming calls could be
- disconnected by pjproject while these calls are being
- processed by asterisk which could use the session's memory pools.
- If the session in the disconnected state then the session memory
- pools were already freed, so we get segfault.
-
- This patch adds a lifetime control on an INVITE session to pjproject.
- The lifetime of the session is manipulated by calling
- pjsip_inv_add_ref/pjsip_inv_dec_ref.
- This patch uses these functions to inform pjproject that the
- session is in use.
-
- This patch adds check if the session state is not disconnected
- and also checks if the memory pool is not NULL.
-
- This patch also places tasks 'session_end' and 'session_end_completion'
- into session's serializer to avoid race condition.
-
- ASTERISK-26291 #close
-
- Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
-
-2016-08-10 15:14 +0000 [63feffa126] Mark Michelson <mmichelson@digium.com>
-
- * ConfBridge: Make some announcements asynchronous.
-
- Confbridge announcements tend to block a channel while they are being
- played. In some circumstances, this is warranted since you want that
- particular channel not to hear the announcement (Example: "John Doe has
- entered the conference"). For others it makes less sense.
-
- This change first introduces methods for playing sounds asynchronously
- into the conference. This is very similar to how synchronous sounds are
- played, except the channel initiating the playback does not wait for the
- sound to complete before moving on.
-
- Asynchronous announcements are used for two circumstances:
- * Sounds played for a user after they have left the bridge
- * Sounds that play first to a single user and then the rest of the
- conference (if the channel and conference use the same language)
-
- ASTERISK-26289 #close
- Reported by Mark Michelson
-
- Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
-
-2016-08-31 12:23 +0000 [a002a4d2db] Michael Kuron <m.kuron@gmx.de>
-
- * app_mp3: Use correct buffer size and the same sample rate as the channel
-
- Previously, the buffer used for MP3 streamed from HTTP servers had a size of
- 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1
- minute. Only when the buffer is full does audio start to play.
- For MP3 files streamed from a server, that is usually not a big deal as long as
- the connection to the server is fast enough to supply that much data within a
- second or two. For MP3 live streams however, it takes 1 minute to download 1
- minute of audio, so without this change, app_mp3 wasn't really usable for MP3
- live streams.
- This commit changes the buffer size so that it covers 6 seconds of an MP3 file
- streamed from a server and 0.5 seconds of an MP3 live stream. The latter is
- identified by the use of a .m3u file extension.
-
- app_mp3 so far only supported 8 kHz audio.
- Now it always runs at the sample rate of the channel.
-
- ASTERISK-26085 #close
-
- Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0
-
-2016-08-26 10:39 +0000 [308a65fe6c] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: qualify/unqualify added/deleted realtime endpoints
-
- If the PJSIP endpoint's AOR with the permanent contact
- was deleted from the realtime storage the res_pjsip module
- continues trying to qualify this contact.
- The error 'Unable to find an endpoint to qualify contact'
- appeares every 'qualify_frequency' seconds.
- This patch deletes this contact in this case.
-
- The PJSIP endpoint's AOR with the permanent contact
- is never qualified if it is added to realtime storage
- after asterisk started.
- This patch adds qualifying for the AOR's permanent contacts
- on the first handling of this AOR.
-
- ASTERISK-26319 #close
-
- Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe
-
-2016-08-17 02:51 +0000 [2fa168348e] chris de rock <chris@derock.de>
-
- * app_macro: Consider '~~s~~' as a macro start extension.
-
- As described in issue ASTERISK-26282 the AEL parser creates macros with
- extension '~~s~~'. app_macro searches only for extension 's' so the
- created extension cannot be found. with this patch app_macro searches for
- both extensions and performs the right extension.
-
- ASTERISK-26282 #close
-
- Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb
-
-2016-08-29 07:10 +0000 [27951792c4] Etienne Lessard <elessard@proformatique.com>
-
- * pbx.c: Prevent infinite recursion in manager_show_dialplan_helper.
-
- Previously, if context A was including context B and context B was including
- context A, i.e. if there was a circular dependency between contexts, then
- calling manager_show_dialplan_helper could lead to an infinite recursion,
- resulting in a crash.
-
- This commit applies the same solution as the one implemented in the
- show_dialplan_helper function. The manager_show_dialplan_helper and
- show_dialplan_helper functions contain lots of code in common, but the former
- was missing the "infinite recursion avoidance" code.
-
- ASTERISK-26226 #close
-
- Change-Id: I1aea85133c21787226f4f8442253a93000aa0897
-
-2016-08-26 14:34 +0000 [fb82fdb013] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Disable srtp use by pjmedia
-
- The reason for the disable is that while Asterisk works fine with older
- libsrtp versions, newer versions of pjproject won't compile with them.
- Debian 6 for instance, has libsrtp 1.4.4 which is older than what
- pjproject is expecting.
-
- We don't use most of pjmedia but we DO use it for SDP negotiation.
- Luckily disabling srtp in pjmedia doesn't interfere with it's ability
- to negitiate a secure channel. The proper crypto attributes are
- negotiated in both directions.
-
- ASTERISK-26279 #close
-
- Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2
-
-2016-08-26 08:41 +0000 [847bd47ff0] Alexander Traud <pabstraud@compuserve.com>
-
- * channel: No hung-up on failing security requirements.
-
- In your Diaplan, if you specify
- same => n,Set(CHANNEL(secure_bridge_media)=1)
- same => n,Set(CHANNEL(secure_bridge_signaling)=1)
- only the SIP channel driver chan_sip supports this. All other channels drivers
- like res_pjsip fail. In case of failure, the original sRTP source code released
- the whole channel, even if not hung-up, yet. This change does not release the
- channel but instead hangs-up the channel.
-
- ASTERISK-26306
-
- Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db
-
-2016-08-20 09:04 +0000 [b59d3b48d0] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations.
-
- When using the migration script sip_to_pjsip.py, and your sip.conf is
- configured with bindaddr=::, two transports are written to pjsip.conf, one for
- 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4
- and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface
- like in chan_sip.
-
- Furthermore, the script internal functions "build_host" and "split_hostport"
- did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change
- makes sure, even such addresses are parsed correctly.
-
- ASTERISK-26309
-
- Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48
-
-2016-08-25 07:06 +0000 [f69f5cd3c4] Joshua Colp <jcolp@digium.com>
-
- * app_queue: Ensure member is removed from pending when hanging up.
-
- When dialing channels it is possible that they may not ever
- leave the not in use state (Local channels in particular) by
- the time we cancel them. If this occurs but we know they were
- dialed we explicitly remove them from the pending members
- container so that subsequent call attempts occur.
-
- ASTERISK-26299 #close
-
- Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65
-
-2016-08-04 20:11 +0000 [5cd583d7a2] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Cache global config options.
-
- We may check a global config option hundreds of times a second or more.
- Asking sorcery for the global configuration from the config files backend
- involves several allocations and container traversals. Using realtime
- without a memory cache is a lot worse because you have to lookup in the
- realtime database each time to reconstitute the sorcery object. With a
- memory cache for realtime, there is about the same amount of overhead as
- for config files. Either way, it is still fairly expensive to access the
- sorcery object that much.
-
- * Cache the global config options so we can access them faster. You must
- now always perform a res_pjsip reload to change the global options.
-
- Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7
-
-2016-08-23 11:02 +0000 [8b4b2500ee] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax: Fix deadlock in ast_channel_get_t38_state().
-
- ast_channel_get_t38_state() calls ast_channel_queryoption() with
- AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
- deadlock can happen if a channel lock is held when called.
-
- * Made ast_channel_get_t38_state() callers not hold a channel lock before
- calling.
-
- * Update ast_channel_get_t38_state() doxygen to note that no channel locks
- can be held when calling the function.
-
- ASTERISK-26203 #close
- Reported by: Etienne Lessard
-
- ASTERISK-24822 #close
- Reported by: David Brillert
-
- ASTERISK-22732 #close
- Reported by: Richard Mudgett
-
- Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
-
-2016-08-23 10:39 +0000 [e8d4f40022] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax: Fix deadlock setting FAXMODE channel variable.
-
- ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
- Unfortunately, it also introduced a deadlock potential because
- set_channel_variables() which sets FAXMODE can be called during a
- masquerade. The ast_channel_get_t38_state() which gets the value used to
- set FAXMODE cannot be called with the channel locked. As a result, local
- channels can deadlock because of how they must acquire the locks necessary
- to operate.
-
- The intent of FAXMODE is for dialplan to know how a fax was transferred
- after the fax completes. However, the previous patch sets FAXMODE to the
- channel's current T.38 state AFTER the fax has completed and where T.38
- may have already disconnected.
-
- * Set FAXMODE based upon T.38 negotiations exchanged either with the fax
- applications or the fax framehooks.
-
- ASTERISK-26203
- Reported by: Etienne Lessard
-
- ASTERISK-24822
- Reported by: David Brillert
-
- ASTERISK-22732
- Reported by: Richard Mudgett
-
- Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
-
-2016-08-22 12:31 +0000 [35cf6c7702] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax.c: Fix deadlock in fax_gateway_indicate_t38().
-
- fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be
- called with any channel locks already held. A deadlock can happen if the
- function is operating on a local channel.
-
- * Made fax_gateway_indicate_t38() unlock the channel before calling
- ast_indicate_data() since fax_gateway_indicate_t38() is always called with
- the channel locked.
-
- * Made fax_gateway_indicate_t38() return void since nothing cared about
- its return value.
-
- ASTERISK-26203
- Reported by: Etienne Lessard
-
- ASTERISK-24822
- Reported by: David Brillert
-
- ASTERISK-22732
- Reported by: Richard Mudgett
-
- Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407
-
-2016-08-23 11:16 +0000 [50b2aa506f] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax.c: Add chan locked precondition comments.
-
- Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7
-
-2016-08-23 10:42 +0000 [038cbc0215] Richard Mudgett <rmudgett@digium.com>
-
- * ast_framehook_detach() must be called with the channel locked.
-
- The framehook container could become corrupted if the channel lock is not
- held before calling.
-
- Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584
-
-2016-08-22 15:01 +0000 [88e9d05ef7] Richard Mudgett <rmudgett@digium.com>
-
- * ast_framehook_attach() must be called with the channel locked.
-
- The framehook container could become corrupted if the channel lock is not
- held before calling.
-
- Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438
-
-2016-08-24 14:42 +0000 [c9e83f6d0b] gtjoseph <gjoseph@digium.com>
-
- * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options
-
- ast_multicast_rtp_create_options now checks for NULL or empty options
-
- Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362
-
-2016-08-19 18:19 +0000 [cb8fd610e2] Corey Farrell <git@cfware.com>
-
- * Fix checks for allocation debugging.
-
- MALLOC_DEBUG should not be used to check if debugging is actually
- enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only
- indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it
- is active.
-
- Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53
-
-2016-08-10 15:14 +0000 [b8b5d52b5e] Mark Michelson <mmichelson@digium.com>
-
- * ConfBridge: Rework announcer channel methodology
-
- NOTE: This patch was submitted earlier and reverted because of a failing
- test. The test has been patched so that it adjusts for the changes here,
- so this is being resubmitted for review.
-
- One feature that confbridge has is the ability to play sounds to all
- participants in the conference. Prior to this commit, the algorithm for
- this was as follows:
-
- * Grab the playback lock
- * Push the conference announcer channel into the bridge
- * Play back the sound
- * Pull the conference announcer channel from the bridge
- * Release the playback lock
-
- The issue here is that the act of adding the playback channel to the
- bridge and removing it for each announcement is expensive. Amongst the
- expenses:
-
- * The announcer channel is imparted into the bridge, meaning a new
- thread is spun up for each playback.
- * When the announcer is added or removed from the bridge, it results
- in the BRIDGEPEER channel variable being set on all channels in the
- bridge. This requires keeping the bridge locked and locking each
- individual channel in order to set it.
- * There's also just the general overhead of adding the channel and
- removing it from the bridge. The bridge potentially has to reconfigure
- every single time
-
- With this commit, the paradigm for playing back announcements has
- shifted.
-
- * The announcer channel is now added to the bridge when the conference
- is allocated, and it is hung up when the conference is destroyed.
- * A taskprocessor is used to queue playbacks onto the announcer channel.
- This keeps the behavior from before where playbacks do not overlap.
- * The announcer channel is no longer placed into the bridge as
- departable. Since we are not constantly removing the channel from
- the bridge, it is safe to add the channel using an independent thread
- and simply hang the channel up when it is time for the conference to
- be destroyed.
-
- The use of the taskprocessor for playbacks opens up the interesting
- possibility of having asynchronous announcements played. In this commit,
- however, the behavior is still exactly the same as it previously was.
-
- ASTERISK-26289
- Reported by Mark Michelson
-
- Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0
-
-2016-08-23 05:54 +0000 [d5d7cbfcfb] Joshua Colp <jcolp@digium.com>
-
- * Revert "ConfBridge: Rework announcer channel methodology"
-
- This reverts commit 0cdeb2bfb0f4203384c08858951af3c77be8b9b3.
-
- Change-Id: I18ba73b6d4dc0b994f4ffb01ae0b6cfad36ac636
-
-2016-08-22 17:08 +0000 [c16ef02318] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Default endpoints to the "offline" status.
-
- A recent change attempted to optimize startup by not updating contact
- status. Instead, code responsible for qualifying contacts updates the
- status as it becomes known. The code even accounts for contacts/AORs
- that are not set to be qualified.
-
- The problem, though, is when there are no contacts associated with an
- endpoint. A common case is when an endpoint is set to register its
- contacts but has not done so yet. In this case, prior to registration,
- the endpoint's device state will appear to be "not in use" and hints
- associated with that device will appear to be "idle". In actuality, the
- device state and hint should both appear as "unavailable". The reason
- for the failure is that the optimization change made all persistent
- endpoint states set to "unknown".
-
- The fix here is to change the hard-coded "unknown" to be "offline"
- instead. The default state will be offline until the qualifying code
- determines that the contact is actually online. This way, if there are
- no contacts at all, then the state stays as offline, and device state
- and hints appear correctly.
-
- ASTERISK-26269 #close
- Reported by nappsoft
-
- Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a
-
-2016-08-20 14:51 +0000 [e54dcf4fd5] David M. Lee <dlee@respoke.io>
-
- * res_odbc_transaction: add dep on generic_odbc
-
- When res_odbc_transaction depended on res_odbc, it got the generic_odbc
- headers and libs implicitly. Now that it no longer depends on res_odbc,
- its dependency on generic_odbc must be explicit.
-
- Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911
-
-2016-08-20 11:18 +0000 [be38c95def] Alexander Traud <pabstraud@compuserve.com>
-
- * pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations.
-
- PJProject supports a lot of platforms even Windows, some with different defaults
- when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS,
- "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows.
-
- Because of this, if configured with just an IPv6 address/transport, PJProject
- listens to both IPv4 and IPv6. However, this is not supported by the PJProject
- team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP,
- incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only
- server which accepts incoming connections on IPv4.
-
- If you try to configure two transports, one with IPv4 and one with IPv6 on the
- same interface, as expected by the PJProject team, the IPv4 transport is not
- able to bind because the IPv6 transport listens to both already.
-
- One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide.
- Then, you are able to configure two transports, one for each IP version on the
- same interface. That way, you get a server which works with IPv4 clients and
- IPv6 clients at the same time over the same interface.
-
- Here, this change sets this parameter directly within PJProject to match the
- expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack
- servers out of the box like in chan_sip. This change was accepted by the
- PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected
- to arrive in the next version, PJProject 2.6.0. Until then, this change is
- incorporated in the bundled PJProject of Asterisk.
-
- ASTERISK-26309
-
- Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae
-
-2016-08-10 15:14 +0000 [0cdeb2bfb0] Mark Michelson <mmichelson@digium.com>
-
- * ConfBridge: Rework announcer channel methodology
-
- One feature that confbridge has is the ability to play sounds to all
- participants in the conference. Prior to this commit, the algorithm for
- this was as follows:
-
- * Grab the playback lock
- * Push the conference announcer channel into the bridge
- * Play back the sound
- * Pull the conference announcer channel from the bridge
- * Release the playback lock
-
- The issue here is that the act of adding the playback channel to the
- bridge and removing it for each announcement is expensive. Amongst the
- expenses:
-
- * The announcer channel is imparted into the bridge, meaning a new
- thread is spun up for each playback.
- * When the announcer is added or removed from the bridge, it results
- in the BRIDGEPEER channel variable being set on all channels in the
- bridge. This requires keeping the bridge locked and locking each
- individual channel in order to set it.
- * There's also just the general overhead of adding the channel and
- removing it from the bridge. The bridge potentially has to reconfigure
- every single time
-
- With this commit, the paradigm for playing back announcements has
- shifted.
-
- * The announcer channel is now added to the bridge when the conference
- is allocated, and it is hung up when the conference is destroyed.
- * A taskprocessor is used to queue playbacks onto the announcer channel.
- This keeps the behavior from before where playbacks do not overlap.
- * The announcer channel is no longer placed into the bridge as
- departable. Since we are not constantly removing the channel from
- the bridge, it is safe to add the channel using an independent thread
- and simply hang the channel up when it is time for the conference to
- be destroyed.
-
- The use of the taskprocessor for playbacks opens up the interesting
- possibility of having asynchronous announcements played. In this commit,
- however, the behavior is still exactly the same as it previously was.
-
- ASTERISK-26289
- Reported by Mark Michelson
-
- Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5
-
-2016-08-19 10:21 +0000 [b494b9f88c] Alexei Gradinari <alex2grad@gmail.com>
-
- * compilation failed with -Werror=maybe-uninitialized
-
- The compilation failed for devmode
- --enable DONT_OPTIMIZE
- --enable BETTER_BACKTRACES
- --enable DO_CRASH
- --enable TEST_FRAMEWORK
-
- res_pjsip/pjsip_configuration.c: In function dtls_handler:
- res_pjsip/pjsip_configuration.c:974:20: error:
- back may be used uninitialized in this function [-Werror=maybe-uninitialized]
- int size = strlen(front);
- ^
- cc1: all warnings being treated as errors
-
- Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
-
-2016-08-19 03:59 +0000 [a628009eb9] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Add cert_file.
-
- When using the migration script sip_to_pjsip.py, cert_file was not migrated to
- pjsip.conf. A previous change regarding this contained a copy/paste error.
-
- ASTERISK-22374
-
- Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b
-
-2016-08-18 09:21 +0000 [b1fe070d0b] Alexander Traud <pabstraud@compuserve.com>
-
- * sip.conf: tlsclientmethod is using sslv23 as default.
-
- When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL
- SSLv23_method. This was documented incorrectly in the file sip.conf.sample.
-
- SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method
- enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that
- function should have been called 'secure_method' or 'automatic_method' back in
- the 90s.
-
- Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if
- you face a server which has problems like not falling back to TLSv1.0
- automatically.
-
- ASTERISK-24425
-
- Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3
-
-2016-08-18 17:16 +0000 [ff2378c735] Kevin Harwell <kharwell@digium.com>
-
- * rest-api: Swagger scripts were not replacing format variable in file brief
-
- Given resource paths did not have 'json' substituted in for the '{format}'. For
- some auto generated documentation/comment strings it resulted in something like
- the following:
-
- "... REST handler for /api-docs/sounds.{format}"
-
- This patch makes sure the resource api's path is properly substituted.
-
- ASTERISK-25472 #close
-
- Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23
-
-2016-08-16 15:57 +0000 [43f400ef95] Jason Parker (license 4993)
-
- * res_format_attr_g729: Add annexb=no format parameter to SDPs
-
- Historically, Asterisk has always specified annexb=no for the g729 format.
- However, when using res_pjsip no format attribute was specified. This patch
- makes it so the SDP now contains a format attribute line with annexb=no.
-
- Note, that this means only g729a is negotiated. Even for pass through support.
- According to rfc7261 the type of annex used (a or b) is dependent upon the
- answerer. However, Asterisk being a back to back user agent makes this tricky
- to support at this time, thus we only allow annex 'a' for now.
-
- ASTERISK-26228 #close
- patches:
- res_format_attr_g729.c submitted by Jason Parker (license 4993)
-
- Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0
-
-2016-08-18 15:15 +0000 [4c1ae07d51] gtjoseph <gjoseph@digium.com>
-
- * res_odbc: Correct the dependency relationship with res_odbc_transaction
-
- The MODULEINFO dependencies between these 2 modules was reversed.
- res_odbc should depend on res_odbc_transaction, not the other way
- around.
-
- ASTERISK-25984 #close
-
- Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f
-
-2016-08-18 12:04 +0000 [cab6975b02] Kevin Harwell <kharwell@digium.com>
-
- * sip_to_pjsip: Set correct tls transport method
-
- A recent update had a copy/paste error where the unused variable 'val' was
- being passed to the set_value function instead of the 'method' value itself.
-
- This patch passes in the right variable.
-
- ASTERISK-22374
-
- Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06
-
-2016-08-18 08:19 +0000 [2381ddde63] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Map the TLS method correctly.
-
- When using the migration script sip_to_pjsip.py and tlsclientmethod is not set
- in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to
- overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is
- offering/using not just TLSv1.0 but TLSv1.2 as well.
-
- ASTERISK-22374
-
- Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f
-
-2016-08-18 08:17 +0000 [6500f5e138] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent.
-
- When using the migration script sip_to_pjsip.py, no section of type=system or
- type=general were created. Therefore the keys compactheaders, timerb, timert1,
- and useragent were not migrated to pjsip.conf.
-
- ASTERISK-22374
-
- Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1
-
-2016-08-18 08:16 +0000 [21e9c69e56] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Map (session-)timers correctly.
-
- When using the migration script sip_to_pjsip.py, session-timers=accept and
- session-timers=refuse were mapped to wrong values.
-
- ASTERISK-22374
-
- Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092
-
-2016-08-18 08:15 +0000 [c9a97398f7] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Write username even without authname.
-
- When using the migration script sip_to_pjsip.py, now the (mandatory) username is
- written to pjsip.conf, even if there was no (optional) authname in the register
- string in sip.conf.
-
- ASTERISK-22374
-
- Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f
-
-2016-08-18 08:14 +0000 [60275359bc] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Parse register even with transport.
-
- When using the migration script sip_to_pjsip.py and the register string
- started with a transport in sip.conf - like tls://... - register was not parsed
- correctly and therefore not migrated correctly to pjsip.conf.
-
- ASTERISK-22374
-
- Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2
-
-2016-08-18 08:13 +0000 [0d479232eb] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit.
-
- When using the migration script sip_to_pjsip.py, those keys got missing. These
- keys might appear several times and the function "merge_value" tried to collect
- those. However, because these keys have different names in sip.conf and
- pjsip.conf, "merge_value" was not able to find the new key name in sip.conf.
- This change lets "merge_value" search with the old key name in sip.conf and
- write with the new key name in pjsip.conf.
-
- ASTERISK-22374
-
- Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2
-
-2016-08-18 08:11 +0000 [cbc1b2d020] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Map externhost/ip to Transports.
-
- When using the migration script sip_to_pjsip.py, the externhost or externip of
- sip.conf were erroneously written to Endpoints instead to Transports.
-
- ASTERISK-22374
-
- Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4
-
-2016-08-18 08:04 +0000 [5f33e99534] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry.
-
- When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
- minexpiry were not migrated to pjsip.conf.
-
- ASTERISK-22374
-
- Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b
-
-2016-08-18 08:03 +0000 [231ea0350d] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Write media_encryption.
-
- When using the migration script sip_to_pjsip.py, encryption=yes got missing and
- media_encryption=sdes was not written to pjsip.conf, because of a typo.
-
- ASTERISK-22374
-
- Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05
-
-2016-08-18 08:02 +0000 [23eb065121] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Write cos and tos.
-
- When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got
- missed, because of a typo. Therefore, cos and tos were not written to
- pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused
- by a copy-and-paste error.
-
- ASTERISK-22374
-
- Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2
-
-2016-08-18 07:55 +0000 [0b675a208b] Alexander Traud <pabstraud@compuserve.com>
-
- * sip_to_pjsip: Add cert_file and ca_list_path.
-
- When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were
- not migrated to pjsip.conf.
-
- ASTERISK-22374
-
- Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825
-
-2016-08-17 14:13 +0000 [1cd12d73a6] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Fix unbound srv failover tests.
-
- Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover
- functionality if a TCP connection gets disconnected. Under these
- conditions, session_inv_on_state_changed() gets a
- PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new
- transport. Unfortunately, session_inv_on_tsx_state_changed() also gets
- the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates
- the session.
-
- * Made session_inv_on_tsx_state_changed() complete terminating the session
- on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still
- PJSIP_INV_STATE_DISCONNECTED.
-
- ASTERISK-26305 #close
- Reported by: Richard Mudgett
-
- Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d
-
-2016-08-16 15:36 +0000 [329507fe20] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip: Add contact_user to endpoint
-
- contact_user, when specified on an endpoint, will override the user
- portion of the Contact header on outgoing requests.
-
- Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
-
-2016-08-17 08:10 +0000 [6f448f32fe] Torrey Searle <torrey@voxbone.com>
-
- * res_ari: Add http prefix to generated docs
-
- updated the uri handler to include the url prefix of the http server
- this enables res_ari to add it to the uris when generating docs
-
- Change-Id: I279335a2625261a8492206c37219698f42591c2e
-
-2016-08-17 06:12 +0000 [56e0aed177] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Detect ca_list_path capabilities in external PJProject.
-
- Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the
- bundled PJProject but not in an external PJProject. Therefore, ca_list_path
- could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2
- is detected again to enable ca_list_path again.
-
- ASTERISK-26303 #close
-
- Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0
-
-2016-08-16 12:24 +0000 [2edcfcf1eb] gtjoseph <gjoseph@digium.com>
-
- * ari: Add documentation that path parameters are case-sensitive
-
- Added to api.wiki.mustache so that the generated object pages
- have the notation in the table header as well as under each method
- that has path parameters.
-
- ASTERISK-25492 #close
-
- Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf
-
-2016-08-15 15:29 +0000 [f4e28b3a09] Corey Farrell <git@cfware.com>
-
- * Refactor usage pattern of xmldoc info tag.
-
- This updates func_channel.c and main/message.c to use a generic xpointer
- include instead of including info from each channel driver. Now the
- name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
- documentation for func_channel. Setting the name attribute of info to
- MessageToInfo or MessageFromInfo causes it to be included in the
- MessageSend application and AMI action.
-
- Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea
-
-2016-08-04 20:00 +0000 [a8d9a53bae] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_config.c: Cleanup ao2 container usage idioms.
-
- Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a
-
-2016-08-04 15:57 +0000 [74a91b9ee5] Richard Mudgett <rmudgett@digium.com>
-
- * sorcery.c: Minor optimizations.
-
- * Remove some unused parameters from internal functions:
- sorcery_wizard_create()
- sorcery_wizard_update()
- sorcery_wizard_delete()
-
- * Created the struct sorcery_observer_invocation ao2 object without a lock
- since it is not needed in sorcery_observer_invocation_alloc().
-
- * Cleanup generic ao2 container sorcery object id hash, sort, and cmp
- functions.
-
- Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e
-
-2016-08-01 11:04 +0000 [29beb2890c] Richard Mudgett <rmudgett@digium.com>
-
- * sorcery.c: Tweak some container declaration formatting.
-
- * Tweak sorcery_object_type_alloc() formatting.
- * Tweak ast_sorcery_init() formatting.
-
- Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944
-
-2016-08-11 23:30 +0000 [9b822293bd] Corey Farrell <git@cfware.com>
-
- * pbx.c: Additional fixes to ast_context_remove_extension_callerid2.
-
- Do not check registrar of the first extension head. We should only check
- the registrar when we match the priority.
-
- Additionally fix a couple calls to strcmp which used the input callerid
- instead of the clean version ex.cidmatch.
-
- ASTERISK-26233
-
- Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1
-
-2016-08-10 14:41 +0000 [403c794684] Alexei Gradinari <alex2grad@gmail.com>
-
- * core: Entity ID is not set or invalid
-
- The Exchanging Device and Mailbox States could not working
- if the Entity ID (EID) is not set manually and can't be obtained
- from ethernet interface.
-
- This patch replaces debug message to warning
- and addes missing description about option 'entityid' to
- asterisk.conf.sample.
-
- With this patch the asterisk also:
- (1) decline loading the modules which won't work without EID:
- res_corosync and res_pjsip_publish_asterisk.
- (2) warn if EID is empty on loading next modules:
- pbx_dundi, res_xmpp
-
- Starting with v197 systemd/udev will automatically assign "predictable"
- names for all local Ethernet interfaces.
- This patch also addes some new ethernet prefixes "eno" and "ens".
-
- ASTERISK-26164 #close
-
- Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
-
-2016-06-15 17:10 +0000 [93332cb1d0] Evgeniy Tsybra <cjack@yandex.ru>
-
- * chan_sip: Fix lastrtprx always updated
-
- Packets are read regulary, when there is no data in buffer fr->frametype
- is AST_FRAME_NULL. There was no check of frametype and lastrtprx always
- updated and, therefore, rtptimeout did not work at all.
-
- ASTERISK-25270 #close
-
- Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d
-
-2016-08-15 07:17 +0000 [2735ec899a] Joshua Colp <jcolp@digium.com>
-
- * manager: Clarify that dialplan manipulation actions are under system class.
-
- ASTERISK-26246 #close
-
- Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657
-
-2016-08-13 22:02 +0000 [f59bd47ed3] Matt Jordan <mjordan@digium.com>
-
- * app_dial: Improve documentation
-
- * Add some helpful <literal> and other embedded paragraph tags
-
- * Document some of the lesser known channel variables set by Dial
-
- * Add examples for some common Dial uses, along with some more
- challenging but useful options
-
- Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1
-
-2016-08-13 20:16 +0000 [4facaac408] Matt Jordan <mjordan@digium.com>
-
- * manager: Add <see-also> tags to relate interrelated events/actions together
-
- Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4
-
-2016-08-13 20:15 +0000 [232d4fe24f] Matt Jordan <mjordan@digium.com>
-
- * manager: Add <see-also> tags to relate Bridge related events,actions, and apps
-
- Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85
-
-2016-08-13 20:14 +0000 [63c0b2f7c9] Matt Jordan <mjordan@digium.com>
-
- * manager: Add <see-also> tags to relate AoC events and actions
-
- Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e
-
-2016-08-13 20:13 +0000 [0422667d6c] Matt Jordan <mjordan@digium.com>
-
- * manager: Add <see-also> tags to relate UserEvent actions/apps/events
-
- Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4
-
-2016-08-12 15:53 +0000 [f9e734974b] Matt Jordan <mjordan@digium.com>
-
- * res_agi: Improve documentation
-
- * Groups of AGI commands that have similar functionality now reference
- each other, and all reference the AGI application for ease of wiki
- reference.
-
- * The documentation for the AGI application has been improved, in
- particular noting the various AGI types and how they are invoked.
-
- * A warning message has been added to DeadAGI, noting that it is
- deprecated.
-
- Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d
-
-2016-08-12 13:53 +0000 [781bb410d0] Matt Jordan <mjordan@digium.com>
-
- * manager: Add <see-also> links between related events
-
- This patch adds some see-also references between related AMI events. It
- focuses primarily on those events that are guaranteed to come in pairs,
- such as DTMFBegin/DTMFEnd, as well as those that occur during the life
- cycle of an Asterisk channel, such as Newchannel/Hangup.
-
- Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3
-
-2016-08-12 11:15 +0000 [cfd6852d39] Matt Jordan <mjordan@digium.com>
-
- * func_channel: Reorganize documentation
-
- * Following the example of the PJSIP channel driver, the channel
- technology specific documentation has been moved to the respective
- channel drivers that provide that functionality. This has the benefit
- of locating the documentation of items with those modules that provide
- it.
-
- * Examples of using the CHANNEL function for both standard items as well
- as for PJSIP have been added.
-
- * The 'max_forwards' standard item has been documented.
-
- Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b
-
-2016-08-11 22:11 +0000 [cb043249b6] Corey Farrell <git@cfware.com>
-
- * Run mandatory cleanup when startup fails.
-
- Errors during startup result in an exit. These error branches should be
- calling ast_run_atexit(0) to ensure mandatory cleanup is run.
-
- ASTERISK-26267 #close
-
- Change-Id: If226f2326ae2df7add20040696132214cf2bb680
-
-2016-08-11 11:24 +0000 [4d5e96ab53] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip_caller_id: Copy header name to short header name
-
- When compact_headers was set, we were sending a zero-length header name
- for PAI and RPID because we always forced the short header name length
- to 0. We did this because we cloned the header from "From" and wanted
- to clear "f" from the sname. By cloning however, we bypass pjproject's
- automatic logic that sets sname to name if there's no compact form of
- the header, which there isn't for PAI and RPID. So now we force sname
- to be the same as name right after we set name.
-
- res_pjsip_diversion needed the same treatment for the Diversion header.
-
- ASTERISK-26241 #close
-
- Change-Id: I633ec139630cd83809aae00336cee4a10077e467
-
-2016-08-11 12:18 +0000 [143df33110] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip: Fail global load if debug or default_from_user are empty
-
- If debug was specified in the global configuration but left blank,
- the logger would treat it as a wildcard and log all hosts. If
- default_from_user was empty, a crash would result.
-
- The global apply handler now checks for empty strings.
-
- ASTERISK-26239 #close
- ASTERISK-26238 #close
-
- Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336
-
-2016-08-01 15:07 +0000 [1fc5c90014] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip res_pjsip_mwi: Misc fixes and cleanups.
-
- * Eliminated RAII_VAR() usage in
- ast_sip_persistent_endpoint_update_state().
-
- * Added a missing allocation failure check to
- persistent_endpoint_find_or_create().
-
- * Made persistent_endpoint_find_or_create() create the new object without
- a lock as it isn't needed.
-
- * Cleaned up some ao2 container allocation idioms.
-
- * Reordered res_pjsip_mwi.c load_module() and unload_module()
-
- Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8
-
-2016-08-04 18:03 +0000 [73052e5732] Richard Mudgett <rmudgett@digium.com>
-
- * location.c: Misc fixes and cleanups.
-
- * Eliminated most RAII_VAR() usage.
-
- * Added several missing allocation failure checks.
-
- * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without
- a lock as it is not needed.
-
- Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc
-
-2016-08-02 13:53 +0000 [9d4bd3d763] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Tweak high water checks.
-
- * The high water check in ast_taskprocessor_alert_set_levels() would
- trigger immediately if the new high water level is zero and the queue was
- empty.
-
- * The high water check in taskprocessor_push() was off by one.
-
- Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d
-
-2016-08-03 16:24 +0000 [e1248c3075] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Make aor named lock a mutex.
-
- The named aor lock was always being locked for writes so a rwlock adds no
- benefit and may be slower because rwlocks are biased toward read locking.
-
- Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
-
-2016-07-29 17:41 +0000 [6e40334d89] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_distributor.c: Add missing allocation failure check.
-
- Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28
-
-2016-08-11 11:13 +0000 [a3c5488ff4] Matt Jordan <mjordan@digium.com>
-
- * app_queue: Prevent crash when a call is forwarded to an invalid location
-
- When a call forward attempt is made from a Queue member, the current
- code will hang up the forwarding channel in an off-nominal condition
- prior to raising the Stasis events informing the rest of Asterisk that
- the call was forwarded. This will result in a slew of dreaded FRACKs,
- most likely leading to a crash.
-
- This patch modifies the code such that we don't hang up the forwarding
- channel even in an off-nominal condition until we've safely raised the
- Stasis messages.
-
- ASTERISK-25797 #close
-
- Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38
-
-2016-08-11 10:50 +0000 [5913929d31] Kevin Harwell <kharwell@digium.com>
-
- * alembic: add auth_username to endpoint's identify_by enum
-
- A new identify_by option was added recently, auth_username. However, this
- setting was not added as an allowable choice in the database enumeration
- value.
-
- This patch updates the current enumeration, adding in the new setting.
-
- ASTERISK-26268 #close
-
- Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8
-
-2016-08-06 10:57 +0000 [1589452fdc] Alexei Gradinari <alex2grad@gmail.com>
-
- * pjsip: Fix deadlock with suspend taskprocessor on masquerade
-
- If both channels which should be masqueraded
- are in the same serializer:
- 1st channel will be locked waiting condition 'complete'
- 2nd channel will be locked waiting condition 'suspended'
-
- On heavy load system a chance that both channels will be in
- the same serializer 'pjsip/distibutor' is very high.
-
- To reproduce compile res_pjsip/pjsip_distributor.c with
- DISTRIBUTOR_POOL_SIZE=1
-
- Steps to reproduce:
- 1. Party A calls Party B (bridged call 'AB')
- 2. Party B places Party A on hold
- 3. Party B calls Voicemail app (non-bridged call 'BV')
- 4. Party B attended transfers Party A to voicemail using REFER.
- 5. When asterisk masquerades calls 'AB' and 'BV',
- a deadlock is happened.
-
- This patch adds a suspension indicator to the taskprocessor.
- When a session suspends/unsuspends the serializer
- it sets the indicator to the appropriate state.
- The session checks the suspension indicator before
- suspend the serializer.
-
- ASTERISK-26145 #close
-
- Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
-
-2016-08-09 12:07 +0000 [f6ec94cca6] Kevin Harwell <kharwell@digium.com>
-
- * alembic/sqlalchemy: auto increment only allowed on a single column
-
- The extensions table defined two columns (id and priority) as primary key
- autoincrement columns. However only one is allowed when defining the primary
- key.
-
- This patch removes the autoincrement attribute from the priority column since
- it does not need to be as such and really should not have been on there in the
- first place.
-
- This patch also removes 'context', 'exten', and 'priority' from the primary key
- index and creates a new combined unique contraint index on them.
-
- ASTERISK-26183 #close
-
- Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b
-
-2016-08-07 09:58 +0000 [5f815f9dba] Matt Jordan <mjordan@digium.com>
-
- * channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH
-
- This patch adds a new PJSIP specific dialplan function,
- PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
- session will be refreshed via either an UPDATE or re-INVITE request.
- When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
- the formats in use on a PJSIP channel can be re-negotiated and changed
- dynamically after call setup.
-
- ASTERISK-26277 #close
-
- Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
-
-2016-08-09 16:19 +0000 [a119bab6a6] Mark Michelson <mmichelson@digium.com>
-
- * res_rtp_asterisk: Cache local RTCP address.
-
- When an RTCP packet is sent or received, res_rtp_asterisk generates a
- Stasis event that contains the RTCP report as well as the local and
- remote addresses that the report pertains to.
-
- The addresses are determined using ast_find_ourip(). For the local
- address, this will typically result in a lookup of the hostname of the
- server, and then a DNS lookup of that hostname. If you do not have the
- host in /etc/hosts, then this results in a full DNS lookup, which can
- potentially block for some time.
-
- This is especially problematic when performing RTCP reads, since those
- are done on the same thread responsible for reading and writing media.
-
- This patch addresses the issue by performing a lookup of the local
- address when RTCP is allocated. We then use this cached local address
- for the Stasis events when necessary.
-
- ASTERISK-26280 #close
- Reported by Mark Michelson
-
- Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556
-
-2016-08-08 12:53 +0000 [a06a1af0eb] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack
-
- The PJSIP taskprocessors could be overflowed on startup
- if there are many (thousands) realtime endpoints
- configured with unsolicited mwi.
- The PJSIP stack could be totally unresponsive for a few minutes
- after boot completed.
-
- This patch creates a separate PJSIP serializers pool for mwi
- and makes unsolicited mwi use serializers from this pool.
- This patch also adds 2 new global options to tune taskprocessor
- alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
-
- This patch also adds new global option 'mwi_disable_initial_unsolicited'
- to disable sending unsolicited mwi to all endpoints on startup.
- If disabled then unsolicited mwi will start processing
- on next endpoint's contact update.
-
- ASTERISK-26230 #close
-
- Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
-
-2016-08-04 10:16 +0000 [485fd27f7c] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_outbound_publish: Use a serializer shutdown group for unload.
-
- This change replaces the custom unload process for the outbound
- publish module with the common serializer shutdown group.
-
- ASTERISK-25217 #close
-
- Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6
-
-2016-08-03 15:39 +0000 [805f105f88] Corey Farrell <git@cfware.com>
-
- * Add missing checks during startup.
-
- This ensures startup is canceled due to allocation failures from the
- following initializations.
- * channel.c: ast_channels_init
- * config_options.c: aco_init
-
- ASTERISK-26265 #close
-
- Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
-
-2016-07-22 16:37 +0000 [ea71bd6e3e] Alexei Gradinari <alex2grad@gmail.com>
-
- * app_voicemail: Add taskprocessor alert level options.
-
- On heavy loaded system with IMAP or DB storage,
- 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
- It could happen when the IMAP or DB server dies or is unreachable.
- It could happen on startup when there are many (thousands)
- realtime endpoints configured with unsolicited mwi.
- If the taskprocessor queue reaches the high water level
- then the alert is triggered and pjsip stops processing new requests
- until the queue reaches the low water level to clear the alert.
-
- This patch adds 2 new 'general' configuration options
- to tune taskprocessor alert levels:
- 'tps_queue_high' - Taskprocessor high water alert trigger level.
- 'tps_queue_low' - Taskprocessor low water clear alert level
-
- ASTERISK-26229 #close
-
- Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8
-
-2016-08-03 09:47 +0000 [9dc8cfabd5] Joshua Colp <jcolp@digium.com>
-
- * astconfigparser: Really handle case where line is simply a comment.
-
- The regular expression would match causing the code that handled
- the line if it was merely a comment to never get executed.
-
- Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819
-
-2016-07-23 08:51 +0000 [ad3e65433c] gtjoseph <gjoseph@digium.com>
-
- * asterisk.c: Add auto generation and persistence of UUID
-
- Upcoming features will require the generation and persistence
- of a UUID.
-
- Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d
-
-2016-08-02 12:55 +0000 [efc4034d72] Kevin Harwell <kharwell@digium.com>
-
- * rest-api: Code out of sync with the model
-
- Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006
-
-2016-07-29 13:13 +0000 [f6821fbaec] Mark Michelson <mmichelson@digium.com>
-
- * Remove SILK payload mappings from Asterisk core.
-
- SILK is a bit of a hog when it comes to using up our limited number of
- dynamic payload types in the RTP engine. By freeing up four slots, it
- allows for other codecs to potentially take the place.
-
- Now, codec_silk.so will dynamically use the payload slots in the RTP
- engine when it loads.
-
- A better fix would be make RTP dynamic payload types actually
- dynamic. However, at this stage of Asterisk 14 development, this is a
- risky move that would be imprudent.
-
- Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612
- (cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817)
-
-2016-08-01 11:08 +0000 [102d28c11a] Joshua Colp <jcolp@digium.com>
-
- * sorcery: Use more compatible regex for local expressions.
-
- This changes the use of an empty regex for both res_sorcery_config
- and res_sorcery_memory to "." instead. This is a more compatible
- regular expression which also works on FreeBSD.
-
- ASTERISK-26206 #close
-
- Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388
-
-2016-08-02 03:08 +0000 [b78d10a2df] Alexander Traud <pabstraud@compuserve.com>
-
- * res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports.
-
- ASTERISK-26256 #close
-
- Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058
-
-2016-08-01 16:13 +0000 [1f95c011c7] gtjoseph <gjoseph@digium.com>
-
- * menuselect: Add an opaque "member_data" string to the acceptable xml
-
- Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe
-
-2016-07-27 09:56 +0000 [df42f64d62] David M. Lee <dlee@respoke.io>
-
- * Replace strdupa with more portable ast_strdupa
-
- The strdupa function is a GNU extension, and not widely portable. We
- have an ast_strdupa function used within Asterisk which is preferred.
- I pulled the definition up from menuselect.c into the menuselect.h
- header file so it can be shared across menuselect.
-
- Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e
-
-2016-07-24 18:27 +0000 [56a07fbab9] gtjoseph <gjoseph@digium.com>
-
- * menuselect: Various menuselect enhancements
-
- * Add 'external' as a support level.
- * Add ability for module directories to add entries to the menu
- by adding members to the <module_prefix>/<module_prefix>.xml file.
- * Expand the description field to 3 lines in the ncurses implementation.
- * Allow the description field to wrap in the newt implementation.
- * Add description field to the gtk implementation.
-
- Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808
- (cherry picked from commit 90f445729d5d86050d9d379485ff0a99f4a006c1)
-
-2016-07-29 04:48 +0000 [7f9369c1b6] Joshua Colp <jcolp@digium.com>
-
- * astconfigparser: Handle case where line is simply a comment.
-
- Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5
-
-2016-07-28 14:10 +0000 [57e9c66819] Corey Farrell <git@cfware.com>
-
- * pbx.c: Fix handling of '-' in extension name and callerid
-
- This adds a two strings to ast_exten. name to go with exten and
- cidmatch_display to go with cidmatch. The new fields contain input used
- to add the extension in the first place. The existing fields now
- contain stripped input that excludes insignificant spaces and dashes.
- These stripped fields should always be used for comparisons. The
- unstripped fields should normally be used for display, but displaying
- stripped values will not cause runtime errors.
-
- Note the actual string is only stored twice if it contains dashes. If
- no dashes are found then both 'char *' fields point to the same memory.
- So this change has a minimum effect on memory usage.
-
- The existing functions ast_get_extension_name and
- ast_get_extension_cidmatch return unstripped values as they did before
- this change. Other similar bugs likely still exist where unstripped
- extensions are saved outside pbx.c then passed back in.
-
- ASTERISK-26233 #close
-
- Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f
-
-2016-07-27 17:17 +0000 [873fc0fda5] Richard Mudgett <rmudgett@digium.com>
-
- * pbx.c: Allow dangerous functions when adding a hint to dialplan.
-
- We can allow dangerous functions when adding a hint since altering
- dialplan is itself a privileged activity. Otherwise, we could never
- execute dangerous functions.
-
- ASTERISK-25996 #close
- Reported by: Andrew Nagy
-
- Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba
-
-2016-07-21 10:36 +0000 [f00525a6f6] Alexei Gradinari <alex2grad@gmail.com>
-
- * pjproject: fixed a few bugs
-
- This patch fixes the issue in pjsip_tx_data_dec_ref()
- when tx_data_destroy can be called more than once,
- and checks if invalid value (e.g. NULL) is passed to.
-
- This patch updates array limit checks and docs
- in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability().
-
- Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40
-
-2016-07-17 18:28 +0000 [972cee2e4c] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Update for pjproject 2.5.5
-
- Add more --disable-* switches to Makefile.rules including
- --disable-opus which was causing bundled pjproject to fail with
- "undefined reference" errors in libasteriskpj.
-
- Changed PJ_ENABLE_EXTRA_CHECK to 1.
-
- Removed 2 obsolete patches and added a new one.
- The new one was merged by Teluu on 6/27/2016.
-
- ASTERISK-26148 #close
-
- Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063
-
-2016-07-27 10:33 +0000 [8902a51d59] David M. Lee <dlee@respoke.io>
-
- * Portably sscanf tv_usec
-
- In a timeval, tv_usec is defined as a suseconds_t, which could be
- different underlying types on different platforms. Instead of trying to
- scanf directly into the timeval, scanf into a long int, then copy that
- into the timeval.
-
- Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95
-
-2016-07-27 12:36 +0000 [852e763571] Kevin Harwell <kharwell@digium.com>
-
- * rtp_engine: Failed assertion and wrong name given for codec
-
- Fixed an assert check that would trigger when the passed in value was negative.
- The negative value was being cast to an unsigned value. This resulted in the
- check failing.
-
- Also fixed another problem when loading formats in the engine. When setting the
- mime type the format's name was being passed in instead of the codec's name.
-
- Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c
-
-2016-07-21 22:44 +0000 [e8c34680ca] Richard Mudgett <rmudgett@digium.com>
-
- * dsp.c: Add fax and DTMF detection unit tests.
-
- * Add fax amplitude and frequency sweep tests.
- * Add DTMF amplitude and twist unit tests.
-
- Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7
-
-2016-07-21 11:56 +0000 [c1f240b818] Richard Mudgett <rmudgett@digium.com>
-
- * dsp.c: Added descriptive comments to Goertzel calculations.
-
- * Added doxygen to describe some struct members and what is going on in
- the code.
-
- Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d
-
-2016-07-13 13:48 +0000 [003a52fd62] Richard Mudgett <rmudgett@digium.com>
-
- * dsp.c: Fix incorrect format reference typo.
-
- Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896
-
-2016-07-25 21:18 +0000 [4c0a0cbe02] Richard Mudgett <rmudgett@digium.com>
-
- * dsp.c: Correct DTMF twist dsp.conf documentation.
-
- Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae
-
-2016-07-22 04:43 +0000 [87433c2566] Joshua Colp <jcolp@digium.com>
-
- * astconfigparser.py: Update with realtime fixes.
-
- When configuring SIP URIs in the pjsip.conf file it is
- necessary to escape the semicolon so the parser does not
- treat it as a comment. This change allows this to work in
- the astconfigparser implementation.
-
- A secondary bug where some data was lost if a configuration
- option included a "=" in its value was also fixed.
-
- A bug where sections would be considered equal despite
- being different has also been fixed.
-
- Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8
-
-2016-07-28 14:32 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.11.0-rc1 Released.
-
-2016-07-28 09:29 +0000 [3bfaf6b172] Mark Michelson <mmichelson@digium.com>
-
- * Release summaries: Add summaries for 13.11.0-rc1
-
-2016-07-28 09:27 +0000 [ca145e1807] Mark Michelson <mmichelson@digium.com>
-
- * .version: Update for 13.11.0-rc1
-
-2016-07-28 09:27 +0000 [918ebf79ff] Mark Michelson <mmichelson@digium.com>
-
- * .lastclean: Update for 13.11.0-rc1
-
-2016-07-28 09:27 +0000 [d7afc1cf9d] Mark Michelson <mmichelson@digium.com>
-
- * realtime: Add database scripts for 13.11.0-rc1
-
-2016-07-21 22:28 +0000 [159e437e5a] Richard Mudgett <rmudgett@digium.com>
-
- * dsp.c: Fix erroneous fax tone detection.
-
- The Goertzel calculations get less accurate the lower the signal level
- being worked with becomes because there is less resolution remaining.
- If it is too low we can erroneously detect a tone where none really
- exists. The searched for fax frequencies not only need to be so much
- stronger than the background noise they must also be a minimum strength.
-
- * Add needed minimum threshold test to tone_detect().
-
- * Set TONE_THRESHOLD to allow low volume frequency spread detection.
-
- ASTERISK-26237 #close
- Reported by: Richard Mudgett
-
- Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc
-
-2016-07-22 14:44 +0000 [eda95236d1] Mark Michelson <mmichelson@digium.com>
-
- * Fix sqlalchemy error regarding identifier length.
-
- sqlalchemy was complaining:
-
- sqlalchemy.exc.IdentifierError: Identifier
- 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30
- characters
-
- This fixes the problem by changing the index name to be
- "ps_contacts_qualifyfreq_exp" instead.
-
- ASTERISK-26227 #close
- Reported by Mark Michelson
-
- Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9
-
-2016-07-22 05:46 +0000 [66c9dfb272] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Enable Session-Timers for SIP over TCP (and TLS).
-
- Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
- scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
- Session-Timers for SIP over TCP (and for SIP over TLS).
-
- However with longer international calls via TCP, the SIP channel might break,
- because all hops on the Internet route must stay online (have not a single power
- outage, for example). Therefore with Session-Timers enabled (which are enabled
- at default), you might see dropped calls. Consequently even with this change,
- you might be better-off going for session-timers=refuse in your sip.conf.
-
- ASTERISK-19968 #close
-
- Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
-
-2016-07-15 16:16 +0000 [33716106e0] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Whitespace and comment cleanup.
-
- Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
-
-2016-07-21 09:05 +0000 [52ab0bf258] gtjoseph <gjoseph@digium.com>
-
- * chan_sip: Prevent deadlock when issuing "sip show channels"
-
- sip_show_channels locks the dialogs container first then locks each
- sip_pvt so it can spit out the details. The rest of sip dialog
- processing locks the sip_pvt first then locks the dialogs container
- if it needs to. Both lock in the order they need but deadlocks can
- result. To fix, sip_show_channels and sip_show_channelstats have
- been converted to use an iterator rather than ao2_callback. This way
- the container is locked only while getting the next entry and is
- unlocked when the callback is called.
-
- ASTERISK-23013 #close
-
- Change-Id: Id9980419909e811f89484950ed46ef117b9eb990
-
-2016-07-19 15:22 +0000 [5997ec7c9e] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice.
-
- This patch removed call of pjsip_tx_data_dec_ref in send_notify
- if send_request failed.
- The pjsip_dlg_send_request deletes the message on error by itself.
-
- It seems this patch fixes next issues:
- ASTERISK-26199
- ASTERISK-26166
- ASTERISK-26174
-
- Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a
-
-2016-07-18 22:46 +0000 [7fdf7c3d4c] Corey Farrell <git@cfware.com>
-
- * Add conditional support for noreturn functions.
-
- This adds support for tagging functions with the noreturn attribute.
- If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE
- and DO_CRASH are enabled then failed assertions never return. This can
- resolve a large number of false positives with static analyzers.
-
- ASTERISK-26220 #close
-
- Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753
-
-2016-07-19 13:18 +0000 [dcb8aa8c1c] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi.c: Fix deadlock potential in fax redirection.
-
- The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to
- deadlock if an incoming fax happens during the Playback or similar
- application.
-
- * Fixed the potential deadlock by not calling ast_async_goto() with the
- channel lock held.
-
- ASTERISK-26216 #close
- Reported by: Richard Mudgett
-
- Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa
-
-2016-07-13 18:49 +0000 [fa91cf3eec] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix deadlock potential in fax redirection.
-
- The sip_read() has the potential to deadlock if an incoming fax happens
- during the Playback or similar application.
-
- * Fixed the potential deadlock by not calling ast_async_goto() with the
- channel lock held.
-
- * Made always eat the fax detection frame whether there is a fax extension
- or not.
-
- ASTERISK-26216
- Reported by: Richard Mudgett
-
- Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e
-
-2016-07-13 18:48 +0000 [2e1bdc3775] Richard Mudgett <rmudgett@digium.com>
-
- * chan_pjsip.c: Fix deadlock potential in fax redirection.
-
- The chan_pjsip_cng_tone_detected() has the potential to deadlock if an
- incoming fax happens during the Playback or similar application.
-
- * Fixed the potential deadlock by not calling ast_async_goto() with the
- channel lock held.
-
- * Made always eat the fax detection frame whether there is a fax extension
- or not.
-
- ASTERISK-26216
- Reported by: Richard Mudgett
-
- Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5
-
-2016-07-12 17:33 +0000 [628e8c91d5] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook.
-
- The fax_detect_framehook() has the potential to deadlock if an incoming
- fax happens during the Playback or similar application.
-
- * Fixed the potential deadlock by not calling ast_async_goto() with the
- channel lock held.
-
- * Made always eat the fax detection frame whether there is a fax extension
- or not.
-
- * Made only detach the framehook if we detected a fax and not on other
- possible frames.
-
- ASTERISK-26216
- Reported by: Richard Mudgett
-
- Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d
-
-2016-07-12 17:24 +0000 [676aeede36] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax: Fix FAXOPT(faxdetect) timeout option.
-
- The fax detection timeout option did not work because basically the wrong
- variable was checked in fax_detect_framehook(). As a result, the timer
- would timeout immediately and disable fax detection.
-
- * Fixed ignoring negative timeout values. We'd complain and then go right
- on using the negative value.
-
- * Fixed destroy_faxdetect() in the off-nominal case of an incomplete
- object creation.
-
- * Added more range checking to FAXOPT(gateway) timeout parameter.
-
- ASTERISK-26214 #close
- Reported by: Richard Mudgett
-
- Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976
-
-2016-07-18 16:16 +0000 [652130feb2] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi: Add faxdetect_timeout option.
-
- The new option allows the channel driver's faxdetect option to timeout on
- a call after the specified number of seconds into a call. The new feature
- is disabled if the timeout is set to zero. The option is disabled by
- default.
-
- * Don't clear dsp_features after passing them to the dsp code in
- my_pri_ss7_open_media(). We should still remember them especially for the
- new faxdetect_timeout option.
-
- ASTERISK-26214
- Reported by: Richard Mudgett
-
- Change-Id: Ieffd3fe788788d56282844774365546dce8ac810
-
-2016-07-15 20:44 +0000 [851b1c3a17] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Add fax_detect_timeout endpoint option.
-
- The new endpoint option allows the PJSIP channel driver's fax_detect
- endpoint option to timeout on a call after the specified number of
- seconds into a call. The new feature is disabled if the timeout is set
- to zero. The option is disabled by default.
-
- ASTERISK-26214
- Reported by: Richard Mudgett
-
- Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
-
-2016-07-19 04:48 +0000 [021d4892cd] Alexander Traud <pabstraud@compuserve.com>
-
- * Makefile: Retain XML Declaration and DTD in docs.
-
- Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo,
- the XML Declaration and DTD were overwritten by this.
-
- ASTERISK-26212 #close
-
- Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd
-
-2016-07-18 18:39 +0000 [c8e41d14a1] Corey Farrell <git@cfware.com>
-
- * Unit tests: Use AST_TEST_DEFINE in conditional code only.
-
- If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
- code. This places all existing unit tests into a conditional block if
- they weren't already.
-
- ASTERISK-26211 #close
-
- Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
-
-2016-07-18 05:13 +0000 [e404f51b42] Alexander Traud <pabstraud@compuserve.com>
-
- * res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.
-
- With this change, the initial RTP sequence number is randomly chosen not between
- 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
- counter (ROC) synchronization is not lost for sRTP, when the very first RTP
- packets get lost; see http://srtp.sourceforge.net/faq.html#Q6
-
- ASTERISK-26207 #close
-
- Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
-
-2016-07-18 04:14 +0000 [5f24874ebb] Alexander Traud <pabstraud@compuserve.com>
-
- * Makefile: Suppress echoing of target 'config' again.
-
- ASTERISK-26038 #close
-
- Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f
-
-2016-07-14 03:25 +0000 [76d4983c15] Corey Farrell <git@cfware.com>
-
- * features.c: Remove unneeded adsi.h include.
-
- adsi.h is no longer used by features.c since parking was moved to a
- module.
-
- Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59
-
-2016-07-14 18:06 +0000 [cb58f853e1] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_mwi: remove unneeded check on endpoint's contacts.
-
- The function create_mwi_subscriptions_for_endpoint checks
- if there is active contacts by retrieving aors and contacts.
-
- This function is used to create all unsolicited mwi subscriptions
- on startup and is used when contact added.
-
- In both cases it's not necessary to check if there are contacts.
- The contacts are needed when asterisk sends mwi.
-
- ASTERISK-26200 #close
-
- Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa
-
-2016-06-30 15:58 +0000 [28501051b4] Mark Michelson <mmichelson@digium.com>
-
- * Update support for SILK format.
-
- This commit adds scaffolding in order to support the SILK audio format
- on calls. Roughly, this is what is added:
-
- * Cached silk formats. One for each possible sample rate.
- * ast_codec structures for each possible sample rate.
- * RTP payload mappings for "SILK".
-
- In addition, this change overhauls the res_format_attr_silk file in the
- following ways:
-
- * The "samplerate" attribute is scrapped. That's native to the format.
- * There are far more checks to ensure that attributes have been
- allocated before attempting to reference them.
- * We do not SDP fmtp lines for attributes set to 0.
-
- These changes make way to be able to install a codec_silk module and
- have it actually work. It also should allow for passthrough silk calls
- in Asterisk.
-
- Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
-
-2016-07-14 07:45 +0000 [43b5f8d57b] Richard Miller (license 5685)
-
- * app_queue: Only remove queue member from pending when state changes.
-
- It is possible for a not in use state change to occur multiple
- times causing a queue member to be removed from the pending call
- container prematurely.
-
- The first not in use state change will remove the queue member
- from the container. At this moment the member may be called and
- placed in the pending container. After this another not in use
- state change can be received which will remove it from the
- container. Despite being called at this point the code will
- incorrectly see that there are no pending calls to it.
-
- This change only removes it from the pending container if the
- state has actually changed.
-
- ASTERISK-26133 #close
- patches:
- app_queue.diff submitted by Richard Miller (license 5685)
-
- Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0
-
-2016-07-14 02:40 +0000 [a17b071e36] Corey Farrell <git@cfware.com>
-
- * pbx: Fix leak of timezone for time based includes.
-
- Create include_free to run ast_destroy_timing and ast_free, use that in
- all places that freed an ast_include structure. This fixes a couple of
- paths that previously did not run ast_destroy_timing.
-
- ASTERISK-26196 #close
-
- Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838
-
-2016-07-13 17:45 +0000 [8cef8f35e7] Kevin Harwell <kharwell@digium.com>
-
- * translate: explicit format destination not properly set
-
- If the destination format's name differed from the codec name then the
- translator's explict_dst field would be improperly set. In some circumstances
- it would end up setting it to a newly created format that has the same name
- as the codec when it actually needed to be the given destination codec.
-
- This could cause the translation path to use the wrong format. For instance,
- if an endpoint had specified 'myulaw' as a format the translator could end up
- using a 'ulaw' format (with whatever/default settings) instead. If the format
- attribute settings differed between the two then there may unexpected results
- during processing.
-
- This patch removes the name check when building the translation path. This
- should make it always set the translator's explicit_dst to the given destination
- format as long as the sample rate and types match.
-
- Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5
-
-2016-07-08 11:46 +0000 [afbd10b0c5] Richard Mudgett <rmudgett@digium.com>
-
- * stasis_endpoint.c: Fix contactstatus_to_json().
-
- The roundtrip_usec json member is optional. If it isn't present then
- don't put it into the converted json structure where ast_json_pack()
- will choke on it.
-
- Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0
-
-2016-07-13 13:45 +0000 [2be13d62fd] Corey Farrell <git@cfware.com>
-
- * chan_sip: Fix reference leak in mwi_event_cb
-
- Cleanup the peer reference when stasis_subscription_final_message is
- true. Also free peer_name even if peer exists, after reload a new
- peer_name will be allocated.
-
- ASTERISK-26193 #close
-
- Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69
-
-2016-06-22 07:13 +0000 [332beb27d8] Eugene Voityuk <eugene@thirdlane.com>,Alexander Traud <pabstraud@compuserve.com>
-
- * res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS.
-
- Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
- support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
- for DTLS. The source code from main/tcptls.c should have been re-used to ease
- security audits. Therefore, this change rolls back the change from July 2015 and
- re-uses the code from July 2014. This has the additional benefits to work under
- CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.
-
- ASTERISK-25659 #close
- Reported by: StefanEng86, urbaniak, pay123
- Tested by: sarumjanuch, traud
- patches:
- res_rtp_asterisk.patch submitted by sarumjanuch
- dtls_centos_step_1.patch submitted by traud
- dtls_centos_step_2.patch submitted by traud
-
- Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
-
-2016-07-13 11:30 +0000 [672a64bda3] Corey Farrell <git@cfware.com>
-
- * threadpool: Fix leak in ast_threadpool_serializer_group error path.
-
- ast_threadpool_serializer_group leaks a reference to ser when listener
- is allocated but tps is not. Although listener takes the reference to
- ser cleanup functions are not run without tps.
-
- ASTERISK-26191 #close
-
- Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585
-
-2016-07-11 10:22 +0000 [fea201f7e6] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_options.c: Fix container operation.
-
- aor_observer_deleted() needs to operate on all contacts found for the
- deleted AOR instead of only the first one found. This is really only a
- problem if there is more than one contact for the AOR.
-
- Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1
-
-2016-07-11 10:21 +0000 [02877b4b4f] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_configuration.c: Misc cleanups.
-
- * Fix some whitespace in various routines.
-
- * Rename i to iter in persistent_endpoint_update_state().
-
- * Fix off-nominal copy/paste message wording in
- persistent_endpoint_contact_deleted_observer()
-
- Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8
-
-2016-07-13 08:57 +0000 [148cd1b319] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.
-
- Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version.
-
- ASTERISK-26046 #close
-
- Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7
-
-2016-07-11 10:25 +0000 [97b4c7a5b4] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Fix statsd regression.
-
- The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
- patch introduced several regressions when the newly created "Updated"
- state goes out for each endpoint registration refresh.
-
- 1) It restarted any OPTIONS RTT ping cycle.
-
- 2) It would interfere with a currently active ping and throw off that
- ping's resulting RTT calculation.
-
- 3) It cleared the RTT time each time the endpoint was refreshed.
-
- 4) The cleared RTT time was sent out as a statsd update each time.
-
- 5) It created two AMI events for each update.
-
- * Revert the original patch and reimplement it. Now the current contact
- status state is re-sent instead of the state being momentarily toggled
- every time the endpoint refreshes its registration. The statsd events are
- not created for the re-sent refresh because they are sent after every
- OPTIONS ping.
-
- ASTERISK-26160 #close
- Reported by: Matt Jordan
-
- Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
-
-2016-07-12 03:50 +0000 [3be6fa1e4b] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Allow own CFLAGS on ./configure.
-
- Before this change, make failed with the error
- Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH
- when CFLAGS were supplied to the configure script. This was introduced with
- <https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when
- CFLAGS were supplied. Those who need different -march= values, please, go for
- ./configure
- make menuselect.makeopts or make menuselect
- ./menuselect/menuselect --disable BUILD_NATIVE
-
- ASTERISK-25289 #close
-
- Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc
-
-2016-07-11 13:42 +0000 [5ee205d8bb] Richard Mudgett <rmudgett@digium.com>
-
- * ast_expr2: Fix off-nominal memory leak.
-
- Thanks to ibercom for pointing out a memory leak that was missed
- in the earlier patch for the issue.
-
- ASTERISK-26119
- Reported by: Alexei Gradinari
-
- Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71
-
-2016-07-11 10:17 +0000 [f5e9872016] Alexander Traud <pabstraud@compuserve.com>
-
- * install_prereq: Checkout of libSRTP 1.5.x.
-
- Since 5th November 2014, the master branch of libSRTP changed the prefix of
- several member names and is not compatible with the source code in Asterisk
- anymore. Therefore instead, this change checks out the latest version of the
- libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as
- backend. This makes AES-GCM and AES-IN possible.
-
- ASTERISK-22131 #close
-
- Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6
-
-2016-07-10 19:08 +0000 [17efed6cf7] Joshua Colp <jcolp@digium.com>
-
- * func_odbc: Fix connection deadlock.
-
- The func_odbc module was modified to ensure that the
- previous behavior of using a single database connection
- was maintained. This was done by getting a single database
- connection and holding on to it. With the new multiple
- connection support in res_odbc this will actually starve
- every other thread from getting access to the database as
- it also maintains the previous behavior of having only
- a single database connection.
-
- This change disables the func_odbc specific behavior if
- the res_odbc module is running with only a single database
- connection active. The connection is only kept for the
- duration of the request.
-
- ASTERISK-26177 #close
-
- Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f
-
-2016-07-09 13:32 +0000 [06ba533bc7] Corey Farrell <git@cfware.com>
-
- * chan_sip: Fix reference leaks in error paths.
-
- * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error.
- * build_peer leaks peer on failure to allocate the endpoint.
-
- This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed
- with an unref in the appropriate place.
-
- ASTERISK-26184 #close
-
- Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12
-
-2016-07-07 12:41 +0000 [9d4e664f62] Corey Farrell <git@cfware.com>
-
- * REF_DEBUG: Prevent logging of container node objects.
-
- Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being
- recorded to the refs log for the node being replaced. This prevents
- logging of those unrefs since they would produce errors in
- refcounter.py.
-
- ASTERISK-26181 #close
-
- Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4
-
-2016-07-07 10:55 +0000 [e26bd15e7a] Scott Griepentrog <scott@griepentrog.com>
-
- * PJSIP: provide valid tcp nodelay option for reuse
-
- When using TCP transport with chan_pjsip, the TCP_NODELAY
- option value was allocated on the stack, then passed as a
- pointer to the tcp transport configuration structure, and
- later re-used on subsequently created sockets when it was
- no longer valid. This patch changes the allocation to be
- a static.
-
- ASTERISK-26180 #close
- Reported by: Scott Griepentrog
-
- Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0
-
-2016-07-07 10:38 +0000 [77b0145a25] Joshua Colp <jcolp@digium.com>
-
- * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.
-
- Some T.38 implementations may send another re-invite after the initial
- one which adds additional negotiation details (such as the max bitrate).
- Currently this will fail when passthrough is being done in chan_sip as we
- do nothing if T.38 is already active.
-
- Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
- scenario so this change adds support for it to chan_sip and res_pjsip_t38.
- If a request to negotiate is received while T.38 is already enabled a
- new re-INVITE is sent and negotiation is done again.
-
- ASTERISK-26179 #close
-
- Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c
-
-2016-07-04 16:38 +0000 [b4a9fa2c9e] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_sorcery_realtime: fix bug when successful UPDATE is treated as failed
-
- If the SQL UPDATE statement changes nothing then SQLRowCount returns 0.
- This value should be treated as success.
- But the function sorcery_realtime_update treats it as failed.
-
- This bug was found using stress tests on PJSIP.
- If there are 2 consecutive SIP REGISTER requests with the same contact data
- during 1 second then res_pjsip_registrar adds contact location on 1st request
- and tries to update contact location on 2nd.
- The update fails and res_pjsip_registrar even removes correct contact location.
-
- The test "object_update_uncreated" was removed from test_sorcery_realtime.c
- because it's now a valid situation.
-
- This patch also adds missing debug of extra SQL parameter.
-
- ASTERISK-26172 #close
-
- Change-Id: I05a7f3051455336c9dda29efc229decf86071303
-
-2016-06-24 19:55 +0000 [1dfd3fc995] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_session: Check for presence of an active negotiator
-
- It is possible in a hypothetical situation for a session refresh to be
- invoked on a PJSIP when the negotiatior on the INVITE session has not
- yet been established. While this shouldn't occur with existing uses of
- ast_sip_session_refresh, the crashes that occur due to improperly
- calling PJSIP functions that expect a non-NULL negotiatior are
- avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this
- means that simply checking for the presence of the negotiator before
- passing it to other PJSIP functions that use it is allowable. As such,
- this patch adds checks for the presence of the negotiator before calling
- PJSIP functions that assume it is non-NULL.
-
- Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d
-
-2015-10-19 18:55 +0000 [9dd0aeeb44] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_pubsub: Add additional debug statements
-
- When something very sad and wrong occurs, it's challenging sometimes to
- figure out why. This patch adds some additional debug statements on
- off-nominal paths to try and make debugging easier.
-
- Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640
-
-2015-10-19 18:55 +0000 [1ec4f8dd00] Matt Jordan <mjordan@digium.com>
-
- * res/res_corosync: Raise a Stasis message on node join/leave events
-
- When res_corosync detects that a node leaves or joins, it currently is
- informed of this via Corosync callbacks. However, there are a few
- limitations with the information presented:
- (1) While we have information that Corosync is aware of - such as the
- Corosync nodeid - that information is really only useful inside of
- Corosync or res_corosync. There's no way to translate a Corosync
- nodeid to some other internally useful unique identifier for the
- Asterisk instance that just joined or left the cluster.
- (2) While res_corosync is notified of the instance joining or leaving
- the cluster, it has no mechanism to inform the Asterisk core or
- other modules of this event. This limits the usefulness of res_corosync
- as a heartbeat mechanism for other modules.
-
- This patch addresses both issues.
-
- First, it adds the notion of a cluster discovery message both within the
- Stasis message bus, as well as the binary event messages that
- res_corosync uses to transmit data back and forth within the cluster.
- When Asterisk joins the cluster, it sends a discovery message to the other
- nodes in the cluster, which correlates the Corosync nodeid along with
- the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids
- to Asterisk EIDs, such that it can map changes in cluster state with the
- Asterisk instance that has that nodeid. Likewise, when an Asterisk
- instance receives a discovery message from a node in the cluster, it now
- sends its own discovery message back to the originating node with the
- local Asterisk EID. This lets Asterisk instances within the cluster
- build a complete picture of the other Asterisk instances within the
- cluster.
-
- Second, it publishes the discovery messages onto the Stasis message bus.
- Said messages are published whenever a node joins or leaves the cluster.
- Interested modules can subscribe for the ast_cluster_discovery_type()
- message under the ast_system_topic() and be notified when changes in
- cluster state occur.
-
- Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465
-
-2016-07-04 13:54 +0000 [2c16a81dd5] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: Added "subscribe_context" to endpoint
-
- If specified, incoming SUBSCRIBE requests will be searched for the matching
- extension in the indicated context. If no "subscribe_context" is specified,
- then the "context" setting is used.
-
- ASTERISK-25471 #close
-
- Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
-
-2016-07-04 05:58 +0000 [a1bd57884d] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf.
-
- Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This
- avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is
- using AS_HELP_STRING everywhere else already.
-
- ASTERISK-26046
-
- Change-Id: I8299faf504ceaeee3e39930c59293809e116c631
-
-2016-06-30 15:17 +0000 [640fbbbe28] Richard Mudgett <rmudgett@digium.com>
-
- * features: Fix channel datastore access.
-
- Found as a result of the testsuite tests/callparking test crashing.
-
- Several calls to ast_get_chan_featuremap_config() and
- ast_get_chan_features_xfer_config() did not lock the channel before
- calling so the channel's datastore list was accessed without the lock's
- protection. Apparently another thread deleted a datastore on the
- channel's list while the crashing thread was walking the list. Crash at
- 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result.
-
- * Add missing channel locks to calls that were not already protected
- as the doxygen for those calls indicates.
-
- Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1
-
-2016-06-22 17:26 +0000 [359134c8d3] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Don't send extra BYE if SDP invalid.
-
- When an answer SDP is invalid we were disconnecting the outgoing call and
- sending two BYE requests. The first BYE was sent by PJPROJECT because of
- the invalid SDP answer. The second BYE was sent by Asterisk because it
- thought the canceled call was the result of the RFC5407 section 3.1.2 race
- condition.
-
- * Made not send the BYE on a canceled session if the SDP negotiation is
- incomplete because PJPROJECT has already sent a BYE for the failed
- negotiation.
-
- ASTERISK-25772 #close
- Reported by: Dmitriy Serov
-
- Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836
-
-2016-06-27 17:19 +0000 [5fabcf2ca1] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: End call on initial invalid SDP negotiation.
-
- When an incoming call defers SDP negotiation and then sends us an invalid
- SDP in the ACK, we need to send a BYE to disconnect the call. In this
- case SDP negotiation has failed and we don't have valid media streams
- negotiated.
-
- ASTERISK-25772
-
- Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8
-
-2016-06-23 15:13 +0000 [38a4e983dc] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip.c: Register PJMEDIA error code decoder.
-
- Registering the PJMEDIA error codes allows errors found when parsing an
- incoming SDP to be easier to figure out.
-
- "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
- is much easier to understand than "Unknown error 220030".
-
- ASTERISK-25772
-
- Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0
-
-2016-06-27 16:56 +0000 [1952434df5] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Remove unused parameter from handle_incoming().
-
- Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa
-
-2016-06-22 18:02 +0000 [28928ba5c4] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Add missing NULL checks when using pjsip_inv_end_session().
-
- pjsip_inv_end_session() is documented as being able to return the
- passed in tdata parameter set to NULL on success.
-
- Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047
-
-2016-06-30 08:25 +0000 [43a78100c0] gtjoseph <gjoseph@digium.com>
-
- * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject
-
- There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK
- from getting set when using an external pjproject.
-
- ASTERISK-26099 #close
- Reported-by: Ross Beer
-
- Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae
-
-2016-06-29 15:31 +0000 [99eff80e76] Matt Jordan <mjordan@digium.com>
-
- * hep.conf.sample: Default 'enabled' to 'no'
-
- Following the principle of least surprise, we should not be sending
- massive numbers of PJSIP and RTCP HEP packets out into the ether to some
- only-slightly-random IP address. Having 'enabled' set to 'no' in the
- sample configuration file should prevent this from happening for those
- who run 'make samples'.
-
- ASTERISK-26159 #close
-
- Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1
-
-2016-06-29 15:09 +0000 [78960975f2] Matt Jordan <mjordan@digium.com>
-
- * pjproject/patches/config_site: Increase the max number of ICE candidates
-
- When negotiating ICE candidates with WebRTC capable endpoints, many
- networks will result in a browser offering ICE candidates that exceeds
- the default number of max candidates, 16. This patch bumps the max
- candidates to 32, with the max checks at twice the number of candidates.
- In practice, this has shown to be sufficient for browser/WebRTC
- negotiation.
-
- Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5
-
-2016-06-28 09:00 +0000 [d07c8a0504] gtjoseph <gjoseph@digium.com>
-
- * codecs: Fix ABI incompatibility created by adding format_name to ast_codec
-
- Adding format_name even to the end of ast_codec caused issued with
- binary codec modules because the pointer would be garbage in asterisk
- when they registered. So, the ast_codec structure was reverted and an
- internal_ast_codec structure was created just for use in codec.c. A new
- internal-only API was also added (__ast_codec_register_with_format) so
- that codec_builtin could register codecs with the format_name in a
- separate parameter rather than in the ast_codec structure.
-
- ASTERISK-26144 #close
- Reported-by: Alexei Gradinari
-
- Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba
-
-2016-06-28 08:22 +0000 [f3d236ca7f] gtjoseph <gjoseph@digium.com>
-
- * BuildSystem: Fix a few issues hightlighted by gcc 6.x
-
- gcc 6.1.1 caught a few more issues.
- Made sure the unit tests still pass for the func_env and stdtime
- issues.
-
- ASTERISK-26157 #close
-
- Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e
-
-2016-06-28 10:33 +0000 [9d5b0934d9] Matt Jordan <mjordan@digium.com>
-
- * configs/basic-pbx/modules.conf: Remove 'bad' modules
-
- This patch removes the following modules:
- - pbx_functions: It never existed.
- - res_pjsip_log_forwarder: It no longer exists.
- - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
- aren't going to be installing HOMER
- - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
- loaded, and we aren't configured to make use of the
- module
-
- Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5
-
-2016-06-22 11:19 +0000 [1dfc286418] Joshua Colp <jcolp@digium.com>
-
- * siren: Add format attribute modules for Siren7 and Siren14.
-
- This change removes hardcoded SDP parsing and generation for
- Siren7 and Siren14 from chan_sip and moves it to format attribute
- modules so it can also be used by chan_pjsip.
-
- With this the fmtp lines for both are added with the bitrate
- information.
-
- ASTERISK-26021
-
- Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037
-
-2016-06-23 04:33 +0000 [5f0a098243] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf.
-
- Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C
- but requires ANSI C anyway.
-
- ASTERISK-26046
-
- Change-Id: I914c014385e1862102d90fe7650621def78db02e
-
-2016-06-02 17:26 +0000 [b3c787d1dd] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: improve realtime performance #2
-
- The patch removes updating all Endpoints' status on startup.
- Instead, only non-qualified aors with static contact
- and non-qualified non-expired contacts are retrieved from the realtime to
- update the endpoint status to ONLINE.
- The endpoint name was added to the contact object to simply find the endpoint
- that created this contact.
-
- The status of endpoints with qualified aors will be updated by 'qualify'
- functions.
-
- ASTERISK-26061 #close
-
- Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
-
-2016-06-23 13:47 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.10.0-rc1 Released.
-
-2016-06-23 08:42 +0000 [62349ac1b4] Mark Michelson <mmichelson@digium.com>
-
- * Release summaries: Add summaries for 13.10.0-rc1
-
-2016-06-23 08:38 +0000 [8da6ba4328] Mark Michelson <mmichelson@digium.com>
-
- * .version: Update for 13.10.0-rc1
-
-2016-06-23 08:38 +0000 [170b85e3ae] Mark Michelson <mmichelson@digium.com>
-
- * .lastclean: Update for 13.10.0-rc1
-
-2016-06-23 08:38 +0000 [4af7049b8f] Mark Michelson <mmichelson@digium.com>
-
- * realtime: Add database scripts for 13.10.0-rc1
-
-2016-06-22 15:04 +0000 [3d904659ec] Corey Farrell <git@cfware.com>
-
- * res_fax: Fix reference leak in fax_v21_session_new.
-
- fax_v21_session_new created a session details object but only released
- the allocation reference during error conditions. fax_session_new adds
- it's own reference to details if needed so the caller is always
- responsible for cleaning it's own reference.
-
- ASTERISK-26141 #close
-
- Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88
-
-2016-06-22 13:41 +0000 [48db4c2159] gtjoseph <gjoseph@digium.com>
-
- * res_rtp_asterisk: Fix a self-comparison identified by gcc 6
-
- gcc 6 caught a previously unidentified self-comparison in
- ice_candidate_cmp. Fixed it and re-ordered the predicates for better
- short-circuiting.
-
- ASTERISK-26140 #close
-
- Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7
-
-2016-06-22 10:37 +0000 [bc69b03316] gtjoseph <gjoseph@digium.com>
-
- * chan_unistim: Fix memcpy in get_to_address
-
- A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
- was using a pointer to a pointer as the destination of a memcpy and a
- '&' instead of '*' in the sizeof.
-
- ASTERISK-26138 #close
-
- Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708
-
-2016-06-20 13:18 +0000 [1b79e2deff] Mark Michelson <mmichelson@digium.com>
-
- * Fix Alembic upgrades.
-
- A non-existent constraint was being referenced in the upgrade script.
- This patch corrects the problem by removing the reference.
-
- This patch fixes another realtime problem as well. Our Alembic scripts
- store booleans as yes or no values. However, Sorcery tries to insert
- "true" or "false" instead. This patch updates Sorcery to use "yes" and
- "no"
-
- ASTERISK-26128 #close
-
- Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec
-
-2016-06-22 10:55 +0000 [e30602587c] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf.
-
- Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not
- support the platform SVR2 from the year 1987 anymore.
-
- ASTERISK-26046
-
- Change-Id: I28161b037feb2d29ab46ed20e785928460226c22
-
-2016-06-22 10:51 +0000 [77da168e58] gtjoseph <gjoseph@digium.com>
-
- * test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO
-
- Since the file was missing the depends on pjproject, it wasn't
- picking up the pjproject related include path. If there was no
- system installed pjproject and pjproject-bundled was used, a compile
- would fail because pjsip.h wasn't found.
-
- ASTERISK-26139 #close
-
- Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757
-
-2016-06-21 06:52 +0000 [dfcd466bf0] Torrey Searle <torrey@voxbone.com>
-
- * res_rtp_asterisk: fix memory leak in dtls
-
- ensure that cert bios get freed after creating the fingerprint
-
- ASTERISK-26129 #close
-
- Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451
-
-2016-06-21 17:42 +0000 [c982da0641] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro.
-
- Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c
-
-2016-06-12 11:19 +0000 [6a568bcc66] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription
-
- Occasionally under load we'll attempt to send a final NOTIFY on a
- subscription that's already been terminated and a SEGV will occur
- down in pjproject's evsub_destroy function. This is a result of a
- race condition between all the paths that can generate a notify
- and/or destroy the underlying pjproject evsub object:
-
- * The client can send a SUBSCRIBE with Expires: 0.
- * The client can send a SUBSCRIBE/refresh.
- * The subscription timer can expire.
- * An extension state can change.
- * An MWI event can be generated.
- * The pjproject transaction timer (timer_b) can expire.
-
- Normally when our pubsub_on_evsub_state is called with a terminate,
- we push a task to the serializer and return at which point the dialog
- is unlocked. This is usually not a problem because the task runs
- immediately and locks the dialog again. When the system is heavily
- loaded though, there may be a delay between the unlock and relock
- during which another event may occur such as the subscription timer
- or timer_b expiring, an extension state change, etc. These may also
- cause a terminate to be processed and if so, we could cause pjproject
- to try to destroy the evsub structure twice. There's no way for us to
- tell that the evsub was already destroyed and the evsub's group lock
- can't tolerate this and SEGVs.
-
- The remedy is twofold.
-
- * A patch has been submitted to Teluu and added to the bundled
- pjproject which adds add/decrement operations on evsub's group lock.
-
- * In res_pjsip_pubsub:
- * configure.ac and pjproject-bundled's configure.m4 were updated
- to check for the new evsub group lock APIs.
- * We now add a reference to the evsub group lock when we create
- the subscription and remove the reference when we clean up the
- subscription. This prevents evsub from being destroyed before
- we're done with it.
- * A state has been added to the subscription tree structure so
- termination progress can be tracked through the asyncronous tasks.
- * The pubsub_on_evsub_state callback has been split so it's not doing
- double duty. It now only handles the final cleanup of the
- subscription tree. pubsub_on_rx_refresh now handles both client
- refreshes and client terminates. It was always being called for
- both anyway.
- * The serialized_on_server_timeout task was removed since
- serialized_pubsub_on_rx_refresh was almost identical.
- * Missing state checks and ao2_cleanups were added.
- * Some debug levels were adjusted to make seeing only off-nominal
- things at level 1 and nominal or progress things at level 2+.
-
- ASTERISK-26099 #close
- Reported-by: Ross Beer.
-
- Change-Id: I779d11802cf672a51392e62a74a1216596075ba1
-
-2016-06-21 07:05 +0000 [ef97911a1c] Alexander Traud <pabstraud@compuserve.com>
-
- * res_rtp_asterisk: Use latest DTLS version available by underlying platform.
-
- Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the
- underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for
- WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based
- cipher-suites.
-
- ASTERISK-26130 #close
-
- Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0
-
-2016-06-21 10:53 +0000 [69d58a1e37] Scott Griepentrog <scott@griepentrog.com>
-
- * PJSIP: provide transport type with received messages
-
- The receipt of a SIP MESSAGE may occur over any transport including TCP
- and TLS. When the message is received, the original URI is added to the
- message in the field PJSIP_RECVADDR, but this is insufficient to ensure
- a reply message can reach the originating endpoint. This patch adds the
- PJSIP_TRANSPORT field populated with the transport type.
-
- ASTERISK-26132 #close
-
- Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e
-
-2016-06-21 08:01 +0000 [cbfa9f771e] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf.
-
- Some configure scripts used both AC_HELP_STRING and its replacement
- AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were
- changed to AS_HELP_STRING.
-
- ASTERISK-26046
-
- Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f
-
-2016-06-20 10:29 +0000 [ba0d9e7f7a] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_session: Handle race condition at shutdown with timer.
-
- When shutting down res_pjsip_session will get unloaded before res_pjsip.
- The act of unloading unregisters all the PJSIP services and sets
- their module IDs to -1. In some cases it is possible for a timer to
- occur after this happens which calls into res_pjsip_session. The
- res_pjsip_session module can then try to get the session from the
- INVITE session using the module ID. Since the module ID is now -1
- this fails.
-
- This change stores a copy of the module ID and uses it for the timer
- callback scenario. If the module ID is -1 the callback immediately
- returns but if the module ID is valid then it continues as normal.
-
- This works as the original ID of the module is guaranteed to still
- be valid when used with the INVITE session.
-
- ASTERISK-26127 #close
-
- Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573
-
-2016-06-20 12:13 +0000 [c1512f4108] Richard Mudgett <rmudgett@digium.com>
-
- * app_voicemail.c: Fix IMAP compile error.
-
- Fix compile error introduced by the patch for
- ASTERISK-26045
-
- Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3
-
-2016-06-16 15:56 +0000 [5134a8043a] Alexei Gradinari <alex2grad@gmail.com>
-
- * fix: memory leaks, resource leaks, out of bounds and bugs
-
- ASTERISK-26119 #close
-
- Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c
-
-2016-06-13 17:40 +0000 [cfebe3b94a] Mark Michelson <mmichelson@digium.com>
-
- * ARI: Ensure announcer channels are destroyed.
-
- Announcer channels were not being destroyed because the
- stasis_app_control structure that referenced them was not being
- destroyed. The control structure was not being destroyed because it was
- not being unlinked from its container. It was not being unlinked from
- its container because the after bridge callback for the announcer
- channel was not being run. The after bridge callback was not being run
- because the after bridge datastore was not being removed from the
- channel on destruction. The channel was not being destroyed because the
- hangup that used to destroy the channel was now only reducing the
- reference count to one. The reference count of the channel was only
- being reduced to one because the stasis_app_control structure was
- holding the final reference...
-
- The control structure used to not keep a reference to the channel, so
- that loop described above did not happen.
-
- The solution is to manually remove the control structure from its
- container when the playback on a bridge is complete.
-
- ASTERISK-26083 #close
- Reported by Joshua Colp
-
- Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4
-
-2016-06-20 08:05 +0000 [76516bd79d] Alexander Traud <pabstraud@compuserve.com>
-
- * http: leverage 'bindaddr' for TLS in http.conf
-
- The internal HTTP/WebSocket server supports both TCP and TLS, which can be
- activated separately via the file http.conf. The source code intends to re-use
- the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified
- explicitly. This did not work because of a typo. This change resolves this typo.
-
- ASTERISK-26126 #close
-
- Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f
-
-2016-05-31 09:10 +0000 [89cc86fc38] Vasil Kolev <vasil.kolev@securax.org>
-
- * chan_sip: bigger buffers for headers, better failure mode
-
- Currently chan_sip can give weird messages if the contacts don't
- fit in the From: or To: headers. This fix changes the from,to and
- invite variables to use ast_str, allocates and deallocates them and
- resizes them if needed.
-
- ASTERISK-26069 #close
-
- Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3
-
-2016-05-18 17:37 +0000 [d53a36ff33] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_transport_management.c: Misc cleanups to survive shutdown.
-
- * In unload_module(), reordered destroying things to minimize the window
- that the global transports container could be used by other threads on
- shutdown. When shutting down you need to stop things in the opposite
- order of creation.
-
- * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to
- eliminate the crash potential by other threads using the container on
- shutdown.
-
- * Made struct monitored_transport.sip_received not use
- ast_atomic_fetchadd_int() since it is used as a boolean value that is only
- set TRUE. It was previously incremented for every received SIP message
- and could theoretically overflow.
-
- * In monitored_transport_state_callback(), allocated the monitored
- transport object without a lock since the lock was unused.
-
- * In keepalive_global_loaded(), removed releasing the transports container
- if the keepalive_thread could not be started. I set it up to be tried
- again if the user reloads the configuration.
-
- Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff
-
-2016-01-05 19:08 +0000 [03953d8034] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip.c: Add check that timer actually got scheduled.
-
- Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1
-
-2016-06-13 13:33 +0000 [32ab98116e] Richard Mudgett <rmudgett@digium.com>
-
- * res_rtp_multicast.c: Fix warning message typo.
-
- Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3
-
-2016-02-11 18:15 +0000 [0429c53368] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Reorganize ast_sip_session_terminate().
-
- Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b
-
-2016-06-10 12:35 +0000 [5823f279f3] Richard Mudgett <rmudgett@digium.com>
-
- * chan_rtp: Backport changes from master.
-
- * Deprecate chan_multicast_rtp.
-
- Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e
-
-2016-06-10 16:13 +0000 [dde58df318] Richard Mudgett <rmudgett@digium.com>
-
- * chan_rtp.c: Copy file from chan_multicast_rtp.c
-
- Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef
-
-2016-06-08 06:15 +0000 [ca38a3cbb4] Alexander Traud <pabstraud@compuserve.com>
-
- * core: Not the configured but granted number of possible file descriptors.
-
- With CLI "core show settings", simply the parameter maxfiles of the file
- asterisk.conf was shown. If that parameter was not set, nothing was displayed
- although the environment might have set a default number itself. Or if maxfiles
- were not granted (completely), still maxfiles was shown. Now, the maximum number
- of possible file descriptors in the environment is shown.
-
- ASTERISK-26097
-
- Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b
-
-2016-06-07 18:45 +0000 [caf6cccc5c] Joshua Colp <jcolp@digium.com>
-
- * cel: Ensure only one dial status per channel exists.
-
- CEL wrongly assumed that a channel would only have a single dial
- event on it. This is incorrect. Particularly in a queue each
- call attempt to a member will result in a dial event, adding
- a new dial status in CEL without removing the old one. This
- would cause the container to grow with only one dial status
- being removed when the channel went away. The other dial status
- entries would remain leaking memory.
-
- This change fixes the memory leak by ensuring that only one dial
- status will only ever exist for each channel.
-
- The behavior during the scenario where multiple events are received
- has also been improved. For failure cases the first failure will
- be the dial status. If an answer dial status is received, though,
- it will take priority and the dial status for the channel will be
- answer.
-
- Memory usage has also been decreased by storing the minimal
- amount of information and the code has been cleaned up slightly.
-
- ASTERISK-25262 #close
-
- Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe
-
-2016-06-09 10:37 +0000 [715ef071a1] Mark Michelson <mmichelson@digium.com>
-
- * chan_pjsip: Lock channel when checking for RTP changes.
-
- bridge_native_rtp can call into an RTP-capable channel driver in order
- for the driver to update information about who the channel is
- communicating with. For SIP channel drivers, this means deactivating
- RTCP and sending a reinvite so that the endpoints can communicate
- directly.
-
- bridge_native_rtp does the right thing and has the channel locked when
- calling into the channel driver. chan_pjsip can't alter session
- properties in this thread, though. chan_pjsip queues a task on the
- session serializer in order to update properties there.
-
- The problem is that this queued task was not locking the channel. This
- meant that the queued task could attempt to deactivate RTCP at the same
- time that the channel thread was attempting to process an incoming RTCP
- packet. This could lead to a crash.
-
- This patch fixes the issue by locking the channel in the queued task
- when altering RTP properties.
-
- ASTERISK-26092 #close
- Reported by Niklas Larsson
-
- Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159
-
-2016-06-09 09:20 +0000 [a99ddc6a0d] gtjoseph <gjoseph@digium.com>
-
- * build: Fix ast_sockaddr initialization to be more portable
-
- A change to glibc 2.22 changed the order of the sockadddr_storage
- members which caused the places where we do an initialization of
- ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those
- initializers (which we shouldn't have been using anyway) have been
- replaced with memsets.
-
- Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4
-
-2016-06-08 12:26 +0000 [eabb398d71] Matt Jordan <mjordan@digium.com>
-
- * res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded
-
- A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not
- loaded and does not have a configuration file. Previously when this
- occurred, checks were put in to see if the configuration was loaded
- successfully. While this is a good idea - and has been added to the
- offending function in res_hep - the reality is res_hep_pjsip and
- res_hep_rtcp have no business running if res_hep isn't also running.
-
- As such, this patch also adds a function to res_hep that returns whether
- or not it successfully loaded. Oddly enough, ast_module_check returns
- "everything is peachy" even if a module declined its load - so it cannot
- be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this
- function to see if they should continue to load; if it fails, they
- decline their load as well.
-
- ASTERISK-26096 #close
-
- Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea
-
-2016-06-08 05:58 +0000 [0d84421f93] Alexander Traud <pabstraud@compuserve.com>
-
- * astfd: Not maximum size of a single file but maximum file descriptors.
-
- With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a
- single file was shown. Now, the maximum number of possible file descriptors is
- shown.
-
- ASTERISK-26097
-
- Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3
-
-2016-06-02 14:53 +0000 [9c5a0b814b] Timo Teräs <timo.teras@iki.fi>
-
- * Fix #include poll.h and sys/cdefs.h
-
- POSIX defines poll.h, sys/poll.h should not be used at is c-library
- internal header which may or may not exist. Notable in musl it
- generates warning of being incorrect. And add explict include of
- sys/cdefs.h where needed.
-
- Change-Id: I142930df53fe7585a06b854b6faddc5301e024be
-
-2016-06-03 22:44 +0000 [9c35f34301] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition.
-
- This patch fixes a race condition processing received REGISTER requests
- and their retransmissions caused by REGISTER requests being processed by
- two threads. The "sip_transaction Unable to register REGISTER transaction
- (key exists)" message is a notable symptom of this issue.
-
- This issue was more likely to happen before the pjsip/distributor
- serializers were created. Instead of steps one and two below placing the
- REGISTER messages into the same pjsip/distributor they were placed in
- random pjsip/default serializers.
-
- 1) REGISTER requests come in and get placed on the pjsip/distributor
- serializer.
-
- 2) Before the first request is processed a retransmission comes in and is
- placed on the same pjsip/distributor serializer.
-
- 3) The first request goes up the pjsip stack and is then shunted off to
- the pjsip/aor/<aor> serializer.
-
- 4) Before the first request is completed processing in the pjsip/aor/<aor>
- serializer, the second request goes up the pjsip stack and is also shunted
- off to the pjsip/aor/<aor> serializer.
-
- 5) The first request completes processing and sends out its response.
-
- 6) The second request completes processing and tries to send out its
- response but pjlib complains that the REGISTER transaction key already
- exists.
-
- 7) Sadness ensues.
-
- * The race is eliminated by removing the pjsip/aor/<aor> serializer and
- continuing the processing in the pjsip/distributor serializer. Now any
- retransmissions queued in the pjsip/distributor serializer will be
- processed after the first message is completely processed.
-
- ASTERISK-26088 #close
- Reported by: Richard Mudgett
-
- Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a
-
-2016-06-03 11:35 +0000 [557333ea4c] Richard Mudgett <rmudgett@digium.com>
-
- * stasis: Add setting subscription congestion levels.
-
- Stasis subscriptions and message routers create taskprocessors to process
- the event messages. API calls are needed to be able to set the congestion
- levels of these taskprocessors for selected subscriptions and message
- routers.
-
- * Updated CDR, CEL, and manager's stasis subscription congestion levels
- based upon stress testing. Increased the congestion levels to reduce the
- potential for bursty call setup/teardown activity from triggering the
- taskprocessor overload alert. CDRs in particular need an extra high
- congestion level because they can take awhile to process the stasis
- messages.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: Id0a716394b4eee746dd158acc63d703902450244
-
-2016-06-02 18:19 +0000 [110d772467] Richard Mudgett <rmudgett@digium.com>
-
- * sorcery: Add setting object type congestion levels.
-
- Sorcery creates taskprocessors for object types to process object observer
- callbacks. An API call is needed to be able to set the congestion levels
- of these taskprocessors for selected object types.
-
- * Updated PJSIP's contact and contact_status sorcery object type observer
- default congestion levels based upon stress testing. Increased the
- congestion levels to reduce the potential for bursty register/unregister
- and subscribe/unsubscribe activity from triggering the taskprocessor
- overload alert.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6
-
-2016-06-02 16:08 +0000 [610eee2a36] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessors: Implement high/low water mark alerts.
-
- When taskprocessors get backed up, there is a good chance that we are
- being overloaded and need to defer adding new work to the system.
-
- * Implemented a high/low water alert mechanism for modules to check if the
- system is being overloaded and take appropriate action. When a
- taskprocessor is created it has default congestion levels set. A
- taskprocessor can later have those congestion levels altered for specific
- needs if stress testing shows that the taskprocessor is a symptom of
- overloading or needs to handle bursty activity without triggering an
- overload alert.
-
- * Add CLI "core show taskprocessor" low/high water columns.
-
- * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was
- never a good thing to use when creating a taskprocessor because of the
- nature of how its references needed to be cleaned up on a partial
- creation.
-
- * Made res_pjsip's distributor check if the taskprocessor overload alert
- is active before placing a message representing brand new work onto a
- distributor serializer.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I182f1be603529cd665958661c4c05ff9901825fa
-
-2016-05-27 17:31 +0000 [26e3492246] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session: Use distributor serializer for incoming calls.
-
- We must continue using the serializer that the original INVITE came in on
- for the dialog. There may be retransmissions already enqueued in the
- original serializer that can result in reentrancy and message sequencing
- problems.
-
- Outgoing call legs create the pjsip/outsess/<endpoint> serializers for
- their dialogs.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc
-
-2016-05-27 16:28 +0000 [ceb1007ed7] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer.
-
- * Resolves potential reentrancy problems if system restarted in the middle
- of subscription message transactions.
-
- * Fixes memory leak recreating persistent subscriptions when the
- subscription resource tree could not be created.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be
-
-2016-05-27 12:50 +0000 [27bafc3a8b] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions.
-
- We must continue using the serializer that the original SUBSCRIBE came in
- on for the dialog. There may be retransmissions already enqueued in the
- original serializer that can result in reentrancy and message sequencing
- problems. The "sip_transaction Unable to register SUBSCRIBE transaction
- (key exists)" message is a notable symptom of this issue.
-
- Outgoing subscriptions still create the pjsip/pubsub/<endpoint>
- serializers for their dialogs.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0
-
-2016-05-26 17:35 +0000 [16b08444da] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_distributor.c: Consistently pick a serializer for messages.
-
- Incoming messages that are not part of a dialog or a recognized response
- to one of our requests need to be sent to a consistent serializer. Under
- load we may be queueing retransmissions before we can process the original
- message. We don't need to throw these messages onto random serializers
- and cause reentrancy and message sequencing problems.
-
- * Created a pool of pjsip/distributor serializers that get picked by
- hashing the call-id and remote tag strings of the received messages.
-
- * Made ast_sip_destroy_distributor() destroy items in the reverse order of
- creation.
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I2ce769389fc060d9f379977f559026fbcb632407
-
-2016-06-02 12:51 +0000 [993b769524] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_distributor.c: Ignore messages until fully booted.
-
- We should not be processing any incoming messages until we are fully
- booted. We may not have dialplan or other needed configuration loaded
- yet.
-
- ASTERISK-26089 #close
- Reported by: Scott Griepentrog
-
- ASTERISK-26088
- Reported by: Richard Mudgett
-
- Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264
-
-2016-06-02 12:04 +0000 [321a9b128f] Joshua Colp <jcolp@digium.com>
-
- * res_odbc: Implement a connection pool.
-
- Testing has shown that our usage of UnixODBC is problematic
- due to bugs within UnixODBC itself as well as the heavy weight
- cost of connecting and disconnecting database connections, even
- when pooling is enabled.
-
- For users of UnixODBC 2.3.1 and earlier crashes would occur due
- to insufficient protection of the disconnect operation. This was
- fixed in UnixODBC 2.3.2 and above.
-
- For users of UnixODBC 2.3.3 and higher a slow-down would occur
- under heavy database use due to repeated connection establishment.
- A regression is present where on each connection the database
- configuration is cached again, with the cache growing out of
- control.
-
- The connection pool implementation present in this change helps
- to mitigate these issues by reducing how much we connect and
- disconnect database connections. We also solve the issue of
- crashes under UnixODBC 2.3.1 by defaulting the maximum number of
- connections to 1, returning us to the previous working behavior.
- For users who may have a fixed version the maximum concurrent
- connection limit can be increased helping with performance.
-
- The connection pool works by keeping a list of active connections.
- If the connection limit has not been reached a new connection is
- established. If the connection limit has been reached then the
- request waits until a connection becomes available before
- continuing.
-
- ASTERISK-26074 #close
- ASTERISK-26054 #close
-
- Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff
-
-2016-06-07 05:45 +0000 [c6ee4a0f44] Alexander Traud <pabstraud@compuserve.com>
-
- * res_srtp: Instead of libSRTP use OpenSSL as random source.
-
- Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore.
- Therefore, the symbol RAND_bytes is used instead of crypto_get_random.
-
- ASTERISK-24436 #close
-
- Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96
-
-2016-06-07 02:16 +0000 [d38b8e6399] Alexander Traud <pabstraud@compuserve.com>
-
- * BuildSystem: Avoid 'ar cru' and use 'ar cr' instead.
-
- In several internal library projects, the files are archived with the help of
- 'ar cr'. Only the projects editline and the Objective Open H.323 stack
- implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms
- changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier
- ignored since `D' is the default (see `U')". For consistency and to avoid this
- message all projects use 'ar cr' now.
-
- ASTERISK-26091 #close
-
- Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40
-
-2016-05-27 14:49 +0000 [c27c232057] gtjoseph <gjoseph@digium.com>
-
- * ari/resource_channels: Add 'formats' to channel create/originate
-
- If you create a local channel and don't specify an originator channel
- to take capabilities from, we automatically add all audio formats to
- the new channel's capabilities. When we try to make the channel
- compatible with another, the "best format" functions pick the best
- format available, which in this case will be slin192. While this is
- great for preserving quality, it's the worst for performance and
- overkill for the vast majority of applications.
-
- In the absense of any other information, adding all formats is the
- correct thing to do and it's not always possible to supply an
- originator so a new parameter 'formats' has been added to the channel
- create/originate functions. It's just a comma separated list of formats
- to make availalble for the channel. Example: "ulaw,slin,slin16".
- 'formats' and 'originator' are mutually exclusive.
-
- To facilitate determination of format names, the format name has been
- added to "core show codecs".
-
- ASTERISK-26070 #close
-
- Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b
-
-2016-06-02 04:59 +0000 [cda3385409] Joshua Colp <jcolp@digium.com>
-
- * alembic: Fix migration.
-
- The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting
- to use UniqueConstraint and failing. It was not imported and after
- importing it also continued to fail.
-
- I've changed the script to use the explicit name of the constraint
- instead.
-
- Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9
-
-2016-06-01 13:57 +0000 [e2132dd358] Richard Mudgett <rmudgett@digium.com>
-
- * logging,cdr,cel: Fix stringfield memory leak.
-
- The stringfields refactor to allow adding stringfields to the end of a
- structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some
- incomplete cleanup code by some stringfield users.
-
- The most noticeable leaker is the logging system where there is a leak for
- every log message generated.
-
- ASTERISK-26078 #close
- Reported by: Etienne Lessard
- Patches:
- jira_asterisk_26078_v13.patch (license #5621) patch uploaded
- by Richard Mudgett
-
- Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782
-
-2016-05-25 10:34 +0000 [2de58c6d01] Alexei Gradinari <alex2grad@gmail.com>
-
- * core/dial: New channel variable FORWARDERNAME
-
- Added a new channel variable FORWARDERNAME which indicates which
- channel was responsible for a forwarding requests received on dial attempt.
-
- Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
-
- ASTERISK-26059 #close
-
- Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
-
-2016-05-31 13:02 +0000 [b2ce0e354b] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_distributor.c: Use correct rdata info access method (Part 2).
-
- The pjproject doxygen for rdata->msg_info.info says to call
- pjsip_rx_data_get_info() instead of accessing the struct member directly.
- You need to call the function mostly because the function will generate
- the struct member value if it is not already setup.
-
- Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799
-
-2016-05-30 19:27 +0000 [fe305ccf01] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip_mwi_body_generator: Re-order the body items
-
- Re-ordered the body items so Message-Account is second.
-
- Messages-Waiting: no
- Message-Account: sip:1571@<IP Removed>:5060
- Voice-Message: 0/0 (0/0)
-
- ASTERISK-26065 #close
- Reported-by: Ross Beer
-
- Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3
-
-2016-05-30 10:58 +0000 [e8abfdcdc5] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Move to pjproject 2.5
-
- Although all the patches we had against 2.4.5 were applied by Teluu,
- a new bug was introduced preventing re-use of tcp and tls transports
- This patch removes all the previous patches against 2.4.5, updates
- the version to 2.5, and adds a new patch to correct the transport
- re-use problem.
-
- Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068
-
-2016-05-27 12:25 +0000 [37d039fdf3] Rusty Newton <rnewton@digium.com>
-
- * res_pjsip: Add clarifying documentation to PJSIP_HEADER help text
-
- Added notes about when you can read or write headers. Specifically
- about being able to read on the inbound channel and write on an
- outbound channel.
-
- ASTERISK-26063 #close
- Reported by: Private Name
- Tested by: Rusty Newton
-
- Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5
-
-2016-05-25 18:30 +0000 [03d5b3ce5c] Richard Mudgett <rmudgett@digium.com>
-
- * pjsip_distributor.c: Use correct rdata info access method.
-
- The pjproject doxygen for rdata->msg_info.info says to call
- pjsip_rx_data_get_info() instead of accessing the struct member directly.
- You need to call the function mostly because the function will generate
- the struct member value if it is not already setup.
-
- Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2
-
-2016-05-20 13:56 +0000 [859bbec09b] Alexei Gradinari <alex2grad@gmail.com>
-
- * app_voicemail: fix bugs, imap mm_status log change to debug
-
- Fixed some bugs:
- - create dirpath when save downloading message from IMAP storage.
- - create IMAP folder if not exists when saving to IMAP storage
- - check if file successfully opened before write to it
- - some IMAP checks
- - remove non-standard flag 'Unseen'
- etc
-
- Change to debug IMAP mm_status log instead of verbose.
-
- Remove unused X-Asterisk-VM-Caller-channel message header
- for security reason. The clients should not know name of peer/endpoint.
-
- ASTERISK-26045 #close
-
- Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b
-
-2016-05-19 14:56 +0000 [230686f4ec] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: add "via_addr", "via_port", "call_id" to contact
-
- As res_pjsip_nat rewrites contact's address, only the last Via header
- can contain the source address of registered endpoint.
- Also Call-Id header may contain the source address of registered
- endpoint.
-
- Added "via_addr", "via_port", "call_id" to contact.
- Added new fields ViaAddress, CallID to AMI event ContactStatus.
-
- ASTERISK-26011
-
- Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
-
-2016-05-24 16:56 +0000 [04c12561a7] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: chatty verbose messages
-
- There are a lot of verbose messages about Endpoint and Contact status
- changes if there are many dynamic endpoints.
- The patch sets verbose level 2 for Endpoint status changes
- and verbose level 3 for Contact status changes.
-
- ASTERISK-26055 #close
-
- Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7
-
-2016-02-12 09:59 +0000 [a42bea3314] Corey Farrell <git@cfware.com>
-
- * threadpool: Fix potential data race.
-
- worker_start checked for ZOMBIE status without holding a lock. All
- other read/write of worker status are performed with a lock, so this
- check should do the same.
-
- ASTERISK-25777 #close
-
- Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781
-
-2016-05-18 10:58 +0000 [a32616d60c] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * Makefile: remove OSARCH check for init install
-
- There are more specific checks for the platform.
-
- Specifically this allows installing OS/X init scripts.
-
- ASTERISK-26038 #close
-
- Change-Id: If08933621145b10362a0cfe73c079301d9c13f50
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-21 05:42 +0000 [9ddaab789e] Jesper (License 5518)
-
- * func_curl: Don't trim response text on non-ASCII characters
-
- The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of
- a signed comparison.
-
- ASTERISK-25669 #close
- Reported by: Jesper
- patches:
- strings.curl.trim.patch submitted by Jesper (License 5518)
-
- Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a
-
-2016-05-20 16:59 +0000 [9453d1187a] Richard Mudgett <rmudgett@digium.com>
-
- * parking.h: Update ast_parking_park_call() doxygen to reality.
-
- ASTERISK-26029
-
- Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260
-
-2016-05-10 14:30 +0000 [cd89501d48] Alexei Gradinari <alex2grad@gmail.com>
-
- * func_odbc: single database connection should be optional
-
- func_odbc was changed in Asterisk 13.9.0
- to make func_odbc use a single database connection per DSN
- because of reported bug ASTERISK-25938
- with MySQL/MariaDB LAST_INSERT_ID().
-
- This is drawback in performance when func_odbc is used
- very often in dialplan.
-
- Single database connection should be optional.
-
- ASTERISK-26010
-
- Change-Id: I57d990616c957dabf7597dea5d5c3148f459dfb6
-
-2016-05-20 09:39 +0000 [c0b190dd9a] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Match dialogs on responses better.
-
- When receiving an incoming response to a dialog-starting INVITE, we were
- not matching the response to the INVITE dialog. Since we had not
- recorded the to-tag to the dialog structure, the PJSIP-provided method
- to find the dialog did not match.
-
- Most of the time, this was not a problem, because there is a fall-back
- that makes the response get routed to the same serializer that the
- request was sent on. However, in cases where an asynchronous DNS lookup
- occurs in the PJSIP core, the thread that sends the INVITE is not
- actually a threadpool serializer thread. This means we are unable to
- record a serializer to handle the incoming response.
-
- Now, imagine what happens when an INVITE is sent on a non-serialized
- thread, and an error response (such as a 486) arrives. The 486 ends up
- getting put on some random threadpool thread. Eventually, a hangup task
- gets queued on the INVITE dialog serializer. Since the 486 is being
- handled on a different thread, the hangup task can execute at the same
- time that the 486 is being handled. The hangup task assumes that it is
- the sole owner of the INVITE session and channel, so it ends up
- potentially freeing the channel and NULLing the session's channel
- pointer. The thread handling the 486 can crash as a result.
-
- This change has the incoming response match the INVITE transaction, and
- then get the dialog from that transaction. It's the same method we had
- been using for matching incoming CANCEL requests. By doing this, we get
- the INVITE dialog and can ensure that the 486 response ends up being
- handled by the same thread as the hangup, ensuring that the hangup runs
- after the 486 has been completely handled.
-
- ASTERISK-25941 #close
- Reported by Javier Riveros
-
- Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0
-
-2016-05-19 11:41 +0000 [ddcf983e39] Joshua Colp <jcolp@digium.com>
-
- * res_sorcery_astdb: Filter fields to only the registered ones.
-
- This change introduces the same filtering that is done in res_sorcery_realtime
- to the res_sorcery_astdb module. This allows persisted sorcery objects
- that may contain unknown fields to still be read in from the AstDB
- and used. This is particularly useful when switching between different
- versions of Asterisk that may have introduced additional fields.
-
- ASTERISK-26014 #close
-
- Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2
-
-2016-05-09 21:40 +0000 [39fedfa423] snuffy <snuffy22@gmail.com>
-
- * res_pjsip_empty_info: Respond to empty SIP INFO packets
-
- Some SBCs require responses to empty SIP INFO packets
- after establishing call via INVITE, if not responded to
- they may drop your call after unspecified timeout of X minutes.
-
- They are identified by having no Content-Type, check for this
- and respond with 200 - OK message.
-
- ASTERISK-24986 #close
- Reported-by: Ilya Trikoz, Federico Santulli
-
- Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
-
-2016-05-18 07:54 +0000 [935e0496c4] gtjoseph <gjoseph@digium.com>
-
- * udptl: Don't eat sequence numbers until OK is received
-
- Scenario:
- Local fax -> Asterisk w/ firewall -> Provider -> Remote fax
-
- * Local fax starts rtp call to remote fax
- * Remote fax starts t38 call back to local fax.
- * Local fax sends t38 no-signal to Asterisk before sending an OK.
- * udptl processes the frame and increments the expected sequence number.
- * chan_sip drops the frame because the call isn't up so nothing goes out
- the external interface to open the port for incoming packets.
- * Local fax sends OK and Asterisk sends OK to the remote fax.
- * Remote fax sends t38 packets which are dropped by the firewall.
- * Local fax re-sends t38 no-signal with the same sequence number.
- * udptl drops the frame because it thinks it's a dup.
- * Still no outgoing packets to open the firewall.
- * t38 negotiation fails.
-
- The patch drops frames t38 received before udptl sequence processing
- when the call hasn't been answered yet. The second no-signal frame
- is then seen as new and is relayed out the external interface which
- opens the port and allows negotiation to continue.
-
- ASTERISK-26034 #close
-
- Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9
-
-2016-05-17 11:14 +0000 [77e8ec162b] gtjoseph <gjoseph@digium.com>
-
- * chan_sip: Prevent extra Session-Expires headers from being added
-
- When chan_sip does a re-INVITE to refresh a session and authentication
- is required, the INVITE with the Authorization header containes a
- second Session-Expires header without the ";refersher=" parameter.
- This is causing some proxies to return a 400. Also, when Asterisk is
- the uas and the refresher, it is including the Session-Expires and
- Min-SE headers in OPTIONS messages which is not allowed per RFC4028.
-
- This patch (based on the reporter's) Checks to see if a Session-Expires
- header is already in the message before adding another one. It also
- checks that the method is INVITE or UPDATE.
-
- ASTERISK-26030 #close
-
- Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9
-
-2016-05-16 15:29 +0000 [3f6ef63099] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip_outbound_registration: Clean up state when registration is deleted
-
- Nothing was cleaning up the registration state object when ast_sorcery_delete
- was called on a registration. So, the registration was deleted from sorcery
- but the state object went right on refreshing the registration (or failing
- to refresh the registration) with the peer.
-
- * Added a 'deleted' observer on registration that removes the state object.
-
- ASTERISK-25964 #close
- Reported-by Matt Jordan
-
- Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23
-
-2016-05-15 19:05 +0000 [b6f9392a12] gtjoseph <gjoseph@digium.com>
-
- * res_pjsip: Set TCP_NODELAY on TCP transports
-
- Although it's perfectly legal to place multiple SIP messages in the same packet,
- it can cause problems because the Linux default is to enable Path MTU Discovery
- which sets the Don't Fragment bit on the packets. If adding a second message to
- the packet causes the MTU to be exceeded, and the destination isn't equipped to
- send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
- dropped.
-
- We can't specifically tell the stack to send only 1 message per packet, but we
- can turn on TCP_NODELAY when we create the transport. This will at least tell
- the stack to send packets as soon as possible.
-
- ASTERISK-26005 #close
- Reported-by: Ross Beer
-
- Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd
-
-2016-05-14 21:48 +0000 [361a16f316] Matt Jordan <mjordan@digium.com>
-
- * configs/samples/pjsip.conf.sample: Fix typo
-
- A ':' is not a valid token for starting a comment.
-
- Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad
-
-2016-05-12 07:08 +0000 [f91a7dc993] Matt Jordan <mjordan@digium.com>
-
- * res/res_hep_pjsip: Fix reported local IP address when bound to 'any'
-
- When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its
- local address the 'any' address, as opposed to the IP address we
- actually received the packet on. This can cause some confusion in Homer,
- as it will dutifully report what we send it.
-
- This patch uses the PJSIP inspection routines to determine which IP
- address we probably received the packet on based on the remote party's
- IP address. In the event that this fails, it falls back to the IP
- address natively reported by the transport.
-
- Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3
-
-2016-05-14 12:29 +0000 [9de5cd209e] Sean Bright <sean.bright@gmail.com>
-
- * res_ari: Correct Location headers returned by some ARI resources
-
- The Location headers returned by:
-
- * /bridges/{bridgeId}/play
- * /bridges/{bridgeId}/record
- * /channels/{channelId}/play
- * /channels/{channelId}/record
-
- Did not have the '/ari' prefix, and in the case of the 'play' resources, were
- using 'playback' instead of 'playbacks.'
-
- Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c
-
-2016-05-13 11:38 +0000 [524a302974] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: Endpoint IP Access Controls
-
- With the old SIP module we can use IP access controls per peer.
- PJSIP module missing this feature.
-
- This patch added next configuration Endpoint options:
- "acl" - list of IP ACL section names in acl.conf
- "deny" - List of IP addresses to deny access from
- "permit" - List of IP addresses to permit access from
- "contact_acl" - List of Contact ACL section names in acl.conf
- "contact_deny" - List of Contact header addresses to deny
- "contact_permit" - List of Contact header addresses to permit
-
- This patch also better logging failed request:
- add custom message instead of "No matching endpoint found"
- add SIP method to logging
-
- ASTERISK-25900
-
- Change-Id: I456dea3909d929d413864fb347d28578415ebf02
-
-2016-05-11 20:17 +0000 [89ae4466ea] Matt Jordan <mjordan@digium.com>
-
- * res_hep: Provide an option to pick the UUID type
-
- At one point in time, it seemed like a good idea to use the Asterisk
- channel name as the HEP correlation UUID. In particular, it felt like
- this would be a useful identifier to tie PJSIP messages and RTCP
- messages together, along with whatever other data we may eventually send
- to Homer. This also had the benefit of keeping the correlation UUID
- channel technology agnostic.
-
- In practice, it isn't as useful as hoped, for two reasons:
- 1) The first INVITE request received doesn't have a channel. As a
- result, there is always an 'odd message out', leading it to be
- potentially uncorrelated in Homer.
- 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
- This causes RTCP information to be uncorrelated to the SIP message
- traffic seen by those capture nodes.
-
- In order to support both (in case someone is trying to use res_hep_rtcp
- with a non-PJSIP channel), this patch adds a new option, uuid_type, with
- two valid values - 'call-id' and 'channel'. The uuid_type option is used
- by a module to determine the preferred UUID type. When available, that
- source of a correlation UUID is used; when not, the more readily available
- source is used.
-
- For res_hep_pjsip:
- - uuid_type = call-id: the module uses the SIP Call-ID header value
- - uuid_type = channel: the module uses the channel name if available,
- falling back to SIP Call-ID if not
- For res_hep_rtcp:
- - uuid_type = call-id: the module uses the SIP Call-ID header if the
- channel type is PJSIP and we have a channel,
- falling back to the Stasis event provided
- channel name if not
- - uuid_type = channel: the module uses the channel name
-
- ASTERISK-25352 #close
-
- Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
-
-2016-05-10 02:56 +0000 [a73d79c22f] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * basic-cfg: asterisk.conf: remove [directories]
-
- A minimal configuration does not need to explicitly spell out the
- directories. The built-in defaults will do just fine. In many cases
- they are wrong.
-
- Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-10 03:06 +0000 [1c56de9453] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * basic-cfg: asterisk.conf: defaults of options
-
- Note the default of remmed-out options. To clarify that those values are
- not the defaults.
-
- Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-10 03:08 +0000 [d7af591c59] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * basic-cfg: asterisk.conf: debug level 5 spams
-
- Don't suggest users to use debug level 5, which spews (usually
- non-useful) debug information. Reduce the suggestion to (an
- arbitrarily-selected) level 2.
-
- Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-10 03:10 +0000 [9b7db18fc1] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * basic-cfg: asterisk.conf: don't set languages
-
- * No need to set language in a miniml configuration. 'en' will do just
- fine.
- * It would be useful to have an example of setting it to a different
- language.
- * Setting the documentation language explicitly is likewise not
- required. Setting it to a different value is not common. At least
- until there is a set of translated documentation.
-
- Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-10 08:17 +0000 [eec539a46e] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * followme: delete the right recorded name file
-
- FollowMe with the option a records the name of the caller and plays it
- to the callee. However it has failed to clean up that recorded file
- as it tried to delete the file name without the '.sln' extension.
-
- ASTERISK-26008 #close
-
- Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec
- Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
-2016-05-12 14:36 +0000 [02d30e171e] Mark Michelson <mmichelson@digium.com>
-
- * Use doubles instead of floats for conversions when comparing strings.
-
- In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
- be deleted at seemingly random times.
-
- One reason this was happening was because of an operation to retrieve
- the contacts whose expiration time was less than or equal to the current
- time. When retrieving existing contacts, the contact's expiration time
- and the current time were converted from a string to a float, and those
- two floats were compared.
-
- On some systems, including mine, this conversion was horribly off. For
- instance, I could regularly see the string "1463079214" get converted
- into 1463079168.000000. When switching from using a float to using a
- double, the conversion was as expected.
-
- Why was the conversion to float off? My best guess is that the
- conversion to float was attempting to store the entire value in the 23
- bit significand of the IEEE-754 floating point number. In particular, if
- you take only the 23 most significant bits of 1463079214, you get the
- messed up 1463079168 that we were seeing in the conversion. It likely
- was possible to get a more precise value by composing the number using
- an exponent, but the conversion did not work that way. With a double,
- you have a 52 bit significand, allowing the entire value to fit there,
- and thereby allowing an accurate conversion.
-
- ASTERISK-26007 #close
- Reported by Greg Siemon
-
- Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070
-
-2016-05-12 09:13 +0000 [e2df15bae9] gtjoseph <gjoseph@digium.com>
-
- * pjsip_distributor: Add missing newline to NOTICE
-
- There was a newline missing from the end of the "no matching endpoint" notice.
-
- Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181
-
-2016-05-10 10:19 +0000 [a94a12bbf7] Sebastian Damm <damm@sipgate.de>
-
- * res_pjsip_outbound_registration: generate correct Contact URI for TLS
-
- There are two types of SIP URIs indicating a secure transport:
- * sips:user@example.org
- * sip:user@example.org;transport=tls
-
- When using a sips URI, Asterisk checks incoming INVITEs and answers from
- the other side for sips URIs, and rejects the packet if there are only
- sip URIs. So Asterisk should only generate a sips Contact URI if the
- other side supports it.
-
- This patch makes Asterisk generate either a sip or sips Contact URI
- depending on the format of the server URI.
-
- If you want a sip URI, use:
- server_uri=sip:example.org\;transport=tls
-
- If you want a sips URI, use:
- server_uri=sips:example.org
-
- ASTERISK-25990 #close
- Reported-by: Sebastian Damm
-
- Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2
-
-2016-05-05 16:41 +0000 [36d66a23e0] Alexei Gradinari <alex2grad@gmail.com>
-
- * logger: Add PID to syslog messages.
-
- During refactoring of this support the addition of
- the PID to messages was removed. This change adds it
- back in.
-
- ASTERISK-25538 #close
-
- Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36
-
-2016-05-11 14:07 +0000 [37214b0bdf] Matt Jordan <mjordan@digium.com>
-
- * configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER
-
- When running on a system that does not support or use AST_UNDEFINED_SANITIZER
- or AST_LEAK_SANITIZER, the configure script would incorrectly set those
- constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
- cause menuselect to error out, complaining that a blank value is not a
- valid option. This patch corrects the issue by setting the value to 0 if
- the options that those constants enable/disable is not found.
-
- Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba
-
-2016-05-03 15:43 +0000 [49b25a0956] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches
-
- When reloading, or fetching realtime data, if the "apply" failed for any
- numerous reasons the current state object would not be maintained. This
- potentially resulted in publishes being stopped for some states/clients when
- they should not have been.
-
- This patch makes it so the current state object is kept upon any type of reload/
- fetch failures.
-
- Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30
-
-2016-05-03 15:31 +0000 [1b5c91b7be] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_publish: Potential crash due to off nominal path
-
- It was possible for the explicit publish destroy function to be called without
- the pjsip client ever being initialized. This fix checks to make sure there is
- a client to destroy before attempting.
-
- Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c
-
-2016-05-03 15:35 +0000 [10de553c9d] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_publishing: After unloading the library won't load again
-
- The same thing was happening in res_pjsip_publish_asterisk. When the library
- was unloaded it did not unregister the object type from sorcery. Subsequent
- loads resulted in a failed load due to the sorcery type already existing.
-
- Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9
-
-2016-05-03 14:59 +0000 [1a833b9739] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_publish: Ref leak in off nominal callback paths
-
- There were a few spots where the client object's reference was being leaked in
- sip_outbound_publish_callback. This patch cleans up those leaks.
-
- Change-Id: I485d0bc9335090f373026f77c548042e258461df
-
-2016-05-03 15:39 +0000 [4752ef02e0] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_publish: Won't unload if condition wait times out
-
- When res_pjsip_outbound_publish unloads it has to wait for all current
- publishing objects to get done. However if the wait condition times out
- then it does not fail the unload. This sometimes results in an infinite
- loop check while unloading. This patch now fails the unload operation if
- the condition times out.
-
- Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec
-
-2016-05-05 11:37 +0000 [4d063814ba] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_authenticator_digest: Don't use source port in nonce verification
-
- From the issue reporter:
- "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
- the timestamp, the source address, the source port, a server UUID that is
- calculated at startup, and the authentication realm.
-
- Rather than caching nonces that we create, we instead attempt to re-calculate
- the nonce when receiving an incoming request with authentication. We then
- compare the re-calculated nonce to the incoming nonce, and if they don't match,
- then authentication has failed early.
-
- The problem is that it is possible, especially when using TCP, to receive two
- requests from the same endpoint but have differing source ports for those
- requests. Asterisk itself commonly will use different source ports for
- outbound TCP requests."
-
- This patch removes the source port dependency when building the nonce.
-
- ASTERISK-25978 #close
-
- Change-Id: I871b5f4adce102df1c4988066283095ec509dffe
-
-2016-05-07 14:39 +0000 [fb6227a372] gtjoseph <gjoseph@digium.com>
-
- * config_transport: Tell pjproject to allow all SSL/TLS protocols
-
- The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2.
- SSL is not allowed. So, even if you specify "sslv3" for a transport method,
- it's silently ignored and one of the TLS protocols is used. This was a new
- behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that
- we never caught.
-
- Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default().
- This tells pjproject to set the socket protocol to match the method.
-
- ASTERISK-26004 #close
-
- Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078
-
-2016-05-04 02:40 +0000 [2db17a793c] Jaco Kroon <jaco@uls.co.za>
-
- * app_confbridge: Add a regcontext option for confbridge bridge profiles.
-
- This patch allows for having app_confbridge register the name of the
- conference as an extension into a specific context, similar to
- regcontext for chan_sip. This variant is not quite as involved as the
- one in chan_sip and doesn't allow for multiple contexts or custom
- extensions, you can only specify the context and the conference name
- will always be used as the extension to register.
-
- ASTERISK-25989 #close
-
- Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
-
-2016-05-08 20:19 +0000 [2a7130b8b0] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Check for python-dev and TEST_FRAMEWORK
-
- The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
- The python bindings are now built only if TEST_FRAMEWORK is set and a
- python development package is installed.
-
- libresample was also disabled.
-
- ASTERISK-25993 #close
- Reported-by: Joshua Colp
-
- Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
-
-2016-05-04 15:16 +0000 [72eb7c8301] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: module load priority
-
- The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
- and res_pjsip_registrar modules should load ASAP
- to avoid "No matching endpoint found" for legitimate endpoint.
-
- ASTERISK-25994
-
- Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b
-
-2016-05-04 03:17 +0000 [dd00c71aae] Chris Trobridge <christ.trobridge@ultra-aep.com>
-
- * config_options.c: Expand #ifdef to contain whole if statement.
-
- ASTERISK-25956 #close
-
- Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38
-
-2016-05-05 15:16 +0000 [e6eb17efd9] Alexei Gradinari <alex2grad@gmail.com>
-
- * stasis_endpoints: Add new Status and Headers to ContactStatus
-
- ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
- ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
- These additions should be also in stasis_endpoints
- to include in command "manager show event ContactStatus"
-
- Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a
-
-2016-05-05 05:07 +0000 [fa11f4c920] Joshua Colp <jcolp@digium.com>
-
- * file: Ensure nativeformats remains valid for lifetime of use.
-
- It is possible for the nativeformats of a channel to change
- throughout its lifetime. As a result a user of it needs to either
- ensure the channel is locked when accessing the formats or keep
- a reference to the nativeformats themselves.
-
- This change fixes the file playback support so it keeps a
- reference to the nativeformats when accessing things.
-
- ASTERISK-25998 #close
-
- Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915
-
-2016-04-15 09:32 +0000 [9c2032240e] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: improve realtime performance
-
- This patch modified pjsip_options to retrieve only
- permament contacts for aor if the qualify_frequency is > 0
- and persisted contacts if the qualify_frequency is > 0.
-
- This patch also fixed a bug in res_sorcery_astdb.
- res_sorcery_astdb doesn't save object data retrived from astdb.
-
- ASTERISK-25826
-
- Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05
-
-2016-05-04 16:11 +0000 [fe38d21c2a] Alexei Gradinari <alex2grad@gmail.com>
-
- * pjsip: Added "reg_server" to contacts (fixed alembic)
-
- ASTERISK-25931
-
- Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549
-
-2016-04-07 16:33 +0000 [7a14e669f0] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip/AMI: add contact.updated event
-
- With the old SIP module AMI sends PeerStatus event on every
- successfully REGISTER requests, ie, on start registration,
- update registration and stop registration.
-
- With PJSIP AMI sends ContactStatus only when status is changed.
- Regarding registration:
- on start registration - Created
- on stop registration - Removed
- but on update registration nothing
-
- This patch added contact.updated event.
-
- ASTERISK-25904
-
- Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
-
-2016-05-02 16:08 +0000 [06d4ac0355] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_fax: add FAXMODE variable
-
- The app_fax set FAXMODE variable, but res_fax missing this feature.
- This patch add FAXMODE variable which is set to either "audio" or "T38".
-
- ASTERISK-25980
-
- Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b
-
-2016-05-02 16:52 +0000 [2d17fe06c5] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_fax/t38_gateway: Peer V.21 session is created on wrong channel
-
- The channel and peer V.21 sessions are created on the same channel now.
- The peer V.21 session should be created only on peer channel
- when one of channel can handle T.38.
-
- Also this patch enable debug for T.38 gateway session
- if global fax debug enabled.
-
- ASTERISK-25982
-
- Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e
-
-2016-05-01 02:21 +0000 [a2f19d82a8] Diederik de Groot <dkgroot@talon.nl>
-
- * configs/basic-pbx/asterisk.conf: contains incorrect path separator
-
- Note: When packagers use these files (as an example) the paths are never
- really used when they are split using '='.
-
- Note: Thirdparty applications will also have trouble parsing the file when
- expecting '=>'.
-
- Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004
-
-2016-04-30 17:52 +0000 [f39089f17c] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Various fixes discovered during testing of OSes
-
- For all OSes:
- * Disabled third-party codecs in pjproject and added
- '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
- configure options since we don't use the pjsip codec capability.
-
- FreeBSD:
- * Added FreeBSD support to install_prereq.
- * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
- * Added __progname and environ to asterisk.exports.in.
- * Reverted the use of ldconfig to create shared library symlinks to ln.
- * Only enable epoll in pjproject if `uname -s` is Linux.
- * Added a patch to pjproject to take the name of the 'make' command from
- an environment variable if supplied. This is needed for the python bindings.
- (merged by Teluu into pjproject trunk 5/3/2016)
- FreeBSD support isn't complete. Still some general issues regarding
- make/gmake having nothing to do with pjproject. With some handholding it DOES
- build successfully.
-
- CentOS:
- Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
- CentOS 6/7 32/64 build and run the pjsip testsuite successfully.
-
- Ubuntu:
- No changes required.
- Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.
-
- Debian:
- No changes required.
- Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.
-
- There will utimately be a follow-up patch to create an install_prereq for
- the testsuite as I've discovered a few missing requirements.
-
- ASTERISK-25968 #close
-
- Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c
-
-2016-03-17 14:29 +0000 [8028fc7585] Andrew Nagy <andrew.nagy@the159.com>
-
- * app_voicemail: always copy dynamic struct to avoid race condition
-
- Voicemail email addresses can be corrupt or voicemail
- emails can end up being sent to the wrong email address if asterisk is
- reading voicemail.conf during a reload and processing an email at the
- same time. This patch always copies the struct that would otherwise only
- be copied once.
-
- ASTERISK-24463 #close
- Reported by: John Campbell
- Tested by: Etienne Lessard
- Tested by: Andrew Nagy
- Change-Id: I3a0643813116da84e2617291903d0d489b7425fb
-
-2016-04-15 14:26 +0000 [3cb8934de0] Alexei Gradinari <alex2grad@gmail.com>
-
- * pjsip: Added "reg_server" to contacts.
-
- If the Asterisk system name is set in asterisk.conf, it will be stored
- into the "reg_server" field in the ps_contacts table to facilitate
- multi-server setups.
-
- ASTERISK-25931
-
- Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
-
-2016-04-28 11:35 +0000 [7992923c70] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Start body generator users after suppliers.
-
- Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb
-
-2016-04-28 16:06 +0000 [5dc0e082b2] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Add useful information to some messages.
-
- Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a
-
-2016-04-26 15:58 +0000 [f9e416f053] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Fix body generator registration race.
-
- Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67
-
-2016-04-26 15:13 +0000 [b1b2019046] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.h: Fix doxygen association.
-
- Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632
-
-2016-04-25 16:00 +0000 [b7f07fdff5] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_publish.c: Remove redundant flag check.
-
- Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353
-
-2016-04-28 16:54 +0000 [719ece5659] gtjoseph <gjoseph@digium.com>
-
- * pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE
-
- When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled,
- the input uri string will become corrupted if it contains escape sequences.
- It's not possible to automatically strdup or strdupa the input string because
- the output uri pj_str_t's will have pointers to chunks of the input string.
- Getting around this would require more memory management code and wouldn't
- be worth the savings of doing the unescape in place.
-
- ASTERISK-25970 #close
- Reported-by: Dmitriy Serov
-
- Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88
-
-2016-03-07 18:34 +0000 [38bed4515d] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add ability to identify by Authorization username
-
- A feature of chan_sip that service providers relied upon was the ability to
- identify by the Authorization username. This is most often used when customers
- have a PBX that needs to register rather than identify by IP address. From my
- own experiance, this is pretty common with small businesses who otherwise
- don't need a static IP.
-
- In this scenario, a register from the customer's PBX may succeed because From
- will usually contain the PBXs account id but an INVITE will contain the caller
- id. With nothing recognizable in From, the service provider's Asterisk can
- never match to an endpoint and the INVITE just stays unauthorized.
-
- The fixes:
-
- A new value "auth_username" has been added to endpoint/identify_by that
- will use the username and digest fields in the Authorization header
- instead of username and domain in the the From header to match an endpoint,
- or the To header to match an aor. This code as added to
- res_pjsip_endpoint_identifier_user rather than creating a new module.
-
- Although identify_by was always a comma-separated list, there was only
- 1 choice so order wasn't preserved. So to keep the order, a vector was added
- to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
- to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
- globals/endpoint_identifier_order.
-
- Along the way, the logic in res_pjsip_registrar was corrected to match
- most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
-
- The order is:
-
- username@domain
- username@domain_alias
- username
-
- Auth by username does present 1 problem however, the first INVITE won't have
- an Authorization header so the distributor, not finding a match on anything,
- sends a securty_alert. It still sends a 401 with a challenge so the next
- INVITE will have the Authorization header and presumably succeed. As a result
- though, that first security alert is actually a false alarm.
-
- To address this, a new feature has been added to pjsip_distributor that keeps
- track of unidentified requests and only sends the security alert if a
- configurable number of unidentified requests come from the same IP in a
- configurable amout of time. Those configuration options have been added to
- the global config object. This feature is only used when auth_username
- is enabled.
-
- Finally, default_realm was added to the globals object to replace the hard
- coded "asterisk" used when an endpoint is not yet identified.
-
- The testsuite tests all pass but new tests are forthcoming for this new
- feature.
-
- ASTERISK-25835 #close
- Reported-by: Ross Beer
-
- Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
-
-2016-04-27 13:23 +0000 [677d5b5151] Mark Michelson <mmichelson@digium.com>
-
- * func_odbc: Check connection status before executing queries.
-
- A recent change to func_odbc made it so that a single connection was
- maintained per DSN. The problem was that the code was optimistic about
- the health of the connection after initially opening it and did nothing
- to re-connect in case the connection had died.
-
- This change adds a check before executing a query to ensure that the
- connection to the database is still up and running.
-
- ASTERISK-25963 #close
- Reported by Ross Beer
-
- Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d
-
-2016-04-15 11:59 +0000 [df3639700a] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: disable multi domain to improve realtime performace
-
- This patch added new global pjsip option 'disable_multi_domain'.
- Disabling Multi Domain can improve Realtime performance by reducing
- number of database requests.
-
- ASTERISK-25930 #close
-
- Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
-
-2016-04-26 11:13 +0000 [949bf6b282] Joshua Colp <jcolp@digium.com>
-
- * chan_sip: Give more time for TCP/TLS threads to stop.
-
- The unload process currently tells each TCP/TLS to terminate but
- does not wait for them to do so. This introduces a race condition
- where the container holding the threads may be destroyed before
- the threads are able to remove themselves from it. When they
- finally do the container is invalid and can't be used causing a
- crash.
-
- A previous change existed which waited a bit to wait for any
- stranglers to finish. This change extends this and waits longer.
-
- ASTERISK-25961 #close
-
- Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6
-
-2016-04-26 05:48 +0000 [6959f5484b] Joshua Colp <jcolp@digium.com>
-
- * app_queue: Fix crash when unloading module.
-
- When unloading the app_queue module the members in each queue are
- destroyed and as part of this they are removed from the pending
- members container. Unfortunately a crash would occur as the container
- was destroyed before the members were removed.
-
- This change tweaks ordering so the container destruction occurs
- after the members are destroyed.
-
- ASTERISK-16115
-
- Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b
-
-2016-04-24 22:51 +0000 [b38f1146e5] gtjoseph <gjoseph@digium.com>
-
- * config: Fix ast_config_text_file_save2 writability check for missing files
-
- A patch I did back in 2014 modified ast_config_text_file_save2 to check the
- writability of the main file and include files before truncating and re-writing
- them. An unintended side-effect of this was that if a file doesn't exist,
- the check fails and the write is aborted.
-
- This patch causes ast_config_text_file_save2 to check the writability of the
- parent directory of missing files instead of checking the file itself. This
- allows missing files to be created again. A unit test was also added to
- test_config to test saving of config files.
-
- The regression was discovered when app_voicemail's passwordlocation=spooldir
- feature stopped working.
-
- ASTERISK-25917 #close
- Reported-by: Jonathan Rose
-
- Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80
-
-2016-04-21 23:53 +0000 [29bab0d1a4] Kirill Katsnelson <kkm@smartaction.com>
-
- * chan_sip: Make autocreated peers send PeerStatus events
-
- Since Stasis has been introduced, an attempt to send AMI messages by an
- autocreated peer caused a crash, and all events from autocreated peers were
- semi-inadvertently disabled altogether in 0b83761. This change restores the
- disabled functionality.
-
- ASTERISK-25950
-
- Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974
-
-2016-04-21 14:23 +0000 [c345e530f4] Kevin Harwell <kharwell@digium.com>
-
- * app_queue: queue members can receive multiple calls
-
- It was possible for a queue member that is a member of at least 2 or more
- queues to receive mulitiple calls at the same time. This happened because
- of a race between when a member was being rung and when the device state
- notified the other queue(s) member object of the state change.
-
- This patch makes it so when a queue member is being rung it gets added to
- a global pool of queue members. If that same member is tried again, e.g.
- from another queue, and it is found to already exist in the pending member
- container then it will not ring that member.
-
- ASTERISK-16115 #close
-
- Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48
-
-2016-04-25 08:11 +0000 [c0688a6398] Javier Acosta <javier.acosta@beeonline.es>
-
- * Fix case sensitive actions in AMI QueueSummary and QueueStatus
-
- ASTERISK-25954 #close
- Reported by: Javier Acosta
-
- Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256
-
-2016-04-22 15:25 +0000 [ebf0724a83] Richard Mudgett <rmudgett@digium.com>
-
- * test_message.c: Wait longer in case dialplan also processes the test message.
-
- Bumped the wait from 1 second to 5 seconds. The test message was hitting my
- default call handler and failing the test because it took longer.
-
- Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba
-
-2016-04-12 15:29 +0000 [ba63aa7c9e] Richard Mudgett <rmudgett@digium.com>
-
- * Manager: Short circuit AMI message processing.
-
- Improve AMI message processing performance if there are no consumers
- listening for the messages. We now skip creating the AMI event message
- text strings.
-
- Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3
-
-2016-04-13 17:54 +0000 [d5ee6acf28] Richard Mudgett <rmudgett@digium.com>
-
- * manager.c: Eliminate most RAII_VAR usage.
-
- * Made ast_manager_event_blob_create() not allocate the ao2 event object
- with a lock as it is not needed.
-
- Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c
-
-2016-04-13 17:09 +0000 [7303e3dc96] Richard Mudgett <rmudgett@digium.com>
-
- * manager_channels.c: Fix allocation failure crash.
-
- An earlier allocation failure failed to create a channel snapshot for the
- AMI HangupRequest/SoftHangupRequest event which resulted in a crash in
- channel_hangup_request_cb(). Where the stasis message gets generated
- cannot tell if the NULL snapshot returned was because of an allocation
- failure or the channel was a dummy channel.
-
- * Made channel_hangup_request_cb() check if the channel blob has a
- snapshot and exit if it doesn't.
-
- * Eliminated the RAII_VAR usage in channel_hangup_request_cb().
-
- Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24
-
-2016-04-13 13:50 +0000 [1e93f3d723] Richard Mudgett <rmudgett@digium.com>
-
- * Bridge system: Fix memory leaks and double frees on impart failure.
-
- You cannot reference the passed in features struct after calling
- ast_bridge_impart(). Even if the call fails.
-
- Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21
-
-2016-04-13 13:20 +0000 [5e388d4188] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_softmix.c: Fix crash if channel fails to join mixing tech.
-
- softmix_bridge_join() failed because of an allocation failure. To address
- this, the softmix bridge technology now checks if the channel failed to
- join softmix successfully. In addition, the bridge now begins the process
- of kicking the channel out of the bridge so we don't have channels
- partially in the bridge for very long.
-
- * Fix the test_channel_feature_hooks.c unit tests. The test channel must
- have a valid codec to join the simple_bridge technology. This patch makes
- joining a bridge more strict by not allowing partially joined channels to
- remain in the bridge.
-
- Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b
-
-2016-03-17 12:28 +0000 [9740277713] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)
-
- There are several places that do scheduled tasks or periodic housecleaning,
- each with its own implementation:
-
- * res_pjsip_keepalive has a thread that sends keepalives.
- * pjsip_distributor has a thread that cleans up expired unidentified requests.
- * res_pjsip_registrar_expire has a thread that cleans up expired contacts.
- * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task.
- * res_pjsip_sdp_rtp also uses ast_sched to send keepalives.
-
- There are also places where we should be doing scheduled work but aren't.
- A good example are the places we have sorcery observers to start registration
- or qualify. These don't work when changes are made to a backend database
- without a pjsip reload. We need to check periodically.
-
- As a first step to solving these issues, a new ast_sip_sched facility has
- been created.
-
- ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue.
- When a task is ready to run, ast_sip_task_pusk is called for it. This ensures
- that the task is executed in a PJLIB registered thread and doesn't hold up the
- ast_sched thread so it can immediately continue processing the queue. The
- serializer used by ast_sip_sched is one of your choosing or a random one from
- the res_pjsip pool if you don't choose one.
-
- Another feature is the ability to automatically clean up the task_data when the
- task expires (if ever). If it's an ao2 object, it will be dereferenced, if
- it's a malloc'd object it will be freed. This is selectable when the task is
- scheduled. Even if you choose to not auto dereference an ao2 task data object,
- the scheduler itself maintains a reference to it while the task is under it's
- control. This prevents the data from disappearing out from under the task.
-
- There are two scheduling models.
-
- AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at
- the specific interval. That is, every "interval" milliseconds, regardless of
- how long the task takes. If the task takes longer than the interval, it will
- be scheduled at the next available multiple of interval. For exmaple: If the
- task has an interval of 60 secs and the task takes 70 secs (it better not),
- the next invocation will happen at 120 seconds.
-
- AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should
- start "interval" milliseconds after the current invocation has finished.
-
- Also, the same ast_sched facility for fixed or variable intervals exists. The
- task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or
- AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time.
-
- One res_pjsip.h housekeeping change was made. The pjsip header files were
- added to the top. There have been a few cases lately where I've needed
- res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because
- I didn't add the pjsip header files to my source even though I never referenced
- any pjsip calls.
-
- Finally, a few new convenience APIs were added to astobj2 to make things a
- little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and
- ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros
- were also copied from res_phoneprov because I got tired of having to duplicate
- the same hash, sort and compare functions over and over again. The
- AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for
- aor_container_alloc into your source.
-
- This facility can be used immediately for the situations where we already have
- a thread that wakes up periodically or do some scheduled work. For the
- registration and qualify issues, additional sorcery and schema changes would
- need to be made so that we can easily detect changed objects on a periodic
- basis without having to pull the entire database back to check. I'm thinking
- of a last-updated timestamp on the rows but more on this later.
-
- Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c
-
-2016-04-25 21:43 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.9.0-rc1 Released.
-
-2016-04-25 16:42 +0000 [5237b7cd47] Joshua Colp <jcolp@digium.com>
-
- * Release summaries: Add summaries for 13.9.0-rc1
-
-2016-04-25 16:40 +0000 [5a3850ecba] Joshua Colp <jcolp@digium.com>
-
- * .version: Update for 13.9.0-rc1
-
-2016-04-25 16:40 +0000 [60b39040e4] Joshua Colp <jcolp@digium.com>
-
- * .lastclean: Update for 13.9.0-rc1
-
-2016-04-25 16:40 +0000 [5078454464] Joshua Colp <jcolp@digium.com>
-
- * realtime: Add database scripts for 13.9.0-rc1
-
-2016-04-22 17:53 +0000 [eb7c581806] gtjoseph <gjoseph@digium.com>
-
- * res_agi: Prevent run_agi from eating frames it shouldn't
-
- The run_agi function is eating control frames when it shouldn't be. This is
- causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond
- transfer.
-
- Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie
- answers.
-
- Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE
- and is left thinking he's connected to Bob.
-
- In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls
- an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on
- Charlie's channel.
-
- The fix was to accumulate deferrable frames in the "forever" loop instead of
- dropping them, and re-queue them just before running the actual agi command
- or exiting.
-
- ASTERISK-25951 #close
-
- Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645
-
-2016-04-22 13:49 +0000 [068ae54c76] Mark Michelson <mmichelson@digium.com>
-
- * func_odbc: Use one connection per DSN.
-
- res_odbc was changed in Asterisk 13.8.0 to remove connection management,
- opting instead to let unixodbc maintain open connections and return
- those to Asterisk as requested.
-
- This was a boon for realtime, since it meant that multiple threads could
- potentially run parallel queries since they could each be using their
- own database connections.
-
- However, on the user-facing side, func_odbc, there were some inherent
- behaviors being relied on that no longer hold true after the change.
- One such reported behavior was that MySQL's LAST_INSERTED_ID() works
- per-connection. This means that if Asterisk uses separate connections
- for every database operation, whereas before it used one connection for
- everything, we have broken expectations and functionality.
-
- The fix provided in this patch is to make func_odbc use a single
- database connection per DSN. This way, user-facing database usage will
- have the same behavior as it did pre-13.8.0. However, realtime, which is
- the real workhorse of database interaction, will continue to let
- unixodbc manage connections.
-
- ASTERISK-25938 #close
- Reported by Edwin Vandamme
-
- Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc
-
-2016-04-22 13:02 +0000 [6aeefa89bc] Leif Madsen <leif@leifmadsen.com>
-
- * Remove reference to non-existent sip.conf option
-
- Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f
-
- ASTERISK-25927 #close
-
- Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8
-
-2016-04-21 08:26 +0000 [e750ea9b5b] Diederik de Groot <dkgroot@talon.nl>
-
- * lock.c: Check *lt before dereferencing it
-
- *lt is NULL if t->tracking == 0
-
- ASTERISK-25948 #close
-
- Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba
-
-2016-04-15 14:36 +0000 [a036c35903] Richard Mudgett <rmudgett@digium.com>
-
- * res_stasis: Handle re-enter stasis bridge with swap channel.
-
- We lose the fact that there is a swap channel if there is one. We
- currently wind up rejoining the stasis bridge as a normal join after the
- swap channel has already been kicked from the bridge.
-
- This patch preserves the swap channel so the AMI/ARI events can note that
- the channel joining the bridge is swapping with another channel. Another
- benefit to swaqpping in one operation is if there are any channels that
- get lonely (MOH, bridge playback, and bridge record channels). The lonely
- channels won't leave before the joining channel has a chance to come back
- in under stasis if the swap channel is the only reason the lonely channels
- are staying in the bridge.
-
- ASTERISK-25947 #close
- Reported by: Richard Mudgett
-
- ASTERISK-24649
- Reported by: John Bigelow
-
- ASTERISK-24782
- Reported by: John Bigelow
-
- Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee
-
-2016-04-19 16:58 +0000 [9942d50aa5] Richard Mudgett <rmudgett@digium.com>
-
- * bridge: Hold off more than one imparting channel at a time.
-
- An earlier patch blocked the ast_bridge_impart() call until the channel
- either entered the target bridge or it failed. Unfortuantely, if the
- target bridge is stasis and the imprted channel is not a stasis channel,
- stasis bounces the channel out of the bridge to come back into the bridge
- as a proper stasis channel. When the channel is bounced out, that
- released the block on ast_bridge_impart() to continue. If the impart was
- a result of a transfer, then it became a race to see if the swap channel
- would get hung up before the imparted channel could come back into the
- stasis bridge. If the imparted channel won then everything is fine. If
- the swap channel gets hung up first then the transfer will fail because
- the swap channel is leaving the bridge.
-
- * Allow a chain of ast_bridge_impart()'s to happen before any are
- unblocked to prevent the race condition described above. When the channel
- finally joins the bridge or completely fails to join the bridge then the
- ast_bridge_impart() instances are unblocked.
-
- ASTERISK-25947
- Reported by: Richard Mudgett
-
- ASTERISK-24649
- Reported by: John Bigelow
-
- ASTERISK-24782
- Reported by: John Bigelow
-
- Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1
-
-2016-04-19 17:52 +0000 [516c626a7d] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_callerid: Clear out display name if id->name is not valid
-
- When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning
- the From header, then it overwrites the display name and uri from the channel's
- connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was
- leaving the display name from the From header in the new RPID or PAI header.
- On an attended transfer where the originator had a caller id number set but not
- a display name, the re-INVITE to the final transferee had the number of the
- originator but the display name of the transferer.
-
- Added a check to clear out the display name in the new header if
- connected.id.name was invalid.
-
- ASTERISK-25942 #close
-
- Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b
-
-2016-04-19 13:02 +0000 [ded3794fc6] Joshua Colp <jcolp@digium.com>
-
- * app_talkdetect: Make the module core supported.
-
- This module is used as part of testsuite tests to confirm
- stuff works. I'm accordingly marking it as core as it is
- required by those tests.
-
- Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88
-
-2016-04-18 12:12 +0000 [efae187217] Mark Michelson <mmichelson@digium.com>
-
- * PJSIP: Remove PJSIP parsing functions from uri length validation.
-
- The PJSIP parsing functions provide a nice concise way to check the
- length of a hostname in a SIP URI. The problem is that in order to use
- those parsing functions, it's required to use them from a thread that
- has registered with PJLib.
-
- On startup, when parsing AOR configuration, the permanent URI handler
- may not be run from a PJLib-registered thread. Specifically, this could
- happen when Asterisk was started in daemon mode rather than
- console-mode. If PJProject were compiled with assertions enabled, then
- this would cause Asterisk to crash on startup.
-
- The solution presented here is to do our own parsing of the contact URI
- in order to ensure that the hostname in the URI is not too long. The
- parsing does not attempt to perform a full SIP URI parse/validation,
- since the hostname in the URI is what is important.
-
- ASTERISK-25928 #close
- Reported by Joshua Colp
-
- Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
-
-2016-04-18 17:00 +0000 [f436b9ab11] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_registrar: Fix bad memory-ness with user_agent.
-
- Recent changes to the PJSIP registrar resulted in tests failing due to
- missing AOR_CONTACT_ADDED test events. The reason for this was that the
- user_agent string had junk values in it, resulting in being unable to
- generate the event.
-
- I'm going to be honest here, I have no idea why this was happening. Here
- are the steps needed for the user_agent variable to get messed up:
- * REGISTER is received
- * First contact in the REGISTER results in a contact being removed
- * Second contact in the REGISTER results in a contact being added
- * The contact, AOR, expiration, and user agent all have to be passed as
- format parameters to the creation of a string. Any subset of those
- parameters would not be enough to cause the problem.
-
- Looking into what was happening, the thing that struck me as odd was
- that the user_agent variable was meant to be set to the value of the
- User-Agent SIP header in the incoming REGISTER. However, when removing a
- contact, the user_agent variable would be set (via ast_strdupa inside a
- loop) to the stored contact's user_agent. This means that the
- user_agent's value would be incorrect when attempting to process further
- contacts in the incoming REGISTER.
-
- The fix here is to use a different variable for the stored user agent
- when removing a contact. Correcting the behavior to be correct also
- means the memory usage is less weird, and the issue no longer occurs.
-
- ASTERISK-25929 #close
- Reported by Joshua Colp
-
- Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
-
-2016-04-18 13:41 +0000 [49bfdc9ac0] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_transport_management: Allow unload to occur.
-
- At shutdown it is possible for modules to be unloaded that wouldn't
- normally be unloaded. This allows the environment to be cleaned up.
-
- The res_pjsip_transport_management module did not have the unload
- logic in it to clean itself up causing the res_pjsip module to not
- get unloaded. As a result the res_pjsip monitor thread kept going
- processing traffic and timers when it shouldn't.
-
- Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
-
-2016-04-15 11:41 +0000 [f4693d1897] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_channel.c: Ignore role setup failure in channel push.
-
- We have to setup the channel roles after the bridge class push is called
- because the bridge class push callback may have set roles on the incoming
- channel. Since we have already partially pushed the channel into the
- bridge and reversing what we have already done could be problematic, the
- only thing we can do is press on to complete pushing the channel into the
- bridge.
-
- * Ignore any channel role setup errors after pushing the channel into a
- bridge. The channel may behave incorrectly in the bridge but we can no
- longer abort the push at this time.
-
- Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00
-
-2016-04-17 15:37 +0000 [22335fe18a] Jaco Kroon <jaco@uls.co.za>
-
- * chan_sip: Don't verify table if rtupdate=no
-
- If rtupdate=no do not verify sipregs/peers table has updatable fields.
-
- ASTERISK-25934 #close
-
- Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d
-
-2016-04-18 04:53 +0000 [3b9d8b60b2] ibercom <ibercom123@gmail.com>
-
- * app_queue: Frequent segfaults in function can_ring_entry()
-
- ASTERISK-25888 #close
-
- Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117
-
-2016-04-15 16:51 +0000 [724acb6ce7] Richard Mudgett <rmudgett@digium.com>
-
- * stasis_bridge.c: Update stasis bridge push diagnostic messages.
-
- Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a
-
-2016-04-14 13:49 +0000 [5f78801859] Mark Michelson <mmichelson@digium.com>
-
- * transport management: Register thread with PJProject.
-
- The scheduler thread that kills idle TCP connections was not registering
- with PJProject properly and causing assertions if PJProject was built in
- debug mode.
-
- This change registers the thread with PJProject the first time that the
- scheduler callback executes.
-
- AST-2016-005
-
- Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283
-
-2016-03-08 12:12 +0000 [7fb3724a77] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_transport_management: Kill idle TCP connections.
-
- "Idle" here means that someone connects to us and does not send a SIP
- request. PJProject will not automatically time out such connections, so
- it's up to Asterisk to do it instead.
-
- When we receive an incoming TCP connection, we will start a timer
- (equivalent to transaction timer D) waiting to receive an incoming
- request. If we do not receive a request in that timeframe, then we will
- shut down the TCP connection.
-
- ASTERISK-25796 #close
- Reported by George Joseph
-
- AST-2016-005
-
- Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6
-
-2016-03-08 10:52 +0000 [707fd4dcd0] Mark Michelson <mmichelson@digium.com>
-
- * Rename res_pjsip_keepalive res_pjsip_transport_management
-
- ASTERISK-25796
- Reported by George Joseph
-
- AST-2016-005
-
- Change-Id: Id322a05f927392293570599730050bc677d99433
-
-2016-04-14 07:15 +0000 [0b4bb19e0b] Mark Michelson <mmichelson@digium.com>
-
- * AST-2016-004: Fix crash on REGISTER with long URI.
-
- Due to some ignored return values, Asterisk could crash if processing an
- incoming REGISTER whose contact URI was above a certain length.
-
- ASTERISK-25707 #close
- Reported by George Joseph
-
- Patches:
- 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
-
- AST-2016-004
-
- Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55
-
-2016-04-12 13:10 +0000 [f6e080c6a4] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_softmix.c: Fix crash if could not allocate the dsp.
-
- Fix off nominal crash where we could not setup the channel to process
- frames for the softmix bridge technology because of allocation failure.
-
- Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372
-
-2016-04-12 15:41 +0000 [cf15a2f2d3] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject: Add patch for removing strip of '[]' from header params
-
- From the patch submitted to Teluu on 4/12/2016
- <<<<<<<<<
- The wholesale stripping of '[]' from header parameters causes issues if
- something (like a port) occurs after the final ']'.
-
- '[2001:a::b]' will correctly parse to '2001:a::b'
- '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left
- with ':8080' and parsing stops with a syntax error.
-
- I can't even find a case where stripping the '[]' is a good thing anyway. Even
- if you continued to parse and resulted in a string that looks like this...
- '2001:a::b:8080', it's not valid.
-
- This came up in Asterisk because Kamailio sends us a Contact with an alias
- URI parameter that has an IPv6 address in it like this:
- Contact: <sip:1171@127.0.0.1:5080;alias=[2001:1:2::3]~43691~6>
- which should be legal but causes a syntax error because of the characters
- after the final ']'. Even if it didn't, the '[]' should still not be stripped.
-
- I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6
- enabled. No issues were caused by removing the code that strips the '[]'.
- >>>>>>>>>>>
-
- ASTERISK-25123 #close
- Reported-by: Anthony Messina
-
- Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a
-
-2016-04-12 09:10 +0000 [daa086fae4] Joshua Colp <jcolp@digium.com>
-
- * app_voicemail: Fix test_voicemail_notify_endl test.
-
- The test_voicemail_notify_endl test checks the end-of-line
- characters of an email message to confirm that they are consistent.
- The test wrongfully assumed that reading from the email message
- into a buffer will always result in more than 1 character being
- read. This is incorrect. If only 1 character was read the test
- would go outside of the buffer and access other memory causing
- a crash.
-
- The test now checks to ensure that 2 or more characters are read
- in ensuring the test stays within the buffer.
-
- ASTERISK-25874 #close
-
- Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710
-
-2016-04-07 12:02 +0000 [f896136460] Alexei Gradinari <alex2grad@gmail.com>
-
- * app_voicemail/IMAP: function 'save_to_folder' creates wrong folder
-
- If try to move message to Cust1 (number 5)
- the function 'save_to_folder' tries to create Greeting folder instead of Cust1.
-
- This patch fixed it by setting GREETINGS_FOLDER = -1
-
- ASTERISK-24927 #close
-
- Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51
-
-2016-04-07 16:18 +0000 [70b7673f09] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip: Add headers to AMI Event ContactStatusDetail
-
- * Added Useragent and RegExpire headers to AMI Event
- ContactStatusDetail with associated documentation.
-
- ASTERISK-25903 #close
-
- Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239
-
-2016-04-11 16:20 +0000 [64ecd41c8f] Alexei Gradinari <alex2grad@gmail.com>
-
- * Codecs: strip codec name while parsing allow/disallow options
-
- Failed registration using PJSIP/Realtime if one of the codec name
- in allow/disallow option is wrong or contains space.
-
- This patch strip codec name.
-
- ASTERISK-25914
-
- Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d
-
-2016-04-11 14:26 +0000 [3f6c4667b8] Jaco Kroon <jaco@uls.co.za>
-
- * core_unreal: Fix hangupcauses not getting set on Local channels
-
- ASTERISK-25912 #close
-
- Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa
-
-2016-04-01 13:30 +0000 [fe7e48db03] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip contact: Lock expiration/addition of contacts
-
- Contact expiration can occur in several places: res_pjsip_registrar,
- res_pjsip_registrar_expire, and automatically when anyone calls
- ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar
- may also be attempting to renew or add a contact. Since none of this was locked
- it was possible for one thread to be renewing a contact and another thread to
- expire it immediately because it was working off of stale data. This was the
- casue of intermittent registration/inbound/nominal/multiple_contacts test
- failures.
-
- Now, the new named lock functionality is used to lock the aor during contact
- expire and add operations and res_pjsip_registrar_expire now checks the
- expiration with the lock held before deleting the contact.
-
- ASTERISK-25885 #close
- Reported-by: Josh Colp
-
- Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059
-
-2016-04-10 14:16 +0000 [0c414eaf35] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject: Add patch to fix Via IPv6 parsing
-
- There's a bug in pjproject's sip_parser where the ":" wasn't correctly
- interpreted. This is causing IPv6 addresses in the "received" parameter of the
- Via header to cause a syntax check failure.
-
- This patch was submitted to Teluu on 4/10/2016.
-
- ASTERISK-25910 #close
- Reported-by: Anthony Messina
-
- Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922
-
-2016-03-31 20:04 +0000 [772ff3048f] gtjoseph <george.joseph@fairview5.com>
-
- * lock: Add named lock capability
-
- Locking some objects like sorcery objects can be tricky because the underlying
- ao2 object may not be the same for all callers. For instance, two threads that
- call ast_sorcery_retrieve_by_id on the same aor name might actually get 2
- different ao2 objects if the underlying wizard had to rehydrate the aor from a
- database. Locking one ao2 object doesn't have any effect on the other even if
- those objects had locks in the first place.
-
- Named locks allow access control by keyspace and key strings. Now an "aor"
- named "1000" can be locked and any other thread attempting to lock "aor" "1000"
- will wait regardless of whether the underlying ao2 object is the same or not.
- Mutex and rwlocks are supported.
-
- This capability will initially be used to lock an aor when multiple threads may
- be attempting to prune expired contacts from it.
-
- Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45
-
-2016-04-05 16:56 +0000 [fd601f26f7] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_outbound_publish: Add transport for outbound PUBLISH
-
- The first available transport of the appropriate type is used now.
- This patch adds new config option 'transport' for outbound-publish.
- If transport is set then outbound PUBLISH requests will use this transport.
-
- ASTERISK-25901 #close
-
- Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151
-
-2016-04-07 16:39 +0000 [5f768d2a9c] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event
-
- BLF pickup isn't working on Cisco SPA and Snom phones
- if the direction="recipient" attribute is missing in 'dialog' tag.
-
- This patch adds direction="recipient" if extension state is
- Ringing.
-
- ASTERISK-24601 #close
-
- Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c
-
-2016-04-07 10:59 +0000 [82638fb0c7] Richard Mudgett <rmudgett@digium.com>
-
- * pbx.c: Minor code rearangements.
-
- * Pull out a loop invariant.
-
- * Convert an else-if ladder to a switch statement.
-
- Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95
-
-2016-04-07 11:37 +0000 [bc320df173] Alexei Gradinari <alex2grad@gmail.com>
-
- * app_voicemail/IMAP: IMAP access FATAL error: Out of memory
-
- Sometimes uw-imap function 'mail_fetchbody' returns huge len
- which then pass to uw-imap function 'rfc822_base64'.
- uw-imap tries to allocate huge memory and abort() on fail.
-
- This patch check the len.
- If the len more than max size (128 Mbytes) log error.
- This patch also set variables len, newlen to avoid uninizialezed len.
- This patch also check pointer returned by rfc822_base64.
-
- ASTERISK-25899 #close
-
- Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca
-
-2016-04-07 12:26 +0000 [2ef8a954b3] Richard Mudgett <rmudgett@digium.com>
-
- * pbx: Update doxygen for extension state watchers.
-
- Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e
-
-2016-04-07 11:49 +0000 [d312fdeb1b] gtjoseph <george.joseph@fairview5.com>
-
- * alembic: Remove batch operations (and sqlite support)
-
- Because SQLite doesn't support full ALTER capabilities, alembic scripts
- require batch operations. However, that capability wasn't available until
- 0.7.0 which some distributions haven't reached yet. Therefore, the batch
- operations introduced in commit 86d6e44cc (review 2319) have been reverted
- and SQLite is unsupported again, for now anyway.
-
- Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql.
-
- ASTERISK-25890 #close
- Reported-by: Harley Peters
-
- Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80
-
-2016-04-07 11:05 +0000 [901e8d78c4] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_registrar_expire: Fix race condition at shutdown.
-
- When shutting down, the PJSIP sorcery is destroyed. The registrar
- expiration module queries the PJSIP sorcery to determine what
- to expire. As there was no synchronization between termination
- of the expiration thread and the unloading of the module it was
- possible for the thread to try to access the PJSIP sorcery after
- it had been destroyed.
-
- This change ensures that the thread is shut down before allowing
- the module to be considered unloaded.
-
- Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b
-
-2016-04-06 16:28 +0000 [8207372e66] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Fix configuration setting of "regcontext".
-
- Due to a merge problem two options were swapped causing the
- regcontext setting to not get set.
-
- Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1
-
-2016-04-06 08:01 +0000 [0735a4d6d7] Jacek Konieczny <jkonieczny@eggsoft.pl>
-
- * frame.c: Copy the whole subclass in ast_frdup().
-
- The problem is ast_frdup() does not copy whole frame.subclass for voice,
- video and image frames, only the format is copied. For video frames, the
- subclass structure contains the .frame_ending flag used to put the RTP
- marker where it needs to be.
-
- ASTERISK-25894 #close
-
- Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33
-
-2016-04-05 14:23 +0000 [c61dca6419] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Handle deferred SDP hold/unhold properly.
-
- Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
- other words, they provide no SDP in the reinvite.
-
- A typical transaction that starts hold might look something like this:
-
- * Device sends reinvite with no SDP
- * Asterisk sends 200 OK with SDP indicating sendrecv on streams.
- * Device sends ACK with SDP indicating sendonly on streams.
-
- At this point, PJMedia's SDP negotiator saves Asterisk's local state as
- being recvonly.
-
- Now, when the device attempts to unhold, it again uses a deferred SDP
- reinvite, so we end up doing the following:
-
- * Device sends reinvite with no SDP
- * Asterisk sends 200 OK with SDP indicating recvonly on streams
- * Device sends ACK with SDP indicating sendonly on streams
-
- The problem here is that Asterisk offered recvonly, and by RFC 3264's
- rules, if an offer is recvonly, the answer has to be sendonly. The
- result is that the device is not taken off hold.
-
- What is supposed to happen is that Asterisk should indicate sendrecv in
- the 200 OK that it sends. This way, the device has the freedom to
- indicate sendrecv if it wants the stream taken off hold, or it can
- continue to respond with sendonly if the purpose of the reinvite was
- something else (like a session timer refresher).
-
- The fix here is to alter the SDP negotiator's state when we receive a
- reinvite with no SDP. If the negotiator's state is currently in the
- recvonly or inactive state, then we alter our local state to be
- sendrecv. This way, we allow the device to indicate the stream state as
- desired.
-
- ASTERISK-25854 #close
- Reported by Robert McGilvray
-
- Change-Id: I7615737276165eef3a593038413d936247dcc6ed
-
-2016-03-27 23:33 +0000 [50b0922a22] gtjoseph <george.joseph@fairview5.com>
-
- * config: Allow filters when appending to a category
-
- In sorcery based config files where there are multiple categories with the same
- name, you can't use the (+) operator to reliably append to a category because
- config.c stops looking when it finds the first one with the same name.
-
- Example:
-
- [1000]
- type = endpoint
-
- [1000]
- type = aor
-
- [1000](+)
- authenticate_qualify = yes
-
- This config will fail because config.c appends authenticate_qualify to the
- first category it finds, the endpoint, and that's not valid for endpoint.
-
- Solution:
-
- The capability to find a category that contains a certain variable already
- exists so the only real change was to parse anything after the '+' that's not a
- comma, as a filter string.
-
- [1000]
- type = endpoint
-
- [1000]
- type = aor
-
- [1000](+type=aor)
- authenticate_qualify = yes
-
- This now works as expected.
-
- Although the following example doesn't make any sense for pjsip, you can even
- specify multiple filters:
-
- [1000](+type=aor&qualify_frequency=10)
-
- ASTERISK-25868 #close
- Reported-by: Nick Repin
-
- Change-Id: I10773da4c79db36fbf1993961992af63d3441580
-
-2016-04-05 10:21 +0000 [cb56ef8069] Joshua Colp <jcolp@digium.com>
-
- * res_http_websocket: Make core supported.
-
- Websockets are a core part of ARI support and as such this
- module should also be core supported.
-
- Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c
-
-2016-03-25 23:22 +0000 [f6f4cf459f] gtjoseph <george.joseph@fairview5.com>
-
- * stringfields: Refactor to allow fields to be added to the end of structures
-
- String fields are great, except that you can't add new ones without breaking
- ABI compatibility because it shifts down everything else in the structure.
- The only alternative is to add your own char * field to the end of the
- structure and manage the memory yourself which isn't ideal, especially since
- you then can't use the OPT_STRINGFIELD_T type.
-
- Background:
-
- The reason string fields had to be declared inside the
- AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared
- fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS
- declared the pool, then the fields, then the manager, you could use the offsets
- of the pool and manager and iterate over the sequential addresses in between to
- access the fields. The actual pool, field allocation and field set operations
- don't actually care where the field is. It's just iteration over the fields
- that was the problem.
-
- Solution: Extended String Fields
-
- An extended string field is one that is declared outside the
- AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent
- structure. Other than using AST_STRING_FIELD_EXTENDED instead of
- AST_STRING_FIELD, it looks the same as other string fields. It's storage comes
- from the pool and it participates in string field compare and copy operations
- peformed on the parent structure. It's also a valid target for the
- OPT_STRINGFIELD_T aco option type.
-
- Implementation:
-
- To keep track of the extended fields and make sure that ABI isn't broken, the
- existing embedded_pool pointer in the manager structure was repurposed to be a
- pointer to a separate header structure that contains the embedded_pool pointer
- plus a vector of fields. The length of the manager structure didn't change and
- the embedded_pool pointer isn't used in the macros, only the stringfields C
- code. A side benefit of this is that changing the header structure in the
- future won't break ABI.
-
- ast_string_fields_init initializes the normal string fields and appends them to
- the vector, and subsequent calls to ast_string_field_init_extended initialize
- and append the extended fields. Cleanup, ast_string_fields_cmp, and
- ast_string_fields_copy can now work on the vector instead of sequentially
- traversing the addresses between the pool and manager.
-
- The total size of a structure using string fields didn't change, whether using
- extended fields or not, nor have the offsets of any structure members, either
- inside the original block or outside. Adding an extended field to the end of a
- structure is the same as adding a char *.
-
- Details:
-
- The stringfield C code was pulled out from utils.c and into stringfields.c.
- It just made sense.
-
- Additional work was done in ast_string_field_init and
- ast_calloc_with_stringfields to handle the allocation of the new header
- structure and the vector, and the associated cleanup. In the process some
- additional NULL pointer checking was added.
-
- A lot of work was done in stringfields.h since the logic for compare and copy
- is there. Documentation was added as well as somne additional NULL checking.
-
- The ability to call ast_calloc_with_stringfields with a number of structures
- greater than 1 never really worked. Well, the calloc worked but there was no
- way to access the additional structures or clean them up. It was agreed that
- there was no use case for requesting more than 1 structure so an ast_assert
- was added to prevent it and the iteration code removed.
-
- Testing:
-
- The stringfield unit tests were updated to test both normal and extended
- fields. Tests for ast_string_field_ptr_set_by_fields and
- ast_calloc_with_stringfields were also added.
-
- As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except
- res_pjsip itself, saved off. The patch was then added and a full compile and
- install was performed. Then the older res_pjsip_* moduled were copied over the
- installed versions so res_pjsip was new and the rest were old. No issues.
-
- contact->aor, which is a char * at the end of contact, was then changed to an
- extended string field and a recompile and reinstall was performed, again
- leaving stock versions of the the res_pjsip_* modules. Again, no issues with
- the res_pjsip_* modules using the old stringfield implementation and with
- contact->aor as a char *, and res_pjsip itself using the new stringfield
- implementation and contact->aor being an extended string field.
-
- Finally, several existing string fields were converted to extended string
- fields to test OPT_STRINGFIELD_T. Again, no issues.
-
- Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61
-
-2016-04-04 18:02 +0000 [fe448ac8a7] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7
-
- I forgot the new voicemail_extension wasn't a stringfield and didn't check
- for NULL where I should have.
-
- Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb
-
-2016-04-03 11:47 +0000 [9d4318f798] gtjoseph <george.joseph@fairview5.com>
-
- * install_prereq: Fix check_installed_debs remove subversion
-
- check_installed_debs wasn't handling virtual packages like libsrtp-dev and
- libresample-dev and on multiarch systems it was accidentally filtering out all
- packages if any :i386 packages were found instead of just filtering out the
- :i386 packages themselves.
-
- Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda
-
-2016-04-01 13:09 +0000 [566601837e] gtjoseph <george.joseph@fairview5.com>
-
- * utils.c: Fix typo in handle_show_locks
-
- ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e).
-
- Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf
-
-2016-03-30 18:34 +0000 [964f54bd5d] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject_bundled: Fix use of LDCONFIG for shared library link creation
-
- LDCONFIG apparently isn't set to something sane on all systems so the creation
- of the shared library links fails. Instead of just testing for non-blank,
- main/Makefile now checks that LDCONFIG is actually executable and reverts to
- LN if it isn't.
-
- This applies to both libasteriskpj and libasteriskssl.
-
- Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG.
-
- ASTERISK-25873 #close
- Reported-by: Hans van Eijsden
-
- Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729
-
-2016-03-30 13:31 +0000 [5f73c2ef0a] Richard Mudgett <rmudgett@digium.com>
-
- * res_stasis.c: Protect channel datastore list from stasis end.
-
- Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95
-
-2016-03-29 18:06 +0000 [74d63f56ee] Richard Mudgett <rmudgett@digium.com>
-
- * res_ari: Cannot get control also means channel is unavailable.
-
- The only caller of ari_bridges_play_found() has this note:
-
- If ari_bridges_play_found fails because the channel is unavailable for
- playback, The channel will be removed from the playback list soon. We can
- keep trying to get channels from the list until we either get one that
- will work or else there isn't a channel for this bridge anymore, in which
- case we'll revert to ari_bridges_play_new.
-
- Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6
-
-2016-03-29 14:29 +0000 [cf49b44090] Richard Mudgett <rmudgett@digium.com>
-
- * res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name().
-
- Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7
-
-2016-03-28 14:23 +0000 [7f53f1d89e] Richard Mudgett <rmudgett@digium.com>
-
- * core_unreal.c: Add clarification comment about channel ref.
-
- Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3
-
-2016-03-29 13:47 +0000 [ecf4102d02] Richard Mudgett <rmudgett@digium.com>
-
- * res_stasis: Add control ref to playback and recording structs.
-
- The stasis_app_playback and stasis_app_recording structs need to have a
- struct stasis_app_control ref. Other threads can get a reference to the
- playback and recording structs from their respective global container.
- These other threads can then use the control pointer they contain after
- the control struct has gone.
-
- * Add control ref to stasis_app_playback and stasis_app_recording structs.
-
- With the refs added, the control command queue can now have a circular
- control reference which will cause the control struct to never get
- released if the control's command queue is not flushed when the channel
- leaves the Stasis application. Also the command queue needs better
- protection from adding commands if the control->is_done flag is set.
-
- * Flush the control command queue on exit.
-
- ASTERISK-25882 #close
-
- Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d
-
-2016-03-28 18:10 +0000 [a179aba65e] Richard Mudgett <rmudgett@digium.com>
-
- * res_stasis: Fix crash on a hanging up channel.
-
- * Give the struct stasis_app_control ao2 object a ref to the channel held
- in the object. Now the channel will still be around if a thread needs to
- post a stasis message instead of crash because the topic was destroyed.
-
- * Moved stopping any lingering silence generator out of the struct
- stasis_app_control destructor and made it a part of exiting the Stasis
- application. Who knows which thread the destructor will be called under
- so it cannot affect the channel's silence generator. Not only was the
- channel unprotected when the silence generator was stopped, stasis may no
- longer even control the channel.
-
- ASTERISK-25882
-
- Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4
-
-2016-03-30 12:38 +0000 [16c7d8e74a] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_mwi: Allow subscribe to vm access extension as an alias
-
- Background:
-
- If your extension is 1000 and the voicemail access extension is 1571 and you
- dial 1571, usually a dialplan rule calls voicemailmain with your extension and
- you are placed directly in your mailbox. Therefore most admins program the
- voicemail (or other speed dial) button on their phones to the access extension.
- Some phones (Snom at least) use whatever is programmed there to also subscribe
- for MWI and so can't dial one number and subscribe to another. This works fine
- in chan_sip because chan_sip completely ignores the user portion of the
- SUBSCRIBE message request URI. If it can match the peer, is subscribes to the
- peer's mailbox. The user could be set to anything or nothing and you'd still
- get subscribed to your mailbox.
-
- Issue:
-
- chan_pjsip actually uses the user portion of the URI to find an aor and its
- mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can
- create an aor for 1571 but you certainly can't add your entire voicemail
- system's mailboxes to it and everyone would get notified of every MWI.
-
- Solution:
-
- When an MWI subscribe comes in and an aor can't be found that matches the
- resource directly, check the resource against the endpoint's aors. If an aor
- is found that has a voicemail_extension that matches the resource, use it.
-
- ASTERISK-25865
- Reported-by: Ross Beer
-
- Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e
-
-2016-03-24 22:55 +0000 [d8f0bc3572] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited
-
- res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
- the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes
- on endpoints for unsolicited mwi and on aors for subscriptions required
- that the admin know in advance which the client wanted. If you specified
- mailboxes on the endpoint, subscriptions were rejected even if you also
- specified mailboxes on the aor.
-
- Voicemail extension:
- * Added a global default_voicemail_extension which defaults to "".
- * Added voicemail_extension to both endpoint and aor.
- * Added ast_sip_subscription_get_dialog for support.
- * Added ast_sip_subscription_get_sip_uri for support.
-
- When an unsolicited NOTIFY is constructed, the From header is parsed, the
- voicemail extension from the endpoint is substituted for the user, and the
- result placed in the Message-Account field in the body.
-
- When a subscribed NOTIFY is constructed, the subscription dialog local uri
- is parsed, the voicemail_extension from the aor (looked up from the
- subscription resource name) is substituted for the user, and the result
- placed in the Message-Account field in the body.
-
- If no voicemail extension was defined, the Message-Account field is not added
- to the NOTIFY body.
-
- mwi_subscribe_replaces_unsolicited:
- * Added mwi_subscribe_replaces_unsolicited to endpoint.
-
- The previous behavior was to reject a subscribe if a previous internal
- subscription for unsolicited MWI was found for the mailbox. That remains the
- default. However, if there are mailboxes also set on the aor and the client
- subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
- subscription is removed and replaced with the external subscription. This
- allows an admin to configure mailboxes on both the endpoint and aor and allows
- the client to select which to use.
-
- ASTERISK-25865 #close
- Reported-by: Ross Beer
-
- Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
-
-2016-03-30 09:46 +0000 [8dc8d6ceb8] gtjoseph <george.joseph@fairview5.com>
-
- * res_rtp_asterisk: Fix placement of txcount increment
-
- Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount
- for rtcp packets as well as rtp packets and that was causing sender reports
- to be generated instead of receiver reports in cases where no rtp was actually
- being sent.
-
- Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp,
- to rtp_sento which only handles rtp packets.
-
- Discovered by the hep/rtcp-receiver test.
-
- Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5
-
-2016-03-26 22:33 +0000 [c7eb18d865] gtjoseph <george.joseph@fairview5.com>
-
- * chan_pjsip: Add 'pjsip show channelstats'
-
- Added the ability to show channel statistics to chan_pjsip (cli_functions.c)
-
- Moved the existing 'pjsip show channel(s)' functionality from
- pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's
- private header so it made sense to move the existing channel commands as well.
-
- Now using stasis_cache_dump to get the channel snapshots rather than retrieving
- all endpoints, then getting each one's channel snapshots. Much more efficient.
-
- Change-Id: I03b114522126d27434030b285bf6d531ddd79869
-
-2016-03-10 19:52 +0000 [1583559a06] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS
-
- No one seemed to notice but every time an OPTIONS goes out, it goes
- out with a From of "asterisk" (or whatever the default from_user is set to),
- even if you specify an endpoint.
-
- The issue had several causes...
- qualify_contact is only called with an endpoint if called from the CLI.
- If the endpoint is NULL, qualify_contact only looks up the endpoint if
- authenticate_qualify=yes. Even then, it never passes it on to
- ast_sip_create_request where the From header is set. Therefore From
- is always "asterisk" (or whatever the default from_user is set to).
- Even if ast_sip_create_request were to get an endpoint, it only sets
- the From if endpoint->from_user is set.
-
- The fix is 4 parts...
-
- First, create_out_of_dialog_request was modified to use the endpoint id
- if endpoint was specified and from_user is not set.
-
- Second, qualify_contact was modified to always look up an endpoint if
- one wasn't specified regardless of authenticate_qualify. It then passes
- the endpoint on to create_out_of_dialog_request.
-
- Third (and most importantly), find_an_endpoint was modified to find
- an endpoint by using an "aors LIKE %contact->aor%" predicate with
- ast_sorcery_retrieve_by_fields. As such, this patch will only work
- if the sorcery realtime optimizations patch goes in. Otherwise we'd
- be pulling the entire endpoints database every time we send an OPTIONS.
- Since we already know the contact's aor, the on_endpoint callback was also
- modified to just check if the contact->aor is an exact match to one of
- the endpoint's.
-
- Finally, since we now have an endpoint for every OPTIONS request,
- res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was
- updated to get the transport from the endpoint and set it on tdata.
- Now the correct transport is used.
-
- Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af
-2016-03-25 10:59 +0000 [0cfab30b28] Jacek Konieczny <jkonieczny@eggsoft.pl>
-
- * res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS
-
- Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764
- explicitly states:
-
- There MUST be a separate DTLS-SRTP session for each distinct pair of
- source and destination ports used by a media session
-
- This means RTP keying material cannot be used for DTLS RTCP, which was
- the reason why RTCP encryption would fail.
-
- ASTERISK-25642
-
- Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a
-
-2016-03-25 10:42 +0000 [6a9c18fb59] Jacek Konieczny <jkonieczny@eggsoft.pl>
-
- * app_echo: forward and generate VIDUPDATE frames
-
- When using app_echo via WebRTC with VP8 video the video would appear
- only after a few minutes, because there would be nothing to request
- a full reference frame.
-
- This fixes the problem in both ways:
- - echos any VIDUPDATE frames received on the channel
- - sends one such frame when first video frame is to be forwarded
-
- This makes the echo work with Firefox and Chrome WebRTC implementation.
-
- ASTERISK-25867 #close
-
- Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e
-
-2016-03-27 12:53 +0000 [1bce690ccb] gtjoseph <george.joseph@fairview5.com>
-
- * res_rtp_asterisk: Fix packet stats on bridged connection
-
- rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated
- for bridged streams because the calulations were being done after the
- bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated.
-
- Moved the calculations so they occur for all valid received packets and
- all transmitted packets. Also added rxoctetcount and txoctetcount to
- ast_rtp_instance_stat.
-
- Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb
-
-2016-03-25 23:19 +0000 [50f90d4099] Philip Correia
-
- * res_parking: Fix blind transfer dynamic lots creation.
-
- Blind transfers to a recognized parking extension need to use the parker's
- channel variable values to create the dynamic parking lot. This is
- because there is always only one parker while the parkee may actually be a
- multi-party bridge. A multi-party bridge can never supply the needed
- channel variables to create the dynamic parking lot. In the multi-party
- bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and
- channel variables are inherited by the local channel used to park the
- bridge.
-
- * In park_common_setup(), make use the parker instead of the parkee to
- supply the dynamic parking lot channel variable values. In all but one
- case, the parkee is the same as the parker. However, in the recognized
- parking extension blind transfer scenario for a two party bridge they are
- different channels. For consistency, we need to use the parker channel.
-
- * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the
- local channel when blind transferring a multi-party bridge to a recognized
- parking extension.
-
- * When a local channel starts a call, the Local;2 side needs to inherit
- the CHANNEL(parkinglot) value from Local;1.
-
- The DTMF one-touch parking case wasn't even trying to create dynamic
- parking lots before it aborted the attempt.
-
- * In parking_park_call(), add missing code to create a dynamic parking
- lot.
-
- A DTMF bridge hook is documented as returning -1 to remove the hook.
- Though the hook caller is really coded to accept non-zero. See the
- ast_bridge_hook_callback typedef.
-
- * In feature_park_call(), don't remove the DTMF one-touch parking hook
- because of an error.
-
- ASTERISK-24605 #close
- Reported by: Philip Correia
- Patches:
- call_park.patch (license #6672) patch uploaded by Philip Correia
-
- Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9
-
-2016-03-08 15:55 +0000 [5aa5c49413] gtjoseph <george.joseph@fairview5.com>
-
- * sorcery/res_pjsip: Refactor for realtime performance
-
- There were a number of places in the res_pjsip stack that were getting
- all endpoints or all aors, and then filtering them locally.
-
- A good example is pjsip_options which, on startup, retrieves all
- endpoints, then the aors for those endpoints, then tests the aors to see
- if the qualify_frequency is > 0. One issue was that it never did
- anything with the endpoints other than retrieve the aors so we probably
- could have skipped a step and just retrieved all aors. But nevermind.
-
- This worked reasonably well with local config files but with a realtime
- backend and thousands of objects, this was a nightmare. The issue
- really boiled down to the fact that while realtime supports predicates
- that are passed to the database engine, the non-realtime sorcery
- backends didn't.
-
- They do now.
-
- The realtime engines have a scheme for doing simple comparisons. They
- take in an ast_variable (or list) for matching, and the name of each
- variable can contain an operator. For instance, a name of
- "qualify_frequency >" and a value of "0" would create a SQL predicate
- that looks like "where qualify_frequency > '0'". If there's no operator
- after the name, the engines add an '=' so a simple name of
- "qualify_frequency" and a value of "10" would return exact matches.
-
- The non-realtime backends decide whether to include an object in a
- result set by calling ast_sorcery_changeset_create on every object in
- the internal container. However, ast_sorcery_changeset_create only does
- exact string matches though so a name of "qualify_frequency >" and a
- value of "0" returns nothing because the literal "qualify_frequency >"
- doesn't match any name in the objset set.
-
- So, the real task was to create a generic string matcher that can take a
- left value, operator and a right value and perform the match. To that
- end, strings.c has a new ast_strings_match(left, operator, right)
- function. Left and right are the strings to operate on and the operator
- can be a string containing any of the following: = (or NULL or ""), !=,
- >, >=, <, <=, like or regex. If the operator is like or regex, the
- right string should be a %-pattern or a regex expression. If both left
- and right can be converted to float, then a numeric comparison is
- performed, otherwise a string comparison is performed.
-
- To use this new function on ast_variables, 2 new functions were added to
- config.c. One that compares 2 ast_variables, and one that compares 2
- ast_variable lists. The former is useful when you want to compare 2
- ast_variables that happen to be in a list but don't want to traverse the
- list. The latter will traverse the right list and return true if all
- the variables in it match the left list.
-
- Now, the backends' fields_cmp functions call ast_variable_lists_match
- instead of ast_sorcery_changeset_create and they can now process the
- same syntax as the realtime engines. The realtime backend just passes
- the variable list unaltered to the engine. The only gotcha is that
- there's no common realtime engine support for regex so that's been noted
- in the api docs for ast_sorcery_retrieve_by_fields.
-
- Only one more change to sorcery was done... A new config flag
- "allow_unqualified_fetch" was added to reg_sorcery_realtime.
- "no": ignore fetches if no predicate fields were supplied.
- "error": same as no but emit an error. (good for testing)
- "yes": allow (the default);
- "warn": allow but emit a warning. (good for testing)
-
- Now on to res_pjsip...
-
- pjsip_options was modified to retrieve aors with qualify_frequency > 0
- rather than all endpoints then all aors. Not only was this a big
- improvement in realtime retrieval but even for config files there's an
- improvement because we're not going through endpoints anymore.
-
- res_pjsip_mwi was modified to retieve only endpoints with something in
- the mailboxes field instead of all endpoints then testing mailboxes.
-
- res_pjsip_registrar_expire was completely refactored. It was retrieving
- all contacts then setting up scheduler entries to check for expiration.
- Now, it's a single thread (like keepalive) that periodically retrieves
- only contacts whose expiration time is < now and deletes them. A new
- contact_expiration_check_interval was added to global with a default of
- 30 seconds.
-
- Ross Beer reports that with this patch, his Asterisk startup time dropped
- from around an hour to under 30 seconds.
-
- There are still objects that can't be filtered at the database like
- identifies, transports, and registrations. These are not going to be
- anywhere near as numerous as endpoints, aors, auths, contacts however.
-
- Back to allow_unqualified_fetch. If this is set to yes and you have a
- very large number of objects in the database, the pjsip CLI commands
- will attempt to retrive ALL of them if not qualified with a LIKE.
- Worse, if you type "pjsip show endpoint <tab>" guess what's going to
- happen? :) Having a cache helps but all the objects will have to be
- retrieved at least once to fill the cache. Setting
- allow_unqualified_fetch=no prevents the mass retrieve and should be used
- on endpoints, auths, aors, and contacts. It should NOT be used for
- identifies, registrations and transports since these MUST be
- retrieved in bulk.
-
- Example sorcery.conf:
-
- [res_pjsip]
- endpoint=config,pjsip.conf,criteria=type=endpoint
- endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error
-
- ASTERISK-25826 #close
- Reported-by: Ross Beer
- Tested-by: Ross Beer
-
- Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67
-
-2016-03-23 14:24 +0000 [05fc3a96d1] Richard Mudgett <rmudgett@digium.com>
-
- * res_parking: Cleanup find_channel_parking_lot_name() usage.
-
- Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02
-
-2016-03-18 14:01 +0000 [a4189763ab] Richard Mudgett <rmudgett@digium.com>
-
- * res_parking: Misc fixes.
-
- res/parking/parking_applications.c:
-
- * Add malloc fail checks in setup_park_common_datastore().
-
- * Fix playing parking failed announcement to only happen on non-blind
- transfers in park_app_exec(). It could never go out before because a test
- was provedly always false.
-
- res/parking/parking_bridge.c:
-
- * Fix NULL tolerance in generate_parked_user() because
- bridge_parking_push() can theoretically pass a NULL parker channel if the
- parker channel went away for some reason.
-
- * Clarify some weird code dealing with blind_transfer in
- bridge_parking_push().
-
- res/parking/parking_bridge_features.c:
-
- * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel
- which will be bulk copied to the Local;2 channel on the subsequent
- ast_call(). The additional advantage is if the parker channel has the
- BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed
- to be overridden.
-
- res/parking/parking_manager.c:
-
- * Fix AMI Park action input range checking of the Timeout header in
- manager_park().
-
- * Reduced locking scope to where needed in manager_park().
-
- res/res_parking.c:
-
- * Fix some off nominal missing unlocks by eliminating the returns.
-
- Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca
-
-2014-12-15 05:23 +0000 [6f95b5eda1] Philip Correia
-
- * res_parking: Update parking documentation for dynamic parking lots.
-
- * Remove duplicate res_parking.conf courtesytone config option
- documentation.
-
- ASTERISK-24596 #close
- Reported by: Philip Correia
-
- ASTERISK-24605
- Reported by: Philip Correia
- Patches:
- call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia
-
- Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5
-
-2016-03-24 14:08 +0000 [81ce60f6d4] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.
-
- Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those
- codecs, which the caller did not request/support. That fix was not complete
- because on the second Session Timer all codecs were sent again. Some VoIP/SIP
- clients interpreted that complete codec-list as a change in the SIP session.
- Because of that, Asterisk did not send the RTP audio via NAT anymore which
- created a non-audio scenario after the second Session Timer fired.
-
- ASTERISK-24543 #close
-
- Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
-
-2016-03-19 07:34 +0000 [c6e4c48e67] Gianluca Merlo <gianluca.merlo@gmail.com>
-
- * config: fix flags in uint option handler
-
- The configuration unsigned integer option handler sets flags for the
- parser as if the option should be a signed integer (PARSE_INT32),
- leading to errors on "out of range" values. Fix flags (PARSE_UINT32).
-
- A fix to res_pjsip is also present which stops invalid flags from
- being passed when registering sorcery object fields for qualify
- status.
-
- ASTERISK-25612 #close
-
- Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e
-
-2016-03-10 16:58 +0000 [59c8e189fd] Mark Michelson <mmichelson@digium.com>
-
- * Restrict CLI/AMI commands on shutdown.
-
- During stress testing, we have frequently seen crashes occur because a
- CLI or AMI command attempts to access information that is in the process
- of being destroyed.
-
- When addressing how to fix this issue, we initially considered fixing
- individual crashes we observed. However, the changes required to fix
- those problems would introduce considerable overhead to the nominal
- case. This is not reasonable in order to prevent a crash from occurring
- while Asterisk is already shutting down.
-
- Instead, this change makes it so AMI and CLI commands cannot be executed
- if Asterisk is being shut down. For AMI, this is absolute. For CLI,
- though, certain commands can be registered so that they may be run
- during Asterisk shutdown.
-
- ASTERISK-25825 #close
-
- Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990
-2016-03-24 07:45 +0000 [ff3eebf454] Walter Doekes <walter+asterisk@wjd.nu>
-
- * musiconhold: Only warn if music class is not found in memory and database.
-
- The log message when a MusicOnHold music class was not found was changed
- from debug level to WARNING level in Asterisk 11.19 and 13.5. For those
- using realtime musiconhold, this message is wrong because it warns
- before checking the database.
-
- This changeset delays the warning until after the database has been
- checked.
-
- Reported-by: Conrad de Wet
- ASTERISK-25444 #close
-
- Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf
-
-2016-03-24 05:38 +0000 [82e55e4883] Walter Doekes <walter+asterisk@wjd.nu>
-
- * core/logging: Fix broken syslog levels on older glibc.
-
- The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However
- this macro is broken in older glibc (< 2.17); it would left-shift the
- facility a second time, causing the resultant priority to become
- invalid.
-
- The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this:
-
- The priority argument is formed by ORing the facility and the level
- values [...].
-
- ASTERISK-25510 #close
- Reported by: Michael Newton
-
- Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03
-
-2016-03-23 08:59 +0000 [d963a33749] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject-bundled: Cleanups for reported issues
-
- PortAudio should no longer be required
- PJSIP_MAX_PKT_LEN is now 6000
- Older autoconf issue fixed. (CentOS 6)
-
- Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd
-
-2015-11-20 08:02 +0000 [c5170677e7] Francesco Castellano <francesco.castellano@messagenet.it>
-
- * chan_sip.c: Space after port causes unnecessary resolution attempt
-
- check_via() already skips leading blanks where the sent-by address (with the
- optional port) should be placed.
-
- Since RFC 3261 allows for blanks between the port ant the Via parameters:
- > https://tools.ietf.org/html/rfc3261#section-20.42
- (actually it allows a lot of blanks more ;-)). I just switched from
- ast_skip_blanks() to ast_strip() on the local copy of the string.
-
- ASTERISK-21301 #close
-
- Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06
-
-2016-03-19 17:49 +0000 [51deadee38] gtjoseph <george.joseph@fairview5.com>
-
- * progdocs: Exclude ./third-party from documentation generation
-
- We don't need pjproject's documentation embedded in Asterisk's.
-
- Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17
-
-2016-03-18 20:32 +0000 [aa2fcd244e] Gianluca Merlo <gianluca.merlo@gmail.com>
-
- * func_aes: fix misuse of strlen on binary data
-
- The encryption code for AES_ENCRYPT evaluates the length of the data to
- be encoded in base64 using strlen. The data is binary, thus the length
- of it can be underestimated at the first NULL character.
- Reuse the write pointer offset to evaluate it, instead.
-
- ASTERISK-25857 #close
-
- Change-Id: If686b5d570473eb926693c73461177b35b13b186
-
-2016-04-20 10:46 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.8.2 Released.
-
-2016-04-20 05:45 +0000 [26d67ce885] Joshua Colp <jcolp@digium.com>
-
- * Release summaries: Remove previous versions
-
-2016-04-20 05:45 +0000 [d9909232ed] Joshua Colp <jcolp@digium.com>
-
- * .version: Update for 13.8.2
-
-2016-04-20 05:45 +0000 [fc57bb9b15] Joshua Colp <jcolp@digium.com>
-
- * .lastclean: Update for 13.8.2
-
-2016-04-20 05:45 +0000 [ac04474f38] Joshua Colp <jcolp@digium.com>
-
- * realtime: Add database scripts for 13.8.2
-
-2016-04-18 17:00 +0000 [91a3e1184f] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_registrar: Fix bad memory-ness with user_agent.
-
- Recent changes to the PJSIP registrar resulted in tests failing due to
- missing AOR_CONTACT_ADDED test events. The reason for this was that the
- user_agent string had junk values in it, resulting in being unable to
- generate the event.
-
- I'm going to be honest here, I have no idea why this was happening. Here
- are the steps needed for the user_agent variable to get messed up:
- * REGISTER is received
- * First contact in the REGISTER results in a contact being removed
- * Second contact in the REGISTER results in a contact being added
- * The contact, AOR, expiration, and user agent all have to be passed as
- format parameters to the creation of a string. Any subset of those
- parameters would not be enough to cause the problem.
-
- Looking into what was happening, the thing that struck me as odd was
- that the user_agent variable was meant to be set to the value of the
- User-Agent SIP header in the incoming REGISTER. However, when removing a
- contact, the user_agent variable would be set (via ast_strdupa inside a
- loop) to the stored contact's user_agent. This means that the
- user_agent's value would be incorrect when attempting to process further
- contacts in the incoming REGISTER.
-
- The fix here is to use a different variable for the stored user agent
- when removing a contact. Correcting the behavior to be correct also
- means the memory usage is less weird, and the issue no longer occurs.
-
- ASTERISK-25929 #close
- Reported by Joshua Colp
-
- Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
- (cherry picked from commit f436b9ab111f1ff57c6dd3970051f123b42c1103)
-
-2016-04-18 13:41 +0000 [70e25ced60] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_transport_management: Allow unload to occur.
-
- At shutdown it is possible for modules to be unloaded that wouldn't
- normally be unloaded. This allows the environment to be cleaned up.
-
- The res_pjsip_transport_management module did not have the unload
- logic in it to clean itself up causing the res_pjsip module to not
- get unloaded. As a result the res_pjsip monitor thread kept going
- processing traffic and timers when it shouldn't.
-
- Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
- (cherry picked from commit 49bfdc9ac029e0ef17cd8e85d8d7b7731387a34e)
-
-2016-04-18 12:12 +0000 [856931edc2] Mark Michelson <mmichelson@digium.com>
-
- * PJSIP: Remove PJSIP parsing functions from uri length validation.
-
- The PJSIP parsing functions provide a nice concise way to check the
- length of a hostname in a SIP URI. The problem is that in order to use
- those parsing functions, it's required to use them from a thread that
- has registered with PJLib.
-
- On startup, when parsing AOR configuration, the permanent URI handler
- may not be run from a PJLib-registered thread. Specifically, this could
- happen when Asterisk was started in daemon mode rather than
- console-mode. If PJProject were compiled with assertions enabled, then
- this would cause Asterisk to crash on startup.
-
- The solution presented here is to do our own parsing of the contact URI
- in order to ensure that the hostname in the URI is not too long. The
- parsing does not attempt to perform a full SIP URI parse/validation,
- since the hostname in the URI is what is important.
-
- ASTERISK-25928 #close
- Reported by Joshua Colp
-
- Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
- (cherry picked from commit efae187217750e322cd6167705a33f888b631427)
-
-2016-04-14 20:26 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.8.1 Released.
-
-2016-04-14 15:26 +0000 [18e6f12e83] Kevin Harwell <kharwell@digium.com>
-
- * Release summaries: Remove previous versions
-
-2016-04-14 15:26 +0000 [625c07711a] Kevin Harwell <kharwell@digium.com>
-
- * .version: Update for 13.8.1
-
-2016-04-14 15:26 +0000 [584f1fb3c7] Kevin Harwell <kharwell@digium.com>
-
- * .lastclean: Update for 13.8.1
-
-2016-04-14 15:26 +0000 [1e37a63379] Kevin Harwell <kharwell@digium.com>
-
- * realtime: Add database scripts for 13.8.1
-
-2016-04-14 13:49 +0000 [dcf1b3c098] Mark Michelson <mmichelson@digium.com>
-
- * transport management: Register thread with PJProject.
-
- The scheduler thread that kills idle TCP connections was not registering
- with PJProject properly and causing assertions if PJProject was built in
- debug mode.
-
- This change registers the thread with PJProject the first time that the
- scheduler callback executes.
-
- AST-2016-005
-
- Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283
-
-2016-03-08 12:12 +0000 [efafbb1319] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_transport_management: Kill idle TCP connections.
-
- "Idle" here means that someone connects to us and does not send a SIP
- request. PJProject will not automatically time out such connections, so
- it's up to Asterisk to do it instead.
-
- When we receive an incoming TCP connection, we will start a timer
- (equivalent to transaction timer D) waiting to receive an incoming
- request. If we do not receive a request in that timeframe, then we will
- shut down the TCP connection.
-
- ASTERISK-25796 #close
- Reported by George Joseph
-
- AST-2016-005
-
- Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6
-
-2016-03-08 10:52 +0000 [159f639770] Mark Michelson <mmichelson@digium.com>
-
- * Rename res_pjsip_keepalive res_pjsip_transport_management
-
- ASTERISK-25796
- Reported by George Joseph
-
- AST-2016-005
-
- Change-Id: Id322a05f927392293570599730050bc677d99433
-
-2016-04-14 07:15 +0000 [c164ff004d] Mark Michelson <mmichelson@digium.com>
-
- * AST-2016-004: Fix crash on REGISTER with long URI.
-
- Due to some ignored return values, Asterisk could crash if processing an
- incoming REGISTER whose contact URI was above a certain length.
-
- ASTERISK-25707 #close
- Reported by George Joseph
-
- Patches:
- 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
-
- AST-2016-004
-
- Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55
-
-2016-03-29 19:39 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.8.0 Released.
-
-2016-03-29 14:39 +0000 [0f885f0076] Mark Michelson <mmichelson@digium.com>
-
- * Release summaries: Add summaries for 13.8.0
-
-2016-03-29 14:34 +0000 [a1fa37aebd] Mark Michelson <mmichelson@digium.com>
-
- * Release summaries: Remove previous versions
-
-2016-03-29 14:34 +0000 [e7de5fd439] Mark Michelson <mmichelson@digium.com>
-
- * .version: Update for 13.8.0
-
-2016-03-29 14:34 +0000 [8baf813848] Mark Michelson <mmichelson@digium.com>
-
- * .lastclean: Update for 13.8.0
-
-2016-03-29 14:34 +0000 [42469df205] Mark Michelson <mmichelson@digium.com>
-
- * realtime: Add database scripts for 13.8.0
-
-2016-03-22 18:31 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.8.0-rc1 Released.
-
-2016-03-22 13:26 +0000 [a698424678] Mark Michelson <mmichelson@lunkwill>
-
- * Release summaries: Add summaries for 13.8.0-rc1
-
-2016-03-22 13:21 +0000 [e395a0b973] Mark Michelson <mmichelson@lunkwill>
-
- * .version: Update for 13.8.0-rc1
-
-2016-03-22 13:21 +0000 [38a86b2dbf] Mark Michelson <mmichelson@lunkwill>
-
- * .lastclean: Update for 13.8.0-rc1
-
-2016-03-22 13:21 +0000 [e0c8c8bf4a] Mark Michelson <mmichelson@lunkwill>
-
- * realtime: Add database scripts for 13.8.0-rc1
-
-2016-03-18 14:31 +0000 [6a40520fe9] Kevin Harwell <kharwell@digium.com>
-
- * chan_pjsip: ref leak when checking direct_media_glare
-
- Fix the reference leak introduced in the following commit:
-
- 9444ddadf8525d1ce66a1faf1db97f9f6c265ca4
-
- ASTERISK-25849
-
- Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85
-2016-03-16 12:37 +0000 [9444ddadf8] Kevin Harwell <kharwell@digium.com>
-
- * chan_pjsip: transfers with direct media reinvite has wrong address/port
-
- During a transfer involving direct media a race occurs between when the
- transferer channel is swapped out, initiating rtp changes/updates, and the
- subsequent reinvites.
-
- When Alice, after speaking with Charlie (Bob is on hold), connects Bob and
- Charlie invites are sent to each in order to establish the call between them.
- Bob is taken off hold and Charlie is told to have his media flow through
- Asterisk. However, if before those invites go out the bridge updates Bob's
- and/or Charlie's rtp information with direct media data (i.e. address, port)
- then the invite(s) will contain the remote data in the SDP instead of the
- Asterisk data.
-
- The race occurs in the native bridge glue code when updating the peer. The
- direct_media_address can get set twice before sending out the first invite
- during call connection. This can happen because the checking/setting of the
- direct_media_address happened in one thread while the sending of the invite(s)
- happened in another thread.
-
- This fix removes the race condition by moving the checking/setting of the
- direct_media_address to be in the same thread as the sending of the invites(s).
- This serializes the checking/setting and sending so they can no longer happen
- out of order.
-
- ASTERISK-25849 #close
-
- Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873
-
-2015-10-19 07:11 +0000 [88240f98d9] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>
-
- * install_prereq: Update repositories before install on Debian systems
-
- When to install packages the indexed local is more old of the
- version of software on the repository they have been upgraded by security
- update then get the package will give 404 not found.
-
- The patch prevent by update local index to repository for aptitude before
- install.
-
- ASTERISK-25495 #close
-
- Reporte by: Rodrigo Ramírez Norambuena
-
- Change-Id: I645959e553aac542805ced394cac2dca964051fa
- (cherry picked from commit 88f3dbaec9509bfba8bc1de7799aa0dc65304bb5)
-
-2015-06-03 20:12 +0000 [efcf9a96db] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * install_prereq: Check if is installed aptitude otherwise to install.
-
- If in Debian or system based, dont have aptitude installed the script do
- nothing. This patch checked if aptitude installed, if not installed.
-
- Also, if execute script with all packages installed yet, the script not show
- nothing and return exit 1 because the command 'grep' get nothing from pipe from
- 'awk'.
-
- ASTERISK-25113 #close
- Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f
- (cherry picked from commit 6737ded0581a9e1256bdfe30c1d747e7ca93f8b3)
-
-2016-03-03 04:43 +0000 [2b1b8e382a] Sergio Medina Toledo <lumasepa@gmail.com>
-
- * res_pjsip_refer.c: Fix seg fault in process of Refer-to header.
-
- The "Refer-to" header of an incoming REFER request is parsed by
- pjsip_parse_uri(). That function requires the URI parameter to be NULL
- terminated. Unfortunately, the previous code added the NULL terminator by
- overwriting memory that may not be safe. The overwritten memory results
- could be benign, memory corruption, or a segmentation fault. Now the URI
- is NULL terminated safely by copying the URI to a new chunk of memory with
- the correct size to be NULL terminated.
-
- ASTERISK-25814 #close
-
- Change-Id: I32565496684a5a49c3278fce06474b8c94b37342
-
-2016-03-11 12:22 +0000 [de04308ae4] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix mwi resub deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023 #close
-
- Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6
-
-2016-03-10 17:01 +0000 [5f6627a8a4] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix registration timeout and expire deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023
-
- Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508
-
-2016-03-10 12:17 +0000 [32bd7a64f9] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix t38id deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023
-
- Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f
-
-2016-03-09 16:34 +0000 [43556b800b] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix reinviteid deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023
-
- Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1
-
-2016-03-09 16:32 +0000 [38c1cdab2c] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix packet retransid deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- * Fix retrans_pkt() to call check_pendings() with both the owner channel
- and the private objects locked as required.
-
- * Refactor dialog retransmission packet list to safely remove packet
- nodes. The list nodes are now ao2 objects. The list has a ref and the
- scheduled entry has a ref.
-
- ASTERISK-25023
-
- Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641
-
-2016-03-09 16:26 +0000 [e4ad55c888] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix waitid deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- * Made always run check_pendings() under the scheduler thread so scheduler
- ids can be checked safely.
-
- ASTERISK-25023
-
- Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52
-
-2016-03-08 15:08 +0000 [98d5669c28] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix session timers deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023
-
- Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900
-
-2016-03-07 13:21 +0000 [9cb8f73226] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix autokillid deadlock potential.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- * Fix clearing autokillid in __sip_autodestruct() even though we could
- reschedule.
-
- ASTERISK-25023
-
- Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f
-
-2016-03-07 18:28 +0000 [c5c7f48a15] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix provisional_keepalive_sched_id deadlock.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Stopping a scheduled event can result in a deadlock if the scheduled event
- is running when you try to stop the event. If you hold a lock needed by
- the scheduled event while trying to stop the scheduled event then a
- deadlock can happen. The general strategy for resolving the deadlock
- potential is to push the actual starting and stopping of the scheduled
- events off onto the scheduler/do_monitor() thread by scheduling an
- immediate one shot scheduled event. Some restructuring may be needed
- because the code may assume that the start/stop of the scheduled events is
- immediate.
-
- ASTERISK-25023
-
- Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48
-
-2016-03-09 11:22 +0000 [f959d84dfd] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- * Make dialog_unlink_all() unschedule all items at once in the sched
- thread.
-
- ASTERISK-25023
-
- Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4
-
-2016-03-10 21:54 +0000 [5f3225ddcc] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Clear scheduled immediate events on unload.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- The reordering of chan_sip's shutdown is to handle any immediate events
- that get put onto the scheduler so resources aren't leaked. The typical
- immediate events at this time are going to be concerned with stopping
- other scheduled events.
-
- ASTERISK-25023
-
- Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20
-
-2016-03-15 14:51 +0000 [7a74971771] Richard Mudgett <rmudgett@digium.com>
-
- * sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- Delaying destruction of the chan_sip sip_pvt structures caused the
- /channels/chan_sip/test_sip_rtpqos unit test to crash. That test
- registers a special test ast_rtp_engine with the rtp engine module. When
- the unit test completes it cleans up by unregistering the test
- ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt
- happens after the unit test returns, the destructor tries to call the rtp
- engine destroy callback of the test ast_rtp_engine auto variable which no
- longer exists on the stack.
-
- * Change the test ast_rtp_engine auto variable to a static variable. Now
- the variable can still exist after the unit test exits so the delayed
- sip_pvt destruction can complete successfully.
-
- ASTERISK-25023
-
- Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13
-
-2016-03-15 13:31 +0000 [d2c09ed73b] Andrew Nagy <andrew.nagy@the159.com>
-
- * app_stasis: Don't hang up if app is not registered
-
- This prevents pbx_core from hanging up the channel if the app isn't
- registered.
-
- ASTERISK-25846 #close
-
- Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce
-2016-03-07 15:50 +0000 [b2d2906445] Richard Mudgett <rmudgett@digium.com>
-
- * sched.c: Ensure oldest expiring entry runs first.
-
- This patch is part of a series to resolve deadlocks in chan_sip.c.
-
- * Updated sched unit test to check new behavior.
-
- ASTERISK-25023
-
- Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3
-
-2016-03-04 18:25 +0000 [9ae21b510f] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full().
-
- Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12
-
-2016-03-07 18:56 +0000 [56bcb97a3c] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Simplify sip_pvt destructor call levels.
-
- Remove destructor calling destroy_it calling really_destroy_it
- for no benefit. Just make the destructor the really_destroy_it
- function.
-
- Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a
-
-2016-03-14 08:59 +0000 [677a65fcbb] Joshua Colp <jcolp@digium.com>
-
- * build: Add configure check for proto field of PJSIP TLS transport setting.
-
- Older versions of PJSIP do not have the proto field on the TLS transport
- setting structure. This change adds a configure check so even if it is
- not present we will still be able to build.
-
- Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9
-
-2016-03-12 16:02 +0000 [32f0a3d52a] gtjoseph <george.joseph@fairview5.com>
-
- * build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE
-
- I can't ever recall actually needing the intermediate files or the checking
- that a double compile produces. What I CAN remember is every DONT_OPTIMIZE
- build needing 3 invocations of gcc instead of 1 just to do the checks and
- produce those intermediate files.
-
- Having said that, Richard pointed out that the reason for the double compile
- was that there were cases in the past where a submitted patch failed to compile
- because the submitter never tried it with the optimizations turned on.
-
- To get the best of both worlds, COMPILE_DOUBLE has been split into its own
- option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected
- BUT you can then turn it off if all you need are the debugging symbols. This
- way you have to make an informed decision about disabling COMPILE_DOUBLE.
-
- To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature
- was added to menuselect. The <use> element can now contain an "autoselect"
- attribute which will turn the used member on but not create a hard dependency.
- The cflags.xml implementation for COMPILE_DOUBLE looks like this...
-
- <member name="DONT_OPTIMIZE" displayname="Disable Optimizations ...">
- <use autoselect="yes">COMPILE_DOUBLE</use>
- <support_level>core</support_level>
- </member>
- <member name="COMPILE_DOUBLE" displayname="Pre-compile with ...>
- <depend>DONT_OPTIMIZE</depend>
- <support_level>core</support_level>
- </member>
-
- When DONT_OPTIMIZE is turned on, COMPILE_DOUBLE is turned on because
- of the use.
- When DONT_OPTIMIZE is turned off, COMPILE_DOUBLE is turned off because
- of the depend.
- When COMPILE_DOUBLE is turned on, DONT_OPTIMIZE is turned on because
- of the depend.
- When COMPILE_DOUBLE is turned off, DONT_OPTIMIZE is left as is because
- it only uses COMPILE_DOUBLE, it doesn't depend on it.
-
- I also made a few tweaks to the ncurses implementation to move things
- left a bit to allow longer descriptions.
-
- Change-Id: Id49ca930ac4b5ec4fc2d8141979ad888da7b1611
-
-2016-03-10 13:09 +0000 [38499e7125] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject: Pass (dont_)optimize flags to pjproject and fix pjsua
-
- The pjproject Makefile now uses the Asterisk optimization flags which
- are determined by the setting of the DONT_OPTMIZE menuselect flag.
- The Makefile was also restructured so a change to the top level
- menuselect.makeopts will result in a rebuild of pjproject.
-
- Also, "--disable-resample" was removed from the pjproject configure
- options. Without resample, pjsua (which is used by the testsuite)
- can't make audio calls. When it can't, it segfaults.
-
- Change-Id: I24b0a4d0872acef00ed89b3c527a713ee4c2ccd4
-
-2016-03-11 16:03 +0000 [336cae73cc] Walter Doekes <walter+asterisk@wjd.nu>
-
- * app_chanspy: Fix occasional deadlock with ChanSpy and Local channels.
-
- Channel masquerading had a conflict with autochannel locking.
-
- When locking autochannel->channel, the channel is fetched from the
- autochannel and then locked. During the fetch, the autochannel -- which
- has no locks itself -- can be modified by someone who owns the channel
- lock. That means that the value of autochan->channel cannot be trusted
- until you hold the lock.
-
- In practice, this caused problems with Local channels getting
- masqueraded away while the ChanSpy attempted to get info from that
- channel. The old channel which was about to get removed got locked, but
- the new (replaced) channel got unlocked (no-op). Because the replaced
- channel was now locked (and would never get unlocked), it couldn't get
- removed from the channel list in a timely manner, and would now cause
- deadlocks when iterating over the channel list.
-
- This change checks the autochannel after locking the channel for changes
- to the autochannel. If the channel had been changed, the lock is
- reobtained on the new channel.
-
- In theory it seems possible that after this fix, the lock attempt on the
- old (wrong) channel can be on an already destroyed lock, maybe causing
- a crash. But that hasn't been observed in the wild and is harder induce
- than the current deadlock.
-
- Thanks go to Filip Frank for suggesting a fix similar to this and
- especially to IRC user hexanol for pointing out why this deadlock was
- possible and testing this fix. And to Richard for catching my rookie
- while loop mistake ;)
-
- ASTERISK-25321 #close
-
- Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def
-
-2016-03-07 21:34 +0000 [875d5e9872] gtjoseph <george.joseph@fairview5.com>
-
- * pjproject_bundled: Remove --with-external-pa from configure options.
-
- Not sure why it was there in the first place as we already specify
- --disable-sound.
-
- Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9
-
-2016-03-06 14:38 +0000 [530cff5f5f] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Strip spaces from items parsed from comma-separated lists
-
- Configurations like "aors = a, b, c" were either ignoring everything after "a"
- or trying to look up " b". Same for mailboxes, ciphers, contacts and a few
- others.
-
- To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To
- facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were
- updated to handle null pointers.
-
- In some cases, an ast_strlen_zero() test was added to skip consecutive commas.
-
- There was also an attempt to ast_free an ast_strdupa'd string in
- ast_sip_for_each_aor which was causing a SEGV. I removed it.
-
- Although this issue was reported for realtime, the issue was in the res_pjsip
- modules so all config mechanisms were affected.
-
- ASTERISK-25829 #close
- Reported-by: Mateusz Kowalski
-
- Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2
-
-2016-03-04 20:37 +0000 [3c8076a83b] gtjoseph <george.joseph@fairview5.com>
-
- * install_prereq: Add packages for bundled pjproject
-
- RedHat/CentOS needs python-devel
- Debian/Ubuntu needs automake, libsrtp-dev and python-dev
-
- Ubuntu also needed libncurses5-dev for cmenuselect so while not
- needed for pjproject, I adedd it anyway.
-
- Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089
-
-2016-02-24 17:25 +0000 [27f32cd0a6] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited
-
- Per RFC3325, the 'From' header is now anonymized on outgoing calls when
- caller id presentation is prohibited.
-
- TID = trust_id_outbound
- PRO = Set(CALLERID(pres)=prohib)
- USR = endpoint/from_user
- DOM = endpoint/from_domain
- PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)
-
- Conditions |Result
- --------------------|----------------------------------------------------
- TID PRO USR DOM |PAI FROM
- --------------------|----------------------------------------------------
- Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi>
- Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid>
- Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi>
- Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid>
-
- Y N abc def.ghi |YES <sip:abc@def.ghi>
- Y N abc |YES <sip:abc@<ip_address>>
- Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
- Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
-
- N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi>
- N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid>
- N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi>
- N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid>
-
- N N abc def.ghi |YES <sip:abc@def.ghi>
- N N abc |YES <sip:abc@<ip_address>>
- N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
- N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
-
- ASTERISK-25791 #close
- Reported-by: Anthony Messina
-
- Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9
-
-2016-03-03 17:34 +0000 [7cf7b0a4f9] gtjoseph <george.joseph@fairview5.com>
-
- * third_party/Makefile.rules: Replace unsupported != operator with $(shell ...)
-
- Apparently the != operator is fairly new so I've replaced it with
- the old $(shell ...) syntax.
-
- Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479
- Reported-by: Richard Mudgett
-2016-01-23 15:50 +0000 [53f57001f2] gtjoseph <george.joseph@fairview5.com>
-
- * loader: Retry dlopen when loading fails
-
- Although we use the RTLD_LAZY flag when calling dlopen
- the first time on a module, this only defers resolution
- for function calls. Pointer references to functions are
- determined at link time so dlopen expects them to be there.
- Since we don't cross-module link, pointers to functions
- in other modules won't be available and dlopen will fail.
-
- Doing a "hardened" build also causes problems because it
- typically sets "-z now" on the ld command line which
- overrides RTLD_LAZY at run time.
-
- If the failing module isn't a GLOBAL_SYMBOLS module, then
- dlopen will be called again after all the GLOBAL_SYMBOLS
- modules have been loaded and they'll eventually resolve.
-
- If the calling module IS a GLOBAL_SYMBOLS module itself
- and a third module depends on it, then there's an issue
- because the second time through the dlopen loop,
- GLOBAL_SYMBOLS modules aren't given any special treatment
- and since the order in which dlopen is called isn't
- deterministic, the dependent may again be tried before the
- module it needs is loaded.
-
- Simple solution: Save modules that fail load_resource
- because of a dlopen error in a list and retry them
- immediately after the first pass. Keep retrying until
- the failed list is empty or we reach a #defined max
- retries. Error messages are suppressed until the final
- pass which also gets rid of those confusing error messages
- about module failures that are later corrected.
-
- Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb
-
-2016-03-01 16:18 +0000 [40d9e9e238] Kevin Harwell <kharwell@digium.com>
-
- * bridge.c: Crash during attended transfer when missing a local channel half
-
- It's possible for the transferer channel to get hung up early during the
- attended transfer process. For instance, a phone may send a "bye" immediately
- upon receiving a sip notify that contains a sip frag 100 (I'm looking at you
- Jitsi). When this occurs a race begins between the transferer being hung up
- and completion of the transfer code.
-
- If the channel hangs up too early during a transfer involving stasis bridging
- for instance, then when the created local channel goes to look up its swap
- channel (and associated datastore) it can't find it (since it is no longer in
- the bridge) thus it fails to enter the stasis application. Consequently, the
- created local channel(s) hang up as well. If the timing is just right then the
- bridging code attempts to add the message link with missing local channel(s).
- Hence the crash.
-
- Unfortunately, there is no great way to solve the problem of the unexpected
- "bye". While we can't guarantee we won't receive an early hangup, and in this
- case still fail to enter the stasis application, we can make it so asterisk
- does not crash.
-
- This patch does just that by locking the local channel structure, checking
- that the local channel's peer has not been lost, and then continuing. This
- keeps the local channel's peer from being ripped out from underneath it by
- the local/unreal hangup code while attempting to set the stasis message link.
-
- ASTERISK-25771
-
- Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
-
-2016-03-01 18:08 +0000 [ff3da61c35] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100
-
- During the transfer process, some phones (okay it was the Jitsi softphone,
- but maybe others are out there) send a "bye" immediately after receiving a
- SIP Notify. When a "bye" is received early for some types of transfers the
- transferer channel may no longer be available during late stage transfer
- processing.
-
- For instance, during an attended transfer involving stasis bridging at one
- point the created local channel looks for an associated swap channel in
- order to retrieve the stasis application name. If the transferer has hung
- up then the local channel will fail to find it. The local channel then has
- no way to know which stasis app to enter, so it fails and hangs up as well.
- Thus the transfer does not complete as expected.
-
- This patch delays the sending of the initial notify in order to give the
- transfer process enough time to gather the necessary data for a successful
- transfer.
-
- ASTERISK-25771
-
- Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16
-
-2016-03-03 08:26 +0000 [26b8f2692e] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_dtmf_info: NULL terminate the message body.
-
- PJSIP does not ensure that when printing the message body the
- buffer will be NULL terminated. This is problematic when searching
- for the signal and duration values of the DTMF.
-
- This change ensures the buffer is always NULL terminated.
-
- Change-Id: I52653a1a60c93092d06af31a27408d569cc98968
-
-2016-03-01 20:03 +0000 [86d6e44cc1] gtjoseph <george.joseph@fairview5.com>
-
- * alembic: Fix downgrade and tweak for sqlite
-
- Downgrade had a few issues. First there was an errant 'update' statement in
- add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we
- weren't cleaning up the ENUMs so subsequent upgrades on postgres failed
- because the types already existed.
-
- For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly.
- Fortunately alembic batch_operations takes care of this for us if we
- use it so the alter and drops were converted to use batch operations.
-
- Here's an example downgrade:
-
- with op.batch_alter_table('ps_endpoints') as batch_op:
- batch_op.drop_column('tos_audio')
- batch_op.drop_column('tos_video')
- batch_op.add_column(sa.Column('tos_audio', yesno_values))
- batch_op.add_column(sa.Column('tos_video', yesno_values))
- batch_op.drop_column('cos_audio')
- batch_op.drop_column('cos_video')
- batch_op.add_column(sa.Column('cos_audio', yesno_values))
- batch_op.add_column(sa.Column('cos_video', yesno_values))
-
- with op.batch_alter_table('ps_transports') as batch_op:
- batch_op.drop_column('tos')
- batch_op.add_column(sa.Column('tos', yesno_values))
- # Can't cast integers to YESNO_VALUES, so dropping and adding is required
- batch_op.drop_column('cos')
- batch_op.add_column(sa.Column('cos', yesno_values))
-
- Upgrades from base to head and downgrades from head to base were tested
- repeatedly for postgresql, mysql/mariadb, and sqlite3.
-
- Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8
-
-2016-03-02 15:55 +0000 [6f0d7ce9db] gtjoseph <george.joseph@fairview5.com>
-
- * config_transport: Fix objects returned by ast_sip_get_transport_states
-
- ast_sip_get_transport_states was returning a container of internal_state
- objects instead of ast_sip_transport_state objects. This was causing
- transport lookups to fail, most noticably in res_pjsip_nat, which
- couldn't find the correct external addresses. This was causing contacts
- to go out with internal ip addresses.
-
- ASTERISK-25830 #close
- Reported-by: Sean Bright
-
- Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e
-
-2016-03-02 11:17 +0000 [1ea7a5a774] Scott Griepentrog <scott@griepentrog.com>
-
- * CHAOS: cleanup possible null vars on msg alloc failure
-
- In message.c, if msg_alloc fails to init the string field,
- vars may be null, so use a null tolerant cleanup.
-
- In res_pjsip_messaging.c, if msg_data_create fails, mdata
- will be null, so use a null tolerant cleanup.
-
- ASTERISK-25323
-
- Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56
-
-2016-03-02 09:34 +0000 [3c37c7071f] Scott Griepentrog <scott@griepentrog.com>
-
- * CHAOS: prevent crash on failed strdup
-
- This patch avoids crashing on a null pointer
- if the strdup() allocation fails.
-
- ASTERISK-25323
-
- Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5
-
-2016-02-29 18:11 +0000 [9633be9d25] Richard Mudgett <rmudgett@digium.com>
-
- * func_callerid.c: Update REDIRECTING reason documentation.
-
- Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386
-
-2016-02-26 18:57 +0000 [4165ea7778] Richard Mudgett <rmudgett@digium.com>
-
- * SIP diversion: Fix REDIRECTING(reason) value inconsistencies.
-
- Previous chan_sip behavior:
-
- Before this patch chan_sip would always strip any quotes from an incoming
- reason and pass that value up as the REDIRECTING(reason). For an outgoing
- reason value, chan_sip would check the value against known values and
- quote any it didn't recognize. Incoming 480 response message reason text
- was just assigned to the REDIRECTING(reason).
-
- Previous chan_pjsip behavior:
-
- Before this patch chan_pjsip would always pass the incoming reason value
- up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip
- would send the reason value as passed down.
-
- With this patch:
-
- Both channel drivers match incoming reason values with values documented
- by REDIRECTING(reason) and values documented by RFC5806 regardless of
- whether they are quoted or not. RFC5806 values are mapped to the
- equivalent REDIRECTING(reason) documented value and is set in
- REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
- quoted string version ('"unconditional"') is converted to
- REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
- with 'cfu' instead of any of the aliases.
-
- The incoming 480 response reason text supported by chan_sip checks for
- known reason values and if not matched then puts quotes around the reason
- string and assigns that to REDIRECTING(reason).
-
- Both channel drivers send outgoing known REDIRECTING(reason) values as the
- unquoted RFC5806 equivalent. User custom values are either sent as is or
- with added quotes if SIP doesn't allow a character within the value as
- part of a RFC3261 Section 25.1 token. Note that there are still
- limitations on what characters can be put in a custom user value. e.g.,
- embedding quotes in the middle of the reason string is silly and just
- going to cause you grief.
-
- * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
- e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
- 'cfu' value.
-
- * Added missing malloc() NULL return check in res_pjsip_diversion.c
- set_redirecting_reason().
-
- * Fixed potential read from a stale pointer in res_pjsip_diversion.c
- add_diversion_header(). The reason string needed to be copied into the
- tdata memory pool to ensure that the string would always be available.
- Otherwise, if the reason string returned by reason_code_to_str() was a
- user's reason string then the string could be freed later by another
- thread.
-
- Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87
-
-2016-02-26 18:54 +0000 [41f4af4ce5] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason.
-
- Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd
-
-2016-02-29 20:41 +0000 [4c5998ff55] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref.
-
- * Fix double unref of other_party channel in off nominal path.
-
- * This is unlikely to be a real problem. However, for safety,
- in handle_incoming_request() keep the datastore ref with the
- other_party channel ref until we are finished with the other_party
- channel.
-
- Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821
-
-2016-01-18 21:54 +0000 [b59956a875] gtjoseph <george.joseph@fairview5.com>
-
- * build-system: Allow building with static pjproject
-
- Background here:
- http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html
-
- From CHANGES:
- * To help insure that Asterisk is compiled and run with the same known
- version of pjproject, a new option (--with-pjproject-bundled) has been
- added to ./configure. When specified, the version of pjproject specified
- in third-party/versions.mak will be downloaded and configured. When you
- make Asterisk, the build process will also automatically build pjproject
- and Asterisk will be statically linked to it. Once a particular version
- of pjproject is configured and built, it won't be configured or built
- again unless you run a 'make distclean'.
-
- To facilitate testing, when 'make install' is run, the pjsua and pjsystest
- utilities and the pjproject python bindings will be installed in
- ASTDATADIR/third-party/pjproject.
-
- The default behavior remains building with the shared pjproject
- installation, if any.
-
- Building:
-
- All you have to do is include the --with-pjproject-bundled option on
- the ./configure command line (and remove any existing --with-pjproject
- option if specified). Everything else is automatic.
-
- Behind the scenes:
-
- The top-level Makefile was modified to include 'third-party' in the
- list of MOD_SUBDIRS.
-
- The third-party directory was created to contain any third party
- packages that may be needed in the future. Its Makefile automatically
- iterates over any subdirectories passing on targets.
-
- The third-party/pjproject directory was created to house the pjproject
- source distribution. Its Makefile contains targets to download, patch
- configure, generate dependencies, compile libs, apps and python bindings,
- sanitized build.mak and generate a symbols list.
-
- When bootstrap.sh is run, it automatically includes the configure.m4
- file in third-party/pjproject. This file has a macro to download and
- conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR
- and PJPROJECT_BUNDLED. It also tests for the capabilities like
- PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to
- trying to compile. Of course, bootstrap.sh is only run once and the
- configure file is incldued in the patch.
-
- When configure is run with the new options, the macro in configure.m4
- triggers the download, patch, conifgure and tests. No compilation is
- performed at this time. The downloaded tarball is cached in /tmp so
- it doesn't get downloaded again on a distclean.
-
- When make is run in the top-level Asterisk source directory, it will
- automatically descend all the subdirectories in third_party just as it
- does for addons, apps, etc. The top-level Makefile makes sure that
- the 'third-party' is built before 'main' so that dependencies from the
- other directories are built first.
-
- When main does build, a new shared library (libasteriskpj) is created that
- links statically to the pjproject .a files and exports all their symbols.
- The asterisk binary links to that, just as it does with libasteriskssl.
-
- When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject
- python bindings are installed in ASTDATADIR/third-party/pjproject. This
- will facilitate testing, including running the testsuite which will be
- updated to check that directory for the pjsua module ahead of the system
- python library.
-
- Modules should continue to depend on pjproject if they use pjproject APIs
- directly. They should not care about the implementation. No changes to any
- res_pjsip modules were made.
-
- Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103
-
-2016-02-22 16:59 +0000 [18a323e542] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Fix T.38 issues caused by leaving a bridge.
-
- chan_sip could not handle AST_T38_TERMINATED frames being sent to it when
- the channel left the bridge. The action resulted in overlapping outgoing
- reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not
- happy.
-
- * Force T.38 to be remembered as locally bridged. Now when the channel
- leaves the native RTP bridge after T.38, the channel remembers that it has
- already reINVITEed the media back to Asterisk. It just needs to terminate
- T.38 when the AST_T38_TERMINATED arrives.
-
- * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant
- AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if
- they happen before the T.38 state changes to disabled. Now the T.38 state
- is set to disabled before the reINVITE is sent.
-
- ASTERISK-25582 #close
-
- Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce
-
-2016-02-18 18:27 +0000 [263a39f2cc] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_t38.c: Back out part of an earlier fix attempt.
-
- This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d
- commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame
- that it puts into the bridge may or may not be processed by the time the
- bridged peer is kicked out of the bridge. If it is processed then all is
- well. However, if it is not processed then that channel is stuck in fax
- mode until it hangs up or maybe if it joins another bridge for T.38
- faxing.
-
- ASTERISK-25582
-
- Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7
-
-2016-02-22 13:54 +0000 [221422be50] Richard Mudgett <rmudgett@digium.com>
-
- * bridge core: Add owed T.38 terminate when channel leaves a bridge.
-
- The channel is now going to get T.38 terminated when it leaves the
- bridging system and the bridged peers are going to get T.38 terminated as
- well.
-
- ASTERISK-25582
-
- Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7
-
-2016-02-19 16:01 +0000 [0a5bc64491] Richard Mudgett <rmudgett@digium.com>
-
- * channel api: Create is_t38_active accessor functions.
-
- ASTERISK-25582
-
- Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b
-
-2016-02-19 19:06 +0000 [513638a5f4] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_channel: Don't settle owed events on an optimization.
-
- Local channel optimization could cause DTMF digits to be duplicated.
- Pending DTMF end events would be posted to a bridge when the local channel
- optimizes out and is replaced by the channel further down the chain. When
- the real digit ends, the channel would get another DTMF end posted to the
- bridge.
-
- A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B
-
- 1) LocalA has the /n flag to prevent optimization.
- 2) B is sending DTMF to A through the local channel chain.
- 3) When LocalB optimizes out it can move B to the position of LocalB;1
- 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would
- settle an owed DTMF end to the bridge toward LocalA;2.
- 5) When B finally ends its DTMF it sends the DTMF end down the chain.
- 6) Without this patch, A would hear the DTMF digit end when LocalB
- optimizes out and when B ends the original digit.
-
- ASTERISK-25582
-
- Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251
-
-2016-02-22 12:15 +0000 [7c4495cb70] Richard Mudgett <rmudgett@digium.com>
-
- * channel.c: Route all control frames to a channel through the same code.
-
- Frame hooks can conceivably return a control frame in exchange for an
- audio frame inside ast_write(). Those returned control frames were not
- handled quite the same as if they were sent to ast_indicate(). Now it
- doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a
- channel or ast_indicate().
-
- ASTERISK-25582
-
- Change-Id: I5775f41421aca2b510128198e9b827bf9169629b
-
-2016-02-25 15:13 +0000 [48d713a832] gtjoseph <george.joseph@fairview5.com>
-
- * sorcery: Refactor create, update and delete to better deal with caches
-
- The ast_sorcery_create, update and delete function have been refactored
- to better deal with caches and errors.
-
- The action is now called on all non-caching wizards first. If ANY succeed,
- the action is called on all caching wizards and the observers are notified.
- This way we don't put something in the cache (or update or delete) before
- knowing the action was performed in at least 1 backend and we only call the
- observers once even if there were multiple writable backends.
-
- ast_sorcery_create was never adding to caches in the first place which
- was preventing contacts from getting added to a memory_cache when they
- were created. In turn this was causing memory_cache to emit errors if
- the contact was deleted before being retrieved (which would have
- populated the cache).
-
- ASTERISK-25811 #close
- Reported-by: Ross Beer
-
- Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46
-2016-02-25 15:39 +0000 [ee947d4a7a] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s.
-
- There are a few cases where we're emitting notices or warnings
- for things that really need neither, like a client retrying to subscribe
- to mwi when they're not conifgured for it. They get a 404 so there's no
- need for non-debug messages.
-
- Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f
-2016-02-25 14:17 +0000 [6e70e8ccdb] gtjoseph <george.joseph@fairview5.com>
-
- * res_sorcery_memory_cache: Fix SEGV in some CLI commands
-
- A few of the CLI commands weren't checking for enough arguments
- and were SEGVing.
-
- Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413
-
-2016-02-25 10:29 +0000 [4417f64d83] Leif Madsen <leif@leifmadsen.com>
-
- * Add initial support to build Docker images
-
- This work-in-progress is the first step to being able to reliably
- build Asterisk containers from the Asterisk source. I'm submitting
- this based on feedback gained at AstriDevCon 2015.
-
- Information about how to use this is provided in contrib/docker/README.md
- and will result in a local Asterisk container being built right from
- your source. I believe this can eventually be automated via
- hub.docker.com.
-
- Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1
-
-2016-02-22 19:31 +0000 [e7a6abbbd3] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.h: Remove extraneous semicolons.
-
- Change-Id: Ib462633d396fa941379dfef648dcd2245e350084
-
-2016-02-23 14:57 +0000 [6656afffa0] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Suppress T.38 SDP c= line if addr is the same.
-
- Use the correct comparison function since we only care if the address
- without the port is the same.
-
- Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0
-
-2016-02-16 08:14 +0000 [ea9deff996] Christof Lauber <christof.lauber@annax.ch>
-
- * res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables
-
- Introduced realloaction of ast_str buf in sqlite3_escape functions in case
- the returned buffer from threadstorage was actually too small.
-
- Change-Id: I3c5eb43aaade93ee457943daddc651781954c445
-
-2016-02-11 11:01 +0000 [d2a1457e0b] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/config_transport: Allow reloading transports.
-
- The 'reload' mechanism actually involves closing the underlying
- socket and calling the appropriate udp, tcp or tls start functions
- again. Only outbound_registration, pubsub and session needed work
- to reset the transport before sending requests to insure that the
- pjsip transport didn't get pulled out from under them.
-
- In my testing, no calls were dropped when a transport was changed
- for any of the 3 transport types even if ip addresses or ports were
- changed. To be on the safe side however, a new transport option was
- added (allow_reload) which defaults to 'no'. Unless it's explicitly
- set to 'yes' for a transport, changes to that transport will be ignored
- on a reload of res_pjsip. This should preserve the current behavior.
-
- Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
-
-2016-02-07 17:34 +0000 [6b921f706d] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjproject: Add ability to map pjproject log levels to Asterisk log levels
-
- Warnings and errors in the pjproject libraries are generally handled by
- Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings
- or errors so the messages emitted by pjproject directly are either superfluous
- or misleading. A good exampe of this are the level-0 errors pjproject emits
- when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't
- consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS
- client be treated any differently?
-
- A config file for res_pjproject has bene added (pjproject.conf) and a new
- log_mappings object allows mapping pjproject levels to Asterisk levels
- (or nothing). The defaults if no pjproject.conf file is found are the same
- as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR,
- 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level>
-
- Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898
-
-2016-02-18 10:55 +0000 [f295088764] Alexei Gradinari <alex2grad@gmail.com>
-
- * res_pjsip_outbound_publish: Fix processing 412 response
-
- When Asterisk receives a 412 (Conditional Request Failed) response
- it has to recreate publish session.
- There is bug in res_pjsip_outbound_publish.c
- The function sip_outbound_publish_client_alloc is called with wrong object
- while processing 412 (Conditional Request Failed) response.
- This patch fixes it.
-
- ASTERISK-25229 #close
-
- Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359
-2016-02-18 11:15 +0000 [f1f79812c1] Mark Michelson <mmichelson@digium.com>
-
- * Fix failing threadpool_auto_increment test.
-
- The threadpool_auto_increment test fails infrequently for a couple of
- reasons
- * The threadpool listener was notified of fewer tasks being pushed than
- were actually pushed
- * The "was_empty" flag was set to an unexpected value.
-
- The problem is that the test pushes three tasks into the threadpool.
- Test expects the threadpool to essentially gather those three tasks, and
- then distribute those to the threadpool threads. It also expects that as
- the tasks are pushed in, the threadpool listener is alerted immediately
- that the tasks have been pushed. In reality, a task can be distributed
- to the threadpool threads quicker than expected, meaning that the
- threadpool has already emptied by the time each subsequent task is
- pushed. In addition, the internal threadpool queue can be delayed so
- that the threadpool listener is not alerted that a task has been pushed
- even after the task has been executed.
-
- From the test's point of view, there's no way to be able to predict
- exactly the order that task execution/listener notifications will occur,
- and there is no way to know which listener notifications will indicate
- that the threadpool was previously empty.
-
- For this reason, the test has been updated to only check the things it
- can check. It ensures that all tasks get executed, that the threads go
- idle after the tasks are executed, and that the listener is told the
- proper number of tasks that were pushed.
-
- Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c
-
-2016-02-16 23:37 +0000 [79dc5e2f00] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>
-
- * app_queue: fix Calculate talktime when is first call answered
-
- Fix calculate of average time for talktime is wrong when is completed the
- first call beacuse the time for talked would be that call.
-
- ASTERISK-25800 #close
-
- Change-Id: I94f79028935913cd9174b090b52bb300b91b9492
-
-2016-02-17 13:30 +0000 [5a3a857dd6] Richard Mudgett <rmudgett@digium.com>
-
- * cel.c: Fix mismatch in ast_cel_track_event() return type.
-
- The return type of ast_cel_track_event() is not large enough to return all
- 64 potential bits of the event enable mask. Fortunately, the defined CEL
- events do not really need all 64 bits and the return value is only used to
- determine if the requested CEL event is enabled.
-
- * Made the ast_cel_track_event() return 0 or 1 only so the return value
- can fit inside an int type instead of zero or a truncated 64 bit non-zero
- value.
-
- Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c
-
-2016-02-16 16:37 +0000 [87ab65c557] gtjoseph <george.joseph@fairview5.com>
-
- * res_odbc: Fix exports.in for missing symbols
-
- res_odbc.exports.in was missing a few symbols.
- Changed to wildcards.
-
- Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c
-
-2016-02-16 12:20 +0000 [c0f3062031] gtjoseph <george.joseph@fairview5.com>
-
- * res_statsd: Fix exports.in for missing symbols
-
- res_statsd.export.in was missing the _va variations of the log
- functions causing Asterisk to crash in res_pjsip if OPTIONAL_API
- wasn't enabled.
-
- ASTERISK-25727 #close
- Reported-by: Gergely Dömsödi
-
- Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b
-
-2016-02-15 21:31 +0000 [5e848dae7b] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_config_wizard: Add command to export primitive objects
-
- A new command (pjsip export config_wizard primitives) has been added that
- will export all the pjsip objects it created to the console or a file
- suitable for reuse in a pjsip.conf file.
-
- ASTERISK-24919 #close
- Reported-by: Ray Crumrine
-
- Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b
-
-2016-02-15 15:37 +0000 [34c64707d1] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header
-
- If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid
- or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify
- the header added by the dialplan function. Since the header added by the
- dialplan function is generic string, there are no virtual functions to parse
- the uri and we get a segfault when we try. Since the modify, was really only
- an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER
- and recreate it.
-
- This raises a question for another time though: What should happen with
- duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups
- so if it's session supplement is loaded after res_pjsip_caller_id's (or any
- other module that adds headers), there'll be dups in the message.
-
- ASTERISK-25337 #close
-
- Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa
-
-2016-02-15 13:08 +0000 [ebe167f792] Mark Michelson <mmichelson@digium.com>
-
- * Fix creation race of contact_status structures.
-
- It is possible when processing a SIP REGISTER request to have two
- threads end up creating contact_status structures in sorcery.
- contact_status is created using a "find or create" function. If two
- threads call into this at the same time, each thread will fail to find
- an existing contact_status, and so both will end up creating a new
- contact status.
-
- During testing, we would see sporadic failures because the
- PJSIP_CONTACT() dialplan function would operate on a different
- contact_status than what had been updated by res_pjsip/pjsip_options.
-
- The fix here is two-fold:
- 1) The "find or create" function for contact_status now has a lock
- around the entire operation. This way, if two threads attempt the
- operation simultaneously, the first to get there will create the object,
- and the second will find the object created by the first thread.
-
- 2) res_sorcery_memory has had its create callback updated so that it
- will not allow for objects with duplicate IDs to be created.
-
- Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97
-
-2016-02-15 12:52 +0000 [1c4f2a920d] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_pubsub: Move where the subscription is stored to after initialized.
-
- A problem arose when testing the AMI subscription listing actions where it
- was possible for a subscription that had not been fully initialized to be
- listed. This was problematic as the underlying listing code would crash.
-
- This change makes it so the subscription tree is fully set up before it is
- added to the list of subscriptions. This ensures that when the listing actions
- get the subscription it is valid.
-
- ASTERISK-25738 #close
-
- Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48
-
-2015-02-20 20:51 +0000 [ac00c6bc2d] Corey Farrell <git@cfware.com>
-
- * main/asterisk.c: Reverse #if statement in listener() to fix code folding.
-
- listener() opens the same code block in two places (#if and #else). This
- confuses some folding editors causing it to think that an extra code block
- was opened. Folding in 'geany' causes all code after listener() to be
- folded as if it were part of that procedure.
-
- ASTERISK-24813 #close
-
- Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961
-
-2016-02-09 17:34 +0000 [b1b797e0e7] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Refactor load_module/unload_module
-
- load_module was just too hairy with every step having to clean up all
- previous steps on failure.
-
- Some of the pjproject init calls have now been moved to a separate
- load_pjsip function and the unload_pjsip function was enhanced to clean
- up everything if an error happened at any stage of the load process.
-
- In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
- and ast_threadpool_shutdowns were also corrected.
-
- Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302
-
-2016-02-09 22:42 +0000 [20e9792fbc] Badalyan Vyacheslav <slavon.net@gmail.com>
-
- * Resources/res_phoneprov: fix memory leak and heap-use-after-free
-
- * heap-use-after-free happens when we free "cfg"
- but then use "value" which refers to it
-
- * A memory leak occurs because in some cases
- it is not released "defaults"
-
- ASTERISK-25721 #close
- Reported by: Badalyan Vyacheslav
- Tested by: Badalyan Vyacheslav
-
- Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469
-
-2016-02-11 11:21 +0000 [962a9d61f8] Etienne Lessard (license #6394)
-
- * func_iconv: Ensure output strings are properly terminated.
-
- ASTERISK-25272 #close
- Reported by: Etienne Lessard
- patches:
- AST-25272.patch submitted by Etienne Lessard (license #6394)
-
- Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17
-
-2016-02-10 16:16 +0000 [c1bf014ea0] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Handle pjsip_dlg_create_uas deprecation
-
- Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
- pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
- increments the lock on the returned dialog. To account for this, configure.ac
- now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
- has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
- the original call or the new one. If the new one was used, the ref count is
- decremented before returning.
-
- ASTERISK-25751 #close
- Reported-by Josh Colp
-
- Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8
-
-2016-02-09 23:40 +0000 [bd07b6f0dd] Badalyan Vyacheslav <slavon.net@gmail.com>
-
- * Build: Added testing compiler to support the system sanitizes
-
- In older versions of the compiler was not sanitizes.
- Compilers other than GCC can not support the Usan and TSAN
- or have other options for *FLAGS.
-
- ASTERISK-25767 #close
- Reported by: Badalyan Vyacheslav
- Tested by: Badalyan Vyacheslav
-
- Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916
-
-2016-02-09 20:57 +0000 [e9e896abd1] Badalyan Vyacheslav <v.badalyan@open-bs.ru>
-
- * Build: Fix menuselect USAN conflicts
-
- USAN can be used together with other sanitizers.
-
- Reported by: Badalyan Vyacheslav
- Tested by: Badalyan Vyacheslav
-
- Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f
-
-2016-02-09 14:21 +0000 [93e8ed0154] Corey Farrell <git@cfware.com>
-
- * Simplify and fix conditional in FD_SET.
-
- FD_SET contains a conditional statement to protect against buffer
- overruns. The statement was overly complicated and prevented use
- of the last array element of ast_fdset. We now just verify the fd
- is less than ast_FDMAX.
-
- Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40
-
-2016-02-09 07:11 +0000 [a7c8d4cd6b] Joshua Colp <jcolp@digium.com>
-
- * tests/test_sorcery_memory_cache_thrash: Improve termination process.
-
- When terminating the threads thrashing a sorcery memory cache each
- would be told to stop and then we would wait on them. During at
- least one thrashing test this was problematic due to the specific
- usage pattern in use. It would take some time for termination of the
- thread to occur.
-
- This would occur due to contention between the threads retrieving
- and the threads updating the cache. As the retrieving threads are
- given priority it may be some time before the updating threads
- are able to proceed.
-
- This change makes it so all threads are told to stop and then each
- are joined to ensure they stop. This way all the threads should
- stop at around the same time instead of waiting for one to stop,
- the next to stop, then the next, and so on. As a result of this
- the execution time for each thrash test is much closer to their
- expected value than previously seen as well.
-
- Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827
-2016-01-29 17:56 +0000 [2451d4e455] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Fix infinite recursion when loading transports from realtime
-
- Attempting to load a transport from realtime was forcing asterisk into an
- infinite recursion loop. The first thing transport_apply did was to do a
- sorcery retrieve by id for an existing transport of the same name. For files,
- this just returns the previous object from res_sorcery_config's internal
- container, if any. For realtime, the res_sourcery_realtime driver looks in the
- database and finds the existing row but now it has to rehydrate it into a
- sorcery object which means calling... transport_apply. And so it goes.
-
- The main issue with loading from realtime (apart from the loop) was that
- transport stores structures and pointers directly in the ast_sip_transport
- structure instead of the separate ast_transport_state structure. This patch
- separates those items into the ast_sip_transport_state structure. The pattern
- is roughly the same as res_pjsip_outbound_registration.
-
- Although all current usages of ast_sip_transport and ast_sip_transport_state
- were modified to use the new ast_sip_get_transport_state API, the original
- items are left in ast_sip_transport and kept updated to maintain ABI
- compatability for third-party modules. They are marked as deprecated and
- noted that they're now in ast_sip_transport_state.
-
- ASTERISK-25606 #close
- Reported-by: Martin Moučka
-
- Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19
-
-2016-01-25 17:36 +0000 [6f978fbfe5] Richard Mudgett <rmudgett@digium.com>
-
- * app_confbridge: Only use b_profile options from the conference.
-
- A user cannot set new bridge options after the conference is created by
- the first user. Attempting to do so is documented as undefined behavior.
-
- This patch ensures that the bridge profile options used are from the
- conference and not what a subsequent user may have tried to set.
-
- Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266
-
-2016-02-05 10:29 +0000 [ec8fd6714d] gtjoseph <george.joseph@fairview5.com>
-
- * chan_misdn: Fix a few issues causing compile errors
-
- Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98
-
-2016-02-04 16:17 +0000 [6a799cd78f] Mark Michelson <mmichelson@digium.com>
-
- * Check for OpenSSL defines before trying to use them.
-
- The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
- to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
- these options, which can cause problems on systems with older OpenSSL
- installations.
-
- This commit adds a configure script check for those defines and will not
- attempt to make use of those if they do not exist. We will print a
- warning urging the user to upgrade their OpenSSL installation if those
- defines are not present.
-
- Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
-2016-02-03 14:25 +0000 [953d1cc11a] gtjoseph <george.joseph@fairview5.com>
-
- * pjsip/alembic: Add missing columns to system and registration
-
- ps_systems needed disable_tcp_switch
- ps_registrations needed line and endpoint
-
- ASTERISK-25737 #close
-
- Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19
-
-2016-02-04 11:39 +0000 [23829b3253] Mark Michelson <mmichelson@digium.com>
-
- * res_stasis_device_state: Fix refcounting error.
-
- Device state subscription lifetimes were governed by when the
- subscription was established and unsubscribed from. However, it is
- possible that at the time of unsubscription, there could be device state
- events still in flight. When those device state events occur, the device
- state callback could attempt to dereference a freed pointer. Crash.
-
- This change ensures that the lifetime of the device state subscription
- does not end until the underlying stasis subscription has confirmed that
- its final message has been sent.
-
- Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2
-
-2016-01-27 10:44 +0000 [4e8e6d3922] Sean Bright <sean.bright@gmail.com>
-
- * res_rtp_asterisk: Allow ICE host candidates to be overriden
-
- During ICE negotiation the IPs of the local interfaces are sent to the remote
- peer as host candidates. In many cases Asterisk is behind a static one-to-one
- NAT, so these host addresses will be internal IP addresses.
-
- To help in hiding the topology of the internal network, this patch adds the
- ability to override the host candidates by matching them against a
- user-defined list of replacements.
-
- Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f
-
-2015-12-07 12:46 +0000 [c6b1b2b1c8] Richard Mudgett <rmudgett@digium.com>
-
- * AST-2016-003 udptl.c: Fix uninitialized values.
-
- Sending UDPTL packets to Asterisk with the right amount of missing
- sequence numbers and enough redundant 0-length IFP packets, can make
- Asterisk crash.
-
- ASTERISK-25603 #close
- Reported by: Walter Doekes
-
- ASTERISK-25742 #close
- Reported by: Torrey Searle
-
- Change-Id: I97df8375041be986f3f266ac1946a538023a5255
-2016-02-03 12:05 +0000 [f8acadde2c] Joshua Colp <jcolp@digium.com>
-
- * AST-2016-001 http: Provide greater control of TLS and set modern defaults.
-
- This change exposes the configuration of various aspects of the TLS
- support and sets the default to the modern standards.
-
- The TLS cipher is now set to the best values according to the
- Mozilla OpSec team, different TLS versions can now be disabled, and
- the cipher order can be forced to be that of the server instead of
- the client.
-
- ASTERISK-24972 #close
-
- Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
-2015-09-28 17:07 +0000 [3c81a052c8] Richard Mudgett <rmudgett@digium.com>
-
- * AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow.
-
- Setting the sip.conf timert1 value to a value higher than 1245 can cause
- an integer overflow and result in large retransmit timeout times. These
- large timeout times hold system file descriptors hostage and can cause the
- system to run out of file descriptors.
-
- NOTE: The default sip.conf timert1 value is 500 which does not expose the
- vulnerability.
-
- * The overflow is now detected and the previous timeout time is
- calculated.
-
- ASTERISK-25397 #close
- Reported by: Alexander Traud
-
- Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
-2016-02-03 14:07 +0000 [2a6ee8caeb] gtjoseph <george.joseph@fairview5.com>
-
- * logging: Remove/fix some message annoyances
-
- test_dlinklists doesn't need to NOTICE everyone that every macro worked.
-
- res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or
- provider was registered.
-
- res_odbc was missing a newline at the end of one message.
-
- Change-Id: I6c06361518ef3711821795e535acd439782a995e
-
-2016-02-02 10:52 +0000 [32fc784284] Alexei Gradinari License #5691
-
- * res_sorcery_realtime: Fix regex regression.
-
- A regression was introduced where searching for realtime PJSIP objects
- by regex by starting the regex with a leading "^" would cause no items
- to be returned.
-
- This was due to a change which attempted to drop the requirement for a
- leading "^" to be present due to how some CLI commands formulate their
- regexes. However, the change, rather than simply eliminating the
- requirement, caused any regexes that did begin with "^" to end up not
- returning the expected results.
-
- This change fixes the problem by inspecting the regex and formulating
- the realtime query differently depending on if it begins with "^".
-
- ASTERISK-25702 #close
- Reported by Nic Colledge
-
- Patches:
- realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691
-
- Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693
-
-2016-02-02 04:05 +0000 [0405c31756] Karsten Wemheuer <kwe-digium@iptam.com>
-
- * res_xmpp: Does not connect in component mode
-
- The module res_xmpp does not accept usernames in the form used in component
- mode (XEP-0114). In component mode there is no @something in the name.
- In component mode the connection is now not dropped anymore.
-
- If the xmpp server sends out a "stream" tag before handshake is finished,
- the connection gets dropped in res_xmpp. Now this tag will be ignored and
- the connection will be established.
-
- After connecting there will be an exchange of presence states. This does
- not work as expected in component mode. The responsible function
- "xmpp_pak_presence" is left before the states get sent out. Sending
- presence states in component mode is now moved to the top of the function.
-
- ASTERISK-25735 #close
-
- Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c
-2016-02-01 13:04 +0000 [8804d0973c] gtjoseph <george.joseph@fairview5.com>
-
- * build_system: Fix some warnings highlighted by clang
-
- Fix some warnings found with clang.
-
- Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd
-
-2016-02-01 13:16 +0000 [109b0aff6b] gtjoseph <george.joseph@fairview5.com>
-
- * res/Makefile: Fix bug in "clean" target for ari
-
- The "clean" target was attempting to clean res/ari from inside
- the res directory which doesn't remove anything. Removed the res/
- prefix.
-
- Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6
-
-2016-01-31 20:13 +0000 [a85fab7c44] gtjoseph <george.joseph@fairview5.com>
-
- * pjsip/alembic: Fix definition of qualify_timeout
-
- A recent commit set qualify_timeout to Decimal which isn't supported.
- This path corrects it to Float.
-
- Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf
-
-2016-01-29 07:39 +0000 [aa9348ab9a] Stefan Engström <stefanen@kth.se>
-
- * chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip.
-
- When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a)
- AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect
- asterisk to include the same value for its own ip in both cases a) and b),
- but it seems a) produces a contact header like Contact:
- <sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like
- <sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf
-
- My guess is that manager_sipnotify should call
- ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does,
- because after applying this patch, both cases a) and b) produce
- the contact header that I expect: <sip:asterisk@192.168.1.227:8060>
-
- Reported by: Stefan Engström
- Tested by: Stefan Engström
-
- Change-Id: I86af5e209db64aab82c25417de6c768fb645f476
-2015-12-23 15:07 +0000 [65bd4fcc3f] Mark Michelson <mmichelson@digium.com>
-
- * res_odbc: Remove connection management
-
- Asterisk by default will create a single database connection and share
- it among all threads that attempt to access the database. In previous
- versions of Asterisk, this was tolerable, because the most used channel
- driver, chan_sip, mostly accessed the database from a single thread.
- With PJSIP, however, many threads may be attempting to perform database
- operations, and there is the potential for many more database accesses,
- meaning the concurrency is a horrible bottleneck if only one connection
- is shared.
-
- Asterisk has a connection pooling facility built into it, but the
- implementation has flaws. For one, there is a strict limit on the number
- of simultaneous connections that could be made to the database. Anything
- beyond the maximum would result in a failed operation. Attempting to
- predict what the maximum should be is nearly impossible even for someone
- intimately familiar with Asterisk's threading model. In addition, use of
- transactions in the dialplan can cause some severe bugs if connection
- pooling is enabled.
-
- This commit seeks to fix the concurrency problem by removing all
- connection management code from Asterisk and leaving that to the
- underlying unixODBC code instead. Now, Asterisk does not share a single
- connection, nor does it try to maintain a connection pool. Instead, all
- Asterisk ever does is request a connection from unixODBC and allow
- unixODBC to either allocate those connections or retrieve them from a
- pool.
-
- Doing this has a bit of a ripple effect. For one, since connections are
- not long-lived objects, several of the safeguards that previously
- existed have been removed. We don't have to worry about trying to use a
- connection that has gone stale. In every case, when we request a
- connection, it has just been made and we don't need to perform any
- sanity checks to be sure it's still active.
-
- Another major player affected by this change is transactions.
- Transactions and their respective connections were so tightly coupled
- that it was almost pornographic. This code change moves
- transaction-related code to its own file separate from the core ODBC
- functionality. This way, the core of ODBC does not even have to know
- that transactions exist.
-
- In making this large change, I had to look at a lot of code and
- understand it. When making this change, I discovered several places
- where the behavior is definitely not ideal, but it seemed outside the
- scope of this change to be fixing it. Instead, any place where I saw
- some sort of room for improvement has had a XXX comment added explaining
- what could be altered to improve it.
-
- Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf
-
-2016-01-28 12:44 +0000 [2a9e623ff9] Richard Mudgett <rmudgett@digium.com>
-
- * config_options.c: Fix warning message wording.
-
- Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4
-
-2016-01-25 17:34 +0000 [ed3c9c1512] Richard Mudgett <rmudgett@digium.com>
-
- * app_confbridge.c: Replace inlined code with existing function.
-
- Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51
-
-2016-01-25 16:05 +0000 [1d0abf86e7] Richard Mudgett <rmudgett@digium.com>
-
- * app_confbridge: Add ability to get the muted conference state.
-
- * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.
-
- * Added Muted header to AMI ConfbridgeListRooms action response list
- events to indicate the muted conference state.
-
- * Added Muted column to CLI "confbridge list" output to indicate the muted
- conference state and made the locked column a yes/no value instead of a
- locked/unlocked value.
-
- ASTERISK-20987
- Reported by: hristo
-
- Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1
-
-2016-01-26 17:59 +0000 [f0d40afa69] Richard Mudgett <rmudgett@digium.com>
-
- * app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
-
- Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7
-
-2016-01-25 15:48 +0000 [3e51e5c7fd] Richard Mudgett <rmudgett@digium.com>
-
- * app_confbridge: Make non-admin users join a muted conference muted.
-
- ASTERISK-20987 #close
- Reported by: hristo
-
- Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38
-
-2016-01-27 13:02 +0000 [9da18af992] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add res_pjproject dependency to UPGRADE.txt and samples
-
- Since res_pjsip now depends on res_pjproject, this is now mentioned
- in UPGRADE.txt and the basic-pbx modules.conf has been updated.
-
- Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d
-2016-01-27 10:29 +0000 [aee8448bc2] gtjoseph <george.joseph@fairview5.com>
-
- * build_system: Prevent goals needing makeopts from running when it's missing
-
- The Makefile only optionally includes makeopts so when goals like uninstall that
- dont depend on anything else are run after a distclean, rules like
- 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts
- to remove everything in the root directory.
-
- Although there's a rule defined for makeopts which prints a message and does
- an 'exit 1', since '-include makepopts' was specified (with the -), the exit
- was ignored letting the rest of the rules run.
-
- This patch makes makeopts required unless the goal has the string 'clean' in it.
-
- ASTERISK-25730 #close
- Reported-by: George Joseph
-
- Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7
-
-2016-01-25 09:35 +0000 [f22074e5d9] Joshua Colp <jcolp@digium.com>
-
- * config: Allow options to register when documentation is unavailable.
-
- The config options framework is strict in that configuration options must
- be documented unless XML documentation support is not available. In
- practice this is useful as it ensures documentation exists however in
- off-nominal cases this can cause strange problems.
-
- If it is expected that a config option has a non-zero or non-empty
- default value but the config option documentation is unavailable
- this reasonable expectation will not be met. This can cause obscure
- crashes and weirdness depending on how the code handles it.
-
- This change tweaks the behavior to ensure that the config option
- is still allowed to register, apply default values, and be set when
- devmode is not enabled. If devmode is enabled then the option can
- NOT be set.
-
- This also does not remove the initial documentation error message that
- is output on load when registering the configuration option.
-
- ASTERISK-25725 #close
-
- Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8
-
-2016-01-25 10:23 +0000 [4a3275abb9] Mark Michelson <mmichelson@digium.com>
-
- * Stasis: Use custom structure when setting variables.
-
- A recent change to queue channel variable setting to the Stasis control
- queue caused a regression. When setting channel variables, it is
- possible to give a NULL channel variable value in order to unset the
- variable (i.e. remove it from the channel variable list). The change
- introduced a call to ast_variable_new(), which is not tolerant of NULL
- channel variable values.
-
- This new change switches from using ast_variable to using a custom
- channel variable struct that is lighter weight and NULL value-tolerant.
-
- Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d
-
-2016-01-25 16:56 +0000 [b2c8a99f9e] Rusty Newton <rnewton@digium.com>
-
- * sounds/Makefile: Incremented core and extra sounds versions to 1.5
-
- Core and extra sounds 1.5 was recently released! The tarballs contain
- change descriptions however I figure more people will see this one so
- I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra
- to Core for en, en_GB, fr and added for languages that didn't already
- have Extra sound sets (it,ja,ru).
-
- In addition all of the English and Russian sounds have been completely
- re-recorded.
-
- Sounds moved and added:
- activated,added,all-circuits-busy-now,astcc-followed-by-pound
- at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy
- ,call-fwd-unconditional,calling,call-waiting,cancelled,
- cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated
- ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist
- ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello
- ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to
- ,location,number,number-not-answering,num-was-successfully,one-moment-please
- ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option
- ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached
- ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial
- ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension
- ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered
- ,your
-
- There were also a few random fixes here and there to file names for a few
- of the languages.
-
- ASTERISK-25068 #close
-
- Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3
-2016-01-25 16:51 +0000 [8261bda1bf] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Prevent crash from AMI command on freed subscription.
-
- A test recently uncovered that running an ill-timed AMI command to show
- inbound subscriptions could cause a crash since Asterisk will try to
- operate on a freed subscription.
-
- The fix for this is to remove the subscription tree from the list of
- subscriptions at the time that we are sending our final NOTIFY request
- out. This way, as the subscription is in the process of dying, it is
- inaccessible from AMI.
-
- Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23
-
-2016-01-25 11:03 +0000 [a6823bb0c4] Corey Farrell <git@cfware.com>
-
- * chan_sip: Fix buffer overrun in sip_sipredirect.
-
- sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer
- of 256 characters. This patch reduces the copy to 255 characters to leave
- room for the string null terminator.
-
- ASTERISK-25722 #close
-
- Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab
-
-2016-01-22 15:08 +0000 [1003c2eb05] Mark Michelson <mmichelson@digium.com>
-
- * Stasis: Fix potential memory leak of control data.
-
- When queuing tasks onto the Stasis control queue, you can pass an
- arbitrary data pointer and a function to free that data. All ARI
- commands that use the Stasis control queue made the assumption that the
- destructor function would be called in all paths, whether the task was
- queued successfully or not. However, this was not correct. If a task was
- queued onto a control structure that was already completed, the
- allocated data would not be freed properly.
-
- This patch corrects this by making sure that all return paths call the
- data destructor.
-
- Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb
-
-2016-01-21 10:58 +0000 [eedd77fda0] Mark Michelson <mmichelson@digium.com>
-
- * Stasis: Use control queue to prevent crash.
-
- A crash occurred when attempting to set a channel variable on a channel
- that had already been hung up. This is because there is a small window
- between when a control is grabbed and when the channel variable is set
- that the channel can be hung up.
-
- The fix here is to queue the setting of the channel variable onto the
- control queue. This way, the manipulation of the channel happens in a
- thread where it is safe to be done.
-
- In this change, I also noticed that the setting of bridge roles on
- channels was being done outside of the control queue, so I also changed
- those operations to be done in the control queue.
-
- ASTERISK-25709 #close
- Reported by Mark Michelson
-
- Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78
-
-2016-01-22 11:48 +0000 [1c95b211a0] Richard Mudgett <rmudgett@digium.com>
-
- * logger.c: Fix buffer overrun found by address sanitizer.
-
- The null terminator of the tail struct member was not being allocated
- when no logger.conf config file is installed.
-
- ASTERISK-25714 #close
- Reported by: Badalian Vyacheslav
-
- Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30
-
-2016-01-21 16:40 +0000 [6ff945ab87] Corey Farrell <git@cfware.com>
-
- * Build System: Add support for checking alembic branches.
-
- * Add 'check-alembic' target to root Makefile.
- * Create build_tools/make_check_alembic to do the actual checks.
-
- ASTERISK-25685
-
- Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004
-
-2016-01-19 18:20 +0000 [02035212de] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case.
-
- ASTERISK-25712 #close
- Reported by: Richard Mudgett
-
- Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f
-
-2016-01-18 03:49 +0000 [c68c66c61f] Diederik de Groot <ddegroot@talon.nl>
-
- * main/asterisk.c: ast_el_read_char
-
- Make sure buf[res] is not accessed at res=-1 (buffer underrun).
- Address Sanitizer will complain about this quite loudly.
-
- ASTERISK-24801 #close
-
- Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9
-
-2016-01-13 16:49 +0000 [f87c3275cc] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Add CLI "pjsip dump endpt [details]"
-
- Dump the res_pjsip endpt internals.
-
- In non-developer mode we will not document or make easily accessible the
- "details" option even though it is still available. The user has to know
- it exists to use it. Presumably they would also be aware of the potential
- crash warning below.
-
- Warning: PJPROJECT documents that the function used by this CLI command
- may cause a crash when asking for details because it tries to access all
- active memory pools.
-
- Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb
-
-2016-01-18 17:16 +0000 [46b2de55f9] Matt Jordan <mjordan@digium.com>
-
- * funcs/func_cdr: Correctly report high precision values for duration and billsec
-
- When CDRs were refactored, func_cdr's ability to report high precision values
- for duration and billsec (the 'f' option) was broken. This was due to func_cdr
- incorrectly interpreting the duration/billsec values provided by the CDR engine
- in milliseconds, as opposed to seconds. Since the CDR engine only provides
- duration and billsec in seconds, and does not expose either attribute with
- sufficient precision to merely pass back the underlying value, this patch fixes
- the bug by re-calculating duration and billsec with microsecond precision based
- on the start/answer/end times on the CDR.
-
- ASTERISK-25179 #close
-
- Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841
-
-2016-01-18 19:20 +0000 [137fe5ae01] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjproject: Add module providing pjproject logging and utils
-
- res_pjsip_log_forwarder has been renamed to res_pjproject
- and enhanced as follows:
-
- As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch,
- a new ast_pjproject_get_buildopt function has been added. It
- allows the caller to get the value of one of the buildopts.
-
- The initial use case is retrieving the runtime value of
- PJ_MAX_HOSTNAME to insure we don't send a hostname greater
- than pjproject can handle. Since it can differ between
- the version of pjproject that Asterisk was compiled against
- and the version of pjproject that Asterisk is running against,
- we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk
- source code.
-
- Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e
-
-2016-01-19 17:15 +0000 [b5c13c1545] Joshua Colp <jcolp@digium.com>
-
- * test_threadpool: Wait for each task to complete and fix memory leak.
-
- This change makes the thread_timeout_thrash unit test wait for
- each task to complete. This fixes the problem where the test would
- prematurely end when all threads were gone and a new one had to be
- started to handle the last task. It also increases the thrasing as
- it is now more likely for each task to encounter the above scenario.
-
- This also fixes a memory leak where the data for each task was not
- being freed.
-
- ASTERISK-25611 #close
-
- Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6
-
-2016-01-18 19:44 +0000 [0ab89182d9] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Increase CLI "core ping taskprocessor" timeout.
-
- Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029
-
-2016-01-18 19:43 +0000 [a2a8ea3330] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Fix some taskprocessor unrefs.
-
- You have to call ast_taskprocessor_unref() outside of the taskprocessor
- implementation code. Taskprocessor use since v12 has become more
- transient than just the singleton uses in earlier versions.
-
- Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb
-
-2016-01-19 13:44 +0000 [d604a9afc8] Richard Mudgett <rmudgett@digium.com>
-
- * Fix alembic branches on v13.
-
- Change-Id: I313449b609ede18ad1e1763a655dd23b9210a8e0
-
-2016-01-18 18:45 +0000 [a0c79f3a4f] gtjoseph <george.joseph@fairview5.com>
-
- * pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject
-
- Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b
-
-2016-01-14 09:26 +0000 [018ccf680b] Rusty Newton <rnewton@digium.com>
-
- * func_channel: Add help text for undocumented CHANNEL function arguments
-
- Adding help text documentation for:
- * hangupsource
- * appname
- * appdata
- * exten
- * context
- * channame
- * uniqueid
- * linkedid
-
- ASTERISK-24097 #close
- Reported by: Steven T. Wheeler
- Tested by: Rusty Newton
-
- Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d
-
-2016-01-16 13:18 +0000 [5644bca9f9] Daniel Journo <dan@keshercommunications.com>
-
- * Update version number in features.conf.sample
-
- Update the version number in the comments from Asterisk 12 to Asterisk 12+
-
- Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b
-
-2016-01-15 19:52 +0000 [3f5f30cf82] Corey Farrell <git@cfware.com>
-
- * main/config: Clean config maps on shutdown.
-
- ASTERISK-25700 #close
-
- Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808
-
-2016-01-14 14:42 +0000 [660fedecb7] Kevin Harwell <kharwell@digium.com>
-
- * bridge_basic: don't cache xferfailsound during an attended transfer
-
- The xferfailsound was read from the channel at the beginning of the transfer,
- and that value is "cached" for the duration of the transfer. Therefore, changing
- the xferfailsound on the channel using the FEATURE() dialplan function does
- nothing once the transfer is under way.
-
- This makes it so the transfer code instead gets the xferfailsound configuration
- options from the channel when it is actually going to be used.
-
- This patch also fixes a potential memory leak of the props object as well as
- making sure the condition variable gets initialized before being destroyed.
-
- ASTERISK-25696 #close
-
- Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4
-
-2015-07-10 10:37 +0000 [9cda1de34d] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Simplify ast_taskprocessor_get() return code.
-
- Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1
-
-2016-01-13 18:20 +0000 [a79af2b312] Richard Mudgett <rmudgett@digium.com>
-
- * astmm.c: Add more stats to CLI "memory show" commands.
-
- * Add freed regions totals to allocations and summary.
-
- * Add totals for all allocations and not just the selected allocations.
-
- Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a
-
-2016-01-14 16:00 +0000 [83feb7db3b] Kevin Harwell <kharwell@digium.com>
-
- * bridge_basic: don't play an attended transfer fail sound after target hangs up
-
- If the attended transfer destination answers (picks call up or goes to
- voicemail) and then hangs up on the transferer then transferer hears the
- fail sound.
-
- This patch makes it so the fail sound is not played when the transfer
- destination/target hangs up after answering.
-
- ASTERISK-25697 #close
-
- Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded
-
-2016-01-14 13:22 +0000 [935d641f3b] Mark Michelson <mmichelson@digium.com>
-
- * Remove res/ari/* content during 'make clean'.
-
- 'make clean' and 'make distclean' can leave behind .o files in the
- res/ari/ directory. One observed consequence of this is that running
- Asterisk with MALLOC_DEBUG can cause Asterisk to crash immediately on
- startup sometimes.
-
- By ensuring that we are making a clean build, we can be sure that stale
- files are not being included in the build and causing problems when
- build options should have caused files to be re-built.
-
- ASTERISK-25683 #close
- Reported by yaron nahum
-
- Change-Id: I1f48baa904d2468eddeefb42ee68a56af7adc7b7
-
-2016-01-13 15:58 +0000 [46f21df302] Daniel Journo <dan@keshercommunications.com>
-
- * pjsip/alembic: Fix qualify_timeout column definition
-
- Corrects the qualify_timeout column type from Integer to Decimal
-
- ASTERISK-25686 #close
- Reported-by: Marcelo Terres
-
- Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8
-
-2016-01-12 11:14 +0000 [32b29d7b02] Joshua Colp <jcolp@digium.com>
-
- * app: Queue hangup if channel is hung up during sub or macro execution.
-
- This issue was exposed when executing a connected line subroutine.
- When connected or redirected subroutines or macros are executed it is
- expected that the underlying applications and logic invoked are fast
- and do not consume frames. In practice this constraint is not enforced
- and if not adhered to will cause channels to continue when they shouldn't.
- This is because each caller of the connected or redirected logic does not
- check whether the channel has been hung up on return. As a result the
- the hung up channel continues.
-
- This change makes it so when the API to execute a subroutine or
- macro is invoked the channel is checked to determine if it has hung up.
- If it has then a hangup is queued again so the caller will see it
- and stop.
-
- ASTERISK-25690 #close
-
- Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea
-
-2016-01-13 07:20 +0000 [e7cfda0b38] Sean Bright <sean.bright@gmail.com>
-
- * res_musiconhold: Prevent multiple simultaneous reloads.
-
- There are two ways in which the reload() function in res_musiconhold can be
- called from the CLI:
-
- * module reload res_musiconhold.so
- * moh reload
-
- In the former case, the module loader holds a lock that prevents multiple
- concurrent calls, but in the latter there is no such protection.
-
- This patch changes the 'moh reload' CLI command to invoke the module loader
- directly, rather than call reload() explicitly.
-
- ASTERISK-25687 #close
-
- Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c
-2016-01-12 14:25 +0000 [5586abc957] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts".
-
- PJPROJECT has a function available to dump the compile time
- options used when building the library.
-
- * Add CLI "pjsip show buildopts" command.
-
- * Update contrib/scripts/autosupport to get pjproject information.
-
- Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748
-
-2016-01-12 10:36 +0000 [4cd58c3b20] Mark Michelson <mmichelson@digium.com>
-
- * res_sorcery_realtime: Remove leading ^ requirement.
-
- res_sorcery_realtime's search-by-regex callback performed a check to
- ensure that the passed-in regex began with a caret (^). If it did not,
- then no results would be returned.
-
- This callback only started to become used when "like" support was added
- to PJSIP CLI commands. The CLI command for listing objects would pass an
- empty regex ("") to the sorcery backend if no "like" statement was
- present. For most sorcery backends, this resulted in returning all
- objects. However, for realtime, this resulted in returning no objects.
-
- This commit seeks to fix the regression by removing the requirement from
- res_sorcery_realtime for the passed-in-regex to begin with a caret.
-
- ASTERISK-25689 #close
- Reported by Marcelo Terres
-
- Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20
-
-2016-01-07 11:57 +0000 [219c204a41] gtjoseph <george.joseph@fairview5.com>
-
- * pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address
-
- On a system with multiple ip addresses in the same subnet, if a
- transport is bound to a specific ip address and endpoint/media_address
- is set, the SIP/SDP will have the correct address in all fields but
- the rtp stream MAY still originate from one of the other ip addresses,
- most probably the "primary" ip address. This happens because
- res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
- the "all" ip address (0.0.0.0 or ::).
-
- The new option causes res_pjsip_sdp_rtp/create_rtp to call
- ast_rtp_instance_new with the endpoint's media_address (if specified)
- instead of the "all" address. This causes the packets to originate from
- the specified address.
-
- ASTERISK-25632
- ASTERISK-25637
- Reported-by: Olivier Krief
- Reported-by: Dan Journo
-
- Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
-
-2016-01-10 16:22 +0000 [22801a06ee] Daniel Journo <dan@keshercommunications.com>
-
- * pjsip: Add option global/regcontext
-
- Added new global option (regcontext) to pjsip. When set, Asterisk will
- dynamically create and destroy a NoOp priority 1 extension
- for a given endpoint who registers or unregisters with us.
-
- ASTERISK-25670 #close
- Reported-by: Daniel Journo
-
- Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62
-
-2016-01-08 15:22 +0000 [1600ebca7d] Kevin Harwell <kharwell@digium.com>
-
- * pbx: Deadlock between contexts container and context_merge locks
-
- Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5)
- introduced the possibility of a deadlock. Due to the mentioned modifications
- ast_change_hints now needs to keep both merge/delete and state callbacks from
- occurring while it executes. Unfortunately, sometimes ast_change_hints can be
- called with the contexts container locked. When this happens it's possible for
- another thread to grab the context_merge_lock before the thread calling into
- ast_change_hints does and then try to obtain the contexts container lock. This
- of course causes a deadlock between the two threads. The thread calling into
- ast_change_hints waits for the other thread to release context_merge_lock and
- the other thread is waiting on that one to release the contexts container lock.
-
- Unfortunately, there is not a great way to fix this problem. When hints change,
- the subsequent state callbacks cannot run at the same time as a merge/delete,
- nor when the usual state callbacks do. This patch alleviates the problem by
- having those particular callbacks (the ones run after a hint change) occur in a
- serialized task. By moving the context_merge_lock to a task it can now safely be
- attempted or held without a deadlock occurring.
-
- ASTERISK-25640 #close
- Reported by: Krzysztof Trempala
-
- Change-Id: If2210ea241afd1585dc2594c16faff84579bf302
-
-2016-01-10 17:08 +0000 [0fc3dad965] Corey Farrell <git@cfware.com>
-
- * devicestate: Cleanup engine thread during graceful shutdown.
-
- ASTERISK-25681 #close
-
- Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1
-
-2016-01-10 13:51 +0000 [f34dd10495] Corey Farrell <git@cfware.com>
-
- * manager: Cleanup manager_channelvars during shutdown.
-
- ASTERISK-25680 #close
-
- Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446
-
-2016-01-10 13:27 +0000 [1d3a1167fc] Corey Farrell <git@cfware.com>
-
- * res_calendar: Cleanup scheduler context at unload.
-
- ASTERISK-25679 #close
-
- Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f
-
-2016-01-08 11:49 +0000 [3a160cdbf6] Joshua Colp <jcolp@digium.com>
-
- * res_rtp_asterisk: Revert DTLS negotiation changes.
-
- Due to locking issues within pjnath these changes are being
- reverted until pjnath can be changed.
-
- ASTERISK-25645
-
- Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays."
-
- This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d.
-
- Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705
-
- Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation"
-
- This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b.
-
- Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe
-
-2016-01-09 17:57 +0000 [4b10fc9173] gtjoseph <george.joseph@fairview5.com>
-
- * Revert "pjsip_location: Delete contact_status object when contact is deleted"
-
- This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45.
-
- Matt,
-
- This patch causes another problem and should not have been needed.
- Before this patch, persistent_endpoint_contact_deleted_observer WAS
- deleting the contact_status when ast_sip_location_delete_contact was
- called. By deleting it yourself in ast_sip_location_delete_contact
- it was gone before the observer could run and the observer therefore
- was throwing an error and not sending stasis/AMI/statsd messages.
-
- So, I don't think this was the cause of your original issue. I also
- had verified the contact AMI and statsd lifecycle and it was working.
- I'll double check now though.
-
- ASTERISK-25675
- Reported-by: Daniel Journo
-
- Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a
-
-2016-01-09 18:04 +0000 [79b4309881] Corey Farrell <git@cfware.com>
-
- * pbx_dundi: Run cleanup on failed load.
-
- During failed startup of pbx_dundi no cleanup was performed. Add a call
- to unload_module before returning AST_MODULE_LOAD_DECLINE.
-
- ASTERISK-25677 #close
-
- Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29
-
-2016-01-09 13:28 +0000 [a5406b1f9e] Corey Farrell <git@cfware.com>
-
- * res_crypto: Perform cleanup at shutdown.
-
- This change causes res_crypto to unregister CLI at shutdown while still
- preventing the module from being unloaded.
-
- ASTERISK-25673 #close
-
- Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc
-
-2016-01-06 19:10 +0000 [cf8e7a580b] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Create human friendly serializer names.
-
- PJSIP name formats:
- pjsip/aor/<aor>-<seq> -- registrar thread pool serializer
- pjsip/default-<seq> -- default thread pool serializer
- pjsip/messaging -- messaging thread pool serializer
- pjsip/outreg/<registration>-<seq> -- outbound registration thread pool
- serializer
- pjsip/pubsub/<endpoint>-<seq> -- pubsub thread pool serializer
- pjsip/refer/<endpoint>-<seq> -- REFER thread pool serializer
- pjsip/session/<endpoint>-<seq> -- session thread pool serializer
- pjsip/websocket-<seq> -- websocket thread pool serializer
-
- Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084
-
-2016-01-06 19:09 +0000 [4276f185f0] Richard Mudgett <rmudgett@digium.com>
-
- * Sorcery: Create human friendly serializer names.
-
- Sorcery name formats:
- sorcery/<type>-<seq> -- Sorcery thread pool serializer
-
- Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47
-
-2016-01-06 19:09 +0000 [f02ac1b7f9] Richard Mudgett <rmudgett@digium.com>
-
- * Stasis: Create human friendly taskprocessor/serializer names.
-
- Stasis name formats:
- subm:<topic>-<seq> -- Stasis subscription mailbox task processor
- subp:<topic>-<seq> -- Stasis subscription thread pool serializer
-
- Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd
-
-2016-01-07 16:15 +0000 [ec1f1c6742] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: New API for human friendly taskprocessor names.
-
- * Add new API call to get a sequence number for use in human friendly
- taskprocessor names.
-
- * Add new API call to create a taskprocessor name in a given buffer and
- append a sequence number.
-
- Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9
-
-2016-01-06 17:19 +0000 [d8bc3e0c8b] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Fix CLI "core show taskprocessors" output format.
-
- Update the CLI "core show taskprocessors" output format to not be
- distorted because UUID names are longer than previously used taskprocessor
- names.
-
- Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601
-
-2016-01-07 21:07 +0000 [2c4b7502de] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Fix CLI "core show taskprocessors" unref.
-
- Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5
-
-2016-01-07 20:44 +0000 [3b33ac7a46] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Sort CLI "core show taskprocessors" output.
-
- Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e
-
-2016-01-06 19:00 +0000 [0fc32c4dd3] Richard Mudgett <rmudgett@digium.com>
-
- * ccss.c: Replace space in taskprocessor name.
-
- The CLI "core ping taskprocessor" command does not work very
- well with taskprocessor names that have spaces in them. You
- have to put quotes around the name so using tab completion
- becomes awkward.
-
- Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0
-
-2016-01-05 16:54 +0000 [0e0c24ad78] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock.
-
- Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b
-
-2016-01-07 03:33 +0000 [0f79c8839b] Diederik de Groot <ddegroot@talon.nl>
-
- * main: Use ast_strdup instead of strdup
-
- Fix compile error in main/utils.c because strdup was used in dummy_start
-
- Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793
-
-2016-01-07 03:21 +0000 [4285dee778] Diederik de Groot <ddegroot@talon.nl>
-
- * include/asterisk/time.h: Renamed global declaration:tv
-
- Renamed global declaration:tv to dummy_tv_var_for_types,
- which would oltherwise cause 'shadow' warnings when 'tv'
- was declared as a local variable elsewhere.
-
- Added comment to note that dummy_tv_var_for_types is never
- really exported and only used as a place holder.
-
- ASTERISK-25627 #close
-
- Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28
-
-2016-01-07 15:37 +0000 [96094feab6] Mark Michelson <mmichelson@digium.com>
-
- * PJSIP: Prevent deadlock due to dialog/transaction lock inversion.
-
- A deadlock was observed where the monitor thread was stuck, therefore
- resulting in no incoming SIP traffic being processed.
-
- The problem occurred when two 200 OK responses arrived in response to a
- terminating NOTIFY request sent from Asterisk. The first 200 OK was
- dispatched to a threadpool worker, who locked the corresponding
- transaction. The second 200 OK arrived, resulting in the monitor thread
- locking the dialog. At this point, the two threads are at odds, because
- the monitor thread attempts to lock the transaction, and the threadpool
- thread loops attempting to try to lock the dialog.
-
- In this case, the fix is to not have the monitor thread attempt to hold
- both the dialog and transaction locks at the same time. Instead, we
- release the dialog lock before attempting to lock the transaction.
-
- There have also been some debug messages added to the process in an
- attempt to make it more clear what is going on in the process.
-
- ASTERISK-25668 #close
- Reported by Mark Michelson
-
- Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a
-
-2016-01-07 09:39 +0000 [52e9de0016] Corey Farrell <git@cfware.com>
-
- * ast_format_cap_append_by_type: Resolve codec reference leak.
-
- This resolves a reference leak caused by ASTERISK-25535. The pointer
- returned by ast_format_get_codec is saved so it can be released.
-
- ASTERISK-25664 #close
-
- Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec
-
-2016-01-04 04:26 +0000 [86eae38d7e] Aaron An <anjb@ti-net.com.cn>
-
- * cel/cel_radius: Fix wrong pointer.
-
- The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter
- y not the address of y.
-
- I capture the radius UDP packet via tcpdump, and the AV pairs are not correct,
- then i review the source code and compare it with cdr/cdr_radius.c. Fix it and
- it works.
-
- ASTERISK-25647 #close
- Reported by: Aaron An
- Tested by: Aaron An
-
- Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0
-
-2016-01-05 14:52 +0000 [881dc862e0] gtjoseph <george.joseph@fairview5.com>
-
- * asterisk.h: Add ASTERISK_REGISTER_FILE macro
-
- The 11/13 branches and master use 2 different file version macros. 11/13
- uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This
- means a new file added to 11/13 can't just be cherry-picked to master
- because the macro has to be changed.
-
- To make cherry-picking possible, ASTERISK_REGISTER_FILE was added
- to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL)
- The "$Revision$" tag doesn't do anything since Asterisk moved to git so
- just passing NULL as the verison works fine. asterisk.h was also
- annotated to deprecate ASTERISK_FILE_VERSION and suggest using
- ASTERISK_REGISTER_FILE for all new files.
-
- Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c,
- were modified to use the new macro to make sure it actually worked.
- 'core show file version' showed the correct output.
-
- Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5
-
-2016-01-05 11:06 +0000 [d228b62fd4] gtjoseph <george.joseph@fairview5.com>
-
- * stasis_cache_pattern: Backport to 13
-
- Somehow stasis_cache_pattern got out of sync between 13 and master
- and it was causing duplicate channel message issues in 13 when
- related to a specific endpoint. I.E. from statsd,
- 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice.
-
- Backporting stasis_cache_pattern from master to 13 solved
- the issue and running the unit and testsuite tests confirmed
- that no new ones were created.
-
- ASTERISK-25317 #close
-
- Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548
-2016-01-04 20:23 +0000 [e462f0063f] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move hangup handler routines to pbx_hangup_handler.c.
-
- This is the sixth patch in a series meant to reduce the bulk of pbx.c.
- This moves hangup handler management functions to their own source.
-
- Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104
-
-2016-01-04 19:46 +0000 [ab191d124c] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move dialplan application management routines to pbx_app.c.
-
- This is the sixth patch in a series meant to reduce the bulk of pbx.c.
- This moves dialplan application management functions to their own source.
-
- Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c
-
-2016-01-04 18:20 +0000 [09a9b93896] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move switch routines to pbx_switch.c.
-
- This is the fifth patch in a series meant to reduce the bulk of pbx.c.
- This moves ast_switch functions to their own source.
-
- Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e
-
-2016-01-04 18:00 +0000 [c608274a39] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move timing routines to pbx_timing.c.
-
- This is the fourth patch in a series meant to reduce the bulk of pbx.c.
- This moves pbx timing functions to their own source.
-
- Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6
-
-2015-12-29 04:31 +0000 [338a8ffed6] Martin Tomec <tomec.martin@gmail.com>
-
- * app_queue: Add member flag "in_call" to prevent reading wrong lastcall time
-
- Member lastcall time is updated later than member status. There was chance to
- check wrapuptime for available member with wrong (old) lastcall time.
- New boolean flag "in_call" is set to true right before connecting call, and
- reset to false after update of lastcall time. Members with "in_call" set to true
- are treat as unavailable.
-
- ASTERISK-19820 #close
-
- Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500
-
-2015-12-28 17:23 +0000 [e13719bff1] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>
-
- * app_queue: Added reason pause of member
-
- In app_queue added value Paused Reason on QueueMemberStatus when a member
- on queue is paused and the reason was set.
-
- ASTERISK-25480 #close
- Reporte by: Rodrigo Ramírez Norambuena
-
- Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e
-
-2015-12-30 10:49 +0000 [4ec85a9f07] gtjoseph <george.joseph@fairview5.com>
-
- * voicemail: Move app_voicemail / res_mwi_external conflict to runtime
-
- The menuselect conflict between app_voicemail and res_mwi_external
- makes it hard to package 1 version of Asterisk. There no actual
- build dependencies between the 2 so moving this check to runtime
- seems like a better solution.
-
- The ast_vm_register and ast_vm_greeter_register functions in app.c
- were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there
- is already a voicemail module registered. The modules' load_module
- functions were then modified to return DECLINE instead of -1 to the
- loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE,
- the modules were incorrectly causing Asterisk to stop so this needed
- to be cleaned up anyway.
-
- Now you can build both and use modules.conf to decide which voicemail
- implementation to load.
-
- The default menuselect options still build app_voicemail and not
- res_mwi_external but if both ARE built, res_mwi_external will load
- first and become the voicemail provider unless modules.conf rules
- prevent it. This is noted in CHANGES.
-
- Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247
-
-2016-01-04 16:22 +0000 [7fdcfd7724] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move variable routines to pbx_variables.c.
-
- This is the third patch in a series meant to reduce the bulk of pbx.c.
- This moves channel and global variable routines to their own source.
-
- Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6
-
-2015-12-04 17:22 +0000 [80a8b2a4cd] Richard Mudgett <rmudgett@digium.com>
-
- * app_dial: Immediately exit dial if the caller is already hung up.
-
- If a caller hangs up before dial is executed within an AGI then the AGI
- has likely eaten all queued frames before executing the dial in DeadAGI
- mode. With the caller hung up and no pending frames from the caller's
- read queue, dial would not know that the call has hung up until a called
- channel answers. It is rather annoying to whoever just answered the
- non-existent call.
-
- Dial should not continue execution in DeadAGI mode, hangup handlers, or
- the h exten.
-
- * Added a check early in dial to abort dialing if the caller has hungup.
-
- ASTERISK-25307 #close
- Reported by: David Cunningham
-
- Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418
-
-2016-01-02 10:26 +0000 [1087b0c6ed] Matt Jordan <mjordan@digium.com>
-
- * main/cdr: Allow setting properties on a finalized CDR if it is the last one
-
- Prior to this patch, we explicitly disallowed setting any properties on a
- finalized CDR. This seemed like a good idea at the time; in practice, it was
- more restrictive.
-
- There are weird and strange scenarios where setting a property on a finalized
- CDR is definitely wrong. For example, we may Fork a CDR, finalizing the
- previous one, then change a property. In said case, the old CDR is supposed
- to now be 'immutable' (so to speak), and should not be updated. From the
- perspective of the code, a forked CDR that is finalized is just finalized.
- Hence why we decided these should not be updated.
-
- In practice, it is much more common to want to set a property on a CDR in
- the h extension or in a hangup handler. Disallowing a common scenario to make
- an esoteric behaviour work isn't good. This patch fixes this by allowing
- callers to set a property IF we are the last CDR in the chain. This preserves
- the finalized CDR if it was forked, while allowing the more common case to
- function.
-
- ASTERISK-25458 #close
-
- Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9
-
-2016-01-02 10:23 +0000 [1f23e65b89] Matt Jordan <mjordan@digium.com>
-
- * main/cdr: Set the end time on a CDR if endbeforehexten is Yes
-
- Prior to this patch, the CDR engine attempted to set the end time on a CDR
- that was executing hangup logic and with endbeforehexten set to Yes by
- calling a function that inspects the properties on the Party A snapshot to
- determine if we are ready to set the end time. That always failed. This is
- because a Party A snapshot is not updated for CDRs that are executing hangup
- logic with endbeforehexten=Yes.
-
- Instead of calling a function that looks at the Party A snapshot, we just
- simply set the end time on the CDR. This is safe to call multiple times, and is
- safe to call at this point as we know that (a) we are executing hangup logic,
- and (b) we are supposed to set the end time at this point.
-
- ASTERISK-25458
-
- Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3
-
-2015-12-30 20:51 +0000 [2ffade4574] Corey Farrell <git@cfware.com>
-
- * main/pbx: Move custom function routines to pbx_functions.c.
-
- This is the second patch in a series meant to reduce the bulk of pbx.c.
- This moves custom function management routines to their own source.
-
- Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177
-
-2015-12-28 19:18 +0000 [20b8474f20] gtjoseph <george.joseph@fairview5.com>
-
- * main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c
-
- We joked about splitting pbx.c into multiple files but this first step was
- fairly easy. All of the pbx_builtin dialplan applications have been moved
- into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
- is called by asterisk.c just after load_pbx().
-
- A few functions were renamed and are cross-exposed between the 2 source files.
-
- Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a
-
-2015-12-24 20:26 +0000 [e4a566918a] Matt Jordan <mjordan@digium.com>
-
- * tests/test_stasis_endpoints: Remove expected duplicate events
-
- The cache_clear test was written to expect duplicate Stasis messages
- sent from the technology endpoint to the all caching topic. This patch
- fixes the test to no longer expect these duplicate messages.
-
- ASTERISK-25137
-
- Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981
-
-2015-12-28 14:02 +0000 [a280400758] Joshua Colp <jcolp@digium.com>
-
- * test_time: Provide a timeout when waiting.
-
- The test_timezone_watch unit test is written to expect a
- condition to be signaled when the inotify daemon thread runs.
- There exists a small window where the test_timezone_watch
- thread can signal the inotify daemon thread while it is not
- reading on the underlying file descriptor. If this occurs
- the test_timezone_watch thread will wait indefinitely for a
- signal that will never arrive.
-
- This change adds a timeout to the condition so it will return
- regardless after a period of time.
-
- Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390
-
-2015-05-27 13:22 +0000 [3a1c4885be] gtjoseph <george.joseph@fairview5.com>
-
- * endpoint/stasis: Eliminate duplicate events on endpoint status change
-
- When an endpoint is created, its messages are forwarded to both the tech
- endpoint topic and the all endpoints topic. This is done so that various
- parties interested in endpoint messages can subscribe to just the tech
- endpoint and receive all messages associated with that particular technology,
- as opposed to subscribing to the all endpoints topic. Unfortunately, when the
- tech endpoint is created, it also forwards all of its messages to the all
- topic. This results in duplicate messages whenever an endpoint publishes its
- messages.
-
- This patch resolves the duplicate message issue by creating a new function
- for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts
- as a normal caching topic, save that it no longer forwards messages it receives
- to the all endpoints topic. This allows it to act as an aggregation "sink",
- while preserving the necessary caching behaviour.
-
- ASTERISK-25137 #close
- Reported-by: Vitezslav Novy
-
- ASTERISK-25116 #close
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
- Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b
-2015-12-24 22:19 +0000 [136c537695] Dade Brandon <dade@xencall.com>
-
- * res_http_websocket.c: prevent avoidable disconnections caused by write errors
-
- Updated ast_websocket_write to encode the entire frame in to one
- write operation, to ensure that we don't end up with a situation
- where the websocket header has been sent, while the body can not
- be written.
-
- Previous to August's patch in commit b9bd3c14, certain network
- conditions could cause the header to be written, and then the
- sub-sequent body to fail - which would cause the next successful
- write to contain a new header, and a new body (resulting in
- the peer receiving two headers - the second of which would be
- read as part of the body for the first header).
-
- This was patched to have both write operations individually fail
- by closing the websocket.
-
- In a case available to the submitter of this patch, the same
- body which would consistently fail to write, would succeed
- if written at the same time as the header.
-
- This update merges the two operations in to one, adds debug messages
- indicating the reason for a websocket connection being closed during
- a write operation, and clarifies some variable names for code legibility.
-
- Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598
-
-2015-12-27 22:38 +0000 [f2efbb5d75] Corey Farrell <git@cfware.com>
-
- * Remove res_jabber file that was left behind.
-
- Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b
-
-2015-12-13 13:09 +0000 [dde7f3c1c4] Matt Jordan <mjordan@digium.com>
-
- * res_pjsip_history: Add a module that provides PJSIP history for debugging
-
- This patch adds a new module, res_pjsip_history, that provides a slightly
- better way of debugging SIP message traffic on a busy Asterisk system. The
- existing mechanisms all rely on passively dumping a SIP message to the CLI.
- While this is perfectly fine for logging purposes and well controlled
- environments, on many installations, the amount of SIP messages Asterisk
- receives will quickly swamp the CLI. This makes it difficult to view/capture
- those messages that you want to diagnose in real time.
-
- This patch provides another way of handling this. When enabled, the module
- will store SIP message traffic in memory. This traffic can then be queried
- at leisure.
-
- In order to make the querying useful, a CLI command has been implemented,
- 'pjsip show history', that supports a basic expression syntax similar to
- SQL or other query languages. A small number of useful fields have been
- added in this initial patch; additional fields can easily be added in
- later improvements. Those fields are:
- - number: The entry index in the history
- - timestamp: The time the message was recieved
- - addr: The source/destination address of the message
- - sip.msg.request.method: The request method
- - sip.msg.call-id: The Call-ID header
-
- Note - this is a resurrection of the module initially proposed on Review Board
- here: https://reviewboard.asterisk.org/r/4053/
-
- Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36
-
-2015-12-25 09:56 +0000 [be050f2638] Dade Brandon <dade@xencall.com>
-
- * chan_sip.c: fix websocket_write_timeout default value
-
- websocket_write_timeout was not being set to its default value
- during sip config reload, which meant that prior to this commit,
- 1) the default value of 100 was not used, unless an invalid value
- (or 1) was specified in sip.conf for websocket_write_timeout, and
- 2) if the websocket_write_timeout directive was removed from sip.conf
- without a full restart of asterisk, then the previous value would
- continue to be used indefinitely.
-
- This essentially lead to a 0ms write timeout (the first write attempt
- in ast_careful_fwrite must have succeeded) in websocket write requests
- from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf.
-
- Changes to websocket_write_timeout still only apply to new websocket
- sessions, after the sip reload -- timeouts on existing sessions are
- not adjusted during sip reload.
-
- Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953
-
-2015-12-23 17:40 +0000 [b3024cad10] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_basic.c: Fix GOTO_ON_BLINDXFR
-
- Use of GOTO_ON_BLINDXFR would not work at all. The target location would
- never be executed by the transferring channel.
-
- * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with
- valid context, exten, and priority parameters from the transferring
- channel.
-
- * Renamed some feature_blind_transfer() local variables for clarity.
-
- ASTERISK-25641 #close
- Reported by Dmitry Melekhov
-
- Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac
-
-2015-12-24 12:19 +0000 [0a9941de9d] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_location: Delete contact_status object when contact is deleted
-
- In 450579e908, a change was made that removed the deletion of the
- 'contact_status' object when a 'contact' object is deleted in sorcery.
- This unfortunately means that the 'contact_status' object persists, even when
- something has explicitly removed a contact. The result is that the state of
- the contact will not be regenerated if that contact is re-created, and the
- stale state will be reported/used for that contact. It also results in
- no ContactStatusChanged events being generated for either ARI or AMI.
-
- This patch restores the deletion logic that was removed. Doing so now
- results in the expected events being generated again.
-
- Change-Id: I28789a112e845072308b5b34522690e3faf58f07
-
-2015-12-24 10:18 +0000 [1e24a0ca8a] Kevin Harwell <kharwell@digium.com>
-
- * res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef
-
- Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151
-
-2015-12-20 21:33 +0000 [1d3d20dd68] Dade Brandon <dade@xencall.com>
-
- * app_amd: Correct documentation to reflect functionality
-
- Update documentation to reflect that maximum_number_of_words
- has functionality inconsistent with the variable name (and inconsistent
- with prior documentation.)
-
- Update documentation for silence_threshold, which previously implied
- that it was measuring time, rather than noise averages in the sample.
-
- Update the comments in amd.conf.sample.
-
- ASTERISK-25639 #close
- Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093
-
-2015-12-17 19:05 +0000 [965a0eee46] Dade Brandon <dade@xencall.com>
-
- * res_rtp_asterisk: Resolve further timing issues with DTLS negotiation
-
- Resolves an edge case dtls negotiation delay for certain networks which
- somehow manage to drop the rtcp side's packet when these are both sent
- ast_rtp_remote_address_set, causing it to have to time-out and restart
- the handshake.
-
- Move dtls pending bio flush in to it's own function, and call it from
- ast_rtp_on_ice_complete, when we're rtp->ice, rather than when
- ast_rtp_remote_address_set.
-
- Keep the existing flush from the recent change to res_rtp_remote_address_set
- if ice is not being used.
-
- ASTERISK-25614 #close
- Reported-by: XenCALL
- Tested by: XenCALL
-
- Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5
-
-2015-12-18 09:54 +0000 [ae428d8460] Carlos Oliva <carlos.oliva@invoxcontact.com>
-
- * app_queue: update RT members when the 1st call joins a queue with no agents
-
- If a call enters on a queue and the members on that queue are updated in
- realtime (ex: using mysql inserting a new agent) the queue members are
- never refreshed and the call will stay in the queue until other event occurs.
- This happens only if this is the first call of the queue and there is no
- agents servicing.
- This patch prevent this issue, ensuring realtime members are updated if
- there is one call in the queue and no available agents
-
- ASTERISK-25442 #close
-
- Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682
-
-2015-12-05 10:01 +0000 [59d5bb0613] Joshua Colp <jcolp@digium.com>
-
- * res_sorcery_memory_cache: Add support for a full backend cache.
-
- This change introduces the configuration option 'full_backend_cache'
- which changes the cache to be a full mirror of the backend instead
- of a per-object cache. This allows all sorcery retrieval operations
- to be carried out against it and is useful for object types which
- are used in a "retrieve all" or "retrieve some" pattern.
-
- ASTERISK-25625 #close
-
- Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5
-
-2015-12-17 10:25 +0000 [0cefcabd58] Joshua Colp <jcolp@digium.com>
-
- * rtp_engine: Ignore empty filenames in DTLS configuration.
-
- When applying an empty DTLS configuration the filenames in the
- configuration will be empty. This is actually valid to do and
- each filename should simply be ignored.
-
- Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539
-
-2015-12-17 08:10 +0000 [158a0a5422] Joshua Colp <jcolp@digium.com>
-
- * chan_sip: Enable WebSocket support by default.
-
- Per the documentation the WebSocket support in chan_sip is
- supposed to be enabled by default but is not. This change
- corrects that.
-
- Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423
-
-2015-12-14 12:04 +0000 [a9d6fc571d] Joshua Colp <jcolp@digium.com>
-
- * json: Audit ast_json_* usage for thread safety.
-
- The JSON library Asterisk uses, jansson, is not thread
- safe for us in a few ways. To help with this wrappers for JSON
- object reference count increasing and decreasing were added
- which use a global lock to ensure they don't clobber over
- each other. This does not extend to reference count manipulation
- within the jansson library itself. This means you can't safely
- use the object borrowing specifier (O) in ast_json_pack and
- you can't share JSON instances between objects.
-
- This change removes uses of the O specifier and replaces them
- with the o specifier and an explicit ast_json_ref. Some cases
- of instance sharing have also been removed.
-
- ASTERISK-25601 #close
-
- Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1
-
-2015-12-16 11:28 +0000 [53bd5a539a] Mark Michelson <mmichelson@digium.com>
-
- * Alembic: Increase column size of PJSIP AOR "contact".
-
- When running the PJSIP AMI "show_endpoint" test with automatic
- conversion to realtime, the test would fail. This was because the AOR
- "contact" column was sized at 40, and the configured contact was larger
- than that.
-
- This commit increases the size of the contact column to 255 characters.
-
- Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1
-
-2015-12-16 11:25 +0000 [da17dc4d75] Mark Michelson <mmichelson@digium.com>
-
- * Alembic: Add PJSIP global keep_alive_interval.
-
- The keep_alive_interval option was added about a year ago, but no
- alembic revision was created to add the appropriate column to the
- database.
-
- This commit fixes the problem and adds the column. This was discovered
- by running the testsuite with automatic conversion to realtime enabled.
-
- Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a
-
-2015-12-08 13:04 +0000 [fe8011cc50] sungtae kim <pchero21@gmail.com>
-
- * AMI: Fixed OriginateResponse message
-
- When the asterisk sending OriginateResponse message,
- it doesn't set the "Uniqueid".
- And it didn't support correct response message for
- Application originate.
-
- ASTERISK-25624 #close
-
- Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d
-
-2015-12-15 18:01 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.7.0-rc1 Released.
-
-2015-12-15 11:57 +0000 [0370acecfc] Kevin Harwell <kharwell@lunkwill>
-
- * Release summaries: Add summaries for 13.7.0-rc1
-
-2015-12-15 11:54 +0000 [d1bb33fe0b] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:54 +0000 [d06a65de01] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:54 +0000 [fb37b44660] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-15 11:48 +0000 [20b7164b8c] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:48 +0000 [6cbf2414c3] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:48 +0000 [ba1794464d] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-15 11:39 +0000 [b3e9753a23] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:39 +0000 [b0df64b5f0] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:39 +0000 [ce9a59faf6] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-15 11:28 +0000 [2e26bef5bb] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:28 +0000 [5e9b47516d] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:28 +0000 [034112c574] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-15 11:19 +0000 [d1f8ff1789] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:19 +0000 [9376488bef] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:19 +0000 [a894c9e7a9] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-15 11:12 +0000 [52afb0f112] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.7.0-rc1
-
-2015-12-15 11:12 +0000 [2de343eb85] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.7.0-rc1
-
-2015-12-15 11:12 +0000 [184de2a160] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.7.0-rc1
-
-2015-12-14 13:53 +0000 [24ae124e4f] server-pandora <server-pandora@xencall.com>
-
- * res_rtp_asterisk.c: Fix DTLS negotiation delays.
-
- - Trigger pending DTLS packets to send out, once the RTP instance's remote
- address is set.
- - Avoids locking the DTLS structure unnecessarily by only doing this if
- DTLS is passive.
- - Add DTLS locks around the structurally sensitive calls in the SSL
- portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock
- inside of itself, and we're dealing with the SSL BIO in at least two
- threads.
-
- WebRTC channels may receive a DTLS handshake before
- ast_rtp_remote_address_set is called, which causes there to be a pending
- response to send out. Previous to 1ad827, this was handled by calling
- dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP
- packet could trigger the pending handshake response. Since that was
- rightfully removed, whenever the DTLS handshake is received before the
- remote address is set, we would have to wait until another SSL packet
- arrives.
-
- As of Chrome M47's optimizations to their handshake process, WebRTC
- conversations between Chrome M47+ and Asterisk, where Asterisk is passive,
- experience a 1 second delay without this patch, because the SSL handshake
- is received before ICE negotation stores the remote_address, and the next
- SSL packet isn't received until after a 1 second timeout in Chrome, which
- causes a new handshake request.
-
- ASTERISK-25614 #close
-
- Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908
-
-2015-12-14 15:25 +0000 [36097a185d] Richard Mudgett <rmudgett@digium.com>
-
- * Fix sscanf() format string type mismatch.
-
- ASTERISK-25615
- Reported by: George Joseph
-
- Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b
-
-2015-12-13 13:13 +0000 [94f9927784] Matt Jordan <mjordan@digium.com>
-
- * main/utils: Don't emit an ERROR message if the read end of a pipe closes
-
- An ERROR or WARNING message should generally indicate that something has gone
- wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not
- in control of when the far end closes its reading on a file descriptor. If the
- far end does close the file descriptor in an unclean fashion, this isn't a bug
- or error in Asterisk, particularly when the situation can be gracefully
- handled in Asterisk.
-
- Currently, when this happens, a user would see the following somewhat cryptic
- ERROR message:
-
- "utils.c: write() returned error: Broken pipe"
-
- There's a few problems with this:
- (1) It doesn't provide any context, other than 'something broke a pipe'
- (2) As noted, it isn't actually an error in Asterisk
- (3) It can get rather spammy if the thing breaking the pipe occurs often, such
- as a FastAGI server
- (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even
- mask legitimate issues
-
- This patch changes ast_carefulwrite to only log an ERROR if we actually had one
- that was reasonably under our control. For debugging purposes, we still emit
- a debug message if we detect that the far side has stopped reading.
-
- Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566
-
-2015-12-12 11:08 +0000 [5b867fa904] gtjoseph <george.joseph@fairview5.com>
-
- * pjsip/config_transport: Check pjproject version at runtime for async ops
-
- pjproject < 2.5.0 will segfault on a tls transport if async_operations
- is greater than 1. A runtime version check has been added to throw
- an error if the version is < 2.5.0 and async_operations > 1.
-
- To assist in the check, a new api "ast_compare_versions" was added
- to utils which compares 2 major.minor.patch.extra version strings.
-
- ASTERISK-25615 #close
-
- Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98
- Reported-by: George Joseph
- Tested-by: George Joseph
-
-2015-12-10 11:44 +0000 [14b41115e3] Jonathan Rose <jrose@digium.com>
-
- * chan_sip: Add TCP/TLS keepalive to TCP/TLS server
-
- Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously
- this option was only being set on session sockets.
- http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/
- According to the link above, the SO_KEEPALIVE option is useful for knowing
- when a TCP connected endpoint has severed communication without indicating
- it or has become unreachable for some reason. Without this patch, keep
- alive is not set on the socket listening for incoming TCP sessions and
- in Komatsu's report this resulted in the thread listening for TCP becoming
- stuck in a waiting state.
-
- ASTERISK-25364 #close
- Reported by: Hiroaki Komatsu
-
- Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36
-2015-12-09 09:48 +0000 [cd119ed4a2] Tyler Cambron <tcambron@digium.com>
-
- * res_chan_stats: Fix bug to send correct statistics to StatsD
-
- Fixed a bug that originally would show a negative number of
- active calls occuring in Asterisk. A gauge is persistent so
- incrementing and decrementing it results in a more consistent
- performance. Also changed to the call to StatsD to use
- ast_statsd_log_string() so that a "+" could be sent to StatsD.
-
- ASTERISK-25619 #close
-
- Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7
-
-2015-12-07 13:07 +0000 [ddf4dddf4f] Corey Farrell <git@cfware.com>
-
- * app_meetme: Set default value for audio_buffers.
-
- The default value was never set for audio_buffers, causing bad
- audio quality. This ensures the default is always set.
-
- ASTERISK-25569 #close
-
- Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44
-2015-12-08 01:57 +0000 [142d4fefb8] Filip Jenicek <phill@janevim.cz>
-
- * chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c)
-
- Asterisk may crash when calling ast_channel_get_t38_state(c)
- on a locked channel which is being hung up.
-
- ASTERISK-25609 #close
-
- Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b
-
-2015-12-08 17:49 +0000 [21962dad93] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add existence and readablity checks for tls related files
-
- Both transport and endpoint now check for the existence and readability
- of tls certificate and key files before passing them on to pjproject.
- This will cause the object to not load rather than waiting for pjproject
- to discover that there's a problem when a session is attempted.
-
- NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located
- in build_peer which is gigantic and I didn't want to disturb it.
- Error messages will emit but it won't interrupt chan_sip loading.
-
- ASTERISK-25618 #close
-
- Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9
- Reported-by: George Joseph
- Tested-by: George Joseph
-
-2015-12-02 12:42 +0000 [28d9243079] Eugene Voityuk <eugene@thirdlane.com>
-
- * chan_sip.c: Start ICE negotiation when response is sent or received.
-
- The current logic for ICE negotiation starts it
- when receiving an SDP with ICE candidates. This is
- incorrect as ICE negotiation can only start when each
- call party have at least one pair of local and remote
- candidate. Starting ICE negotiation early would result
- in negotiation failure and ultimately no audio.
-
- This change makes it so ICE negotiation is only started
- when a response with SDP is received or when a response
- with SDP is sent.
-
- ASTERISK-24146
-
- Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca
-2015-12-08 11:03 +0000 [e03582a1c2] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls
-
- See ASTERISK-25615.
- If the transport protocol is tls and async_operations > 1, pjproject
- will segfault if more than one operation is attempted on the same socket.
- Until this is fixed upstream, a check has been added to throw an error
- if a tls transport config has async_operations set to > 1.
-
- ASTERISK-25615
-
- Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6
- Reported-by: George Joseph
- Tested-by: George Joseph
-
-2015-12-08 08:39 +0000 [876600ce6e] Alexander Traud <pabstraud@compuserve.com>
-
- * codec_resample: Increase buffer for Opus Codec with FEC.
-
- ASTERISK-25599 #close
-
- Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e
-
-2015-12-08 03:46 +0000 [69e3d40ad7] Alexander Traud <pabstraud@compuserve.com>
-
- * translate: Avoid a warning message when doing FEC within Opus Codec.
-
- ASTERISK-25616 #close
-
- Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319
-
-2015-12-04 15:36 +0000 [2b992014dc] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip: Fix crash involving the bogus peer during sip reload.
-
- A crash happens sometimes when performing a CLI "sip reload". The bogus
- peer gets refreshed while it is in use by a new call which can cause the
- crash.
-
- * Protected the global bogus peer object with an ao2 global object
- container.
-
- ASTERISK-25610 #close
-
- Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed
-
-2015-12-06 16:32 +0000 [529535f0c2] Matt Jordan <mjordan@digium.com>
-
- * Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state"
-
- This reverts commit 6614babea27fbafbe11820ea03737dd5c4f9ecec.
-
- Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks
- in core_unreal/chan_local. Local channels attempt to reach across both their
- peer and the peer's bridge to inspect T.38 state. Given the propensity of
- Local channel chains, managing the locking situation in such a scenario is
- practically infeasible.
-
- Change-Id: Ic687397ffea08dfb899345a443bd990ec3d0416a
-
-2015-12-04 16:23 +0000 [450579e908] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/contacts/statsd: Make contact lifecycle events more consistent
-
- It will never be perfect or even pretty, mostly because of the differences
- between static and dynamic contacts.
-
- Created:
-
- Can't use the contact or contact_status alloc functions
- because the objects come and go regardless of the actual state.
-
- Can't use the contact_apply_handler, ast_sip_location_add_contact or
- a sorcery created handler because they only get called for dynamic
- contacts. Similarly, permanent_uri_handler only gets called for
- static contacts.
-
- So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is
- the only place it can go and not have duplicated code. Both
- permanent_uri_handler and contact_apply_handler call find_or_create.
-
- Removed:
-
- Can't use the destructors for the same reason as above. The only
- place to put this is in persistent_endpoint_contact_deleted_observer
- which I believe is the "correct" place but even that will handle only
- dynamic contacts. This doesn't called on shutdown however. There is
- no hook to use for static contacts that may be removed because of a
- config change while asterisk is in operation.
-
- I moved the cleanup of contact_status from ast_sip_location_delete_contact
- to the handler as well.
-
- Status Change and RTT:
-
- Although they worked fine where they were (in update_contact_status) I
- moved them to persistent_endpoint_contact_status_observer to make it
- more consistent with removed. There was logic there already to detect
- a state change.
-
- Finally, fixed a nit in permanent_uri_handler rmudgett reported
- eralier.
-
- ASTERISK-25608 #close
-
- Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d
- Reported-by: George Joseph
- Tested-by: George Joseph
-
-2015-11-21 06:02 +0000 [5a18193dc0] Alexander Traud <pabstraud@compuserve.com>
-
- * res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.
-
- ASTERISK-25584 #close
-
- Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91
-
-2015-11-21 05:21 +0000 [3e2178c05e] Alexander Traud <pabstraud@compuserve.com>
-
- * res_format_attr_opus: Update to latest RFC 7587.
-
- Beside that, the format-attribute module sends only non-default values in the
- line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore,
- previously the parameter stereo was not parsed when being the first parameter.
-
- ASTERISK-25583 #close
-
- Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73
-2015-12-02 14:11 +0000 [072d94183c] Jonathan Rose <jrose@digium.com>
-
- * Fix crash in audiohook translate to slin
-
- This patch fixes a crash which would occur when an audiohook was
- applied to a channel using an audio codec that could not be translated
- to signed linear (such as when using pass-through codecs like OPUS or
- when the codec translator module for the format in use is not loaded).
-
- ASTERISK-25498 #close
- Reported by: Ben Langfeld
-
- Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384
-
-2015-12-03 12:07 +0000 [9184fbeb34] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Use a MD5 hash for static Contact IDs
-
- When 90d9a70789 was merged, it mostly tested dynamic contacts created as
- a result of registering a PJSIP endpoint. Contacts generated in this
- fashion typically have a long alphanumeric string as their object identifier,
- which maps reasonably well for StatsD. Unfortunately, this doesn't work in the
- general case. StatsD treats both '.' and ':' characters as special characters.
- In particular, having a ':' appear in the middle of a StatsD metric will
- result in the metric being rejected.
-
- This causes some obvious issues with SIP URIs.
-
- The StatsD API should not be responsible for escaping the metric name passed
- to it. The metric is treated as a single long string, and it would be
- challenging to know what to escape in the string passed to the function.
- Likewise, we don't want to escape the metric in PJSIP, as that involves
- overhead that is wasted when either res_statsd isn't loaded or enabled.
-
- This patch takes an alternative approach. The Contact ID has been changed
- to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the
- aforementioned special characters, (b) can be done on Contact creation,
- which has minimal impact on run-time performance, and (c) also conforms to an
- earlier commit that changed the ID for dynamic contacts.
-
- The downside of this is that StatsD users will have to map SHA1 hashes back to
- the Contacts that are emitting the statistics. To that end, the CLI commands
- have been updated to include the first 10 characters of the MD5 hash, which
- should be enough to match what is shown in Graphite (or some other StatsD
- backend).
-
- ASTERISK-25595 #close
-
- Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2
- Reported-by: Matt Jordan
- Tested-by: George Joseph
-
-2015-11-30 22:19 +0000 [ed9134282e] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Update logging to show contact->uri in messages
-
- An earlier commit changed the id of dynamic contacts to contain
- a hash instead of the uri. This patch updates status change
- logging to show the aor/uri instead of the id. This required
- adding the aor id to contact and contact_status and adding
- uri to contact_status. The aor id gets added to contact and
- contact_status in their allocators and the uri gets added to
- contact_status in pjsip_options when the contact_status is
- created or updated.
-
- ASTERISK-25598 #close
-
- Reported-by: George Joseph
- Tested-by: George Joseph
-
- Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511
-
-2015-12-01 16:11 +0000 [eadad24b59] Jonathan Rose <jrose@digium.com>
-
- * Unset BRIDGEPEER when leaving a bridge
-
- Currently if a channel is transferred out of a bridge, the BRIDGEPEER
- variable (also BRIDGEPVTCALLID) remain set even once the channel is
- out of the bridge. This patch removes these variables when leaving
- the bridge.
-
- ASTERISK-25600 #close
- Reported by: Mark Michelson
-
- Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da
-
-2015-11-30 14:22 +0000 [bb0b60619d] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Fix off nominal ref leak.
-
- Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49
-
-2015-11-30 16:42 +0000 [e7c88e11aa] Richard Mudgett <rmudgett@digium.com>
-
- * sched.c: Make not return a sched id of 0.
-
- According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0
- was never returned historically and several users incorrectly coded usage
- of the returned sched ID assuming that 0 was invalid.
-
- ASTERISK-25476
-
- Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20
-
-2015-11-25 12:23 +0000 [4aed349a7b] Richard Mudgett <rmudgett@digium.com>
-
- * Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions)
-
- chan_sip.c:
- * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to
- ao2 conversion.
-
- * Initialize register scheduler ids earlier because of ASTOBJ to ao2
- conversion.
-
- chan_skinny.c:
- * Fix more scheduler usage for the valid 0 id value.
-
- ASTERISK-25476
-
- Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95
-
-2015-11-24 12:44 +0000 [6d9156d10f] Richard Mudgett <rmudgett@digium.com>
-
- * Audit improper usage of scheduler exposed by 5c713fdf18f.
-
- channels/chan_iax2.c:
- * Initialize struct chan_iax2_pvt scheduler ids earlier because of
- iax2_destroy_helper().
-
- channels/chan_sip.c:
- channels/sip/config_parser.c:
- * Fix initialization of scheduler id struct members. Some off nominal
- paths had 0 as a scheduler id to be destroyed when it was never started.
-
- chan_skinny.c:
- * Fix some scheduler id comparisons that excluded the valid 0 id.
-
- channel.c:
- * Fix channel initialization of the video stream scheduler id.
-
- pbx_dundi.c:
- * Fix channel initialization of the packet retransmission scheduler id.
-
- ASTERISK-25476
-
- Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8
-
-2015-12-01 07:55 +0000 [b76c196e13] Alexander Traud <pabstraud@compuserve.com>
-
- * codec_resample: Increase buffer for Opus Codec.
-
- ASTERISK-25599 #close
-
- Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10
-
-2015-11-28 08:46 +0000 [6614babea2] Matt Jordan <mjordan@digium.com>
-
- * bridges/bridge_t38: Add a bridging module for managing T.38 state
-
- When 4875e5ac32 was merged, it fixed several issues with a direct media bridge
- transitioning to handling a T.38 fax. However, it uncovered a race condition
- caused by the bridging core. When a channel involved in a T.38 fax leaves a
- bridge, the frame queued by the channel driver that should inform the far side
- that it is no longer in a T.38 fax may not make it across the bridge. The
- bridging framework is *extremely* aggressive in tearing down the bridge, and
- control frames that are currently in flight *may* get dropped.
-
- This patch adds a new module to the bridging framework, bridge_t38. This module
- maintains some notion of the T.38 state for the two channels in a bridge. When
- the bridge detects that it is being torn down or when one of the two channels
- leaves, it informs the respective channel(s) that they should stop faxing. This
- ensures that channels switch back to audio if they survive and are ejected out
- of a bridge while faxing.
-
- ASTERISK-25582
-
- Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0
-
-2015-11-27 07:39 +0000 [3fcf160fae] Niklas Larsson <niklas@tese.se>
-
- * CHANGES: Fix a typo
-
- Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7
-2015-11-25 15:26 +0000 [45efbf8503] Kevin Harwell <kharwell@digium.com>
-
- * fastagi: record file closed after sending result
-
- The fastagi record-file testsuite test sometimes fails reporting an empty
- recorded file. This was happening because Asterisk was sending the agi result
- notification prior to actually closing the file and the data, being buffered,
- had not been written to the file yet when the test attempts to check the file
- size.
-
- This patch makes it so the record file stream is closed prior to sending the
- agi result notification.
-
- ASTERISK-25593 #close
-
- Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde
-
-2015-11-25 13:29 +0000 [b2787876d6] Walter Doekes <walter+asterisk@wjd.nu>
-
- * main: Slight refactor of main. Improve color situation.
-
- Several issues are addressed here:
- - main() is large, and half of it is only used if we're not rasterisk;
- fixed by spliting up the daemon part into a separate function.
- - Call ast_term_init from rasterisk as well.
- - Remove duplicate code reading/writing asterisk history file.
- - Attempt to tackle background color issues and color changes that
- occur. Tested by starting asterisk -c until the colors stopped
- changing at odd locations.
-
- ASTERISK-25585 #close
-
- Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f
-
-2015-11-24 13:54 +0000 [59881fbb99] David M. Lee <dlee@respoke.io>
-
- * Fixed some typos
-
- Fixes some minor typos in the CHANGES file, plus an embarrasing typo in
- the StatsD API.
-
- Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7
-
-2015-11-24 13:07 +0000 [b75f587d15] Corey Farrell <git@cfware.com>
-
- * res_pjsip_notify: Fix CLI usage info
-
- The usage info for 'pjsip send notify' previously referenced the
- chan_sip configuration sip_notify.conf. Fix this to reference
- the correct configuration pjsip_notify.conf.
-
- ASTERISK-25590 #close
-
- Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea
-
-2015-11-23 14:27 +0000 [fc45f4040d] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_realtime.c: Fix crash from NULL sorcery object type.
-
- If the sorcery object type is not found a NULL is returned.
- Unfortunately, sorcery_realtime_filter_objectset() will crash after
- complaining about not finding the object type and saying to expect errors.
-
- * Use ao2_cleanup() instead of ao2_ref() to prevent the crash.
-
- ASTERISK-25165
- Reported by Corey Farrell
-
- Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97
-
-2015-11-20 21:08 +0000 [4875e5ac32] Matt Jordan <mjordan@digium.com>
-
- * chan_pjsip: Handle T.38 faxes with direct media bridges
-
- When a channel is in a direct media bridge, a re-INVITE may arrive that forces
- Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
- must change its technology to a simple bridge, and re-INVITE the media back
- to Asterisk.
-
- Generally, this logic mostly already exists in Asterisk. However, prior to this
- patch, there were a few bugs:
- (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
- ever entering into a direct media bridge. This applies even when the only
- media being passed over the channel is audio. This patch fixes this bug
- by having the framehook specify that it defers caring about any frame type.
- This allows the channels to enter into a direct media bridge, which will
- be broken when a re-INVITE is received.
- (2) When a re-INVITE is received, nothing instructed the bridging layer to
- re-inspect the allowed bridging technology. This now occurs when either
- a re-INVITE is received from a peer, or when a response is received from
- the far end (that is, when the T.38 state changes to either
- T38_PEER_REINVITE or T38_LOCAL_REINVITE).
- (3) chan_pjsip needs to do a small amount of work to prevent a direct media
- bridge from being chosen when a T.38 session is in progress. When a T.38
- session supplement has a t38 datastore - which is added when we detect
- we should start thinking about T.38 on a channel - we now refuse a native
- RTP bridge.
- (4) When a BYE request is received, we don't terminate the T.38 session. If
- the other side of a T.38 fax survives the hangup (due to the 'g' flag
- in Dial, for example), we don't currently re-INVITE the media on the
- other channel back to audio. This patch now has res_pjsip_t38 intercept
- BYE requests and inform the far side that the T.38 session is terminated.
- This naturally causes the correct re-INVITEs to be sent.
-
- ASTERISK-25582
-
- Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb
-
-2015-11-20 21:07 +0000 [2b94d9a10d] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_t38: Add debug statements
-
- This patch adds some debug statements to res_pjsip_t38. These statements help
- to determine which SDP negotiation callbacks are being executed, and, when
- a particular callback exits, why a callback may not have applied its logic
- to the local or remote SDP.
-
- Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77
-
-2015-10-22 09:44 +0000 [af288b2d96] Matt Jordan <mjordan@digium.com>
-
- * main/cli: Use proper string methods to check existence of context/exten/app
-
- Because the context, extension, and application are stored in stringfields,
- checking for them being NULL doesn't work so well. This patch uses the
- appropriate string library call, ast_strlen_zero, to see if there is a value
- in the context/exten/app values.
-
- Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23
-
-2015-11-18 09:43 +0000 [d27aac0a9d] Matt Jordan <mjordan@digium.com>
-
- * res/res_endpoint_stats: Add module to emit endpoint StatsD statistics
-
- This patch adds a module that emits StatsD statistics about Asterisk
- endpoints. This includes:
- * A GUAGE statistic for endpoint states, tracking how many endpoints are in
- a particular state.
- * A GUAGE statistic for each endpoint, counting the number of channels
- currently associated with an endpoint.
-
- ASTERISK-25572
-
- Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305
-
-2015-11-18 10:07 +0000 [90d9a70789] Matt Jordan <mjordan@digium.com>
-
- * res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts
-
- This patch adds the ability to send StatsD statistics related to the
- state of PJSIP contacts. This includes:
- * A GUAGE statistic measuring the count of contacts in a particular state.
- This measures how many contacts are reachable, unreachable, etc.
- * The RTT time for each contact, if those contacts are qualified. This
- provides StatsD engines useful time-based data about each contact.
-
- ASTERISK-25571
-
- Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c
-
-2015-11-13 10:34 +0000 [75097a0955] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_outbound_registration: Add registration statistics for StatsD
-
- This patch adds outbound registration statistics for StatsD. This includes
- the following:
- * A GUAGE metric for the overall count of outbound registrations.
- * A GUAGE metric for each state an outbound registration can be in. As the
- outbound registrations change state, the overall count of how many
- outbound registrations are in the particular state is changed.
-
- These statistics are particularly useful for systems with a large number of
- SIP trunks, and where measuring the change in state of the trunks is useful
- for monitoring.
-
- ASTERISK-25571
-
- Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37
-
-2015-11-19 09:40 +0000 [8f71263e72] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_outbound_registration: Apply configuration on object type load
-
- When Asterisk is configured to use a dynamic sorcery backend (such as
- res_sorcery_astdb) with 'registration' objects, it will fail to create the
- internal state objects associated with the registration objects on module
- load. This is due to nothing actually querying for the specific objects
- and calling their sorcery apply handler during module load.
-
- This patch fixes that by calling get_registrations in the sorcery observer's
- object_type_loaded handler. Doing this causes the sorcery backends to be
- asked for the current state of all registration objects, which causes the
- apply handler to be called and the internal run-time state to be created.
-
- ASTERISK-25575 #close
-
- Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23
-
-2015-11-11 11:51 +0000 [0b508789ab] Alexander Traud <pabstraud@compuserve.com>
-
- * translate: Provide translation modules the result of SDP negotiation.
-
- Previously, a trancoding module did not have access to the joint but cached
- format. Therefore, the module did not have access to the attributes negotiated
- via SDP (line fmtp). Now, a translation module receives the joint format.
-
- ASTERISK-25545 #close
-
- Change-Id: Id6878a989b50573298dab115d3371ea369e1a718
-
-2015-11-19 01:14 +0000 [1aa552b2a2] Alexander Traud <pabstraud@compuserve.com>
-
- * res_format_attr_h264: Do not reset string buffer.
-
- When no parameter is present, Asterisk does not generate the line fmtp, as
- expected. However, because a buffer was reset, even rtpmap and fmtp of previous
- media codecs got removed. Now, Asterisk does not reset other codecs in case of
- no parameter for H.264.
-
- ASTERISK-25573 #close
-
- Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286
-
-2015-11-18 10:05 +0000 [3354b325c6] Matt Jordan <mjordan@digium.com>
-
- * res_statsd: Add functions that support variable arguments
-
- Often, the metric names of statistics we are generating for StatsD have some
- dynamic component to them. This can be the name of a particular resource, or
- some internal status label in Asterisk. With the current set of functions,
- callers of the statsd API must first build the metric name themselves, then
- pass this to the API functions. This results in a large amount of boilerplate
- code and usage of either fixed length static buffers or dynamic memory
- allocation, neither of which is desireable.
-
- This patch adds two new functions to the StatsD API that support a printf
- style format specifier for constructing the metric name. A dynamic string,
- allocated in threadstorage, is used to build the metric name. This eases
- the burden on users of the StatsD API.
-
- Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea
-
-2015-11-17 14:53 +0000 [d4a522d587] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts.
-
- Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d
-
-2015-11-17 14:53 +0000 [e44ab3816c] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Fix 423 response handling.
-
- Receiving a 423 Interval Too Brief response after authentication for an
- outbound registration attempt results in assuming that the registrar has
- rejected the registration permanently. If there are no configured retries
- for fatal responses then the outbound registration is stopped for that
- endpoint.
-
- For registrations, PJSIP/PJPROJECT intercepts the handling of 423
- responses and does not include any authentication in the updated
- registration request. When the updated request is challenged then the
- Asterisk code assumes that we were challenged again because the peer
- rejected the authentication we sent earlier.
-
- * Made registration challenges keep track of the CSeq number to determine
- if the received challenge response was for the request we thought we sent.
- If the response's CSeq number differs from the CSeq number we last sent
- with authentication then authenticate again because it is a challenge to a
- different request.
-
- Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09
-
-2015-11-03 14:36 +0000 [1e0040b88f] Tyler Cambron <tcambron@digium.com>
-
- * StatsD: Add res_statsd compatibility
-
- Added a new api to res_statsd.c to allow it to receive a
- character pointer for the value argument. This allows for a
- '+' and a '-' to easily be sent with the value.
-
- ASTERISK-25419
- Reported By: Ashley Sanders
-
- Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611
-
-2015-11-16 13:56 +0000 [f62b642fe3] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip: Fix off nominal crash with requests that fail and have a timer
-
- When a request is sent using pjsip_endpt_send_request and fails, a condition
- exists where the request wrapper, which is an AO2 object, may be de-ref'd
- more times than it should. This occurs when the request's callback is called,
- and, in the callback, the timer on the PJSIP heap is cancelled. When that
- occurs, the request wrapper's lifetime is decremented. When
- pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of
- the request wrapper again, even though we've already cancelled the reference
- associated with the timer.
-
- This patch checks the return result of pj_timer_heap_cancel_if_active before
- removing the reference associated with the timer. We now only decrement it
- in this case if a timer is cancelled as a result of the function call.
-
- Change-Id: I21332343a1a019c1117076f9bf2df27be2850102
-
-2015-11-13 14:03 +0000 [fdd2afcd16] Mark Michelson <mmichelson@digium.com>
-
- * Confbridge: Add a user timeout option
-
- This option adds the ability to specify a timeout, in seconds, for a
- participant in a ConfBridge. When the user's timeout has been reached,
- the user is ejected from the conference with the CONFBRIDGE_RESULT
- channel variable set to "TIMEOUT".
-
- The rationale for this change is that there have been times where we
- have seen channels get "stuck" in ConfBridge because a network issue
- results in a SIP BYE not being received by Asterisk. While these
- channels can be hung up manually via CLI/AMI/ARI, adding some sort of
- automatic cleanup of the channels is a nice feature to have.
-
- ASTERISK-25549 #close
- Reported by Mark Michelson
-
- Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
-
-2015-11-16 04:29 +0000 [7debb986a5] Alec Davis <sivad.a@paradise.net.nz>
-
- * app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!
-
- commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525)
- refer ASTERISK-24958
-
- above commit removed ast_channel_lock(qe->chan);
- but failed to remove corresponding ast_channel_unlock(qe->chan);
-
- ASTERISK-25561 #close
- Reported Alec Davis
-
- Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a
-
-2015-11-14 07:02 +0000 [afd9a89e5a] Joshua Colp <jcolp@digium.com>
-
- * hashtab: Add NULL check when destroying iterator.
-
- The hashtab API is pretty NULL tolerant which has resulted
- in remaining callers not doing much checks themselves.
- Unfortunately the function to destroy an iterator does not
- do a NULL check and will result in a crash if passed NULL.
- This change fixes that.
-
- ASTERISK-25552 #close
-
- Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619
-
-2015-11-13 14:32 +0000 [c0f2f8de45] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_rfc3326.c: Fix crash when channel goes away.
-
- If an authenticated incoming caller does not respond to our 200 OK INVITE
- response with an ACK then PJSIP will hangup the call. Unfortunately,
- there is a chance that the session's channel will go away between one use
- of the channel pointer and another when building the BYE request because
- the BYE is being built by the monitor thread and not the call's serializer
- thread.
-
- * Added a check to ensure that the thread trying to add the Reason header
- is the call's serializer thread. This ensures that the channel will not
- go away on us.
-
- Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89
-
-2015-11-13 14:19 +0000 [4f43b85c92] Mark Michelson <mmichelson@digium.com>
-
- * Taskprocessors: Increase high-water mark
-
- In practical tests, we have seen certain taskprocessors, specifically
- Stasis subscription taskprocessors, cross the recently-added high-water
- mark and emit a warning. This high-water mark warning is only intended
- to be emitted when things have tanked on the system and things are
- heading south quickly. In the practical tests, the Stasis taskprocessors
- sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in
- any danger at all.
-
- As such, this ups the high-water mark to 500 tasks instead. It also
- redefines the SIP threadpool request denial number to be a multiple of
- the taskprocessor high-water mark.
-
- Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce
-
-2015-11-11 11:46 +0000 [d8d3991390] Alexander Traud <pabstraud@compuserve.com>
-
- * format: Register format-attribute module with cached formats.
-
- In Asterisk 13, cached formats are created before their corresponding format-
- attribute module is registered. Cached formats are involved when a local
- extension is called. Therefore, ast_format_generate_sdp_fmtp did not work
- on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264,
- and format-attribute modules provided externally.
-
- ASTERISK-25160 #close
-
- Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354
-
-2015-11-12 11:17 +0000 [367972e42d] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip distributor: Don't send 503 response to responses.
-
- When the SIP threadpool is backed up with tasks, we send 503 responses
- to ensure that we don't try to overload ourselves. The problem is that
- we were not insuring that we were not trying to send a 503 to an
- incoming SIP response.
-
- This change makes it so that we only send the 503 on incoming requests.
-
- Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404
-
-2015-11-11 17:11 +0000 [2f9cb7d62b] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Deny requests when threadpool queue is backed up.
-
- We have observed situations where the SIP threadpool may become
- deadlocked. However, because incoming traffic is still arriving, the SIP
- threadpool's queue can continue to grow, eventually running the system
- out of memory.
-
- This change makes it so that incoming traffic gets rejected with a 503
- response if the queue is backed up too much.
-
- Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
-
-2015-11-12 06:24 +0000 [4e5bf12b33] Joshua Colp <jcolp@digium.com>
-
- * format_cap: Don't append the 'none' format when appending all.
-
- When appending all formats of a type all the codecs are iterated
- and added. This operation was incorrectly adding the ast_format_none
- format which is special in that it is supposed to be used when no
- format is present. It shouldn't be appended.
-
- ASTERISK-25535
-
- Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c
-
-2015-11-11 04:16 +0000 [07583c2888] Steve Davies <steve@one47.co.uk>
-
- * Further fixes to improper usage of scheduler
-
- When ASTERISK-25449 was closed, a number of scheduler issues mentioned in
- the comments were missed. These have since beed raised in ASTERISK-25476
- and elsewhere.
-
- This patch attempts to collect all of the scheduler issues discovered so
- far and address them sensibly.
-
- ASTERISK-25476 #close
-
- Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b
-
-2015-11-11 11:04 +0000 [b818d70533] Joshua Colp <jcolp@digium.com>
-
- * threadpool: Handle worker thread transitioning to dead when going active.
-
- This change adds handling of dead worker threads when moving them
- to be active. When this happens the worker thread is removed from
- both the active and idle threads container. If no threads are able
- to be moved to active then the pool grows as configured.
-
- A unit test has also been added which thrashes the idle timeout
- and thread activation to exploit any race conditions between the
- two.
-
- ASTERISK-25546 #close
-
- Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143
-
-2015-11-10 09:27 +0000 [4bf84459c7] Alexander Traud <pabstraud@compuserve.com>
-
- * rtp_engine: Init a format-attribute module to its RFC defaults.
-
- Previously, format-attribute modules relied on an existing fmtp line in SDP
- negotiation. However, fmtp is optional for several formats like the Opus Codec.
- Now, the format-attribute module is called with an empty fmtp, which allows the
- module to initialise itself to RFC defaults. Furthermore now, Asterisk is able
- to differentiate between internally and externally created formats.
-
- ASTERISK-25537 #close
-
- Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52
-
-2015-11-09 03:04 +0000 [1bff400df7] Alexander Traud <pabstraud@compuserve.com>
-
- * ast_format_cap_get_names: To display all formats, the buffer was increased.
-
- ASTERISK-25533 #close
-
- Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a
-
-2015-11-09 07:04 +0000 [f3ac4d8090] Alexander Traud <pabstraud@compuserve.com>
-
- * ast_format_cap: Avoid format creation on module load, use cache instead.
-
- Since Asterisk 13, formats are immutable and cached. However while loading a
- module like chan_sip, some formats were created instead using cached ones.
-
- ASTERISK-25535 #close
-
- Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b
-
-2015-11-06 07:54 +0000 [6d1bdb9d3b] Walter Doekes <walter+asterisk@wjd.nu>
-
- * func_callerid: Document that CALLERID(pres) is available.
-
- CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres)
- and CALLERID(name-pres). But for channel driver that don't make a
- distinction between the two (e.g. SIP), it makes more sense to get/set
- both at once. This change reveals the availability of CALLERID(pres),
- CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and
- REDIRECTING(from-pres).
-
- ASTERISK-25373 #close
-
- Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a
-2015-11-06 07:52 +0000 [8410336681] Walter Doekes <walter+asterisk@wjd.nu>
-
- * docs: Fix a few typo's in app docs (more then, resourse).
-
- Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7
-
-2015-11-06 07:36 +0000 [0d425f2eb4] Walter Doekes <walter+asterisk@wjd.nu>
-
- * xmldoc: Improve xmldoc wrapping of 'core show ...' output.
-
- Previously, the wrapping did both lookahead and lookback, which,
- together with color escape sequences, caused some lines to be wrapped
- way earlier than other lines. This led to inconsistent output.
-
- This simplifies the wrapping code and makes it more sane: if maxcolumns
- is hit, we simply jump back to the last space and wrap there.
-
- ASTERISK-25527 #close
-
- Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957
-
-2015-11-06 06:57 +0000 [33752e0837] Sean Bright (license #5060)
-
- * res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.
-
- In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual
- amount of channels is negotiated in-band. Therefore now, the Opus codec and its
- attribute rtpmap are registered with two channels.
-
- ASTERISK-24779 #close
- Reported by: PowerPBX
- Tested by: Alexander Traud
- patches:
- asterisk-24779.patch submitted by Sean Bright (license #5060)
-
- Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b
-
-2015-11-03 16:19 +0000 [6ff48319d9] Jonathan Rose <jrose@digium.com>
-
- * taskprocessor: Add high water mark warnings
-
- If a taskprocessor's queue grows large, this can indicate that there
- may be a problem with tasks not leaving the processor or else that
- the number of available task processors for a given type of task is
- too low. This patch makes it so that if a taskprocessor's task queue
- grows above 100 queued tasks that it will emit a warning message.
- Warning messages are emitted only once per task processor.
-
- ASTERISK-25518 #close
- Reported by: Jonathan Rose
-
- Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c
-
-2015-11-04 14:31 +0000 [506aea26e6] Matt Jordan <mjordan@digium.com>
-
- * main/dial: Protect access to the format_cap structure of the requesting channel
-
- When a dial attempt is made that involves a requesting channel, we previously
- were not:
- a) Protecting access to the native format capabilities structure on the
- requesting channel. That is inherently unsafe.
- b) Reference bumping the lifetime of the format capabilities structure.
-
- In both cases, something else could sneak in, blow away the format
- capabilities, and we'd be holding onto an invalid format_cap structure. When
- the newly created channel attempts to construct its format capabilities, things
- go poorly.
-
- This patch:
- a) Ensures that we get a reference to the native format capabilities while
- the requesting channel is locked
- b) Holds a reference to the native format capabilities during the creation
- of the new channel.
-
- ASTERISK-25522 #close
-
- Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f
-
-2015-10-30 22:57 +0000 [d098d00424] Corey Farrell <git@cfware.com>
-
- * Fix cli display of build options.
-
- A previous commit reduced the AST_BUILDOPTS compiler define to
- only include options that affected ABI. This included some options
- that were previously displayed by cli "core show settings". This
- change corrects the CLI display while still restricting buildopts.h
- to ABI effecting options only.
-
- ASTERISK-25434 #close
- Reported by: Rusty Newton
-
- Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325
-
-2015-11-03 11:15 +0000 [afec1b1b64] Matt Jordan <mjordan@digium.com>
-
- * res_pjsip/location: Destroy contact_status objects on contact deletion
-
- The contact_status Sorcery objects are currently not destroyed when a contact
- is deleted. This causes the contact's last known RTT/status to be 'sticky'
- when the contact itself may no longer exist. This patch causes the
- contact_status objects associated with both dynamic and static contacts to
- be destroyed if the AoR holding those contacts is also destroyed (or via
- other paths where a contact may be deleted.)
-
- Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e
-
-2015-11-03 10:58 +0000 [715f770c9f] Matt Jordan <mjordan@digium.com>
-
- * pjsip_configuration: On delete, remove the persistent version of an endpoint
-
- When an endpoint is deleted (such as through an API), the persistent endpoint
- currently continues to lurk around. While this isn't harmful from a memory
- consumption perspective - as all persistent endpoints are reclaimed on
- shutdown - it does cause Stasis endpoint related operations to continue
- to believe that the endpoint may or may not exist.
-
- This patch causes the persistent endpoint related to a PJSIP endpoint to be
- destroyed if the PJSIP endpoint is deleted.
-
- Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb
-2015-11-03 08:15 +0000 [f0f190af08] Matt Jordan <mjordan@digium.com>
-
- * main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field
-
- The JSON packing for the ContactStatusChange event forgot to include the
- roundtrip_usec field. As a result, the field never showed up in any event,
- even when the data was available. This patch corrects that error by properly
- packing the JSON blob with the data.
-
- Change-Id: I8df80da659a44010afbd48f645967518ff5daa17
-
-2015-11-02 20:24 +0000 [0393bd6bed] Corey Farrell <git@cfware.com>
-
- * chan_sip: Allow websockets to be disabled.
-
- This patch adds a new setting "websockets_enabled" to sip.conf.
- Setting this to false allows chan_sip to be used without causing
- conflicts with res_pjsip_transport_websocket.
-
- ASTERISK-24106 #close
- Reported by: Andrew Nagy
-
- Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7
-
-2015-11-02 17:19 +0000 [6fbffe42e1] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Set threadpool max size default to 50.
-
- During a stress test of subscriptions, a huge blast of
- subscription-related traffic resulted in the threadpool expanding to a
- ridiculous number of threads. The balooning of threads resulted in an
- increase of memory, which led to a crash due to being out of memory.
-
- An easy fix for the particular test was to limit the size of the
- threadpool, thus reining in the amount of memory that would be used. It
- was decided that there really is no downside to having a non-infinite
- default value for the maximum size of the threadpool, so this change
- introduces 50 threads as the maximum threadpool size for the SIP
- threadpool.
-
- ASTERISK-25513 #close
- Reported by John Bigelow
-
- Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be
-
-2015-11-02 06:57 +0000 [11e54b1932] Matt Jordan <mjordan@digium.com>
-
- * pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction
-
- When an AoR is created or destroyed dynamically, the scheduled OPTIONS
- requests that qualify the contacts on the AoR are not necessarily started
- or destroyed, particularly for persistent contacts created for that AoR.
- This patch adds create/update/delete sorcery observers for an AoR, which
- schedule/unschedule the qualifies as expected.
-
- Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d
-
-2015-10-30 13:22 +0000 [118d628e08] Matt Jordan <mjordan@digium.com>
-
- * Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs
-
- This patch adds a rule for installing the Super Awesome Company based 'Basic
- PBX' configuration files. As part of adding this rule, a bit of the content
- that makes up installing the configuration files under the 'samples' target
- was refactored into a make subroutine for usage by additional later config
- make targets.
-
- Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404
-2015-10-29 08:28 +0000 [9a021a42ad] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.
-
- When compiled with assertions enabled one will occur when destroying
- the subscription tree when UAS dialog creation fails. This is because
- the code assumes that a dialog will always exist on a subscription
- tree when in reality during this specific scenario it won't.
-
- This change makes it so a dialog is not removed from the subscription
- tree if it is not present.
-
- ASTERISK-25505 #close
-
- Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee
-
-2015-10-26 11:42 +0000 [1256aedf66] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Do not send all codecs on INVITE.
-
- Since version 13, Asterisk sent all allowed codecs as callee, even when the
- caller did not request/support them. In case of dynamic RTP payloads, this led
- to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the
- intersection between the requested and the supported codecs is send again.
-
- ASTERISK-24543 #close
-
- Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287
-
-2015-10-24 13:08 +0000 [5f593e7c38] gtjoseph <george.joseph@fairview5.com>
-
- * build: GCC 5.1.x catches some new const, array bounds and missing paren issues
-
- Fixed 1 issue in each of the affected files.
-
- ASTERISK-25494 #close
- Reported-by: George Joseph
- Tested-by: George Joseph
-
- Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77
-
-2015-10-20 16:02 +0000 [162acd45f7] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add "like" processing to pjsip list and show commands
-
- Add the ability to filter output from pjsip list and show commands
- using the "like" predicate like chan_sip.
-
- For endpoints, aors, auths, registrations, identifyies and transports,
- the modification was a simple change of an ast_sorcery_retrieve_by_fields
- call to ast_sorcery_retrieve_by_regex. For channels and contacts a
- little more work had to be done because neither of those objects are
- true sorcery objects. That was just removing the non-matching object
- from the final container. Of course, a little extra plumbing in the
- common pjsip_cli code was needed to parse the "like" and pass the regex
- to the get_container callbacks.
-
- Some of the get_container code in res_pjsip_endpoint_identifier was also
- refactored for simplicity.
-
- ASTERISK-25477 #close
- Reported by: Bryant Zimmerman
- Tested by: George Joseph
-
- Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1
-
-2015-10-21 11:51 +0000 [c58091737d] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip_outbound_registration: registration stops due to fatal 4xx response
-
- During outbound registration it is possible to receive a fatal (any permanent/
- non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due
- to a problem with the registrar itself. Upon receiving the failure response
- Asterisk terminates outbound registration for the given endpoint.
-
- This patch adds an option, 'fatal_retry_interval', that when set continues
- outbound registration at the given interval up to 'max_retries' upon receiving
- a fatal response.
-
- ASTERISK-25485 #close
-
- Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2
-
-2015-10-22 17:07 +0000 [ebe69dee0d] Mark Michelson <mmichelson@digium.com>
-
- * format_cap: Detect vector allocation failures.
-
- A crash was seen on a system that ran out of memory due to Asterisk not
- checking for vector allocation failures in format_cap.c. With this
- change, if either of the AST_VECTOR_INIT calls fail, we will return a
- value indicating failure.
-
- Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8
-
-2015-10-02 15:32 +0000 [3b19efefef] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.
-
- A certain situation can result in our attempting to send a NOTIFY on a
- destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but
- that subscriber has dropped off the network. We end up retransmitting
- that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY
- transaction. When the pjsip evsub code is told that the transaction has
- been terminated, it responds in kind by alerting us that the
- subscription has been terminated, destroying the subscription, and then
- removing its reference to the dialog, thus destroying the dialog.
-
- The problem is that when we get told that the subscription is being
- terminated, we detect that we have not sent a terminating NOTIFY
- request, so we queue up such a NOTIFY to be sent out. By the time that
- queued NOTIFY gets sent, the dialog has been destroyed, so attempting to
- send that NOTIFY can result in a crash.
-
- The fix being introduced here is actually a reintroduction of something
- the pubsub code used to employ. We hold a reference to the dialog and
- wait to decrement our reference to the dialog until our subscription
- tree object is destroyed. This way, we can send messages on the dialog
- even if the PJSIP evsub code wants to terminate earlier than we would
- like.
-
- In doing this, some NULL checks for subscription tree dialogs have been
- removed since NULL dialogs are no longer actually possible.
-
- Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5
-
-2015-09-29 14:53 +0000 [0a346f095f] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Ensure dialog lock balance.
-
- When sending a NOTIFY, we lock the dialog and then unlock the dialog
- when finished. A recent change made it so that the subscription tree's
- dialog pointer will be set NULL when sending the final NOTIFY request
- out. This means that when we attempt to unlock the dialog, we pass a
- NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog
- remains locked after we think we have unlocked it. When a response to
- the NOTIFY arrives, the monitor thread attempts to lock the dialog, but
- it cannot because we never released the dialog lock. This results in
- Asterisk being unable to process incoming SIP traffic any longer.
-
- The fix in this patch is to use a local pointer to save off the pointer
- value of the subscription tree's dialog when locking and unlocking the
- dialog. This way, if the subscription tree's dialog pointer is NULLed
- out, the local pointer will still have point to the proper place and the
- dialog lock will be unlocked as we expect.
-
- Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a
-
-2015-09-28 16:36 +0000 [ad39508095] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Prevent crashes on final NOTIFY.
-
- The SIP dialog is removed from the subscription tree when the final
- NOTIFY is sent. However, after the final NOTIFY is sent, the persistence
- update function still attempts to access the cseq from the dialog,
- resulting in a crash.
-
- This fix removes the subscription persistence at the same time that the
- dialog is removed from the subscription tree. This way, there is no
- attempt to update persistence when the subscription is being destroyed.
-
- Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb
-
-2015-09-17 17:28 +0000 [067f408760] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Remove serializer when sending final NOTIFY.
-
- There have been crashes seen where a taskprocessor's listener is NULL
- unexpectedly.
-
- Looking at backtraces, the problem was specifically seen in PJSIP
- serializers.
-
- Subscriptions make the mistake of removing a serializer from a dialog
- during subscription tree destruction. Since subscription trees are
- reference-counted, guaranteeing the circumstances behind the destruction
- are not possible. This makes it so that the dialog serializer can be
- removed while not holding the dialog lock. This makes it possible for
- the distributor to get a pointer to the dialog serializer and have that
- serializer get freed out from under it.
-
- The fix for this is to remove the serializer from a subscription dialog
- when sending the final NOTIFY. This guarantees that the serializer is
- removed with the dialog lock held. By doing this, we guarantee that if
- the distributor gains access to the dialog's serializer, it will not be
- possible for the serializer to get freed by another thread.
-
- Change-Id: I21f5dac33529f65cec45679bdace60670800ff66
-
-2015-09-02 09:14 +0000 [1bcc592765] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.
-
- If an old persistent subscription is recreated but then immediately
- destroyed because it is out of date, the subscription tree will have no
- leaf subscriptions on it. This was resulting in a crash when attempting
- to destroy the subscription tree.
-
- A simple NULL check fixes this problem.
-
- Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac
-
-2015-09-01 15:47 +0000 [b3cc2bd7df] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Solidify lifetime and ownership of objects.
-
- There have been crashes and general instability seen in the pubsub code,
- so this patch introduces three changes to increase the stability.
-
- First, the ownership model for subscriptions has been modified. Due to
- RLS, subscriptions are stored in memory as a tree structure. Prior to my
- patch, the PJSIP subscription was the owner of the subscription tree.
- When the PJSIP subscription told us that it was terminating, we started
- destroying the subscription tree along with all of the individual leaf
- subscriptions that belong to the tree. The problem with this model is
- that the two actors in play here, the PJSIP subscription and the
- individual leaf subscriptions, need to have joint ownership of the
- subscription tree. So now, the PJSIP subscription and the individual
- leaf subscriptions each have a reference to the subscription tree. This
- way, we will not actually free memory until no players are left that
- care. The PJSIP subscription is a bigger stakeholder, in that if the
- PJSIP subscription's reference to the subscription tree is removed, the
- subscription tree instructs the leaf subscriptions to shut down and drop
- their references to the subscription tree when possible. The individual
- leaf subscriptions, upon being told to shut down, can drop their stasis
- subscriptions or whatever they use to learn of new state, and then drop
- their reference to the subscription tree once they are ready to die.
-
- Second, the lifetime of a PJSIP subscription's reference to our
- subscription tree has been altered. As I learned from doing a deep dive,
- the PJSIP evsub code can tell Asterisk multiple times that the
- subscription has been terminated, and not all of these times
- are especially helpful. I have altered the message flow that we use for
- SIP subscriptions such that we will always drop the PJSIP subscription's
- reference to the subscription tree when we send the NOTIFY that
- terminates a SIP subscription. This also means that we will now queue
- NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so
- that we can have predictable state changes from the PJSIP evsub code.
-
- Third, the synchronization of operations has been improved. PJSIP can
- call into our code from a serializer thread (e.g. upon receiving an
- incoming request) or from the monitor thread (e.g. when a subscription
- times out). Because of this, there is the possibility of competing
- threads stepping on each other. PJSIP attempts to do some
- synchronization on its own by always keeping the dialog lock held when
- it calls into us. However, since we end up pushing tasks into the
- serializer, the result was that serialized operations were not grabbing
- the dialog lock and could, as a result, step on something that was being
- attempted by a different thread. Now we ensure that serialized
- operations grab the dialog lock, then check for extenuating
- circumstances, then proceed with their operation if they can.
-
- Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5
-
-2015-10-19 15:28 +0000 [c8c65dfa41] Richard Mudgett <rmudgett@digium.com>
-
- * strings.c: Fix __ast_str_helper() to always return a terminated string.
-
- Users of functions which call __ast_str_helper() such as the ones listed
- below are likely to not check the return value for failure so ensuring
- that the string is always nil terminated is a good safety measure.
-
- ast_str_set_va()
- ast_str_append_va()
- ast_str_set()
- ast_str_append()
-
- Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07
-
-2015-10-19 15:27 +0000 [b271d4a28a] Richard Mudgett <rmudgett@digium.com>
-
- * Add missing failure checks to ast_str_set_va() callers.
-
- Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f
-
-2015-10-21 11:44 +0000 [f2725c8b77] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Move URI validation to use time.
-
- In a realtime based system with a limited number of threadpool threads
- it is possible for a deadlock to occur. This happens when permanent
- endpoint state is updated, which will cause database queries to be done.
- These queries may result in URI validation being done which is done
- synchronously using a PJSIP thread. If all PJSIP threads are in use
- processing traffic they themselves may be blocked waiting to get the
- permanent endpoint container lock when identifying an endpoint.
-
- This change moves URI validation to occur at use time instead of
- configuration time. While this comes at a cost of not seeing a problem
- until you use it it does solve the underlying deadlock problem.
-
- ASTERISK-25486 #close
-
- Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a
-
-2015-10-21 08:08 +0000 [84ff075d41] Alexander Traud <pabstraud@compuserve.com>
-
- * format: Update the maximum packetization time for iLBC 30.
-
- In September 2006, the maximum packetization time (ptime) were set to such a
- low value, packetization was disabled for many codecs actually. This was fixed
- for many codecs but not for iLBC 30. This enables packetization for iLBC which
- can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf.
-
- ASTERISK-7803
-
- Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12
-2015-10-21 09:51 +0000 [869ef2a8ee] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Fix autoframing=yes.
-
- With Asterisk 13, the structures ast_format and ast_codec changed. Because of
- that, the paketization timing (framing) of the RTP channel moved away from the
- formats/codecs. In the course of that change, the ptime of the callee was not
- honored anymore, when the optional autoframing was enabled.
-
- ASTERISK-25484 #close
-
- Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4
-
-2015-10-20 22:24 +0000 [9fd2adc204] Matt Jordan <mjordan@digium.com>
-
- * rest-api-templates: Wikify error code response reasons
-
- Error response code descriptions may contain wiki markup that need to be
- escaped. Without this patch, Confluence will reject the document being sent
- and the responsible script will raise an exception.
-
- Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5
-
-2015-10-20 12:06 +0000 [72cbb6df55] Matt Jordan <mjordan@digium.com>
-
- * funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function
-
- When ab803ec342 was committed, it accidentally forgot to actually *add* the
- HOLD_INTERCEPT function. This highlights two interesting points:
- * Gerrit forces you to put the patch as it is going to into the repo up for
- review, which Review Board did not. Yay Gerrit.
- * No one apparently bothered to use this feature, or else they don't know about
- it. I'm going to go with the latter explanation.
-
- ASTERISK-24922
-
- Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396
-
-2015-10-19 19:59 +0000 [9fc9777fa3] Matt Jordan <mjordan@digium.com>
-
- * contrib/scripts/autosupport: Update for Asterisk 13
-
- This patch adds some minor tweaks for autosupport to update it for Asterisk 13.
- This includes:
- * Finally removing most references to Zaptel
- * Adding support for some additional 'core' commands, and fixing nomenclature
- that generally hasn't been used for some time
- * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels
-
- Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1
-
-2015-10-14 14:15 +0000 [dc6ec661b3] mdu113 <mulitskiy@acedsl.com>
-
- * res_config_pgsql.c: Fix deadlock loading realtime configuration.
-
- On v13, loading several thousand PJSIP endpoints on Asterisk start causes
- a deadlock most of the time.
-
- Thanks to mdu113 for discovering that there was a call to pgsql_exec() not
- protected by the pgsql_lock reentrancy lock.
-
- {quote}
- I believe a code path exists that attempts to use pgsql connection without
- locking pgsql_lock. I believe what happens during that deadlock that I
- see is two concurrent threads are both attempting to send query to pgsql,
- one of the thread is using a code path without locking pgsql_lock. If
- they managed to send queries at the same time, it seems postgres ignores
- one of the queries and replies only to the one of them. If it happens so
- that the thread holding the lock didn't receive the reply it will wait for
- it (and hold the lock) forever (or at least for very long time), thus
- completely blocking all access to db.
- {quote}
-
- * Added missing reentrancy locking around pgsql_exec() in find_table().
-
- * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion
- between the psql_tables list lock and the pgsql_lock.
-
- ASTERISK-25455 #close
- Reported by: mdu113
- Patches:
- res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113
-
- Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2
-
-2015-10-13 14:13 +0000 [f8707ae9a5] Olle Johansson (License 5267)
-
- * channels/chan_sip: Set cause code to 44 on RTP timeout
-
- To quote Olle:
-
- "When issuing a hangup due to RTP timeouts the cause code is not set. I have
- selected 44 based on Cisco's implementation..."
-
- ASTERISK-25135 #close
- Reported by: Olle Johansson
- patches:
- rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267)
-
- Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc
-
-2015-10-10 15:20 +0000 [486b172b50] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * Build: Add menuselect options for using compiler sanitizers
-
- This patch adds menuselect options for building Asterisk with
- various sanitizers provided by gcc and clang.
-
- When one of *SANITIZER flags is set in menuselect, the appropriate
- option is added to CFLAGS ad LDFLAGS for the build.
-
- Information on sanitizers in the project wiki:
- https://github.com/google/sanitizers/wiki
-
- GCC Manual:
- https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html
-
- Clang Compiler User's Manual:
- http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation
-
- ASTERISK-24718 #close
- Reported by: Badalian Vyacheslav
-
- Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0
-
-2015-10-12 11:21 +0000 [e14023ca35] Richard Mudgett <rmudgett@digium.com>
-
- * config.c: Fix off-nominal memory leak.
-
- Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0
-
-2015-10-12 11:20 +0000 [a99e821520] Richard Mudgett <rmudgett@digium.com>
-
- * config.c: Fix potential memory corruption after [section](+).
-
- The memory corruption could happen if the [section](+) is the last section
- in the file with trailing comments. In this case process_text_line() has
- left *last_cat is set to newcat and newcat is destroyed.
-
- Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93
-
-2015-10-12 11:21 +0000 [8d31d2526b] Richard Mudgett <rmudgett@digium.com>
-
- * config.c: Fix #include after [section](+).
-
- An #include right after a [section](+) would associate any variable
- assignments before a new section in the #include with the wrong section.
-
- * Fix section association by setting the current section to the appended
- section.
-
- * Fix '+' and '!' section flag interaction corner case depending upon
- which flag came first. If the '!' came first then it would be ignored.
- If the '!' came after then it would affect the appended section. The '!'
- will now no longer be ignored.
-
- ASTERISK-25461 #close
- Reported by: Sean Pimental
-
- Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3
-
-2015-10-06 18:01 +0000 [3329c714f7] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Fix deadlock when sending out-of-dialog requests.
-
- The struct send_request_wrapper has a pjsip lock associated with it that
- is created non-recursive. There is a code path for the struct
- send_request_wrapper lock that will attempt to lock it recursively. The
- reporter's deadlock showed that the thread calling endpt_send_request()
- deadlocked itself right after the wrapper object got created.
-
- Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited
- MWI NOTIFY messages can hit this deadlock.
-
- * Replaced the struct send_request_wrapper pjsip lock with the mutex lock
- that can come with an ao2 object since all of Asterisk's mutexes are
- recursive. Benefits include removal of code maintaining the pjsip
- non-recursive lock since ao2 objects already know how to maintain their
- own lock and the lock will show up in the CLI "core show locks" output.
-
- ASTERISK-25435 #close
- Reported by: Dmitriy Serov
-
- Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d
-
-2015-10-06 11:05 +0000 [a1435aa3fa] Stefan Engström <stefanen@kth.se>
-
- * res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending
-
- In ast_rtp_read, the value of the variable 'mark' which we try to assign to a
- frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate
- it to 0 or 1.
-
- ASTERISK-25451 #close
- Change-Id: I53bdf5c026041730184a6a809009c028549ce626
-
-2015-10-07 01:24 +0000 [3357678b94] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * func_presencestate: Return "not_set" when no data is set in AstDB
-
- Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not
- exist, i.e. when a new CustomPresence is added in the dialplan.
-
- ASTERISK-25400 #close
- Reported by: Andrew Nagy
-
- Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a
-
-2015-10-06 20:43 +0000 [b714b2152d] Matt Jordan <mjordan@digium.com>
-
- * res/res_rtp_asterisk: Fix assignment after ao2 decrement
-
- When we decide we will no longer schedule an RTCP write, we remove the
- reference to the RTP instance, then assign -1 to the stored scheduler ID
- in case something else comes along and wants to see if anything is scheduled.
-
- That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to
- fix the regression introduced by 3cf0f29310, this improper assignment on a
- potentially destroyed object started getting tripped on the build agents.
-
- Frankly, this should have been crashing a lot more often earlier. I can only
- assume that the timing was changed just enough by both changes to start
- actually hitting this problem.
-
- As it is, simply moving the assignment prior to the ao2 deference is sufficient
- to keep the RTP instance from being referenced when it is very, truly,
- aboslutely dead.
-
- (Note that it is still good practice to assign -1 to the scheduler ID when we
- know we won't be scheduling it again, as the ao2 deref *may* not always destroy
- the ao2 object.)
-
- ASTERISK-25449
-
- Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
-
-2015-10-06 12:40 +0000 [f939e2bd48] Florian Sauerteig <ffs@ccn.net>
-
- * chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers.
-
- If a Via header containes an IPv6 address and a port number is ommitted,
- as it is the standard port, we now leave the port empty and to not set it
- to the value after the first colon of the IPv6 address.
-
- ASTERISK-25443 #close
-
- Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70
-
-2015-10-05 16:53 +0000 [426263a64d] Richard Mudgett <rmudgett@digium.com>
-
- * chan_pjsip: Fix crash on reINVITE before initial INVITE completes.
-
- Apparently some endpoints attempt to send a reINVITE before completing the
- initial INVITE transaction. In this case PJSIP responds appropriately to
- the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip
- is using the initial INVITE transaction state to determine if an INVITE is
- the initial INVITE or a reINVITE. Since the initial INVITE transaction
- has not been confirmed yet chan_pjsip thinks the reINVITE is an initial
- INVITE and starts another PBX thread on the channel. The extra PBX thread
- ensures that hilarity ensues.
-
- * Fix checks for a reINVITE on incoming requests to look for the presence
- of a to-tag instead of the initial INVITE transaction state.
-
- * Made caller_id_incoming_request() determine what to do if there is a
- channel on the session or not. After a channel is created it is too late
- to just store the new party id on the session because the session's party
- id has already been copied to the channel's caller id.
-
- ASTERISK-25404 #close
- Reported by: Chet Stevens
-
- Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be
-
-2015-10-05 21:34 +0000 [50fa9ff997] Matt Jordan <mjordan@digium.com>
-
- * Fix improper usage of scheduler exposed by 5c713fdf18f
-
- When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of
- '0' returned. While this was valid per the documentation for the API, it was
- apparently never returned previously. As a result, several users of the
- scheduler API viewed the result as being invalid, causing them to reschedule
- already scheduled items or otherwise fail in interesting ways.
-
- This patch corrects the users such that they view '0' as valid, and a returned
- ID of -1 as being invalid.
-
- Note that the failing HEP RTCP tests now pass with this patch. These tests
- failed due to a duplicate scheduling of the RTCP transmissions.
-
- ASTERISK-25449 #close
-
- Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39
-2015-08-26 16:58 +0000 [8f777ab584] Debian Amtelco <dan@amtelco.com>
-
- * chan_pjsip: Add Referred-By header to the PJSIP REFER packet.
-
- Some systems require the REFER packet to include a Referred-By header.
- If the channel variable SIPREFERREDBYHDR is set, it passes that value as the
- Referred-By header value. Otherwise, it adds the current dialog’s local info.
-
- Reported by: Dan Cropp
- Tested by: Dan Cropp
-
- Change-Id: I3d17912ce548667edf53cb549e88a25475eda245
-
-2015-10-03 06:27 +0000 [74635b5638] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * manager: Fix GetConfigJSON returning invalid JSON
-
- When GetConfigJSON was introduced back in 1.6, it returned each
- section as an array of strings: ["key=value", "key2=value2"].
- Afterwards, it was changed a few times and became
- ["key": "value", "key2": "value2"], which is not a correct JSON.
- This patch fixes that by constructing a JSON object {} instead of
- an array [].
-
- Also, the keys "istemplate" and "tempates" that are used to
- indicate templates and their inherited categories are now wrapped in
- quotes.
-
- ASTERISK-25391 #close
- Reported by: Bojan Nemčić
-
- Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8
-
-2015-09-30 17:28 +0000 [40c69e78f5] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Fix deadlock with scheduler.
-
- A deadlock can happen when a sorcery object is being expired from the
- memory cache when at the same time another object is being placed into the
- memory cache. There are a couple other variations on this theme that
- could cause the deadlock. Basically if an object is being expired from
- the sorcery memory cache at the same time as another thread tries to
- update the next object expiration timer the deadlock can happen.
-
- * Add a deadlock avoidance loop in expire_objects_from_cache() to check if
- someone is trying to remove the scheduler callback from the scheduler.
-
- ASTERISK-25441 #close
-
- Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc
-
-2015-10-01 14:30 +0000 [dfeb513e85] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Replace inline code with function.
-
- Make sorcery_memory_cache_close() call remove_all_from_cache() instead of
- partially inlining it.
-
- ASTERISK-25441
-
- Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c
-
-2015-10-01 14:27 +0000 [ced0a2d71b] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Shutdown in a less crash potential order.
-
- Basically you should shutdown in the opposite order of how you setup since
- later setup pieces likely depend on earlier setup pieces. e.g.,
- Registering your external API with the rest of the system should be the
- last thing setup and the first thing unregistered during shutdown.
-
- Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e
-
-2015-09-30 17:27 +0000 [cc279eea11] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Misc tweaks.
-
- Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160
-
-2015-09-30 17:27 +0000 [9af3b613f6] Richard Mudgett <rmudgett@digium.com>
-
- * res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK.
-
- Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c
-
-2015-09-30 13:42 +0000 [56ed7b9dd5] Joshua Colp <jcolp@digium.com>
-
- * res_rtp_asterisk: Move "Set role" warning to be debug.
-
- In practice the set_role API callback can be invoked even
- when no ICE is present on an RTP instance. This can occur
- if ICE has not been enabled on it.
-
- ASTERISK-25438 #close
-
- Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69
-
-2015-09-28 15:31 +0000 [ddebb217f0] Richard Mudgett <rmudgett@digium.com>
-
- * sched.c: Add warning about negative time interval request.
-
- Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc
-
-2015-09-29 21:15 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.6.0-rc1 Released.
-
-2015-09-29 16:12 +0000 [bba1c4066b] Kevin Harwell <kharwell@lunkwill>
-
- * Release summaries: Add summaries for 13.6.0-rc1
-
-2015-09-29 16:08 +0000 [82c4aecdbb] Kevin Harwell <kharwell@lunkwill>
-
- * .version: Update for 13.6.0-rc1
-
-2015-09-29 16:08 +0000 [bc18db7388] Kevin Harwell <kharwell@lunkwill>
-
- * .lastclean: Update for 13.6.0-rc1
-
-2015-09-29 16:08 +0000 [b9c53f95e3] Kevin Harwell <kharwell@lunkwill>
-
- * realtime: Add database scripts for 13.6.0-rc1
-
-2015-09-29 14:53 +0000 [d30939b6e8] Kevin Harwell <kharwell@digium.com>
-
- * ARI: Changed version from 1.8.0 to 1.9.0
-
- Change-Id: I510991c60d28d171f47c4b58bba4947f7fc71b13
-
-2015-09-25 18:37 +0000 [5f19c9bade] Richard Mudgett <rmudgett@digium.com>
-
- * res/ari/config.c: Fix user sort compare function.
-
- Made use the ao2 sort compare template function and OBJ_SEARCH_xxx
- identifiers.
-
- Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c
-
-2015-09-25 17:26 +0000 [3a85764039] Richard Mudgett <rmudgett@digium.com>
-
- * res/ari/config.c: Optimize conf_alloc() object init.
-
- * Now conf_alloc() has more off nominal error checking.
-
- * Eliminated RAII_VAR() use in conf_alloc().
-
- * Eliminated a dubius shortcut when destroying cfg->general in
- conf_destructor() that would cause a crash if cfg->general failed to get
- allocated.
-
- * Add some ACO registration section comments.
-
- Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a
-
-2015-09-25 16:48 +0000 [028033e5a8] Richard Mudgett <rmudgett@digium.com>
-
- * res/ari/config.c: Fix conf_alloc() object init.
-
- Need to finish initializing the string fields in the ao2 object before
- putting any default strings into them.
-
- ASTERISK-25383 #close
- Reported by: yaron nahum
-
- Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84
-
-2015-09-27 20:45 +0000 [90165e306d] Matt Jordan <mjordan@digium.com>
-
- * res/res_stasis: Fix accidental subscription to 'all' bridge topic
-
- When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a
- NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to
- all bridges. Unfortunately, the res_stasis control loop did not check that
- a bridge changing on a channel's control object was actually also non-NULL.
- As a result, app_subscribe_bridge will be called with a NULL bridge when a
- channel leaves a bridge. This causes a new subscription to be made to the
- bridge. If an application has also subscribed to the bridge, the application
- will now have two subscriptions:
- (1) The explicit one created by the app
- (2) The implicit one accidentally created by the control structure
-
- As a result, the 'BridgeDestroyed' event can be sent multiple times. This
- patch corrects the control loop such that it only subscribes an application
- to a new bridge if the bridge pointer is non-NULL.
-
- ASTERISK-24870
-
- Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f
-
-2015-09-04 13:51 +0000 [e1223ff6db] Scott Griepentrog <scott@griepentrog.com>
-
- * Scripts: check file versions of Asterisk and dependencies
-
- To help in diagnosing mismatched modules and libraries, this
- script scans for version, repository, and source information
- and reports what is found.
-
- ASTERISK-25376 #close
- Reported by: Ashley Sanders
-
- Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6
-
-2015-09-24 14:56 +0000 [6b1e7583c1] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Force COLP update if outgoing channel name changed.
-
- * When a call is answered and the outgoing channel name has changed then
- force a connected line update because the channel is no longer the same.
- The channel was masqueraded into by another channel. This is usually
- because of a call pickup.
-
- Note: Forwarded calls are handled in a controlled manner so the original
- channel name is replaced with the forwarded channel.
-
- ASTERISK-25423 #close
- Reported by: John Hardin
-
- Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172
-
-2015-09-24 14:20 +0000 [6bf304bf25] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Factor out a connected line update routine.
-
- Replace inlined code with update_connected_line_from_peer().
-
- ASTERISK-25423
- Reported by: John Hardin
-
- Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3
-
-2015-09-24 13:27 +0000 [e36b5f1e8e] Richard Mudgett <rmudgett@digium.com>
-
- * app_dial.c: Make 'A' option pass COLP updates.
-
- While the 'A' option is playing the announcement file allow the caller and
- peer to exchange COLP update frames.
-
- ASTERISK-25423
- Reported by: John Hardin
-
- Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9
-
-2015-09-24 12:59 +0000 [747bfac895] Richard Mudgett <rmudgett@digium.com>
-
- * app_dial.c: Force COLP update if outgoing channel name changed.
-
- * When a call is answered and the outgoing channel name has changed then
- force a connected line update because the channel is no longer the same.
- The channel was masqueraded into by another channel. This is usually
- because of a call pickup.
-
- Note: Forwarded calls are handled in a controlled manner so the original
- channel name is replaced with the forwarded channel.
-
- ASTERISK-25423
- Reported by: John Hardin
-
- Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c
-
-2015-09-24 12:37 +0000 [14481d9aa0] Richard Mudgett <rmudgett@digium.com>
-
- * app_dial.c: Factor out a connected line update routine.
-
- Replace inlined code with update_connected_line_from_peer().
-
- ASTERISK-25423
- Reported by: John Hardin
-
- Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091
-
-2015-09-23 17:41 +0000 [bbeda190c3] Richard Mudgett <rmudgett@digium.com>
-
- * app_dial.c: Remove some no-op code.
-
- Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54
-
-2015-09-23 14:02 +0000 [f050fa76eb] Mark Michelson <mmichelson@digium.com>
-
- * logger: Prevent duplicate dynamic channels from being added.
-
- There was a problem observed where the "logger add channel" CLI command
- would allow for a channel with the same name to be added multiple times.
- This would result in each message being written out to the same file
- multiple times.
-
- The problem was due to the difference in how logger channel filenames
- are stored versus the format they are allowed to be presented when they
- are added. For instance, if adding the logger channel "foo" through the
- CLI, the result would be a logger channel with the file name
- /var/log/asterisk/foo being stored. So when trying to add another "foo"
- channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily
- add the duplicate channel.
-
- The fix presented here is to introduce two new methods in the logger
- code:
- * make_filename(): given a logger channel name, this creates the
- filename for that logger channel.
- * find_logchannel(): given a logger channel name, this calls
- make_filename() and then traverses the list of logchannels in order
- to find a match.
-
- This change has made use of make_filename() and find_logchannel()
- throughout to more consistently behave.
-
- ASTERISK-25305 #close
- Reported by Mark Michelson
-
- Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36
-
-2015-09-24 14:49 +0000 [629458d349] Mark Michelson <mmichelson@digium.com>
-
- * Do not swallow frames on channels leaving bridges.
-
- When leaving a bridge, indications on a channel could be swallowed by
- the internal indication logic because it appears that the channel is on
- its way to be hung up anyway. One such situation where this is
- detrimental is when channels on hold are redirected out of a bridge. The
- AST_CONTROL_UNHOLD indication from the bridging code is swallowed,
- leaving the channel in question to still appear to be on hold.
-
- The fix here is to modify the logic inside ast_indicate_data() to not
- drop the indication if the channel is simply leaving a bridge. This way,
- channels on hold redirected out of a bridge revert to their expected "in
- use" state after the redirection.
-
- ASTERISK-25418 #close
- Reported by Mark Michelson
-
- Change-Id: If6115204dfa0551c050974ee138fabd15f978949
-
-2015-09-22 17:08 +0000 [5f15cd93f0] Richard Mudgett <rmudgett@digium.com>
-
- * app_page.c: Fix crash when forwarding with a predial handler.
-
- Page uses the async method of dialing with the dial API. When a call gets
- forwarded there is no calling channel available. If the predial handler
- was set then the calling channel could not be put into auto-service
- for the forwarded call because it doesn't exist. A crash is the result.
-
- * Moved the callee predial parameter string processing to before the
- string is passed to the dial API rather than having the dial API do it.
- There are a few benefits do doing this. The first is the predial
- parameter string processing doesn't need to be done for each channel
- called by the dial API. The second is in async mode and the forwarded
- channel is to have the predial handler executed on it then the
- non-existent calling channel does not need to be present to process the
- predial parameter string.
-
- * Don't start auto-service on a non-existent calling channel to execute
- the predial handler when the dial API is in async mode and forwarding a
- call.
-
- ASTERISK-25384 #close
- Reported by: Chet Stevens
-
- Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981
-
-2015-09-03 21:19 +0000 [b50e372394] Matt Jordan <mjordan@digium.com>
-
- * ARI: Add events for Contact and Peer Status changes
-
- This patch adds support for receiving events regarding Peer status changes
- and Contact status changes. This is particularly useful in scenarios where
- we are subscribed to all endpoints and channels, where we often want to know
- more about the state of channel technology specific items than a single
- endpoint's state.
-
- ASTERISK-24870
-
- Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9
-
-2015-09-04 12:24 +0000 [3502c0431d] Matt Jordan <mjordan@digium.com>
-
- * res/res_stasis_device_state: Allow for subscribing to 'all' device state
-
- This patch adds support for subscribing to all device state changes. This is
- done either by subscribing to an empty device, e.g., 'eventSource=deviceState:',
- or by the WebSocket connection specifying that it wants all state in the
- system.
-
- ASTERISK-24870
-
- Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32
-
-2015-09-04 12:25 +0000 [4c9f613309] Matt Jordan <mjordan@digium.com>
-
- * ARI: Add the ability to subscribe to all events
-
- This patch adds the ability to subscribe to all events. There are two possible
- ways to accomplish this:
- (1) On initial WebSocket connection. This patch adds a new query parameter,
- 'subscribeAll'. If present and True, Asterisk will subscribe the
- applications to all ARI events.
- (2) Via the applications resource. When subscribing in this manner, an ARI
- client should merely specify a blank resource name, i.e., 'channels:'
- instead of 'channels:12354'. This will subscribe the application to all
- resources of the 'channels' type.
-
- ASTERISK-24870 #close
-
- Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6
-
-2015-09-21 08:16 +0000 [ec514ad64d] Elazar Broad <elazar@thebroadfamily.com>
-
- * core/logging: Fix logging to more than one syslog channel
-
- Currently, Asterisk will log to the last configured syslog
- channel in logger.conf. This is due to the fact that the
- final call to openlog() supersedes all of the previous calls.
- This commit removes the call to openlog() and passes the
- facility to ast_log_vsyslog(), along with utilizing the
- LOG_MAKEPRI macro to ensure that the message is routed to
- the correct facility and with the correct priority.
-
- ASTERISK-25407 #close
- Reported by: Elazar Broad
- Tested by: Elazar Broad
-
- Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2
-
-2015-09-21 18:06 +0000 [aeddee39fb] Kevin Harwell <kharwell@digium.com>
-
- * app_record: RECORDED_FILE variable not being populated
-
- The RECORDED_FILE variable is empty unless a '%d' is specified in the filename.
- This patch makes it so the variable is always set to the filename.
-
- ASTERISK-25410 #close
-
- Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653
-
-2015-09-16 08:22 +0000 [2bd27d1222] Joshua Colp <jcolp@digium.com>
-
- * pbx: Update device and presence state when changing a hint extension.
-
- When changing a hint extension without removing the hint first the
- device state and presence state is not updated. This causes the state
- of the hint to be that of the previous extension and not the current
- one. This state is kept until a state change occurs as a result of
- something (presence state change, device state change).
-
- This change updates the hint with the current device and presence
- state of the new extension when it is changed. Any state callbacks
- which may have been added before the hint extension is changed are
- also informed of the new device and presence state if either have
- changed.
-
- ASTERISK-25394 #close
-
- Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f
-
-2015-09-17 16:34 +0000 [c94f46080f] Scott Griepentrog <scott@griepentrog.com>
-
- * CHAOS: avoid crash if string create fails
-
- Validate string buffer allocation before using them.
-
- ASTERISK-25323
-
- Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0
-
-2015-09-17 04:52 +0000 [b59c4d82b5] Walter Doekes <walter+asterisk@wjd.nu>
-
- * chan_sip: Fix From header truncation for extremely long CALLERID(name).
-
- The CALLERID(num) and CALLERID(name) and other info are placed into the
- `char from[256]` in initreqprep. If the name was too long, the addr-spec
- and params wouldn't fit.
-
- Code is moved around so the addr-spec with params is placed there first,
- and then fitting in as much of the display-name as possible.
-
- ASTERISK-25396 #close
-
- Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260
-
-2015-09-17 16:59 +0000 [4cc59533b9] Richard Mudgett <rmudgett@digium.com>
-
- * CHAOS: res_pjsip_diversion avoid crash if allocation fails
-
- Validate ast_malloc buffer returned before using it in
- set_redirecting_value().
-
- ASTERISK-25323
-
- Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253
-
-2015-09-17 16:47 +0000 [4fb95bbc4e] Kevin Harwell <kharwell@digium.com>
-
- * app_queue: AgentComplete event has wrong reason
-
- When a queued caller transfers an agent to another extension sometimes the
- raised AgentComplete event has a reason of "caller" and sometimes "transfer".
- Since a transfer has taken place this should always be transfer. This occurs
- because sometimes the stasis hangup event arrives before the transfer event
- thus writing a different reason out.
-
- With this patch, when a hangup event is received during a transfer it will
- check to see if the channel that is hanging up is part of a transfer. If so
- it will return and let the subsequently received transfer event handler take
- care of the cleanup.
-
- ASTERISK-25399 #close
-
- Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d
-
-2015-09-17 13:09 +0000 [fb6b5c684b] Scott Griepentrog <scott@griepentrog.com>
-
- * PJSIP: avoid crash when getting rtp peer
-
- Although unlikely, if the tech private is returned as
- a NULL, chan_pjsip_get_rtp_peer() would crash.
-
- ASTERISK-25323
-
- Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a
-
-2015-09-17 11:31 +0000 [6409e7b11a] Kevin Harwell <kharwell@digium.com>
-
- * app_queue: Crash when transferring
-
- During some transfer scenarios involving queues Asterisk would sometimes
- crash when trying to obtain a channel snapshot (could happen on caller or
- member channels). This occurred because the underlying channel had already
- disappeared when trying to obtain the latest snapshot.
-
- This patch adds a reference to both the member and caller channels that
- extends to the lifetime of the queue'd call, thus making sure the channels
- will always exist when retrieving the latest snapshots.
-
- ASTERISK-25185 #close
- Reported by: Etienne Lessard
-
- Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128
-
-2015-09-16 17:36 +0000 [fe5077b1f8] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Eliminate race during initial NOTIFY.
-
- There is a slim chance of a race condition occurring where two threads
- can both attempt to manipulate the same area.
-
- Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
- lets the specific subscription handler know that the subscription has
- been established.
-
- At this point, Thread B may detect a state change on the subscribed
- resource and queue up a notification task on Thread C, the subscription
- serializer thread.
-
- Now Thread A attempts to generate the initial NOTIFY request to send to
- the subscriber at the same time that Thread C attempts to generate a
- state change NOTIFY request to send to the subscriber.
-
- The result is that Threads A and C can step on the same memory area,
- resulting in a crash. The crash has been observed as happening when
- attempting to allocate more space to hold the body for the NOTIFY.
-
- The solution presented here is to queue the subscription establishment
- and initial NOTIFY generation onto the subscription serializer thread
- (Thread C in the above scenario). This way, there is no way that a state
- change notification can occur before the initial NOTIFY is sent, and if
- there is a quick succession of NOTIFYs, we can guarantee that the two
- NOTIFY requests will be sent in succession.
-
- Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815
-
-2015-08-28 15:42 +0000 [b88c54fa4b] Alexander Traud <pabstraud@compuserve.com>
-
- * translate: Fix transcoding while different in frame size.
-
- When Asterisk translates between codecs, each with a different frame size (for
- example between iLBC 30 and Speex-WB), too large frames were created by
- ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
- length, creating several frames when necessary. Affects all transcoding modules
- which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.
-
- ASTERISK-25353 #close
-
- Change-Id: I2e229569d73191d66a4e43fef35432db24000212
-
-2015-09-10 17:19 +0000 [5c713fdf18] Mark Michelson <mmichelson@digium.com>
-
- * scheduler: Use queue for allocating sched IDs.
-
- It has been observed that on long-running busy systems, a scheduler
- context can eventually hit INT_MAX for its assigned IDs and end up
- overflowing into a very low negative number. When this occurs, this can
- result in odd behaviors, because a negative return is interpreted by
- callers as being a failure. However, the item actually was successfully
- scheduled. The result may be that a freed item remains in the scheduler,
- resulting in a crash at some point in the future.
-
- The scheduler can overflow because every time that an item is added to
- the scheduler, a counter is bumped and that counter's current value is
- assigned as the new item's ID.
-
- This patch introduces a new method for assigning scheduler IDs. Instead
- of assigning from a counter, a queue of available IDs is maintained.
- When assigning a new ID, an ID is pulled from the queue. When a
- scheduler item is released, its ID is pushed back onto the queue. This
- way, IDs may be reused when they become available, and the growth of ID
- numbers is directly related to concurrent activity within a scheduler
- context rather than the uptime of the system.
-
- Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2
-
-2015-08-21 21:50 +0000 [865377fc38] Rodrigo Ramírez Norambuena <a@rodrigoramirez.com>
-
- * chan_sip.c: Validation on module reload
-
- Change validation on reload module because now used the cli function for
- reload. The sip_reload() function never fail and ever return NULL for this
- reason on reload() now use the call the sip_reload() and return
- AST_MODULE_LOAD_SUCCESS.
-
- This problem is dectected on reload by PUT method on ARI, getting always
- 404 http code when the module is reloaded.
-
- ASTERISK-25325 #close
- Reporte by: Rodrigo Ramírez Norambuena
-
- Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb
-
-2015-08-21 17:39 +0000 [e75aff53e6] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.
-
- Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093
-
-2015-09-09 12:24 +0000 [4d91d01df1] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Add some notification comments.
-
- Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20
-
-2015-08-21 18:01 +0000 [f36a9d1221] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.
-
- We should not try to send a SIP response message because we may be
- restoring a persistent subscription where we are not responding to a SIP
- request.
-
- Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec
-
-2015-08-21 17:40 +0000 [94582f8fab] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Fix off-nominal memory leak.
-
- Fix off-nominal visited vector leak in build_resource_tree().
-
- Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c
-
-2015-08-21 15:26 +0000 [8b3ed52239] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Fix one byte buffer overrun error.
-
- ast_sip_pubsub_register_body_generator() did not account for the null
- terminator set by sprintf() in the allocated output buffer.
-
- Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2
-
-2015-08-21 15:25 +0000 [4329bd1e4c] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().
-
- Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3
-
-2015-08-21 11:04 +0000 [a456a20ecf] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_pubsub.c: Add missing error return in load_module().
-
- Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc
-
-2015-08-21 11:03 +0000 [f58f4c6e27] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip/location.c: Use the builtin ao2_callback() match function instead.
-
- Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f
-
-2015-09-10 09:49 +0000 [9d1f176e29] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Copy default_from_user to avoid crash.
-
- The default_from_user retrieval function was pulling the
- default_from_user from the global configuration struct in an unsafe way.
- If using a database as a backend configuration store, the global
- configuration struct is short-lived, so grabbing a pointer from it
- results in referencing freed memory.
-
- The fix here is to copy the default_from_user value out of the global
- configuration struct.
-
- Thanks go to John Hardin for discovering this problem and proposing the
- patch on which this fix is based.
-
- ASTERISK-25390 #close
- Reported by Mark Michelson
-
- Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c
-
-2015-09-10 08:39 +0000 [1dd0e220bf] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route
-
- We will only rewrite the Contact header if there is no Record-Route header in
- the received request. If a malfunctioning proxy places a Record-Route header
- into a REGISTER request, we will decide that we shouldn't update the IP/port
- in the Contact header, and we will end up storing a contact with an AoR that
- contains the NAT'd IP address.
-
- While it is nice to have the proxy *not* send a Record-Route in a REGISTER
- request, it's also a good idea to not process the header in a non-dialog
- message. This patch updates the code to explicitly ignore the Record-Route
- header in REGISTER requests.
-
- ASTERISK-25387 #close
-
- Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f
-
-2015-09-03 21:15 +0000 [4eedd9ef9d] Matt Jordan <mjordan@digium.com>
-
- * main/config_options: Check for existance of internal object before derefing
-
- Asterisk can load and register an object type while still having an invalid
- sorcery mapping. This can cause an issue when a creation call is invoked.
- For example, mis-configuring PJSIP's endpoint identifier by IP address mapping
- in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a
- subsequent ARI invocation to create the object will cause a crash, as the
- internal type may not be registered as sorcery expects.
-
- Merely checking for a NULL pointer here solves the issue.
-
- Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac
-2015-09-09 16:46 +0000 [71408df2b8] Alexander Anikin <may213@yandex.ru>
-
- * chan_ooh323: Add ProgressIndicator IE with inband info available
-
- Add ProgressIndicator IE with inband info present to Progress and
- Alerting Q.931 message
-
- ASTERISK-25227 #close
- Reported by: Alexandr Dranchuk
-
- Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203
-2015-09-08 10:35 +0000 [f72f9ceefc] Scott Griepentrog <scott@griepentrog.com>
-
- * pjsip: avoid possible crash req_caps allocation failure
-
- Make certain that the pjsip session has not failed to
- allocate the format capabilities structure, which can
- otherwise cause a crash when referenced.
-
- ASTERISK-25323
-
- Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750
-
-2015-09-03 14:07 +0000 [fbf720db91] Jonathan Rose <jrose@digium.com>
-
- * ParkAndAnnounce: Add variable inheritance
-
- In Asterisk 11, the announcer channel would receive channel variables
- from the channel being parked by means of normal channel inheritance.
- This functionality was lost during the big res_parking project in
- Asterisk 12. This patch restores that functionality.
-
- ASTERISK-25369 #close
- Review: https://gerrit.asterisk.org/#/c/1180/
-
- Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e
-
-2015-09-04 16:33 +0000 [695f26cbb7] David M. Lee <dlee@respoke.io>
-
- * res_rtp_asterisk: Add more ICE debugging
-
- In working through a recent ICE negotiation bug, I found the debug
- logging in res_rtp_asterisk to be lacking. This patch adds a number of
- debug and warning statements that were helpful.
-
- Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80
-2015-09-01 10:16 +0000 [4ed9c9a280] Guido Falsi <madpilot@freebsd.org>
-
- * Core/General: Add #ifdef needed on FreeBSD.
-
- pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD
- too.
-
- ASTERISK-25310 #close
- Reported by: Guido Falsi
-
- Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4
-
-2015-09-08 07:21 +0000 [5469caa9dd] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Use hash for contact object identity instead of Contact URI.
-
- In the wild it is possible for Contact URIs to be quite long as
- parameters can exist on them. This can present a problem when storing
- them in the AstDB as the URI is used as part of the object name and
- there is a fixed length limit for the AstDB. This will cause
- the contact to not get stored.
-
- This change uses the MD5 hash of the Contact URI as part of the
- object name instead. This has a fixed length which is guaranteed
- to not exceed the AstDB length limit.
-
- ASTERISK-25295 #close
-
- Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02
-
-2015-09-07 13:19 +0000 [480c443e26] Alexander Anikin <may213@yandex.ru>
-
- * chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy
-
- Call ast_rtp_instance_stop on ooh323_destroy to free resources
- allocated by rtp instance
-
- ASTERISK-25299 #close
- Report by: Alexandr Dranchuk
-
- Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f
-
-2015-09-07 11:15 +0000 [c3e6debdb9] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip: Purge contacts when an AoR is deleted
-
- When an AoR is deleted by an external mechanism, such as through ARI, we
- currently do not remove dynamic contacts that were created for that AoR as a
- result of a received REGISTER request. As a result, re-creating the AoR will
- cause the dynamic contact to be interpreted as a persistent contact, leading
- to some rather strange state being created for the contacts/endpoints.
-
- This patch adds a sorcery observer for the 'aor' object. When a delete is
- issued on the underlying sorcery object, the observer is called, and all
- contacts created and persisted in sorcery for that AoR are also removed. Note
- that we don't want to perform this action when an AO2 object that is an AoR is
- destroyed, as the AoR can still exist in the backing storage (and we would
- thus be removing valid contacts from an AoR that still "exists".)
-
- ASTERISK-25381 #close
-
- Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328
-
-2015-09-05 14:58 +0000 [78d0b9d97e] Matt Jordan <mjordan@digium.com>
-
- * channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id
-
- This patch adds a new option to the CHANNEL function that allows for the
- extraction of the SIP call-id. It is used in conjunction with the 'pjsip'
- option, and will return the Call-ID of the INVITE request that established
- the PJSIP channel.
-
- ASTERISK-25352
-
- Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
-
-2015-09-04 16:06 +0000 [61c6c6aa6c] David M. Lee <dlee@respoke.io>
-
- * Fix when remote candidates exceed PJ_ICE_MAX_CAND
-
- We were passing the wrong count into pj_ice_sess_create_check_list(),
- causing the create to fail if we ever received more than PJ_ICE_MAX_CAND
- candidates.
-
- Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378
-
-2015-09-04 14:40 +0000 [ac62928d6b] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Change default from user value.
-
- When Asterisk sends an outbound SIP request, if there is no direct
- reason to place a specific value for the username in the From header,
- Asterisk would generate a UUID. For example, this would happen when
- sending outbound OPTIONS requests when qualifying or when sending
- outbound INVITE requests when originating (if no explicit caller ID were
- provided). The issue is that some SIP providers reject these sorts of
- requests with a "Name too long" error response.
-
- This patch aims to fix this by changing the default outbound username in
- From headers to "asterisk". This value can be overridden by changing the
- default_from_user option in the global options if desired.
-
- ASTERISK-25377 #close
- Reported by Mark Michelson
-
- Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190
-
-2015-09-04 09:26 +0000 [6002472a62] Scott Griepentrog <scott@griepentrog.com>
-
- * endpoint snapshot: avoid second cleanup on alloc failure
-
- In ast_endpoint_snapshot_create(), a failure to init the
- string fields results in two attempts to ao2_cleanup the
- same pointer. Removed RAII_VAR to eliminate problem.
-
- ASTERISK-25375 #close
- Reported by: Scott Griepentrog
-
- Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979
-
-2015-09-04 05:33 +0000 [d32e516c7c] Martin Tomec <tomec.martin@gmail.com>
-
- * res/pjsip: Mark WSS transport as secure
-
- Pjsip is refusing to use unsecure transport with "sips" in url.
- WSS should be considered as secure transport.
-
- ASTERISK-24602 #comment Partially fixed by setting WSS as secure
-
- Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353
-
-2015-09-02 17:26 +0000 [ad9cb6c2ce] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Fix contact refleak on stateful responses.
-
- When sending a stateful response, creation of the transaction can fail,
- most commonly because we are trying to create a transaction from a
- retransmitted request. When creation of the transaction fails, we end up
- leaking a reference to a contact that was bumped when the response was
- created.
-
- This patch adds the missing deref and fixes the reference leak.
-
- Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07
-
-2015-09-02 12:41 +0000 [cc1363209e] Joshua Colp <jcolp@digium.com>
-
- * pbx: Fix crash when issuing "core show hints" with long pattern match.
-
- When issuing the "core show hints" CLI command a combination of both
- the hint extension and context is created. This uses a fixed size
- buffer expecting that the extension will not exceed maximum extension
- length. When the extension is actually a pattern match this constraint
- does not hold true, and the extension may exceed the maximum extension
- length. In this case extra characters are written past the end of the
- fixed size buffer.
-
- This change makes it so the construction of the combined hint extension
- and context can not exceed the size of the buffer.
-
- ASTERISK-25367 #close
-
- Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499
-
-2015-09-01 09:05 +0000 [d58c8d73af] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: re-re-fix persistent subscription storage.
-
- A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
- a means of writing an appropriate packet to persistent storage. While
- this partially solved the issue, it had its own problems.
- pjsip_msg_print will always add a Content-Length header to the message
- it prints. Frequent restarts of Asterisk can result in persistent
- subscriptions being written with five or more Content-Length headers. In
- addition, sometimes some apparent corruption of individual headers could
- be seen.
-
- This aims to fix the problem by not running a parsed message through an
- interpreter but rather by taking the raw message and saving it. The
- logic for what to save is going to be different depending on whether a
- SUBSCRIBE was received from the wire or if it was pulled from
- persistence. When receiving a packet from the wire, when using a
- streaming transport, the rdata->pkt_info.packet may contain multiple SIP
- messages or fragments. However, the rdata->msg_info.msg_buf will always
- contain the current SIP message to be processed. When pulling from
- persistence, though, the rdata->msg_info.msg_buf will be NULL since no
- transport actually handled the packet. However, since we know that we
- will always ever pull one SIP message from persistence, we are free to
- save directly from rdata->pkt_info.packet instead.
-
- ASTERISK-25365 #close
- Reported by Mark Michelson
-
- Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b
-
-2015-08-31 15:24 +0000 [03fe79f29e] Mark Michelson <mmichelson@digium.com>
-
- * Fix deadlock on presence state changes.
-
- A deadlock was observed where three threads were competing for different
- locks:
-
- * One thread held the hints lock and was attempting to lock a specific
- hint.
- * One thread was holding the specific hint's lock and was attempting to
- lock the contexts lock
- * One thread was holding the contexts lock and attempting to lock the
- hints lock.
-
- Clearly the second thread was doing the wrong thing here. The fix for
- this is to make sure that the hint's lock is not held on presence state
- changes. Something similar is already done (and commented about) for
- device state changes.
-
- ASTERISK-25362 #close
- Reported by Mark Michelson
-
- Change-Id: I15ec2416b92978a4c0c08273b2d46cb21aff97e2
-
-2015-08-29 10:36 +0000 [a676ba2aad] Joshua Colp <jcolp@digium.com>
-
- * taskprocessor: Fix race condition between unreferencing and finding.
-
- When unreferencing a taskprocessor its reference count is checked
- to determine if it should be unlinked from the taskprocessors
- container and its listener shut down. In between the time when the
- reference count is checked and unlinking it is possible for
- another thread to jump in, find it, and get a reference to it. If
- the thread then uses the taskprocessor it may find that it is not
- in the state it expects.
-
- This change locks the taskprocessors container during almost the
- entire unreference operation to ensure that any other thread which
- may attempt to find the taskprocessor has to wait.
-
- ASTERISK-25295
-
- Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c
-
-2015-08-28 20:22 +0000 [1b1561f4c8] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.
-
- The keepalive support in res_pjsip_sdp_rtp currently assumes
- that a stream will only be negotiated once. This is false.
- If the stream is replaced and later added back it can be
- negotiated again causing multiple keepalive scheduled items
- to exist. This change explicitly deletes the existing
- keepalive scheduled item before adding the new one.
-
- The res_pjsip_sdp_rtp module also does not stop RTP
- keepalives or timeout timer if the stream has been
- replaced. This change adds a callback to the session media
- interface to allow a media stream to be stopped without
- the resources being destroyed. This allows the scheduled
- items and RTP to be stopped when the stream no longer
- exists.
-
- ASTERISK-25356 #close
-
- Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de
-
-2015-08-28 19:57 +0000 [85e1cb51b2] Joshua Colp <jcolp@digium.com>
-
- * sched: ast_sched_del may return prematurely due to spurious wakeup
-
- When deleting a scheduled item if the item in question is currently
- executing the ast_sched_del function waits until it has completed.
- This is accomplished using ast_cond_wait. Unfortunately the
- ast_cond_wait function can suffer from spurious wakeups so the
- predicate needs to be checked after it returns to make sure it has
- really woken up as a result of being signaled.
-
- This change adds a loop around the ast_cond_wait to make sure that
- it only exits when the executing task has really completed.
-
- ASTERISK-25355 #close
-
- Change-Id: I51198270eb0b637c956c61aa409f46283432be61
-
-2015-08-27 12:26 +0000 [c2c7319082] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_session: Don't invoke session supplements twice for BYE requests.
-
- When a BYE request is received the PJSIP invite session implementation
- creates and sends a 200 OK response before we are aware of it. This
- causes the INVITE session state callback to be called into and ultimately
- the session supplements run on the BYE request. Once this response has
- been sent the normal transaction state callback is invoked which
- invokes the session supplements on the BYE request again. This can
- be problematic in particular with res_pjsip_rfc3326 as it may
- attempt to update the hangup cause code on the channel while it is
- in the process of being hung up.
-
- This change makes it so the session supplements are only invoked
- once by the INVITE session state callback.
-
- ASTERISK-25318 #close
-
- Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a
-
-2015-08-26 15:26 +0000 [6862c2a167] Scott Griepentrog <scott@griepentrog.com>
-
- * Chaos: handle failed allocation in get_media_encryption_type
-
- If the ast_strndup() call fails to allocate a copy of the
- transport string for parsing, fail gracefully.
-
- ASTERISK-25323
- Reported by: Scott Griepentrog
-
- Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28
-
-2015-08-26 14:25 +0000 [f1cd636658] Scott Griepentrog <scott@griepentrog.com>
-
- * Chaos: make hangup NULL tolerant
-
- In chan_pjsip_new, if allocation of the pvt
- structure fails, ast_hangup is called. But
- it was written to assume pvt was valid, and
- this change corrects that.
-
- ASTERISK-25323
- Reported by: Scott Griepentrog
-
- Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87
-
-2015-08-26 05:40 +0000 [c01111223f] Joshua Colp <jcolp@digium.com>
-
- * chan_sip: Allow call pickup to set the hangup cause.
-
- The call pickup implementation in chan_sip currently sets the channel
- hangup cause to "normal clearing" if call pickup is successfully
- performed. This action overwrites the "answered elsewhere" hangup cause
- set by the call pickup code and can result in the SIP device in
- question showing a missed call when it should not.
-
- This change sets the hangup cause to "normal clearing" as a
- default initially but allows the call pickup to change it as
- needed.
-
- ASTERISK-25346 #close
-
- Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff
-
-2015-08-25 07:17 +0000 [2a4eee0cd9] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Add common ast_sip_get_host_ip API.
-
- Modules commonly used the pj_gethostip function for retrieving the
- IP address of the host. This function does not cache the result and may
- result in a DNS lookup occurring, or additional work. If the DNS
- server is unreachable or network issues arise this can cause the
- pj_gethostip function to block for a period of time.
-
- This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
- function which does the same thing but caches the host IP address at
- module load time. This results in no additional work being done each
- time the local host IP address is needed.
-
- ASTERISK-25342 #close
-
- Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e
-
-2015-08-24 11:04 +0000 [7c4d0c3506] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced
-
- When recreating a subscription it is possible for a freed sub_tree
- to be referenced when the initial NOTIFY fails to be created.
-
- Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788
-
-2015-08-24 06:21 +0000 [6c2dab1e88] Joshua Colp <jcolp@digium.com>
-
- * bridge: Kick channel from bridge if hung up during action.
-
- When executing an action in a bridge it is possible for the
- channel to be hung up without the bridge becoming aware of it.
- This is most easily reproducible by hanging up when the bridge
- is streaming DTMF due to a feature timeout. This change makes
- it so after action execution the channel is checked to determine
- if it has been hung up and if it has it is kicked from the bridge.
-
- ASTERISK-25341 #close
-
- Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062
-
-2015-08-23 18:26 +0000 [bc6fe07f5c] Matt Jordan <mjordan@digium.com>
-
- * res_pjsip/pjsip_configuration: Disregard empty auth values
-
- When an endpoint is backed by a non-static conf file backend (such as
- the AstDB or Realtime), the 'auth' object may be returned as being an
- empty string. Currently, res_pjsip will interpret that as being a valid
- auth object, and will attempt to authenticate inbound requests. This
- isn't desired; is an auth value is empty (which the name of an auth
- object cannot be), we should instead interpret that as being an invalid
- auth object and skip it.
-
- ASTERISK-25339 #close
-
- Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7
-
-2015-08-19 12:10 +0000 [0582776f7f] Richard Mudgett <rmudgett@digium.com>
-
- * ari/ari_websockets.c: Fix ast_debug parameter type mismatch.
-
- This is a type mismatch fix of the debugging commit
- c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why
- a testsuite test was failing only on one of the continuous
- integration build agents.
-
- Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75
-
-2015-08-19 10:30 +0000 [504213f542] Scott Griepentrog <scott@griepentrog.com>
-
- * contrib: script install_prereq should install sqlite3
-
- Asterisk needs the sqlite 3 library, which is package
- sqlite-devel in CentOS. By adding this package to the
- script, a problem with configure failing is resolved.
-
- ASTERISK-25331 #close
- Reported by: Kevin Harwell
-
- Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec
-
-2015-08-18 16:06 +0000 [77518d5434] Richard Mudgett <rmudgett@digium.com>
-
- * res_http_websocket.c: Fix some off nominal path cleanup.
-
- * Remove extraneous unlock on off-nominal path.
- * Add missing HTTP error reply.
-
- Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b
-
-2015-08-18 14:46 +0000 [c61547fee6] Richard Mudgett <rmudgett@digium.com>
-
- * res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().
-
- Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf
-
-2015-08-17 16:41 +0000 [bd867cd078] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Extract some functions for simpler code.
-
- * Extract set_queue_member_pause() from set_member_paused() for simpler
- and more consistent code.
-
- * Extract set_queue_member_ringinuse() from
- set_member_ringinuse_help_members() for simpler code.
-
- Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306
-
-2015-08-14 12:55 +0000 [e5f5b9f384] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.
-
- Setting the 'paused' and 'ringinuse' options on a queue member using the
- dialplan function QUEUE_MEMBER did not behave the same way as the
- equivalent dialplan applications or AMI actions.
-
- * Made queue_function_mem_write() call the set_member_paused() and
- set_member_value() for the 'paused' and 'ringinuse' options respectively.
- A beneficial side effect is that the queue name is now optional and sets
- the value in all queues the interface is a member.
-
- * Update QUEUE_MEMBER XML documentation.
-
- * Fix error checking in QUEUE_MEMBER() write.
-
- ASTERISK-25215 #close
- Reported by: Lorne Gaetz
-
- Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb
-
-2015-08-17 13:34 +0000 [ded51e3d77] Richard Mudgett <rmudgett@digium.com>
-
- * app_queue.c: Fix error checking in QUEUE_MEMBER() read.
-
- Change-Id: I7294e13d27875851c2f4ef6818adba507509d224
-
-2015-08-17 11:00 +0000 [ab373f2cef] Scott Griepentrog <scott@griepentrog.com>
-
- * CHAOS: prevent sorcery object with null id
-
- When allocating a sorcery object, fail if the
- id value was not allocated.
-
- ASTERISK-25323
- Reported by: Scott Griepentrog
-
- Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e
-
-2015-08-14 15:46 +0000 [b719f56c72] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_sdp_rtp: Restore removed NULL check.
-
- When sending an RTP keepalive, we need to be sure we're not dealing with
- a NULL RTP instance. There had been a NULL check, but the commit that
- added the rtp_timeout and rtp_hold_timeout options removed the NULL
- check.
-
- Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64
-
-2015-08-13 12:30 +0000 [cea5dc7b8a] Richard Mudgett <rmudgett@digium.com>
-
- * audiohook.c: Simplify variable usage in audiohook_read_frame_both().
-
- Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c
-
-2015-08-13 12:22 +0000 [b3a56bee83] Richard Mudgett <rmudgett@digium.com>
-
- * audiohook.c: Fix MixMonitor crash when using the r() or t() options.
-
- The built frame format in audiohook_read_frame_both() is now set to a
- signed linear format before the rx and tx frames are duplicated instead of
- only for the mixed audio frame duplication.
-
- ASTERISK-25322 #close
- Reported by Sean Pimental
-
- Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538
-
-2015-08-12 12:59 +0000 [25af2d71c8] Kevin Harwell <kharwell@digium.com>
-
- * chan_sip.c: wrong peer searched in sip_report_security_event
-
- In chan_sip, after handling an incoming invite a security event is raised
- describing authorization (success, failure, etc...). However, it was doing
- a lookup of the peer by extension. This is fine for register messages, but
- in the case of an invite it may search and find the wrong peer, or a non
- existent one (for instance, in the case of call pickup). Also, if the peers
- are configured through realtime this may cause an unnecessary database lookup
- when caching is enabled.
-
- This patch makes it so that sip_report_security_event searches by IP address
- when looking for a peer instead of by extension after an invite is processed.
-
- ASTERISK-25320 #close
-
- Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4
-2015-08-13 05:26 +0000 [e18c300550] Joshua Colp <jcolp@digium.com>
-
- * res_http_websocket: When shutting down a session don't close closed socket
-
- Due to the use of ast_websocket_close in session termination it is
- possible for the underlying socket to already be closed when the
- session is terminated. This occurs when the close frame is attempted
- to be written out but fails.
-
- Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b
-2015-08-11 05:24 +0000 [b4e9416138] Joshua Colp <jcolp@digium.com>
-
- * res_http_websocket: Forcefully terminate on write errors.
-
- The res_http_websocket module will currently attempt to close
- the WebSocket connection if fatal cases occur, such as when
- attempting to write out data and being unable to. When the
- fatal cases occur the code attempts to write a WebSocket close
- frame out to have the remote side close the connection. If
- writing this fails then the connection is not terminated.
-
- This change forcefully terminates the connection if the
- WebSocket is to be closed but is unable to send the close frame.
-
- ASTERISK-25312 #close
-
- Change-Id: I10973086671cc192a76424060d9ec8e688602845
-
-2015-08-10 13:43 +0000 [256bc52b66] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.
-
- Pressing DTMF digits on a phone to go out on a DAHDI channel can result in
- the digit not being recognized or even heard by the peer.
-
- Phone -> Asterisk -> DAHDI/channel
-
- Turns out the DAHDI behavior with DTMF generation (and any other generated
- tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When
- Asterisk requests to start sending DTMF then DAHDI waits until its write
- buffer is empty before generating any samples for the DTMF tones. When
- Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI
- immediately stops generating the DTMF samples. As a result, the more
- samples there are in the DAHDI write buffer the shorter the time DTMF
- actually gets sent on the wire. If there are more samples in the write
- buffer than the time DTMF is supposed to be sent then no DTMF gets sent on
- the wire. With the "buffers=12,half" setting and each buffer representing
- 20 ms of samples then the DAHDI write buffer is going to contain around
- 120 ms of samples. For DTMF to be recognized by the peer the actual sent
- DTMF duration needs to be a minimum of 40 ms. Therefore, the intended
- duration needs to be a minimum of 160 ms for the peer to receive the
- minimum DTMF digit duration to recognize it.
-
- A simple and effective solution to work around the DAHDI behavior is for
- Asterisk to flush the DAHDI write buffer when sending DTMF so the full
- duration of DTMF is actually sent on the wire. When someone is going to
- send DTMF they are not likely to be talking before sending the tones so
- the flushed write samples are expected to just contain silence.
-
- * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting
- to send a DTMF digit.
-
- ASTERISK-25315 #close
- Reported by John Hardin
-
- Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a
-
-2015-08-05 14:21 +0000 [800e0ea48d] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi.c: Lock private struct for ast_write().
-
- There is a window of opportunity for DTMF to not go out if an audio frame
- is in the process of being written to DAHDI while another thread starts
- sending DTMF. The thread sending the audio frame could be past the
- currently dialing check before being preempted by another thread starting
- a DTMF generation request. When the thread sending the audio frame
- resumes it will then cause DAHDI to stop the DTMF tone generation. The
- result is no DTMF goes out.
-
- * Made dahdi_write() lock the private struct before writing to the DAHDI
- file descriptor.
-
- ASTERISK-25315
- Reported by John Hardin
-
- Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb
-
-2015-08-10 18:23 +0000 [c126afe18f] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.
-
- If the saved SUBSCRIBE message is not parseable for whatever reason then
- Asterisk could crash when libpjsip tries to parse the message and adds an
- error message to the parse error list.
-
- * Made ast_sip_create_rdata() initialize the parse error rdata list. The
- list is checked after parsing to see that it remains empty for the
- function to return successful.
-
- ASTERISK-25306
- Reported by Mark Michelson
-
- Change-Id: Ie0677f69f707503b1a37df18723bd59418085256
-
-2015-08-10 07:40 +0000 [f68c995bc9] Alexander Traud <pabstraud@compuserve.com>
-
- * chan_sip: Fix negotiation of iLBC 30.
-
- iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is
- supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5,
- only iLBC 30 is negotiated now.
-
- ASTERISK-25309 #close
-
- Change-Id: I92d724600a183eec3114da0ac607b994b1a793da
-
-2015-08-09 18:42 +0000 [8e194047ac] Matt Jordan <mjordan@digium.com>
-
- * res/res_format_attr_silk: Expose format attributes to other modules
-
- This patch adds the .get callback to the format attribute module, such
- that the Asterisk core or other third party modules can query for the
- negotiated format attributes.
-
- Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c
-
-2015-08-09 17:56 +0000 [a0f451c35e] Matt Jordan <mjordan@digium.com>
-
- * main/format: Add an API call for retrieving format attributes
-
- Some codecs that may be a third party library to Asterisk need to have
- knowledge of the format attributes that were negotiated. Unfortunately,
- when the great format migration of Asterisk 13 occurred, that ability
- was lost.
-
- This patch adds an API call, ast_format_attribute_get, to the core
- format API, along with updates to the unit test to check the new API
- call. A new callback is also now available for format attribute modules,
- such that they can provide the format attribute values they manage.
-
- Note that the API returns a void *. This is done as the format attribute
- modules themselves may store format attributes in any particular manner
- they like. Care should be taken by consumers of the API to check the
- return value before casting and dereferencing. Consumers will obviously
- need to have a priori knowledge of the type of the format attribute as
- well.
-
- Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3
-
-2015-08-07 22:11 +0000 [26f0559a94] David M. Lee <dlee@respoke.io>
-
- * Replace htobe64 with htonll
-
- We don't have a compatability function to fill in a missing htobe64; but
- we already have one for the identical htonll.
-
- Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac
-
-2015-08-07 14:20 +0000 [df9ce36366] Scott Emidy <jemidy@digium.com>
-
- * ARI: Retrieve existing log channels
-
- An http request can be sent to get the existing Asterisk logs.
-
- The command "curl -v -u user:pass -X GET 'http://localhost:8088
- /ari/asterisk/logging'" can be run in the terminal to access the
- newly implemented functionality.
-
- * Retrieve all existing log channels
-
- ASTERISK-25252
-
- Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808
-
-2015-08-07 11:14 +0000 [e9f1bc08cb] Scott Emidy <jemidy@digium.com>
-
- * ARI: Creating log channels
-
- An http request can be sent to create a log channel
- in Asterisk.
-
- The command "curl -v -u user:pass -X POST
- 'http://localhost:088/ari/asterisk/logging/mylog?
- configuration=notice,warning'" can be run in the terminal
- to access the newly implemented functionality for ARI.
-
- * Ability to create log channels using ARI
-
- ASTERISK-25252
-
- Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782
-
-2015-08-06 15:18 +0000 [78364132ce] Scott Emidy <jemidy@digium.com>
-
- * ARI: Deleting log channels
-
- An http request can be sent to delete a log channel
- in Asterisk.
-
- The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
- /ari/asterisk/logging/mylog'" can be run in the terminal
- to access the newly implemented functionally for ARI.
-
- * Able to delete log channels using ARI
-
- ASTERISK-25252
-
- Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6
-
-2015-08-06 12:48 +0000 [e25569ef95] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: More accurately persist packet.
-
- The pjsip_rx_data structure has a pkt_info.packet field on it that is
- the packet that was read from the transport. For datagram transports,
- the packet read from the transport will correspond to the SIP message
- that arrived. For streamed transports, however, it is possible to read
- multiple SIP messages in one packet.
-
- In a recent case, Asterisk crashed on a system where TCP was being used.
- This is because at some point, a read from the TCP socket resulted in a
- 200 OK response as well as an incoming SUBSCRIBE request being stored in
- rdata->pkt_info.packet. When the SUBSCRIBE was processed, the
- combination 200 OK and SUBSCRIBE was saved in persistent storage. Later,
- a restart of Asterisk resulted in the crash because the persistent
- subscription recreation code ended up building the 200 OK response
- instead of a SUBSCRIBE request, and we attempted to access
- request-specific data.
-
- The fix here is to use the pjsip_msg_print() function in order to
- persist SUBSCRIBE requests. This way, rather than using the raw socket
- data, we use the parsed SIP message that PJSIP has given us. If we
- receive multiple SIP messages from a single read, we will be sure only
- to save off the relevant SIP message. There also is a safeguard put in
- place to make sure that if we do end up reconstructing a SIP response,
- it will not cause a crash.
-
- ASTERISK-25306 #close
- Reported by Mark Michelson
-
- Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2
-
-2015-08-04 16:12 +0000 [8521a86367] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip: Ensure sanitized XML is NULL terminated.
-
- The ast_sip_sanitize_xml function is used to sanitize
- a string for placement into XML. This is done by examining
- an input string and then appending values to an output
- buffer. The function used by its implementation, strncat,
- has specific behavior that was not taken into account.
- If the size of the input string exceeded the available
- output buffer size it was possible for the sanitization
- function to write past the output buffer itself causing
- a crash. The crash would either occur because it was
- writing into memory it shouldn't be or because the resulting
- string was not NULL terminated.
-
- This change keeps count of how much remaining space is
- available in the output buffer for text and only allows
- strncat to use that amount.
-
- Since this was exposed by the res_pjsip_pidf_digium_body_supplement
- module attempting to send a large message the maximum allowed
- message size has also been increased in it.
-
- A unit test has also been added which confirms that the
- ast_sip_sanitize_xml function is providing NULL terminated
- output even when the input length exceeds the output
- buffer size.
-
- ASTERISK-25304 #close
-
- Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302
-
-2015-08-05 05:23 +0000 [9a12804e59] Joshua Colp <jcolp@digium.com>
-
- * res_rtp_asterisk: Don't leak temporary key when enabling PFS.
-
- A change recently went in which enabled perfect forward secrecy for
- DTLS in res_rtp_asterisk. This was accomplished two different ways
- depending on the availability of a feature in OpenSSL. The fallback
- method created a temporary instance of a key but did not free it.
- This change fixes that.
-
- ASTERISK-25265
-
- Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396
-2015-08-04 09:47 +0000 [27dc2094e9] Mark Michelson <mmichelson@digium.com>
-
- * res_http_websocket: Debug write lengths.
-
- Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a
- test failure observed on 32 bit test agents by ensuring that a cast from
- a 32 bit unsigned integer to a 64 bit unsigned integer was happening in
- a predictable place. As it turns out, this did not cause test runs to
- succeed.
-
- This commit adds several redundant debug messages that print the payload
- lengths of websocket frames. The idea here is that this commit will not
- cause tests to succeed for the faulty test agent, but we might deduce
- where the fault lies more easily this way by observing at what point the
- expected value (537) changes to some ungangly huge number.
-
- If you are wondering why something like this is being committed to the
- branch, keep in mind that in commit
- 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test
- failures only happen when automated tests are run. Attempts to run the
- tests by hand manually on the test agent result in the tests passing.
-
- Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d
-
-2015-08-03 11:06 +0000 [39cc28f6ea] Mark Michelson <mmichelson@digium.com>
-
- * res_http_websocket: Avoid passing strlen() to ast_websocket_write().
-
- We have seen a rash of test failures on a 32-bit build agent. Commit
- 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where
- we were not encoding a 64-bit value correctly over the wire. This
- commit, however, did not solve the test failures.
-
- In the failing tests, ARI is attempting to send a 537 byte text frame
- over a websocket. When sending a frame this small, 16 bits are all that
- is required in order to encode the payload length on the websocket
- frame. However, ast_websocket_write() thinks that the payload length is
- greater than 65535 and therefore writes out a 64 bit payload length.
- Inspecting this payload length, the lower 32 bits are exactly what we
- would expect it to be, 537 in hex. The upper 32 bits, are junk values
- that are not expected to be there.
-
- In the failure, we are passing the result of strlen() to a function that
- expects a uint64_t parameter to be passed in. strlen() returns a size_t,
- which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit
- unsigned value to somewhere where a 64-bit unsigned value is expected
- would cause no problems. In fact, in manual runs of failing tests, this
- works just fine. However, ast_websocket_write() uses the Asterisk
- optional API, which means that rather than a simple function call, there
- are a series of macros that are used for its declaration and
- implementation. These macros may be causing some sort of error to occur
- when converting from a 32 bit quantity to a 64 bit quantity.
-
- This commit changes the logic by making existing ast_websocket_write()
- calls use ast_websocket_write_string() instead. Within
- ast_websocket_write_string(), the 64-bit converted strlen is saved in a
- local variable, and that variable is passed to ast_websocket_write()
- instead.
-
- Note that this commit message is full of speculation rather than
- certainty. This is because the observed test failures, while always
- present in automated test runs, never occur when tests are manually
- attempted on the same test agent. The idea behind this commit is to fix
- a theoretical issue by performing changes that should, at the least,
- cause no harm. If it turns out that this change does not fix the failing
- tests, then this commit should be reverted.
-
- Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67
-
-2015-07-28 05:33 +0000 [aed068844c] Mark Duncan <mark@syon.co.jp>
-
- * res/res_rtp_asterisk: Add ECDH support
-
- This will add ECDH support to Asterisk. It will
- detect auto ECDH support in OpenSSL
- (1.0.2b and above) during ./configure. If this is
- available, it will use it,
- otherwise it will fall back to prime256v1 (this
- behavior is consistent with
- other projects such as Apache and nginx).
-
- This fixes WebRTC being broken in Firefox 38+ due
- to Firefox now only supporting
- ciphers with perfect forward secrecy.
-
- ASTERISK-25265 #close
-
- Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b
-
-2015-07-29 14:17 +0000 [1ae762634c] Benjamin Ford <bford@digium.com>
-
- * ARI: Rotate log channels.
-
- An http request can be sent to rotate a specified log channel.
- If the channel does not exist, an error response will be
- returned.
-
- The command "curl -v -u user:pass -X PUT 'http://localhost:8088
- /ari/asterisk/logging/logChannelName/rotate'" can be run in the
- terminal to access this new functionality.
-
- * Added the ability to rotate log files through ARI
-
- ASTERISK-25252
-
- Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
-
-2015-07-29 13:49 +0000 [aeeb170fc4] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.c: Fix performance issue with several channel drivers that use RTP.
-
- ast_rtp_codecs_get_payload() gets called once or twice for every received
- RTP frame so it would be nice to not allocate an ao2 object to then have
- it destroyed shortly thereafter. The ao2 object gets allocated only if
- the payload type is not set by the channel driver as a negotiated value.
- The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323.
-
- * Made static_RTP_PT[] an array of ao2 objects that
- ast_rtp_codecs_get_payload() can return instead of an array of structs
- that must be copied into a created ao2 object.
-
- ASTERISK-25296 #close
- Reported by: Richard Mudgett
-
- Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0
-
-2015-07-29 17:00 +0000 [84262749d2] Richard Mudgett <rmudgett@digium.com>
-
- * res_rtp_asterisk.c: Fix off-nominal crash potential.
-
- ASTERISK-25296
- Reported by: Richard Mudgett
-
- Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b
-
-2015-07-29 13:48 +0000 [1519eb44a7] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.c: Must protect mime_types_len with mime_types_lock.
-
- Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e
-
-2015-07-24 18:42 +0000 [a93b7a927c] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.
-
- Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2
-
-2015-07-24 18:38 +0000 [741fa0d26d] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_sdp_rtp.c: Fixup some whitespace.
-
- Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973
-
-2015-07-27 19:10 +0000 [89b21fd9a3] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.h: No sense allowing payload types larger than RFC allows.
-
- * Tweaked add_static_payload() to not use magic numbers.
-
- Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b
-
-2015-07-23 14:04 +0000 [7427c7f13b] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.c: Minor tweaks.
-
- * Fix off nominial ref leak of new_type in
- ast_rtp_codecs_payloads_set_m_type().
-
- * No need to lock static_RTP_PT_lock in
- ast_rtp_codecs_payloads_set_m_type() and
- ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type
- parameter sanity check.
-
- * No need to create ast_rtp_payload_type ao2 objects with a lock since the
- lock is not used.
-
- Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4
-
-2015-07-23 12:41 +0000 [e20f435b60] Richard Mudgett <rmudgett@digium.com>
-
- * rtp_engine.h: Misc comment fixes.
-
- Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43
-
-2015-07-17 16:23 +0000 [bc5d7f9c37] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip.c: Tweak glue->update_peer() parameter nil value.
-
- Change glue->update_peer() parameter from 0 to NULL to better indicate it
- is a pointer.
-
- Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd
-
-2015-07-30 17:05 +0000 [13eb491e35] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Fix crashes seen when call cancelled.
-
- Two testsuite tests crashed in the same place as a result of an INVITE
- being CANCELed.
-
- tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified
- tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp
-
- The session pointer is no longer in the inv->mod_data[session_module.id]
- location because the INVITE transaction has reached the terminated state.
-
- ASTERISK-25297 #close
- Reported by: Richard Mudgett
-
- Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427
-
-2015-07-29 14:35 +0000 [48698a5e21] Mark Michelson <mmichelson@digium.com>
-
- * res_http_websocket: Properly encode 64 bit payload
-
- A test agent was continuously failing all ARI tests when run against
- Asterisk 13. As it turns out, the reason for this is that on those test
- runs, for some reason we decided to use the super extended 64 bit
- payload length for websocket text frames instead of the extended 16 bit
- payload length. For 64-bit payloads, the expected byte order over the
- network is
-
- 7, 6, 5, 4, 3, 2, 1, 0
-
- However, we were sending the payload as
-
- 3, 2, 1, 0, 7, 6, 5, 4
-
- This meant that we were saying to expect an absolutely MASSIVE payload
- to arrive. Since we did not follow through on this expected payload
- size, the client would sit patiently waiting for the rest of the payload
- to arrive until the test would time out.
-
- With this change, we use the htobe64() function instead of htonl() so
- that a 64-bit byte-swap is performed instead of a 32 bit byte-swap.
-
- Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a
-
-2015-07-29 12:23 +0000 [10ba72a927] Mark Michelson <mmichelson@digium.com>
-
- * Add a test event for inband ringing.
-
- This event is necessary for the bridge_wait_e_options test to be able to
- confirm that ringing is being played on the local channel that runs the
- BridgeWait() application with the e(r) option.
-
- ASTERISK-25292 #close
- Reported by Kevin Harwell
-
- Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e
-
-2015-07-16 12:16 +0000 [8458b8d441] Jonathan Rose <jrose@digium.com>
-
- * holding_bridge: ensure moh participants get frames
-
- Currently, if a blank musiconhold.conf is used, musiconhold will fail
- to start for a channel going into a holding bridge with an anticipation
- of getting music on hold. That being the case, no frames will be written
- to the channel and that can pose a problem for blind transfers in PJSIP
- which may rely on frames being written to get past the REFER framehook.
- This patch makes holding bridges start a silence generator if starting
- music on hold fails and makes it so that if no music on hold functions
- are installed that the ast_moh_start function will report a failure so
- that consumers of that function will be able to respond appropriately.
-
- ASTERISK-25271 #close
-
- Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99
-
-2015-07-24 22:20 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.5.0-rc1 Released.
-
-2015-07-24 17:15 +0000 [a4b527393b] Matt Jordan <mjordan@digium.com>
-
- * Release summaries: Add summaries for 13.5.0-rc1
-
-2015-07-24 17:11 +0000 [158b0b8ebf] Matt Jordan <mjordan@digium.com>
-
- * .version: Update for 13.5.0-rc1
-
-2015-07-24 17:11 +0000 [a0a7650e34] Matt Jordan <mjordan@digium.com>
-
- * .lastclean: Update for 13.5.0-rc1
-
-2015-07-24 17:11 +0000 [4d238af086] Matt Jordan <mjordan@digium.com>
-
- * realtime: Add database scripts for 13.5.0-rc1
-
-2015-07-24 12:56 +0000 [f78a4b52b8] Matt Jordan <mjordan@digium.com>
-
- * Bump the ARI version to 1.8.0
-
- Due to backwards compatible changes, the ARI version should be bumped to
- 1.8.0 prior to the release of 13.5.0. Note that a previous patch already
- bumped the version of AMI for this release.
-
- Change-Id: I419033bfbbc0d3533a29ccb32b2981f39e0883e7
-
-2015-07-18 11:16 +0000 [2749721791] Joshua Colp <jcolp@digium.com>
-
- * pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
-
- This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
- endpoint options. These allow the channel to be hung up if RTP
- is not received from the remote endpoint for a specified number of
- seconds.
-
- ASTERISK-25259 #close
-
- Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
-
-2015-07-24 09:46 +0000 [b4e19e414a] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Add rtp_keepalive to sample config file.
-
- Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
-
-2015-07-23 13:11 +0000 [f635520527] Mark Michelson <mmichelson@digium.com>
-
- * Local channels: Alternate solution to ringback problem.
-
- Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a
- specific scenario involving local channels and a native local RTP bridge
- could result in ringback still being heard on a calling channel even
- after the call is bridged.
-
- That commit caused many tests in the testsuite to fail with alarming
- consequences, such as not sending DialBegin and DialEnd events, and
- giving incorrect hangup causes during calls.
-
- This commit reverts the previous commit and implements and alternate
- solution. This new solution involves only passing AST_CONTROL_RINGING
- frames across local channels if the local channel is in AST_STATE_RING.
- Otherwise, the frame does not traverse the local channels. By doing
- this, we can ensure that a playtones generator does not get started on
- the calling channel but rather is started on the local channel on which
- the ringing frame was initially indicated.
-
- ASTERISK-25250 #close
- Reported by Etienne Lessard
-
- Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39
-
-2015-07-22 12:24 +0000 [f509730cb9] Joshua Colp <jcolp@digium.com>
-
- * audiohook: Use manipulated frame instead of dropping it.
-
- Previous changes to sample rate support in audiohooks accidentally
- removed code responsible for allowing the manipulate audiohooks
- to work. Without this code the manipulated frame would be dropped
- and not used. This change restores it.
-
- ASTERISK-25253 #close
-
- Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13
-
-2015-07-22 09:46 +0000 [54b25c80c8] Mark Michelson <mmichelson@digium.com>
-
- * Local channels: Do not block control -1 payloads.
-
- Control frames with a -1 payload are used as a special signal to stop
- playtones generators on channels. This indication is sent both by
- app_dial as well as by ast_answer() when a call is answered in case any
- tones were being generated on a calling channel.
-
- This control frame type was made to stop traversing local channel pairs
- as an optimization, because it was thought that it was unnecessary to
- send these indications, and allowing such unnecessary control frames to
- traverse the local channels would cause the local channels to optimize
- away less quickly.
-
- As it turns out, through some special magic dialplan code, it is
- possible to have a tones being played on a non-local channel, and it is
- important for the local channel to convey that the tones should be
- stopped. The result of having tones continue to be played on the
- non-local channel is that the tones play even once the channel has been
- bridged. By not blocking the -1 control frame type, we can ensure that
- this situation does not happen.
-
- ASTERISK-25250 #close
- Reported by Etienne Lessard
-
- Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815
-
-2015-07-22 05:16 +0000 [f1493f900e] Joshua Colp <jcolp@digium.com>
-
- * audiohook: Read the correct number of samples based on audiohook format.
-
- Due to changes in audiohooks to support different sample rates the
- underlying storage of samples is in the format of the audiohook
- itself and not of the format being requested. This means that if a
- channel is using G722 the samples stored will be at 16kHz. If
- something subsequently reads from the audiohook at a format which
- is not the same sample rate as the audiohook the number of samples
- needs to be adjusted.
-
- Given the following example:
- 1. Channel writing into audiohook at 16kHz (as it is using G722).
- 2. Chanspy reading from audiohook at 8kHz.
-
- The original code would read 160 samples from the audiohook for
- each 20ms of audio. This is incorrect. Since the audio in the
- audiohook is at 16kHz the actual number needing to be read is 320.
- Failure to read this much would cause the audiohook to reset
- itself constantly as the buffer became full.
-
- This change adjusts the requested number of samples by determining
- the duration of audio requested and then calculating how many
- samples that would be in the audiohook format.
-
- ASTERISK-25247 #close
-
- Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d
-
-2015-07-20 12:39 +0000 [62c64c3bd1] Rusty Newton <rnewton@digium.com>
-
- * Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c
-
- * In sip.conf.sample fix sentence where we said that WS or WSS are supported
- transports for use in an outbound register definition. They are not
- supported in that case.
- * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used
- to enable CDR on a channel.
-
- ASTERISK-24867 #close
- Reported by: Rusty Newton
-
- ASTERISK-24853 #close
- Reported by: PSDK
-
- Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca
-
-2015-07-09 14:17 +0000 [d9094ddd73] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Add rtp_keepalive endpoint option.
-
- This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
- chan_sip option, this specifies an interval, in seconds, at which we
- will send RTP comfort noise frames. This can be useful for keeping RTP
- sessions alive as well as keeping NAT associations alive during lulls.
-
- ASTERISK-25242 #close
- Reported by Mark Michelson
-
- Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b
-
-2015-07-16 09:13 +0000 [a23adcca3d] Michael Cargile <mikec@vicidial.com>
-
- * res/res_musiconhold: Add a warning when MOH does not exist
-
- Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b
-
-2015-07-19 09:11 +0000 [03064daeb2] Matt Jordan <mjordan@digium.com>
-
- * res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf
-
- Misconfiguring sorcery.conf with a 'config' wizard with no extra data
- will currently crash Asterisk on startup, as the wizard requires a comma
- delineated list to parse. This patch updates res_sorcery_config to check
- for the presence of the data before it starts manipulating it.
-
- Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847
-
-2015-07-16 09:46 +0000 [2c626ceb64] Joshua Colp <jcolp@digium.com>
-
- * chan_pjsip: Don't change formats when frame of unsupported format is received.
-
- Receipt of an RTP packet currently causes the formats on an PJSIP channel to
- change to the format of the RTP packet. In some off-nominal cases it's possible
- for this to be a format that has not been configured or negotiated. This change
- makes it so only formats explicitly configured on the endpoint are allowed.
-
- ASTERISK-25258 #close
-
- Change-Id: If93d641fb6418a285928839300d7854cab8c1020
-
-2015-07-17 04:59 +0000 [abb14ac5b8] Patric Marschall <patric.marschall@1und1.de>
-
- * sig_pri.h: force_restart_unavailable_chans in wrong scope
-
- In channels/sig_pri.h, struct sig_pri_span, the field
- force_restart_unavailable_chans is only defined if
-
- #if defined(HAVE_PRI_MCID) is true.
-
- All other occurences of force_restart_unavailable_chans are outside of the
-
- #if defined(HAVE_PRI_MCID)
- endif
-
- scope.
-
- ASTERISK-25257 #close
- Reported by: Patric Marschall
-
- Change-Id: I071de89cc2cd0d85927a013036e235851f672549
-2015-07-14 16:55 +0000 [875aee4c09] Richard Mudgett <rmudgett@digium.com>
-
- * pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable.
-
- ASTERISK-25256 #close
- Reported by: Richard Mudgett
-
- Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3
-
-2015-07-08 16:39 +0000 [8bcf6d2801] Matt Jordan <mjordan@digium.com>
-
- * ARI: Add support for push configuration of dynamic object
-
- This patch adds support for push configuration of dynamic, i.e.,
- sorcery, objects in Asterisk. It adds three new REST API calls to the
- 'asterisk' resource:
- * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current
- object given its ID. This returns back a list of ConfigTuples, which
- define the fields and their present values that make up the object.
- * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an
- object. A body may be passed with the request that contains fields to
- populate in the object. The same format as what is retrieved using
- the GET operation is used for the body, save that we specify that the
- list of fields to update are contained in the "fields" attribute.
- * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic
- object from its backing storage.
-
- Note that the success/failure of these operations is somewhat
- configuration dependent, i.e., you must be using a sorcery wizard that
- supports the operation in question. If a sorcery wizard does not support
- the create or delete mechanisms, then the REST API call will fail with a
- 403 forbidden.
-
- ASTERISK-25238 #close
-
- Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c
-
-2015-07-15 15:40 +0000 [e31cb6b248] Richard Mudgett <rmudgett@digium.com>
-
- * strings.h: Fix issues with escape string functions.
-
- Fixes for issues with the ASTERISK-24934 patch.
-
- * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is
- an empty string. If it were an empty string the functions returned NULL
- as if there were a memory allocation failure. This failure caused the AMI
- VarSet event to not get posted if the new value was an empty string.
-
- * Fixed dest buffer overwrite potential in ast_escape() and
- ast_escape_c(). If the dest buffer size is smaller than the space needed
- by the escaped s parameter string then the dest buffer would be written
- beyond the end by the nul string terminator. The num parameter was really
- the dest buffer size parameter so I renamed it to size.
-
- * Made nul terminate the dest buffer if the source string parameter s was
- an empty string in ast_escape() and ast_escape_c().
-
- * Updated ast_escape() and ast_escape_c() doxygen function description
- comments to reflect reality.
-
- * Added some more unit test cases to /main/strings/escape to cover the
- empty source string issues.
-
- ASTERISK-25255 #close
- Reported by: Richard Mudgett
-
- Change-Id: Id77fc704600ebcce81615c1200296f74de254104
-
-2015-07-14 14:29 +0000 [243c0d1609] Richard Mudgett <rmudgett@digium.com>
-
- * parking_applications.c: Fix ast_verb() line terminator.
-
- Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775
-
-2015-07-14 14:36 +0000 [c782320c68] Richard Mudgett <rmudgett@digium.com>
-
- * res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park.
-
- setup_park_common_datastore() was assuming that a non-NULL string returned
- for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty
- strings. Things got crashy as a result.
-
- * Made setup_park_common_datastore() treat the channel variable values the
- same whether they are NULL or empty for ATTENDEDTRANSFER and
- BLINDTRANSFER.
-
- ASTERISK-25254 #close
- Reported by: Richard Mudgett
-
- Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2
-
-2015-07-10 18:01 +0000 [2735dd5b2d] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer().
-
- Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb
-
-2015-07-10 10:42 +0000 [3d0ca343ca] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Add some helpful comments and minor tweaks.
-
- Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743
-
-2015-07-10 10:43 +0000 [8d08bb179c] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Fix off nominal crash potential in debug message.
-
- Change-Id: I09928297927ee85f7655289acee3a586816466bc
-
-2015-07-15 10:31 +0000 [0a1a550593] Matt Jordan <mjordan@digium.com>
-
- * apps/app_dictate: Fix typo in attribution
-
- Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian
- (GameGamer43) for pointing that out.
-
- Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106
-
-2015-07-15 10:28 +0000 [3384e64ef6] Benjamin Ford <bford@digium.com>
-
- * ARI: Fixed unload mode for unload module.
-
- Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM,
- which would unload a module even if it was in use.
-
- * Changed unload mode to proper mode
-
- ASTERISK-25173
-
- Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533
-
-2015-07-08 16:38 +0000 [0b6ff77afb] Matt Jordan <mjordan@digium.com>
-
- * res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails
-
- Having a debug message tell us that we attempted to look up an item but
- failed is nice in circumstances when it isn't clear if the wizard was
- queried correctly or not.
-
- Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7
-
-2015-07-08 16:37 +0000 [2f0d6d346c] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_outbound_registration: Fix WARNING message
-
- Newlines are nice.
-
- Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42
-
-2015-07-08 16:35 +0000 [cd2213f1ae] Matt Jordan <mjordan@digium.com>
-
- * res_pjsip/configuration: Fix a variety of default value problems
-
- This patch fixes some bad default value handling in the following
- settings:
-
- * The 'message_context' and 'accountcode' settings are not mandatory. As
- such, we can allow their stringfield values to be empty.
- * The 'media_encryption' setting applies a default value of 'none' to
- the setting, which it then can't parse or understand. Since the value
- is documented to be 'no', this will now apply that as the default
- value.
-
- Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83
-
-2015-07-08 16:32 +0000 [2e4bdbd78a] Matt Jordan <mjordan@digium.com>
-
- * main/sorcery: Provide log messages when a wizard does not support an operation
-
- If a sorcery wizard does not support one of the 'optional' CRUD
- operations (namely the CUD), log a WARNING message so we are aware of
- why the operation failed. This also removes an assert in this case, as
- the CUD operation may have been triggered by an external system, in
- which case it is not a programming error but a configuration error.
-
- Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53
-
-2015-07-10 18:17 +0000 [653f2087e0] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session.c: Fix crash on call disconnect.
-
- The crash fix for ASTERISK-25183 backported some code from master to try
- to make sure that a BYE response is processed by the same serializer used
- by the BYE request. The identified race condition causing that backport
- was the BYE request code had not finished processing after sending the BYE
- before the BYE response came in for processing under a different thread.
- Unfortunately, there is still a race condition. Now the race condition is
- between destroying the call session's serializer in
- ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a
- reference to the serializer for a BYE response. Even worse, the new race
- condition is a design limitation of the taskprocessor implementation that
- didn't matter in versions before v12. Back then, taskprocessors were only
- destroyed when a module unloaded. Now res_pjsip can destroy them when a
- call ends.
-
- However, as noted on the ASTERISK-25183 commit,
- session_inv_on_state_changed() is disassociating the dialog from the
- session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED.
- This is a tad too soon because our BYE request transaction has not
- completed yet.
-
- * Split session_end() that is called by session_inv_on_state_changed() to
- hold off session destruction until the BYE transaction timeout occurs or a
- failed initial INVITE transaction timeout occurs in
- session_inv_on_tsx_state_changed().
-
- ASTERISK-25201 #close
- Reported by: Matt Jordan
-
- Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961
-
-2015-07-14 13:12 +0000 [1aafadf814] Benjamin Ford <bford@digium.com>
-
- * ARI: Added new functionality to reload a single module.
-
- An http request can be sent to reload an Asterisk module. If the
- module can not be reloaded or is not already loaded, an error
- response will be returned.
-
- The command "curl -v -u user:pass -X PUT 'http://localhost:8088
- /ari/asterisk/modules/{moduleName}'" (or something similar, based
- on configuration) can be run in the terminal to access this new
- functionality.
-
- For more information, see:
- https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
-
- * Added new ARI functionality
- * Asterisk modules can be reloaded through http requests
-
- ASTERISK-25173
-
- Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1
-
-2015-07-14 08:55 +0000 [9dcae23cfc] Benjamin Ford <bford@digium.com>
-
- * ARI: Added new functionality to unload a single module.
-
- An http request can be sent to unload an Asterisk module. If the
- module can not be unloaded or is already unloaded, an error response
- will be returned.
-
- The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
- /ari/asterisk/modules/{moduleName}'" (or something similar, depending
- on configuration) can be run in the terminal to access this new
- functionality.
-
- For more information, see:
- https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
-
- * Added new ARI functionality
- * Asterisk modules can be unloaded through http requests
-
- ASTERISK-25173
-
- Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57
-
-2015-07-13 16:00 +0000 [c219a98d2b] Benjamin Ford <bford@digium.com>
-
- * ARI: Added new functionality to load a single module.
-
- An http request can be sent to load an Asterisk module. If the
- module can not be loaded or is loaded already, an error response
- will be returned.
-
- The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
- /asterisk/modules/{moduleName}'" (or something similar, depending on
- configuration) can be run in the terminal to access this new
- functionality.
-
- For more information, see:
- https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
-
- * Added new ARI functionality
- * Asterisk modules can be loaded through http requests
-
- ASTERISK-25173
-
- Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33
-
-2015-07-13 10:54 +0000 [73e35d20de] Benjamin Ford <bford@digium.com>
-
- * ARI: Added new functionality to get information on a single module.
-
- An http request can be sent to retrieve information on a single
- module, including the resource name, description, use count, status,
- and support level.
-
- The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
- /asterisk/modules/{moduleName}'" (or something similar, depending on
- configuration) can be run in the terminal to access this new
- functionality.
-
- For more information, see:
- https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
-
- * Added new ARI functionality
- * Information on a single module can now be retrieved
-
- ASTERISK-25173
-
- Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463
-
-2015-07-08 14:56 +0000 [97ee0ee6c6] Kevin Harwell <kharwell@digium.com>
-
- * bridge.c: Fixed race condition during attended transfer
-
- During an attended transfer a thread is started that handles imparting the
- bridge channel. From the start of the thread to when the bridge channel is
- ready exists a gap that can potentially cause problems (for instance, the
- channel being swapped is hung up before the replacement channel enters the
- bridge thus stopping the transfer). This patch adds a condition that waits
- for the impart thread to get to a point of acceptable readiness before
- allowing the initiating thread to continue.
-
- ASTERISK-24782
- Reported by: John Bigelow
-
- Change-Id: I08fe33a2560da924e676df55b181e46fca604577
-
-2015-07-08 16:28 +0000 [bb76b88baf] Matt Jordan <mjordan@digium.com>
-
- * main/sorcery: Don't fail object set creation from JSON if field fails
-
- Some individual fields may fail their conversion due to their default
- values being invalid for their custom handlers. In particular,
- configuration values that depend on others being enabled (and thus have
- an empty default value) are notorious for tripping this routine up. An
- example of this are any of the DTLS options for endpoints. Any of the
- DTLS options will fail to be applied (as DTLS is not enabled), causing
- the entire object set to be aborted.
-
- This patch makes it so that we log a debug message when skipping a
- field, and rumble on anyway.
-
- ASTERISK-25238
-
- Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76
-
-2015-07-08 16:21 +0000 [5f13c2226a] Matt Jordan <mjordan@digium.com>
-
- * main/format_cap: Parse capabilities generated by ast_format_cap_get_names
-
- We have a strange relationship between the parsing of format
- capabilities from a string and their representation as a string. We
- expect the format capabilities to be expressed as a string in the
- following format:
-
- allow = !all,ulaw,alaw
- disallow = g722
-
- While we would generate the string representation of those formats as:
-
- allow = (ulaw|alaw)
- disallow = (ulaw|alaw|g729...)
-
- When the configuration framework needs to store values as a string, it
- generates the format capabilities using the second representation; this
- representation however cannot be parsed when the entry is rehydrated.
- This patch fixes that by updating
- ast_format_cap_update_by_allow_disallow to parse an entry as if it were
- in the generated format if it has a leading '(' and a trailing ')'.
-
- ASTERISK-25238
-
- Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca
-
-2015-06-27 17:53 +0000 [2325b106fd] Matt Jordan <mjordan@digium.com>
-
- * tests/test_devicestate: Add additional tests for the device state API
-
- This patch adds more tests that exercise the device state API. This includes:
-
- * Tests that cover adding a device state provider, as well as deleting a
- device state provider. This also verifies that you cannot add an
- already added device state provider, and cannot delete an already
- deleted device state provider.
- * A test that covers changing device state and receiving said updates
- from a device state subscriber. This also covers hitting both the
- device state cache as well as a custom device state provider.
- * A test that covers converting device state to channel state and device
- state values to a string representation and back.
- * A test that covers obtaining device state from an active channel and a
- channel driver that provides its own device state.
-
- Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d
-
-2015-06-27 17:51 +0000 [328f0be806] Matt Jordan <mjordan@digium.com>
-
- * main/devicestate: Prevent duplicate registration of device state providers
-
- Currently, the device state provider API will allow you to register a
- device state provider with the same case insensitive name more than
- once. This could cause strange issues, as the duplicate device state
- providers will not be queried when a device's state has to be polled.
- This patch updates the API such that a device state provider with the
- same name as one that has already registered will be rejected.
-
- Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2
-
-2015-07-10 22:25 +0000 [bee41eec62] Matt Jordan <mjordan@digium.com>
-
- * res/res_sorcery_memory_cache: Fix test registration issues
-
- Again, tests now need to not end with a newline. This patch makes it so
- the tests can register again, unit tests will actually pass, and we can
- stop wasting time trying to figure out why builds are failing when they
- really aren't failing.
-
- Change-Id: Ide519fbeba89f413c733446c5ff7b224fc4ce840
-
-2015-07-10 21:42 +0000 [4d738e9026] Matt Jordan <mjordan@digium.com>
-
- * tests/test_sorcery_memory_cache_thrash: Fix test loading problems
-
- Because unit tests now want descriptions to not end with a newline, the
- sorcery memory cache thrash tests failed to register. This patch
- corrects their descriptions.
-
- Change-Id: Id004b1becfdeed8ee3c846f49beab76a5c0f68b6
-
-2015-06-26 10:57 +0000 [47ea312b24] Benjamin Ford <bford@digium.com>
-
- * ARI: Added new functionality to get all module information.
-
- An http request can be sent to retrieve a list of all existing modules,
- including the resource name, description, use count, status, and
- support level.
-
- The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
- asterisk/modules" (or something similar, depending on configuration)
- can be run in the terminal to access this new functionality.
-
- For more information, see:
- https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
-
- * Added new ARI functionality
- * Information on modules can now be retrieved
-
- Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0
-
-2015-07-09 09:18 +0000 [d558b00c85] Joshua Colp <jcolp@digium.com>
-
- * bridge_native_rtp.c: Don't start native RTP bridging after attended transfer.
-
- The bridge_native_rtp module adds a frame hook to channels which are in
- a native RTP bridge. This frame hook is used to intercept when a hold
- or unhold frame traverses the bridge so native RTP can be stopped or
- started as appropriate. This is expected but exposes a specific bug
- when attended transfers are involved.
-
- Upon completion of an attended transfer an unhold frame is queued up
- to take one of the channels involved off hold. After this is done
- the channel is moved between bridges.
-
- When the frame hook is involved in this case for the unhold it
- releases the channel lock and acquires the bridge lock. This
- allows the bridge core to step in and move the channel
- (potentially changing the bridging techology) from another thread.
- Once completed the bridge lock is released by the bridge core.
- The frame hook is then able to acquire the bridge lock and
- wrongfully starts native RTP again, despite the channel no longer
- being in the bridge or needing to start native RTP. In fact at
- this point the frame hook is no longer attached to the channel.
-
- This change makes it so the native RTP bridge data is available to
- the frame hook when it is invoked. Whether the frame hook has
- been detached or not is stored on the native RTP bridge data and
- is checked by the frame hook before starting or stopping native
- RTP bridging. If the frame hook has been detached it does nothing.
-
- ASTERISK-25240 #close
-
- Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2
-
-2015-05-16 17:02 +0000 [b74b071369] Joshua Colp <jcolp@digium.com>
-
- * res_sorcery_memory_cache: Backport to 13
-
- Gerrit is complaining of conflicts when trying to create a patch series
- of all of the cherry-picked master commits, so I have instead squashed
- it all into one commit.
-
- ASTERISK-25067 #close
- Reported by: Matt Jordan
-
- Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9
-
-2015-07-08 04:21 +0000 [7ff1ac8797] Joshua Colp <jcolp@digium.com>
-
- * res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used.
-
- This change fixes a bug where the DTLS timeout timer would be
- initialized to 0 if DTLS was not used for an RTP session.
-
- ASTERISK-25103
-
- Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac
-
-2015-07-01 07:55 +0000 [05e8e14982] Joshua Colp <jcolp@digium.com>
-
- * res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.
-
- This change moves logic for setting up the DTLS SSL contexts to
- when the SDP is done being processed instead of when ICE negotiation
- completes. It also stops handshakes from being initiated when we
- are acting as a server.
-
- Manipulating the SSL context when ICE negotiation has completed
- is problematic as the SSL context is not protected and if acting
- as a client the remote side may have started DTLS negotiation
- already.
-
- The retransmission timeout timer code has also been split up
- and simplified some. Both RTP and RTCP now have their own timers
- and the points at which the timer is stopped and started is now
- more specific. When a packet is sent the timer is started. When
- a response is received but before it is processed the timer is
- stopped. This provides a guarantee that the timeout is not
- occurring while the response is processed.
-
- ASTERISK-22805 #close
- ASTERISK-24550 #close
- ASTERISK-24651 #close
- ASTERISK-24832 #close
- ASTERISK-25103 #close
- ASTERISK-25127 #close
-
- Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91
-
-2015-06-26 16:10 +0000 [38bace4fbb] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_t38.c: Fix always false if test.
-
- Calling t38_change_state() sets the t38 state so it makes little sense to
- then check the state right after the call for something else.
-
- * Made the code in t38_interpret_parameters() reject or exit T.38 mode as
- intended but not implemented.
-
- Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2
-
-2015-06-30 11:17 +0000 [2f7688c788] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str().
-
- Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907
-
-2015-06-30 11:14 +0000 [74be3a50d7] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_mwi.c: Eliminate a simple RAII_VAR.
-
- Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da
-
-2015-06-30 11:11 +0000 [589e93617a] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_mwi.c: Fix mid-line log message line breaks.
-
- * Add create_mwi_subscriptions_for_endpoint() doxygen comment.
-
- Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2
-
-2015-06-26 18:48 +0000 [0d67e04359] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_mwi.c: Fix MWI subscription memory corruption crash.
-
- MWI subscriptions can crash or corrupt memory when using the subscription
- datastore to access the MWI subscription object because the datastore is
- not holding a reference to the object.
-
- * Give the subscription datastore a ref to the MWI subscription object.
- It is unfortunate that the ref causes a circular ref chain that must be
- explicitly broken to allow the memory to get released. The loop is broken
- when the subscription is shutdown and if the subscription setup fails.
-
- ASTERISK-25168 #close
- Reported by: Carl Fortin
-
- Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f
-
-2015-07-02 14:51 +0000 [0422433f47] Richard Mudgett <rmudgett@digium.com>
-
- * PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error.
-
- When res_pjsip body generator modules were generating XML or XPIDF
- response bodies, there was a chance that the generated body would be the
- exact size of the supplied buffer. Adding the nul string terminator would
- then write beyond the end of the buffer and potentially corrupt memory.
-
- * Fix MALLOC_DEBUG high fence violations caused by adding a nul string
- terminator on the end of a buffer for XML or XPIDF response bodies.
-
- * Made calls to pj_xml_print() safer if the XML prolog is requested. Due
- to a bug in pjproject, the return value could be -1 _or_
- AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
-
- * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
- return value of pj_xml_print() when the supplied buffer is not large
- enough.
-
- ASTERISK-25168
- Reported by: Carl Fortin
-
- Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de
-
-2015-06-26 10:36 +0000 [8ea214aed7] Richard Mudgett <rmudgett@digium.com>
-
- * PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences.
-
- When a caller calls a FAX number and then hangs up right after the call is
- answered then the T.38 re-INVITE automatic reject timer may still be
- running after the channel goes away.
-
- * Added session NULL channel checks on the code paths that get executed by
- t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
- automatic reject timer expires.
-
- ASTERISK-25168
- Reported by: Carl Fortin
-
- Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403
-
-2015-06-05 15:37 +0000 [ada7346792] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip: Need to use the same serializer for a pjproject SIP transaction.
-
- All send/receive processing for a SIP transaction needs to be done under
- the same threadpool serializer to prevent reentrancy problems inside
- pjproject and res_pjsip.
-
- * Add threadpool API call to get the current serializer associated with
- the worker thread.
-
- * Pick a serializer from a pool of default serializers if the caller of
- res_pjsip.c:ast_sip_push_task() does not provide one.
-
- This is a simple way to ensure that all outgoing SIP request messages are
- processed under a serializer. Otherwise, any place where a pushed task is
- done that would result in an outgoing out-of-dialog request would need to
- be modified to supply a serializer. Serializers from the default
- serializer pool are picked in a round robin sequence for simplicity.
-
- A side effect is that the default serializer pool will limit the growth of
- the thread pool from random tasks. This is not necessarily a bad thing.
-
- * Made pjsip_distributor.c save the thread's serializer name on the
- outgoing request tdata struct so the response can be processed under the
- same serializer.
-
- This is a cherry-pick from master.
-
- **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a
-
- NOTE: session_inv_on_state_changed() is disassociating the dialog from the
- session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED.
- Unfortunately this is a tad too soon because our BYE request transaction
- has not completed yet.
-
- ASTERISK-25183 #close
- Reported by: Matt Jordan
-
- Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a
-
-2015-07-04 18:22 +0000 [55137c3d12] Joshua Colp <jcolp@digium.com>
-
- * res/res_http_websocket: Don't send HTTP response fragmented.
-
- This change makes it so that when accepting a WebSocket
- connection the HTTP response is sent as one packet instead of
- fragmented. Browsers don't like it when you send it fragmented.
-
- ASTERISK-25103
-
- Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69
-
-2015-06-27 18:47 +0000 [49f81ddb85] Matt Jordan <mjordan@digium.com>
-
- * Makefile: Remove coverage files on 'make clean'
-
- This patch updates a variety of Makefiles in Asterisk's build system to
- remove .gcda and .gcno files when 'make clean' is executed. These files
- are generated when '--enable-coverage' is passed to the Asterisk
- configure script.
-
- Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
-
-2015-07-02 09:08 +0000 [e0f565663b] Walter Doekes <walter+asterisk@wjd.nu>
-
- * chan_sip: Fix early call pickup channel leak.
-
- When handle_invite_replaces() was called, and either ast_bridge_impart()
- failed or there was no bridge (because the channel we're picking up was
- still ringing), chan_sip would leak a channel.
-
- Thanks Matt and Corey for checking the bridge path.
-
- ASTERISK-25226 #close
-
- Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8
-
-2015-07-02 06:19 +0000 [a5a262be78] Walter Doekes <walter+asterisk@wjd.nu>
-
- * chan_mgcp: Don't call close on fd -1.
-
- ASTERISK-25220 #close
-
- Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3
-
-2015-07-02 06:10 +0000 [b835312b4c] Walter Doekes <walter+asterisk@wjd.nu>
-
- * rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format.
-
- When running valgrind on Asterisk, it complained about:
-
- ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304)
- ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...)
- ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292)
- ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437)
-
- The code in question is a struct assignment, which may be performed by
- memcpy as a compiler optimization. It is changed to only copy the struct
- contents if source and destination are different.
-
- ASTERISK-25219 #close
-
- Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a
-
-2015-07-02 05:16 +0000 [6551e16e03] Walter Doekes <walter+asterisk@wjd.nu>
-
- * astfd: Fix buffer overflow in DEBUG_FD_LEAKS.
-
- If DEBUG_FD_LEAKS was used and more file descriptors than the default of
- 1024 were available, some DEBUG_FD_LEAKS-patched functions would
- overwrite memory past the fixed-size (1024) fdleaks buffer.
-
- This change:
- - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe
- - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024
- - stores pointers to constants instead of copying the contents
- - reorders the fdleaks struct for possibly tighter packing
- - adds a tiny bit of documentation
-
- ASTERISK-25212 #close
-
- Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5
-
-2015-07-02 04:57 +0000 [f4dd9560cf] Walter Doekes <walter+asterisk@wjd.nu>
-
- * res_timing: Don't close FD 0 when out of open files.
-
- This fixes so a failure to get a timer file descriptor does not cascade
- to closing FD 0.
-
- On error, both res_timing_kqueue and res_timing_timerfd would call the
- destructor before setting the file handle. The file handle had been
- initialized to 0, causing FD 0 to be closed. This in turn, resulted in
- floods of "CLI>" messages and an unusable terminal.
-
- ASTERISK-19277 #close
- Reported by: Barry Chern
-
- For the 13 branch, this was already fixed. This patch only ensures that
- we do not attempt to close a negative file descriptor.
-
- Change-Id: I147d7e33726c6e5a2751928d56561494f5800350
-
-2015-07-01 17:25 +0000 [78a1f4aa46] Richard Mudgett <rmudgett@digium.com>
-
- * chan_vpb.cc: Fix compiler warning Jenkins found.
-
- Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0
-
-2015-07-01 13:34 +0000 [6b16fbfc22] Scott Griepentrog <scott@griepentrog.com>
-
- * Channel alert pipe: improve diagnostic error return
-
- When a frame is queued on a channel, any failure in
- ast_channel_alert_write is logged along with errno.
-
- This change improves the diagnostic message through
- aligning the errno value with actual failure cases.
-
- ASTERISK-25224
- Reported by: Andrey Biglari
-
- Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b
-
-2015-07-01 16:04 +0000 [8e07ab145d] Matt Jordan <mjordan@digium.com>
-
- * sorcery/realtime: Add a bit of debug and warning messages for bad configs
-
- When a mapping does not exist between a sorcery.conf defined object and
- a realtime mapping in extconf, currently, the user will receive a slew
- of ERROR messages that don't really tell what is happening. Some ERROR
- messages may even be misleading, as they occur after the sorcery API has
- already given up on the attempt to load and create the sorcery object.
-
- This patch adds a bit of debug and a useful WARNING message for when a
- wizard's open callback fails for a particular object type. In the bad
- configurations that resulted in this patch, this provided a 'root cause'
- WARNING message that pointed in the right direction of the configuration
- problem.
-
- Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b
-2015-06-29 12:45 +0000 [156395e743] Mark Michelson <mmichelson@digium.com>
-
- * res_sorcery_realtime: Fix leak of sorcery object type.
-
- This prevents a leak of a sorcery object type when realtime sorcery
- objects are retrieved by fields or when multiple objects are retrieved.
-
- The extent of this leak is that sorcery object types would be leaked.
- These are allocated whenever an object type is registered with sorcery,
- meaning that on module shutdown, these objects would be leaked. This
- could be problematic if many reloads were performed, but it is not as
- severe as if every sorcery object retrieved from realtime were being
- leaked.
-
- ASTERISK-25165 #close
- Reported by Corey Farrell
-
- Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8
-
-2015-06-26 22:02 +0000 [a5e9c4e9b2] Matt Jordan <mjordan@digium.com>
-
- * res/res_corosync: Always decline module load, instead of failing
-
- Returns a 'failure' from the module load routine indicates to Asterisk
- that it should abort loading completely. This is rarely - in fact,
- really, never - a good option. Aborting load of Asterisk from a dynamic
- module implies that the core, and the rest of the dynamic modules, don't
- matter: we should abandon all processing.
-
- res_corosync is really not that important.
-
- This patch updates the module such that, if it fails to load, it
- politely declines (emitting ERROR messages along the way), and allows
- Asterisk to continue to function.
-
- Note that this issue was keeping Asterisk unit tests from running on
- certain build agents.
-
- Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f
-
-2015-06-26 20:38 +0000 [399cd8bcd9] Matt Jordan <mjordan@digium.com>
-
- * main/pbx: Resolve case sensitivity regression in PBX hints
-
- When 8297136f was merged for ASTERISK-25040, a regression was introduced
- surrounding the case sensitivity of device names within hints.
- Previously, device names - such as 'sip/foo' - were compared in a case
- insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
- that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
- As a result, some dialplan hints stopped working.
-
- This patch re-introduces case insensitive matching for device names in
- hints.
-
- ASTERISK-25040
-
- ASTERISK-25202 #close
-
- Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c
- (cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03)
-
-2015-06-26 16:12 +0000 [24eec5a10b] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_nat: Adjust when contact should be rewritten.
-
- A previous change made the contact only get rewritten if the dialog's
- route set was not marked frozen. Unfortunately, while the intent of this
- is correct, the dialog's route set actually gets marked as frozen
- earlier than expected, especially for UAS dialogs.
-
- Instead, the idea is that the contact needs to not be rewritten if there
- is a pre-existing route set on the dialog. This is now accomplished by
- checking the dialog's route set list instead of checking if the route
- set is frozen.
-
- Doing this causes some broken tests to begin passing again.
-
- ASTERISK-25196
- Reported by Mark Michelson
-
- Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e
-
-2015-06-19 18:27 +0000 [0ec461a637] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Add a serializer shutdown group.
-
- The client_state objects contain a serializer used to send the outbound
- REGISTER messages. Once all those message transactions are complete then
- the module can shutdown.
-
- ASTERISK-24907 #close
- Reported by: Kevin Harwell
-
- Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547
-
-2015-06-26 10:41 +0000 [05a2cc1293] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_refer: Prevent sending duplicate headers.
-
- res_pjsip_refer will attempt to add Referred-By or Replaces headers to
- outbound INVITEs at times. If the INVITE gets challenged for
- authentication, then we will resend the INVITE. Prior to this patch, the
- Referred-By or Replaces header would be re-added to the outbound INVITE,
- resulting in duplicated headers.
-
- ASTERISK-25204 #close
- Reported by Mark Michelson
-
- Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d
-
-2015-06-23 17:43 +0000 [028fa54620] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_nat: Rewrite route set when required.
-
- When performing some provider testing, the rewrite_contact option was
- interfering with proper construction of a route set when sending an ACK
- after receiving a 200 OK response to an INVITE.
-
- The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
- header with URI sip:bar. In addition, the 200 OK had Record-Route
- headers for sip:baz and sip:foo, in that order. Since the Record-Route
- headers had the lr parameter, the result should have been:
-
- * Set R-URI of the ACK to sip:bar.
- * Add Route headers for sip:foo and sip:baz, in that order.
-
- However, the rewrite_contact option resulted in our rewriting the
- Contact header on the 200 OK to sip:foo. The result was:
-
- * R-URI remained sip:foo.
- * We added Route headers for sip:foo and sip:baz, in that order.
-
- The result was that sip:bar was not indicated in the ACK at all, so the
- far end never received our ACK. The call eventually dropped.
-
- The intention of rewrite_contact is to rewrite the most immediate
- destination of our SIP request to be the same address on which we
- received a request or response. In the case of processing a SIP response
- with Record-Route headers, this means that instead of rewriting the
- Contact header, we should instead rewrite the bottom-most Record-Route
- header. In the case of processing a SIP request with Record-Route
- headers, this means we rewrite the top-most Record-route header.
- Like when we rewrite the Contact header, we also ensure to update
- the dialog's route set if it exists.
-
- ASTERISK-25196 #close
- Reported by Mark Michelson
-
- Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f
-2015-06-19 16:16 +0000 [84c12f9e0c] Richard Mudgett <rmudgett@digium.com>
-
- * threadpool, res_pjsip: Add serializer group shutdown API calls.
-
- A module trying to unload needs to wait for all serializers it creates and
- uses to complete processing before unloading.
-
- ASTERISK-24907
- Reported by: Kevin Harwell
-
- Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059
-
-2015-06-16 15:06 +0000 [602c4b74b5] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs
-
- * handle_client_state_destruction() must always be passed a ref to
- client_state because it will always unref client_state.
- handle_registration_response() was not passing a client_state ref.
-
- * Made the final un-REGISTER message get sent normally using the pjproject
- register control structure in handle_client_state_destruction(). The
- previous code attempted to short circuit the response handling for the
- module to unload. That doesn't work for a couple reasons. One,
- pjsip_regc_send() may call the registered callback before it returns and
- unbalance the client_state ref count. Two, the registered callback
- handles any authentication for the un-REGISTER message.
-
- * Made the distinction between internal registration state and external
- registration status with sip_outbound_registration_status_str(). This is
- necessary to avoid altering documented AMI messages with internal
- changes.
-
- * Removed references to client_state->client outside of the serializer
- thread. When handle_client_state_destruction() destroys the pjproject
- register control structure that memory is freed and cannot be referenced
- anymore. These accesses were to provide information for debug and
- off-nominal warning messages.
-
- * In sip_outbound_registration_timer_cb() you should not access entry->id
- after unrefing client_state because the passed in entry is normally
- pointing to the timer entry in the client_state object.
-
- ASTERISK-24907
- Reported by: Kevin Harwell
-
- Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f
-
-2015-06-15 15:28 +0000 [8c6a95a9ac] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API
-
- The sorcery pjsip 'registration' config object needs to be destroyed on
- module unload. Otherwise, a reload of res_pjsip could try to use
- callbacks for a previously unloaded instance of the module provided by
- ast_sorcery_object_register() or one of the variants. Also, if
- res_pjsip_outbound_registration were subsequently reloaded, the sorcery
- config field objects would be registered in sorcery twice.
-
- ASTERISK-24907
- Reported by: Kevin Harwell
-
- Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697
-
-2015-06-25 06:42 +0000 [e4a2ef9e4e] Joshua Colp <jcolp@digium.com>
-
- * channel: Remove ignore of answer on non-outgoing channels.
-
- Due to the way that channels can now be moved around inside of
- Asterisk it is possible for the outgoing flag of a channel to get
- cleared before it has been answered. This results in the bridge
- not receiving notification that the outgoing leg has been answered.
-
- This most easily exhibits itself with DTMF based blond transfers.
- Since the answer of the outgoing leg is ignored the other party
- continues to receive both a locally generated ringing and the
- media stream of the outgoing leg upon its answer. This results
- in no media being heard.
-
- This change removes the ignore of the answer and allows it
- to pass through.
-
- ASTERISK-25171 #close
-
- Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e
-
-2015-06-15 15:28 +0000 [20f3d77ab9] Richard Mudgett <rmudgett@digium.com>
-
- * sorcery: Add ast_sorcery_object_unregister() API call.
-
- Find and unlink the specified sorcery object type to complement
- ast_sorcery_object_register(). Without this function you cannot
- completely unload individual modules that use sorcery for configuration.
-
- ASTERISK-24907
- Reported by: Kevin Harwell
-
- Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88
-
-2015-06-15 13:38 +0000 [4313f32969] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Reorder load_module() and unload_module().
-
- It is best if the loading code creates and initializes the module's
- infrastructure before letting the system know of its existence. The
- unloading code needs to reverse the actions of the loading code and in the
- reverse order.
-
- ASTERISK-24907
- Reported by: Kevin Harwell
-
- Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4
-
-2015-06-23 14:34 +0000 [890c923786] Richard Mudgett <rmudgett@digium.com>
-
- * AMI: Add Linkedid to the standard channel snapshot AMI event headers.
-
- * The AMI version is bumped to 2.8.0.
-
- ASTERISK-25189 #close
- Reported by: John Hardin
-
- Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac
-
-2015-06-24 14:30 +0000 [2602a7484b] Richard Mudgett <rmudgett@digium.com>
-
- * test.c: Add unit test registration checks for summary and description.
-
- Added checks when a unit test is registered to see that the summary and
- description strings do not end with a new-line '\n' for consistency.
-
- The check generates a warning message and will cause the
- /main/test/registrations unit test to fail.
-
- * Updated struct ast_test_info member doxygen comments.
-
- Change-Id: I295909b6bc013ed9b6882e85c05287082497534d
-
-2015-06-24 14:39 +0000 [2b0482d699] Richard Mudgett <rmudgett@digium.com>
-
- * Unit tests: Fix unit test description strings.
-
- Analyzing the code shows that the unit test summary and description
- strings should not end with a new-line character. Where these strings are
- used in the code a new-line is provided for output.
-
- Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba
-
-2015-06-23 11:21 +0000 [e99e654d75] Joshua Colp <jcolp@digium.com>
-
- * app_dial: Hold reference to calling channel formats when dialing outbound.
-
- Currently when requesting a channel the native formats of the
- calling channel are provided to the core for usage when dialing
- the outbound channel. This occurs without holding the channel lock
- or keeping a reference to the formats. This is problematic as
- the channel driver may end up changing the formats during this time.
- In the case of chan_sip this happens when an SDP negotiation
- completes.
-
- This change makes it so app_dial keeps a reference to the native
- formats of the calling channel which guarantees that they will
- remain valid for the period of time needed.
-
- ASTERISK-25172 #close
-
- Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db
-2015-06-17 05:04 +0000 [80e82dc97f] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_mwi: Set up unsolicited MWI upon registration.
-
- The res_pjsip_mwi previously required a reload to set up the proper
- subscriptions to allow unsolicited MWI to work. This change
- makes it so the act of registering will also cause this to occur.
- This is particularly useful if realtime is involved as no reload
- needs to occur within Asterisk to cause the MWI information
- to get sent.
-
- ASTERISK-25180 #close
-
- Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
-
-2015-06-22 15:11 +0000 [35a99b6394] Kevin Harwell <kharwell@digium.com>
-
- * bridge.c: Hangup attended transfer target if bridged
-
- After completing an attended transfer the transfer target channel was not being
- hung up after leaving the bridge. Added an explicit softhangup to hangup said
- channel, but only if it was previously bridged.
-
- ASTERISK-24782 #close
- Reported by: John Bigelow
-
- Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada
-
-2015-06-17 16:23 +0000 [036bc0012f] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Add missing line endings to CLI commands
-
- Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7
-
-2015-06-12 14:29 +0000 [bec7435945] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage.
-
- Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e
-
-2015-06-12 13:33 +0000 [c2519fdf1c] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_outbound_registration.c: Misc code cleanups.
-
- * Break some long lines.
-
- * Fix doxygen comment.
-
- Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305
-
-2015-06-22 09:26 +0000 [a419c69def] Alexander Traud (License 6520)
-
- * chan_sip: Reload peer without its old capabilities.
-
- On reload, previously allowed codecs were not removed. Therefore, it was not
- possible to remove codecs while Asterisk was running. Furthermore, newly added
- codecs got appended behind the previous codecs. Therefore, it was not possible
- to add a codec with a priority of #1. This change removes the old capabilities
- before the current ones are added.
-
- ASTERISK-25182 #close
- Reported by: Alexander Traud
- patches:
- asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520)
-
- Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802
-
-2015-06-20 19:38 +0000 [74616ae43d] Joshua Colp <jcolp@digium.com>
-
- * chan_sip: Destroy peers without holding peers container lock.
-
- Due to the use of stasis_unsubscribe_and_join in the peer destructor
- it is possible for a deadlock to occur when an event callback is
- occurring at the same time.
-
- This happens because the peer may be destroyed while holding the
- peers container lock. If this occurs the event callback will never
- be able to acquire the container lock and the unsubscribe will
- never complete.
-
- This change makes it so the peers that have been removed from the
- peers container are not destroyed with the container lock held.
-
- ASTERISK-25163 #close
-
- Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33
-
-2015-06-18 13:16 +0000 [9015bb4c8c] Mark Michelson <mmichelson@digium.com>
-
- * Resolve race conditions involving Stasis bridges.
-
- This resolves two observed race conditions.
-
- First, a bit of background on what the Stasis application does:
-
- 1a Creates a stasis_app_control structure. This structure is linked into
- a global container and can be looked up using a channel's unique ID.
- 2a Puts the channel in an event loop. The event loop can exit either
- because the stasis_app_control structure has been marked done, or
- because of some other factor, such as a hangup. In the event loop, the
- stasis_app_control determines if any specific ARI commands need to be
- run on the channel and will run them from this thread.
- 3a Checks if the channel is bridged. If the channel is bridged, then
- ast_bridge_depart() is called since channels that are added to Stasis
- bridges are always imparted as departable.
- 4a Unlink the stasis_app_control from the container.
-
- When an ARI command is received by Asterisk, the following occurs
- 1b A thread is spawned to handle the HTTP request
- 2b The stasis_app_control(s) that corresponds to the channel(s) in the
- request is/are retrieved. If the stasis_app_control cannot be
- retrieved, then it is assumed that the channel in question has exited
- the Stasis app or perhaps was never in Stasis in the first place.
- 3b A command is queued onto the stasis_app_control, and the channel's
- event loop thread is signaled to run the command.
- 4b While most ARI commands do nothing further, some, such as adding or
- removing channels from a bridge, will block until the command they
- issued has been completed by the channel's event loop.
-
- The first race condition that is solved by this patch involves a crash
- that can occur due to faulty detection of the channel's bridged status
- in step 3a. What can happen is that in step 2a, the event loop may run
- the ast_bridge_impart() function to asynchronously place the channel
- into a bridge, then immediately exit the event loop because the channel
- has hung up. In step 3a, we would detect that the channel was not
- bridged and would not call ast_bridge_depart(). The reason that the
- channel did not appear to be bridged was that the depart_thread that is
- spawned by ast_bridge_impart() had not yet started. That is the thread
- where the channel is marked as being bridged. Since we did not call
- ast_bridge_depart(), the Stasis application would exit, and then the
- channel would be destroyed Then the depart_thread would start up and
- try to manipulate the destroyed channel, causing a crash.
-
- The fix for this is to switch from using ast_channel_is_bridged() to
- checking the NULLity of ast_channel_internal_bridge_channel() to
- determine if ast_bridge_depart() needs to be called. The channel's
- internal bridge_channel is set when ast_bridge_impart() is called and
- is NULLed by the call to ast_bridge_depart(). If the channel's internal
- bridge_channel is non-NULL, then the channel must have been imparted
- into the bridge and needs to be departed, even if the actual bridging
- operation has not yet started. By departing the channel when necessary,
- the thread that is running the Stasis application will block until the
- bridge gives the okay that the depart_thread has exited.
-
- The second race condition that is solved by this patch involves a leak
- of HTTP handler threads. The problem was that step 2b would successfully
- retrieve a stasis_app_control structure. Then step 2a would exit the
- channel from the event loop due to a hangup. Steps 3a and 4a would
- execute, and then finally steps 3b and 4b would. The problem is that at
- step 4b, when attempting to add a channel to a bridge, the thread would
- block forever since the channel would never execute the queued command
- since it was finished with the event loop. This meant that the HTTP
- handling thread would be leaked, along with any references that thread
- may have owned (in my case, I was seeing bridges leaked).
-
- The fix for this is to hone in better on when the channel has exited the
- event loop. The stasis_app_control structure has an is_done field that
- is now set at each point where the channel may exit the event loop. If
- step 2b retrieves a valid stasis_app_control structure but the control
- is marked as done, then the attempted operation exits immediately since
- there will be nothing to service the attempted command.
-
- ASTERISK-25091 #close
- Reported by Ilya Trikoz
-
- Change-Id: If66265b73b4c9f8f58599124d777fedc54576628
-2015-06-16 11:13 +0000 [723a9d4225] Mark Michelson <mmichelson@digium.com>
-
- * Parking: Add documentation for AMI ParkedCallSwap event.
-
- This event was added some time ago in order to clarify when a channel
- took the place of another channel in a parking lot. However, there was
- no XML documentation added for the event. This patch adds the XML
- documentation.
-
- ASTERISK-24900 #close
- Reported by Rusty Newton
-
- Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac
-2015-06-15 16:40 +0000 [79bf56c78a] Corey Farrell <git@cfware.com>
-
- * func_pjsip_aor: Fix leaked contact from iterator.
-
- ASTERISK-25162 #close
-
- Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e
-
-2015-06-12 16:58 +0000 [31c77b157b] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
-
- Some phones send g.726 audio packed for AAL2, which differs from what is
- recommended by RFC 3351. If Asterisk receives audio formatted as such when
- negotiating g.726 then it sounds a bit distorted. Added an option to
- res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
- AAL2 packed.
-
- ASTERISK-25158 #close
- Reported by: Steve Pitts
-
- Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
-
-2015-06-14 19:48 +0000 [de8c7f46ed] Matt Jordan <mjordan@digium.com>
-
- * main/cdr: Carry over the disable flag when 'disable all' is specified
-
- The CDR_PROP function (as well as the NoCDR application) set the
- 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This
- flag is supposed to be applied to all CDRs that are currently in the
- chain, as well as all CDRs that may be created in the future. Currently,
- however, the flag is only applied to the existing CDRs in the chain; new
- CDRs do not receive the 'disable all' flag. In particular, this affects
- parallel dials, which generate new CDRs for each pair of channels in
- the dial attempt.
-
- This patch carries over the 'disable all' flag when it is specified on a
- CDR and a new CDR is generated for the chain.
-
- ASTERISK-24344 #close
-
- Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e
-2015-06-12 14:28 +0000 [78ea356e78] Matt Jordan <mjordan@digium.com>
-
- * main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines
-
- When a parallel dial occurs, a new CDR will be created for each dial
- attempt that is made. In most circumstances, the act of creating each
- CDR in the chain will include a step that updates the Party A snapshot,
- which causes the context/extension of the Party A to be copied onto the
- CDR object.
-
- However, when the Party A is in a subroutine, we explicitly do *not*
- copy the context/extension onto the CDR. This prevents the Macro or
- GoSub routine name from blowing away the context/extension that the
- channel was originally executing in. For the original CDR, this is not a
- problem: the original CDR already recorded the last known 'good' state
- of the channel just prior to it going into the subroutine. However, for
- newly generated CDRs in a chain, there is no context/extension set on
- them. Since we are in a subroutine, we will never set the Party A's
- context/extension on the CDR, and we end up with a CDR with no
- destination recorded on it.
-
- This patch updates the creation of a chained CDR such that it copies
- over the original CDR's context/extension. This is the last known "good"
- state of the CDR, and is a reasonable starting point for the newly
- generated CDR. In the case where we are not in a subroutine, subsequent
- code will update the location of the CDR from the Party A information;
- in the case where we are in a subroutine, the context/extension on the
- original CDR is the correct information.
-
- ASTERISK-24443 #close
-
- Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a
-
-2015-06-11 08:18 +0000 [3f57f3f8ec] Damian Ivereigh <damo@launtel.net.au>
-
- * chan_sip.c: Update dialog fromtag after request with auth
-
- If a client sends and INVITE which is 401 rejected, then subsequently
- sends a new INVITE with the auth info and uses a different fromtag
- from the first INVITE, Asterisk will accept the new INVITE as part of
- the original dialog - match_req_to_dialog() specifically ignores the
- fromtag. However it does not update the stored dialog with the new
- fromtag.
-
- This results in Asterisk being unable to match future packets that are
- part of this dialog (such as the ACK to the OK or the OK to the BYE),
- and the call is dropped.
-
- This problem was originally found when using an NEC-i SV8100-GE (NEC SIP
- Card).
-
- * After a successful match of a packet to the dialog, if the packet is
- not a SIP_RESPONSE, authentication is present and the fromtags are
- different, the stored fromtag is updated with the one from the recent
- INVITE.
-
- ASTERISK-25154 #close
- Reported by: Damian Ivereigh
- Tested by: Damian Ivereigh
-
- Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e
-
-2015-06-11 18:52 +0000 [30a0f2d9ac] Matt Jordan <mjordan@digium.com>
-
- * chan_pjsip: Set the context and extension on the channel when created
-
- Prior to this patch, chan_pjsip was failing to pass the endpoint's
- context and the desired extension to the ast_channel_alloc_* routine.
- This caused a new channel snapshot to be issued without a context and
- extension, which can cause some reporting issues for users of AMI, CEL,
- and other APIs. The channel driver would later set the context and
- extension on the channel such that the channel would start in the
- correct location in the dialplan, but the information reported in the
- initial event would be incorrect.
-
- This patch modifies the channel driver such that it now passes the
- context and extension directly into the allocation routine. This
- provides the information in the new channel snapshot published over
- Stasis.
-
- ASTERISK-25156 #close
- Reported by: cloos
-
- Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e
-
-2015-06-10 18:28 +0000 [dbb067279e] Joshua Colp <jcolp@digium.com>
-
- * bridge: When performing a blonde transfer update connected line information.
-
- When performing a blonde transfer the code uses the old masquerade
- mechanism to move a channel around. As a result of this certain information,
- such as connected line, is moved between the channels involved. Upon
- completion of the move a frame is queued which is supposed to update the
- connected line information on the channel. This does not occur as the
- code considers it a redundant update since the masquerade operation
- updated the channel (but did not inform it of the new connected line
- information). The code also does not queue a connected line update
- to be handled by the thread handling the channel. Without this any
- other channel that may be loosely involved does not know it is
- talking to a different caller.
-
- This change does the following to resolve this:
-
- 1. The indicated connected line information is cleared upon
- completion of the masquerade operation when doing a blonde transfer.
- This prevents the connected line update from being considered
- redundant.
-
- 2. A connected line update frame is now queued upon the completion
- of the masquerade operation so any other channel loosely involved
- knows that there is a different caller.
-
- ASTERISK-25157 #close
- Reported by: Joshua Colp
-
- Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20
-
-2015-06-11 14:39 +0000 [a2f4d03c87] Richard Mudgett <rmudgett@digium.com>
-
- * app_directory: Fix crash when using the alias option 'a'.
-
- The voicemail.conf mailbox key/value pair is defined as:
- <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
- Where all fields in the value including the field values are optional.
-
- Since the parsing code for the mailbox key/value pair is sloppy, this
- patch tightens the parsing for the directory information.
-
- * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
- respectively in search_directory_sub(). Those names make more sense.
-
- * Made sure that search_directory_sub() is dealing with the voicemail.conf
- mailbox options field if it even exists when looking for the 'hidefromdir'
- and 'alias' options.
-
- * Fix crash if a voicemail.conf mailbox is just
- <mailbox>=<password>,<name> when the 'a' option is used. If there were no
- fields after the name then the 'options' pointer was not checked for NULL.
-
- * Fix users.conf alias processing if the 'a' option is used. The wrong
- variable was used.
-
- ASTERISK-25087 #close
- Reported by: Chet Stevens
-
- Change-Id: I86052ea77307beddddba5279824d39dc0d593374
-
-2015-06-09 15:31 +0000 [a2b718f4f6] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip.h: Fix some doxygen comments.
-
- Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976
-
-2015-06-05 13:46 +0000 [32ddf6d86b] Richard Mudgett <rmudgett@digium.com>
-
- * taskprocessor.c: Remove extra unref from off-nominal path.
-
- Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60
-
-2015-04-20 16:00 +0000 [cf98c744d5] Yousf Ateya <y.ateya@starkbits.com>
-
- * chan_iax2: Prevent deadlock between hangup and sending lagrq/ping
-
- channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/
- send_ping. This deadlock happens because the scheduled task send_lagrq(or
- send_ping) starts execution after the call hangup procedure starts but before
- it deletes the tasks in the scheduler.
-
- The solution is to delete scheduled lagrq (and ping) task asynchronously
- (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will
- be called in a new context (doesn't have callno locked).
-
- This commit also cleans up the procedure of sending LAGRQ and PING.
-
- main/sched.c: Do not assert when deleting non existant entry from scheduler.
- This assert seems to be the reason for a lot of awkward code to avoid it.
-
- ASTERISK-24983 #close
- Reported by: Y Ateya
-
- Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c
-
-2015-05-31 12:37 +0000 [8af6c9cf6b] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * res_pjsip_transport_websocket: Fix use-after-free bugs.
-
- This patch fixes use-after-free bugs caught by AddressSanitizer.
-
- 1. PJSIP transport manager may decide to destroy transport on its own.
- For example, when the contact registered via websocket has not renewed
- its registration in time. The transport was destoyed, but the websocket
- listener thread was still active until the socket closes, and then tried
- to call transport_shutdown on transport that has been freed.
-
- Also, the transport destructor accessed wstransport->rdata.tp_info.pool
- right after freeing memory that contained wstransport itself.
-
- This patch converts transport to an ao2 object, allowing it to be
- refcounted, so that it is available until both websocket listener and
- pjsip transport manager are finished with it.
-
- 2. The websocket listener deletes the last reference on websocket session
- when the tcp connection is closed, and it gets destroyed, but
- the transport manager may still use it, for example when disconnect
- happens in the middle of a SIP transaction.
-
- A new reference to websocket session has been added that is released
- with the transport to prevent this.
-
- ASTERISK-25096 #close
- Reported by: Josh Kitchens
-
- ASTERISK-24963 #close
- Reported by: Badalian Vyacheslav
-
- Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b
-
-2015-06-09 13:41 +0000 [3046bc17ed] ibercom <ibercom123@gmail.com>
-
- * weakref attribute detection broken with gcc 4.6 and higher
-
- GCC 4.7 Manual:
- http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html
-
- weakref ("target")
-
- A weak reference is an alias that does not by itself require a definition
- to be given for the target symbol.
-
- ASTERISK-22559 #close
- Reported by: Ibercom
-
- Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf
-
-2015-06-08 10:09 +0000 [55c8daf88b] Corey Farrell <git@cfware.com>
-
- * Fix unsafe uses of ast_context pointers.
-
- Although ast_context_find, ast_context_find_or_create and
- ast_context_destroy perform locking of the contexts table,
- any context pointer can become invalid at any time that the
- contexts table is unlocked. This change adds locking around
- all complete operations involving these functions.
-
- Places where ast_context_find was followed by ast_context_destroy
- have been replaced with calls ast_context_destroy_by_name.
-
- ASTERISK-25094 #close
- Reported by: Corey Farrell
-
- Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
-
-2015-06-04 07:14 +0000 [e0090216db] ibercom <ibercom123@gmail.com>
-
- * CLI: Cosmetic issue - core show uptime
-
- Show uptime information ends with an unnecessary space.
-
- Now NEEDCOMMA is better defined.
-
- Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1
-
-2015-06-03 17:41 +0000 [88212ccb7f] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip: Prevent access of NULL channels.
-
- It is possible to receive incoming requests or responses after the channel
- on an ast_sip_session has been destroyed and NULLed out. Handlers of these
- sorts of requests or responses need to be prepared for the possibility
- that the channel is NULL or else they could cause a crash.
-
- While several places have been amended to deal with NULL channels, there
- were still a couple of places that needed updating.
-
- res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
- return early if there is no channel on the session.
-
- res_pjsip_session.c: When handling a 302 response, we need to stop the
- redirecting attempt if there is no channel on the session.
-
- ASTERISK-25148 #close
- reported by Mark Michelson
-
- Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9
-
-2015-06-01 11:45 +0000 [f5d5aa67dc] Kevin Harwell <kharwell@digium.com>
-
- * AMI: Escape string values.
-
- So this issue is a bit complicated. Since it is possible to pass values to AMI
- that contain a '\r\n' (or other similar sequences) these values need to be
- escaped. One way to solve this is to escape the values and then pass the escaped
- values to the AMI variable parameter string building function. However, this
- puts the onus on the pre-build function to escape all string values. This
- potentially requires a fair amount of changes along with a lot of string
- allocations/freeing for all values.
-
- Surely there is a way to push this complexity down a level into the string
- building function itself? This of course is possible, but ends up requiring a
- way to distinguish between strings that need to be escaped and those that don't.
- The best way to handle this is by introducing a new format specifier in the
- format string. For instance a %s (no escape) and %S (escape). However, that is
- a bit weird and unexpected.
-
- So faced with those possibilities this patch implements a limited version of the
- first option. Instead of attempting to escape all string values this patch only
- escapes those values that make sense. This approach limits the number of changes
- and doesn't suffer from the odd format specifier problem.
-
- ASTERISK-24934 #close
- Reported by: warren smith
-
- Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0
-
-2015-06-03 13:17 +0000 [5dc9fb4198] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/location: Fix ref leak in contact_apply_handler
-
- contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status
- to force the creation of a contact_status object whenever a new
- contact is added but it didn't unref the returned object.
-
- Added an ao2_cleanup(status) to plug the leak.
-
- ASTERISK-25141
-
- Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40
- Reported-by: Corey Farrell
-
-2015-06-02 15:07 +0000 [d908272b7e] David M. Lee <dlee@respoke.io>
-
- * Fixes for OS X
-
- * Add some type casting so tv_usec can really be a long, instead of
- some strange platform specific type.
-
- * Add some .dylib style files to .gitignore.
-
- * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer
- versions of GCC, when compiling the Homebrew formula for Asterisk,
- are not properly passing the -Xlinker options to the linker. Given
- that -Wl, does exactly the [same thing][], and does it properly, this
- patch changes the -Xlinker options to use -Wl, instead.
-
- [reasons unknown]: http://bit.ly/1SUbEYx
- [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html
-
- Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd
-
-2015-05-30 20:22 +0000 [9e7827e3ac] Corey Farrell <git@cfware.com>
-
- * pjsip_configuration: Fix leak in persistent_endpoint_update_state.
-
- The loop to find the first available contact of an endpoint grabbed
- contact from the iterator, then checked for offline state. This
- caused the first contact after the state was found to leak a reference.
-
- ASTERISK-25141
-
- Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08
-2015-05-31 11:33 +0000 [888bb49618] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * Fix buffer overflow in slin sample frames generation.
-
- The length of frames retured by sample functions was twice as large as
- real, what caused global buffer overflow caught by AddressSanitizer.
-
- ASTERISK-24717 #close
- Reported by: Badalian Vyacheslav
-
- Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6
-
-2015-05-29 16:19 +0000 [857166b5e5] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip/location: Fix memory leak in permanent_uri_handler
-
- When permanent_uri_handler was creating the contact status
- object for each contact, it wasn't unreffing it at the
- end of the loop.
-
- ASTERISK-25141 #close
- Reported-by: Corey Farrell
-
- Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12
-
-2015-05-29 14:52 +0000 [1558a89129] gtjoseph <george.joseph@fairview5.com>
-
- * Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change"
-
- This reverts commit 35c699086ae2fd81b2473307ccb2ae79ad32375a.
-
- Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7
-
-2015-05-27 13:22 +0000 [35c699086a] gtjoseph <george.joseph@fairview5.com>
-
- * endpoint/stasis: Eliminate duplicate events on endpoint status change
-
- When an endpoint was created, it's messages were being forwarded to
- both the tech endpoint topic and the all endpoints topic. Since
- the tech topic was also forwarded to all, this was resulting in
- duplicate messages whenever an endpoint published. This patch
- causes the endpoint to only forward to the tech topic and lets
- the tech topic forward to all.
-
- To accomplish this, the existing stasis_cp_single_create function
- (which both creates and forwards) was cloned and split into 2
- functions, one that creates the topic and one that sets up the
- forwarding. This allows endpoint_internal_create to create
- the topic from the endpoint_all cache without forwarding it there,
- then allows it to do the forward to the tech's topic.
-
- ASTERISK-25137 #close
- Reported-by: Vitezslav Novy
- ASTERISK-25116 #close
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
- Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c
-
-2015-05-26 13:56 +0000 [fe21f2e52f] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_session: Fix in-dialog authentication.
-
- When the remote peer requires authentication for in-dialog requests then
- re-INVITEs to the peer cause the call to be disconnected and other
- in-dialog requests to the peer like MESSAGE just don't go through.
-
- * Made session_inv_on_tsx_state_changed() handle in-dialog authentication
- for re-INVITEs and other methods. Initial INVITEs cannot be handled here
- because the INVITE transaction must be restarted earlier.
-
- * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in
- preparation for removing the file. The generic outbound authentication
- code did not work as well as anticipated.
-
- * Created outbound_invite_auth() to only handle initial outbound INVITEs.
- Re-INVITEs cannot be handled here. The re-INVITE transaction is still in
- progress and the PJSIP library cannot handle the overlapping INVITE
- transactions. Other method types should not be handled here as this code
- only works on outgoing calls and we need to handle incoming and outgoing
- calls.
-
- ASTERISK-25131 #close
- Reported by: Richard Mudgett
-
- Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0
-
-2015-05-21 17:21 +0000 [262d590819] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
-
- Add a new ContactStatus AMI event.
- Publish the following status/state changes:
- Created
- Removed
- Reachable
- Unreachable
- Unknown
-
- Contact URI, new status/state, aor and endpoint names, and the
- last qualify rtt result are included in the event.
-
- ASTERISK-25114 #close
-
- Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
-2015-05-26 07:44 +0000 [5a42397018] Joshua Colp <jcolp@digium.com>
-
- * sorcery: Fix cache creation callback.
-
- The cache creation callback function expects to receive a sorcery_details
- structure and not just a standalone object.
-
- Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450
-
-2015-05-24 13:47 +0000 [97a6ce1717] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * Astobj2: Correctly treat hash_fn returning INT_MIN
-
- The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0.
- However, abs(INT_MIN) = INT_MIN and is still negative, as well as
- abs(INT_MIN) % num_buckets, and as a result this led to a crash.
-
- One way to trigger the bug is using host=::80 or 0.0.0.128 in peer
- configuration section in chan_sip or chan_iax.
-
- This patch takes the remainder before applying abs, so that bucket
- number is always in range.
-
- ASTERISK-25100 #close
- Reported by: Mark Petersen
-
- Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899
-2015-05-23 04:36 +0000 [554bd1e39c] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * res_pjsip_transport_websocket: Fix crash on receiving large SIP packets
-
- Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves
- truncated before passing to pjsip_tpmgr_receive_packet, but the length
- was passed unaltered, thus causing memory corruption and segfault.
-
- ASTERISK-25122 #close
-
- Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab
-
-2015-05-22 21:50 +0000 [0d266cbe02] Corey Farrell <git@cfware.com>
-
- * Stasis: Fix unsafe use of stasis_unsubscribe in modules.
-
- Many uses of stasis_unsubscribe in modules can be reached through unload.
- These have been switched to stasis_unsubscribe_and_join.
-
- Some subscription callbacks do nothing, for these I've created a noop
- callback function in stasis.c. This is used by some modules that monitor
- MWI topics in order to enable cache, since the callback does not become
- invalid after dlclose it is safe to use stasis_unsubscribe on these, even
- during module unload.
-
- ASTERISK-25121 #close
-
- Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
-
-2015-05-22 12:22 +0000 [51ffed5e61] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS
-
- In addition to specifying lists of 'presence' and 'message-summary',
- users can also create lists of type 'dialog'. These should be treated in
- the same fashion as 'presence'.
-
- Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e
-
-2015-05-22 12:18 +0000 [7950b65e4f] Matt Jordan <mjordan@digium.com>
-
- * res/res_pjsip_exten_state: Fix confusing NOTICE message
-
- When a SUBSCRIBE request is made to a dialplan hint that doesn't exist,
- the current NOTICE message informing users of this swaps the context and
- extension parameters. This can cause a bit of confusion.
-
- Thanks to CptBurger in #asterisk for helping to point this out.
-
- Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43
-
-2015-05-17 20:36 +0000 [5ac65ddfb4] Matt Jordan <mjordan@digium.com>
-
- * res/ari: Register Stasis application on WebSocket attempt
-
- Prior to this patch, when a WebSocket connection is made, ARI would not
- be informed of the connection until after the WebSocket layer had
- accepted the connection. This created a brief race condition where the
- ARI client would be notified that it was connected, a channel would be
- sent into the Stasis dialplan application, but ARI would not yet have
- registered the Stasis application presented in the HTTP request that
- established the WebSocket.
-
- This patch resolves this issue by doing the following:
- * When a WebSocket attempt is made, a callback is made into the ARI
- application layer, which verifies and registers the apps presented in
- the HTTP request. Because we do not yet have a WebSocket, we cannot
- have an event session for the corresponding applications. Some
- defensive checks were thus added to make the application objects
- tolerant to a NULL event session.
- * When a WebSocket connection is made, the registered application is
- updated with the newly created event session that wraps the WebSocket
- connection.
-
- ASTERISK-24988 #close
- Reported by: Joshua Colp
-
- Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636
-
-2015-05-20 11:11 +0000 [60e2fbfe62] gtjoseph <george.joseph@fairview5.com>
-
- * res_pjsip: Refactor endpt_send_transaction (qualify_timeout)
-
- This patch refactors the transaction timeout processing to eliminate
- calling the lower level public pjsip functions and reverts to calling
- pjsip_endpt_send_request again. This is the result of me noticing
- a possible incompatibility with pjproject-2.4 which was causing
- contact status flapping.
-
- The original version of this feature used the lower level calls to
- get access to the tsx structure in order to cancel the transaction
- when our own timer expires. Since we no longer have that access,
- if our own timer expires before the pjsip timer, we call the callbacks
- and just let the pjsip transaction take it's own course. When the
- transaction ends, it discovers the callbacks have already been run
- and just cleans itself up.
-
- A few messages in pjsip_configuration were also added/cleaned up.
-
- ASTERISK-25105 #close
-
- Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-2015-05-20 00:45 +0000 [42476e6633] demon-ru <serov.d.p@gmail.com>
-
- * res_pjsip_outbound_registration: Check request URI for line.
-
- When an inbound call is received the To header is checked
- for the "line" option. Some remote servers will place this
- in the request URI instead. This adds an additional check for
- the option in the request URI.
-
- ASTERISK-25072 #close
- Reported by: Dmitriy Serov
-
- Change-Id: Id4e44debbb80baad623b914a88574371575353c8
-
-2015-05-21 17:51 +0000 [e7edb59db6] Corey Farrell <git@cfware.com>
-
- * res_mwi_external_ami: Use module version of AMI registration.
-
- Use ast_manager_register_xml for res_mwi_external_ami manager
- actions. This ensures the module is held open while any of
- the actions are being run.
-
- ASTERISK-25117 #close
- Reported by: Corey Farrell
-
- Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7
-2015-05-21 19:59 +0000 Asterisk Development Team <asteriskteam@digium.com>
-
- * asterisk 13.4.0-rc1 Released.
-
-2015-05-21 14:56 +0000 [3fb2b375fe] Matt Jordan <mjordan@digium.com>
-
- * Release summaries: Remove previous versions
-
-2015-05-21 14:56 +0000 [9d9ae03842] Matt Jordan <mjordan@digium.com>
-
- * .version: Update for 13.4.0-rc1
-
-2015-05-21 14:56 +0000 [53a39083e5] Matt Jordan <mjordan@digium.com>
-
- * .lastclean: Update for 13.4.0-rc1
-
-2015-05-21 14:56 +0000 [7af8ef9346] Matt Jordan <mjordan@digium.com>
-
- * realtime: Add database scripts for 13.4.0-rc1
-
-2015-05-21 14:52 +0000 [20982c68d4] Matt Jordan <mjordan@digium.com>
-
- * Release summaries: Correct summaries for 13.4.0-rc1
-
-2015-05-21 13:17 +0000 [1bb62b037f] mjordan <mjordan@lunkwill>
-
- * ChangeLog: Updated for 13.4.0-rc1
-
-2015-05-21 13:17 +0000 [1e98a36699] mjordan <mjordan@lunkwill>
-
- * Release summaries: Add summaries for 13.4.0-rc1
-
-2015-05-21 13:15 +0000 [5c12e5ba72] mjordan <mjordan@lunkwill>
-
- * .version: Update for 13.4.0-rc1
-
-2015-05-21 13:15 +0000 [69292a9f11] mjordan <mjordan@lunkwill>
-
- * .lastclean: Update for 13.4.0-rc1
-
-2015-05-21 13:15 +0000 [628680803a] mjordan <mjordan@lunkwill>
-
- * realtime: Add database scripts for 13.4.0-rc1
-
-2015-05-21 13:05 +0000 [9d8a462356] Matt Jordan <mjordan@digium.com>
-
- * ARI: Update version to 1.7.0
-
- This patch updates the version of ARI to 1.7.0 to reflect the backwards
- compatible changes that will be introduced in 13.4.0.
-
- Change-Id: I6c36e6144da426412f25828a868e4df916bff60a
-
-2015-05-21 07:22 +0000 [620054c527] Matt Jordan <mjordan@digium.com>
-
- * Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13
-2015-05-21 07:21 +0000 [f5e195b44e] Matt Jordan <mjordan@digium.com>
-
- * Merge "Logger: Reset defaults before processing config." into 13
-2015-05-21 07:20 +0000 [e8a4e01c32] Matt Jordan <mjordan@digium.com>
-
- * Merge "res/res_http_websocket: Add a pre-session established callback" into 13
-2015-05-21 05:15 +0000 [3c98544543] Joshua Colp <jcolp@digium.com>
-
- * Merge "main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits" into 13
-2015-05-20 20:53 +0000 [9b6e228419] Corey Farrell <git@cfware.com>
-
- * Logger: Reset defaults before processing config.
-
- Reset options to default values before reloading config. This ensures
- that if a setting is removed or commented out of the configuration file
- it is unset on reload.
-
- ASTERISK-25112 #close
- Reported by: Corey Farrell
-
- Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd
-
-2015-05-20 19:05 +0000 [7fcf0a97b8] George Joseph <george.joseph@fairview5.com>
-
- * app_playback: Suppress warnings on playback if channel hung up
-
- If a channel hangs up while an audio file is playing, there's
- no need to clutter up the logs with a warning so suppress it
- if ast_check_hangup returns true.
-
- Also, change warning to debug/2 in file.c if writing a frame
- fails. Same reasoning.
-
- Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
-2015-05-14 15:21 +0000 [b1e8c0b9eb] Kevin Harwell <kharwell@digium.com>
-
- * audiohook.c: Difference in read/write rates caused continuous buffer resets
-
- Currently, everytime a sample rate change occurs (on read or write) the
- associated factory buffers are reset. If the requested sample rate on a
- read differed from that of a write then the buffers are continually reset
- on every read and write. This has the side effect of emptying the buffer,
- thus there being no data to read and then write to a file in the case of
- call recording.
-
- This patch fixes it so that an audiohook_list's rate always maintains the
- maximum sample rate among hooks and formats. Audiohook sample rates are
- only overwritten by this value when slin native compatibility is turned on.
- Also, the audiohook sample rate can only overwrite the list's sample rate
- when its rate is greater than that of the list or if compatibility is
- turned off. This keeps the rate from constantly switching/resetting.
-
- ASTERISK-24944 #close
- Reported by: Ronald Raikes
-
- Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f
-
-2015-05-20 15:22 +0000 [4a450f863b] Matt Jordan <mjordan@digium.com>
-
- * Merge "Fix potential crash after unload of func_periodic_hook or test_message." into 13
-2015-05-19 13:01 +0000 [17d6ede337] Corey Edwards <tensai@zmonkey.org>
-
- * main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits
-
- ASTERISK-24887 #close
- Reported by: Makoto Dei
- Tested by: tensai
-
- Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf
-
-2015-05-13 09:55 +0000 [31cc24aad6] Matt Jordan <mjordan@digium.com>
-
- * res/res_http_websocket: Add a pre-session established callback
-
- This patch updates http_websocket and its corresponding implementation
- with a pre-session established callback. This callback allows for
- WebSocket server consumers to be notified when a WebSocket connection is
- attempted, but before we accept it. Consumers can choose to reject the
- connection, if their application specific logic allows for it.
-
- As a result, this patch pulls out the previously private
- websocket_protocol struct and makes it public, as
- ast_websocket_protocol. In order to preserve backwards compatibility
- with existing modules, the existing APIs were left as-is, and new APIs
- were added for the creation of the ast_websocket_protocol as well as for
- adding a sub-protocol to a WebSocket server.
-
- In particular, the following new API calls were added:
- * ast_websocket_add_protocol2 - add a protocol to the core WebSocket
- server
- * ast_websocket_server_add_protocol2 - add a protocol to a specific
- WebSocket server
- * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object.
- Consumers can populate this with whatever callbacks they wish to
- support, then add it to the core server or a specified server.
-
- ASTERISK-24988
- Reported by: Joshua Colp
-
- Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2
-
-2015-05-14 22:05 +0000 [f9114179e6] snuffy <snuffy22@gmail.com>
-
- * chan_pjsip: Fix crash during off-nominal when no endpoint specified.
-
- Add missing return -1 when no endpoint name is specified.
-
- ASTERISK-25086 #close
- Reported by: snuffy
-
- Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e
-2015-05-14 18:01 +0000 [dd78ab42e4] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip_config_wizard/config: Fix template processing
-
- The config wizard was always pulling the first occurrence of
- a variable from an ast_variable list but this gets the template
- value from the list instead of any overridden value. This patch
- creates ast_variable_find_last_in_list() in config.c and updates
- res_pjsip_config_wizard to use it instead of
- ast_variable_find_in_list. Now the overridden values, where they
- exist, are used instead of template variables.
-
- Updated test_config to test the new API.
-
- ASTERISK-25089 #close
-
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
- Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4
-
-2015-05-15 01:54 +0000 [091b436007] snuffy <snuffy22@gmail.com>
-
- * cdr: Fix 'core show channel' CDR variable truncation.
-
- When the new Bridging API was implemented, the workspace variable
- changed to a malloc'd string, causing sizeof() to always be 8 (char).
-
- Revert back to stored on stack string for workspace.
-
- ASTERISK-25090 #close
-
- Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7
-2015-05-14 15:20 +0000 [8697a49ef9] Joshua Colp <jcolp@digium.com>
-
- * Merge "sorcery: Add API to insert/remove a wizard to/from an object type's list" into 13
-2015-05-14 15:19 +0000 [aea349a87e] Joshua Colp <jcolp@digium.com>
-
- * Merge "Message.c: Clear message channel frames on cleanup" into 13
-2015-05-14 00:06 +0000 [6b7282ca40] Corey Farrell <git@cfware.com>
-
- * Fix potential crash after unload of func_periodic_hook or test_message.
-
- These modules save a pointer to the context they create on load, and
- use that pointer to destroy the context at unload. It is not safe
- to save this pointer, it is replaced during load of pbx_config,
- pbx_lua or pbx_ael.
-
- This change causes the modules to pass NULL to ast_context_destroy,
- a safer way to perform the unregistration since it does not use
- a pointer that could become invalid.
-
- ASTERISK-25085 #close
- Reported by: Corey Farrell
-
- Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835
-2015-05-14 05:02 +0000 [8f8d54a18e] Joshua Colp <jcolp@digium.com>
-
- * Merge "main/manager.c: Bugfix sort action_manager by alphabetically" into 13
-2015-05-13 15:41 +0000 [02c5130589] Jonathan Rose <jrose@digium.com>
-
- * Message.c: Clear message channel frames on cleanup
-
- The message channel is a special channel that doesn't actually process frames.
- However, certain actions can cause frames to be placed in the channel's read
- queue including the Hangup application which is called on the channel after
- each message is processed. Since the channel will continually be reused for
- many messages, it's necessary to flush these frames at some point.
-
- ASTERISK-25083 #close
- Reported by: Jonathan Rose
-
- Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f
-
-2015-05-13 15:44 +0000 [586da882bc] Joshua Colp <jcolp@digium.com>
-
- * Merge "app_voicemail: fix moving when old messages full" into 13
-2015-05-12 17:45 +0000 [d49d64b79c] Jonathan Rose <jrose@digium.com>
-
- * app_voicemail: fix moving when old messages full
-
- When completing voicemail playback of a message in the 'INBOX', the
- message gets moved to the 'Old' messages folder. Without this patch, if
- the 'Old' folder is already at its set limit, then the 'INBOX' message will
- simply be deleted. With this patch, the flag to delete the message will be
- removed if the save_to_folder function indicates that the message could
- not be moved due to a full folder.
-
- ASTERISK-25082 #close
- Reported by: Jonathan Rose
- Review: https://gerrit.asterisk.org/#/c/448/
-
- Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f
-2015-05-13 14:20 +0000 [51478575e4] Joshua Colp <jcolp@digium.com>
-
- * Merge "General: Fix recent menuselect-related cross compile regression" into 13
-2015-05-13 12:26 +0000 [5fcaf727cc] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_config_mysql: Fix broken column type checking" into 13
-2015-05-13 12:24 +0000 [6a12b0634b] Joshua Colp <jcolp@digium.com>
-
- * Merge "chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision." into 13
-2015-05-04 20:11 +0000 [9b13536fed] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * main/manager.c: Bugfix sort action_manager by alphabetically
-
- Fix the alphabetic order added on ast_manager_register_struct. The order
- for struct manager_action added is not working, this change fixes the
- problem.
-
- Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b
-
-2015-05-08 18:01 +0000 [e67e8d5c7f] Alexandre Fournier <alexandre.fournier@kiplink.fr>
-
- * res_config_mysql: Fix broken column type checking
-
- MySQL configuration engine contains a bug in require_mysql(). This
- function is used for column type checking in tables. This bug only
- affects DATETIME, DATE and FLOAT types.
-
- It came from mixing the first condition (switch-case-like
- if/then/else), to check the expected column type, with the second
- condition, to check the actual column type against the expected column
- type. Both conditions must be checked separately in order to avoid the
- execution of the wrong block.
-
- ASTERISK-18252 #comment This patch might fix the issue
- Reported by: Gareth Blades
-
- ASTERISK-25041 #close
- Reported by: Alexandre Fournier
- Tested by: Alexandre Fournier
-
- Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa
-
-2015-05-10 07:37 +0000 [16f602f5c2] Yousf Ateya <y.ateya@starkbits.com>
-
- * res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS.
-
- First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC
- https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values.
-
- Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31
-
-2015-05-13 04:35 +0000 [62422712f7] Joshua Colp <jcolp@digium.com>
-
- * Merge "cdr_pgsql: Use PQescapeStringConn for escaping names." into 13
-2015-05-12 17:34 +0000 [c780b6e431] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision.
-
- If an ISDN call is hungup by both sides at the same time a crash could
- happen.
-
- * Added missing NULL checks for the owner channel after calling
- pri_queue_pvt_cause_data() in two places. Code after those calls need to
- check the owner channel pointer for NULL before use because
- pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the
- owner and the owner may get hung up.
-
- ASTERISK-21893 #close
- Reported by: Alexandr Gordeev
-
- Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a
-
-2015-05-10 02:26 +0000 [6627de830b] Sebastian Kemper <sebastian_ml@gmx.net>
-
- * General: Fix recent menuselect-related cross compile regression
-
- MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the
- target platform. But menuselect is to be run on the build system, so
- BUILD_CC needs to be used instead - like it was in the past, before the
- recent changes (https://reviewboard.asterisk.org/r/4370/). This is the
- patch for ASTERISK-25074.
-
- ASTERISK-25074 #close
- Reported by: Sebastian Kemper
- Tested by: Sebastian Kemper
-
- Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8
-
-2015-05-05 15:32 +0000 [637c8f065e] George Joseph <george.joseph@fairview5.com>
-
- * sorcery: Add API to insert/remove a wizard to/from an object type's list
-
- Currently you can 'apply' a wizard to an object type but the wizard
- always goes at the end of the object type's wizard list. This patch
- adds a new ast_sorcery_insert_wizard_mapping function that allows
- you to insert a wizard anyplace in the list. I.E. You could
- add a caching wizard to an object type and place it before all
- wizards.
-
- ast_sorcery_get_wizard_mapping_count and
- ast_sorcery_get_wizard_mapping were added to allow examination
- of the mapping list.
-
- ast_sorcery_remove_mapping was added to remove a mapping by name.
-
- As part of this patch, the object type's wizard list was converted
- from an ao2_container to an AST_VECTOR_RW.
-
- A new test was added to test_sorcery for this capability.
-
- ASTERISK-25044 #close
-
- Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57
-
-2015-05-12 01:31 +0000 [3cdb7950f0] Corey Farrell <git@cfware.com>
-
- * Fix processing of asterisk.conf debug=yes.
-
- The code which reads asterisk.conf supports processing the debug
- option with ast_true, but ast_true returns -1. This causes debug
- to still be off, convert to 1 so debug will be on as requested.
-
- ASTERISK-25042
- Reported by: Corey Farrell
-
- Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6
-
-2015-05-01 23:43 +0000 [6553a00770] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * cdr_pgsql: Use PQescapeStringConn for escaping names.
-
- Use function PQescapeStringConn for escaping the name
- of the table and schema instead of doing it manually.
-
- Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599
-
-2015-05-12 05:38 +0000 [8523a5ed09] Joshua Colp <jcolp@digium.com>
-
- * Merge "vector: Add REMOVE, ADD_SORTED and RESET macros" into 13
-2015-05-09 16:58 +0000 [ea917fefaf] George Joseph <george.joseph@fairview5.com>
-
- * vector: Add REMOVE, ADD_SORTED and RESET macros
-
- Based on feedback from Corey Farrell and Y Ateya, a few new
- macros have been added...
-
- AST_VECTOR_REMOVE which takes a parameter to indicate if
- order should be preserved.
-
- AST_VECTOR_ADD_SORTED which adds an element to
- a sorted vector.
-
- AST_VECTOR_RESET which cleans all elements from the vector
- leaving the storage intact.
-
- Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14
-
-2015-05-11 07:07 +0000 [d5864a358c] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * pbx/pbx_spool: Fix issue when call files were executed too early
-
- pbx_spool used to delete/move the call file upon successful outgoing
- call completion, but did not delete it from in-memory list of files
- (dirlist, used only when compiled with inotify/kqueue support).
- That resulted in an extra attempt to process that filename after
- retrytime seconds.
- Then, if a new file with the same name appears that is scheduled
- in future further than the completed one plus its retrytime,
- then it gets executed earlier than expected.
-
- This patch fixes remove_from_queue function to also remove the entry
- from the dirlist.
-
- ASTERISK-17069 #close
- Reported by: Jeremy Kister
-
- ASTERISK-24442 #close
- Reported by: tootai
-
- Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b
-
-2015-05-08 14:47 +0000 [4dbd4021c9] Rusty Newton <rnewton@digium.com>
-
- * configs/basic-pbx: Modified main IVR to play new Allison prompt.
-
- The main IVR was playing demo-congrats. I've switched it over to the
- basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
- has Allison prompting the user with the actual IVR menu.
-
- ASTERISK-24892 #close
-
- Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d
-
-2015-05-08 15:55 +0000 [7111ba6df4] Matt Jordan <mjordan@digium.com>
-
- * Merge "tcptls: Avoiding ERR_remove_state in OpenSSL." into 13
-2015-05-08 10:39 +0000 [613a461c3d] Sean Bright <sean@malleable.com>
-
- * res_rtp_asterisk: Issue ERROR if res_srtp is not found.
-
- While trying to get WebRTC working with chan_pjsip, I was running
- into the following error:
-
- Attempted to set an invalid DTLS-SRTP configuration on RTP
- instance...
-
- Josh helpfully pointed out that res_srtp.so might not be loaded, and
- sure enough, it wasn't. This patch adds a ERROR indiciating as much
- to hopefully help others having a similar problem.
-
- Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f
-
-2015-05-07 17:49 +0000 [394fcb5eab] Rusty Newton <rnewton@digium.com>
-
- * sounds: Add Swedish sounds to Makefile and XML
-
- Added the necessary lines to the Makefile and sounds.xml so we'll have the
- Swedish sounds in all available formats in menuselect.
-
- See also: Swedish sounds were added into the core sounds release 1.4.27.
-
- ASTERISK-24744 #close
-
- Reported by: Tove Hjelm
- Tested by: Rusty Newton
-
- Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908
-
-2015-05-08 09:54 +0000 [30c3b254c5] Joshua Colp <jcolp@digium.com>
-
- * Merge "doc: Make progdocs play nice with git" into 13
-2015-05-05 11:35 +0000 [2115f11b54] Alexander Traud (License 6520)
-
- * tcptls: Avoiding ERR_remove_state in OpenSSL.
-
- ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by
- ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were
- called by SSL_load_error_strings already and got removed. These changes allow
- OpenSSL forks like BoringSSL to be used with Asterisk.
-
- ASTERISK-25043 #close
- Reported by: Alexander Traud
- patches:
- asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520)
-
- Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629
- (cherry picked from commit 247fef66537b59649e7571d64e2c574a106dbd65)
-
-2015-05-07 14:54 +0000 [5392e970d0] George Joseph <george.joseph@fairview5.com>
-
- * doc: Make progdocs play nice with git
-
- Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in
-
- Changed /Makefile to copy asterisk-ng-doxygen.in to
- asterisk-ng-doxygen then modify it with version instead of
- modifying asterisk-ng-doxygen directly. Updated clean
- targets as well.
-
- Updated /.gitignore and doc/.gitignore.
-
- Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622
-
-2015-05-07 15:10 +0000 [1e44d1bef9] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination" into 13
-2015-05-04 14:43 +0000 [608f0a94ee] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * contrib/editors: Fix vim syntax highlighting of comments in config files
-
- * Added a lookbehind to one-line comment matcher to skip escaped
- semicolons.
- * Added support for block comments.
-
- Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7
-
-2015-05-07 13:30 +0000 [22c6c12af2] Matt Jordan <mjordan@digium.com>
-
- * Merge "vector: Additional enhancements and fixes" into 13
-2015-05-06 13:24 +0000 [d649d682c4] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination
-
- The res_pjsip_exten_state module currently has a race condition between
- processing the extension state callback from the PBX core and processing
- the subscription shutdown callback from res_pjsip_pubsub. There is currently
- no synchronization between the two. This can present a problem as while
- the SIP subscription will remain valid the tree it points to may not.
- This is in particular a problem as a task to send a NOTIFY may get queued
- which will try to use the tree that may no longer be valid.
-
- This change does the following to fix this problem:
-
- 1. All access to the subscription tree is done within the task that
- sends the NOTIFY to ensure that no other thread is modifying or
- destroying the tree. This task executes on the serializer for the
- subscriptions.
-
- 2. A reference to the subscription serializer is kept to ensure it
- remains valid for the lifetime of the extension state subscription.
-
- 3. The NOTIFY task has been changed so it will no longer attempt
- to send a NOTIFY if the subscription has already been terminated.
-
- ASTERISK-25057 #close
- Reported by: Matt Jordan
-
- Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643
-
-2015-05-07 07:02 +0000 [9322bc6ff6] Matt Jordan <mjordan@digium.com>
-
- * Merge "chan_dahdi: Improve force_restart_unavailable_chans option description." into 13
-2015-05-07 06:39 +0000 [b1514362ef] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_stasis_snoop: Spying on a single direction continually increases CPU" into 13
-2015-05-07 06:28 +0000 [652ee2ff83] Joshua Colp <jcolp@digium.com>
-
- * Merge "features: Fix crash when transferee hangs up during DTMF attended transfer." into 13
-2015-05-05 20:22 +0000 [5f9aea8e3c] George Joseph <george.joseph@fairview5.com>
-
- * vector: Additional enhancements and fixes
-
- After using the new vector stuff for real I found...
-
- A bug in AST_VECTOR_INSERT_AT that could cause a seg fault.
-
- The callbacks needed to be closer to ao2_callback in behavior
- WRT to CMP_MATCH and CMP_STOP behavior and the ability to return
- a vector of matched entries.
-
- A pre-existing issue with APPEND and REPLACE was also fixed.
-
- I also added a new macro to test.h that acts like ast_test_validate
- but also accepts a return code variable and a cleanup label. As well
- as printing the error, it sets the rc variable to AST_TEST_FAIL and
- does a goto to the specified label on error. I had a local version
- of this in test_vector so I just moved it.
-
- ASTERISK-25045
-
- Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc
-
-2015-05-04 17:28 +0000 [68513e00f7] Kevin Harwell <kharwell@digium.com>
-
- * res_stasis_snoop: Spying on a single direction continually increases CPU
-
- Creating a snoop channel in ARI and spying only on a single direction (in or
- out) results in CPU utilization continually increasing until the CPU is fully
- consumed. This occurs because frames are being put in the opposing direction's
- slin factory queue, but not being removed.
-
- Fixed the problem by always reading and disposing of frames from the opposite
- queue of the direction selected.
-
- ASTERISK-24938 #closes
-
- Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60
-2015-05-06 16:00 +0000 [904f5d98f6] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi: Improve force_restart_unavailable_chans option description.
-
- ASTERISK-25034
- Reported by: Richard Mudgett
-
- Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30
-
-2015-05-06 07:42 +0000 [d6ffbe39b0] Joshua Colp <jcolp@digium.com>
-
- * Merge "app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter" into 13
-2015-05-06 06:13 +0000 [dfb292ce3e] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_ari_bridges: Add missing dependencies." into 13
-2015-05-05 21:05 +0000 [50e90f9121] Matt Jordan <mjordan@digium.com>
-
- * Merge "pbx_config: Register manager actions with module version of macro." into 13
-2015-05-05 18:17 +0000 [be1260a35f] Richard Mudgett <rmudgett@digium.com>
-
- * features: Fix crash when transferee hangs up during DTMF attended transfer.
-
- A crash happens with this sequence of steps:
- 1) Party A is connected to party B.
- 2) Party B starts a DTMF attended transfer.
- 3) Party A hangs up while party B is dialing party C.
-
- When party A hangs up the bridge that party A and party B are in is
- dissolved and party B is kicked out of the bridge. When party B finishes
- dialing party C he attempts to move to the new bridge with party C. Since
- party B is no longer in a bridge the attempted move dereferences a NULL
- bridge_channel pointer and crashes.
-
- * Made the hold(), unhold(), ringing(), and the bridge_move() functions
- tolerant of the channel not being in a bridge. The assertion that party B
- is always in a bridge is not true if the bridged peer of party B hangs up
- and dissolves the bridge. Being tolerant of not being in a bridge allows
- the peer hangup stimulus to be processed by the FSM.
-
- * Made the bridge_move() function return void since where the return value
- for a failed move was checked generated a FSM coding ERROR message for a
- normal off-nominal condition.
-
- * Eliminated most uses of RAII_VAR in bridge_basic.c.
-
- ASTERISK-25003 #close
- Reported by: Artem Volodin
-
- Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada
-
-2015-05-05 15:40 +0000 [8b0f85ac06] George Joseph <george.joseph@fairview5.com>
-
- * test_vector: Fix build breakage caused by ASTERISK_REGISTER_FILE
-
- My 13 version of test_vector had an ASTERISK_REGISTER_FILE() macro
- call at the top which is only supported in master. Once removed
- builds are successful.
-
- Change-Id: I7cac8b669bed6de543bbf4e2eec3cffc9741acdd
-
-2015-05-05 14:48 +0000 [87263b47b5] Ivan Poddubny <ivan.poddubny@gmail.com>
-
- * app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter
-
- This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3
- parameters: position, original position and waiting time.
-
- ASTERISK-25038 #close
- Reported by: Etienne Lessard
-
- Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618
-
-2015-05-05 13:13 +0000 [2d9081b5ec] Matt Jordan <mjordan@digium.com>
-
- * Merge "stasis: Fix dial masquerade datastore lifetime" into 13
-2015-05-05 12:45 +0000 [8ca25dfd7e] Matt Jordan <mjordan@digium.com>
-
- * Merge "vector: Traversal, retrieval, insert and locking enhancements" into 13
-2015-05-05 09:47 +0000 [366ea63438] Corey Farrell <git@cfware.com>
-
- * res_ari_bridges: Add missing dependencies.
-
- Missed this module in the previous commit. res_ari_bridges uses symbols
- from res_stasis_playback and res_stasis_recording.
-
- ASTERISK-25027 #close
- Reported by: Corey Farrell
-
- Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f
-
-2015-05-05 09:27 +0000 [69ae8cf0a4] Corey Farrell <git@cfware.com>
-
- * pbx_config: Register manager actions with module version of macro.
-
- Switch manager actions in pbx_config to use the registration macro that
- passes the module pointer, allowing pbx_config reference to be bumped
- while the manager actions run.
-
- ASTERISK-25061 #close
- Reported by: Corey Farrell
-
- Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e
-
-2015-05-04 12:16 +0000 [181ae3b8d9] Joshua Colp <jcolp@digium.com>
-
- * stasis: Fix dial masquerade datastore lifetime
-
- A recent change went into Asterisk which added reference counts to the
- channels stored in a dial masquerade datastore. Unfortunately this
- included a reference to the caller in a dialing operation. While all
- of the dialed targets have the datastore removed from them upon dialing
- completion this did not occur for the caller, causing it to have a
- reference to itself that could go never go away (as it depended on
- the destruction of the datastore which only happened when the channel
- was destroyed). This resulted in the caller channel remaining on the
- system despite it having hung up.
-
- This change does the following to fix this issue:
-
- 1. The dial masquerade datastore is now removed from the caller upon
- dialing completion, just like the dialed targets.
- 2. Upon destruction of the caller all the dialed targets are also
- removed from the dial masquerade datastore (just in case).
- 3. The reference to the caller has been removed as it should not be
- possible for the datastore to now be valid/useful after the lifetime
- of the caller has ended.
-
- ASTERISK-25025 #close
-
- Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f
-
-2015-05-01 19:25 +0000 [7a7e9733c2] George Joseph <george.joseph@fairview5.com>
-
- * vector: Traversal, retrieval, insert and locking enhancements
-
- Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really
- does replace not insert. The few users of AST_VECTOR_INSERT were
- refactored. Because these are macros, there should be no ABI
- compatibility issues.
-
- Added AST_VECTOR_INSERT_AT that actually inserts an element into the
- vector at a specific index pushing existing elements to the right.
-
- Added AST_VECTOR_GET_CMP that can retrieve from the vector based
- on a user-provided compare function.
-
- Added AST_VECTOR_CALLBACK function that will execute a function
- for each element in the vector. Similar to ao2_callback and
- ao2_callback_data functions although the vector callback can take
- a variable number of arguments. This should allow easy migration
- to a vector where a container might be too heavy.
-
- Added read/write locked vector and lock manipulation macros.
-
- Added unit tests.
-
- ASTERISK-25045 #close
-
- Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0
-
-2015-05-03 13:55 +0000 [040d2f8558] Corey Farrell <git@cfware.com>
-
- * main/test.c: Add test to verify there were no registration errors.
-
- This adds a test that will fail if any test failed to register. Also fail
- if any test registration produced a warning about missing a leading or
- trailing slash.
-
- ASTERISK-25053 #close
- Reported by: Corey Farrell
-
- Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3
-
-2015-05-04 09:26 +0000 [626bffc4c2] Matt Jordan <mjordan@digium.com>
-
- * Merge "contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update" into 13
-2015-05-04 09:26 +0000 [87fb7fc165] Matt Jordan <mjordan@digium.com>
-
- * Merge "cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8" into 13
-2015-05-04 09:25 +0000 [81c27127aa] Matt Jordan <mjordan@digium.com>
-
- * Merge "Format Interfaces: Prevent unload except by shutdown." into 13
-2015-05-04 07:46 +0000 [743fed71fc] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_odbc: Use negative connection cache for all connections" into 13
-2015-04-21 11:52 +0000 [3dcec04ab5] Martin Tomec <tomec.martin@gmail.com>
-
- * res_odbc: Use negative connection cache for all connections
-
- Apply the negative connection cache setting to all connections,
- even those that are not pooled. This ensures that the connection
- will not be re-established before the negative connection cache
- time is met.
-
- ASTERISK-22708 #close
-
- Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271
-2015-05-04 04:03 +0000 [74799b3fe2] Matt Jordan <mjordan@digium.com>
-
- * Merge "Remove unneeded uses of optional_api providers." into 13
-2015-05-04 04:03 +0000 [78c02f8e88] Matt Jordan <mjordan@digium.com>
-
- * Merge "Update configure.ac/Makefile for clang" into 13
-2015-05-03 21:03 +0000 [f38066fcad] Corey Farrell <git@cfware.com>
-
- * Format Interfaces: Prevent unload except by shutdown.
-
- Format interfaces cannot be unregistered, so the modules that provide them
- need to be held open except by shutdown.
-
- ASTERISK-25054 #close
- Reported by: Corey Farrell
-
- Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3
-
-2015-05-03 20:28 +0000 [e76a6a97bf] Matt Jordan <mjordan@digium.com>
-
- * contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update
-
- The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755
- failed to add ENUM support for Postgres databases. This requires a
- specific import from the sqlalchemy.dialects.postgresql package. This
- patch corrects this error, which allows for Postgres update scripts to
- be generated.
-
- ASTERISK-24706
-
- Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015
-
-2015-05-01 19:50 +0000 [92120247e9] D Tucny <d@tucny.com>
-
- * term: send proper reset sequence when black background is forced
-
- When using the force black background command-line option or configuration
- option an invalid reset sequence is sent following a coloured output item
- in the CLI, the result is that the colour is not 'turned off' and continues
- until the next non-default coloured text output.
-
- A reset sequence is already defined in term.c, but the ast_term_reset
- function doesn't use it, instead building it's own invalid sequence and
- returning that.
-
- This patch changes that behaviour, removing the building of a reset sequence
- and instead using the pre-built constant 'enddata' which is a suitable reset
- sequence for this purpose.
-
- ASTERISK-24896 #close
- Reported by: Dan Tucny
-
- Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43
-
-2015-05-03 09:20 +0000 [13819a34c4] Matt Jordan <mjordan@digium.com>
-
- * Merge "Build System: Prevent unneeded changes to asterisk/buildopts.h." into 13
-2015-05-03 09:19 +0000 [b518ba1c6c] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip_dlg_options: Fix MODULEINFO section." into 13
-2015-05-02 18:58 +0000 [ad6ea29697] Corey Farrell <git@cfware.com>
-
- * Remove unneeded uses of optional_api providers.
-
- A few cases exist where headers of optional_api provders are included but
- not needed. This causes unneeded calls to ast_optional_api_use.
-
- * Don't include optional_api.h from sip_api.h.
- * Move 'struct ast_channel_monitor' to channel.h.
- * Don't include monitor.h from chan_sip.c, channel.c or features.c.
-
- The move of struct ast_channel_monitor is needed since channel.c depends on
- it. This has no effect on users of monitor.h since channel.h is included
- from monitor.h.
-
- ASTERISK-25051 #close
- Reported by: Corey Farrell
-
- Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478
-
-2015-05-02 10:19 +0000 [9888562c8c] Matt Jordan <mjordan@digium.com>
-
- * Merge "include/asterisk/channel.h: Fix typo" into 13
-2015-05-02 10:17 +0000 [b4000f2d44] Matt Jordan <mjordan@digium.com>
-
- * Merge "Astobj2: Fix initialization order of refdebug and AO2_DEBUG." into 13
-2015-04-30 02:07 +0000 [525c8c8689] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * include/asterisk/channel.h: Fix typo
-
- Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3
-
-2015-05-02 02:15 +0000 [63196a8256] Corey Farrell <git@cfware.com>
-
- * res_pjsip_dlg_options: Fix MODULEINFO section.
-
- Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options.
- This extra space prevented any of the dependencies from being seen by
- menuselect, so building with default options would fail if PJSIP was
- not installed.
-
- This also makes the tool that extracts information for menuselect
- tolerant of multiple spaces in the future.
-
- ASTERISK-25033 #close
- Reported by: Peter Whisker
-
- Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698
-
-2015-04-29 03:03 +0000 [ac1f0090eb] Corey Farrell <git@cfware.com>
-
- * Build System: Prevent unneeded changes to asterisk/buildopts.h.
-
- * Add AST_DEVMODE to BUILDOPTS
- * Remove CFLAGS that do not effect ABI from BUILDOPTS.
- * Use BUILDOPTS to generate AST_BUILDOPT_SUM.
- * Remove loop that defined AST_MODULE_*
-
- These changes ensure that only ABI effecting options are considered for
- AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused
- by enabling or disabling one module that another is dependent on.
-
- ASTERISK-25028
- Reported by: Corey Farrell
-
- Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2
-
-2015-05-01 13:22 +0000 [5875bf183c] Corey Farrell <git@cfware.com>
-
- * Astobj2: Fix initialization order of refdebug and AO2_DEBUG.
-
- This ensures that refdebug is initialized before AO2_DEBUG if
- both are enabled, since AO2_DEBUG allocates a container.
-
- This change also makes AO2_DEBUG initialization critical, a
- failure will abort Asterisk startup. This is needed since
- the failure would be caused by reg_containers allocation
- failure, and that would result in a segmentation fault by
- ao2_container_register later in startup.
-
- ASTERISK-25048 #close
- Reported by: Corey Farrell
-
- Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244
-
-2015-04-29 14:49 +0000 [1b19c15f17] Matt Jordan <mjordan@digium.com>
-
- * main/pbx: Improve performance of dialplan reloads with a large number of hints
-
- The PBX core maintains two hash tables for hints: a container of the
- actual hints (hints), along with a container of devices that are watching that
- hint (hintdevices). When a dialplan reload occurs, each hint in the hints
- container is destroyed; this requires a lookup in the container of devices to
- find the device => hint mapping object. In the current code, this performs an
- ao2_callback, iterating over each of the device to hint objects in the
- hintdevices container. For a large number of hints, this is extremely
- expensive: dialplan reloads with 20000 hints could take several minutes
- in just this phase.
-
- This patch improves the performance of this step in the dialplan reloads
- by caching which devices are watching a hint on the hint object itself.
- Since we don't want to create a circular reference, we just cache the
- name of the device. This allows us to perform a smarter ao2_callback on
- the hintdevices container during hint removal, hashing on the name of the
- device and returning an iterator to the matching names. The overall
- performance improvement is rather large, taking this step down to a number of
- seconds as opposed to minutes.
-
- In addition, this patch also registers the hint containers in the PBX
- core with the astobj2 library. This allows for reasonable debugging to
- hash collisions in those containers.
-
- ASTERISK-25040 #close
- Reported by: Matt Jordan
-
- Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360
-
-2015-05-01 06:55 +0000 [ec0f80b6e8] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback." into 13
-2015-05-01 06:55 +0000 [ed51fbbe9c] Matt Jordan <mjordan@digium.com>
-
- * Merge "Prevent potential crash on blond transfer." into 13
-2015-04-30 15:54 +0000 [3efe0df044] Corey Farrell <git@cfware.com>
-
- * Sample Configs: Fix syntax error in pjsip.conf
-
- The sample pjsip.conf has a few comment lines that are missing the
- semicolons at the start of the comment, causing the config to fail
- load.
-
- Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352
-
-2015-04-30 15:20 +0000 [077979618b] Mark Michelson <mmichelson@digium.com>
-
- * Prevent potential crash on blond transfer.
-
- Scenario:
- Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects
- the incoming call (or some other immediate circumstance causes Carol not
- to answer the call)
-
- What occurs in this case is that when the bridge between Alice and Bob
- breaks, Alice is told to masquerade into Bob's channel that had placed
- the call to Carol. The actual masquerade goes down without a hitch.
- However, a channel fixup callback that attempts to publish dial events
- over Stasis has a crash. The reason for this crash is that the datastore
- on Bob's channel that placed the outbound call to Carol only had a bare
- pointer to Carol's channel. Since Carol rejected the incoming call,
- Carol's channel has been hung up and freed, meaning accessing her
- channel results in a crash.
-
- The fix here is simple. The dial fixup code has been altered to hold
- references to the involved channels and to drop those references when
- freeing data.
-
- ASTERISK-25025 #close
- Reported by Chet Stevens
-
- Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197
-
-2015-04-30 14:09 +0000 [4b8cddfb36] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback.
-
- The Asterisk 13 version of the fix for outbound registration was missing
- a key component that set the outbound authenticator's callback that
- creates an authenticated request based on an old request. This was
- picked up by some outbound registration tests failing in the testsuite.
-
- Change-Id: I5ca9379698c606da36bc38eaffccedaf64211ce3
-2015-04-30 13:42 +0000 [415a0d0745] Joshua Colp <jcolp@digium.com>
-
- * res_ari_device_states: Fix dependency on res_stasis_device_state.
-
- The res_ari_device_states module depends on res_stasis_device_state,
- not res_stasis_device_states.
-
- Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de
-
-2015-04-30 11:11 +0000 [e0c6f88010] Mark Michelson <mmichelson@digium.com>
-
- * Merge "chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option." into 13
-2015-04-30 10:53 +0000 [d1bc86fc99] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip_outbound_registration: Add virtual line support." into 13
-2015-04-29 14:29 +0000 [d3c310a28c] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option.
-
- Some telco switches occasionally ignore ISDN RESTART requests. The fix
- for ASTERISK-19608 added an escape clause for B channels in the restarting
- state if the telco ignores a RESTART request. If the telco fails to
- acknowledge the RESTART then Asterisk will assume the telco acknowledged
- the RESTART on the second call attempt requesting the B channel by the
- telco. The escape clause is good for dealing with RESTART requests in
- general but it does cause the next call for the restarting B channel to be
- rejected if the telco insists the call must go on that B channel.
-
- chan_dahdi doesn't really need to issue a RESTART request in response to
- receiving a cause 44 (Requested channel not available) code. Sending the
- RESTART in such a situation is not required (nor prohibited) by the
- standards. I think chan_dahdi does this for historical reasons to deal
- with buggy peers to get channels unstuck in a similar fashion as the
- chan_dahdi.conf resetinterval option.
-
- * Add the chan_dahdi.conf force_restart_unavailable_chans compatability
- option that when disabled will prevent chan_dahdi from trying to RESTART
- the channel in response to a cause 44 code.
-
- ASTERISK-25034 #close
- Reported by: Richard Mudgett
-
- Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
-2015-04-30 06:38 +0000 [7f611fa0e8] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8
-
- This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
- columns added in Asterisk 1.8. The columns are:
- * peeraccount
- * linkedid
- * sequence
- When enabled, the columns in the database entry will be populated with the data
- from the CDR.
-
- ASTERISK-24976 #close
-
- Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b
-
-2015-04-30 06:04 +0000 [e332c7ed5e] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_outbound_registration: Fix double unref on error return.
-
- When the PJSIP pjsip_regc_send function is invoked and an error
- status returned the caller currently decrements the reference count
- of the client state that it just incremented, assuming the
- registration callback would not have been invoked. In practice
- this is not correct. If the failure happens after the transaction
- has been set up the callback will still be invoked. This will
- cause the reference count to be incorrectly decremented twice, once
- by the registration callback and second by the caller of
- pjsip_regc_send.
-
- This change makes it so that whether the callback is invoked or
- not is known by the caller of pjsip_regc_send. Depending on
- this it can know whether it is responsible for decrementing the
- reference count of the client state or not.
-
- ASTERISK-25037 #close
- Reported by: Joshua Colp
-
- Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de
-
-2015-04-20 13:03 +0000 [9c3ed42875] Diederik de Groot <ddegroot@talon.nl>
-
- * Update configure.ac/Makefile for clang
-
- Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which
- checks compiler requirements for RAII:
- gcc: -fnested-functions support
- clang: -fblocks (and if required -lBlocksRuntime)
- The original check was implemented in configure.ac and now has it's
- own file. This function also sets C_COMPILER_FAMILY to either gcc or
- clang for use by makefile
-
- Created autoconf/ast_check_strsep_array_bounds.m4 (contains
- AST_CHECK_STRSEP_ARRAY_BOUNDS):
- which checks if clang is able to handle the optimized strsep & strcmp
- functions (linux). If not, the standard libc implementation should be
- used instead. Clang + the optimized macro's work with:
- strsep(char *, char []), but not with strsepo(char *, char *).
- Instead of replacing all the occurences throughout the source code,
- not using the optimized macro version seemed easier
-
- See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h':
- llvm-comment: Normally, this array-bounds warning are suppressed for
- macros, so that unused paths like the one that accesses __s1[3] are
- not warned about. But if you preprocess manually, and feed the
- result to another instance of clang, it will warn about all the
- possible forks of this particular if statement. Instead of switching
- of this optimization, another solution would be to run the preproces-
- sing step with -frewrite-includes, which should preserve enough
- information so that clang should still be able to suppress the diag-
- nostic at the compile step later on.
-
- See also "https://llvm.org/bugs/show_bug.cgi?id=20144"
- See also "https://llvm.org/bugs/show_bug.cgi?id=11536"
-
- Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning
- suppressions:
- -Wno-unused-value
- -Wno-parentheses-equality
- In an earlier review (reviewboard: 4550 and 4554), they were deemed a
- nuisace and less than benefitial.
-
- configure.ac:
- Added AST_CHECK_RAII() see earlier
- Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier
- Removed moved content
-
- ASTERISK-24917
- Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb
-
-2015-04-29 16:43 +0000 [37a193da18] Matt Jordan <mjordan@digium.com>
-
- * Merge "ARI: Fix missing dependencies." into 13
-2015-04-29 16:42 +0000 [6a86b3555b] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_fax: allow 2400 transmission rate according to v.27ter standard" into 13
-2015-04-29 16:15 +0000 [d4e207e27e] Matt Jordan <mjordan@digium.com>
-
- * main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8
-
- The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS
- structures in the RTP engine. However, when a 'core reload' is issued, a
- double free of the memory pointed to by the char *'s in the DTLS
- configuration struct can occur, as ast_rtp_dtls_cfg_free does not set
- the pointers to NULL when they are freed.
-
- This patch sets those pointers to NULL, preventing a second call to
- ast_rtp_dtls_cfg_free from corrupting memory.
-
- ASTERISK-25022
-
- Change-Id: I820471e6070a37e3c26f760118c86770e12f6115
-
-2015-04-29 13:05 +0000 [3fb6daeb55] Kevin Harwell <kharwell@digium.com>
-
- * res_fax: allow 2400 transmission rate according to v.27ter standard
-
- A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so
- a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits
- per second. This reverts all or some of those patches since according to the
- v.27ter standard a rate of 2400 bits per second is also supported.
-
- One of the original patches also added 9600 bits per second support for v.27.
- This patch also removes that since v.27ter only supports 2400/4800 bits per
- second.
-
- Also, since Asterisk specifically supports v.27ter the enum was renamed to
- better reflect this.
-
- ASTERISK-24955 #close
- Reported by: Matt Jordan
-
- Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733
-
-2015-04-29 10:46 +0000 [49ef81c15c] Joshua Colp <jcolp@digium.com>
-
- * res_sorcery_config: Fix build issue due to syntax error.
-
- Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac
-
-2015-04-28 00:29 +0000 [3278fe5327] Ashley Sanders <asanders@digium.com>
-
- * chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR
- Sections Exist in pjsip.conf
-
- This patch modifies the current loading strategy of the pjsip configuration. If
- duplicate sections (e.g. sections containing the same [id/type]) are defined in
- [pjsip.conf], the loader will consider the configuration for the given type as
- invalid when the duplicate section is encountered. The entire configuration
- (including what was previously loaded) for the duplicate [id/type] sections
- will be rejected and destroyed, an error message is logged and the load
- processing for the given stops.
-
- ASTERISK-24996
- Reported By: Ashley Sanders
-
- Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef
-
-2014-11-04 06:03 +0000 [89f6719f7a] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_outbound_registration: Add virtual line support.
-
- Virtual line support establishes a relationship between messages
- related to an outbound registration and a local endpoint. This is
- accomplished by attaching a parameter to the Contact of the outbound
- registration and looking for it on any received requests. If the
- parameter exists and can be matched to an outbound registration
- the configured endpoint is associated with the request.
-
- ASTERISK-24949 #close
- Reported by: Joshua Colp
-
- Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d
-
-2015-04-29 06:39 +0000 [d61f03c4f9] Corey Farrell <git@cfware.com>
-
- * ARI: Fix missing dependencies.
-
- ARI modules that are generated by 'make ari-stubs' are all dependent on
- res_ari_model. Additionally some of the same modules depend on one or more
- res_stasis_* modules.
-
- ASTERISK-25027 #close
- Reported by: Corey Farrell
-
- Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153
-
-2015-04-29 06:26 +0000 [3e4624ad21] Corey Farrell <git@cfware.com>
-
- * res_pjsip: Remove incorrect MODULEINFO from presence_xml.c.
-
- Remove incorrect MODULEINFO block and unneeded header includes
- from presence_xml.c.
-
- ASTERISK-25027
- Reported by: Corey Farrell
-
- Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb
-
-2015-04-29 06:17 +0000 [fed9faab8d] Corey Farrell <git@cfware.com>
-
- * Git Migration: Create doc/rest-api when needed.
-
- Create the directory './doc/rest-api' at the start of 'make ari-stubs'
- to prevent an error when documentation is generated. The directory is
- also added to git ignores.
-
- ASTERISK-25027
- Reported by: Corey Farrell
-
- Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b
-
-2015-04-29 05:17 +0000 [df23c8a86b] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_outbound_registration: Fix build due to removal of transaction.
-
- Change-Id: I7a8a7beec3334cec304943f2dd7597eabe2e3150
-
-2015-04-28 19:18 +0000 [95ab9fdb1a] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_pjsip_outbound_registration: Add debugging messages." into 13
-2015-04-28 19:18 +0000 [0e70dc0dc8] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_pjsip_outbound_registration: Don't fail on delayed processing: 13." into 13
-2015-04-27 16:56 +0000 [e39bd6ba46] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_outbound_registration: Don't fail on delayed processing: 13.
-
- This is the Asterisk 13 version of a change to master that allows for
- registration responses to be processed successfully potentially after
- the original transaction has timed out. The main difference between this
- and the master change is that the master version has API changes that
- are unacceptable for 13. For 13, this is worked around by adding a new
- API call that the outbound registration code uses instead.
-
- The following is the text from the master version of this commit:
-
- Odd behaviors have been observed during outbound registrations. The most
- common problem witnessed has been one where a request with
- authentication credentials cannot be created after receiving a 401
- response. Other behaviors include apparently processing an incorrect SIP
- response.
-
- Inspecting the code led to an apparent issue with regards to how we
- handle transactions in outbound registration code. When a response to a
- REGISTER arrives, we save a pointer to the transaction and then push a
- task onto the registration serializer. Between the time that we save the
- pointer and push the task, it's possible for the transaction to be
- destroyed due to a timeout. It's also possible for the address to be
- reused by the transaction layer for a new transaction.
-
- To allow for authentication of a REGISTER request to be authenticated
- after the transaction has timed out, we now also hold a reference to the
- original REGISTER request instead of the transaction. The function for
- creating a request with authentication has been altered to take the
- original request instead of the transaction where the original request
- was sent.
-
- ASTERISK-25020
- Reported by Mark Michelson
-
- Change-Id: If1ee5f601be839479a219424f0358a229f358f7c
-2015-04-27 14:44 +0000 [1bf008fc76] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_outbound_registration: Add debugging messages.
-
- When problems occur regarding outbound registrations, it currently
- is difficult to debug. Most off-nominal paths had warning messages,
- but sometimes we want to know what's going on before hitting the
- off-nominal path. This patch adds lots of debugging output that
- should give a clearer picture of what is happening with regards
- to outbound registrations.
-
- ASTERISK-25020
- Reported by Mark Michelson
-
- Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45
-
-2015-04-28 07:13 +0000 [7ee05892d6] Joshua Colp <jcolp@digium.com>
-
- * Merge "Example script for scan-build (the llvm static analyzer)" into 13
-2015-04-28 05:38 +0000 [0b6410c4f8] Steve Davies <steve@one47.co.uk>
-
- * res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS
-
- ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created.
- The resources are linked into a table, but the original alloc refs
- are never released. ast_strdup leak in rtp_engine.c. If
- ast_rtp_dtls_cfg_copy() is called twice on the same destination struct,
- a pointer to an alloc'd string is overwritten before the string is free'd.
-
- ASTERISK-25022
- Reported by: one47
-
- Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b
-
-2015-04-28 06:55 +0000 [427209603d] Joshua Colp <jcolp@digium.com>
-
- * Merge "cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version" into 13
-2015-04-27 12:11 +0000 [99fb87ae13] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip: Fix SEGV on pending-qualify contacts
-
- Permanent contacts that hadn't been qualified yet were missing
- their contact_status entries causing SEGVs when running CLI
- commands.
-
- This patch makes sure that contact_statuses are created for
- both dynamic and permanent contacts when they are created.
- It also adds checks in the CLI code to make sure there's a
- contact_status, just in case.
-
- ASTERISK-25018 #close
- Reported-by: Ivan Poddubny
- Tested-by: Ivan Poddubny
- Tested-by: George Joseph
-
- Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029
-
-2015-04-15 18:55 +0000 [d5dd43856e] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version
-
- Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
- * peeraccount
- * linkedid
- * sequence
- This feature is configurable in cdr_odbc.conf using a new configuration
- option, 'newcdrcolumns'.
-
- ASTERISK-24976 #close
-
- Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127
-2015-04-26 17:21 +0000 [e9788056e9] Matt Jordan <mjordan@digium.com>
-
- * channels/chan_skinny: Fix compilation error introduced in f8e21a1adf
-
- A typo in commit f8e21a1adf resulted in a compilation error in
- chan_skinny. This patch fixes the typo.
-
- ASTERISK-24917
-
- Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c
-
-2015-04-26 15:53 +0000 [2d277996b7] Matt Jordan <mjordan@digium.com>
-
- * Merge "Clang: Fix some more tautological-compare warnings." into 13
-2015-04-24 13:07 +0000 [145f65598c] Matt Jordan <mjordan@digium.com>
-
- * Merge "app_confbridge: Default the template option to a compatible default profile." into 13
-2015-04-23 15:11 +0000 [7e5056b393] Kevin Harwell <kharwell@digium.com>
-
- * app_confbridge: Default the template option to a compatible default profile.
-
- Confbridge dynamic profiles did not have a default profile unless you
- explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a
- template was not set prior to the bridge being created then some
- options were left with no default values set. This patch makes it so
- the default templates are set to the default bridge and user profiles.
-
- ASTERISK-24749 #close
- Reported by: philippebolduc
-
- Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a
-
-2015-04-24 09:17 +0000 [1da9ec969d] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_outbound_authenticator: Increase CSeq on authed requests.
-
- The way PJSIP generates an authenticated request is to use a previous
- request as a template. This means that the authenticated request will
- have the same Call-ID, From header (including tag), and CSeq as the
- original request. PJSIP generates a new branch on the Via header to
- indicate that this is a new transaction, though.
-
- There are some SIP implementations, though, that do not notice the
- change in the branch and therefore will match the authed request to the
- original request's transaction. Since the CSeq is the same, the server
- will repeat the response it sent to the original request.
-
- This patch aids interoperability by increasing the CSeq of the authed
- request by one.
-
- ASTERISK-24845 #close
- Reported by: Carl Fortin
- Tested by: Carl Fortin
-
- Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01
-
-2015-04-24 09:24 +0000 [bf3d9db4a6] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into 13
-2015-04-20 13:06 +0000 [cb318f3960] Diederik de Groot <ddegroot@talon.nl>
-
- * Example script for scan-build (the llvm static analyzer)
-
- - Added Pre-amble (Options / Flags / Usage Example / GNU License)
- - Extended Configurability
- - Made Executable
-
- ASTERISK-24917
- Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8
-
-2015-04-23 17:23 +0000 [b3cd5bc77f] Mark Michelson <mmichelson@digium.com>
-
- * Merge "Clang: change previous tautological-compare fixes." into 13
-2015-04-23 12:54 +0000 [eabf3b5a3c] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX.
-
- When Asterisk originates a channel to an application, the channel is
- hung up once the application finishes executing. When the application
- in question is SendFax, the Asterisk PJSIP code will attempt to reinvite
- the T.38 session to audio after the FAX completes. The hangup of the
- channel happens in the midst of this reinvite transaction. In most
- circumstances, this works out okay because the BYE is delayed until the
- reinvite transaction can complete.
-
- However, if the reinvite that Asterisk sends receives a 401/407
- response, then Asterisk's attempt to re-send the reinvite with
- authentication will fail. This is because the session supplement in
- res_pjsip_t38 makes the assumption that the channel on the session will
- always be non-NULL. Since the channel has been hung up, though, the
- channel is now NULL. Attempting to operate on the channel causes a
- crash.
-
- This patch fixes the issue by ensuring that the channel on the session
- is not NULL before attempting to mess with the T.38 framehook.
-
- This patch also contains some corrections for comments that were
- incorrect and really confused me when I first started looking at the
- code.
-
- ASTERISK-25004 #close
- Reported by Mark Michelson
-
- Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0
-2015-04-23 09:16 +0000 [f70d21b2cf] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip: Validate that contact uris start with sip: or sips:
-
- Currently we use pjsip_parse_hdr to validate contact uris but it
- appears that it allows uris without a scheme if there's a port
- supplied. I.E myexample.com will fail but myexample.com:5060 will
- pass even though it has no scheme. This causes SEGVs later on
- whenever the uri is used.
-
- To prevent this, permanent_contact_validate has been updated to check
- that the scheme is either 'sip' or 'sips'.
-
- 2 uses of possibly-null endpoint have also been fixed in
- create_out_of_dialog_request.
-
- ASTERISK-24999
-
- Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2
- Reported-by: Brad Latus
-
-2015-04-23 08:00 +0000 [1bb16bedc7] Diederik de Groot <ddegroot@talon.nl>
-
- * Clang: change previous tautological-compare fixes.
-
- clang can warn about a so called tautological-compare, when it finds
- comparisons which are logically always true, and are therefor deemed
- unnecessary.
-
- Exanple:
- unsigned int x = 4;
- if (x > 0) // x is always going to be bigger than 0
-
- Enum Case:
- Each enumeration is its own type. Enums are an integer type but they
- do not have to be *signed*. C leaves it up to the compiler as an
- implementation option what to consider the integer type of a particu-
- lar enumeration is. Gcc treats an enum without negative values as
- an int while clang treats this enum as an unsigned int.
-
- rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
- The cast does have an effect. For gcc, which seems to treat all enums
- as int, the cast to unsigned int will eliminate the possibility of
- negative values being allowed. For clang, which seems to treat enums
- without any negative members as unsigned int, the cast will have no
- effect. If for some reason in the future a negative value is ever
- added to the enum the assert will still catch the negative value.
-
- ASTERISK-24917
-
- Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a
-
-2015-04-23 06:50 +0000 [a06924e9d9] Matt Jordan <mjordan@digium.com>
-
- * Merge "Astobj2: Ensure all calls to __adjust_lock pass a valid object." into 13
-2015-04-22 16:22 +0000 [1474bb05f6] George Joseph <george.joseph@fairview5.com>
-
- * res_corosync: Add check for config file before calling corosync apis
-
- On some systems, res_corosync isn't compatible with the installed version of
- corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE,
- and Asterisk terminates. The work around has been to remember to add
- res_corosync as a noload in modules.conf. A better solution though is to have
- res_corosync check for its config file before attempting to call corosync apis
- and return LOAD_DECLINE if there's no config file. This lets Asterisk loading
- continue.
-
- If you have a res_corosync.conf file and res_corosync fails, you get the same
- behavior as today and the fatal error tells you something is wrong with the
- install.
-
- ASTERISK-24998
-
- Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889
-2015-04-22 15:17 +0000 [73efb093b8] Corey Farrell <git@cfware.com>
-
- * Astobj2: Ensure all calls to __adjust_lock pass a valid object.
-
- __adjust_lock doesn't check for invalid objects, and doesn't have an
- appropriate return value for invalid objects. Most callers of
- __adjust_lock pass objects that have already been confirmed valid,
- this change adds checks before the remaining calls.
-
- ASTERISK-24997 #close
- Reported by: Corey Farrell
-
- Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f
-
-2015-04-22 16:32 +0000 [b0e929219b] George Joseph <george.joseph@fairview5.com>
-
- * .gitignore: Add .gcno and .gcda
-
- Products of --enable-coverage
-
- Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3
-
-2015-04-22 14:25 +0000 [5a3948a66f] Matt Jordan <mjordan@digium.com>
-
- * Merge "Fix/Update clang-RAII macro implementation" into 13
-2015-04-22 14:07 +0000 [2ef1e1fc68] Mark Michelson <mmichelson@digium.com>
-
- * Merge "res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers." into 13
-2015-04-22 04:17 +0000 [d6dfc85666] Diederik de Groot <ddegroot@talon.nl>
-
- * Clang: Fix some more tautological-compare warnings.
-
- clang can warn about a so called tautological-compare, when it finds
- comparisons which are logically always true, and are therefor deemed
- unnecessary.
-
- Exanple:
- unsigned int x = 4;
- if (x > 0) // x is always going to be bigger than 0
-
- Enum Case:
- Each enumeration is its own type. Enums are an integer type but they
- do not have to be *signed*. C leaves it up to the compiler as an
- implementation option what to consider the integer type of a particu-
- lar enumeration is. Gcc treats an enum without negative values as
- an int while clang treats this enum as an unsigned int.
-
- rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
- The cast does have an effect. For gcc, which seems to treat all enums
- as int, the cast to unsigned int will eliminate the possibility of
- negative values being allowed. For clang, which seems to treat enums
- without any negative members as unsigned int, the cast will have no
- effect. If for some reason in the future a negative value is ever
- added to the enum the assert will still catch the negative value.
-
- ASTERISK-24917
- Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62
-
-2015-04-22 05:45 +0000 [edd9e54818] Joshua Colp <jcolp@digium.com>
-
- * Merge "Check for ao2_alloc failure in __ast_channel_internal_alloc." into 13
-2015-04-14 14:04 +0000 [7b57116833] Joshua Colp <jcolp@digium.com>
-
- * res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers.
-
- Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon
- a mailbox state change (such as a new message being left, or one being deleted).
- In practice this is not sufficient to keep clients aware of the current MWI status.
-
- This change makes the module send unsolicited MWI NOTIFY on startup so that
- clients are guaranteed to have the most up to date MWI information. It also makes
- clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware
- of the current MWI status they receive it.
-
- ASTERISK-24982 #close
- Reported by: Joshua Colp
-
- Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58
-
-2015-04-22 05:29 +0000 [4423d5f755] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." into 13
-2015-04-21 15:17 +0000 [ad1a118632] Corey Farrell <git@cfware.com>
-
- * Check for ao2_alloc failure in __ast_channel_internal_alloc.
-
- Fix a crash that could occur in __ast_channel_internal_alloc if
- ao2_alloc fails.
-
- ASTERISK-24991 #close
-
- Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90
-
-2015-04-20 14:30 +0000 [3327560cb2] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs.
-
- When SUBSCRIBE dialogs were established, we never associated
- the endpoint that created the subscription with the dialog
- we end up creating. In most cases, this ended up not causing
- any problems.
-
- The actual bug that was observed was that when a device that
- was behind NAT established a subscription with Asterisk, Asterisk
- would end up sending in-dialog NOTIFY requests to the device's
- private IP addres instead of the public address of the NAT router.
-
- When Asterisk receives the initial SUBSCRIBE from the device,
- res_pjsip_nat rewrites the contact to the public address on which the
- SUBSCRIBE was received. This allows for the dialog to have its target
- address set to the proper public address. Asterisk then would send a 200
- OK response to the SUBSCRIBE, then a NOTIFY with the initial
- subscription state. The device would then send a 200 OK response to
- Asterisk's NOTIFY.
-
- Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat
- did not rewrite the address in the Contact header. Then, when the PJSIP
- dialog layer processed the 200 OK, PJSIP would perform a comparison
- between the IP address in the Contact header and its saved target
- address for the dialog. Since they differed, PJSIP would update the
- target dialog address to be the address in the Contact header. From this
- point, if Asterisk needed to send a NOTIFY to the device, the result was
- that the NOTIFY would be sent to the private address that the device
- placed in the Contact header.
-
- The reason why res_pjsip_nat did not rewrite the address when it
- received the 200 OK response was that it could not associate the
- incoming response with a configured endpoint. This is because on a
- response, the only way to associate the response to an endpoint is by
- finding the dialog that the response is associated with and then finding
- the endpoint that is associated with that dialog. We do not perform
- endpoint lookups on responses. res_pjsip_pubsub skipped the step of
- associating the endpoint with the dialog we created, so res_pjsip_nat
- could not find the associated endpoint and therefore couldn't rewrite
- the contact.
-
- This commit message is like 50x longer than the actual fix.
-
- ASTERISK 24981 #close
- Reported by Mark Michelson
-
- Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd
-2015-04-20 18:00 +0000 [d08446ec36] Richard Mudgett <rmudgett@digium.com>
-
- * chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels.
-
- The chan_dahdi channel driver is a very old driver. The ability for it to
- support ISDN was added well after the initial analog support. Setting the
- softhangup flags is a carry over from the original analog code. The
- driver was not updated to call ast_queue_hangup() which will post the AMI
- HangupRequest event.
-
- * Changed sig_pri.c to call ast_queue_hangup() instead of setting the
- softhangup flag when the remote party initiates a hangup.
-
- ASTERISK-24895 #close
- Reported by: Andrew Zherdin
-
- Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325
-
-2015-04-20 17:23 +0000 [96e18453f4] Joshua Colp <jcolp@digium.com>
-
- * Merge "pjsip_options: Fix non-qualified contacts showing as unavailable" into 13
-2015-04-20 13:01 +0000 [2be9cc2643] Diederik de Groot <ddegroot@talon.nl>
-
- * Fix/Update clang-RAII macro implementation
-
- - When you need to refer to 'variable XXX' outside a block, it needs
- to be declared as '__block XXX', otherwise it will not be available with-
- in the block, making updating that variable hard to do, and ast_free
- lead to issues.
-
- - Removed the #error message
- because it creates complications when compiling external projects
- against asterisk For example when using a different compiler than the
- one used to compile asterisk. The warning/error should be generated
- during the configure process not the compilation process
-
- ASTERISK-24917
- Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2
-2015-04-20 09:53 +0000 [b74b2cdcda] George Joseph <george.joseph@fairview5.com>
-
- * pjsip_options: Fix format specifier for int64_t rtt.
-
- Contact status rtt is an int64_t and needs the PRId64 macro to
- properly create the format specifier on 32-bit systems.
-
- Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7
-
-2015-04-20 06:29 +0000 [27a122af66] Matt Jordan <mjordan@digium.com>
-
- * Merge "main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple" into 13
-2015-04-20 05:54 +0000 [9581a0ebf3] Joshua Colp <jcolp@digium.com>
-
- * Merge "Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled." into 13
-2015-04-18 13:36 +0000 [63169e00ff] George Joseph <george.joseph@fairview5.com>
-
- * pjsip_options: Fix non-qualified contacts showing as unavailable
-
- The "Add qualify_timeout processing and eventing" patch introduced
- an issue where contacts that had qualify_frequency set to 0 were
- showing Unavailable instead Unknown. This patch checks for
- qualify_frequency=0 and create an "Unknown" contact_status
- with an RTT = 0.
-
- Previously, the lack of contact_status implied Unknown but since
- we're now changing endpoint state based on contact_status, I've
- had to add new UNKNOWN status so that changes could trigger the
- appropriate contact_status observers.
-
- ASTERISK-24977: #close
-
- Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
-
-2015-04-19 15:49 +0000 [f0c82a173a] Matt Jordan <mjordan@digium.com>
-
- * main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple
-
- When a PBX registrar is unloaded, it will fail to remove its extension from
- the context root_table if a dialplan application used by that extension is
- still loaded. This can be the case for AGI, which can be unloaded after several
- of the standard PBX providers. Often, this is harmless; however, if the
- extension's priorities are removed during the failed unloading *and* the
- dialplan application later unregisters, it leaves a ticking timebomb for the
- next PBX provider that attempts to iterate over the extensions. When that
- occurs, the peer_table pointer on the extension will already be set to NULL.
- The current code does not check to see if the pointer is NULL before passing
- it to a hashtab function this is not NULL tolerant.
-
- Since it is possible for the peer_table to be NULL when we normally would not
- expect that to be the case, the solution in this patch is to simply skip over
- processing an extension's priorities if peer_table is NULL.
-
- Prior to this patch, the tests/pbx/callerid_match test would crash during
- module unload. With this patch, the test no longer crashes after running.
-
- ASTERISK-24774 #close
- Reported by: Corey Farrell
-
- Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40
-
-2015-04-17 18:05 +0000 [82bc0fd3ad] Richard Mudgett <rmudgett@digium.com>
-
- * res_fax: Fix latent bug exposed by ASTERISK-24841 changes.
-
- Three fax related tests started failing as a result of changes made for
- ASTERISK-24841:
- tests/fax/pjsip/gateway_t38_g711
- tests/fax/sip/gateway_mix1
- tests/fax/sip/gateway_mix3
-
- Historically, ast_channel_make_compatible() did nothing if the channels
- were already "compatible" even if they had a sub-optimal translation path
- already setup. With the changes from ASTERISK-24841 this is no longer
- true in order to allow the best translation paths to always be picked. In
- res_fax.c:fax_gateway_framehook() code manually setup the channels to go
- through slin and then called ast_channel_make_compatible(). With the
- previous version of ast_channel_make_compatible() this was always a
- no-operation.
-
- * Remove call to ast_channel_make_compatible() in fax_gateway_framehook()
- that now undoes what was just setup when the framehook is attached.
-
- * Fixed locking around saving the channel formats in
- fax_gateway_framehook() to ensure that the formats that are saved are
- consistent.
-
- * Fix copy pasta errors in fax_gateway_framehook() that confuses read and
- write when dealing with saved channel formats.
-
- ASTERISK-24841
- Reported by: Matt Jordan
-
- Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d
-
-2015-04-17 16:19 +0000 [c59a800707] Corey Farrell <git@cfware.com>
-
- * Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled.
-
- When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be
- called as a function. This causes a compile error with raw threadstorage as
- it uses NULL for cleanup. This fix uses a macro that provides NULL when
- DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);"
- with "{};" when DEBUG_THREADLOCALS is enabled.
-
- ASTERISK-24975 #close
- Reported by: Ashley Sanders
-
- Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402
-
-2015-04-17 15:57 +0000 [e05b076827] Matt Jordan <mjordan@digium.com>
-
- * Merge "Detect potential forwarding loops based on count." into 13
-2015-04-15 10:38 +0000 [4f1a8dbe92] Mark Michelson <mmichelson@digium.com>
-
- * Detect potential forwarding loops based on count.
-
- A potential problem that can arise is the following:
-
- * Bob's phone is programmed to automatically forward to Carol.
- * Carol's phone is programmed to automatically forward to Bob.
- * Alice calls Bob.
-
- If left unchecked, this results in an endless loops of call forwards
- that would eventually result in some sort of fiery crash.
-
- Asterisk's method of solving this issue was to track which interfaces
- had been dialed. If a destination were dialed a second time, then
- the attempt to call that destination would fail since a loop was
- detected.
-
- The problem with this method is that call forwarding has evolved. Some
- SIP phones allow for a user to manually forward an incoming call to an
- ad-hoc destination. This can mean that:
-
- * There are legitimate use cases where a device may be dialed multiple
- times, or
- * There can be human error when forwarding calls.
-
- This change removes the old method of detecting forwarding loops in
- favor of keeping a count of the number of destinations a channel has
- dialed on a particular branch of a call. If the number exceeds the
- set number of max forwards, then the call fails. This approach has
- the following advantages over the old:
-
- * It is much simpler.
- * It can detect loops involving local channels.
- * It is user configurable.
-
- The only disadvantage it has is that in the case where there is a
- legitimate forwarding loop present, it takes longer to detect it.
- However, the forwarding loop is still properly detected and the
- call is cleaned up as it should be.
-
- Address review feedback on gerrit.
-
- * Correct "mfgium" to "Digium"
- * Decrement max forwards by one in the case where allocation of the
- max forwards datastore is required.
- * Remove irrelevant code change from pjsip_global_headers.c
-
- ASTERISK-24958 #close
-
- Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
-2015-04-11 16:56 +0000 [674b18bdf0] George Joseph <george.joseph@fairview5.com>
-
- * pjsip_options: Add qualify_timeout processing and eventing
-
- This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
- discussion at
- http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
-
- The basic issues are that changes in contact status don't cause events to be
- emitted for the associated endpoint. Only dynamic contact add/delete actions
- update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
- which is a long time.
-
- This patch makes use of the new transaction timeout feature in r4585 and
- provides the following capabilities...
-
- 1. A new aor/contact variable 'qualify_timeout' has been added that allows the
- user to specify the maximum time in milliseconds to wait for a response to an
- OPTIONS message. The default is 3000ms. When the timer expires, the contact is
- marked unavailable.
-
- 2. Contact status changes are now propagated up to the endpoint as follows...
- When any contact is 'Available', the endpoint is marked as 'Reachable'. When
- all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
- existing endpoint events are generated appropriately.
-
- ASTERISK-24863 #close
-
- Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
- Tested-by: Dmitriy Serov
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
-2015-04-17 15:29 +0000 [f1abf51b73] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13
-2015-04-17 10:30 +0000 [ab5b38e434] Matt Jordan <mjordan@digium.com>
-
- * Merge "res_pjsip: Add global option to limit the maximum time for initial qualifies" into 13
-2015-04-17 10:25 +0000 [ec77b6148f] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced" into 13
-2015-04-16 10:51 +0000 [b56c1914fa] Kevin Harwell <kharwell@digium.com>
-
- * bridge.c: NULL app causes crash during attended transfer
-
- Due to a race condition there was a chance that during an attended transfer the
- channel's application would return NULL. This, of course, would cause a crash
- when attempting to access the memory. This patch retrieves the channel's app
- at an earlier time in processing in hopes that the app name is available.
- However, if it is not then "unknown" is used instead. Since some string value
- is now always present the crash can no longer occur.
-
- ASTERISK-24869 #close
- Reported by: viniciusfontes
- Review:
-
- Change-Id: I5134b84c4524906d8148817719d76ffb306488ac
-
-2015-04-16 13:20 +0000 [8d4ce7cc2b] Scott Griepentrog <scott@griepentrog.com>
-
- * res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced
-
- This change makes the send_notify of the sub_tree
- not happen when the sub_tree has been deleted due
- to the notify call failing, which avoids a crash.
-
- ASTERISK-24970 #close
-
- Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf
-2015-04-11 16:39 +0000 [bf46799f0e] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip: Refactor endpt_send_request to include transaction timeout
-
- This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
- discussion at
- http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
-
- Since we currently have no control over pjproject transaction timeout, this
- patch pulls the pjsip_endpt_send_request function out of pjproject and into
- res_pjsip/endpt_send_transaction in order to implement that capability.
-
- Now when the transaction is initiated, we also schedule our own pj_timer with
- our own desired timeout.
-
- If the transaction completes before either timeout, pjproject cancels its timer,
- and calls our tsx callback where we cancel our timer and run the app callback.
-
- If the pjproject timer times out first, pjproject calls our tsx callback where
- we cancel our timer and run the app callback.
-
- If our timer times out first, we terminate the transaction which causes
- pjproject to cancel its timer and call our tsx callback where we run the app
- callback.
-
- Regardless of the scenario, pjproject is calling the tsx callback inside the
- group_lock and there are checks in the callback to make sure it doesn't run
- twice.
-
- As part of this patch ast_sip_send_out_of_dialog_request was created to replace
- its similarly named private function. It takes a new timeout argument in
- milliseconds (<= 0 to disable the timeout).
-
- ASTERISK-24863 #close
- Reported-by: George Joseph <george.joseph@fairview5.com>
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
- Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
-2015-04-11 17:04 +0000 [1b6f6ff841] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip: Add global option to limit the maximum time for initial qualifies
-
- Currently when Asterisk starts initial qualifies of contacts are spread out
- randomly between 0 and qualify_timeout to prevent network and system overload.
- If a contact's qualify_frequency is 5 minutes however, that contact may be
- unavailable to accept calls for the entire 5 minutes after startup. So while
- staggering the initial qualifies is a good idea, basing the time on
- qualify_timeout could leave contacts unavailable for too long.
-
- This patch adds a new global parameter "max_initial_qualify_time" that sets the
- maximum time for the initial qualifies. This way you could make sure that all
- your contacts are initialy, randomly qualified within say 30 seconds but still
- have the contact's ongoing qualifies at a 5 minute interval.
-
- If max_initial_qualify_time is > 0, the formula is initial_interval =
- min(max_initial_interval, qualify_timeout * random(). If not set,
- qualify_timeout is used.
-
- The default is "0" (disabled).
-
- ASTERISK-24863 #close
-
- Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
-2015-04-15 16:08 +0000 [5d218cde87] George Joseph <george.joseph@fairview5.com>
-
- * More .gitignore updates
-
- Added .pyc and .sha1 to the top-level .gitignore.
-
- Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a
- Tested-by: George Joseph <george.joseph@fairview5.com>
-
-2015-04-15 13:36 +0000 [97f83c4c53] Matt Jordan <mjordan@digium.com>
-
- * Merge "Build System: Replace comment about setting menuselect defaults." into 13
-2015-04-14 13:16 +0000 [abd56db3e0] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * cel_pgsql: Fix name string for log on unable allocate memory.
-
- The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and
- LENGTHEN_BUF2.
-
- ASTERISK-24965 #close
- Reported by: Rodrigo Ramirez Norambuena
-
- Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744
-2015-04-14 13:48 +0000 [222fbe1d9a] Corey Farrell <git@cfware.com>
-
- * Build System: Replace comment about setting menuselect defaults.
-
- The Makefile claims that you can set default menuselect options by creating
- ~/.asterisk.makeopts or /etc/asterisk.makeopts, but those files have never
- been respected in Asterisk 11 or 13. This changes the comment to accurately
- reflect that these files are not automatically used by the build system.
-
- ASTERISK-13721 #close
- Reported by: pj
-
- Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2
-
-2015-04-12 09:08 +0000 [07e729cc7b] Rodrigo Ramírez Norambuena <decipher.hk@gmail.com>
-
- * cdr_pgsql: Fix CLI "cdr show pgsql status" command.
-
- The command always showed the usage information.
-
- * Fix the error in command validation for CLI_SHOWUSAGE.
-
- ASTERISK-24959 #close
- Reported by: Rodrigo Ramirez Norambuena
-
- Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5
- (cherry picked from commit 23a180cade51e84b9def65b05759c3cb9feba225)
-
-2015-04-13 19:06 +0000 [7d43d85bea] George Joseph <george.joseph@fairview5.com>
-
- * .gitignore updates for master/13
-
- Added products of ./bootstrap
-
- Added nmenuselect and gmenuselect to menuselect/
-
- Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35
-
-2015-04-13 14:41 +0000 [3d27c223a5] David M. Lee <dlee@respoke.io>
-
- * Fixing extconf compile
-
- During the mass code deletion for clang support, a stray backslash was
- left behind that was causing utils to fail to compile.
-
- Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
-
-2015-04-13 12:03 +0000 [30045b4e67] Matt Jordan <mjordan@digium.com>
-
- * Merge "build_tools/make_version: Update version parsing for Git migration" into 13
-2015-04-13 10:47 +0000 [88dbf6653e] Joshua Colp <jcolp@digium.com>
-
- * Merge "res_monitor: Add dependency on func_periodic_hook." into 13
-2015-04-13 09:54 +0000 [e996d8f728] Matt Jordan <mjordan@digium.com>
-
- * build_tools/make_version: Update version parsing for Git migration
-
- External systems - such as the Asterisk Test Suite - require knowledge of the
- upstream branch. Unfortunately, after moving to Git, the Asterisk version
- currently consists of only a 'GIT" prefix followed by an object blob,
- e.g., GIT-as08d7. This makes it difficult for such systems to know what
- features are available in a particular check out of Asterisk.
-
- This patch fixes this by hardcoding the branch in a variable in the
- make_version script. Since the mainline branches are not changed often -
- typically only once a year - this is a reasonable approach to solving
- the problem, and is more reliable than parsing the output of 'git branch
- -vv'. Branches that track off of an upstream primary branch will then get the
- benefit of knowing which mainline branch they are currently based off
- of.
-
- ASTERISK-24954 #close
-
- Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799
-
-2015-04-12 12:59 +0000 [d1a6f1a9f9] Matt Jordan <mjordan@digium.com>
-
- * git migration: Remove support for file versions
-
- Git does not support the ability to replace a token with a version
- string during check-in. While it does have support for replacing a
- token on clone, this is somewhat sub-optimal: the token is replaced
- with the object hash, which is not particularly easy for human
- consumption. What's more, in practice, the source file version was often
- not terribly useful. Generally, when triaging bugs, the overall version
- of Asterisk is far more useful than an individual SVN version of a file.
- As a result, this patch removes Asterisk's support for showing source file
- versions.
-
- Specifically, it does the following:
- * main/asterisk:
- - Refactor the file_version structure to reflect that it no longer
- tracks a version field.
- - Alter the "core show file version" CLI command such that it always
- reports the version of Asterisk. The file version is no longer
- available.
-
- * main/manager: The Version key now always reports the Asterisk version.
-
- * UPGRADE: Add notes for:
- - Modification to the ModuleCheck AMI Action.
- - Modification of the "core show file version" CLI command.
-
- Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28
-
-2015-04-13 06:19 +0000 [0e4b997cd7] Corey Farrell <git@cfware.com>
-
- * res_monitor: Add dependency on func_periodic_hook.
-
- OPTIONAL_API has conditionals to define AST_OPTIONAL_API and
- AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined.
- Unfortunately this is inside the include protection block, so only the
- first status of AST_API_MODULE is respected. For example res_monitor
- is an optional API provider, but uses func_periodic_hook. This makes
- func_periodic_hook non-optional to res_monitor.
-
- ASTERISK-17608 #close
- Reported by: Warren Selby
-
- Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679
-
-2015-04-12 15:27 +0000 [91c1ed7ef6] Matt Jordan <mjordan@digium.com>
-
- * Merge "main/editline: Add .gitignore." into 13
-2015-04-12 06:12 +0000 [a77c31b99c] Corey Farrell <git@cfware.com>
-
- * main/editline: Add .gitignore.
-
- This patch adds a .gitignore for main/editline to ignore all build results.
-
- Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d
-
-2015-04-11 23:22 +0000 [d918c3b78e] Matt Jordan <mjordan@digium.com>
-
- * .gitignore: Ignore tarballs (*.gz)
-
- This patch updates the root .gitignore file to ignore files with a .gz
- extension. This will cause git to ignore downloaded sound tarballs in
- the the sounds/ directory.
-
- Change-Id: I1e42fbfa02a8884231507b683e8e49ac3e278aaa
-
-2015-04-11 13:20 +0000 [555b5f5d30] George Joseph <george.joseph@fairview5.com>
-
- * Add .gitignore and .gitreview files
-
- Add the .gitignore and .gitreview files to the asterisk repo.
-
- NB: You can add local ignores to the .git/info/exclude file
- without having to do a commit.
-
- Common ignore patterns are in the top-level .gitignore file.
- Subdirectory-specific ignore patterns are in their own .gitignore
- files.
-
- Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696
- Tested-by: George Joseph
-
-2015-04-11 10:35 +0000 [5807ca519c] Matthew Jordan <mjordan@digium.com>
-
- * Blocked revisions 434708
-
- ........
- main/event: Remove unnecessary assignment of negative value to enum
-
- When cleaning up some clang compiler warnings, the comparison of a negative
- value to an unsigned enum was removed. However, the initial assignment of a
- negative value to said enum remained in the variable declaration. This patch
- removes that assignment.
-
- Thanks to ibercom in #asterisk-bugs for pointing it out.
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-11 10:26 +0000 [d0d78d5732] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix various warnings for tests
-
- This patch fixes a variety of clang compiler warnings for unit tests. This
- includes autological comparison issues, ignored return values, and
- interestingly enough, one embedded function. Fun!
-
- Review: https://reviewboard.asterisk.org/r/4555
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4555.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434705 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-11 10:10 +0000 [4cf7d0bf01] Juergen Spies (License 6698)
-
- * res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram
-
- Prior to this patch, the far_max_datagram value on the UDPTL structure would
- remain -1 if the remote endpoint fails to provide the SDP media attribute
- T38FaxMaxDatagram. This can result in the INVITE request being rejected. With
- this patch, we will now properly initialize the value with either the default
- value or with the value provided by pjsip.conf's t38_udptl_maxdatagram
- parameter.
-
- Review: https://reviewboard.asterisk.org/r/4589
-
- ASTERISK-24928 #close
- Reported by: Juergen Spies
- Tested by: Juergen Spies
- patches:
- pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698)
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 18:29 +0000 [13cd99682d] Richard Mudgett <rmudgett@digium.com>
-
- * chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices.
-
- With this patch, chan_pjsip/res_pjsip now sets the native formats to the
- codecs negotiated by a call.
-
- * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
- formats to include all the negotiated audio codecs instead of only the
- initial preferred audio codec and later the currently received audio
- codec.
-
- * The audio frame handling in channel.c:ast_read() is more streamlined and
- will automatically adjust to changes in received frame formats. The new
- policy is to remove translation and pass the new frame format to the
- receiver except if the translation was to a signed linear format. A more
- long winded version is commented in ast_read() along with some caveats.
-
- * The audio frame handling in channel.c:ast_write() is more streamlined
- and will automatically adjust any needed translation to changes in the
- frame formats sent. Frame formats sent can change for many reasons such
- as a recording is being played back or the bridged peer changed the format
- it sends. Since it is a normal expectation that sent formats can change,
- the codec mismatch warning message is demoted to a debug message.
-
- * Removed the short circuit check in
- channel.c:ast_channel_make_compatible_helper(). Two party bridges need to
- make channels compatible with each other. However, transfers and moving
- channels among bridges can result in otherwise compatible channels having
- sub-optimal translation paths if the make compatible check is short
- circuited. A result of forcing the reevaluation of channel compatibility
- is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
- options take effect consistently now. It is unfortunate that these two
- options are enabled by default and negate some of the benefits to the
- changes in channel.c:ast_read() by forcing translation through signed
- linear on a two party bridge.
-
- * Improved the softmix bridge technology to better control the translation
- of frames to the bridge. All of the incoming translation is now normally
- handled by ast_read() instead of splitting any translation steps between
- ast_read() and the slin factory. If any frame comes in with an unexpected
- format then the translation path in ast_read() is updated for the next
- frame and the slin factory handles the current frame translation.
-
- This is the final patch in a series of patches aimed at improving
- translation path choices. The other patches are on the following reviews:
- https://reviewboard.asterisk.org/r/4600/
- https://reviewboard.asterisk.org/r/4605/
-
- ASTERISK-24841 #close
- Reported by: Matt Jordan
-
- Review: https://reviewboard.asterisk.org/r/4609/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 16:03 +0000 [af458e2e60] Kevin Harwell <kharwell@digium.com>
-
- * chan_sip: make progressinband default to no
-
- After the "progressinband" value setting of "never" was updated to never send a
- 183 this separated its use from the "no" value. Since "never" was the default,
- but most users probably expect "no" this patch updates the default for the
- "progressinband" setting to "no."
-
- ASTERISK-24835 #close
- Reported by: Andrew Nagy
- Review: https://reviewboard.asterisk.org/r/4606/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 12:53 +0000 [88b0fa7755] yaron nahum (License 6676)
-
- * res_pjsip: Add an 'auto' option for DTMF Mode
-
- This patch adds support for automatically detecting the type of DTMF that a
- PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
- the channel created for an endpoint will attempt to determine if RFC 4733
- DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
- for the channel will be set to inband.
-
- Review: https://reviewboard.asterisk.org/r/4438
-
- ASTERISK-24706 #close
- Reported by: yaron nahum
- patches:
- yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 11:59 +0000 [16afee4651] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip_config_wizard: Cleanup load unload
-
- While investigating other unload issues I realized that the load/unload process
- for the config wizard was pretty ugly so I've refactored it as follows...
-
- When the res_pjsip sorcery instance is created the config_wizard bumps it's own
- module reference to prevent it from unloading while the sorcery instance is
- still active. When res_pjsip unloads and it's sorcery instance is destroyed,
- the config wizard unrefs itself which then allows itself to unload cleanly.
- Since the config wizard now can't load after res_pjsip or unload before it
- (which should have been the correct behavior all along), I was able to remove
- the chunks of code in both load_module and unload_module that handled that case.
-
- Ran the testsuite tests to insure there were no functional changes and REF_DEBUG
- to insure that Asterisk was shutting down cleanly with no FRACKs or leaks.
-
- Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4610/
-
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 11:37 +0000 [125acc52fe] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_softmix.c,channel.c: Minor code simplification and cleanup.
-
- * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats()
- and made some debug messages more helpful.
-
- * Made some debug and warning messages more helpful in
- channel.c:set_format().
-
- Review: https://reviewboard.asterisk.org/r/4607/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 11:28 +0000 [a63f7ad04a] Richard Mudgett <rmudgett@digium.com>
-
- * translate.c: Only select audio codecs to determine the best translation choice.
-
- Given a source capability of h264 and ulaw, a destination capability of
- h264 and g722 then ast_translator_best_choice() would pick h264 as the
- best choice even though h264 is a video codec and Asterisk only supports
- translation of audio codecs. When the audio starts flowing, there are
- warnings about a codec mismatch when the channel tries to write a frame to
- the peer.
-
- * Made ast_translator_best_choice() only select audio codecs.
-
- * Restore a check in channel.c:set_format() lost after v1.8 to prevent
- trying to set a non-audio codec.
-
- This is an intermediate patch for a series of patches aimed at improving
- translation path choices for ASTERISK-24841.
-
- This patch is a complete enough fix for ASTERISK-21777 as the v11 version
- of ast_translator_best_choice() does the same thing. However, chan_sip.c
- still somehow tries to call ast_codec_choose() which then calls
- ast_best_codec() with a capability set that doesn't contain any audio
- formats for the incoming call. The remaining warning message seems to be
- a benign transient.
-
- ASTERISK-21777 #close
- Reported by: Nick Ruggles
-
- ASTERISK-24380 #close
- Reported by: Matt Jordan
-
- Review: https://reviewboard.asterisk.org/r/4605/
- ........
-
- Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 09:55 +0000 [c9791dba1f] Matthew Jordan <mjordan@digium.com>
-
- * res/ari: Fix model validation for ChannelHold event
-
- When the ChannelHold event was added, the 'musicclass' parameter was
- erroneously removed. This caused the ChannelHold events to be rejected as
- they failed model validation. This patch updates the Swagger schema such that
- it now properly reflects the event that is being created.
-
- Hooray for tests that catch things like this.
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-10 07:39 +0000 [c39faa4729] Y Ateya (License 6693)
-
- * channels/chan_iax2: Improve POKE expiration time calculation for lossy networks
-
- POKE is used to check for peer availability; however, in networks with packet
- loss, the current calculations may result in POKE expiration times that are too
- short. This patch alters the expiration/retry time logic to take into account
- the last known qualify round trip time, as opposed to always using a static
- value for each peer.
-
- Review: https://reviewboard.asterisk.org/r/4536
-
- ASTERISK-22352 #close
- Reported by: Frederic Van Espen
-
- ASTERISK-24894 #close
- Reported by: Y Ateya
- patches:
- poke_noanswer_duration.diff submitted by Y Ateya (License 6693)
- ........
-
- Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 17:35 +0000 [75c2c85962] George Joseph <george.joseph@fairview5.com>
-
- * res_pjsip_phoneprov_provider: Fix reference leak on unload
-
- res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to
- a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the
- OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator.
- This plugged the leak but exposed an unload order issue between
- res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip.
-
- res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip.
- Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it
- unloads, it's objects are still in the sorcery instance. When res_pjsip
- unloads, it destroys all its objects including res_pjsip_phoneprov_provider's.
- The phoneprov destructor then attempts to unregister the extension from
- res_phoneprov but because res_phoneprov is already cleaned up, its users
- container is gone and we get a FRACK.
-
- Simple solution, check for the NULL users container before attempting to remove
- the entry. Duh.
-
- Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in
- res_pjsip_phoneprov_provider and no FRACKs.
-
- Reported-by: Corey Farrell
- Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4608/
- ASTERISK-24935 #close
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 17:31 +0000 [73c286a393] George Joseph <george.joseph@fairview5.com>
-
- * loader/main: Don't set ast_fully_booted until deferred reloads are processed
-
- Until we have a true module management facility it's sometimes necessary for one
- module to force a reload on another before its own load is complete. If
- Asterisk isn't fully booted yet, these reloads are deferred. The problem is
- that asterisk reports fully booted before processing the deferred reloads which
- means Asterisk really isn't quite ready when it says it is.
-
- This patch moves the report of fully booted after the processing of the deferred
- reloads is complete.
-
- Since the pjsip stack has the most number of related modules, I ran the
- channels/pjsip testsuite to make sure there aren't any issues. All tests
- passed.
-
- Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4604/
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 17:03 +0000 [5737650a67] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip: add CLI command to show global and system configuration
-
- Added a new CLI command for res_pjsip that shows both global and system
- configuration settings: pjsip show settings
-
- ASTERISK-24918 #close
- Reported by: Scott Griepentrog
- Review: https://reviewboard.asterisk.org/r/4597/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 11:07 +0000 [1695a5b85f] Richard Mudgett <rmudgett@digium.com>
-
- * chan_iax2.c: Fix ref leak in iax2_request().
-
- * Increased warning message format capability string buffer size in
- iax2_request().
-
- Review: https://reviewboard.asterisk.org/r/4601/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 10:54 +0000 [92c1688edb] Richard Mudgett <rmudgett@digium.com>
-
- * bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible().
-
- Review: https://reviewboard.asterisk.org/r/4601/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 10:42 +0000 [2679d0100a] yaron nahum (License 6676)
-
- * res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests
-
- This patch adds a new session supplement that handles in-dialog OPTIONS
- requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup
- for the OPTIONS request would already have been done by the time the
- session supplement receives the inbound request.
-
- ASTERISK-24862 #close
- Reported by: yaron nahum
- patches:
- res_pjsip_dlg_options.c submitted by yaron nahum (License 6676)
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-09 07:56 +0000 [6ba6e3dffd] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix autological comparisons
-
- This fixes autological comparison warnings in the following:
- * chan_skinny: letohl may return a signed or unsigned value, depending on the
- macro chosen
- * func_curl: Provide a specific cast to CURLoption to prevent mismatch
- * cel: Fix enum comparisons where the enum can never be negative
- * enum: Fix comparison of return result of dn_expand, which returns a signed
- int value
- * event: Fix enum comparisons where the enum can never be negative
- * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
- negative
- * presencestate: Use the actual enum value for INVALID state
- * security_events: Fix enum comparisons where the enum can never be negative
- * udptl: Don't bother to check if the return value from encode_length is less
- than 0, as it returns an unsigned int
- * translate: Since the parameters are unsigned int, don't bother checking
- to see if they are negative. The cast to unsigned int would already blow
- past the matrix bounds.
- * res_pjsip_exten_state: Use a temporary value to cache the return of
- ast_hint_presence_state
- * res_stasis_playback: Fix enum comparisons where the enum can never be
- negative
- * res_stasis_recording: Add an enum value for the case where the recording
- operation is in error; fix enum comparisons
- * resource_bridges: Use enum value as opposed to -1
- * resource_channels: Use enum value as opposed to -1
-
- Review: https://reviewboard.asterisk.org/r/4533
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4533.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 21:05 +0000 [e05c8ae68e] Stefan Engström (License 6691)
-
- * apps/app_queue: Prevent possible crash when evaluating queue penalty rules
-
- Although it only occurred once, a crash occurred when a queue attempted to
- evaluate a queue penalty rule that appeared to have already been destroyed.
- In many locations in app_queue, a test is done to see if qe->pr is NULL;
- however, when we dispose of a queue's penalty rules, we don't set the pointer
- to NULL after free'ing it. This patch does that to prevent any dangling
- pointers from lingering on the queue object.
-
- Review: https://reviewboard.asterisk.org/r/4522
-
- ASTERISK-23319 #close
- Reported by: Vadim
- patches:
- rb4552.patch submitted by Stefan Engström (License 6691)
- ........
-
- Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 13:15 +0000 [f21b45db49] Jonathan Rose <jrose@digium.com>
-
- * res_pjsip_t38: Fix FAX failures when using PJSIP with authentication
-
- Without this patch, if a PJSIP endpoint with udptl enabled and authentication
- set attempted to use sendFax, the FAX session would fail during setup. This
- was because the invite issued in response to being auth challenged would cause
- the PJSIP channel performing the FAX to receive a second T38 framehook and
- this would cause frames to be consumed in an inappropriate manner.
-
- ASTERISK-24933 #close
- Reported by: Jonathan Rose
- Review: https://reviewboard.asterisk.org/r/4577/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 13:14 +0000 [4441bb6a25] Richard Mudgett <rmudgett@digium.com>
-
- * Bridging: Eliminate the unnecessary make channel compatible with bridge operation.
-
- When a channel enters the bridging system it is first made compatible with
- the bridge and then the bridge technology makes the channel compatible
- with the technology. For all but the DAHDI native and softmix bridge
- technologies the make channel compatible with the bridge step is an
- effective noop because the other technologies allow all audio formats.
- For the DAHDI native bridge technology it doesn't matter because it is not
- an initial bridge technology and chan_dahdi allows only one native format
- per channel. For the softmix bridge technology, it is a noop at best and
- harmful at worst because the wrong translation path could be setup if the
- channel's native formats allow more than one audio format.
-
- This is an intermediate patch for a series of patches aimed at improving
- translation path choices.
-
- * Removed code dealing with the unnecessary step of making the channel
- compatible with the bridge.
-
- ASTERISK-24841
- Reported by: Matt Jordan
-
- Review: https://reviewboard.asterisk.org/r/4600/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 11:40 +0000 [f767440906] mhej (license 6085)
-
- * Security/tcptls: MitM Attack potential from certificate with NULL byte in CN.
-
- When registering to a SIP server with TLS, Asterisk will accept CA signed
- certificates with a common name that was signed for a domain other than the
- one requested if it contains a null character in the common name portion of
- the cert. This patch fixes that by checking that the common name length
- matches the the length of the content we actually read from the common name
- segment. Some certificate authorities automatically sign CA requests when
- the requesting CN isn't already taken, so an attacker could potentially
- register a CN with something like www.google.com\x00www.secretlyevil.net
- and have their certificate signed and Asterisk would accept that certificate
- as though it had been for www.google.com - this is a security fix and is
- noted in AST-2015-003.
-
- ASTERISK-24847 #close
- Reported by: Maciej Szmigiero
- Patches:
- asterisk-null-in-cn.patch submitted by mhej (license 6085)
- ........
-
- Merged revisions 434337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........
-
- Merged revisions 434338 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 11:23 +0000 [1712d16825] Richard Mudgett <rmudgett@digium.com>
-
- * format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear().
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 07:33 +0000 [ae39dd1f46] Matthew Jordan <mjordan@digium.com>
-
- * chan_iax2: Fix compilation issue due to funky merge
-
- Don't mix declarations and code
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 07:00 +0000 [05397ad01e] Jaco Kroon (License 5671)
-
- * chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno]
-
- This patch fixes an access to the peer callnumber that is unprotected by a
- corresponding mutex. The peer->callno value can be changed by multiple threads,
- and all data inside the iaxs array must be procted by a corresponding lock
- of iaxsl.
-
- The patch moves the unprotected access to a location where the mutex is
- safely obtained.
-
- Review: https://reviewboard.asterisk.org/r/4599/
-
- ASTERISK-21211 #close
- Reported by: Jaco Kroon
- patches:
- asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671)
- ........
-
- Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 06:53 +0000 [be13c72142] Valentin Vidić (License 6697)
-
- * chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled
-
- When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will
- attempt to handle both IPv4 and IPv6 addresses, although the information will
- be stored in a struct with an AF_INET6 address type. However, the current
- NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly.
- This patch adds an additional check for the mapped address case, allowing
- the NAT code to handle clients even when the address is IPv6.
-
- Review: https://reviewboard.asterisk.org/r/4563/
-
- ASTERISK-18032 #close
- Reported by: Christoph Timm
- patches:
- nat_with_ipv6.diff submitted by Valentin Vidić (License 6697)
- ........
-
- Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-08 06:44 +0000 [f324870dab] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix pointer-bool-converesion warnings
-
- This patch fixes several warnings pointed out by the clang compiler.
- * chan_pjsip: Removed check for data->text, as it will always be non-NULL.
- * app_minivm: Fixed evaluation of etemplate->locale, which will always
- evaluate to 'true'. This patch changes the evaluation to use
- ast_strlen_zero.
- * app_queue:
- - Fixed evaluation of qe->parent->monfmt, which always evaluates to
- true. Instead, we just check to see if the dereferenced pointer
- evaluates to true.
- - Fixed evaluation of mem->state_interface, wrapping it with a call to
- ast_strlen_zero.
- * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
-
- Review: https://reviewboard.asterisk.org/r/4541
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4541.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-07 14:38 +0000 [a6aed7f6f6] Scott Griepentrog <sgriepentrog@digium.com>
-
- * Revert accidental change in r434261
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-07 14:35 +0000 [0584e29300] Scott Griepentrog <sgriepentrog@digium.com>
-
- * pjsip: resolve compatibility problem with ast_sip_session
-
- A change in r430179 inserted a variable near the top of a
- structure caused a problem when running DPMA in a version
- of Asterisk compiled across the change. This patch moves
- the new variable to the end of the structure, eliminating
- the problem.
-
- Review: https://reviewboard.asterisk.org/r/4574/
- ........
-
- Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-07 11:40 +0000 [d754f70239] Kevin Harwell <kharwell@digium.com>
-
- * bridge.c: Hangup attended transfer target after it has been swapped out
-
- After completing an attended transfer the transfer target channel (the one that
- gets swapped out) was not being hung up after leaving the bridge. This resulted
- in a channel possibly being left around. Added an explicit softhangup for the
- channel in question after the transfer is successfully completed in order to
- make sure the channel is hung up.
-
- ASTERISK-24782 #close
- Reported by: John Bigelow
- Review: https://reviewboard.asterisk.org/r/4575/
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-07 10:33 +0000 [c516981dc7] Mark Michelson <mmichelson@digium.com>
-
- * Do not queue message requests that we do not respond to.
-
- If we receive a MESSAGE request that we cannot send a response
- to, we should not send the incoming MESSAGE to the dialplan.
-
- This commit should help the bouncing message_retrans test to
- pass consistently.
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-07 10:21 +0000 [ab803ec342] Matthew Jordan <mjordan@digium.com>
-
- * ARI: Add the ability to intercept hold and raise an event
-
- For some applications - such as SLA - a phone pressing hold should not behave
- in the fashion that the Asterisk core would like it to. Instead, the hold
- action has some application specific behaviour associated with it - such as
- disconnecting the channel that initiated the hold; only playing MoH to channels
- in the bridge if the channels are of a particular type, etc.
-
- One way of accomplishing this is to use a framehook to intercept the
- hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
- accomplishes that using a new dialplan function, HOLD_INTERCEPT.
-
- In addition, some general cleanup of raising hold/unhold Stasis messages was
- done, including removing some RAII_VAR usage.
-
- Review: https://reviewboard.asterisk.org/r/4549/
-
- ASTERISK-24922 #close
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 21:09 +0000 [488f093e97] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix sometimes-initialized warning in func_math
-
- This patch fixes a bug in a unit test in func_math where a variable could be
- passed to ast_free that wasn't allocated. This patch corrects the issue and
- ensures that we only attempt to free a variable if we previously allocated
- it.
-
- Review: https://reviewboard.asterisk.org/r/4552
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4552.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 21:03 +0000 [c027133f6d] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix non-literal-null-conversion warnings
-
- Clang will flag errors when a char pointer is set to '\0', as opposed to a
- value that the char pointer points to. This patch fixes this warning
- in a variety of locations.
-
- Review: https://reviewboard.asterisk.org/r/4551
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4551.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 14:23 +0000 [2270c40d33] Kevin Harwell <kharwell@digium.com>
-
- * res_pjsip: config option 'timers' can't be set to 'no'
-
- When setting the configuration option 'timers' equal to 'no' the bit flag was
- not properly negated. This patch clears all associated flags and only sets the
- specified one. pjsip will handle any necessary flag combinations. Also went
- ahead and did similar for the '100rel' option.
-
- ASTERISK-24910 #close
- Reported by: Ray Crumrine
- Review: https://reviewboard.asterisk.org/r/4582/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 14:02 +0000 [95de71f247] George Joseph <george.joseph@fairview5.com>
-
- * build: Fixes for gcc 5 compilation
-
- These are fixes for compilation under gcc 5.0...
-
- chan_sip.c: In parse_request needed to make 'lim' unsigned.
- inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99
- inline semantics (same as clang).
- ccss.c: In ast_cc_set_parm, needed to fix weird comparison.
- dsp.c: Needed to work around a possible compiler bug. It was throwing
- an array-bounds error but neither
- sgriepentrog, rmudgett nor I could figure out why.
- manager.c: In action_atxfer, needed to correct an array allocation.
-
- This patch will go to 11, 13, trunk.
-
- Review: https://reviewboard.asterisk.org/r/4581/
- Reported-by: Jeffrey Ollie
- Tested-by: George Joseph
- ASTERISK-24932 #close
- ........
-
- Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 13:18 +0000 [d54ccda3b1] dkdegroot (License 6600)
-
- * clang compiler warnings: Remove large chunks of unused code from extconf
-
- This patch fixes a warning caught by clang, in which it detected that large
- chunks of extconf were unused. Frankly, I wish we could pretend that all of
- extconf was unused, but alas, that is not yet the case.
-
- A few extraneous functions in the parking tests were removed as well, for
- the same reason.
-
- Review: https://reviewboard.asterisk.org/r/4553
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4553.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 13:03 +0000 [0ecd472e4f] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config
-
- This patch fixes a warning caught by clang, in which a char pointer could be
- assigned to before it was initialized. The patch re-organizes the code to
- ensure that the pointer is always initialized, even on off nominal paths.
-
- Review: https://reviewboard.asterisk.org/r/4529
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4529.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434090 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 12:52 +0000 [4e7be5b2dc] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix format specified in framehook
-
- This patch fixes an invalid format specifier used in the formatting of an
- ERROR message in the framehook code. The format specifier specifies a
- type of 'unsigned short', but the argument passed to it is of type 'int'.
- The patch changes the format specifier to 'i'.
-
- Review: https://reviewboard.asterisk.org/r/4540
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4535.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 434087 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 11:02 +0000 [2443b40341] Mark Michelson <mmichelson@digium.com>
-
- * Ensure that a non-zero sample rate is returned for all formats.
-
- Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate
- if one was not provided by a format. In Asterisk 13, this was removed.
- The result was that some calculations which involve dividing by the
- sample rate resulted in dividing by 0. The fix being put in place
- here is to have the same default fallback that was present in previous
- versions of Asterisk.
-
- Asterisk-24914 #close
- Reported by Marcello Ceschia
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 10:16 +0000 [b1102cd642] Corey Farrell <git@cfware.com>
-
- * res_pjsip_phoneprov_provider: Revert 433996 / 433997.
-
- res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then
- ignoring the return. OBJ_NODATA flag was to prevent a reference leak, but
- this caused the module to FRACK on unload. Revert change until this can
- be investigated further.
-
- ASTERISK-24935
- Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4578/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-06 09:50 +0000 [0f25076f67] Mark Michelson (license #5049)
-
- * ParkedCall: Don't allow dialplan fallthrough after retrieving parked call.
-
- This is a change to align behavior with that of Asterisk 11 and previous versions.
- In those versions, if a parked call were retrieved, and the call ended, the parked
- call retriever would be hung up after the ParkedCall application ran. Prior to this
- patch, in Asterisk 13, the same situation would result in the parked call retriever
- falling through to additional priorities in the extension where the ParkedCall
- application was called. With this patch, the behavior between Asterisk 11 and 13
- aligns.
-
- ASTERISK-24899 #close
- Reported by Malcolm Davenport
- Patches:
- ASTERISK-24899.patch uploaded by Mark Michelson(license #5049)
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-05 07:53 +0000 [709fa14b44] Corey Farrell <git@cfware.com>
-
- * res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
-
- res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then
- ignoring the return. Added OBJ_NODATA flag to prevent a reference leak.
-
- ASTERISK-24935 #close
- Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4578/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-03 16:53 +0000 [1ee8424f27] Mark Michelson <mmichelson@digium.com>
-
- * res_pjsip_messaging: Serialize outbound SIP MESSAGEs
-
- Outbound SIP MESSAGEs had the potential to be sent out
- of order from how they were specified in a set of
- dialplan steps.
-
- This change creates a serializer for sending outbound
- MESSAGE requests on. This ensures that the MESSAGEs are
- sent by Asterisk in the same order that they were sent
- from the dialplan.
-
- ASTERISK-24937 #close
- Reported by Mark Michelson
-
- Review: https://reviewboard.asterisk.org/r/4579
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-02 09:56 +0000 [169e57d2e0] Scott Griepentrog <sgriepentrog@digium.com>
-
- * pjsip: resolve compatibility problem with ast_sip_session
-
- A change in r430179 inserted a variable near the top of a
- structure caused a problem when running DPMA in a version
- of Asterisk compiled across the change. This patch moves
- the new variable to the end of the structure, eliminating
- the problem.
-
- Review: https://reviewboard.asterisk.org/r/4574/
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-02 05:31 +0000 [1eb0c5f4e8] Corey Farrell <git@cfware.com>
-
- * Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS.
-
- DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but
- for this to work correctly MALLOC_DEBUG must also be marked
- as conflicting with DEBUG_CHAOS.
-
- Review: https://reviewboard.asterisk.org/r/4557/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-04-01 11:25 +0000 [e301185983] Ashley Sanders <asanders@digium.com>
-
- * stasis: set a channel variable on websocket disconnect error
-
- Resolve compile errors caused by r433863 by fixing the
- documentation xml to comply with the schema.
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-31 22:26 +0000 [a1f12d9231] Ashley Sanders <asanders@digium.com>
-
- * stasis: set a channel variable on websocket disconnect error
-
- Resolve compile errors caused by r433839 by included the missing
- header file, pbx.h.
-
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-31 17:00 +0000 [7293ecd90b] Ashley Sanders <asanders@digium.com>
-
- * stasis: set a channel variable on websocket disconnect error
-
- When an error occurs while writing to a web socket, the web socket is
- disconnected and the event is logged. A side-effect of this, however, is that
- any application on the other side waiting for a response from Stasis is left
- hanging indefinitely (as there is no mechanism presently available for
- notifying interested parties about web socket error states in Stasis).
-
- To remedy this scenario, this patch introduces a new channel variable:
- STASISSTATUS.
-
- The possible values for STASISSTATUS are:
- SUCCESS - The channel has exited Stasis without any failures
- FAILED - Something caused Stasis to croak. Some (not all) possible
- reasons for this:
- - The app registry is not instantiated;
- - The app requested is not registered;
- - The app requested is not active;
- - Stasis couldn't send a start message
-
- ASTERISK-24802
- Reported By: Kevin Harwell
- Review: https://reviewboard.asterisk.org/r/4519/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-31 11:55 +0000 [94949e7f2f] Richard Mudgett <rmudgett@digium.com>
-
- * chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos.
-
- Fix misplaced parentheses in original fabs() expression.
- ........
-
- Merged revisions 433816 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-31 06:47 +0000 [9967739669] Corey Farrell <git@cfware.com>
-
- * Re-add _ast_mem_backtrace_buffer variable for ABI compatibility.
-
- Modules built prior to commit of r4502 expect to link at runtime
- to the variable _ast_mem_backtrace_buffer. This change re-adds
- the variable to the C file only.
-
- Review: https://reviewboard.asterisk.org/r/4558/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-30 06:42 +0000 [2d39bc5528] Corey Farrell <git@cfware.com>
-
- * Fix an ABI compatibility issue with ast_log_safe for modules.
-
- Binary modules are sometimes built against the latest release of
- Asterisk in each branch, and need to be compatible with all
- releases of that branch. This change ensures that utils.h only
- uses ast_log_safe from the core. For modules and utilities ast_log
- is used instead.
-
- Review: https://reviewboard.asterisk.org/r/4548/
- ........
-
- Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-29 21:44 +0000 [5f8faf16af] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Wabsolute-value warnings
-
- This patch fixes several warnings caught by clang - in this case, usage of the
- abs function on non-integer values. This patch uses labs and fabs, as
- appropriate, in the various affected files.
-
- Review: https://reviewboard.asterisk.org/r/4525
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4525.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-29 21:39 +0000 [09b681e344] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix invalid enum conversion
-
- This patch fixes some invalid enum conversion warnings caught by clang. In
- particular:
- * chan_sip: Several functions mixed usage of the st_refresher_param
- enum and st_refresher enum. This patch corrects the functions to use the
- right enum.
- * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
- * strings: Fixed incorrect usage of AO2 flags with strings container.
- * res_stasis: Change a return enumeration to stasis_app_user_event_res.
-
- Review: https://reviewboard.asterisk.org/r/4535
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4535.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-29 21:29 +0000 [7f33abb827] Matthew Jordan <mjordan@digium.com>
-
- * main/stdtime/localtime: Fix warning introduced in r433720
-
- The patch in r433720 caused a warning to be kicked back by gcc. It occurred
- due to this check in unistd.h:
-
- if (__nbytes > __bos0 (__buf))
- return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf));
-
- That is, if __nbytes is greater than the result of GCC's built-in object size
- for the struct, we'll kick back a warning.
-
- As it turns out, this is because there is an error in the code in the patch.
- We are passing the address of the pointer to the struct, not iev, which is a
- pointer to the struct. Hence, the number of bytes is probably going to be lot
- larger than the number of bytes that make up a pointer! This patch changes
- the code just read from the pointer to the struct - which fixes the warning.
-
- ASTERISK-24917
- ........
-
- Merged revisions 433743 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-29 20:56 +0000 [47eeb67e14] dkdegroot (License 6600)
-
- * clang compiler warnings: Ignore -Wunused-command-line-argument
-
- Asterisk's build system has a tendency to pass include directives for libraries
- to everything compiled within a particular group of source files. This means
- we pass the header for libxml2 to things that don't necessarily need it. As a
- result, we ignore this particular warning.
-
- Review: https://reviewboard.asterisk.org/r/4545/
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4545.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433720 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-29 20:52 +0000 [dbb4d6f9e7] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end
-
- This patch fixes a warning caught by clang, wherein a variable sized struct is
- not located at the end of a struct. While the code in question actually
- expected this, this is a good warning to watch for. Hence, this patch refactors
- the code in question to not have two variable length elements in the same
- struct.
-
- Review: https://reviewboard.asterisk.org/r/4530/
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4530.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433717 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:56 +0000 [e126ab9eeb] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix a variety of "unused" warnings
-
- This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable
- errors caught by clang. Specifically:
-
- * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[],
- qsmp_cmd_usage[]
- * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom"
- * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel"
- * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$"
- * funcs/func_env.c:729: Fixed ast_str_append_substr.
- * main/editline/np/strlcat.c: removed unused rcsid variable
- * main/editline/np/strlcpy.c: removed unused rcsid variable
- * main/security_events.c: removed unused TIMESTAMP_STR_LEN
- * utils/conf2ael.c: removed unused cfextension_states
- * utils/extconf.c: removed unused cfextension_states
-
- Review: https://reviewboard.asterisk.org/r/4526
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4526.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:48 +0000 [2f6534527d] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Wself-assign
-
- Assigning a variable to itself isn't super useful. However, the WAV format
- modules make use of this in order to perform byte endian checks. This patch
- works around the warning by only performing the self assignment if we are
- going to do more than just assign it to ourselves. Which is odd, but true.
-
- Review: https://reviewboard.asterisk.org/r/4544/
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4544.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433690 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:40 +0000 [eb70993a50] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Wparantheses-equality warnings
-
- Clang will treat ((a == b)) as a warning, as it reasonably expects that the
- developer may have intended to write (a == b) or ((a = b)). This patch cleans
- up all instances where equality, not assignment, was intended between two
- parantheses.
-
- Review: https://reviewboard.asterisk.org/r/4531/
-
- ASTERISK-24917
- Repoted by: dkdegroot
- patches:
- rb4531.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:31 +0000 [c0ff16036a] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Wbitfield-constant-conversion warning
-
- In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by
- clang, as it will truncate the -1 to a 1 implicitly.
-
- Instead, we just assign the value a '1'.
-
- Review: https://reviewboard.asterisk.org/r/4537/
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4537.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433683 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:27 +0000 [844bc76bef] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Winitializer-overrides
-
- This patch fixes clange compiler warnings for initializer overrides.
- Specifically:
-
- res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration
- value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing
- those enum values, we therefore initialize the value twice to two different
- values, "tlsv1" and "default". This patch changes it to just initialize
- the index in the array to "tlsv1".
-
- Review: https://reviewboard.asterisk.org/r/4539/
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4539.patch submitted by dkdegroot (License 6600)
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-28 07:19 +0000 [5e204042d9] dkdegroot (License 6600)
-
- * clang compiler warnings: Fix -Wunused-function; make inline function static
-
- This patch fixes clang compilers warnings for unused functions. Specifically:
- * channels/chan_iax2: removed user_ref function
- * main/dsp.c: removed goertzel_update function
- * main/config.c: made variable_list_switch static
-
- Review: https://reviewboard.asterisk.org/r/4527
-
- ASTERISK-24917
- Reported by: dkdegroot
- patches:
- rb4527.patch submitted by dkdegroot (License 6600)
- ........
-
- Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 17:34 +0000 [cfbf5fbe91] Jonathan Rose <jrose@digium.com>
-
- * SAC: Add a few basic queues
-
- Review: https://reviewboard.asterisk.org/r/4503/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 17:25 +0000 [1a50d8d4c2] Jonathan Rose <jrose@digium.com>
-
- * SAC: Add conferencing extensions and configuration
-
- Review: https://reviewboard.asterisk.org/r/4504/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 16:15 +0000 [c6c08d755d] Rusty Newton <rnewton@digium.com>
-
- * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2
-
- Example configuration files for a "basic PBX" deployment for the fictitious
- Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
- and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
-
- Patch 4488 includes all functionality needed for SAC's outside connectivity
- and some externally accessed features, as well as outbound dialing.
-
- Reported by: Malcolm Davenport
- Tested by: Rusty Newton
-
- Review: https://reviewboard.asterisk.org/r/4488/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 16:04 +0000 [13557675d4] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage.
-
- * Converted the contact_autoexpire container to use the ao2 template hash
- and cmp functions. Also made use the OBJ_SEARCH_xxx names instead of the
- deprecated names.
-
- * Eliminates several unnecessary uses of RAII_VAR().
-
- Review: https://reviewboard.asterisk.org/r/4524/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 15:30 +0000 [85feac857c] Mark Michelson <mmichelson@digium.com>
-
- * Add stateful PJSIP response API call, and use it for out-of-dialog responses.
-
- Asterisk had an issue where retransmissions of MESSAGE requests resulted in
- Asterisk processing the retransmission as if it were a new MESSAGE request.
-
- This patch fixes the issue by creating a transaction in PJSIP on the incoming
- request. This way, if a retransmission arrives, the PJSIP transaction layer
- will resend the response and Asterisk will not ever see the retransmission.
-
- ASTERISK-24920 #close
- Reported by Mark Michelson
-
- Review: https://reviewboard.asterisk.org/r/4532/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 12:50 +0000 [dc2cf21144] Richard Mudgett <rmudgett@digium.com>
-
- * res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown.
-
- Contact expiration object refs were leaked when the module was unloaded.
-
- * Made empty the scheduler of entries before destroying it to release the
- object ref held by the scheduler entry.
-
- Review: https://reviewboard.asterisk.org/r/4523/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 09:41 +0000 [6e6f5b3a1f] scsiguy (License 6692)
-
- * res/res_timing_kqueue: Update the module to conform to current timer API
-
- This patch updates the kqueue timing module to conform to current timer API.
-
- This fixes issues with using the kqueue timing source on Asterisk 13 on
- FreeBSD 10. These issues include:
-
- - Remove support for kevent64(). The values used to support Asterisk timers
- fit within 32bits and so can be handled on all platforms via kevent().
-
- - Provide debug logging for, but do not track, unacked events. This matches
- the behavior of all other timer implementations.
-
- - Implement continuous mode by triggering and leaving active, a user event.
- This ensures that the file descriptor for the timer returns immediately from
- poll(), without placing the load of a high speed timer on the kernel.
-
- - In kqueue_timer_get_max_rate(), don't overstate the capability of the timer.
- On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer
- type kqueue supports for timers.
-
- - In kqueue_timer_get_event(), assume the caller woke up from poll() and just
- return the mode the timer is currently in. This matches all other timer
- implementations.
-
- - Adjust the test code now that unacked events are not tracked.
-
- Review: https://reviewboard.asterisk.org/r/4465/
-
- ASTERISK-24857 #close
- Reported by: scsiguy
- Tested by: Ed Hynan
- patches:
- rb4465.patch submitted by scsiguy (License 6692)
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 07:26 +0000 [b0df413fb2] Corey Farrell <git@cfware.com>
-
- * Fix link error for utils/aelparse.
-
- Use the standard ast_log instead of ast_log_safe for STANDALONE programs.
-
- Review: https://reviewboard.asterisk.org/r/4538/
- ........
-
- Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-27 02:09 +0000 [d01706ce1e] Corey Farrell <git@cfware.com>
-
- * Improved and portable ast_log recursion avoidance
-
- This introduces a new logger routine ast_log_safe. This routine should be
- used for all error messages in code that can be run as a result of ast_log.
- ast_log_safe does nothing if run recursively. All error logging in
- astobj2.c, strings.c and utils.h have been switched to ast_log_safe.
-
- This required adding support for raw threadstorage. This provides direct
- access to the void* pointer in threadstorage. In ast_log_safe, NULL is used
- to signify that this thread is not already running ast_log_safe, (void*)1 when
- it is already running. This was done since it's critical that ast_log_safe
- do nothing that could log during recursion checking.
-
- ASTERISK-24155 #close
- Reported by: Timo Teräs
- Review: https://reviewboard.asterisk.org/r/4502/
- ........
-
- Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-26 18:07 +0000 [4b225e2104] Corey Farrell <git@cfware.com>
-
- * Fix compile errors caused by r4500 / r4501.
-
- * Add ast_register_cleanup to utils/clicompat.c to deal with
- any utils that copy sources from main.
- * Asterisk 13+: remove unused variables from core_local.c.
-
- Review: https://reviewboard.asterisk.org/r/4534/
- ........
-
- Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-26 17:19 +0000 [6adf26f14d] Corey Farrell <git@cfware.com>
-
- * Replace most uses of ast_register_atexit with ast_register_cleanup.
-
- Since 'core stop now' and 'core restart now' do not stop modules,
- it is unsafe for most of the core to run cleanups. Originally all
- cleanups used ast_register_atexit, and were only changed when it
- was shown to be unsafe. ast_register_atexit is now used only when
- absolutely required to prevent corruption and close child processes.
-
- Exceptions that need to use ast_register_atexit:
- * CDR: Flush records.
- * res_musiconhold: Kill external applications.
- * AstDB: Close the DB.
- * canary_exit: Kill canary process.
-
- ASTERISK-24142 #close
- Reported by: David Brillert
-
- ASTERISK-24683 #close
- Reported by: Peter Katzmann
-
- ASTERISK-24805 #close
- Reported by: Badalian Vyacheslav
-
- ASTERISK-24881 #close
- Reported by: Corey Farrell
-
- Review: https://reviewboard.asterisk.org/r/4500/
- Review: https://reviewboard.asterisk.org/r/4501/
- ........
-
- Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-26 12:46 +0000 [d0df545a44] Corey Farrell <git@cfware.com>
-
- * res_pjsip: Enable unload of all modules at shutdown.
-
- * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes
- caused by running PJSIP functions from non-PJSIP threads.
- * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing
- crashes in some cases. In theory pj_shutdown() should take care of this.
- * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at
- shutdown.
- * Resolve leaked config global in res_pjsip_notify.
- * Unregister pubsub pjsip service module.
- * Implement cleanup for res_pjsip_session.
-
- ASTERISK-24731 #close
- Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4498/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-26 12:04 +0000 [fd434a210f] Kevin Harwell <kharwell@digium.com>
-
- * app_confbridge: file playback blocks dtmf
-
- Attempting to execute DTMF in a confbridge while file playback (prompt,
- announcement, etc) is occurring is not allowed. You have to wait until
- the sound file has completed before entering DTMF. This patch fixes it
- so that app_confbridge now monitors for dtmf key presses during menu
- driven file playback. If a key is pressed playback stops and it executes
- the matched menu option.
-
- ASTERISK-24864 #close
- Reported by: Steve Pitts
- Review: https://reviewboard.asterisk.org/r/4510/
- ........
-
- Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-25 13:37 +0000 [dea885a607] Richard Mudgett <rmudgett@digium.com>
-
- * A couple minor cleanup tweaks.
-
- * In res/res_sorcery_realtime.c: Broke long line.
-
- * In main/bucket.c: Eliminated unnecessary NULL check as
- ast_sorcery_unref() is NULL tolerant and set the global object to NULL
- after unref in the system shutdown bucket_cleanup().
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-25 10:30 +0000 [05de9082a5] Simon Arlott (License 5756)
-
- * res_xmpp: Buddies are always auto-registered when processing the roster
-
- Due to a quirk in the configuration handling of res_xmpp, the 'autoregister'
- setting was never actually processed. This was due to not properly copying
- over the global settings to the client settings when applying the
- configuration to the run-time object.
-
- Review: https://reviewboard.asterisk.org/r/4496/
-
- ASTERISK-14233
- ASTERISK-24780 #close
- Reported by: Simon Arlott
- patches:
- asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756)
- ........
-
- Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-24 14:26 +0000 [b1e9552b08] Richard Mudgett <rmudgett@digium.com>
-
- * chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.
-
- Incoming PJSIP call legs that have not been answered yet send unnecessary
- "180 Ringing" or "183 Progress" messages every time a connected line
- update happens. If the outgoing channel is also PJSIP then the incoming
- channel will always send a "180 Ringing" or "183 Progress" message when
- the outgoing channel sends the INVITE.
-
- Consequences of these unnecessary messages:
-
- * The caller can start hearing ringback before the far end even gets the
- call.
-
- * Many phones tend to grab the first connected line information and refuse
- to update the display if it changes. The first information is not likely
- to be correct if the call goes to an endpoint not under the control of the
- first Asterisk box.
-
- When connected line first went into Asterisk in v1.8, chan_sip received an
- undocumented option "rpid_immediate" that defaults to disabled. When
- enabled, the option immediately passes connected line update information
- to the caller in "180 Ringing" or "183 Progress" messages as described
- above.
-
- * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
- "183 Progress" messages. The default is "no" to disable sending the
- unnecessary messages.
-
- ASTERISK-24781 #close
- Reported by: Richard Mudgett
-
- Review: https://reviewboard.asterisk.org/r/4473/
-
-
- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-
-2015-03-23 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.3.0-rc1 Released.
-
-2015-03-22 23:58 +0000 [r433247-433269] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_queue.c, main/cli.c, main/cdr.c, main/manager.c,
- main/rtp_engine.c, /, funcs/func_cdr.c: Fix compilations errors
- on 64-bit OpenBSD systems In versiong 5.5, OpenBSD went to 64-bit
- time values. This requires a cast to (long) when printing members
- of certain time structs. Review:
- https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close
- Reported by: snuffy Tested by: snuffy patches:
- openbsd-time64.diff uploaded by snuffy (License 5024) ........
- Merged revisions 433268 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * main/asterisk.c, main/loader.c, main/xmldoc.c, /: Fix compilation
- issues for OpenBSD This patch addresses compilation issues for
- OpenBSD. Specifically, it addresses: * It allows including
- <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast
- in xmldoc.c In 13+, it also addresses a conditional inclusion in
- loader.c. Review: https://reviewboard.asterisk.org/r/4506
- ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy
- patches: misc-openbsd.diff uploaded by snuffy (License 5024)
- ........ Merged revisions 433245 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-20 19:52 +0000 [r433199-433222] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_messaging.c, res/res_pjsip/pjsip_options.c,
- res/res_pjsip.c, res/res_pjsip_nat.c: Audit
- ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind
- found some memory leaks associated with
- ast_pjsip_rdata_get_endpoint(). The leaks would manifest when
- sending responses to OPTIONS requests, processing MESSAGE
- requests, and res_pjsip supplements implementing the
- incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint()
- endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(),
- res/res_pjsip/pjsip_options.c:send_options_response(),
- res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and
- res/res_pjsip_messaging.c:send_response(). * Eliminated
- RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in
- res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but
- benign return value in
- res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review:
- https://reviewboard.asterisk.org/r/4511/
-
- * res/res_pjsip_sdp_rtp.c, main/sorcery.c, main/xmldoc.c:
- res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak
- respectively. Valgrind found a memory leak and invalid access. *
- Fix invalid access by sscanf() being fed a non-nul terminated
- string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix
- memory leak in main/sorcery.c:sorcery_object_field_destructor().
- * Fix potential NULL pointer dereference in
- main/xmldoc.c:xmldoc_get_syntax_config_option(). Review:
- https://reviewboard.asterisk.org/r/4513/
-
-2015-03-19 19:19 +0000 [r433174] Matthew Jordan <mjordan@digium.com>
-
- * funcs/func_env.c, tests/test_func_file.c, /: funcs/func_env: Fix
- regression caused in FILE read operation When r432935 was merged,
- it did correctly fix a situation where a FILE read operation on
- the middle of a file buffer would not read the requested length
- in the parameters passed to the FILE function. Unfortunately, it
- would also allow the FILE function to append more bytes than what
- was available in the buffer if the length exceeded the end of the
- buffer length. This patch takes the minimum of the remaining
- bytes in the buffer along with the calculated length to append
- provided by the original patch, and uses that as the length to
- append in the return result. This patch also updates the unit
- tests with the scenarios that were originally pointed out in
- ASTERISK-21765 that the original implementation treated
- incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-19 10:20 +0000 [r433113-433126] Corey Farrell <git@cfware.com>
-
- * main/logger.c, /: logger: Apply default console logging when
- configuration cannot be loaded. When logger.conf is missing or
- invalid enable console logging and display an error message.
- ASTERISK-24817 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4497/ ........ Merged
- revisions 433122 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * channels/sip/include/dialog.h, channels/chan_sip.c,
- channels/sip/include/sip.h: chan_sip: Simplify dialog/peer
- references, improve REF_DEBUG output. * Replace functions for
- ref/undef of dialogs and peers with macro's to call
- ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller
- information to sip_alloc and find_call. ASTERISK-24882 #close
- Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4189/
-
- * /, channels/chan_sip.c: chan_sip: Fix dialog reference leaked to
- scheduler for reinvite_timeout. Release the scheduler reference
- to the dialog for reinvite timeout during dialog_unlink_all.
- ASTERISK-24876 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4491/ ........ Merged
- revisions 433112 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-18 02:34 +0000 [r433088] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_session.c: res_pjsip_session: Fix off-nominal extra
- unref of session.
-
-2015-03-17 22:15 +0000 [r433060-433064] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/asterisk.c, main/config.c, main/xmldoc.c, main/manager.c,
- include/asterisk/config.h, main/utils.c, main/codec_builtin.c,
- main/endpoints.c: Various: bugfixes found via chaos Using
- DEBUG_CHAOS several instances of a null pointer crash, and one
- uninitialized variable were uncovered and fixed. Also added
- details on why Asterisk failed to initialize. Review:
- https://reviewboard.asterisk.org/r/4468/
-
- * build_tools/cflags.xml, include/asterisk/utils.h: core: Introduce
- chaos into memory allocations Locate potential crashes by
- exercising seldom used code paths. This patch introduces a new
- define DEBUG_CHAOS, and mechanism to randomly return an error
- condition from functions that will seldom do so. Functions that
- handle the allocation of memory get the first treatment. Review:
- https://reviewboard.asterisk.org/r/4463/
-
-2015-03-17 21:49 +0000 [r433057] Richard Mudgett <rmudgett@digium.com>
-
- * main/netsock2.c, /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c,
- apps/app_externalivr.c, res/res_pjsip_acl.c: Audit
- ast_sockaddr_resolve() usage for memory leaks. Valgrind found
- some memory leaks associated with ast_sockaddr_resolve(). Most of
- the leaks had already been fixed by earlier memory leak hunt
- patches. This patch performs an audit of ast_sockaddr_resolve()
- and found one more. * Fix ast_sockaddr_resolve() memory leak in
- apps/app_externalivr.c:app_exec(). * Made
- main/netsock2.c:ast_sockaddr_resolve() always set the addrs
- parameter for safety so the pointer will never be uninitialized
- on return. The same goes for
- res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that
- call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs
- variable use ast_free instead of ast_free_ptr to provide better
- MALLOC_DEBUG information. Review:
- https://reviewboard.asterisk.org/r/4509/ ........ Merged
- revisions 433056 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-17 18:34 +0000 [r433028-433031] Kevin Harwell <kharwell@digium.com>
-
- * include/asterisk/res_pjsip.h,
- res/res_pjsip_endpoint_identifier_anonymous.c,
- res/res_pjsip_endpoint_identifier_ip.c, res/res_pjsip.c,
- res/res_pjsip_endpoint_identifier_user.c: res_pjsip: Allow
- configuration of endpoint identifier query order Updated some
- documentation stating that endpoint identifiers registered
- without a name are place at the front of the lookup list. Also
- renamed register method
- 'ast_sip_register_endpoint_identifier_by_name' to
- 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840
- Reported by: Mark Michelson
-
- * configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
- res/res_pjsip_endpoint_identifier_user.c,
- include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- res/res_pjsip_endpoint_identifier_anonymous.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py:
- res_pjsip: Allow configuration of endpoint identifier query order
- This patch fixes previously reverted code that caused binary
- incompatibility problems with some modules. And like the original
- patch it makes sure that no matter what order the endpoint
- identifier modules were loaded, priority is given based on the
- ones specified in the new global 'endpoint_identifier_order'
- option. ASTERISK-24840 Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/4489/
-
-2015-03-17 16:10 +0000 [r433005] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip.c: res_pjsip: Add reason comment.
-
-2015-03-14 02:28 +0000 [r432971] Matthew Jordan <mjordan@digium.com>
-
- * /, main/format_cap.c: main/frame: Don't report empty disallow
- values as an error In realtime, it is normal to have a database
- with both 'allow' and 'disallow' columns in the schema. It is
- perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw'
- and no 'disallow' value. Unlike in static conf files, you can't
- *not* provide the disallow value. Thus, the empty disallow value
- causes a spurious WARNING message, which is kind of annoying.
- This patch makes it so that a 'disallow' value with no ... value
- ... is ignored. Granted, you can still screw this up as well, as
- technically specifying 'disallow=all,!ulaw' allows only ulaw, and
- then you would have no 'allow' value in your database. But
- really, why would you do that? WHY? ASTERISK-16779 #close
- Reported by: Atis Lezdins ........ Merged revisions 432970 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-14 02:00 +0000 [r432945-432949] Joshua Colp <jcolp@digium.com>
-
- * funcs/func_curl.c, /: func_curl: Don't hold exclusive lock when
- performing HTTP request. This code originally kept a lock held
- when performing the HTTP request to ensure that the options
- provided to curl remain valid. This doesn't seem to be necessary
- these days and holding the lock caused requests to happen
- sequentially instead of in parallel. ASTERISK-18708 #close
- Reported by: Dave Cabot ........ Merged revisions 432948 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * main/cli.c, /: core: Fix tab completion of "core set debug
- channel" CLI command. The "core set debug channel" CLI command
- mistakenly had source filenames added to its tab completion. This
- occurred because the CLI generator fell back to the "core set
- debug" command which permits setting debug at a source filename
- level. ASTERISK-21038 #close Reported by: Richard Kenner ........
- Merged revisions 432944 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-14 01:21 +0000 [r432920-432938] Matthew Jordan <mjordan@digium.com>
-
- * /, funcs/func_env.c: FILE: fix retrieval of file contents when
- offset is specified The loop that reads in a file was not
- correctly using the offset when determining what bytes to append
- to the output. This patch corrects the logic such that the
- correct portion of the file is extracted when an offset is
- specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt
- Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by
- Di-Shi Sun (License 5076) ........ Merged revisions 432935 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * apps/app_amd.c, /, configs/samples/amd.conf.sample: apps/app_amd:
- Document maximum_word_length option; fix AMDCAUSE documentation
- This patch corrects the documentation for the AMD application.
- Specifically: * It documents the maximum_word_length option,
- which limits the maximum allowed length of a single utterance. *
- It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH.
- MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was
- undocumented. Thanks to the issue reporter, Frank DiGennaro, for
- pointing out the issues. ASTERISK-19470 #close Reported by: Frank
- DiGennaro ........ Merged revisions 432918 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-13 17:04 +0000 [r432892-432894] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c: chan_pjsip: AMI action
- PJSIPShowEndpoint closes AMI connection on error. Also fixed
- similar problem with AMI action PJSIPShowEndpoints.
- ASTERISK-24872 #close Reported by: Dmitriy Serov Review:
- https://reviewboard.asterisk.org/r/4487/
-
- * channels/chan_pjsip.c, res/res_pjsip_caller_id.c:
- chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and
- consistent. The res_pjsip modules were manually checking both
- name and number presentation values when there is a function that
- determines the combined presentation for a party ID struct. The
- function takes into account if the name or number components are
- valid while the manual code rarely checked if the data was even
- valid. * Made use ast_party_id_presentation() rather than
- manually checking party ID presentation values. * Ensure that
- set_id_from_pai() and set_id_from_rpid() will not return
- presentation values other than what is pulled out of the SIP
- headers. It is best if the code doesn't assume that
- AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. *
- Fixed copy paste error in add_privacy_params() dealing with RPID
- privacy. * Pulled the id->number.valid test from
- add_privacy_header() and add_privacy_params() up into the parent
- function add_id_headers() to skip adding PAI/RPID headers
- earlier. * Made update_connected_line_information() not send out
- connected line updates if the connected line number is invalid.
- Lower level code would not add the party ID information and thus
- the sent message would be unnecessary. * Eliminated RAII_VAR
- usage in send_direct_media_request(). Review:
- https://reviewboard.asterisk.org/r/4472/
-
-2015-03-13 14:48 +0000 [r432868] Kevin Harwell <kharwell@digium.com>
-
- * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- res/res_pjsip_endpoint_identifier_anonymous.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py,
- configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
- res/res_pjsip_endpoint_identifier_user.c: Revert - res_pjsip:
- Allow configuration of endpoint identifier query order Due to a
- break in binary compatibility with some other modules these
- changes are being reverted until the issue can be resolved.
- ASTERISK-24840 Reported by: Mark Michelson
-
-2015-03-12 12:58 +0000 [r432808-432811] Matthew Jordan <mjordan@digium.com>
-
- * /, main/audiohook.c: main/audiohook: Update internal sample rate
- on reads When an audiohook is created (which is used by the
- various Spy applications and Snoop channel in Asterisk 13+), it
- initially is given a sample rate of 8kHz. It is expected,
- however, that this rate may change based on the media that passes
- through the audiohook. However, the read/write operations on the
- audiohook behave very differently. When a frame is written to the
- audiohook, the format of the frame is checked against the
- internal sample rate. If the rate of the format does not match
- the internal sample rate, the internal sample rate is updated and
- a new SLIN format is chosen based on that sample rate. This works
- just fine. When a frame is read, however, we do something quite
- different. If the format rate matches the internal sample rate,
- all is fine. However, if the rates don't match, the audiohook
- attempts to "fix up" the number of samples that were requested.
- This can result in some seriously large number of samples being
- requested from the read/write factories. Consider the worst case
- - 192kHz SLIN. If we attempt to read 20ms worth of audio produced
- at that rate, we'd request 3840 samples (192000 / (1000 / 20)).
- However, if the audiohook is still expecting an internal sample
- rate of 8000, we'll attempt to "fix up" the requested samples to:
- samples_converted = samples * (ast_format_get_sample_rate(format)
- / (float) audiohook->hook_internal_samp_rate); which is: 92160 =
- 3840 * (192000 / 8000) This results in us attempting to read
- 92160 samples from our factories, as opposed to the 3840 that we
- actually wanted. On a 64-bit machine, this miraculously survives
- - despite allocating up to two buffers of length 92160 on the
- stack. The 32-bit machines aren't quite so lucky. Even in the
- case where this works, we will either (a) get way more samples
- than we wanted; or (b) get about 3840 samples, assuming the
- timing is pretty good on the machine. Either way, the calculation
- being performed is wrong, based on the API users expectations. My
- first inclination was to allocate the buffers on the heap. As it
- is, however, there's at least two drawbacks with doing this: (1)
- It's a bit complicated, as the size of the buffers may change
- during the lifetime of the audiohook (ew). (2) The stack is
- faster (yay); the heap is slower (boo). Since our calculation is
- flat out wrong in the first place, this patch fixes this issue by
- instead updating the internal sample rate based on the format
- passed into the read operation. This causes us to read the
- correct number of samples, and has the added benefit of setting
- the audihook with the right SLIN format. Note that this issue was
- caught by the Asterisk Test Suite as a result of r432195 in the
- 13 branch. Because this issue is also theoretically possible in
- Asterisk 11, the change is being made here as well. Review:
- https://reviewboard.asterisk.org/r/4475/ ........ Merged
- revisions 432810 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * Makefile, include/asterisk/utils.h, /, configure, main/Makefile,
- configure.ac, include/asterisk/inline_api.h, makeopts.in: Add
- support for the clang compiler; update RAII_VAR to use
- BlocksRuntime RAII_VAR, which is used extensively in Asterisk to
- manage reference counted resources, uses a GCC extension to
- automatically invoke a cleanup function when a variable loses
- scope. While this functionality is incredibly useful and has
- prevented a large number of memory leaks, it also prevents
- Asterisk from being compiled with clang. This patch updates the
- RAII_VAR macro such that it can be compiled with clang. It makes
- use of the BlocksRuntime, which allows for a closure to be
- created that performs the actual cleanup. Note that this does not
- attempt to address the numerous warnings that the clang compiler
- catches in Asterisk. Much thanks for this patch goes to: * The
- folks on StackOverflow who asked this question and Leushenko for
- providing the answer that formed the basis of this code:
- http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
- * Diederik de Groot, who has been extremely patient in working on
- getting this patch into Asterisk. Review:
- https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133
- ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by:
- Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik
- de Groot (License 6600) ........ Merged revisions 432807 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-11 16:38 +0000 [r432764-432787] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip/config_domain_aliases.c,
- res/res_pjsip/include/res_pjsip_private.h,
- include/asterisk/res_pjsip.h: res_pjsip: Move internal
- init/destroy prototypes to private header file. Done as a
- separate commit from a finding in
- https://reviewboard.asterisk.org/r/4467/
-
- * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix pjsip.conf
- type=global object default value handling. When a type=global
- section is not defined in pjsip.conf the global defaults are not
- applied. As a result the mandatory Max-Forwards header is not
- added to SIP messages for res_pjsip/chan_pjsip. The handling of
- pjsip.conf type=global objects has several problems: 1) If the
- global object is missing the defaults are not applied. 2) If the
- global object is missing the default_outbound_endpoint's default
- value is not returned by
- ast_sip_global_default_outbound_endpoint(). 3) Defines are needed
- so default values only need to be changed in one place. * Added a
- sorcery instance observer callback to check if there were any
- type=global sections loaded. If there were more than one then
- issue an error message. If there were none then apply the global
- defaults. * Fixed ast_sip_global_default_outbound_endpoint() to
- return the documented default when no type=global object is
- defined. * Made defines for the global default values. *
- Increased the default_useragent[] size because SVN version
- strings can get lengthy and 128 characters may not be enough. *
- Fixed an off-nominal code path ref leak in global_alloc() if the
- string fields fail to initialize. * Eliminated RAII_VAR in
- get_global_cfg() and ast_sip_global_default_outbound_endpoint().
- ASTERISK-24807 #close Reported by: Anatoli Review:
- https://reviewboard.asterisk.org/r/4467/
-
- * res/res_pjsip/pjsip_global_headers.c: res_pjsip: Fixed invalid
- empty Server and User-Agent SIP headers. Setting pjsip.conf
- useragent to an empty string results in an empty SIP header being
- sent. * Made not add an empty SIP header item to the global SIP
- headers list. Review: https://reviewboard.asterisk.org/r/4467/
-
-2015-03-10 23:09 +0000 [r432742] Joshua Colp <jcolp@digium.com>
-
- * main/stasis_channels.c, main/endpoints.c, main/stasis_bridges.c:
- core: Don't create snapshots with locks. Snapshots are immutable
- and are never changed. Allocating them with a lock is wasteful.
- Review: https://reviewboard.asterisk.org/r/4469/
-
-2015-03-10 21:33 +0000 [r432693-432721] Matthew Jordan <mjordan@digium.com>
-
- * res/res_config_odbc.c, /: res/res_config_odbc: Fix improper
- escaping of backslashes with MySQL When escaping backslashes with
- MySQL, the proper way to escape the characters in a LIKE clause
- is to escape the '\' four times, i.e., '\\\\'. To quote the MySQL
- manual: "Because MySQL uses C escape syntax in strings (for
- example, “\n” to represent a newline character), you must double
- any “\” that you use in LIKE strings. For example, to search for
- “\n”, specify it as “\\n”. To search for “\”, specify it as
- “\\\\”; this is because the backslashes are stripped once by the
- parser and again when the pattern match is made, leaving a single
- backslash to be matched against." ASTERISK-24808 #close Reported
- by: Javier Acosta patches: res_config_odbc.diff uploaded by
- Javier Acosta (License 6690) ........ Merged revisions 432720
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
- * apps/app_voicemail.c, /: app_voicemail: Fix crash with IMAP
- backends when greetings aren't present When an IMAP backend is in
- use and greetings are set to be used, but aren't present for a
- user in their IMAP folder, Asterisk will crash. This occurs due
- to the mailstream being set to the 'greetings' folder and being
- left in that particular state, regardless of the success/failure
- of the attempt to access the folder the mailstream points to.
- Later access of the mailstream assumes that it points to the
- 'INBOX' (or some other folder), resulting in either a crash (if
- the greetings folder didn't exist and the mailstream is invalid)
- or an inability to read messages from the 'INBOX' folder. This
- patch restores the mailstream to its correct state after
- accessing the greetings. This fixes the crash, and sets the
- mailstream to the state that VoiceMailMain expects. Note that
- while ASTERISK-23390 also contained a patch for this issue, the
- patch on ASTERISK-24786 is the one being merged here. Review:
- https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close
- Reported by: Ben Smithurst ASTERISK-24786 #close Reported by:
- Graham Barnett Tested by: Graham Barnett patches:
- app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett
- (License 6685) ........ Merged revisions 432695 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, main/stdtime/localtime.c: localtime: Fix file descriptor leak
- on kqueue(2) systems The localtime management in the Asterisk
- core contains a thread that watches for changes in the local
- timezone. On systems where the directory containing
- /etc/localtime is modified frequently, the thread monitoring the
- changes will be woken up to determine if any changes in timezone
- have occurred. When using kqueue(2), this can cause a leak of
- file descriptors due to some improper management of resources.
- This patch updates the kqueue(2) handling in localtime, such that
- is no longer leaks resources. Review:
- https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close
- Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed
- Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License
- 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License
- 6680) ........ Merged revisions 432691 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-10 16:04 +0000 [r432668] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_refer.c, res/res_pjsip_session.c,
- include/asterisk/res_pjsip_session.h,
- res/res_pjsip_session.exports.in: res_pjsip_refer: Fix occasional
- unexpected BYE sent after receiving a REFER. A race condition
- happened between initiating a transfer and requesting that a
- dialog termination be delayed. Occasionally, the transferrer
- channels would exit the bridge and hangup before the dialog
- termination delay was requested. * Made request dialog
- termination delay before initiating the transfer action. If the
- transfer fails then cancel the delayed dialog termination
- request. ASTERISK-24755 #close Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/4460/
-
-2015-03-09 16:12 +0000 [r432638] Kevin Harwell <kharwell@digium.com>
-
- * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- res/res_pjsip_endpoint_identifier_anonymous.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py
- (added), configs/samples/pjsip.conf.sample, CHANGES,
- res/res_pjsip.c, res/res_pjsip_endpoint_identifier_user.c:
- res_pjsip: Allow configuration of endpoint identifier query order
- It's possible to have a scenario that will create a conflict
- between endpoint identifiers. For instance an incoming call could
- be identified by two different endpoint identifiers and the one
- chosen depended upon which identifier module loaded first. This
- of course causes problems when, for example, the incoming call is
- expected to be identified by username, but instead is identified
- by ip. This patch adds a new 'global' option to res_pjsip called
- 'endpoint_identifier_order'. It is a comma separated list of
- endpoint identifier names that specifies the order by which
- identifiers are processed and checked. ASTERISK-24840 #close
- Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/4455/
-
-2015-03-08 01:46 +0000 [r432614] Joshua Colp <jcolp@digium.com>
-
- * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix wrongful use of
- USE_PJPROJECT define. As pjproject is now used as a shared
- library a different define, HAVE_PJPROJECT, is used to specify if
- pjproject is present. ASTERISK-24830 #close Reported by: Stefan
- Engström
-
-2015-03-06 22:50 +0000 [r432574-432594] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_refer.c: res_pjsip_refer: Make safely get the
- context for a blind transfer. Made safely get the
- TRANSFER_CONTEXT channel value while the channel is locked in
- refer_incoming_attended_request() and
- refer_incoming_blind_request(). The pointer returned by
- pbx_builtin_getvar_helper() is only valid while the channel is
- locked.
-
- * res/res_pjsip_refer.c: res_pjsip_refer: Made
- refer_attended_alloc() not create the ao2 object with a lock. The
- lock is unused.
-
-2015-03-06 21:11 +0000 [r432556] Jonathan Rose <jrose@digium.com>
-
- * include/asterisk/app.h, main/app.c: app: Add functions to swap
- voicemail function table for testing purposes
-
-2015-03-06 20:18 +0000 [r432528-432534] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, channels/sig_analog.c, /,
- channels/chan_dahdi.h, channels/sig_analog.h, UPGRADE.txt:
- chan_dahdi/sig_analog: Fix distinctive ring detection to suck
- less. The distinctive ring feature interferes with detecting
- Caller ID and appears to have been broken for years. What happens
- is if you have a ring-ring cadence as used in the UK you get too
- many DAHDI events for the distinctive ring pattern array and
- Caller ID detection is aborted. I think when Zapata/DAHDI added
- the ring begin event it broke distinctive ring. More events
- happen than before and the code does no filtering of which event
- times are recorded in the pattern array. * Made distinctive ring
- only record the ringt count when the ring ends instead of on just
- any DAHDI event. Distinctive ring can be ring, ring-ring,
- ring-ring-ring, or different ring durations for the up to three
- rings. * Fixed the distinctive ring detection enable
- (chan_dahdi.conf option usedistinctiveringdetection) to be per
- port instead of somewhat per port and somewhat global. This has
- been broken since v1.8. * Fixed using the default distinctive
- ring context when the detected pattern does not match any
- configured dringX patterns. The default context did not get set
- when the previous call was a matched distinctive ring pattern and
- the current call is not matched. This has been broken since v1.8.
- * Made distinctive ring have no effect on Caller ID detection
- when it is disabled. Caller ID detection just monitors for 10
- seconds before giving up. * Fixed leak of struct callerid_state
- memory when a polarity reversal during Caller ID detection causes
- the incoming call to be aborted. DAHDI-1143 AST-1545
- ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588
- Reported by: Daniel Flounders Review:
- https://reviewboard.asterisk.org/r/4444/ ........ Merged
- revisions 432530 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, channels/chan_sip.c: chan_sip: Fix realtime locking inversion
- when poking a just built peer. When a realtime peer is built it
- can cause a locking inversion when the just built peer is poked.
- If the CLI command "sip show channels" is periodically executed
- then a deadlock can happen because of the locking inversion. *
- Push the peer poke off onto the scheduler thread to avoid the
- locking inversion of the just built realtime peer. AST-1540
- ASTERISK-24838 #close Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/4454/ ........ Merged
- revisions 432526 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-05 16:38 +0000 [r432485] George Joseph <george.joseph@fairview5.com>
-
- * apps/app_voicemail.c, /: app_voicemail: Fix compile breaking in
- app_voicemail with IMAP_STORAGE. There is a leftover "assert" in
- app_voicemail/__messagecount that references variables that don't
- exist. This causes the compile to fail when --enable-dev-mode and
- IMAP_STORAGE are selected. This patch removes the assert.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4461/ ........ Merged
- revisions 432484 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-03-04 18:52 +0000 [r432453] Matthew Jordan <mjordan@digium.com>
-
- * main/translate.c: translate: Prevent invalid memory accesses on
- fast shutdown When a 'core restart now' or 'core stop now' is
- executed and a channel is currently in a media operation, the
- translator matrix can be destroyed while a channel is currently
- blocked on getting the best translation choice (see
- ast_translator_best_choice). When the channel gets the mutex, the
- translation matrix now has invalid memory, and Asterisk crashes.
- This patch does two things: (1) We now only clean up the
- translation matrix on a graceful shutdown. In that case, there
- are no channels, and so there is no risk of this occurring. (2)
- We also now set the __matrix and __indextable to NULL. In some
- initial backtraces when this occurred, it looked as if there was
- a memory corruption occurring, and it wasn't until we determined
- that something had restarted Asterisk that the issue became
- clear. By setting these to NULL on shutdown, it becomes a bit
- easier to determine why a crash is occurring. Note that we could
- litter the code with NULL checks on the __matrix, but the act of
- making the translation matrix cleaned up on shutdown should
- preclude this issue from occurring in the first place, and this
- part of the code needs to be as fast as possible. Review:
- https://reviewboard.asterisk.org/r/4457/
-
-2015-03-02 19:14 +0000 [r432423] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert portion of
- r432195 Unfortunately, while initial testing with ConfBridge did
- not reproduce the audio problem alluded to in the comment in
- res_pjsip_sdp_rtp, further testing did show that bridge_softmix
- and/or ConfBridge has a severe problem bridging two or more
- participants at different sampling rates. Sometimes, it even
- picks odd sampling rates that cause hideous audio problems. This
- patch backs out the offending portion of the code until the
- issues in the affected bridging modules can be more properly
- analyzed. ASTERISK-24841
-
-2015-02-27 18:23 +0000 [r432404] Richard Mudgett <rmudgett@digium.com>
-
- * main/json.c, rest-api/api-docs/endpoints.json,
- res/ari/resource_endpoints.c, res/res_ari_endpoints.c,
- include/asterisk/json.h, res/ari/resource_channels.c: ARI: Fix
- crash if integer values used in JSON payload 'variables' object.
- Sending the following ARI commands caused Asterisk to crash if
- the JSON body 'variables' object passes values of types other
- than strings. POST /ari/channels POST /ari/channels/{channelid}
- PUT /ari/endpoints/sendMessage PUT
- /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated
- RAII_VAR usage in ast_ari_channels_originate_with_id(),
- ast_ari_channels_originate(), ast_ari_endpoints_send_message(),
- and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751
- #close Reported by: jeffrey putnam Review:
- https://reviewboard.asterisk.org/r/4447/
-
-2015-02-26 18:52 +0000 [r432385] Scott Griepentrog <sgriepentrog@digium.com>
-
- * include/asterisk/dial.h, main/dial.c: Dial API: add self destruct
- option when complete This patch adds a self-destruction option to
- the dial api. The usefulness of this is mostly when using async
- mode to spawn a separate thread used to handle the new call,
- while the calling thread is allowed to go on about other
- business. The only alternative to this option would be the
- calling thread spawning a new thread, or hanging around itself
- waiting to destroy the dial struct after completion. Example of
- use (minus error checking): struct ast_dial *dial =
- ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL);
- ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC,
- "Echo"); ast_dial_option_global_enable(dial,
- AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1);
- The dial_run call will return almost immediately after spawning
- the new thread to run and monitor the dial. If the call is
- answered, it is placed into the echo app. When completed, it will
- call ast_dial_destroy() on the dial structure. Note that any
- allocations made to pass values to ast_dial_set_user_data() or
- dial options must be free'd in a state callback function on any
- of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED,
- AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review:
- https://reviewboard.asterisk.org/r/4443/
-
-2015-02-26 17:07 +0000 [r432363] Kevin Harwell <kharwell@digium.com>
-
- * /, apps/app_chanspy.c, main/channel.c: app_chanspy, channel: fix
- frame leaks Fixed a couple of frame leaks that were found during
- testing. ASTERISK-24828 #close Reported by: John Hardin Review:
- https://reviewboard.asterisk.org/r/4445/ ........ Merged
- revisions 432362 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-26 04:58 +0000 [r432321-432342] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/Makefile, channels/Makefile: make: Remove 'res_features'
- from libraries to link against with cygwin/mingw32 Both the apps
- and channels Makefiles still listed 'res_features' as modules to
- link against when compiling for cygwin or mingw32. This module
- hasn't existed for quite some time. ASTERISK-18105 #close
- Reported by: feyfre ........ Merged revisions 432341 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, channels/chan_sip.c: channels/chan_sip: Don't send a BYE after
- final response when PBX thread fails When Asterisk fails to start
- a PBX thread for a new channel - for example, when the maxcalls
- setting in asterisk.conf is exceeded - we currently send a final
- response, and then attempt to send a BYE request to the UA. Since
- that's all sorts of wrong, this patch fixes that by setting
- sipalreadygone on the sip_pvt such that we don't get stuck
- sending BYE requests to something that does not want it. Note
- that this patch is a slight modification of the one on
- ASTERISK-15434. For clarity, it explicitly calls sipalreadygone
- with the calls to transmit a final response. ASTERISK-21845
- ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt
- Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei
- (License 5027) ........ Merged revisions 432320 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-25 23:48 +0000 [r432301] Rusty Newton <rnewton@digium.com>
-
- * configs/basic-pbx/README (added), configs/basic-pbx (added),
- configs/basic-pbx/extensions.conf (added),
- configs/basic-pbx/logger.conf (added),
- configs/basic-pbx/indications.conf (added),
- configs/basic-pbx/musiconhold.conf (added),
- configs/basic-pbx/asterisk.conf (added),
- configs/basic-pbx/pjsip.conf (added),
- configs/basic-pbx/modules.conf (added),
- configs/basic-pbx/voicemail.conf (added): configs/basic-pbx -
- Super Awesome Company example configs Phase 1, Patch 1 Example
- configuration files for a "basic PBX" deployment for the
- fictitious Super Awesome Company. Details at
- https://reviewboard.asterisk.org/r/4379/ and
- https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
- Reported by: Malcolm Davenport Tested by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/4379/
-
-2015-02-25 23:09 +0000 [r432258-432281] Matthew Jordan <mjordan@digium.com>
-
- * /, configure, configure.ac: configure: Promote SQLite3 "not
- installed" warning to error Since Asterisk won't build without
- the library, not having it is definitely an error. Thanks to Kyle
- Kurz for pointing this out. ........ Merged revisions 432280 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, channels/chan_sip.c: channels/chan_sip: Clarify WARNING
- message in mismatched SRTP scenario When we receive an SDP as
- part of an offer/answer for a peer/friend has been configured to
- require encryption, and that SDP offer/answer failed to provide
- acceptable crypto attributes, we currently issue a WARNING that
- uses the phrase "we" and "requested". In this case, both of those
- terms are ambiguous - the user will probably think "we" is
- Asterisk (it most likely isn't) and it may not be a "request", so
- much as an SDP that was received in some fashion. This patch
- makes the WARNING messages slightly less bad and a bit more
- accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton
- ........ Merged revisions 432277 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * main/sdp_srtp.c, /: channels/sip/sdp_crypto: Handle SRTP keys
- negotiated with key lifetime/MKI Prior to this patch, SDP offers
- negotiating SDES-SRTP crypto attributes would be rejected if
- those crypto attributes contained either a key lifetime or a MKI
- parameter. While from a theoretical point of view this was
- defensible - Asterisk does not support key lifetimes or multiple
- crypto keys - from a practical point of view, this is quite a
- problem. A large number of endpoints offer lifetimes/MKI, which
- Asterisk can tolerate so long as it doesn't actually have to
- support anything more than a single key or refresh the key. In
- reality, this is (so far as we've seen) always the case. This
- patch is a forward port of Olle's work in the
- lingon-srtp-key-lifetime-1.8 branch. To quote Olle from
- ASTERISK-17721, it handles lifetime/MKI parameters in the
- following fashion: > The Lingon branch now handle lifetime and
- MKI parameters. > > We only accept lifetimes up to max for the
- crypto and higher than 10 hours > for packetization of 20 ms (50
- pps). > > We only handle MKI with index 1. > > We do not really
- bother with counting packets and reinviting at end of > lifetime,
- so the min of 10 hours kind of takes care of most calls. If there
- > are longer ones, we rely on the other side for re-invites. > >
- It's still not perfect, but I personally think this is an
- improvement. A > configuration option for minimum lifetime
- accepted could be added. When the patch was ported forward, I
- decided against adding a configuration option as Olle's handling
- was more than sufficient for every case I've seen come through
- the issue tracker or through interoperability testing. We can
- revisit that decision if it proves to be false. A few small other
- tweaks were made to the surrounding code to reduce indentation
- and provide better type safety for the 'tag' parameter. Review:
- https://reviewboard.asterisk.org/r/4419/ Review:
- https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close
- Reported by: Terry Wilson ASTERISK-17899 #close Reported by:
- Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff
- uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai
- ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged
- revisions 432239 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-25 20:44 +0000 [r432237] David M. Lee <dlee@digium.com>
-
- * res/res_http_websocket.c, /: Increase WebSocket frame size and
- improve large read handling Some WebSocket applications, like
- [chan_respoke][], require a larger frame size than the default
- 8k; this patch bumps the default to 16k. This patch also fixes
- some problems exacerbated by large frames. The sanity counter was
- decremented on every fread attempt in ws_safe_read(), regardless
- of whether data was read from the socket or not. For large
- frames, this could result in loss of sanity prior to reading the
- entire frame. (16k frame / 1448 bytes per segment = 12 segments).
- This patch changes the sanity counter so that it only decrements
- when fread() doesn't read any bytes. This more closely matches
- the original intention of ws_safe_read(), given that the error
- message is "Websocket seems unresponsive". This patch also
- properly logs EOF conditions, so disconnects are no longer
- confused with unresponsive connections. [chan_respoke]:
- https://github.com/respoke/chan_respoke Review:
- https://reviewboard.asterisk.org/r/4431/ ........ Merged
- revisions 432236 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-24 22:14 +0000 [r432195-432199] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: channels/chan_sip: Fix crash when
- transmitting packet after thread shutdown When the monitor thread
- is stopped, its pthread ID is set to a specific value
- (AST_PTHREADT_STOP) so that later portions of the code can
- determine whether or not it is safe to manipulate the thread.
- Unfortunately, __sip_reliable_xmit failed to check for that
- value, checking instead only for AST_PTHREAD_STOP. Passing the
- invalid yet very specific value to pthread_kill causes a crash.
- This patch adds a check for AST_PTHREADT_STOP in
- __sip_reliable_xmit such that it doesn't attempt to poke the
- thread if the thread has already been stopped. ASTERISK-24800
- #close Reported by: JoshE ........ Merged revisions 432198 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * channels/chan_pjsip.c, main/channel.c, res/res_pjsip_sdp_rtp.c,
- res/ari/resource_channels.c: ARI/PJSIP: Apply requesting
- channel's format cap to created channels This patch addresses the
- following problems: * ari/resource_channels: In ARI, we currently
- create a format capability structure of SLIN and apply it to the
- new channel being created. This was originally done when the PBX
- core was used to create the channel, as there was a condition
- where a newly created channel could be created without any
- formats. Unfortunately, now that the Dial API is being used, this
- has two drawbacks: (a) SLIN, while it will ensure audio will
- flows, can cause a lot of needless transcodings to occur,
- particularly when a Local channel is created to the dialplan.
- When no format capabilities are available, the Dial API handles
- this better by handing all audio formats to the requsted
- channels. As such, we defer to that API to provide the format
- capabilities. (b) If a channel (requester) is causing this
- channel to be created, we currently don't use its format
- capabilities as we are passing in our own. However, the Dial API
- will use the requester channel's formats if none are passed into
- it, and the requester channel exists and has format capabilities.
- This is the "best" scenario, as it is the most likely to create a
- media path that minimizes transcoding. Fixing this simply entails
- removing the providing of the format capabilities structure to
- the Dial API. * chan_pjsip: Rather than blindly picking the first
- format in the format capability structure - which actually *can*
- be a video or text format - we select an audio format, and only
- pick the first format if that fails. That minimizes the weird
- scenario where we attempt to transcode between video/audio. *
- res_pjsip_sdp_rtp: Applied the joint capapbilites to the format
- structure. Since ast_request already limits us down to one format
- capability once the format capabilities are passed along, there's
- no reason to squelch it here. * channel: Fixed a comment. The
- reason we have to minimize our requested format capabilities down
- to a single format is due to Asterisk's inability to convey the
- format to be used back "up" a channel chain. Consider the
- following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u
- That is, we have PJSIP/A dialing a Local channel, where the
- Local;2 dials PJSIP/B. PJSIP/A has native format capabilities
- g722,ulaw,alaw; the Local channel has inherited those format
- capabilities down the line; PJSIP/B supports only ulaw. According
- to these format capabilities, ulaw is acceptable and should be
- selected across all the channels, and no transcoding should
- occur. However, there is no way to convey this: when L;2 and
- PJSIP/B are put into a bridge, we will select ulaw, but that is
- not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A
- <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be
- written to PJSIP/B. Even if we can convey the 'ulaw' choice back
- up the chain (which through some severe hacking in Local channels
- was accomplished), such that the chain looks like: PJSIP/A <=>
- L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's
- *channel driver* to Answer in the SDP back with only 'ulaw'. This
- results in all the channel structures being set up correctly, but
- PJSIP/A *still* sending g722 and causing the chain to fall apart.
- There's a lot of difficulty just in setting this up, as there are
- numerous race conditions in the act of bridging, and no clean
- mechanism to pass the selected format backwards down an
- established channel chain. As such, the best that can be done at
- this point in time is clarifying the comment. Review:
- https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close
- Reported by: Matt Jordan
-
-2015-02-24 18:32 +0000 [r432175] Kevin Harwell <kharwell@digium.com>
-
- * /, bridges/bridge_softmix.c: bridge_softmix: G.729 codec license
- held When more than one call using the same codec type enters
- into a softmix bridge and no audio is present for a channel the
- bridge optimizes the out frame by using the same one for all
- channels with the same codec type. Unfortunately, when that
- number (channels with same codec type) dropped to <= 1 the codec
- was not dereferenced. At least not until all parties left the
- bridge. Thus in the case of G.729 the license was not released.
- This patch ensures that the codec is dereferenced immediately
- when the optimization no longer applies. ASTERISK-24797 #close
- Reported by: Luke Hulsey Review:
- https://reviewboard.asterisk.org/r/4429/ ........ Merged
- revisions 432174 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-21 20:47 +0000 [r432118-432154] Joshua Colp <jcolp@digium.com>
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c: res_ari_channels: Return a 404 response
- when a requested channel variable does not exist. This change
- makes it so that if a channel variable is requested and it does
- not exist a 404 response will be returned instead of an
- allocation failed response. This makes it easier to debug and
- figure out what is going on for a user. ASTERISK-24677 #close
- Reported by: Joshua Colp
-
- * res/res_pjsip_registrar.c: res_pjsip_registrar: Add Expires
- header to 200 OK if present in REGISTER. Some implementations
- don't pay attention to the expires for individual contacts. In
- this case they may consider the lack of an Expires header in the
- 200 OK as unregistered. This change makes it so if an Expires
- header is present in the REGISTER we will add one in the 200 OK.
- ASTERISK-24785 #close Reported by: Ross Beer
-
- * res/res_pjsip.c: res_pjsip: Add a log message when creating a UAC
- dialog to a target URI that is invalid. ASTERISK-24499 #close
- Reported by: Rusty Newton
-
-2015-02-21 17:35 +0000 [r432099] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_voicemail.c, /: apps/app_voicemail: Demote an ERROR
- message to a WARNING message When using IMAP voicemail with
- FreePBX, you will often get ERROR messages complaining about not
- being able to find a mailbox. This is due to how FreePBX handles
- voicemail mailboxes. Unfortunately, app_voicemail has to consider
- this a configuration error, as in any other system it would be
- indicative of someone misconfiguring their system. Regardless, a
- misconfiguration is a WARNING, and not an ERROR. This patch
- demotes the message so that system administrators can hopefully
- reduce some of the noise in their log files. Note that in the
- original patch this was made into a NOTICE, but that's a too
- forgiving. ASTERISK-24790 #close Reported by: Graham Barnett
- patches: app_voicemail.c.patch_noise uploaded by Graham Barnett
- (License 6685) ........ Merged revisions 432098 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-21 14:05 +0000 [r432079] Joshua Colp <jcolp@digium.com>
-
- * main/http.c, /: http: Add missing html tag to 'httpstatus'
- functionality. ASTERISK-24724 #close Reported by: Ashley Sanders
- ........ Merged revisions 432078 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-21 02:56 +0000 [r432055-432059] Corey Farrell <git@cfware.com>
-
- * /, main/bucket.c, main/codec.c, main/loader.c: Allow shutdown to
- unload modules that register bucket scheme's or codec's. * Change
- __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules
- that call ast_bucket_scheme_register or ast_codec_register to be
- unloaded during graceful shutdown only (13+ only). ASTERISK-24796
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4428/ ........ Merged
- revisions 432058 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, include/asterisk/lock.h: asterisk/lock.h: Fix syntax errors
- for non-gcc OSX with 64-bit integers. Add a couple of missing
- closing brackets / parenthesis. ASTERISK-24814 #close Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4436/ ........ Merged
- revisions 432054 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-20 17:51 +0000 [r432034] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/sig_analog.c: chan_dahdi/sig_analog: Put log message
- strings on one line. With the log messages on one line, you can
- search for the log message seen in the log and expect to find it.
- ........ Merged revisions 432032 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-20 17:46 +0000 [r432033] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_publish_asterisk.c, res/res_pjsip_acl.c:
- ASTERISK-24811: Add ast_sorcery_apply_config() to
- res_pjsip_publish_asterisk. Matt Hoskins reported that
- res_pjsip_publish_asterisk wouldn't pull config from realtime.
- Turns out it was just missing a call ast_sorcery_apply_config().
- res_pjsip_acl was missing it as well, so I added it. The other
- pjsip modules looked OK. ASTERISK-24811 #close Reported-by: Matt
- Hoskins Tested-by: George Joseph Tested-by: Matt Hoskins patches:
- res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins
- (license 6688) Review: https://reviewboard.asterisk.org/r/4433/
-
-2015-02-20 15:47 +0000 [r432013] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_voicemail.c, /: apps/app_voicemail: Fix IMAP header
- compatibility issue with Microsoft Exchange When interfacing with
- Microsoft Exchange, custom headers will be returned as all lower
- case. Currently, the IMAP header code will fail to parse the
- returned custom headers, as it will be performing a case
- sensitive comparison. This can cause playback of messages to
- fail, as needed information - such as origtime - will not be
- present. This patch updates app_voicemail's header parsing code
- to perform a case insensitive lookup for the requested custom
- headers. Since the headers are specific to Asterisk, e.g.,
- 'x-asterisk-vm-orig-time', and headers should be unique in an
- IMAP message, this should cause no issues with other systems.
- ASTERISK-24787 #close Reported by: Graham Barnett patches:
- app_voicemail.c.patch_MSExchange uploaded by Graham Barnett
- (License 6685) ........ Merged revisions 432012 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-19 21:25 +0000 [r431956-431993] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi:
- Remove some dead code. ........ Merged revisions 431992 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * main/aoc.c: ISDN AOC: Fix crash from an AOC-E message that
- doesn't have a channel association. Processing an AOC-E event
- that does not or no longer has a channel association causes a
- crash. The problem with posting AOC events to the channel topic
- is that AOC-E events don't always have a channel association and
- posting the event to the all channels topic is just wrong. AOC-E
- events do however have their own charging association method to
- refer to the agreement with the charging entity. * Changed the
- AOC events to post to the AMI manager topic instead of the
- channel topics. If a channel is associated with the event then
- channel snapshot information is supplied with the AMI event. *
- Eliminated RAII_VAR() usage in aoc_to_ami() and
- ast_aoc_manager_event(). This patch supercedes the patch on
- Review: https://reviewboard.asterisk.org/r/4427/ ASTERISK-22670
- #close Reported by: klaus3000 ASTERISK-24689 #close Reported by:
- Marcel Manz ASTERISK-24740 #close Reported by: Panos Gkikakis
- Review: https://reviewboard.asterisk.org/r/4430/
-
- * res/res_pjsip_refer.c: res_pjsip_refer: Handle INVITE with
- Replaces failure after answer. * Fixed hangup handling of the
- session->channel after answer if the ast_channel_move() or
- ast_bridge_impart() fails. We are still the thread controlling
- the session->channel so we need to call ast_hangup() to kill the
- channel. * Fixed debug messages in
- refer_incoming_invite_request() referencing incorrect channnels
- on success. Code comments now say why the session->channel cannot
- be used. Review: https://reviewboard.asterisk.org/r/4422/
-
-2015-02-19 15:28 +0000 [r431937] Matthew Jordan <mjordan@digium.com>
-
- * main/tcptls.c, /: tcptls: Handle new OpenSSL compile time option
- to disable SSLv3 Some distributions are going to disable SSLv3 at
- compile time. This option can be checked using the directive
- OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling
- in Asterisk to look for that directive before attempting to use
- the SSLv3 specific methods. ASTERISK-24799 #close Reported by:
- Alexander Traud patches: no-ssl3-method.patch uploaded by
- Alexander Traud (License 6520) ........ Merged revisions 431936
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-19 02:01 +0000 [r431917] Corey Farrell <git@cfware.com>
-
- * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c:
- Create work around for scheduler leaks during shutdown. * Added
- ast_sched_clean_by_callback for cleanup of scheduled events that
- have not yet fired. * Run all pending peercnt_remove_cb and
- replace_callno events in chan_iax2. Cleanup of replace_callno
- events is only run 11, since it no longer releases any references
- or allocations in 13+. ASTERISK-24451 #close Reported by: Corey
- Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........
- Merged revisions 431916 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-17 15:31 +0000 [r431898] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, res/res_pjsip_messaging.c,
- res/res_pjsip_caller_id.c, res/res_pjsip_refer.c,
- res/res_pjsip_send_to_voicemail.c: res_pjsip_refer: Fix crash
- from a REFER and BYE collision. Analyzing a one-off crash on a
- busy system showed that processing a REFER request had a NULL
- session channel pointer. The only way I can think of that could
- cause this is if an outgoing BYE transaction overlapped the
- incoming REFER transaction in a collision. Asterisk sends a BYE
- while the phone sends a REFER to complete an attended transfer. *
- Made check the session channel pointer before processing an
- incoming REFER request in res_pjsip_refer. * Fixed similar crash
- potential for res_pjsip supplement incoming request processing
- for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE,
- res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail
- REFER messages. * Made res_pjsip_messaging respond to a message
- body too large with a 413 instead of ignoring it. ASTERISK-24700
- #close Reported by: Zane Conkle Review:
- https://reviewboard.asterisk.org/r/4417/
-
-2015-02-16 21:29 +0000 [r431879] Matthew Jordan <mjordan@digium.com>
-
- * res/res_rtp_asterisk.c: res/res_rtp_asterisk: Fix crash in debug
- from RTCP reports without report block When RTCP debugging was
- enabled, an RTCP report without a report block would cause a
- crash. This was due to the verbose output not checking to see if
- the report_block pointer was NULl before dereferencing it. This
- patch adds the necessary check to prevent printing any verbose
- output if the far side hasn't provided us the information they
- should have. ASTERISK-24791 #close Reported by: JoshE Tested by:
- JoshE
-
-2015-02-15 19:00 +0000 [r431807-431860] Joshua Colp <jcolp@digium.com>
-
- * configs/samples/pjsip.conf.sample: pjsip: Remove "contact" type
- from pjsip.conf.sample The "contact" object is not meant to be
- configured from the pjsip.conf configuration file. It is meant to
- be created as a result of a registration and stored elsewhere.
- ASTERISK-24085 #close Reported by: Rusty Newton
-
- * contrib/scripts/install_prereq: install_prereq: Tweak flags when
- configuring pjproject. This change does two things: 1. Disables
- debugging so assertions which can return an error do, instead of
- asserting. 2. Enables IPv6 support. ASTERISK-24632 #close
- Reported by: Rusty Newton
-
- * res/res_sorcery_config.c: res_sorcery_config: Improve object
- lookup times. The res_sorcery_config module currently uses a
- fixed bucket size of 53. This means that depending on the number
- of objects you either end up with excess buckets or a lot of
- collisions. Due to the way that res_sorcery_config is implemented
- it's actually possible to make the bucket size dynamic based on
- the number of objects. This is due to the fact that each loading
- of the config file produces a new container and does not modify
- the existing one. This change uses the number of expected objects
- and finds a prime number near it. In practice depending on the
- number of objects this can speed up lookups anywhere from 2X to
- 15X. This change also removes the lock from the container as it
- is not needed. Review: https://reviewboard.asterisk.org/r/4423/
-
- * res/res_pjsip/pjsip_cli.c: res_pjsip: Add "pjsip show version"
- CLI command. When debugging things it can be useful to know
- absolutely what version of pjproject res_pjsip is running
- against. This change adds a "pjsip show version" CLI command
- which can be used to query for this. ASTERISK-24685 #close
- Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/4424/
-
- * res/res_timing_pthread.c: res_timing_pthread: Fix leaky pipes.
- During some refactoring the way private information for timers
- was stored was changed. As a result of this the action which
- normally removed the timer upon closure in res_timing_pthread was
- also removed causing the timer to remain after it should using up
- resources. This change ensures that the timer is removed upon
- closure. ASTERISK-24768 #close Reported by: Matthias Urlichs
- patches: timer.patch submitted by Matthias Urlichs (license 5508)
-
-2015-02-15 00:32 +0000 [r431789] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/app_mixmonitor.c: apps/app_mixmonitor: Move Test Event
- for MIXMONITOR_END to after it finishes The Test Event for
- MIXMONITOR_END - which signals that a MixMonitor has completed -
- technically fired before the filestream was closed. If a test
- used this to trigger a condition to verify that the file was
- written, it could result in a race condition where the file size
- would not be what the test expected. Luckily, no tests were using
- this (although they should have been). Since the test event
- needed to be moved after the point where the MixMonitor autochan
- has been destroyed, the test event no longer emits the channel
- name. Luckily, nothing needs it. ........ Merged revisions 431788
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-14 19:45 +0000 [r431751-431771] Joshua Colp <jcolp@digium.com>
-
- * main/sorcery.c: sorcery: Output an error message if a wizard is
- specified for an object type and it isn't found. ASTERISK-24612
- #close Reported by: Joshua Colp
-
- * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Improve log
- message when a subscription is attempted to a non-existent
- extension. ASTERISK-24716 #close Reported by: Rusty Newton
-
- * channels/pjsip/dialplan_functions.c: 'information' ends with an
- 'n'.
-
- * channels/pjsip/dialplan_functions.c: chan_pjsip: Fix crash when
- CHANNEL dialplan function is invoked with pjsip argument and no
- type. ASTERISK-24771 #close Reported by: Niklas Larsson
-
-2015-02-13 17:21 +0000 [r431734] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_session.c: res_pjsip_session: Fix double re-INVITE
- collision crash. A multi-asterisk box setup with direct media
- enabled would occasionally crash when two re-INVITE collisions on
- a call leg happen in a row. The re-INVITE logic only had one
- timer struct to defer the re-INVITE. When the second collision
- happens the timer struct is overwritten and put into the timer
- heap again. Resources for the first timer are leaked and the heap
- has two positions occupied by the same timer struct. Now the heap
- ordering is potentially corrupted, the timer will fire twice, and
- any resources allocated for the second timer will be released
- twice. * The solution is to put the collided re-INVITE into the
- delayed requests queue with all the other delayed requests and
- cherry pick the next request that can come off the queue when an
- event happens. * Changed to put delayed BYE requests at the head
- of the delayed queue. There is no sense in processing delayed
- UPDATEs and re-INVITEs when a BYE has been requested. * Made the
- start of a BYE request flush the delayed requests queue to
- prevent a delayed request from overlapping the BYE transaction. I
- saw a few cases where a delayed re-INVITE got started after the
- BYE transaction started. * Changed the delayed_request struct to
- use an enum instead of a string for the request method. Cherry
- picking the queue is easier with an enum than string comparisons
- and the compiler can warn if a switch statement does not cover
- all defined enum values. * Improved the debug output to give more
- information. It helps to know which channel is involved with an
- endpoint. Trunks can have many channels associated with the
- endpoint at the same time. ASTERISK-24727 #close Reported by:
- Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/
-
-2015-02-12 20:32 +0000 [r431717] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_multihomed.c, res/stasis/control.c,
- include/asterisk/stasis_app.h, rest-api/api-docs/channels.json,
- res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c,
- channels/chan_pjsip.c, res/res_pjsip_nat.c,
- res/res_pjsip_transport_websocket.c, res/ari/resource_channels.h:
- ARI/PJSIP: Add the ability to redirect (transfer) a channel in a
- Stasis app This patch adds a new feature to ARI to redirect a
- channel to another server, and fixes a few bugs in PJSIP's
- handling of the Transfer dialplan application/ARI redirect
- capability. *New Feature* A new operation has been added to the
- ARI channels resource, redirect. With this, a channel in a Stasis
- application can be redirected to another endpoint of the same
- underlying channel technology. *Bug fixes* In the process of
- writing this new feature, two bugs were fixed in the PJSIP stack:
- (1) The existing .transfer channel callback had the limitation
- that it could only transfer channels to a SIP URI, i.e., you had
- to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan
- application. While this is still supported, it is somewhat
- unintuitive - particularly in a world full of endpoints. As such,
- we now also support specifying the PJSIP endpoint to transfer to.
- (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a
- 302 redirect by updating its Contact header. Alas, that resulted
- in the forwarding destination set by the dialplan application/ARI
- resource/whatever being rewritten with very incorrect
- information. Hence, we now don't bother updating an outgoing
- response if it is a 302. Since this took a looong time to find,
- some additional debug statements have been added to those modules
- that update the Contact headers. Review:
- https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close
- Reported by: Private Name ASTERISK-24703 #close Reported by: Matt
- Jordan
-
-2015-02-11 18:02 +0000 [r431693-431698] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c: res_pjsip: dtls_handler
- causes Asterisk to crash There have been a couple of times where
- a crash occurred in the dtls_handler section of the code for
- res_pjsip. Unfortunately, in working this issue the problem was
- unable to be reproduced. After looking at the backtraces and
- through the code the current best guess as to why this happened
- might be due to a reentrance problem and the strtok function. So,
- the current fix is to convert the strtok function into the
- reentrant version of the function, strtok_r. ASTERISK-24741
- #close Reported by: Zane Conkle Review:
- https://reviewboard.asterisk.org/r/4409/
-
- * res/ari/ari_websockets.c: ari_websockets: removed extra check on
- websocket session read When merging the websocket timeout issue
- (ASTERISK-24701) an extra, almost duplicate, check was left in
- the code that should not have been. This removes it.
- ASTERISK-24701 #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4412/
-
-2015-02-11 17:28 +0000 [r431692] Richard Mudgett <rmudgett@digium.com>
-
- * main/bridge.c, main/http.c, apps/app_confbridge.c,
- include/asterisk/channel.h, res/res_pjsip/pjsip_options.c,
- res/res_pjsip_pubsub.c, main/asterisk.c, main/channel.c,
- include/asterisk.h, channels/chan_sip.c: HTTP: Stop accepting
- requests on final system shutdown. There are three CLI commands
- to stop and restart Asterisk each. 1) core stop/restart now -
- Hangup all calls and stop or restart Asterisk. New channels are
- prevented while the shutdown request is pending. 2) core
- stop/restart gracefully - Stop or restart Asterisk when there are
- no calls remaining in the system. New channels are prevented
- while the shutdown request is pending. 3) core stop/restart when
- convenient - Stop or restart Asterisk when there are no calls in
- the system. New calls are not prevented while the shutdown
- request is pending. ARI has made stopping/restarting Asterisk
- more problematic. While a shutdown request is pending it is
- desirable to continue to process ARI HTTP requests for current
- calls. To handle the current calls while a shutdown request is
- pending, a new committed to shutdown phase is needed so ARI
- applications can deal with the calls until the system is fully
- committed to shutdown. * Added a new shutdown committed phase so
- ARI applications can deal with calls until the final committed to
- shutdown phase is reached. * Made refuse new HTTP requests when
- the system has reached the final system shutdown phase. Starting
- anything while the system is actively releasing resources and
- unloading modules is not a good thing. * Split the bridging
- framework shutdown to not cleanup the global bridging containers
- when shutting down in a hurry. This is similar to how other
- modules prevent crashes on rapid system shutdown. * Moved
- ast_begin_shutdown(), ast_cancel_shutdown(), and
- ast_shutting_down(). You should not have to include channel.h
- just to access these system functions. ASTERISK-24752 #close
- Reported by: Matthew Jordan Review:
- https://reviewboard.asterisk.org/r/4399/
-
-2015-02-11 17:12 +0000 [r431674] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: channels/chan_sip: Fix RealTime error
- during SIP unregistration with MariaDB When a SIP device that has
- its registration stored in RealTime unregisters, the entry for
- that device is updated with blank values, i.e., "", indicating
- that it is no longer registered. Unfortunately, one of those
- values that is 'blanked' is the device's port. If the column type
- for the port is not a string datatype (the recommended type is
- integer), an ODBC or database error will be thrown. MariaDB does
- not coerce empty strings to a valid integer value. This patch
- updates the query run from chan_sip such that it replaces the
- port value with a value of '0', as opposed to a blank value. This
- is the value that other database backends coerce the empty string
- ("") to already, and the handling of reading a RealTime
- registration value from a backend already anticipates receiving a
- port of '0' from the backends. ASTERISK-24772 #close Reported by:
- Richard Miller patches: chan_sip.diff uploaded by Richard Miller
- (License 5685) ........ Merged revisions 431673 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-11 16:51 +0000 [r431670] Kevin Harwell <kharwell@digium.com>
-
- * res/res_http_websocket.c, res/ari/ari_websockets.c, /:
- res_http_websocket: websocket write timeout fails to fully
- disconnect When writing to a websocket if a timeout occurred the
- underlying socket did not get closed/disconnected. This patch
- makes sure the websocket gets disconnected on a write timeout.
- Also a notice is logged stating that the websocket was
- disconnected. ASTERISK-24701 #close Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged
- revisions 431669 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-11 15:51 +0000 [r431663] Corey Farrell <git@cfware.com>
-
- * include/asterisk/module.h, main/loader.c, /,
- bridges/bridge_builtin_features.c: Enable REF_DEBUG for
- ast_module_ref / ast_module_unref. Add ast_module_shutdown_ref
- for use by modules that can only be unloaded during graceful
- shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to
- struct ast_module. * Allocate ao2 object when the module is
- loaded. * Perform an ao2_ref in each place where mod->usecount is
- manipulated. * ao2_cleanup on module unload. ASTERISK-24479
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4141/ ........ Merged
- revisions 431662 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-02-10 23:16 +0000 [r431643] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_config_wizard.c,
- configs/samples/pjsip_wizard.conf.sample:
- res_pjsip_config_wizard: Add ability to auto-create hints.
- Looking at the Super Awesome Company sample reminded me that
- creating hints is just plain gruntwork. So you can now have the
- pjsip conifg wizard auto-create them for you. Specifying
- 'hint_exten' in the wizard will create 'exten =>
- <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for
- 'hint_context'. Specifying 'hint_application' in the wizard will
- create 'exten => <hint_exten>,1,<hint_application>' in whatever
- is specified for 'hint_context'. The default for 'hint_context'
- is the endpoint's context. There's no default for
- 'hint_application'. If not specified, no app is added. There's no
- default for 'hint_exten'. If not specified, neither the hint
- itself nor the application will be created. Some may think this
- is the slippery slope to users.conf but hints are a basic
- necessity for phones unlike voicemail, manager, etc that
- users.conf creates. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4383/
-
-2015-02-09 03:10 +0000 [r431600-431622] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c:
- res/ari/resource_channels: Add missing 'no_answer' reason to
- DELETE /channels One of the canonical reasons for hanging up a
- channel is because the far end failed to answer - or because
- someone else answered, and we want to get rid of this channel.
- This patch adds the missing value to the 'reason' query parameter
- for the DELETE /channels operation. Review:
- https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close
- Reported by: Ben Merrills patches:
- add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills
- (License 6678)
-
- * /, res/res_odbc.c: res/res_odbc: Remove unneeded queries when
- determining if a table exists This patch modifies the
- ast_odbc_find_table function such that it only performs a lookup
- of the requested table if the table is not already known. Prior
- to this patch, a queries would be executed against the database
- even if the table was already known and cached. Review:
- https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close
- Reported by: ibercom patches: patch.diff uploaded by ibercom
- (License 6599) ........ Merged revisions 431617 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Fix leak of local
- ICE candidates when applying to SDP When an SDP is created for an
- outgoing request/response, the ICE candidates obtained from the
- RTP instance are currently leaked. This causes the ao2 container
- that holds the candidates to never properly be reclaimed when the
- RTP instance is destroyed. This patch properly decrements the ICE
- candidates' container if it is successfully obtained.
- ASTERISK-24769 #close Reported by: Matt Jordan
-
-2015-02-06 21:26 +0000 [r431583] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/utils.c, res/res_pjsip.c, main/config.c: various: cleanup
- issues found during leak hunt In this collection of small patches
- to prevent Valgrind errors are: fixes for reference leaks in
- config hooks, evaluating a parameter beyond bounds, and accessing
- a structure after a lock where it could have been already free'd.
- Review: https://reviewboard.asterisk.org/r/4407/
-
-2015-02-04 01:27 +0000 [r431538-431555] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_keepalive.c: res_pjsip_keepalive: Don't crash if
- PJSIP module is not loaded.
-
- * main/sorcery.c: sorcery: Don't try to load object types which
- haven't been defined. The act of defining wizards for an object
- type in sorcery.conf will create a minimal object type. This can
- cause a problem when a module has multiple sorcery instances
- (which all get the wizards from sorcery.conf applied) but the
- sorcery instances do not all contain full information about the
- object types. Upon loading errors will occur stating that the
- objects can not be created. This is confusing and is actually
- perfectly fine. This change makes it so that only object types
- which have been fully defined will be loaded. ASTERISK-24748
- #close Reported by: Joshua Colp
-
-2015-01-31 16:27 +0000 [r431521] Joshua Colp <jcolp@digium.com>
-
- * res/res_format_attr_h264.c: res_format_attr_h264: Fix crash when
- determining joint capability. The res_format_attr_h264 module
- currently incorrectly attempts to copy SPS and PPS information
- from the wrong attribute. This change fixes that. ASTERISK-24616
- #close Reported by: Yura Kocyuba Review:
- https://reviewboard.asterisk.org/r/4392/
-
-2015-02-06 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.2.0 Released.
-
-2015-01-30 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.2.0-rc1 Released.
-
-2015-01-30 17:44 +0000 [r431492] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_agent_pool.c: app_agent_pool: Fix initial module load
- agent device state reporting. When the app_agent_pool module
- initially loads there is a race condition between the thread
- loading agents.conf and the device state internal processing
- thread. If the device state internal processing thread handles
- the agent creation state updates before the thread that loaded
- agents.conf registers the device state provider callback then the
- cached agent state is "Invalid". When a consumer module like
- app_queue asks for the agent state it gets the cached "Invalid"
- state instead of the real state from the provider. * Moved
- loading the agents.conf configuration to the last thing setup by
- app_agent_pool in load_module(). Now the device state provider
- callback is registered before the config is loaded so the agent
- creation state updates are guaranteed to get the initial device
- state. * Removed some now redundant config cleanup on error in
- load_config(). * Added lock protection when accessing the device
- state in agent_pvt_devstate_get() and eliminated the RAII_VAR()
- usage. ASTERISK-24737 #close Reported by: Steve Pitts Review:
- https://reviewboard.asterisk.org/r/4390/
-
-2015-01-30 17:38 +0000 [r431490] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish:
- eventually crashes when no response is ever received When
- Asterisk attempts to send SIP outbound publish information and no
- response is ever received (no 200 okay, 412, 423) the system
- eventually crashes. A response is never received because the
- system Asterisk is attempting to send publish information to is
- not available. The underlying pjsip framework attempts to send
- publish information. After several attempts it calls back into
- the Asterisk outbound publish code. At this point if the
- "client->queue" is empty Asterisk attempts to schedule a refresh
- which utilizes "rdata" and since no response was received the
- given "rdata" struture is NULL. Attempting to dereference a NULL
- object of course results in a crash. The fix here removes the
- dependency on rdata for schedule_publish_refresh. Instead
- param->expiration is now passed to it as this is set to -1 if no
- response is received. Also added a notification when no response
- is received. ASTERISK-24635 #close Reported by: Marco Paland
- Review: https://reviewboard.asterisk.org/r/4384/
-
-2015-01-30 16:52 +0000 [r431471] asanders <asanders@localhost>:
-
- * include/asterisk/http.h, configs/samples/http.conf.sample,
- main/http.c: HTTP: For httpd server, need option to define server
- name for security purposes Added a new config property
- [servername] to the http.conf file; updated the http server to
- use the new property when sending responses, for showing http
- status through the CLI and when reporting status through the
- 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew
- Nagy Review: https://reviewboard.asterisk.org/r/4374/
-
-2015-01-30 16:47 +0000 [r431468] Mark Michelson <mmichelson@digium.com>
-
- * main/stasis_channels.c, channels/chan_pjsip.c, main/xmldoc.c,
- res/res_pjsip_refer.c, main/pbx.c, main/manager.c,
- pbx/pbx_spool.c, main/bridge_after.c: Fix some memory leaks.
- These memory leaks were found and fixed by John Hardin. I'm just
- committing them for him. ASTERISK-24736 #close Reported by Mark
- Michelson Review: https://reviewboard.asterisk.org/r/4389
-
-2015-01-29 23:02 +0000 [r431450] Scott Griepentrog <sgriepentrog@digium.com>
-
- * include/asterisk/bridge.h, main/bridge.c,
- res/stasis/stasis_bridge.c: stasis transfer: fix stasis bridge
- push race part two When swapping a Local channel in place of one
- already in a bridge (to complete a bridge attended transfer), the
- channel that was swapped out can actually be hung up before the
- stasis bridge push callback executes on the independant transfer
- thread. This results in the stasis app loop dropping out and
- removing the control that has the the app name which the local
- replacement channel needs so it can re-enter stasis. To avoid
- this race condition a new push_peek callback has been added, and
- called from the ast_bridge_impart thread before it launches the
- independant thread that will complete the transfer. Now the
- stasis push_peek callback can copy the stasis app name before the
- swap channel can hang up. ASTERISK-24649 Review:
- https://reviewboard.asterisk.org/r/4382/
-
-2015-01-29 20:58 +0000 [r431420-431426] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_sips_contact.c (added): Use SIPS
- URIs in Contact headers when appropriate. RFC 3261 sections
- 8.1.1.8 and 12.1.1 dictate specific scenarios when we are
- required to use SIPS URIs in Contact headers. Asterisk's
- non-compliance with this could actually cause calls to get
- dropped when communicating with clients that are strict about
- checking the Contact header. Both of the SIP stacks in Asterisk
- suffered from this issue. This changeset corrects the behavior in
- res_pjsip/chan_pjsip.c Review:
- https://reviewboard.asterisk.org/r/4345
-
- * /, channels/chan_sip.c: Use SIPS URIs in Contact headers when
- appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate
- specific scenarios when we are required to use SIPS URIs in
- Contact headers. Asterisk's non-compliance with this could
- actually cause calls to get dropped when communicating with
- clients that are strict about checking the Contact header. Both
- of the SIP stacks in Asterisk suffered from this issue. This
- changeset corrects the behavior in chan_sip. ASTERISK-24646
- #close Reported by Stephan Eisvogel Review:
- https://reviewboard.asterisk.org/r/4346 ........ Merged revisions
- 431423 from http://svn.asterisk.org/svn/asterisk/branches/11
-
- * res/res_pjsip/pjsip_configuration.c: Allow disabling of 100rel
- support on PJSIP endpoints. Due to an inversion error, setting
- 100rel=no would not actually change the current value of the
- setting (which defaulted to "yes"). With this fix, the inversion
- is corrected.
-
-2015-01-29 16:46 +0000 [r431403] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Reduce log
- clutter... change a WARNING to a VERBOSE/2 Reduce log clutter by
- changing the "Watcher for hint %s (removed|deactivated)" message
- from WARNING to VERBOSE/2. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4387/
-
-2015-01-29 12:09 +0000 [r431385] Joshua Colp <jcolp@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix DTLS when used
- with OpenSSL 1.0.1k A recent security fix for OpenSSL broke DTLS
- negotiation for many applications. This was caused by read ahead
- not being enabled when it should be. While a commit has gone into
- OpenSSL to force read ahead on for DTLS it may take some time for
- a release to be made and the change to be present in
- distributions (if at all). As enabling read ahead is a simple one
- line change this commit does that and fixes the issue.
- ASTERISK-24711 #close Reported by: Jared Biel ........ Merged
- revisions 431384 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-28 17:37 +0000 [r431301-431303] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c,
- res/res_pjsip_session.c: Fix file descriptor leak in RTP code.
- SIP requests that offered codecs incompatible with configured
- values could result in the allocation of RTP and RTCP ports that
- would not get reclaimed later. ASTERISK-24666 #close Reported by
- Y Ateya Review: https://reviewboard.asterisk.org/r/4323
- AST-2015-001 ........ Merged revisions 431300 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * funcs/func_curl.c, /: Multiple revisions 431297-431298 ........
- r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan
- 2015) | 17 lines Mitigate possible HTTP injection attacks using
- CURL() function in Asterisk. CVE-2014-8150 disclosed a
- vulnerability in libcURL where HTTP request injection can be
- performed given properly-crafted URLs. Since Asterisk makes use
- of libcURL, and it is possible that users of Asterisk may get
- cURL URLs from user input or remote sources, we have made a patch
- to Asterisk to prevent such HTTP injection attacks from
- originating from Asterisk. ASTERISK-24676 #close Reported by Matt
- Jordan Review: https://reviewboard.asterisk.org/r/4364
- AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49
- -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from
- previous patch. ........ Merged revisions 431297-431298 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 431299 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2015-01-28 12:18 +0000 [r431267] Sean Bright <sean@malleable.com>
-
- * res/res_format_attr_silk.c, res/res_format_attr_opus.c: media
- formats: update res_format_attr_opus & silk In r419044, we
- changed how formats were handled, but the return value of the
- format_parse_sdp_fmtp functions in res_format_attr_opus and
- res_format_attr_silk were not updated, causing calls to fail. Ran
- into this when getting codec_opus working with Asterisk 13. Once
- the return value was corrected, we were crashing in opus_getjoint
- because of NULL format attributes. I've fixed this as well in
- this patch. Review: https://reviewboard.asterisk.org/r/4371/
-
-2015-01-28 04:09 +0000 [r431243] Richard Mudgett <rmudgett@digium.com>
-
- * main/sorcery.c, res/res_pjsip_outbound_registration.c,
- res/res_pjsip.c: res_pjsip_outbound_registration: Fix reload race
- condition. Performing a CLI "module reload" command when there
- are new pjsip.conf registration objects defined frequently failed
- to load them correctly. What happens is a race condition between
- res_pjsip pushing its reload into an asynchronous task processor
- task and the thread that does the rest of the reloads when it
- gets to reloading the res_pjsip_outbound_registration module. A
- similar race condition happens between a reload and the CLI/AMI
- show registrations commands. The reload updates the
- current_states container and the CLI/AMI commands call
- get_registrations() which builds a new current_states container.
- * Made res_pjsip.c reload_module() use
- ast_sip_push_task_synchronous() instead of ast_sip_push_task() to
- eliminate two threads processing config reloads at the same time.
- * Made get_registrations() not replace the global current_states
- container so the CLI/AMI show registrations command cannot
- interfere with reloading. You could never add/remove objects in
- the container without the possibility of the container being
- replaced out from under you by get_registrations(). * Added a
- registration loaded sorcery instance observer to purge any dead
- registration objects since get_registrations() cannot do this job
- anymore. The struct ast_sorcery_instance_observer callbacks must
- be used because the callback happens inline with the load
- process. The struct ast_sorcery_observer callbacks are pushed to
- a different thread. * Added some global current_states NULL
- pointer checks in case the container disappears because of
- unload_module(). * Made sorcery's struct
- ast_sorcery_instance_observer.object_type_loaded callbacks
- guaranteed to be called before any struct
- ast_sorcery_observer.loaded callbacks will be called. * Moved the
- check for non-reloadable objects to before the sorcery instance
- loading callbacks happen to short circuit unnecessary work.
- Previously with non-reloadable objects, the sorcery instance
- loading/loaded callbacks would always happen, the individual
- wizard loading/loaded would be prevented, and the non-reloadable
- type logging message would be logged for each associated wizard.
- ASTERISK-24729 #close Review:
- https://reviewboard.asterisk.org/r/4381/
-
-2015-01-27 22:56 +0000 [r431179-431219] Kevin Harwell <kharwell@digium.com>
-
- * /, main/tcptls.c: tcptls: Bad file descriptor error when
- reloading chan_sip While running through some scenarios using
- chan_sip and tcp a problem would occur that resulted in a flood
- of bad file descriptor messages on the cli: tcptls.c:712
- ast_tcptls_server_root: Accept failed: Bad file descriptor The
- message is received because the underlying socket has been
- closed, so is valid. This is probably happening because unloading
- of chan_sip is not atomic. That however is outside the scope of
- this patch. This patch simply stops the logging of multiple
- occurrences of that message. ASTERISK-24728 #close Reported by:
- Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/
- ........ Merged revisions 431218 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, channels/chan_sip.c: chan_sip: stale nonce causes failure When
- refreshing (with a small expiration) a registration that was sent
- to chan_sip the nonce would be considered stale and reject the
- registration. What was happening was that the initial
- registration's "dialog" still existed in the dialogs container
- and upon refresh the dialog match algorithm would choose that as
- the "dialog" instead of the newly created one. This occurred
- because the algorithm did not check to see if the from tag
- matched if authentication info was available after the 401. So,
- it ended up assuming the original "dialog" was a match and
- stopped the search. The old "dialog" of course had an old nonce,
- thus the stale nonce message. This fix attempts to leave the
- original functionality alone except in the case of a REGISTER. If
- a REGISTER is received if searches for an existing "dialog"
- matching only on the callid. If the expires value is low enough
- it will reuse dialog that is there, otherwise it will create a
- new one. ASTERISK-24715 #close Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/4367/ ........ Merged
- revisions 431187 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c,
- main/stasis_message_router.c, res/res_pjsip/location.c,
- res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_distributor.c,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip/pjsip_global_headers.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
- res/res_pjsip/config_transport.c: res_pjsip: make it unloadable
- (take 2) Due to the original patch causing memory corruptions it
- was removed until the problem could be resolved. This patch is
- the original patch plus some added locking around stasis router
- subcription that was needed to avoid the memory corruption.
- Description of the original problem and patch (still applicable):
- The res_pjsip module was previously unloadable. With this patch
- it can now be unloaded. This patch is based off the original
- patch on the issue (listed below) by Corey Farrell with a few
- modifications. Namely, removed a few changes not required to make
- the module unloadable and also fixed a bug that would cause
- asterisk to crash on unloading. This patch is the first step
- (should hopefully be followed by another/others at some point) in
- allowing res_pjsip and the modules that depend on it to be
- unloadable. At this time, res_pjsip and some of the modules that
- depend on res_pjsip cannot be unloaded without causing problems
- of some sort. The goal of this patch is to get res_pjsip and only
- res_pjsip to be able to unload successfully and/or shutdown
- without incident (crashes, leaks, etc...). Other dependent
- modules may still cause problems on unload. Basically made sure,
- with the patch applied, that res_pjsip (with no other dependent
- modules loaded) could be succesfully unloaded and Asterisk could
- shutdown without any leaks or crashes that pertained directly to
- res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4363/ patches:
- pjsip_unload-broken-r1.patch submitted by Corey Farrell (license
- 5909)
-
-2015-01-27 17:36 +0000 [r431160] Richard Mudgett <rmudgett@digium.com>
-
- * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
- app_confbridge: Repeatedly starting and stopping recording ref
- leaks the recording channel. Starting and stopping conference
- recording more than once causes the recording channels to be
- leaked. For v13 the channels also show up in the CLI "core show
- channels" output. * Reworked and simplified the recording channel
- code to use ast_bridge_impart() instead of managing the recording
- thread in the ConfBridge code. The recording channel's ref
- handling easily falls into place and other off nominal code paths
- get handled better as a result. ASTERISK-24719 #close Reported
- by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/
- Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged
- revisions 431135 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-27 17:32 +0000 [r431157] Joshua Colp <jcolp@digium.com>
-
- * main/bridge_channel.c, res/res_pjsip_sdp_rtp.c: bridge /
- res_pjsip_sdp_rtp: Fix issues with media not being reinvited
- during direct media. This change fixes two issues: 1. During a
- swap operation bridging added the new channel before having the
- swap channel leave. This was not handled in bridge_native_rtp and
- could result in a channel not getting reinvited back to Asterisk.
- After this change the swap channel will leave first and the new
- channel will then join. 2. If a re-invite was received after a
- session had been established any upstream elements (such as
- bridge_native_rtp) were not notified that they may want to
- re-evaluate things. After this change an UPDATE_RTP_PEER control
- frame is queued when this situation occurs and upstream can
- react. AST-1524 #close Review:
- https://reviewboard.asterisk.org/r/4378/
-
-2015-01-27 17:22 +0000 [r431153] Jonathan Rose <jrose@digium.com>
-
- * main/manager.c: Manager: Fix Manager Action ModuleLoad to give
- correct response when reloading Prior to this patch, ModuleLoad
- would respond with an error indicating that the requested module
- wasn't found in spite of finding and reloading the module.
- Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721
- #close
-
-2015-01-27 17:20 +0000 [r431134-431145] Matthew Jordan <mjordan@digium.com>
-
- * res/ari/resource_bridges.c,
- rest-api-templates/asterisk_processor.py,
- res/ari/resource_channels.h, res/res_ari_bridges.c,
- res/ari/resource_bridges.h, rest-api-templates/api.wiki.mustache,
- rest-api/api-docs/channels.json,
- rest-api-templates/swagger_model.py,
- rest-api/api-docs/bridges.json: ARI: Improve wiki documentation
- This patch improves the documentation of ARI on the wiki.
- Specifically, it addresses the following: * Allowed values and
- allowed ranges weren't documented. This was particularly
- frustrating, as Asterisk would reject query parameters with
- disallowed values - but we didn't tell anyone what the allowed
- values were. * The /play/id operation on /channels and /bridges
- failed to document all of the added media resource types. *
- Documentation for creating a channel into a Stasis application
- failed to note when it occurred, and that creating a channel into
- Stasis conflicts with creating a channel into the dialplan. *
- Some other minor tweaks in the mustache templates, including
- italicizing the parameter type, putting the default value on its
- own sub-bullet, and some other nicities. Review:
- https://reviewboard.asterisk.org/r/4351
-
- * apps/confbridge/conf_config_parser.c,
- apps/confbridge/include/confbridge.h: app_confbridge: Restore
- user's menu name to CLI output of 'confbridge list' When issuing
- a 'confbridge list XXXX' CLI command, the resulting output no
- longer displays the menu associated with a ConfBridge
- participant. The issue was caused by ASTERISK-22760. When that
- patch was done, it removed the copying of the menu name
- associated with the user from the actual user profile. This patch
- fixes the issue by copying the menu name over to the user profile
- when the menu hooks are applied to the user. Since that function
- now does a little bit more than just apply the hooks, the name of
- the function has been changed to cover the copying of the menu
- name over as well. In addition, there is a disparity between the
- menu name length as it is stored on the conf_menu structure and
- the confbridge_user structure; this patch makes the lengths match
- so that a strcpy can be used. Review:
- https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close
- Reported by: Steve Pitts
-
-2015-01-27 11:47 +0000 [r431114] Joshua Colp <jcolp@digium.com>
-
- * res/parking/parking_manager.c: res_parking: Fix crash due to race
- condition when unloading. There is currently a race condition
- when unloading the res_parking module. Depending on the will of
- the universe the subscription invocation may occur AFTER the
- module is unloaded. This is because the module does NOT use
- stasis_unsubscribe_and_join when terminating the subscription. It
- merely uses stasis_unsubscribe. This change makes it use
- stasis_unsubscribe_and_join which is documented for usage in this
- exact scenario. AST-1520 #close Review:
- https://reviewboard.asterisk.org/r/4375/
-
-2015-01-26 14:49 +0000 [r431092] David M. Lee <dlee@digium.com>
-
- * channels/sip/include/route.h, funcs/func_presencestate.c,
- main/rtp_engine.c, configure, include/asterisk/autoconfig.h.in,
- include/asterisk/sem.h, configure.ac, main/app.c,
- main/bridge_channel.c, main/sem.c, res/res_timing_kqueue.c,
- main/asterisk.c: Various fixes for OS X This patch addresses
- compilation errors on OS X. It's been a while, so there's quite a
- few things. * Fixed __attribute__ decls in route.h to be
- portable. * Fixed htonll and ntohll to work when they are defined
- as macros. * Replaced sem_t usage with our ast_sem wrapper. *
- Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC
- 4.9 warnings using sig*set() functions. * Fixed some format
- strings for portability. * Fixed compilation issues with
- res_timing_kqueue (although tests still fail on OS X). * Fixed
- menuconfig /sbin/launchd detection, which disables
- res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by:
- George Joseph ASTERISK-24544 #close Reported by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4327/
-
-2015-01-25 13:42 +0000 [r431072] Matthew Jordan <mjordan@digium.com>
-
- * main/config.c: dynamic realtime: Updates fail to work due to
- update fields being passed over When a crash was fixed due to
- usage of the REALTIME function in r423003, a regression was
- introduced into ast_update2_realtime where the update fields
- passed to the function would be skipped and the lookup field
- processed twice. The use of this function is a bit interesting: A
- variable argument list is used with two sentinel values - the
- first marks the end of the lookup fields/values; the second marks
- the end of the update fields/values. Unfortunately,
- ast_update2_realtime parses over the lookup fields twice, as
- opposed to parsing over the update fields. This causes the
- lookups to succeed, but the updates itself to have no effect.
- Note that the most common instance of this problem occurred in
- app_voicemail during the updating of a mailbox password. Thanks
- to the issue reporter, Paddy Grice, for pointing out the problem.
- Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231
- ASTERISK-24626 #close Reported by: Paddy Grice
-
-2015-01-23 20:13 +0000 [r431050-431052] Richard Mudgett <rmudgett@digium.com>
-
- * apps/confbridge/conf_chan_record.c: app_confbridge: Make CBRec
- channel names more unique. Channel names should be different from
- other channels in the system while the channel exists. * Use a
- sequence number for CBRec channels instead of a random number
- because the same random number could be picked again for the next
- CBRec channel.
-
- * /, apps/app_confbridge.c: app_confbridge: Whitespace Because
- there is sometimes no sence to any whitespace. ........ Merged
- revisions 431049 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-23 17:08 +0000 [r431030] David M. Lee <dlee@digium.com>
-
- * res/res_pjsip_config_wizard.c: Add depend on pjproject to
- res_pjsip_config_wizard.c
-
-2015-01-23 15:12 +0000 [r430999] Kevin Harwell <kharwell@digium.com>
-
- * res/parking/parking_applications.c, channels/chan_iax2.c,
- res/res_pjsip/pjsip_global_headers.c, res/res_pjsip_pubsub.c,
- res/res_ari_channels.c, res/res_stasis.c,
- rest-api-templates/param_parsing.mustache,
- res/res_ari_endpoints.c, res/res_ari_events.c,
- include/asterisk/stasis_app.h, res/res_pjsip_mwi.c: Investigate
- and fix memory leaks in Asterisk Fixed memory leaks that were
- found in Asterisk. ASTERISK-24693 #close Reported by: Kevin
- Harwell Review: https://reviewboard.asterisk.org/r/4347/
-
-2015-01-23 15:03 +0000 [r430994-430998] Walter Doekes <walter+asterisk@wjd.nu>
-
- * apps/app_voicemail.c, channels/chan_unistim.c,
- funcs/func_hangupcause.c, main/manager_bridges.c,
- channels/chan_misdn.c, funcs/func_groupcount.c, /,
- addons/ooh323c/src/ooh245.c, channels/chan_sip.c, res/res_fax.c,
- res/res_pjsip_outbound_registration.c, apps/app_minivm.c,
- apps/app_alarmreceiver.c, include/asterisk/channel.h,
- contrib/utils/eagi_proxy.c: Fix typo's (retrieve, specified,
- address). ........ Merged revisions 430996 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * /, channels/chan_sip.c: chan_sip: Case insensitive comparison of
- "defaultuser" parameter. All the other configuration options are
- case insensitive, so this one should be too. ASTERISK-24355
- #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded
- by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions
- 430993 from http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-22 19:24 +0000 [r430957-430975] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/bridge.h,
- include/asterisk/bridge_channel_internal.h, main/bridge.c,
- include/asterisk/bridge_internal.h, main/bridge_channel.c: Bridge
- core: Pass a ref with the swap channel when joining a bridge.
- When code imparts a channel into a bridge to swap with another
- channel, a ref needs to be held on the swap channel to ensure
- that it cannot dissapear before finding it in the bridge. * The
- ast_bridge_join() swap channel parameter now always steals a ref
- for the swap channel. This is the only change to the bridge
- framework's public API semantics. *
- bridge_channel_internal_join() now requires the
- bridge_channel->swap channel to pass in a ref. ASTERISK-24649
- Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/4354/
-
- * res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration.c: Minor code cleanup. * Add an
- allocation failure check and assert in
- sip_outbound_registration_response_cb(). * Made
- sip_outbound_registration_state_destroy() handle partially
- created state objects from
- sip_outbound_registration_state_alloc(). Review:
- https://reviewboard.asterisk.org/r/4366/
-
-2015-01-22 18:09 +0000 [r430939] Scott Griepentrog <sgriepentrog@digium.com>
-
- * res/stasis/app.c, res/stasis/stasis_bridge.c: stasis transfer:
- fix a race condition on stasis bridge push After a bridge
- transfer completes where a local replacement channel is used, a
- stasis transfer message with the details of the transfer is sent.
- This is processed by stasis which then sets the stasis app name
- and replaced channel snapshot on the replacement channel.
- However, since a separate thread was already started to run
- stasis on the new replacement channel, a race was on to see if
- the message processing would be completed before the app name was
- needed, otherwise the channel would be hung up. This change moves
- the calls used to set the stasis app name and the replace
- snapshot to the bridge_stasis_push function callback from the
- bridge transfer logic, allowing the steps to be completed earlier
- and more deterministically, and the race elimianted. NOTE: the
- swap channel parameter to bridge_stasis_push (and thus all bridge
- push callbacks) must always be present when performing a swap
- with another channel. ASTERISK-24649 #close Reported by: John
- Bigelow Review: https://reviewboard.asterisk.org/r/4341/
-
-2015-01-22 14:23 +0000 [r430921] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/app_voicemail.c: apps/app_voicemail: Trigger MWI
- notification with MixMonitor m() option The MixMonitor m() option
- allows a recording to be pushed to a specific voicemail mailbox.
- If the message is delivered to the mailbox's INBOX, however, no
- MWI notification is currently raised. This patch corrects the
- issue by properly calling notify_new_state from the
- msg_create_from_file function. This will cause MWI to be
- triggered if the message was placed in the mailbox's INBOX.
- ASTERISK-24709 #close Reported by: Gareth Palmer patches:
- app_voicemail-430919.patch uploaded by Gareth Palmer (License
- 5169) ........ Merged revisions 430920 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-21 21:53 +0000 [r430902] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration.c: Move unref to a better place.
- Move an unconditional unref of client_state so it doesn't look
- like it could be used after the last ref has destroyed it.
-
-2015-01-21 13:33 +0000 [r430840-430864] Matthew Jordan <mjordan@digium.com>
-
- * channels/chan_sip.c: channels/chan_sip: Fix registration leak
- during reload When the SIP registrations were migrated to using
- ao2 in what was then trunk, the explicit destruction of the
- registrations on module reload was removed and not replaced with
- an ao2 equivalent. Debugging done by Stefan Engström, the issue
- reporter, on ASTERISK-24673 confirmed that the reference in the
- registry_list container was being leaked. Since the purpose of
- cleanup_all_regs is to prep a registration for destruction, this
- function now calls an ao2_callback function callback with the
- OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the
- registrations. This cleans up each registration, and also removes
- it from the registration container registry_list. Review:
- https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close
- Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan
- Engström Tested by: Stefan Engström
-
- * cdr/cdr_manager.c, cel/cel_manager.c: AMI: Add documentation for
- the missing Cdr/CEL events. This patch adds AMI event
- documentation for the Cdr and CEL AMI events. Note that while
- these events do share fields with each other and with other
- channel related events, they do not contain all of the fields in
- a standard channel snapshot, nor is the description of the fields
- identical. As such, the patch opts for documentation for each
- field, for each event. Review:
- https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close
- Reported by: Dan Jenkins
-
- * apps/app_dial.c: apps/app_dial: Don't publish DialEnd twice on
- unexpected GoSub/Macro values The Dial application has some
- interesting options with the mid-call Macro (M) and GoSub (U)
- options. If the MACRO_RESULT/GOSUB_RESULT returns specific
- values, the Dial application will take some action upon the
- channels involved in the dial operation (such as hanging up a
- particular party, etc.) The Dial application ensures that a
- Stasis message is published in the event that
- MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial
- operation, so that there is a corresponding DialEnd event
- published in AMI/ARI for the DialBegin event that preceeded it. A
- bug exists where that same DialEnd event will be published on
- Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is
- not one that the Dial application cares about. This causes two
- DialEnd events to be published - one with the
- MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is
- all sorts of wrong. This patch fixes the bug by ensuring that we
- only publish a DialEnd message to Stasis if the Dial
- application's mid-call Macro/GoSub returns something that Dial
- cares about. Review: https://reviewboard.asterisk.org/r/4336
- ASTERISK-24682 #close Reported by: Matt Jordan
-
- * main/rtp_engine.c: main/rtp_engine: Format NTP timestamps as
- unsigned longs When the RTCP reports are created, the NTP
- timestamps are stored as strings, as JSON does not have an
- integer type long enough to store the value. However, on 32-bit
- systems, a signed long may overflow for some portion of the
- timestamp. This patch corrects the overflow by formatting the
- timestamps as unsigned longs.
-
-2015-01-20 16:51 +0000 [r430818] asanders <asanders@localhost>:
-
- * res/ari/resource_bridges.c: ARI: Fixed crash that occurred when
- updating a bridge when the optional query parameter 'name' was
- not supplied. Prior to this changeset, posting to the:
- /ari/bridges/{bridgeId} endpoint without specifying a value for
- the [name] query parameter, would crash Asterisk if the bridge
- you are attempting to create (or update) had the same ID as an
- existing bridge. The internal mechanism of the POST operation
- interpreted a null value for name, thus resulting in an error
- condition that crashed Asterisk. ASTERISK-24560 #close Reported
- By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/
-
-2015-01-20 16:46 +0000 [r430817] Richard Mudgett <rmudgett@digium.com>
-
- * configs/samples/iax.conf.sample, res/res_fax.c,
- funcs/func_channel.c, UPGRADE.txt, res/snmp/agent.c,
- channels/chan_iax2.c: CHANNEL(peer), chan_iax2, res_fax, SNMP
- agent: Fix deadlock from reaching across a bridge. Calling
- ast_channel_bridge_peer() cannot be done while holding any
- channel locks. The reported issue hit the deadlock in chan_iax2,
- but an audit of the ast_channel_bridge_peer() calls found three
- more locations where the same deadlock can occur. * Made
- CHANNEL(peer), res_fax, and the SNMP agent not call
- ast_channel_bridge_peer() with any channel locked. For
- CHANNEL(peer) I had to rework the logic to not hold the channel
- lock. * Made chan_iax2 no longer call ast_channel_bridge_peer().
- It was done for legacy reasons that no longer apply. * Removed
- the iax.conf forcejitterbuffer option. It is now always enabled
- when the jitterbuffer option is enabled. If you put a jitter
- buffer on a channel it will be on the channel. ASTERISK-24600
- #close Reported by: Jeff Collell Review:
- https://reviewboard.asterisk.org/r/4342/
-
-2015-01-20 02:39 +0000 [r430796-430799] Matthew Jordan <mjordan@digium.com>
-
- * contrib/scripts/install_prereq, /:
- contrib/scripts/install_prereq: Don't install 32-bit packages on
- 64-bit hosts On Debian based systems, the install_prereq tool
- uses a search command on Debian that results in selecting both
- 64-bit and 32-bit packages. Besides the waste of disk space, this
- can actually cause aptitude use 100% of memory on a VM with 1GB
- of RAM as it tried to work out all of the 32-bit package
- dependencies. This patch filters out the 32-bit packages on a
- 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048
- #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan
- patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang
- (License 5876) ........ Merged revisions 430798 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * apps/app_voicemail.c, /: app_voicemail: Temp message left after
- review/hangup with ODBC/IMAP backend When using ODBC or IMAP
- storage, temporary files created on the file system must be
- disposed of using the DISPOSE macro. The DELETE macro will map to
- a deletion function for the backend storage, but does not clean
- up any local files created as a result of the operation. When
- using voicemail with the operator and review options enabled,
- pressing 0 to enter the menu, followed by 1 to save the message,
- followed by any other DTMF press to delete the message, will
- result in the temporary file lingering on the file system. This
- patch properly calls DISPOSE after the DELETE. This causes the
- local file to be disposed of. ASTERISK-24288 #close Reported by:
- LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU
- (License 6640) ........ Merged revisions 430795 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-19 18:05 +0000 [r430776] Mark Michelson <mmichelson@digium.com>
-
- * main/pbx.c: Call extension state callbacks at hint creation. When
- a hint gets created, any subsequent device or presence state
- changes result in extension status events getting sent out to
- interested parties. However, at the time of hint creation, no
- such event gets sent out, so watchers of extension state are
- potentially left in the dark until the first state change after
- hint creation. Patch contributed by John Hardin (License #6512)
-
-2015-01-19 13:18 +0000 [r430755] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_multihomed.c, res/res_pjsip.c: res_pjsip /
- res_pjsip_multihomed: Use the correct transport and addressing
- information on UAS sessions. The first thing this patch fixes is
- UAS dialogs. Previously if a transport was configured on an
- endpoint and an inbound session was created there was no
- guarantee that requests sent on the dialog would use the correct
- transport and address information. This has now been fixed so an
- explicitly configured transport is taken into account. The second
- thing this patch fixes is res_pjsip_multihomed. The
- res_pjsip_multihomed module attempts to determine what transport
- a message should go out on and what addressing information should
- go into the message itself. In a scenario where multiple
- transports exist bound to the same IP address but a different
- port the code would incorrectly alter the transport and change
- the message to the wrong transport. This change makes the
- res_pjsip_multihomed module smarter so it will only change the
- transport and address information in the message when it is
- possible and makes sense. ASTERISK-24615 #close Reported by:
- David Justl Review: https://reviewboard.asterisk.org/r/4331/
-
-2015-01-17 00:31 +0000 [r430734] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip/config_transport.c,
- res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c,
- main/stasis_message_router.c, res/res_pjsip/location.c,
- res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_distributor.c,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip/pjsip_global_headers.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip.c: REVERTING
- res_pjsip: make it unloadable Due to the original patch causing
- memory corruptions the patch is being removed until the problem
- can be resolved.
-
-2015-01-16 22:13 +0000 [r430709-430716] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES: Change PJProject version requirement for ca_list_path
- transport option in CHANGES file.
-
- * channels/chan_pjsip.c, res/res_pjsip_session.c: Fix problem where
- a hung channel could occur on a failed blind transfer. Different
- clients react differently to being told that a blind transfer has
- failed. Some will simply send a BYE and be done with it. Others
- will attempt to reinvite themselves back onto the call. In the
- latter case, we were creating a new channel and then leaving it
- to sit forever doing nothing. With this code change, that new
- channel will not be created and the dialog with the transferring
- channel will be cleaned up properly. ASTERISK-24624 #close
- Reported by Zane Conkle Review:
- https://reviewboard.asterisk.org/r/4339
-
- * include/asterisk/res_pjsip.h, configure,
- include/asterisk/autoconfig.h.in, configure.ac,
- configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
- res/res_pjsip/config_transport.c: Add support for the
- ca_list_path option for PJSIP transports. This allows for a path
- to be specified that has a collection of CA certificates in it.
- ASTERISK-24575 #close Reported by cloos Patches:
- pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review:
- https://reviewboard.asterisk.org/r/4344
-
-2015-01-15 17:35 +0000 [r430685-430687] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_fax_spandsp.c, res/res_fax.c: res_fax.c,
- res_fax_spandsp.c: Remove redundant locking. When FAX was
- developed, apparently the faxregistry.container used to be a
- linked list that was converted to an ao2 container. Some of the
- replacement ao2 container operations still had explicit
- lock/unlocks around them. Three off nominal code paths in
- res_fax.c and res_fax_spandsp.c unlock the channel even though
- the routine did not lock the channel and other code paths in the
- routine do not unlock the channel. Review:
- https://reviewboard.asterisk.org/r/4340/
-
- * res/res_fax_spandsp.c, res/res_fax.c: res_fax.c,
- res_fax_spandsp.c: Fix some curlies on the end of function
- definitions.
-
-2015-01-15 12:09 +0000 [r430664] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Fix race condition when
- reloading and listing registrations. Due to the split of outbound
- registration state from configuration it is possible during a
- reload for a "pjsip show registrations" CLI command to be
- executed which gets an older snapshot of the configuration. This
- configuration may include outbound registrations which have been
- removed due to a reload operation occurring at the same time. The
- code for printing the outbound registration did not take this
- into account but now it does. AST-1506 #close Review:
- https://reviewboard.asterisk.org/r/4338/
-
-2015-01-15 02:18 +0000 [r430646] Matthew Jordan <mjordan@digium.com>
-
- * configure, configure.ac: configure: If cross-compiling, assume we
- have working semaphores The Asterisk 13 configure.ac checks for
- HAS_WORKING_SEMAPHORE but does not have an option for
- cross-compiling so it fails with an exit. Since we're cross-
- compiling, we can't exactly go looking for the header. The
- semaphore.h header is relatively common: * It's part of the POSIX
- standard * It's part of GNU C Library As such, we assume that it
- will be present when cross-compiling. As such, this patch
- defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling is
- detected. If you're cross-compiling to a platform that doesn't
- support this, then make sure you re-define this to 0.
- ASTERISK-24663 #close Reported by: abelbeck patches:
- asterisk-13-anonymous-semaphores.patch uploaded by abelbeck
- (License 5903)
-
-2015-01-14 23:14 +0000 [r430628] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_distributor.c,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip/pjsip_global_headers.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
- res/res_pjsip/config_transport.c,
- res/res_pjsip/pjsip_outbound_auth.c, res/res_pjsip/config_auth.c,
- main/stasis_message_router.c, res/res_pjsip/location.c:
- res_pjsip: make it unloadable The res_pjsip module was previously
- unloadable. With this patch it can now be unloaded. This patch is
- based off the original patch on the issue (listed below) by Corey
- Farrell with a few modifications. Namely, removed a few changes
- not required to make the module unloadable and also fixed a bug
- that would cause asterisk to crash on unloading. This patch is
- the first step (should hopefully be followed by another/others at
- some point) in allowing res_pjsip and the modules that depend on
- it to be unloadable. At this time, res_pjsip and some of the
- modules that depend on res_pjsip cannot be unloaded without
- causing problems of some sort. The goal of this patch is to get
- res_pjsip and only res_pjsip to be able to unload successfully
- and/or shutdown without incident (crashes, leaks, etc...). Other
- dependent modules may still cause problems on unload. Basically
- made sure, with the patch applied, that res_pjsip (with no other
- dependent modules loaded) could be succesfully unloaded and
- Asterisk could shutdown without any leaks or crashes that
- pertained directly to res_pjsip. ASTERISK-24485 #close Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4311/ patches:
- pjsip_unload-broken-r1.patch submitted by Corey Farrell (license
- 5909)
-
-2015-01-14 20:27 +0000 [r430608] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_outbound_publish.c: Prevent slow graceful shutdown
- when outbound publications never started. The code was missing
- the case for explicitly destroying an outbound publication when
- Asterisk had never actually published anything. The result was
- that Asterisk would hang for a while on a graceful shutdown. With
- this change, the case is taken into account, and on a graceful
- shutdown, these publications are destroyed without the need to
- actually send a PUBLISH request. ASTERISK-24655 #close Reported
- by Kevin Harwell Review: https://reviewboard.asterisk.org/r/4325
-
-2015-01-14 15:39 +0000 [r430590] Matthew Jordan <mjordan@digium.com>
-
- * build_tools/mkpkgconfig, /: build_tools/mkpkgconfig: Fix Cflags
- concatenation error in asterisk.pc The mkpkgconfig script
- incorrectly concatenates Cflags options together. As an example,
- the following: Cflags: -I/usr/include/libxml2 -g3 Is instead
- generated as: Cflags: -I/usr/include/libxml2-g3 This patch
- corrects the generation of Cflags in mkpkgconfig such that the
- Cflags options are output correctly. Review:
- https://reviewboard.asterisk.org/r/3707/ ASTERISK-23991 #close
- Reported by: Diederik de Groot patches: fix_mkpkgconfig.diff
- uploaded by Diederik de Groot (License 6600) ........ Merged
- revisions 430589 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-13 18:16 +0000 [r430565] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_macro.c, /: app_macro: Don't restore the calling
- location on a channel redirect. v11: If a channel redirect to a
- macro exten of a macro that is active happens, the redirect
- location doesn't get executed. Instead the original macro
- location is restored and gets reexecuted. v13: An additional
- effect happens if a parked call times out to an extension in the
- macro that parked the call then the macro is reexecuted instead
- of the expected park return location. * Made not restore the
- macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. *
- Increased the locked channel range when setting up the macro
- execution environment to cover things that should be done while
- the channel is locked. * Removed unnecessary NULL tests before
- calling ast_free() in _macro_exec(). ASTERISK-23850 #close
- Reported by: Andrew Nagy Review:
- https://reviewboard.asterisk.org/r/4292/ ........ Merged
- revisions 430564 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-13 12:06 +0000 [r430546] Joshua Colp <jcolp@digium.com>
-
- * channels/pjsip/dialplan_functions.c, configure,
- include/asterisk/autoconfig.h.in, configure.ac: chan_pjsip: Add
- configure check for 'pjsip_get_dest_info' function. The
- 'pjsip_get_dest_info' function is used to determine if the
- signaling transport of the dialog is secure or not. This function
- was added in PJSIP 2.3 and does not exist in earlier versions.
- This configure check allows Asterisk to build and run with older
- versions at the loss of the 'secure' argument for the PJSIP
- CHANNEL dialplan function. Usage of this argument will require
- upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark
- Michelson Review: https://reviewboard.asterisk.org/r/4329/
-
-2015-01-12 18:34 +0000 [r430528] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/manager.h, main/manager.c: AMI: Revert
- non-backwards compatible changes from earlier commit. * Reverted
- the change to astman_send_listack() to not use the listflag
- parameter and always set the value to "Start" so the start
- capitalization is consistent. Unfortunately changing the case of
- a returned value is not a backward compatible change so for now
- FAXSessions is going to have to remain inconsistent with all of
- the other AMI list actions. * Reverted the minor protocol error
- fix in action_getconfig() when no requested categories are found.
- Each line needs to be formatted as "Header: text". Caught by the
- testsuite. ASTERISK-24049
-
-2015-01-12 18:28 +0000 [r430488-430526] Matthew Jordan <mjordan@digium.com>
-
- * configs/samples/features.conf.sample:
- configs/samples/features.conf.sample: Document attended transfer
- DTMF options The sample config was missing the configuration
- options for DTMF attended transfer completion scenarios. The
- configuration options 'atxferabort', 'atxfercomplete',
- 'atxferthreeway', and 'atxferswap' are now documented in the
- appropriate configuration file. ASTERISK-24678 #close Reported
- by: Niklas Larsson patches: features.conf.sample.diff uploaded by
- Niklas Larsson (License 5068)
-
- * main/syslog.c, include/asterisk/syslog.h, /: main/syslog: Allow
- dynamic logs, such as security events, to log to the syslog The
- security event log uses a dynamic log level (SECURITY) that is
- registered with the Asterisk logging core. Unfortunately, the
- syslog would ignore log statements that had a dynamic log level
- associated with them. Because the syslog cannot handle ad hoc
- dynamic log levels, this patch treats any dynamic log entries
- sent to the syslog as logs with a level of NOTICE. ASTERISK-20744
- #close Reported by: Michael Keuter Tested by: Michael L. Young,
- Jacek Konieczny patches:
- asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by
- Michael L. Young (license 5026) ........ Merged revisions 430506
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
- * funcs/func_curl.c, /: funcs/func_curl: Fix memory leak when
- CURLOPT channel datastore is destroyed When the channel datastore
- associated with the usage of CURLOPT on a specific channel is
- freed, the underlying structure holding the list of options is
- not disposed of. This patch properly frees the structure in the
- datastore .destroy callback. ASTERISK-24672 #close Reported by:
- Kristian Hogh patches: func_curl-memory-leak.diff uploaded by
- Kristian Hogh (License 6639) ........ Merged revisions 430487
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-09 22:08 +0000 [r430467-430469] Scott Griepentrog <sgriepentrog@digium.com>
-
- * contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
- contrib/scripts/sip_to_pjsip/astconfigparser.py: sip_to_pjsip:
- improve ability to parse input files General improvements to SIP
- to PJSIP conversion utility: 1) track default section of input
- file to allow parsing an include file that doesn't specify a
- [section] 2) informatively handle case of assignment without
- [section] 3) correctly handle getting sections from included
- files - [section]'s are inherited by included file 4) provide
- null string as default transport bind ip 5) gracefully handle
- missing portions of registration string 6) denote steps of
- operation during conversion and confirm top level files as a
- convenience ASTERISK-24474 #close Review:
- https://reviewboard.asterisk.org/r/4280/ Reported by: John
- Kiniston
-
- * main/features.c: app_bridge: return to the next dialplan priority
- When app_bridge grabs a channel and puts it into a bridge, the
- channel should then continue where it left off in the dialplan
- after the bridge has ended. Although it stores the current
- dialplan location as an after bridge goto on the channel, it was
- executing the same priority again instead of going to the next
- priority. By swapping the "specific" version of
- bridge_set_after_goto with bridge_set_after_go_on, the next
- priority in the dialplan is executed instead. ASTERISK-24637
- #close Review: https://reviewboard.asterisk.org/r/4322/ Reported
- by: John Bigelow
-
-2015-01-09 17:54 +0000 [r430434] Richard Mudgett <rmudgett@digium.com>
-
- * UPGRADE.txt, res/res_mwi_external_ami.c, CHANGES,
- include/asterisk/manager.h, channels/chan_iax2.c,
- apps/app_queue.c, apps/app_agent_pool.c,
- res/res_manager_devicestate.c, main/manager_bridges.c,
- channels/chan_dahdi.c, main/manager.c, channels/chan_skinny.c,
- res/res_pjsip_outbound_registration.c,
- res/res_manager_presencestate.c,
- res/res_pjsip/pjsip_configuration.c, apps/app_confbridge.c,
- res/res_pjsip_pubsub.c, main/db.c, res/parking/parking_manager.c,
- res/res_pjsip_registrar.c, apps/app_voicemail.c, main/pbx.c,
- channels/chan_sip.c, apps/app_meetme.c, main/bridge.c,
- res/res_fax.c: AMI: Make AMI actions that generate event lists
- consistent. * Made the following AMI actions use list API calls
- for consistency: Agents BridgeInfo BridgeList
- BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms
- CoreShowChannels DAHDIShowChannels DBGet DeviceStateList
- ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers
- IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls
- Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints
- PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound
- PJSIPShowResourceLists PJSIPShowSubscriptionsInbound
- PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans
- QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus
- SIPshowregistry SKINNYdevices SKINNYlines Status
- VoicemailUsersList * Incremented the AMI version to 2.7.0. *
- Changed astman_send_listack() to not use the listflag parameter
- and always set the value to "Start" so the start capitalization
- is consistent. i.e., The FAXSessions used "Start" while the rest
- of the system used "start". The corresponding complete event
- always used "Complete". * Fixed ami_show_resource_lists()
- "PJSIPShowResourceLists" to output the AMI ActionID for all of
- its list events. * Fixed off-nominal AMI protocol error in
- manager_bridge_info(), manager_parking_status_single_lot(), and
- manager_parking_status_all_lots(). Use of astman_send_error()
- after responding to the original AMI action request violates the
- action response pattern by sending two responses. * Fixed minor
- protocol error in action_getconfig() when no requested categories
- are found. Each line needs to be formatted as "Header: text". *
- Fixed off-nominal memory leak in
- manager_build_parked_call_string(). * Eliminated unnecessary use
- of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close
- Reported by: Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/4315/
-
-2015-01-09 14:51 +0000 [r430416] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_fax.c, include/asterisk/res_fax.h,
- configs/samples/res_fax.conf.sample, CHANGES: res_fax: Add T.38
- negotiation timeout option This change makes the T.38 negotiation
- timeout configurable via 't38timeout' in res_fax.conf or
- FAXOPT(t38timeout). It was previously hard coded to be 5000
- milliseconds. This change also handles T.38 switch failures by
- aborting the fax since in the case where this can happen, both
- sides have agreed to switch to T.38 and Asterisk is unable to do
- so. Review: https://reviewboard.asterisk.org/r/4320/ ........
- Merged revisions 430415 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2015-01-08 21:40 +0000 [r430373-430397] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Fix persistent
- subscriptions not surviving graceful shutdown If you do a 'core
- (shutdown|restart) graceful' persistent subscriptions won't
- survive. If you do a 'core (shutdown|restart) now' or asterisk
- terminates for some reason, they do. Here's why... When asterisk
- shuts down gracefully, it sends a 'NOTIFY/terminated' to
- subscribers for each subscription. This not only tells the
- subscribers that the dialog/state machine is done, it also frees
- the last reference to the subscription tree which causes the
- persistent subscription to get deleted from astdb. When asterisk
- restarts, nothing's left. Just preventing the delete from astdb
- doesn't work because we already told the subscriber to terminate
- the dialog so we can't restart it even if it was still in astdb.
- Everything works OK if asterisk terminates unexpectedly because
- we never send the 'terminated' message so on restart, the
- subscription is still in astdb and the subscriber is none the
- wiser. This patch suppresses the sending of 'NOTIFY/terminated'
- on shutdown for persistent connections. Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4318/
-
- * res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Fix reference leak. Every time a
- registration started, sip_outbound_registration_response_cb bumps
- the ref count on client_state then pushes a
- handle_registration_response task. handle_registration_response
- never unreffed it though. So every time a registration goes out,
- the ref count goes up by one. This patch adds the unreffs to
- handle_registration_response. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4303/
-
- * res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Fix several reload issues There
- are 2 issues with reloading registrations... 1. The
- 'can_reuse_registration' test wasn't considering the intervals or
- expiration in its determination of whether a registration changed
- or not so if you changed any of the intervals or the expiration
- and reloaded, the object would get reloaded but the actual timers
- wouldn't change. can_reuse_registration now does a sorcery diff
- on the old and new objects instead of discretely testing certain
- fields. Now if you change expiration for instance, and reload,
- the timer is updated and re-registration will occur on the new
- value. 2. If you mung up your password on an outbound
- registration you get a permanent failure. If you fix the password
- (on the outbound_auth object) and reload, nothing tells
- outbound_registration to try again because the registration
- itself didn't change. This patch adds an observer on the "auth"
- object type and if any auth changes, existing registration states
- are searched and those in a REJECTED_PERMANENT state are retried.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4304/
-
-2015-01-07 21:25 +0000 [r430355] Kinsey Moore <kmoore@digium.com>
-
- * res/res_stasis.c: ARI: Allow usage of ASYNCGOTO with Stasis()
- When the AMI Redirect action is used with a channel bridged
- inside Stasis() and not running a pbx, the channel is hung up
- instead of proceeding to the desired location in dialplan. This
- change allows such channels to be Redirected properly by
- detecting the operation used by Redirect (ASYNCGOTO) and using
- the code already established for functionality of the ARI channel
- continue operation. ASTERISK-24591 #close Review:
- https://reviewboard.asterisk.org/r/4271/
-
-2015-01-07 18:53 +0000 [r430337] Mark Michelson <mmichelson@digium.com>
-
- * rest-api/api-docs/channels.json, rest-api/resources.json,
- res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c,
- res/ari/resource_channels.h: Add the ability to continue and
- originate using priority labels. With this patch, the following
- two ARI commands POST /channels POST /channels/{id}/continue
- Accept a new parameter, label, that can be used to continue to or
- originate to a priority label in the dialplan. Because this is
- adding a new parameter to ARI commands, the API version of ARI
- has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of
- Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412
- #close Reported by Nir Simionovich Review:
- https://reviewboard.asterisk.org/r/4285
-
-2015-01-07 18:17 +0000 [r430315-430319] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_exten_state.c: res_pjsip_exten_state: Change 'does
- not exist' warning to notice The 'new_subscribe: Extension <>
- does not exist or has no associated hint' is a config issue and
- doesn't need to clutter up logs with warnings. Changed to notice.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4307/
-
- * res/res_pjsip_mwi.c: res_pjsip_mwi: Change "MWI Subscription
- failed" message from warning to notice The "MWI Subscription
- failed" message means the client is trying to subscribe to a
- mailbox that doesn't exist. There's no need to clutter up logs
- with warnings for a client misconfiguration so I changed it to a
- notice. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4306/
-
- * funcs/func_config.c, tests/test_config.c: func_config: Add
- ability to retrieve specific occurrence of a variable I guess
- nobody uses templates with AST_CONFIG because today if you have a
- context that inherits from a template and you call AST_CONFIG on
- the context, you'll get the value from the template even if
- you've overridden it in the context. This is because AST_CONFIG
- only gets the first occurrence which is always from the template.
- This patch adds an optional 'index' parameter to AST_CONFIG which
- lets you specify the exact occurrence to retrieve, or '-1' to
- retrieve the last. The default behavior is the current behavior.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4313/
-
-2015-01-07 17:35 +0000 [r430313] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_refer.c: Fix ability to perform a remote attended
- transfer with PJSIP. This fix has two parts: * Corrected an error
- message to properly state that external_replaces is an extension.
- The error message also prints what dialplan context the
- external_replaces extension was being looked for in. * Corrected
- the printing of the Replaces: header in an INVITE request. We
- were duplicating "Replaces: " in the header. ASTERISK-24376
- #close Reported by Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4296
-
-2015-01-07 16:55 +0000 [r430295] George Joseph <george.joseph@fairview5.com>
-
- * include/asterisk/config.h, main/config.c, main/manager.c: config:
- Add option to NOT preserve effective context when changing a
- template Let's say you have a template T with variable VAR1 = ON
- and you have a context C(T) that doesn't specify VAR1. If you
- read C, the effective value of VAR1 is ON. Now you change T VAR1
- to OFF and call ast_config_text_file_save. The current behavior
- is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is
- added. Personally, I think this is a bug. It's preserving the
- effective state of C even though I didn't specify C/VAR1 in th
- first place. I believe the behavior should be that if I didn't
- specify C/VAR1 originally, then the effective value of C/VAR1
- should continue to follow the inherited state. Now, if I DID
- explicitly specify C/VAR1, the it should be preserved even if the
- template changes. Even though I think the existing behavior is a
- bug, it's been that way forever so I'm not changing it. Instead,
- I've created ast_config_text_file_save2() that takes a bitmask of
- flags, one of which is to preserve the effective context (the
- current behavior). The original ast_config_text_file_save calls
- *2 with the preserve flag. If you want the new behavior, call *2
- directly without a flag. I've also updated Manager UpdateConfig
- with a new parameter 'PreserveEffectiveContext' whose default is
- 'yes'. If you want the new behavior with UpdateConfig, set
- 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4297/
-
-2015-01-07 02:52 +0000 [r430274] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
- main/rtp_engine.c: Fix dev-mode build on recent gcc
-
-2015-01-06 22:46 +0000 [r430252] Matthew Jordan <mjordan@digium.com>
-
- * contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py:
- contrib/ast-db-manage: Correct down_revision path for
- user_eq_phone When the user_eq_phone patch was backported to 13,
- it referenced the downward revision that the PJSIP optimistic
- encryption option also references. This creates a multi-path
- upgrade Exception when generating the SQL files. This patch
- corrects this in the 13 branch. Note that trunk, which already
- contained both of these features, is unaffected by this problem.
-
-2015-01-06 17:52 +0000 [r430221-430227] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_mwi.c: res_pjsip_mwi: Change warning to notice When
- res_pjsip loads and an endpoint auto-subscribes a mailbox for
- mwi, if a contact hasn't registered yet, res_pjsip_mwi spits out
- a warning. This is a perfectly normal situation though and
- doesn't require something as serious as a warning. It's also self
- correcting. The device will start getting mwi as soon as it
- registers. This patch changes the warning to a notice. Tested-by:
- George Joseph Review: https://reviewboard.asterisk.org/r/4314/
-
- * bridges/bridge_native_rtp.c: bridge_native_rtp: Change
- local/remote message from debug/2 to verb/4 Change the "Locally
- bridged"/"Remotely bridged" messages from dbg/2 to verb/4.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4300/
-
- * res/res_pjsip_outbound_registration.c, CHANGES:
- outbound_registration: Add 'pjsip send register' and update 'send
- unregister' The current behavior of 'pjsip send unregister' is to
- send the unregister (REGISTER with 0 exp) but let the next
- scheduled register proceed normally. I don't think that's a good
- idea. If you unregister, it should stay unregistered until you
- decide to start registrations again. So this patch just adds a
- cancel_registration call to the current unregister_task to cancel
- the timer. Of course, now you need a way to start registration
- again so I've added a 'pjsip send register' command that
- unregisters and cancels any existing registration (the same as
- send unregister), then sends an immediate registration and starts
- the timer back up again. Both changes also ripple to AMI. There's
- a new PJSIPRegister command. There's no harm in calling either
- command repeatedly. They don't care about the actual state.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4301/
-
- * res/res_pjsip/location.c: pjsip cli: Fix sorting of contacts for
- 'pjsip list contacts' For some reason I was using a hash
- container instead of a list to gather the contacts for 'pjsip
- list/show contacts' so even though I had a sort function, the
- output wasn't sorted. This patch just changes the hash container
- to a list container and the contacts now appear sorted in the
- CLI. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4305/
-
-2015-01-05 22:49 +0000 [r430200] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/bridge_basic.c: bridge: avoid leaking channel during
- blond transfer pt2 A blond transfer to a failed destination, when
- followed by a recall attempt, lead to a leak of the reference to
- the destination channel. In addition to correcting the regression
- on the previous attempt (r429826) this fixes the leak and two
- additional reference leaks on failures of bridge_import.
- ASTERISK-24513 #close Review:
- https://reviewboard.asterisk.org/r/4302/ ........ Merged
- revisions 430199 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2015-01-05 17:56 +0000 [r430179-430181] Joshua Colp <jcolp@digium.com>
-
- * CHANGES: pjsip: Document addition of 'PJSIP_AOR' and
- 'PJSIP_CONTACT' in CHANGES file.
-
- * funcs/func_pjsip_contact.c (added), res/res_pjsip_session.c,
- include/asterisk/res_pjsip.h,
- channels/pjsip/dialplan_functions.c,
- include/asterisk/res_pjsip_session.h, funcs/func_pjsip_aor.c
- (added), res/res_pjsip/location.c: pjsip: Add 'PJSIP_AOR' and
- 'PJSIP_CONTACT' dialplan functions. The PJSIP_AOR dialplan
- function allows inspection of configured AORs including what
- contacts are currently bound to them. The PJSIP_CONTACT dialplan
- function allows inspection of contacts in existence. These can
- include both externally added (by way of registration) or
- permanent ones. ASTERISK-24341 Reported by: xrobau Review:
- https://reviewboard.asterisk.org/r/4308/
-
-2014-12-29 13:10 +0000 [r430145] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip.c: PJSIP: Update transport method documentation
- This updates the documentation for the 'method' configuration
- option to be more verbose about the behaviors of values
- 'unspecified' and 'default'. They do exactly the same thing which
- is to select the default as defined by PJSIP which is currently
- TLSv1. Review: https://reviewboard.asterisk.org/r/4264/
-
-2014-12-24 21:27 +0000 [r430127] Kevin Harwell <kharwell@digium.com>
-
- * /, configs/samples/queues.conf.sample: app_queue: Update sample
- conf documenation Updated the queues.conf.sample file to
- explicitly state which channel queue variables are propagated to.
- ASTERISK-24267 Reported by: Mitch Claborn ........ Merged
- revisions 430126 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-24 15:26 +0000 [r430083-430092] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip.c,
- contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
- (added): res_pjsip: Backport missing commits for user_eq_phone
- This backports the following from trunk, which were missed:
- r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2
- lines res_pjsip: Allow + at the beginning of a phone number when
- user_eq_phone is enabled. r427259 | file | 2014-11-04 16:51:32
- -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Apply the
- 'user_eq_phone' setting to the To header as well. It also adds
- the Alembic script for the option. ASTERISK-24643
-
- * CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h,
- res/res_pjsip/config_global.c, res/res_pjsip_keepalive.c (added),
- configs/samples/pjsip.conf.sample: res_pjsip_keepalive: Add
- runtime configurable keepalive module for connection-oriented
- transports. Note that this is backport from trunk of r425825.
- This change adds a module which is configurable using the
- keep_alive_interval setting in the global section that will send
- a CRLF keep alive to all active connection-oriented transports at
- the provided interval. This is useful because it can help keep
- connections open through NATs. This functionality also exists
- within PJSIP but can not be controlled at runtime and requires
- recompiling it. Review: https://reviewboard.asterisk.org/r/4084/
- ASTERISK-24644 #close
-
- * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_caller_id.c,
- CHANGES, res/res_pjsip.c, include/asterisk/res_pjsip.h:
- res_pjsip: Add 'user_eq_phone' option to add a 'user=phone'
- parameter when applicable. Note that this is a backport of
- r425804 from trunk. This change adds a configuration option which
- adds a 'user=phone' parameter if the user portion of the request
- URI or the From URI is determined to be a number. Review:
- https://reviewboard.asterisk.org/r/4073/ ASTERISK-24643 #close
-
-2014-12-23 23:18 +0000 [r430059-430064] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip/pjsip_options.c: pjsip_options: Fix continued
- qualifies after endpoint/aor deletion If you remove an
- endpoint/aor from pjsip.conf then do a core reload, qualifies
- will continue even though the object are gone. This happens
- because nothing clears out the qualify tasks. This patch
- unschedules all existing qualify tasks before scheduling new ones
- on reload. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4290/
-
- * tests/test_astobj2.c: test_astobj2: Fix warning for missing
- trailing slash in category This patch adds a trailing slash to
- the category for this test. No more warning. Tested-by: George
- Joseph Review: https://reviewboard.asterisk.org/r/4295/
-
-2014-12-22 21:18 +0000 [r430010-430034] Richard Mudgett <rmudgett@digium.com>
-
- * main/bridge_basic.c: DTMF atxfer: Setup recall channels as if the
- transferee initiated the call. After the initial DTMF atxfer call
- attempt to the transfer target fails to answer during a blonde
- transfer, the recall callback channels do not get setup with
- information from the initial transferrer channel. As a result,
- the recall callback to the transferrer does not have callid,
- channel variables, datastores, accountcode, peeraccount, COLP,
- and CLID setup. A similar situation happens with the recall
- callback to the transfer target but it is less visible. The
- recall callback to the transfer target does not have callid,
- channel variables, datastores, accountcode, peeraccount, and COLP
- setup. * Added missing information to the recall callback
- channels before initiating the call. callid, channel variables,
- datastores, accountcode, peeraccount, COLP, and CLID * Set callid
- of the transferrer channel on the DTMF atxfer controller thread
- attended_transfer_monitor_thread(). * Added missing channel
- unlocks and props unref to off nominal paths in
- attended_transfer_properties_alloc(). ASTERISK-23841 #close
- Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/4259/
-
- * /, main/logger.c, include/asterisk/_private.h, main/asterisk.c:
- queue_log: Post QUEUESTART entry when Asterisk fully boots. The
- QUEUESTART log entry has historically acted like a fully booted
- event for the queue_log file. When the QUEUESTART entry was
- posted to the log was broken by the change made by
- ASTERISK-15863. * Made post the QUEUESTART queue_log entry when
- Asterisk fully boots. This restores the intent of that log entry
- and happens after realtime has had a chance to load. AST-1444
- #close Reported by: Denis Martinez Review:
- https://reviewboard.asterisk.org/r/4282/ ........ Merged
- revisions 430009 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-22 15:40 +0000 [r429983] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Send CANCEL via original INVITE
- destination even after UPDATE request Given the following
- scenario: * Three SIP phones (A, B, C), all communicating via a
- proxy with Asterisk * A call is established between A and B. B
- performs a SIP attended transfer of A to C. B sets the call on
- hold (A is hearing MOH) and dials the extension of C. While phone
- C is ringing, B transfers the call (that is, what we typically
- call a 'blond transfer'). * When the transfer completes, A hears
- the ringing of phone C, while B is idle. In the SIP messaging for
- the above scenario, a REFER request is sent to transfer the call.
- When "sendrpid=yes" is set in sip.conf, Asterisk may send an
- UPDATE request to phone C to update party information. This
- update is sent directly to phone C, not through the intervening
- proxy. This has the unfortunate side effect of providing route
- information, which is then set on the sip_pvt structure for C. If
- someone (e.g. B) is trying to get the call back (through a
- directed pickup), Asterisk will send a CANCEL request to C.
- However, since we have now updated the route set, the CANCEL
- request will be sent directly to C and not through the proxy. The
- phone ignores this CANCEL according to RFC3261 (Section 9.1).
- This patch updates reqprep such that the route is not updated if
- an UPDATE request is being sent while the INVITE state is
- INV_PROCEEDING or INV_EARLY_MEDIA. This ensures that a subsequent
- CANCEL request is still sent to the correct location. Review:
- https://reviewboard.asterisk.org/r/4279 ASTERISK-24628 #close
- Reported by: Karsten Wemheuer patches: issue.patch uploaded by
- Karsten Wemheuer (License 5930) ........ Merged revisions 429982
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-22 00:17 +0000 [r429914] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_phoneprov_provider.c:
- res_pjsip_phoneprovi_provider: Fix reload Reloading wasn't
- working correctly because on a reload, the sorcery apply handler
- was never being called for unchanged users. So, instead of using
- an apply handler, I'm now iterating over all users. Works much
- more reliably. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4288/
-
-2014-12-20 20:57 +0000 [r429894] Joshua Colp <jcolp@digium.com>
-
- * main/named_acl.c, /: acl: Fix reloading of configuration if
- configuration file does not exist at startup. The named ACL code
- incorrectly destroyed the config options information if loading
- of the configuration file failed at startup. This would result in
- reloading also failing even if a valid configuration file was put
- in place. ASTERISK-23733 #close Reported by: Richard Kenner
- ........ Merged revisions 429893 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-19 20:54 +0000 [r429829-429868] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_http_websocket.c: res_http_websocket.c: Fix incorrect
- use of sizeof in ast_websocket_write(). This won't fix the
- reported issue but it is an incorrect use of sizeof.
- ASTERISK-24566 Reported by: Badalian Vyacheslav ........ Merged
- revisions 429867 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * channels/chan_dahdi.c, /: chan_dahdi: Don't ignore setvar when
- using configuration section scheme. When the configuration
- section scheme of chan_dahdi.conf is used (keyword dahdichan
- instead of channel) all setvar= options are completely ignored.
- No variable defined this way appears in the created DAHDI
- channels. * Move the clearing of setvar values to after the
- deferred processing of dahdichan. AST-1378 #close Reported by:
- Guenther Kelleter Patch by: Guenther Kelleter ........ Merged
- revisions 429825 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-19 17:26 +0000 [r429827] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/bridge_basic.c: bridge: avoid leaking channel during
- blond transfer After a blond transfer (start attended and hang
- up) to a destination that also hangs up without answer, the
- Local;1 channel was leaked and would show up on core show
- channels. This was happening because the attended state
- blond_nonfinal_enter() resetting the props->transfer_target to
- null while releasing it's own reference, which would later
- prevent props from releasing another reference during
- destruction. The change made here is simply to not assign the
- target to NULL. ASTERISK-24513 #close Reported by: Mark Michelson
- Review: https://reviewboard.asterisk.org/r/4262/ ........ Merged
- revisions 429826 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-18 22:38 +0000 [r429784-429805] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_rtp_asterisk.c, channels/chan_dahdi.c, /: chan_dahdi.c,
- res_rtp_asterisk.c: Change some spammy debug messages to level 5.
- ASTERISK-24337 #close Reported by: Rusty Newton ........ Merged
- revisions 429804 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
- * UPGRADE.txt, channels/sig_analog.c, /: chan_dahdi: Populate
- CALLERID(ani2) for incoming calls in featdmf signaling mode. For
- the featdmf signaling mode the incoming MF Caller-ID information
- is formatted as follows:
- *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}# Rather than
- discarding the ani2 digits, populate the CALLERID(ani2) value
- with what is received instead. AST-1368 #close Reported by: Denis
- Martinez Patches: extract_ani2_for_featdmf_v11.patch (license
- #5621) patch uploaded by Richard Mudgett ........ Merged
- revisions 429783 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-18 15:50 +0000 [r429763] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: wrong bridge chosen
- when the DTMF mode is not compatible A native rtp bridge was
- being chosen (it shouldn't have been) when using two pjsip
- channels with incompatible DTMF modes. This patch sets the rtp
- instance property, AST_RTP_PROPERTY_DTMF, for the appropriate
- DTMF mode(s) for pjsip. It was not being set before, meaning all
- DTMF modes for pjsip were being treated as compatible, thus
- native bridging would be chosen as the bridge type when it
- shouldn't have been. ASTERISK-24459 #close Reported by: Yaniv
- Simhi Review: https://reviewboard.asterisk.org/r/4265/
-
-2014-12-18 15:34 +0000 [r429739-429761] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_outbound_registration.c: Prevent potential infinite
- outbound authentication loops in registration. Prior to this
- patch, Asterisk would always respond to 401 responses to
- registration attempts by trying to provide a registration with
- authentication credentials. Even if subsequent attempts were
- rejected with 401 responses, Asterisk would continue this
- behavior. If authentication credentials were incorrect, this
- could continue forever. With this patch, we keep track of whether
- we have attempted authentication on an outbound registration
- attempt. If we already have, we don not try again until the next
- attempt. This prevents the infinite loop scenario. Review:
- https://reviewboard.asterisk.org/r/4273
-
- * main/manager.c: Prevent possible race condition on dual redirect
- of channels in the same bridge. The
- AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent
- bridges from prematurely acting on orphaned channels in bridges.
- The problem with the AMI redirect action was that it was setting
- this flag on channels based on the presence of a PBX, not whether
- the channel was in a bridge. Whether a channel has a PBX is
- irrelevant, so the condition has been altered to check if the
- channel is in a bridge. ASTERISK-24536 #close Reported by Niklas
- Larsson Review: https://reviewboard.asterisk.org/r/4268
-
- * channels/pjsip/dialplan_functions.c: Ensure the correct value is
- returned for CHANNEL(pjsip, secure) Prior to this patch, we were
- using the PJSIP dialog's secure flag to determine if a secure
- transport was being used. Unfortunately, the dialog's secure flag
- was only set if a SIPS URI were in use, as required by RFC 3261
- sections 12.1.1 and 12.1.2. What we're interested in is not
- dialog security, but transport security. This code change
- switches to a model where we use the dialog's target URI to
- determine what transport would be used to communicate, and then
- check if that transport is secure. AST-1450 #close Reported by
- John Bigelow Review: https://reviewboard.asterisk.org/r/4277
-
-2014-12-18 00:10 +0000 [r429699-429719] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: fix
- unload SEGV If certain pjsip modules aren't loaded, the wizard
- causes a SEGV when it unloads. Added a check for the presense of
- the object type wizard before trying to clean it up. Tested-by:
- George Joseph
-
- * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: Change
- FILEUNCHANGED config_load2 flag determination The module now
- applies the FILEUNCHANGED flag when both reloaded is specified
- AND there's no last_config for the object type. Tested-by: George
- Joseph Review: https://reviewboard.asterisk.org/r/4276/
-
-2014-12-17 09:54 +0000 [r429675] Walter Doekes <walter+asterisk@wjd.nu>
-
- * addons/ooh323c/src/printHandler.c, apps/app_adsiprog.c,
- channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c, /,
- channels/chan_sip.c, channels/vcodecs.c, res/res_crypto.c,
- utils/astman.c, utils/smsq.c, main/utils.c, pbx/dundi-parser.c,
- apps/app_getcpeid.c, channels/chan_iax2.c, channels/sig_pri.c,
- res/res_pktccops.c, main/loader.c, channels/iax2/parser.c,
- main/uuid.c, main/manager.c, channels/chan_misdn.c,
- apps/app_osplookup.c, channels/misdn/ie.c, main/http.c,
- apps/app_sms.c: Fix printf problems with high ascii characters
- after r413586 (1.8). In r413586 (1.8) various casts were added to
- silence gcc 4.10 warnings. Those fixes included things like: -out
- += sprintf(out, "%%%02X", (unsigned char) *ptr); +out +=
- sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii
- characters, but for the high range that yields e.g. FFFFFFC3 when
- C3 is expected. This changeset: - fixes those casts to use the
- 'hh' unsigned char modifier instead - consistently uses %02x
- instead of %2.2x (or other non-standard usage) - adds a few 'h'
- modifiers in various places - fixes a 'replcaes' typo -
- dev/urandon typo (in 13+ patch) Review:
- https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close
- Reported by: Stefan27 (on IRC) ........ Merged revisions 429673
- from http://svn.asterisk.org/svn/asterisk/branches/11 ........
- Merged revisions 429674 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-16 17:53 +0000 [r429653] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_config_wizard.c: res_pjsip_config_wizard: fix test
- breakage Fix test breakage caused by not checking for res_pjsip
- before calling ast_sip_get_sorcery. Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4269/
-
-2014-12-16 16:38 +0000 [r429612-429633] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Allow T.38 switch-over when
- SRTP is in use. Previously when SRTP was enabled on a channel it
- was not possible to switch to T.38 as no crypto attributes would
- be present. This change makes it so it is now possible. If a T.38
- re-invite comes in SRTP is terminated since in practice you can't
- encrypt a UDPTL stream. Now... if we were doing T.38 over RTP
- (which does exist) then we'd have a chance but almost nobody does
- that so here we are. ASTERISK-24449 #close Reported by: Andreas
- Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by
- Andreas Steinmetz (license 6523) ........ Merged revisions 429632
- from http://svn.asterisk.org/svn/asterisk/branches/11
-
- * res/res_pjsip_t38.c: res_pjsip_t38: Fix T.38 failure when peer
- reinvites immediately. If a remote endpoint reinvites to T.38
- immediately the state machine will go into a peer reinvite state.
- If a T.38 capable application (such as ReceiveFax) queries it
- will receive this state. Normally the application will then
- indicate so that the channel driver will queue up the T.38 offer
- previously received. Once it receives this offer the application
- will act normally and negotiate. The res_pjsip_t38 module
- incorrectly partially squashed this indication. This would cause
- the application to think the request had failed when in reality
- it had actually worked. This change makes it so that no T.38
- control frames (or indications) are squashed.
-
-2014-12-15 17:07 +0000 [r429592] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_phoneprov_provider.c,
- configs/samples/pjsip_wizard.conf.sample (added), CHANGES,
- res/res_pjsip_config_wizard.c (added): res_pjsip_config_wizard:
- Allow streamlined config of common pjsip scenarios
- res_pjsip_config_wizard ------------------ * This is a new module
- that adds streamlined configuration capability for chan_pjsip.
- It's targetted at users who have lots of basic configuration
- scenarios like 'phone' or 'agent' or 'trunk'. Additional
- information can be found in the sample configuration file at
- config/samples/pjsip_wizard.conf.sample. Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4190/
-
-2014-12-15 15:36 +0000 [r429571] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c: Activate persistent subscriptions when
- they are recreated. Prior to this change, recreating persistent
- subscriptions would create the subscription but would not
- activate it. This led to subscriptions being listed in the "NULL"
- state by diagnostics and not sending NOTIFYs when expected.
- Review: https://reviewboard.asterisk.org/r/4261
-
-2014-12-12 23:54 +0000 [r429542] George Joseph <george.joseph@fairview5.com>
-
- * main/manager.c, include/asterisk/module.h,
- include/asterisk/_private.h: loader: Move definition of
- ast_module_reload from _private.h to module.h No functionality
- change. Just move the definition of ast_module_reload from
- _private.h to module.h so it can be public. Also removed the
- include of _private.h from manager.c since ast_module_load was
- the only reason for including it. Tested-by: George Joseph
- Review: https://reviewboard.asterisk.org/r/4251/
-
-2014-12-12 23:40 +0000 [r429540] Richard Mudgett <rmudgett@digium.com>
-
- * main/lock.c, /, include/asterisk/lock.h: DEBUG_THREADS: Fix
- regression and lock tracking initialization problems. This patch
- started with David Lee's patch at
- https://reviewboard.asterisk.org/r/2826/ and includes a
- regression fix introduced by the ASTERISK-22455 patch. The
- initialization of a mutex's lock tracking structure was not
- protected in a critical section. This is fine for any mutex that
- is explicitly initialized, but a static mutex may have its lock
- tracking double initialized if multiple threads attempt the first
- lock simultaneously. * Added a global mutex to properly serialize
- initialization of the lock tracking structure. The painful global
- lock can be mitigated by adding a double checked lock flag as
- discussed on the original review request. * Defer lock tracking
- initialization until first use. * Don't be "helpful" and
- initialize an uninitialized lock when DEBUG_THREADS is enabled.
- Debug code is not supposed to fix or change normal code behavior.
- We don't need a lock initialization race that would force a
- re-setup of lock tracking. Lock tracking already handles
- initialization on first use. * Properly handle allocation
- failures of the lock tracking structure. * No need to initialize
- tracking data in __ast_pthread_mutex_destroy() just to turn
- around and destroy it. The regression introduced by
- ASTERISK-22455 is the result of manipulating a pthread_mutex_t
- struct outside of the pthread library code. The pthread_mutex_t
- struct seems to have a global linked list pointer member that can
- get changed by other threads. Therefore, saving and restoring the
- contents of a pthread_mutex_t struct is a bad thing. Thanks to
- Thomas Airmont for finding this obscure regression. * Don't
- overwrite the struct ast_lock_track.reentr_mutex member to
- restore tracking data in __ast_cond_wait() and
- __ast_cond_timedwait(). The pthread_mutex_t struct must be
- treated as a read-only opaque variable. Miscellaneous other items
- fixed by this patch: * Match ast_suspend_lock_info() with
- ast_restore_lock_info() in __ast_cond_timedwait(). * Made some
- uninitialized lock sanity checks return EINVAL and try a
- DO_THREAD_CRASH. * Fix bad canlog initialization expressions.
- ASTERISK-24614 #close Reported by: Thomas Airmont Review:
- https://reviewboard.asterisk.org/r/4247/ Review:
- https://reviewboard.asterisk.org/r/2826/ ........ Merged
- revisions 429539 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-12-12 22:53 +0000 [r429518-429519] Matthew Jordan <mjordan@digium.com>
-
- * res/res_agi.c: res/res_agi: Make Verbose message for 'stream
- file' match other playbacks The Verbose message displayed when a
- file is played back via 'stream file' was formatted differently
- than other playbacks: * It didn't include the channel name * It
- didn't include the channel language It does, however, include the
- playback offset as well as any escape digits. That information
- was kept; however, this patch updates the formatting to more
- closely match the Verbose messages displayed when a file is
- played back by 'control stream file', Playback, ControlPlayback,
- or any other file playback operation.
-
- * /: Add 11 merge properties
-
-2014-12-12 16:57 +0000 [r429497] Joshua Colp <jcolp@digium.com>
-
- * main/format.c, main/codec.c, include/asterisk/format.h: media:
- Fix crash when determining sample count of a frame during
- shutdown. When shutting down Asterisk the codecs are cleaned up.
- As a result anything attempting to get a codec based on ID or
- details will find that no codec exists. This currently occurs
- when determining the sample count of a frame. This code did not
- take this situation into account. This change fixes this by
- getting the codec directly from the format and eliminates the
- lookup. This is both faster and also provides a guarantee that
- the codec will exist and will be valid. ASTERISK-24604 #close
- Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4260/
-
-2014-12-12 15:30 +0000 [r429477] Kevin Harwell <kharwell@digium.com>
-
- * channels/chan_pjsip.c: chan_pjsip: Race between channel answer
- and bridge setup when using direct media When direct media is
- enabled and a pjsip channel is answered a race would occur
- between the handling of the answer and bridge setup. Sometimes
- the media negotiation would take place after the native bridge
- was setup. This resulted in a NULL media address, which in turn
- resulted in Asterisk using its address as the remote media
- address when sending a reinvite. This patch makes the chan_pjsip
- answer handler synchronous thus alleviating the race condition
- (the bridge won't start setting things up until after it
- returns). ASTERISK-24563 #close Reported by: Steve Pitts Review:
- https://reviewboard.asterisk.org/r/4257/
-
-2014-12-12 15:00 +0000 [r429457] David M. Lee <dlee@digium.com>
-
- * res/res_pjsip_outbound_publish.c: Fix crash for sorcery
- misconfigs res_pjsip_outbound_publish was missing the
- CHECK_PJSIP_MODULE_LOADED() call in load_module, and would crash
- with a segfault if res_pjsip declined to load. Review:
- https://reviewboard.asterisk.org/r/4258/
-
-2014-12-12 14:12 +0000 [r429430-429433] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive'
- streams for hold This allows use of the 'inactive' stream
- direction identifier to be used for hold where 'sendonly' is
- normally used. Some Seimens phones use 'inactive' and this change
- allows music on hold to operate properly. Review:
- https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts
- ........ Merged revisions 429432 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_sorcery_config.c, /: Sorcery: Log when old config remains
- in use This adds a log message notifying the user that a stale
- configuration is in place upon reload when a config object fails
- to load. This situation can end up causing confusion when the
- object failed to load but exists from a previous config load
- especially when the old config is significantly different from
- the new config. Review: https://reviewboard.asterisk.org/r/4250/
- Reported by: Thomas Thompson ........ Merged revisions 429429
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-12 13:05 +0000 [r429407-429409] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.exports.in, channels/chan_pjsip.c,
- res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h:
- res_pjsip_session: Delay sending BYE if a re-INVITE transaction
- is in progress. Given the scenario where a PJSIP channel is in a
- native RTP bridge with direct media and the channel is then hung
- up the code will currently re-INVITE the channel back to Asterisk
- and send a BYE at the same time. Many SIP implementations dislike
- this greatly. This change makes it so that if a re-INVITE
- transaction is in progress the BYE is queued to occur after the
- completion of the transaction (be it through normal means or a
- timeout). Review: https://reviewboard.asterisk.org/r/4248/
-
- * res/res_pjsip_session.c: res_pjsip_session: Fix issue where a
- declined media stream in a re-INVITE would fail SDP negotiation.
- In the past the SDP negotiation within res_pjsip_session was made
- more tolerant of certain situations. The only case where SDP
- negotiation will fail is when a major error occurs during
- negotiation. Receiving an already declined media stream is not
- considered a major error. When producing the local SDP the logic
- took this into account so on the initial INVITE the declined
- media stream did not cause an SDP negotiation failure.
- Unfortunately the logic for handling media streams with a handler
- did not mirror this logic and considered an already declined
- media stream an error and thus failed the SDP negotiation. This
- change makes the logic between both situations match so only
- under major errors will the SDP negotiation fail. ASTERISK-24607
- #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4254/
-
-2014-12-11 20:31 +0000 [r429387] Kevin Harwell <kharwell@digium.com>
-
- * CHANGES: ARI/AMI: Include language in standard channel snapshot
- output The CHANGES verbiage for the "language" addition had been
- put under the wrong release. This moves it to be under 13.1 to
- 13.2 changes. ASTERISK-24553 Reported by: Matt Jordan
-
-2014-12-11 17:21 +0000 [r429352-429379] Kinsey Moore <kmoore@digium.com>
-
- * /: Recorded merge of revisions 429378 from
- http://svn.asterisk.org/svn/asterisk/branches/12 ........ Fix
- incorrect patch applied in r429354 The patch that was applied was
- another pending patch. This swaps them out.
-
- * /: Recorded merge of revisions 429354 from
- http://svn.asterisk.org/svn/asterisk/branches/12 ........ Stasis:
- Update unittest for channel snapshots This adjusts the unit test
- for channel snapshots to take the new language key into account.
-
- * tests/test_stasis_channels.c: Stasis: Update unittest for channel
- snapshots This adjusts the unit test for channel snapshots to
- take the new language key into account.
-
-2014-12-10 15:42 +0000 [r429326] Kevin Harwell <kharwell@digium.com>
-
- * /, CHANGES: ARI/AMI: Include language in standard channel
- snapshot output Adding information about including "language" in
- the standard channel snapshot output to the CHANGES file. Note
- the actual source changes have already been previously committed.
- ASTERISK-24553 Reported by: Matt Jordan ........ Merged revisions
- 429325 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-10 13:34 +0000 [r429273] Joshua Colp <jcolp@digium.com>
-
- * res/res_http_websocket.c, res/res_pjsip_transport_websocket.c, /,
- channels/chan_sip.c: res_http_websocket: Fix crash due to double
- freeing memory when receiving a payload length of zero. Frames
- with a payload length of 0 were incorrectly handled in
- res_http_websocket. Provided a frame with a payload had been
- received prior it was possible for a double free to occur. The
- realloc operation would succeed (thus freeing the payload) but be
- treated as an error. When the session was then torn down the
- payload would be freed again causing a crash. The read function
- now takes this into account. This change also fixes assumptions
- made by users of res_http_websocket. There is no guarantee that a
- frame received from it will be NULL terminated. ASTERISK-24472
- #close Reported by: Badalian Vyacheslav Review:
- https://reviewboard.asterisk.org/r/4220/ Review:
- https://reviewboard.asterisk.org/r/4219/ ........ Merged
- revisions 429270 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 429272 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-10 13:14 +0000 [r429246] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip/pjsip_options.c: PJSIP: Fix assert on initial
- mass qualify This fixes the MWI test regressions caused by
- r429127 and ensures that contacts have non-zero qualify_frequency
- before attempting scheduling. ........ Merged revisions 429245
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-09 20:46 +0000 [r429223] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/asterisk.c: core: avoid possible asterisk -r crash from long
- id When connecting to the remote console, an id string is first
- provided that consts of the hostname, pid, and version. This is
- parsed by the remote instance using a buffer that may be too
- short, and can allow a buffer overrun because it is not
- terminated. This patch adds termination and a larger buffer.
- Review: https://reviewboard.asterisk.org/r/4182/
-
-2014-12-09 20:19 +0000 [r429175-429206] Kevin Harwell <kharwell@digium.com>
-
- * res/ari/ari_model_validators.h, /, main/stasis_channels.c,
- rest-api/api-docs/channels.json, res/ari/ari_model_validators.c,
- main/manager_channels.c: ARI/AMI: Include language in standard
- channel snapshot output The channel "language" was already part
- of a channel snapshot, however is was not sent out over AMI or
- ARI. This patch makes it so the channel "language" is included in
- the appropriate AMI or ARI events. ASTERISK-24553 #close Reported
- by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4245/
- ........ Merged revisions 429204 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
- main/rtp_engine.c, /, channels/chan_sip.c: Direct Media calls
- within private network sometimes get one way audio When endpoints
- with direct_media enabled, behind a firewall (Asterisk on a
- separate network) and were bridged sometimes Asterisk would send
- the ip address of the firewall in the sdp to one of the phones in
- the reinvite resulting in one way audio. When sending the
- reinvite Asterisk will retrieve the media address from the
- associated rtp instance, but if frames were being read this can
- be overwritten with another address (in this case the
- firewall's). This patch ensures that Asterisk uses the original
- device address when using direct media. ASTERISK-24563 Reported
- by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/
- ........ Merged revisions 429195 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_outbound_publish.c: res_pjsip_outbound_publish:
- stack overflow when using non-default sorcery wizard When using a
- non-default sorcery wizard (in this instance realtime) for
- outbound publishes Asterisk will crash after a stack overflow
- occurs due to the code infinitely recursing. The fix entails
- removing the outbound publish state dependency from the outbound
- publish sorcery object and instead keeping an in memory container
- that can be used to lookup the state when needed. ASTERISK-24514
- #close Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/4178/
-
-2014-12-09 15:44 +0000 [r429153] Joshua Colp <jcolp@digium.com>
-
- * res/ari/resource_channels.h, rest-api/api-docs/channels.json,
- res/ari/resource_channels.c, CHANGES, res/res_ari_channels.c:
- ari: Add support for specifying an originator channel when
- originating. If an originator channel is specified when
- originating a channel the linked ID of it will be applied to the
- newly originated outgoing channel. This allows an association to
- be made between the two so it is known that the originator has
- dialed the originated channel. ASTERISK-24552 #close Reported by:
- Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/
-
-2014-12-09 14:00 +0000 [r429128] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip/pjsip_options.c: PJSIP: Stagger outbound
- qualifies This change staggers initiation of outbound qualify
- (OPTIONS) attempts to reduce instantaneous server load and
- prevent network congestion. Review:
- https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close
- Reported by: Richard Mudgett ........ Merged revisions 429127
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-15 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.1.0 Released.
-
-2014-12-10 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.1.0-rc2 Released.
-
- * AST-2014-019: Fix crash when receiving a WebSocket packet with a
- payload length of zero.
-
- Frames with a payload length of 0 were incorrectly handled in
- res_http_websocket. Provided a frame with a payload had been
- received prior it was possible for a double free to occur. The
- realloc operation would succeed (thus freeing the payload) but be
- treated as an error. When the session was then torn down the payload
- would be freed again causing a crash. The read function now takes
- this into account.
-
- This change also fixes assumptions made by users of
- res_http_websocket. There is no guarantee that a frame received from
- it will be NULL terminated.
-
- ASTERISK-24472 #close
- Reported by: Badalian Vyacheslav
-
-2014-12-08 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.1.0-rc1 Released.
-
-2014-12-08 16:53 +0000 [r429091] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/playbacks.json, UPGRADE.txt,
- rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/resources.json, CHANGES, include/asterisk/manager.h,
- rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json,
- rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
- rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json: AMI/ARI: Update version to
- 2.6.0/1.6.0 respectively for new features AMI/ARI are getting a
- few enhancements in the next release of Asterisk 13. Per semantic
- versioning, that warrants a bump in the minor version number, as
- it reflects a backwards compatible change. Hence, this commit.
-
-2014-12-08 16:41 +0000 [r429064-429089] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_session.c: Fix a crash that would occur when
- receiving a 491 response to a reinvite. The reviewboard
- description does a fine job of summarizing this, so here it is: A
- reporter discovered that Asterisk would crash when attempting to
- retransmit a reinvite that had previously received a 491
- response. The crash occurred because a pjsip_tx_data structure
- was being saved for reuse, but its reference count was not being
- increased. The result was that the pjsip_tx_data was being freed
- before we were actually done with it. When we attempted to re-use
- the structure when re-sending the reinvite, Asterisk would crash.
- The fix implemented here is not to try holding onto the
- pjsip_tx_data at all. Instead, when we reschedule sending the
- reinvite, we create a brand new pjsip_tx_data and send that
- instead. Because of this change, there is no need for an
- ast_sip_session_delayed_request structure to have a pjsip_tx_data
- on it any more. So any code referencing its use has been removed.
- When this initial fix was introduced, I encountered a second
- crash when processing a subsequent 200 OK on a rescheduled
- reinvite. The reason was that when rescheduling the reinvite, we
- gave the wrong location for a response callback. This has been
- fixed in this patch as well. ASTERISK-24556 #close Reported by
- Abhay Gupta Review: https://reviewboard.asterisk.org/r/4233
-
- * main/stasis_channels.c, CHANGES, res/ari/ari_model_validators.c,
- main/manager_channels.c, main/channel.c,
- res/ari/ari_model_validators.h,
- include/asterisk/stasis_channels.h,
- rest-api/api-docs/events.json, res/stasis/app.c: Add new AMI and
- ARI events for connected line changes on a channel. The AMI event
- is called NewConnectedLine and the ARI event is called
- ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt
- Jordan Review: https://reviewboard.asterisk.org/r/4231
-
-2014-12-08 15:43 +0000 [r429062] Kinsey Moore <kmoore@digium.com>
-
- * /, res/stasis/app.c, main/channel_internal_api.c,
- res/stasis/stasis_bridge.c, res/stasis/app.h,
- include/asterisk/channel.h, res/res_stasis.c, main/channel.c:
- Stasis: Fix StasisStart/End order and missing events This
- corrects several bugs that currently exist in the stasis
- application code. * After a masquerade, the resulting channels
- have channel topics that do not match their uniqueids **
- Masquerades now swap channel topics appropriately * StasisStart
- and StasisEnd messages are leaked to observer applications due to
- being published on channel topics ** StasisStart and StasisEnd
- publishing is now properly restricted to controlling apps via app
- topics * Race conditions exist where StasisStart and StasisEnd
- messages due to a masquerade may be received out of order due to
- being published on different topics ** These messages are now
- published directly on the app topic so this is now a non-issue *
- StasisEnds are sometimes missing when sent due to masquerades and
- bridge swaps into and out of Stasis() ** This was due to
- StasisEnd processing adjusting message-sent flags after Stasis()
- had already exited and Stasis() had been re-entered ** This was
- corrected by adjusting these flags prior to sending the message
- while the initial Stasis() application was still shutting down
- Review: https://reviewboard.asterisk.org/r/4213/ ASTERISK-24537
- #close Reported by: Matt DiMeo ........ Merged revisions 429061
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-06 18:16 +0000 [r429029-429033] Matthew Jordan <mjordan@digium.com>
-
- * res/res_monitor.c, /: res/res_monitor: Reset in/out sample counts
- on Monitor start When repeatedly starting/stopping a Monitor on a
- channel, the accumulated in/out sample counts are never reset to
- 0. This can cause inadvertent jumps in the recordings, as the
- code in the channel core will determine incorrectly that a jump
- in the recorded file position should occur. Setting the sample
- counts to 0 simply reflects the initial state a Monitor should be
- in when it is started, as this is the initial count that would be
- on the channels at that time. ASTERISK-24573 #close Reported by:
- Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License
- 6116) ........ Merged revisions 429031 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 429032 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_meetme.c: apps/app_meetme: Apply default values on
- initial load with no config file When the app_meetme module is
- loaded without its configuration file, the module settings aren't
- initialized. In particular, this impacts the use of logging
- realtime members. This patch guarantees that we always set the
- default module settings on initial load. Review:
- https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close
- Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno
- Borges (License 6116) ........ Merged revisions 429027 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 429028 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-05 17:06 +0000 [r429000] George Joseph <george.joseph@fairview5.com>
-
- * tests/test_sorcery.c, main/sorcery.c, include/asterisk/test.h, /,
- include/asterisk/sorcery.h: sorcery: Add additional observer
- capabilities. Add new global, instance and wizard observers.
- instance_created wizard_registered wizard_unregistered
- instance_destroying instance_loading instance_loaded
- wizard_mapped object_type_registered object_type_loading
- object_type_loaded wizard_loading wizard_loaded Tested-by: George
- Joseph Review: https://reviewboard.asterisk.org/r/4215/ ........
- Merged revisions 428999 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-04 17:13 +0000 [r428865-428973] Matthew Jordan <mjordan@digium.com>
-
- * /, main/test.c: main/test: Fix compilation issue on 32-bit
- systems On a 32-bit system, a type of intmax_t will result in a
- compilation warning when formatted as a 'long int'. Use the
- format specifier of %jd (which was what was used originally in
- manager.c) to format the JSON extracted integer on both
- 32-/64-bit systems. ........ Merged revisions 428972 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /, main/test.c: main/test: Fix race condition
- between AMI topic and Test Suite topic This patch fixes a race
- condition between the raising of test AMI events (which drive
- many tests in the Asterisk Test Suite) and other AMI events.
- Prior to this patch, the Stasis messages published to the test
- topic were not forwarded to the AMI topic. Instead, the code in
- manager had a dedicated handler for test messages that was
- independent of the topics forwarded to the AMI topic. This
- results in no synchronization between the test messages and the
- rest of the Stasis messages published out over AMI. In some test
- with very tight timing constraints, this can result in out of
- order messages and spurious test failures. Properly forwarding
- the Test Suite topic to the AMI topic ensures that the messages
- are synchronized properly. This patch does that, and moves the
- message handling to the Stasis definition of the Test Suite
- message in test.c as well. Review:
- https://reviewboard.asterisk.org/r/4221/ ........ Merged
- revisions 428945 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_cel.c, /: tests/test_cel: Add
- test_cel_attended_transfer_bridges_link to racey tests Despite
- failing less often, the ordering of the ATTENDEDTRANSFER event
- and the BRIDGE_EXIT event for the Alice and David channels is not
- defined. This makes the test still fail. ........ Merged
- revisions 428918 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_cel.c, /: tests/test_cel: Fix CEL unit test failures
- caused by attended transfer changes When the publication of
- attended transfer messages were pushed to another thread, some
- subtle race conditions were introduced with the CEL unit tests.
- This patch fixes one of them, and pushes the other to
- ASTERISK-22367, which already exists to fix another bouncy CEL
- unit test. In particular, this patch fixes the
- test_cel_attended_transfer_bridges_link test, and defers the
- test_cel_attended_transfer_bridges_swap test to the
- aforementioned JIRA issue. ASTERISK-22367 ........ Merged
- revisions 428891 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_voicemail.c, /: apps/app_voicemail: Fix crash with IMAP
- when streams are opened simultaneously The UW IMAP library is
- instrinsically not thread-safe, and relies upon higher level
- applications to guarantee thread safety. For the most part, this
- is provided by the vms object, which provides locking for
- individual streams. Unfortunately, this is not sufficient for
- calls to mail_open which create the IMAP stream. mail_open can,
- on some systems, call into a UW IMAP specific function for
- determining the address of a system based on a hostname,
- ip_nametoaddr. In the ip6_unix implementation of this function,
- static variables are used to hold parsing buffers. This can cause
- a crash if multiple threads attempt to convert a hostname to an
- address at the same time. Locking on a single mail stream is not
- sufficient to prevent simultaneous access to these static
- variables. In the IMAP library, this function can be called from
- the mail_open and imap_status functions. As the imap_status
- function is not used by app_voicemail, locking on access to
- mail_open is sufficient to prevent any mangling of the buffers.
- Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516
- #close Reported by: David Duncan Ross Palmer Tested by: David
- Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David
- Duncan Ross Palmer (License 6660) ........ Merged revisions
- 428863 from http://svn.asterisk.org/svn/asterisk/branches/11
- ........ Merged revisions 428864 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-02 21:53 +0000 [r428837] George Joseph <george.joseph@fairview5.com>
-
- * CHANGES, /: CHANGES: Add item for new 'pjsip show identif(y|ies)
- commands Tested-by: George Joseph ........ Merged revisions
- 428836 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-02 19:03 +0000 [r428789-428815] Matthew Jordan <mjordan@digium.com>
-
- * tests/test_stasis.c: tests/test_stasis: Resolve compilation
- issues from Asterisk 12 merge When merging the changes up stream
- in r428687, I missed the fact that the signature for
- stasis_message_type_create was changed. This patch fixes the
- compilation issues introduced by that merge.
-
- * pbx/pbx_loopback.c, /: pbx/pbx_loopback: Speed up switches by
- avoiding unneeded lookups This patch makes a small rearrangement
- to only do dialplan lookups during loopback switches if the
- pattern matches. Prior to this patch, the dialplan lookups were
- always performed, even when the result would be discarded.
- Dialplan lookups can be very costly if remote switches - like
- DUNDi - are present. In those cases extension matching is sped up
- considerably, making the issue of lost digits more manageable. As
- collateral damage, 6 trailing spaces were killed. Review:
- https://reviewboard.asterisk.org/r/4211 ASTERISK-24577 #close
- Reported by: Birger Harzenetter patches: ast-loopback.patch
- uploaded by Birger Harzenetter (License 5870) ........ Merged
- revisions 428787 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428788 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-02 12:20 +0000 [r428761] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_refer.c, /: res_pjsip_refer: Fix issue where native
- bridge may not occur upon completion of a transfer. There are two
- methods within res_pjsip_refer for keeping track of the state of
- a transfer. The first is a framehook which looks at frames
- passing by to determine the state. The second subscribes to know
- when the channel joins a bridge. In the case when the channel
- joins the bridge the framehook is *NOT* removed and this prevents
- the native RTP bridging technology from getting used. This change
- gets the channel and if it still exists remove the framehook.
- Review: https://reviewboard.asterisk.org/r/4218/ ........ Merged
- revisions 428760 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-02 00:38 +0000 [r428731-428734] George Joseph <george.joseph@fairview5.com>
-
- * /, include/asterisk/config.h, main/config.c: config: Create
- ast_variable_find_in_list() Add const char
- *ast_variable_find_in_list(const struct ast_variable *list, const
- char *variable); ast_variable_find() requires a config category
- to search whereas ast_variable_find_in_list() just needs the root
- list element which is useful if you don't have a category.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4217/ ........ Merged
- revisions 428733 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_endpoint_identifier_ip.c,
- res/res_pjsip/pjsip_cli.c: res_pjsip_endpoint_identifier_ip: Add
- 'show identify(ies)' cli commands While troubleshooting other
- things I realized there were no pjsip cli commands for identify.
- This patch adds them. It also also fixes a reference leak when a
- 'show endpoint' displayed identifies and properly sets the return
- code if load_module can't allocate a cli formatter structure.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4212/ ........ Merged
- revisions 428725 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-01 17:57 +0000 [r428687] Matthew Jordan <mjordan@digium.com>
-
- * channels/chan_skinny.c, res/res_pjsip_mwi.c, tests/test_stasis.c,
- res/res_pjsip_pubsub.c, res/res_pjsip_refer.c,
- channels/chan_mgcp.c, main/stasis_cache.c, channels/chan_sip.c,
- include/asterisk/stasis_internal.h, /, include/asterisk/stasis.h,
- UPGRADE.txt, configs/samples/stasis.conf.sample,
- res/parking/parking_applications.c, res/res_xmpp.c,
- channels/chan_iax2.c, apps/app_queue.c,
- res/res_stasis_device_state.c, channels/sig_pri.c,
- include/asterisk/stasis_message_router.h, main/endpoints.c,
- res/parking/parking_bridge_features.c, main/stasis.c,
- channels/chan_dahdi.c, main/stasis_message_router.c: main/stasis:
- Allow subscriptions to use a threadpool for message delivery
- Prior to this patch, all Stasis subscriptions would receive a
- dedicated thread for servicing published messages. In contrast,
- prior to r400178 (see review
- https://reviewboard.asterisk.org/r/2881/), the subscriptions
- shared a thread pool. It was discovered during some initial work
- on Stasis that, for a low subscription count with high message
- throughput, the threadpool was not as performant as simply having
- a dedicated thread per subscriber. For situations where a
- subscriber receives a substantial number of messages and is
- always present, the model of having a dedicated thread per
- subscriber makes sense. While we still have plenty of
- subscriptions that would follow this model, e.g., AMI, CDRs, CEL,
- etc., there are plenty that also fall into the following two
- categories: * Large number of subscriptions, specifically those
- tied to endpoints/peers. * Low number of messages. Some
- subscriptions exist specifically to coordinate a single message -
- the subscription is created, a message is published, the delivery
- is synchronized, and the subscription is destroyed. In both of
- the latter two cases, creating a dedicated thread is wasteful
- (and in the case of a large number of peers/endpoints, harmful).
- In those cases, having shared delivery threads is far more
- performant. This patch adds the ability of a subscriber to Stasis
- to choose whether or not their messages are dispatched on a
- dedicated thread or on a threadpool. The threadpool is
- configurable through stasis.conf. Review:
- https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close
- Reported by: xrobau Tested by: xrobau ........ Merged revisions
- 428681 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-12-01 13:41 +0000 [r428632-428655] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_record.c: app_record: Fix bug where using the 'k'
- option and hanging up would trim 1/4 of a second of the
- recording. The Record dialplan function trims 1/4 of a second
- from the end of recordings in case they are terminated because of
- DTMF. When hanging up, however, you don't want this to happen.
- This change makes it so on hangup this does not occur.
- ASTERISK-24530 #close Reported by: Ben Smithurst patches:
- app_record_v2.diff submitted by Ben Smithurst (license 6529)
- Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged
- revisions 428653 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428654 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/channel.c: channel: Extend size of buffer for codecs in
- "core show channeltype" CLI command. The static buffer for codecs
- when invoking the "core show channeltype" CLI command did not
- have enough room for all codecs. This has been extended so it
- does. ASTERISK-24542 #close Reported by: snuffy patches:
- channeltype-tech.diff submitted by snuffy (license 5024) Review:
- https://reviewboard.asterisk.org/r/4204/
-
-2014-11-24 20:37 +0000 [r428602-428604] Richard Mudgett <rmudgett@digium.com>
-
- * tests/test_channel_feature_hooks.c: test_channel_feature_hooks.c:
- Fix unit test for DTMF hooks. Fix the failing
- /channels/features/test_features_channel_dtmf unit test. DTMF
- emulation does not work without a stream of packets to prod the
- emulation code. Review: https://reviewboard.asterisk.org/r/4199/
-
- * /, main/bridge.c, main/bridge_channel.c: DTMF hooks: Leaving
- channels need to push any collected digits into the bridge. Any
- partially collected DTMF digits for a DTMF hook need to be pushed
- into the bridge when a channel leaves the bridging system as if
- there were a timeout. Review:
- https://reviewboard.asterisk.org/r/4199/ ........ Merged
- revisions 428601 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-21 19:09 +0000 [r428572] Richard Mudgett <rmudgett@digium.com>
-
- * main/manager.c, /: manager: Fix could not extend string messages.
- When shutting down Asterisk that has an active AMI connection,
- you get several "failed to extend from %d to %d" messages because
- use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission
- strings to the event. * Created MAX_AUTH_PERM_STRING to use when
- creating stack based struct ast_str variables used with the
- authority_to_str() and user_authority_to_str() functions instead
- of a variety of magic numbers that could be too small. * Added a
- special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it
- will not attempt to add all permission level strings. Review:
- https://reviewboard.asterisk.org/r/4200/ ........ Merged
- revisions 428570 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428571 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-21 17:45 +0000 [r428544] George Joseph <george.joseph@fairview5.com>
-
- * main/sorcery.c, /, res/res_pjsip_phoneprov_provider.c,
- tests/test_sorcery.c: sorcery: Make is_object_field_registered
- handle field names that are regexes. As a result of
- https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime was
- tossing database fields that didn't have an exact match to a
- sorcery registered field. This broke the ability to use regexes
- as field names which manifested itself as a failure of
- res_pjsip_phoneprov_provider which uses this capability. It also
- broke handling of fields that start with '@' in realtime but I
- don't think anyone noticed. This patch does the following... *
- Modifies ast_sorcery_fields_register to pre-compile the name
- regex. * Modifies ast_sorcery_is_object_field_registered to test
- the regex if it exists instead of doing an exact strcmp. *
- Modifies res_pjsip_phoneprov_provider with a few tweaks to get it
- to work with realtime. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4185/ ........ Merged
- revisions 428543 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-21 02:16 +0000 [r428505] Matthew Jordan <mjordan@digium.com>
-
- * main/bridge_basic.c: main/bridge_basic: Fix features regressions
- introduced by r428165 In r428165, two bugs were introduced: *
- Prior to entering the features retry loop, the buffer that holds
- the collected digits is wiped. However, this inadvertently wipes
- out the first collected digit on the first pass through, which is
- obtained in ast_stream_and_wait. This caused all of the features
- tests to fail. * If ast_app_dtget returns a hangup (-1), the loop
- would retry incorrectly. If we detect a hangup, we have to stop
- trying the feature. This patch fixes both issues. Review:
- https://reviewboard.asterisk.org/r/4196/
-
-2014-11-20 16:36 +0000 [r428425] Mark Michelson <mmichelson@digium.com>
-
- * main/acl.c, /: Fix error with mixed address family ACLs. Prior to
- this commit, the address family of the first item in an ACL was
- used to compare all incoming traffic. This could lead to traffic
- of other IP address families bypassing ACLs. ASTERISK-24469
- #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff
- uploaded by Matt Jordan (License #6283) AST-2014-012 ........
- Merged revisions 428402 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 428417 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428422 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-20 16:34 +0000 [r428413] Kevin Harwell <kharwell@digium.com>
-
- * funcs/func_db.c, /: AST-2014-018 - func_db: DB Dialplan function
- permission escalation via AMI. The DB dialplan function when
- executed from an external protocol (for instance AMI), could
- result in a privilege escalation. Asterisk now inhibits the DB
- function from being executed from an external interface if the
- live_dangerously option is set to no. ASTERISK-24534 Reported by:
- Gareth Palmer patches: submitted by Gareth Palmer (license 5169)
- ........ Merged revisions 428331 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 428363 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428409 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-20 16:13 +0000 [r428343] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_acl.c, /: PJSIP ACLs: Fix ACLs not loading on
- startup and apply/acl issues on contact The biggest problem this
- patch fixes is that ACLs weren't previously being loaded when the
- res_pjsip_acl module was loaded. Yikes. In addition, the ACL
- options contact_permit and contact_acl were effectively
- interpreted as contact_deny and this patch fixes that as well.
- AST-1418 #close Reported by: Thomas Thompson Review:
- https://reviewboard.asterisk.org/r/4120/ ASTERISK-24531 #close
- Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4171/ ........ Merged
- revisions 428333 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-20 15:50 +0000 [r428339] Kevin Harwell <kharwell@digium.com>
-
- * apps/app_confbridge.c, /: AST-2014-017 - app_confbridge:
- permission escalation/ class authorization. Confbridge dialplan
- function permission escalation via AMI and inappropriate class
- authorization on the ConfbridgeStartRecord action. The CONFBRIDGE
- dialplan function when executed from an external protocol (for
- instance AMI), could result in a privilege escalation. Also, the
- AMI action “ConfbridgeStartRecord” could also be used to execute
- arbitrary system commands without first checking for system
- access. Asterisk now inhibits the CONFBRIDGE function from being
- executed from an external interface if the live_dangerously
- option is set to no. Also, the “ConfbridgeStartRecord” AMI action
- is now only allowed to execute under a user with system level
- access. ASTERISK-24490 Reported by: Gareth Palmer ........ Merged
- revisions 428332 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428334 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-20 14:55 +0000 [r428302-428305] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_refer.c, /: AST-2014-016: Fix crash when receiving
- an in-dialog INVITE with Replaces in res_pjsip_refer. The
- implementation of INVITE with Replaces in res_pjsip_refer did not
- expect them to occur in-dialog. As a result it would incorrectly
- attempt to hang up a channel it thought was under its control. In
- reality the channel would be under the control of another thread.
- When the other thread accessed the channel it would be accessing
- freed memory and could crash. This change makes res_pjsip_refer
- not act on an in-dialog INVITE with Replaces. ASTERISK-24528
- #close Reported by: Joshua Colp ........ Merged revisions 428304
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_pjsip.c, /: AST-2014-015: Fix race condition in
- chan_pjsip when sending responses after a CANCEL has been
- received. Due to the serialized architecture of chan_pjsip there
- exists a race condition where a CANCEL may be received and
- processed before responses (such as 180 Ringing, 183 Session
- Progress, and 200 OK) are sent. Since the session is in an
- unexpected state PJSIP will assert when this is attempted. This
- change makes it so that these responses are not sent on
- disconnected sessions. ASTERISK-24471 #close Reported by: yaron
- nahum ........ Merged revisions 428301 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-19 19:31 +0000 [r428273] Corey Farrell <git@cfware.com>
-
- * include/asterisk/stringfields.h, /: stringfields: Fix bug in
- ast_string_fields_copy. ast_string_fields_copy relies on the fact
- that __ast_string_field_release_active never previously zeroed
- pool->used, so keeping the existing pointer was "ok". Now that
- existing pools can be reset to 'empty', it is important to set
- each field to __ast_string_field_empty after releasing the
- memory. ASTERISK-24535 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4186/ ........ Merged
- revisions 428272 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-19 17:13 +0000 [r428246] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_calendar.c, main/manager.c, /, channels/chan_sip.c,
- channels/sip/security_events.c: ast_str: Fix improper member
- access to struct ast_str members. Accessing members of struct
- ast_str outside of the string manipulation API routines is
- invalid since struct ast_str is supposed to be treated as opaque.
- Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged
- revisions 428244 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428245 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-19 12:40 +0000 [r428196-428222] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, include/asterisk/res_pjsip.h,
- include/asterisk/res_pjsip_session.h, res/res_pjsip_sdp_rtp.c,
- res/res_pjsip/pjsip_configuration.c,
- configs/samples/pjsip.conf.sample,
- contrib/ast-db-manage/config/versions/eb88a14f2a_add_media_encryption_optimistic_to_pjsip.py
- (added), CHANGES, res/res_pjsip.c: res_pjsip_sdp_rtp: Add support
- for optimistic SRTP. Optimistic SRTP is the ability to enable
- SRTP but not have it be a fatal requirement. If SRTP can be used
- it will be, if not it won't be. This gives you a better chance of
- using it without having your sessions fail when it can't be.
- Encrypt all the things! Review:
- https://reviewboard.asterisk.org/r/3992/
-
- * res/res_pjsip_refer.c, /: res_pjsip_refer: Ensure Refer-To is
- NULL terminated and parse it as a URI. There is no guarantee that
- when we get a Refer-To that it will be NULL terminated. As the
- URI parsing function requires it to be we now NULL terminate it.
- Additionally parsing the Refer-To as a 'To' header is needless
- and it can simply be done as a URI. This also fixes a problem
- where certain Refer-To headers would not be parsed as a 'To'
- header causing the REFER to fail. ASTERISK-24508 #close Reported
- by: Beppo Mazzucato Review:
- https://reviewboard.asterisk.org/r/4187/ ........ Merged
- revisions 428195 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-18 18:54 +0000 [r428169] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/parking/parking_tests.c: parking_tests.c: Add missing
- newline on a unit test message. ........ Merged revisions 428168
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-17 16:51 +0000 [r428145] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES, main/features_config.c,
- configs/samples/features.conf.sample,
- include/asterisk/features_config.h, main/bridge_basic.c: Allow
- for transferer to retry when dialing an invalid extension. This
- allows for a configurable number of attempts for a transferer to
- dial an extension to transfer the call to. For Asterisk 13, the
- default values are such that upgrading between versions will not
- cause a behaivour change. For trunk, though, the defaults will be
- changed to be more user-friendly. Review:
- https://reviewboard.asterisk.org/r/4167
-
-2014-11-17 16:00 +0000 [r428119] Corey Farrell <git@cfware.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix theoretical leak of
- p->refer. If transmit_refer is called when p->refer is already
- allocated, it leaks the previous allocation. Updated code to
- always free previous allocation during a new allocation. Also
- instead of checking if we have a previous allocation, always
- create a clean record. ASTERISK-15242 #close Reported by: David
- Woolley Review: https://reviewboard.asterisk.org/r/4160/ ........
- Merged revisions 428117 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428118 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-17 15:27 +0000 [r428079-428115] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/confbridge/conf_state_multi_marked.c:
- apps/app_confbridge: Ensure 'normal' users hear message when last
- marked leaves When r428077 was made for ASTERISK-24522, it failed
- to take into account users who are neither wait_marked nor
- end_marked. These users are *also* supposed to hear the 'leader
- has left the conference' message. Granted, this behaviour is a
- bit odd; however, that is how it used to work... and behaviour
- changes are not good. This patch ensures that if there are any
- 'normal' users present when the last marked user leaves the
- conference, the message will still be played to them. Note that
- this regression was caught by the Asterisk Test Suite's
- confbridge_nominal test, which has a quirky combination of users.
- ........ Merged revisions 428113 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428114 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge:
- Don't play leader leaving prompt if no one will hear it Consider
- the following: - A marked user in a conference - One or more
- end_marked only users in the conference When the marked users
- leaves, we will be in the conf_state_multi_marked state. This
- currently will traverse the users, kicking out any who have the
- end_marked flags. When they are kicked, a full ast_bridge_remove
- is immediately called on the channels. At this time, we also
- unilaterally set the need_prompt flag. When the need_prompt flag
- is set, we then playback a sound to the bridge informing everyone
- that the leader has left; however, no one is left in the bridge.
- This causes some odd behaviour for the end_marked users - they
- are stuck waiting for the bridge to be unlocked. This results in
- them waiting for 5 or 6 seconds of dead air before hearing that
- they've been kicked. Unfortunately, we do have to keep the bridge
- locked while we're playing back the 'leader-has-left' prompt. If
- there are any wait_marked users in the conference, this behaviour
- can't be easily changed - but we do make the case of the
- end_marked users better with this patch. Review:
- https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close
- Reported by: Matt Jordan ........ Merged revisions 428077 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 428078 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-16 21:12 +0000 [r427979-428052] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_pjsip.c, /: chan_pjsip: Remove AOR check when
- dialing and one is specified. The AOR value may contain the name
- of an AOR or a full SIP URI. Checking if the AOR exists can't be
- done as a result of this. ........ Merged revisions 428051 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_pjsip.c: chan_pjsip: Add additional log message
- when an AOR is specified when dialing and it does not exist.
- ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged
- revisions 428007 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_motif.c, channels/chan_pjsip.c, /: chan_motif /
- chan_pjsip: Fix incorrect "No such module" messages when
- reloading. For chan_motif the direct return value of the
- underlying config options framework was passed back. This can
- relay various states which the module loader would not interpet
- as success. It has been changed so only on errors will it report
- back an error. For chan_pjsip the code implemented a dummy reload
- function which always returned an error. This has been removed as
- all configuration is held within res_pjsip instead.
- ASTERISK-23651 #close Reported by: Rusty Newton ........ Merged
- revisions 427981 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Enforce
- requirements for session timer minimum expiration period and
- normal expiration period. This change enforces the requirements
- in PJSIP for session timer configuration. The minimum expiration
- period must be 90 seconds or higher and the normal expiration
- period can not be lower than the minimum expiration period. If
- either of these were done the code would assert at session setup
- time. ASTERISK-24336 #close Reported by: Leon Rowland ........
- Merged revisions 427978 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-15 16:56 +0000 [r427927-427954] Matthew Jordan <mjordan@digium.com>
-
- * cel/cel_odbc.c, /: cel/cel_odbc: Provide microsecond precision in
- 'eventtime' column when possible This patch adds microsecond
- precision when inserting a CEL record into a table with an
- "eventtime" column of type timestamp, instead of second
- precision. The documentation (configs/cel_odbc.conf.sample) was
- already saying that the eventtime column included microseconds
- precision, but that was not the case. Also, without this patch,
- if you had a table with an "eventtime" column of type varchar,
- you had millisecond precision. With this patch, you also get
- microsecond precision in this case. Review:
- https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close
- Reported by: Etienne Lessard patches:
- cel_odbc_time_precision.patch uploaded by Etienne Lessard
- (License 6394) ........ Merged revisions 427952 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427953 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_cel.c: tests/test_cel: Unlock bridge on off nominal
- paths If the test fails due to memory allocation errors, we may
- as well attempt to unlock the bridge on the way out.
-
-2014-11-14 17:45 +0000 [r427902] Jonathan Rose <jrose@digium.com>
-
- * configs/samples/cdr.conf.sample, main/cdr.c, /: Documentation:
- Revise explanation of cdr.conf option 'Unanswered' ASTERISK-24279
- #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4109/ ........ Merged
- revisions 427901 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-14 15:51 +0000 [r427876] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/stun.c: stun: correct attribute string padding to match
- rfc When sending the USERNAME attribute in an RTP STUN response,
- the implementation in append_attr_string passed the actual
- length, instead of padding it up to a multiple of four bytes as
- required by the RFC 3489. This change adds separate variables for
- the string and padded attributed lengths, and performs padding
- correctly. Reported by: Thomas Arimont Review:
- https://reviewboard.asterisk.org/r/4139/ ........ Merged
- revisions 427874 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427875 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-14 15:24 +0000 [r427870] Mark Michelson <mmichelson@digium.com>
-
- * main/bridge.c, main/bridge_basic.c,
- include/asterisk/stasis_bridges.h, tests/test_cel.c,
- apps/app_queue.c, main/cel.c, main/stasis_bridges.c, /,
- res/stasis/app.c: Fix race condition that could result in ARI
- transfer messages not being sent. From reviewboard: "During blind
- transfer testing, it was noticed that tests were failing
- occasionally because the ARI blind transfer event was not being
- sent. After investigating, I detected a race condition in the
- blind transfer code. When blind transferring a single channel,
- the actual transfer operation (i.e. removing the transferee from
- the bridge and directing them to the proper dialplan location) is
- queued onto the transferee bridge channel. After queuing the
- transfer operation, the blind transfer Stasis message is
- published. At the time of publication, snapshots of the channels
- and bridge involved are created. The ARI subscriber to the blind
- transfer Stasis message then attempts to determine if the bridge
- or any of the involved channels are subscribed to by ARI
- applications. If so, then the blind transfer message is sent to
- the applications. The way that the ARI blind transfer message
- handler works is to first see if the transferer channel is
- subscribed to. If not, then iterate over all the channel IDs in
- the bridge snapshot and determine if any of those are subscribed
- to. In the test we were running, the lone transferee channel was
- subscribed to, so an ARI event should have been sent to our
- application. Occasionally, though, the bridge snapshot did not
- have any channels IDs on it at all. Why? The problem is that
- since the blind transfer operation is handled by a separate
- thread, it is possible that the transfer will have completed and
- the channels removed from the bridge before we publish the blind
- transfer Stasis message. Since the blind transfer has completed,
- the bridge on which the transfer occurred no longer has any
- channels on it, so the resulting bridge snapshot has no channels
- on it. Through investigation of the code, I found that attended
- transfers can have this issue too for the case where a transferee
- is transferred to an application." The fix employed here is to
- decouple the creation of snapshots for the transfer messages from
- the publication of the transfer messages. This way, snapshots can
- be created to reflect what they are at the time of the transfer
- operation. Review: https://reviewboard.asterisk.org/r/4135
- ........ Merged revisions 427848 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-14 14:56 +0000 [r427846] Joshua Colp <jcolp@digium.com>
-
- * /, apps/confbridge/conf_state_multi_marked.c: app_confbridge:
- Play "leader has left" sound even when musiconhold is enabled.
- Currently if the leader of a conference bridge leaves any
- participant that has musiconhold enabled will not hear the
- "leader has left" sound. This is because musiconhold is started
- and THEN the sound is played. This change makes it so that the
- sound is played and THEN musiconhold is started. This provides a
- better experience for users as they may not have known previously
- why they went back to musiconhold. Review:
- https://reviewboard.asterisk.org/r/4177/ ........ Merged
- revisions 427844 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427845 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-14 14:24 +0000 [r427841] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
- include/asterisk/res_pjsip.h: Fix race condition where duplicated
- requests may be handled by multiple threads. This is the Asterisk
- 13 version of the patch. The main difference is in the pubsub
- code since it was completely refactored between Asterisk 12 and
- 13. Review: https://reviewboard.asterisk.org/r/4175
-
-2014-11-13 22:03 +0000 [r427815] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c: res_pjsip_exten_state:
- PJSIPShowSubscriptionsInbound causes crash When using a
- non-default sorcery wizard (in this instance realtime) for
- outbound registrations and after adding in an appropriate call to
- ast_sorcery_apply_config() (since it is missing) Asterisk will
- crash after a stack overflow occurs due to the code infinitely
- recursing. The fix entails removing the outbound registration
- state dependency from the outbound registration sorcery object
- and instead keeping an in memory container that can be used to
- lookup the state when needed. ASTERISK-24514 Reported by: Mark
- Michelson Review: https://reviewboard.asterisk.org/r/4164/
- ........ Merged revisions 427814 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-13 15:44 +0000 [r427789] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/stasis.h, include/asterisk/stasis_app.h,
- res/stasis/app.h, res/res_stasis.c, /, res/stasis/app.c,
- res/stasis/stasis_bridge.c: Stasis: Fix StasisEnd message
- ordering This change corrects message ordering in cases where a
- channel-related message can be received after a Stasis/ARI
- application has received the StasisEnd message. The StasisEnd
- message was being passed to applications directly without waiting
- for the channel topic to empty. As a result of this fix, other
- bugs were also identified and fixed: * StasisStart messages were
- also being sent directly to apps and are now routed through the
- stasis message bus properly * Masquerade monitor datastores were
- being removed at the incorrect time in some cases and were
- causing StasisEnd messages to not be sent * General refactoring
- where necessary for the above * Unsubscription on StasisEnd
- timing changes to prevent additional messages from following the
- StasisEnd when they shouldn't A channel sanitization function
- pointer was added to reduce processing and AO2 lookups. Review:
- https://reviewboard.asterisk.org/r/4163/ ASTERISK-24501 #close
- Reported by: Matt Jordan ........ Merged revisions 427788 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-13 00:00 +0000 [r427763] Matthew Jordan <mjordan@digium.com>
-
- * main/rtp_engine.c, /: main/rtp_engine: Fix crash when processing
- more than one RTCP report info block Asterisk - in
- res_rtp_asterisk - only understands a single RTCP report info
- block. When the RTCP information was refactored in the RTP Engine
- to be pushed over the Stasis message bus, I put in the hooks into
- the engine to handle multiple RTCP report info blocks, in the
- hope that a future RTP implementation would be able to provide
- that data. Unfortunately, res_rtp_asterisk has a tendency to
- "lie": (1) It will send RTCP reports with a
- reception_report_count greater than 1 (which is pulled directly
- from the RTCP packet itself, so that part is correct) (2) It will
- only provide a single report block When the rtp_engine goes to
- convert this to a JSON blob, hilarity ensues as it looks for a
- report block that doesn't exist. This patch updates the
- rtp_engine to be a bit more skeptical about what it is presented
- with. While this could also be fixed in res_rtp_asterisk, this
- patch prefers to fix it in the engine for two reasons: (1) The
- engine is designed to work with multiple RTP implementation, and
- hence having it be more robust is a good thing (tm) (2)
- res_rtp_asterisk's handling of RTCP information is "fun". It
- should report the correct reception_report_count; ideally it
- should also be giving us all of the blocks - but it is
- *definitely* not designed to do that. Going down that road is a
- non-trivial effort. Review:
- https://reviewboard.asterisk.org/r/4158/ ASTERISK-24489 #close
- Reported by: Gregory Malsack Tested by: Gregory Malsack
- ASTERISK-24498 #close Reported by: Beppo Mazzucato Tested by:
- Beppo Maazucato ........ Merged revisions 427762 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-12 20:39 +0000 [r427737] Corey Farrell <git@cfware.com>
-
- * /, main/features.c: Fix leak in AMI Action Bridge Add missing
- reference cleanup for newly created bridge. ASTERISK-24281
- Reported by: Stefan Engström Review:
- https://reviewboard.asterisk.org/r/4154/ ........ Merged
- revisions 427736 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-12 16:12 +0000 [r427711] Joshua Colp <jcolp@digium.com>
-
- * main/pbx.c, /: pbx: Fix off-nominal case where a freed extension
- may still be used. If during the operation of adding an extension
- a priority is added but fails it is possible for the extension to
- be freed but still exist in the PBX core. If this occurs
- subsequent lookups may try to access the extension and end up in
- freed memory. This change removes the extension from the PBX core
- when the priority addition fails and then frees the extension.
- ASTERISK-24444 #close Reported by: Leandro Dardini Review:
- https://reviewboard.asterisk.org/r/4162/ ........ Merged
- revisions 427709 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427710 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-12 13:46 +0000 [r427684] Corey Farrell <git@cfware.com>
-
- * codecs/ilbc, /, tests, codecs/speex, apps/confbridge,
- Makefile.rules: Fix compiler error when using ./configure
- --enable-dev-mode --enable-coverage When DONT_OPTIMIZE is enabled
- with dev-mode, it causes a shadow compilation to be done with
- output to /dev/null. This can cause errors with coverage when GCC
- attempts to write to /dev/null.gcno. This change disables
- coverage for the shadow compilation. ASTERISK-24502 #close
- Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4151/ ........ Merged
- revisions 427682 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427683 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-09 08:00 +0000 [r427643] Corey Farrell <git@cfware.com>
-
- * main/manager.c, /: manager: Fix HTTP connection reference leaks.
- Fix reference leak that happens if (session && !blastaway).
- ASTERISK-24505 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4153/ ........ Merged
- revisions 427641 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427642 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-09 00:38 +0000 [r427583-427615] Matthew Jordan <mjordan@digium.com>
-
- * channels/chan_mgcp.c, /: channels/chan_mgcp: Fix regression which
- causes gateways to be skipped In r227276, a while loop was turned
- into a for loop. Unfortunately, a portion of the while loop was
- left in the code such that, when a static gateway is encountered
- in the list of MGCP gateways, the next gateway would be skipped.
- At best, we would simply flip past a gateway; at worst, this
- could lead to a crash. ASTERISK-24500 #close Reported by: Xavier
- Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne
- (License 6657) ........ Merged revisions 427613 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427614 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, addons/chan_mobile.c: addons/chan_mobile: Increase buffer size
- of UCS2 encoded SMS messages When UCS2 character encoding is
- used, one symbol in national language can be expanded to 4 bytes.
- The current buffer used for receiving message in do_monitor_phone
- is 256 bytes, which is not large enough for incoming messages.
- For example: * AT+CMGR phone response prefix '+CMGR: "REC
- UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes *
- SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone
- response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null
- character - 1 byte This results in a needed buffer size of 349
- bytes. Hence, this patch opts for a 350 byte buffer.
- ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches:
- chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651)
- chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651)
- ........ Merged revisions 427607 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427610 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_voicemail.c: app_voicemail: Fix enhancement that allowed
- multiple recipients in To: header An issue existed in r420577,
- which added multiple recipients to voicemail emails. The patch,
- when looking at the intended recipients, looked ahead for the '|'
- character inside a while loop which already had pulled out the
- appropriate field parsing on the '|' character. This would cause
- it to skip the recipients. This patch fixes it such that it
- relies completely on the while loop to parse through the e-mail
- fields. Note that the original author of the patch looked at this
- fix and approved it. ASTERISK-24250 #close Reported by: abelbeck
- patches: voicemail-420577-to-comma-fix.diff uploaded by abelbeck
- (License 5903)
-
- * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix T.38
- issues with remote bridges After r425242 the
- fax/sip/directmedia_reinvite_t38 test started failing due to the
- surviving channel not being re-INVITEd back from T.38 to audio.
- This patch fixes that bug - a deeper explanation of what happened
- follows. When two RTP channels are in a native bridge, the
- bridging layer will investigate each via the get_rtp_info glue
- callback. This callback returns the native bridge preference of
- the channel *at that moment in time* (that part is key). At
- different points during the bridging, the native bridging layer
- will inform the RTP capable channels of the status of the bridge
- via the update_peer glue callback. In a T.38 scenario with audio
- direct media, the sequence of events will often look like the
- following: * SIP/A and SIP/B both have audio and enter a native
- bridge. * Asterisk re-INVITEs audio between SIP/A and SIP/B
- directly (via an update_peer callback). * SIP/A sends a re-INVITE
- to T.38, which causes Asterisk to send a re-INVITE to T.38 to
- SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack
- receives UDPTL packets in Asterisk from both endpoints. From the
- perspective of the channels, we are now in a local bridge for
- T.38, even though we are technically still in a remote bridge in
- bridge_native_rtp. (YAY!) * When one side hangs up,
- bridge_native_rtp is told to stop bridging. It then re-evaluates
- the channels and asks them how they are bridged - and since T.38
- is enabled, they reply with a Local bridge (which is correct),
- but is wrong because the audio portion is still technically in a
- remote bridge. * Asterisk releases the surviving channel, whose
- audio is *not* re-INVITED back to Asterisk as bridge_native_rtp
- incorrectly assumes that it was in a local bridge. Ironically,
- prior to r425242, this used to work mostly due to a fluke in the
- bridging layer. The purpose of the get_rtp_info callback
- shouldn't be modified: it should tell the bridging layer what
- kind of bridge the channel prefers at that moment in time. If you
- have T.38 enabled, that *must* be a local bridge, as the UDPTPL
- stack must be in the media path. As such, this patch does not
- modify that part of the code. However, we have to tell the
- channels to re-evaluate themselves when they come out of a native
- bridge, since we can no longer trust the get_rtp_info callbacks
- when the native bridge is being stopped. Something else may have
- changed in the channels, and they may now be lying to us. As
- such, this patch makes it so that we unilaterally tell the
- channels that they are no longer bridged via the update_peer
- callback. This is actually what the channels expect anyway: code
- in both chan_sip and chan_pjsip's callbacks look at the T.38
- state and - if they were in T.38 - send a re-INVITE to get the
- audio back to Asterisk. Review:
- https://reviewboard.asterisk.org/r/4157/ ........ Merged
- revisions 427582 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-08 18:17 +0000 [r427557] Corey Farrell <git@cfware.com>
-
- * /, channels/chan_console.c: chan_console: Fix reference leaks to
- pvt. Fix a bunch of calls to get_active_pvt where the reference
- is never released. ASTERISK-24504 #close Reported by: Corey
- Farrell Review: https://reviewboard.asterisk.org/r/4152/ ........
- Merged revisions 427554 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427555 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-06 19:22 +0000 [r427494-427512] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_agent_pool.c, /: app_agent_pool: Made agent alert
- interruptable by DTMF. Made agent able to interrupt the alerting
- beep playback with DTMF. Any digit can interrupt if the call does
- not need to be acknowledged. Only the first digit of the
- acknowledgement can interrupt if the call needs to be
- acknowledged. The agent interrupting the alerting playback builds
- on the ASTERISK-24447 patch because it knows what digit
- interrupted the playback and needs to be able to pass that digit
- to the DTMF hook digit collection code. ASTERISK-24257 #close
- Reported by: Steve Pitts Review:
- https://reviewboard.asterisk.org/r/4123/ ........ Merged
- revisions 427508 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, include/asterisk/bridge_channel.h, main/bridge_channel.c:
- Bridge DTMF hooks: Made audio pass from the bridge while waiting
- for more matching digits. * Made collecting DTMF digits for the
- DTMF feature hooks pass frames from the bridge. * Made collecting
- DTMF digits possible by other bridge hooks if there is a need.
- ASTERISK-24447 #close Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/4123/ ........ Merged
- revisions 427493 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-06 18:20 +0000 [r427491] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/pjsip_distributor.c: res_pjsip: Ensure in-dialog
- responses have an endpoint associated. When handling incoming
- messages we determine if it is associated with a dialog. If so we
- use that to determine what serializer and endpoint to use for the
- message. Previously this would pass the endpoint to the endpoint
- lookup module to actually place the endpoint completely on the
- message. For in-dialog responses, however, this did not occur as
- dialog processing took over and the endpoint lookup did not
- occur. This change just places the endpoint in the expected spot
- immediately instead of relying on the endpoint lookup module.
- In-dialog responses thus have the expected endpoint. AST-1459
- #close Review: https://reviewboard.asterisk.org/r/4146/ ........
- Merged revisions 427490 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-06 12:13 +0000 [r427384-427466] Corey Farrell <git@cfware.com>
-
- * main/file.c, /: main/file.c: fix possible extra ast_module_unref
- to format modules. fn_wrapper only adds a reference to the
- format's module if the file was able to be opened. If not this
- causes an unmatched ast_module_unref in filestream_destructor.
- Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4149/ ........ Merged
- revisions 427464 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427465 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_hep.c, /: res_hep: fix major leak that occurs when config
- is missing or enabled=no. Add missing unreference in
- hepv3_send_packet. ASTERISK-24491 #close Reported by: Zane Conkle
- Tested by: Zane Conkle Review:
- https://reviewboard.asterisk.org/r/4150/ ........ Merged
- revisions 427400 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/utils.c, include/asterisk/stringfields.h: Fix unintential
- memory retention in stringfields. * Fix missing / unreachable
- calls to __ast_string_field_release_active. * Reset pool->used to
- zero when the current pool->active reaches zero. ASTERISK-24307
- #close Reported by: Etienne Lessard Tested by: ibercom, Etienne
- Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........
- Merged revisions 427380 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 427381 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427382 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-06 02:37 +0000 [r427356] George Joseph <george.joseph@fairview5.com>
-
- * tests/test_strings.c, /: test_strings: Remove string tests that
- exercise asserts. Since unit tests are run with DO_CRASH, those
- tests were causing the test to fail. Tested-by: George Joseph
- ........ Merged revisions 427354 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427355 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-05 19:52 +0000 [r427334] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip/config_system.c, configs/samples/pjsip.conf.sample,
- res/res_pjsip.c: Make the disable_tcp_switch PJSIP system object
- enabled by default. Testing has shown repeatedly that PJSIP's
- default behavior of switching automatically to TCP for large
- messages can cause issues. The most common issues are that
- devices that we are communicating with do not handle the switch
- to TCP gracefully, thus causing situations such as broken calls
- or broken subscriptions. Now, in order to have this behavior
- happen, you must opt into it. The sample file has been updated to
- warn that enabling the TCP switch behavior may cause issues for
- you, so use at your own risk.
-
-2014-11-05 12:18 +0000 [r427303] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Add logging
- during startup to aid debugging if local DNS is misbehaving. This
- change adds a bit of logging so if the local DNS is misbehaving
- it is easier to track down what is going on and where Asterisk
- may be hanging. ASTERISK-24438 #close Reported by: Melissa
- Shepherd Review: https://reviewboard.asterisk.org/r/4148/
- ........ Merged revisions 427300 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-05 00:15 +0000 [r427228-427276] George Joseph <george.joseph@fairview5.com>
-
- * pbx/pbx_config.c, main/config.c, tests/test_strings.c,
- include/asterisk/utils.h, /, main/utils.c: config: Make
- text_file_save and 'dialplan save' escape semicolons in values.
- When a config file is read, an unescaped semicolon signals
- comments which are stripped from the value before it's stored.
- Escaped semicolons are then unescaped and become part of the
- value. Both of these behaviors are normal and expected. When the
- config is serialized either by 'dialplan save' or
- AMI/UpdateConfig however, the now unescaped semicolons are
- written as-is. If you actually reload the file just saved, the
- unescaped semicolons are now treated as start of comments. Since
- true comments are stripped on read, any semicolons in
- ast_variable.value must have been escaped originally. This patch
- re-escapes semicolons in ast_variable.values before they're
- written to file either by 'dialplan save' or
- config/ast_config_text_file_save which is called by
- AMI/UpdateConfig. I also fixed a few pre-existing formatting
- issues nearby in pbx_config.c Tested-by: George Joseph
- ASTERISK-20127 #close Review:
- https://reviewboard.asterisk.org/r/4132/ ........ Merged
- revisions 427275 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/config.c, /: config: BUG: Restore ability for non-templ to
- be used as base objs in config. My recent refactor of config.c
- accidentally removed the capability for an object to inherit from
- a non-template object. This patch restores the capability to
- inherit from both template and non-template objects. Tested-by:
- George Joseph Reported-by: Scott Griepentrog ASTERISK-24487
- #close Review: https://reviewboard.asterisk.org/r/4147/ ........
- Merged revisions 427227 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-04 19:44 +0000 [r427181-427204] Corey Farrell <git@cfware.com>
-
- * funcs/func_talkdetect.c, /: func_talkdetect: Fix stasis message
- leak in audiohook callback. ASTERISK-24482 #close Reported by:
- Corey Farrell Review: https://reviewboard.asterisk.org/r/4142/
- ........ Merged revisions 427203 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_http_websocket.c: res_http_websockets: Fix extra unref
- of module In websocket_add_protocol_internal is used to add the
- "echo" protocol, but ast_websocket_remove_protocol is used to
- remove it. This causes an extra call to ast_module_unref.
- ASTERISK-24480 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4140/ ........ Merged
- revisions 427200 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/app.c: Fix crash caused by merge error on review 4138 When
- merging from 12 to 13 there were conflicts, I mistakenly had the
- loop run ast_closestream(others[0]) when it should be
- ast_closestream(others[x]).
-
-2014-11-03 18:15 +0000 [r427130] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_pjsip/config_system.c, UPGRADE.txt,
- configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip:
- Add disable_tcp_switch option. When a packet exceeds the MTU,
- pjproject will switch from UDP to TCP. In some circumstances (on
- some networks), this can cause some issues with messages not
- getting sent to the correct destination - and can also cause
- connections to get dropped due to quirks in pjproject deciding to
- terminate TCP connections with no messages. While fixing the
- routing/messaging issues is important, having a configuration
- option in Asterisk that tells pjproject to not switch over to TCP
- would be useful. That way, if some glitch is discovered on some
- other network/site, we can at least disable the behavior until a
- fix is put into place. AFS-197 #close Review:
- https://reviewboard.asterisk.org/r/4137/ ........ Merged
- revisions 427129 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-03 02:34 +0000 [r427021-427089] Corey Farrell <git@cfware.com>
-
- * apps/app_voicemail.c, /: Fix compile error caused by review 4138
- There is no procedure called ast_closeframe, fix code to use
- ast_closestream. Reported By: Matt Jordan ........ Merged
- revisions 427087 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427088 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/app.c, apps/app_voicemail.c, /: Fix ast_writestream leaks
- Fix cleanup in __ast_play_and_record where others[x] may be
- leaked. This was caught where prepend != NULL && outmsg != NULL,
- once realfile[x] == NULL any further others[x] would be leaked. A
- cleanup block was also added for prepend != NULL && outmsg ==
- NULL. 11+: Fix leak of ast_writestream recording_fs in
- app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by:
- Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/
- ........ Merged revisions 427023 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 427024 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427025 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/abstract_jb.c: func_jitterbuffer: fix frame leaks. Fix
- code paths where it is possible for frames to leak. Fix
- uninitialized variable in jb_get_fixed and jb_get_adaptive.
- ASTERISK-22409 #related Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4128/ ........ Merged
- revisions 427019 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 427020 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-11-02 01:01 +0000 [r426996] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_stasis.c: res/res_stasis: Fix crash on module unload
- while performing operation When the res_stasis module is
- unloaded, it will dispose of the apps_registry container. This is
- a problem if an ARI operation is in flight that attempts to use
- the registry, as the shutdown occurs in a separate thread. This
- patch adds some sanity checks to the various routines that access
- the registry which cause the operations to fail if the
- apps_registry does not exist. Crash caught by the Asterisk Test
- Suite. ........ Merged revisions 426995 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-31 16:50 +0000 [r426934] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * Makefile, /: install init.d files on GNU/kFreeBSD Review:
- https://reviewboard.asterisk.org/r/4118/ ........ Merged
- revisions 426926 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426927 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426933 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-31 16:40 +0000 [r426924-426930] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c: pjsip:
- clarify tls cert and key file usage A question arose as to
- whether a .pem file could be provided in place of the .crt and
- .key files in a PJSIP TLS configuration. I tested this and
- discovered that although a cert will be read from the pem file, a
- key will not, and thus the priv_key_file entry is still required.
- This update to the fine documentation clarifies the option usage.
- AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/
- Reported by: John Bigelow ........ Merged revisions 426928 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_outbound_registration.c: pjsip: Handle outbound
- unregister correctly This updates the status of the outbound
- registration to reflect when it has been unregistered. Since the
- registration is unregistered but is not stopped, the registration
- schedule remains active as before. The patch also updates the
- documentation of both the AMI and CLI commands. ASTERISK-24411
- #close Review: https://reviewboard.asterisk.org/r/4119/ Reported
- by: John Bigelow patches: unregister-patch1.txt uploaded by John
- Bigelow (License 5091) ........ Merged revisions 426923 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-31 03:26 +0000 [r426865] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/sip/reqresp_parser.c,
- channels/sip/include/reqresp_parser.h:
- channels/sip/reqresp_parser: Fix unit tests for r426594 When
- r426594 was made, it did not take into account a unit test that
- verified that the function properly populated the unsupported
- buffer. The function would previously memset the buffer if it
- detected it had any contents; since this function can now be
- called iteratively on successive headers, the unit tests would
- now fail. This patch updates the unit tests to reset the buffer
- themselves between successive calls, and updates the
- documentation of the function to note that this is now required.
- ........ Merged revisions 426858 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426860 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426863 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-31 03:08 +0000 [r426803-426833] Corey Farrell <git@cfware.com>
-
- * contrib/Makefile (added), Makefile, /: REF_DEBUG: Install
- refcounter.py to $(ASTDATADIR)/scripts This change ensures
- refcounter.py is installed to a place where it can be found by
- the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4094/ ........ Merged
- revisions 426830 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426831 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426832 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_queue.c: app_queue: fix a couple leaks to struct
- call_queue in set_member_value set_member_value has a couple
- leaks to references in the variable q found through testsuite
- tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES
- compiler declaration, this is no longer possible with the updated
- REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged
- revisions 426805 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426806 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/audiohook.c: audiohooks: Clean references to formats Cleanup
- references to in_translate[x].format and out_translate[x].format
- in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by:
- Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/
-
-2014-10-30 21:13 +0000 [r426757-426780] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_exten_state.c, /: res_pjsip_exten_state:
- PJSIPShowSubscriptionsInbound causes crash Currently, it is
- possible for some subscriptions to get into a NULL state. When
- this occurs and the PJSIPShowSubscriptionsInbound ami action is
- issued and a device is subscribed for extension state then the
- associated subscription state object can't be located. The code
- then attempts to dereference a NULL object. Added a NULL check to
- avoid the problem. Reported by: John Bigelow ........ Merged
- revisions 426779 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip/pjsip_options.c, /: res_pjsip: incorrect qualify
- statistics after disabling for contact When removing the
- qualify_frequency from an AoR or a contact the statistics shown
- when issuing "pjsip show aors" from the CLI are incorrect. This
- patch deletes the contact's status object from sorcery,
- disassociating it from the contact, if the qualify_freqency is
- removed from configuration. ASTERISK-24462 #close Reported by:
- Mark Michelson Review: https://reviewboard.asterisk.org/r/4116/
- ........ Merged revisions 426755 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-30 09:20 +0000 [r426702] Walter Doekes <walter+asterisk@wjd.nu>
-
- * apps/app_voicemail.c, /: app_voicemail: Fix unchecked bounds of
- myArray in IMAP_STORAGE. In update_messages_by_imapuser(),
- messages were appended to a finite array which resulted in a
- crash when an IMAP mailbox contained more than 256 entries. This
- memory is now dynamically increased as needed. Observe that this
- patch adds a bunch of XXX's to questionable code. See the review
- (url below) for more information. ASTERISK-24190 #close Reported
- by: Nick Adams Tested by: Nick Adams Review:
- https://reviewboard.asterisk.org/r/4126/ ........ Merged
- revisions 426691 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426692 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426696 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-30 06:09 +0000 [r426668] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * channels/chan_unistim.c, /: Add additional checks for NULL
- pointers to fix several crashes reported. ASTERISK-24304 #close
- Reported by: dhanapathy sathya ........ Merged revisions 426666
- from http://svn.asterisk.org/svn/asterisk/branches/11 ........
- Merged revisions 426667 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-30 01:59 +0000 [r426597-426602] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: channels/chan_sip: Add improved support
- for 4xx error codes This patch adds support for 414, 493, 479,
- and a stray 400 response in REGISTER response handling. This
- helps interoperability in a number of scenarios. Review:
- https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch
- uploaded by oej (License 5267) ........ Merged revisions 426599
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 426600 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426601 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/sip/reqresp_parser.c, /, channels/chan_sip.c:
- channels/chan_sip: Support mutltiple Supported and Required
- headers A SIP request may contain multiple Supported: and
- Required: headers. Currently, chan_sip only parses the first
- Supported/Required header it finds. This patch adds support for
- multiple Supported/Required headers for INVITE requests. Review:
- https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close
- Reported by: Olle Johansson patches: rb2478.patch uploaded by oej
- (License 5267) ........ Merged revisions 426594 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426595 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426596 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-29 10:33 +0000 [r426570] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_phone.c: Fix building chan_phone on big endian
- systems A left over from the formats conversion (Corey Farrell).
- ASTERISK-24458 #close Review:
- https://reviewboard.asterisk.org/r/4117/
-
-2014-10-28 21:26 +0000 [r426552] Richard Mudgett <rmudgett@digium.com>
-
- * /, bridges/bridge_builtin_features.c: bridge_builtin_features:
- Add missing channel locks around
- ast_get_chan_features_general_config(). The feature_automonitor()
- and feature_automixmonitor() functions were not locking the
- channel around ast_get_chan_features_general_config(). Accessing
- the channel datastore list without the channel locked is a good
- way to corrupt the list or follow the pointer chain into
- oblivion. ........ Merged revisions 426531 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-28 21:05 +0000 [r426525-426529] Corey Farrell <git@cfware.com>
-
- * /, res/res_fax.c: res_fax: Resolve T38 gateway frame leak. When
- frames are translated by a fax gateway they need to be freed. The
- existing call to ast_frfree was unreachable. This change
- reorganizes fax_gateway_framehook to ensure that ast_frfree is
- called when needed. ASTERISK-24457 #close Reported by: Corey
- Farrell Review: https://reviewboard.asterisk.org/r/4115/ ........
- Merged revisions 426527 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426528 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: manager: Unsubscribe from acl_change_sub at
- shutdown. ASTERISK-24453 #close Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged
- revisions 426524 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-28 18:09 +0000 [r426459] mdavenport <mdavenport@localhost>:
-
- * configs/samples/manager.conf.sample: ASTERISK-23512, correct
- inaccurate comment in manager.conf.sample
-
-2014-10-28 16:40 +0000 [r426368-426432] Matthew Jordan <mjordan@digium.com>
-
- * /, main/bridge.c: main/bridge: Destroy features struct on off
- nominal path during bridge impart When a channel is imparted to a
- bridge, the invocation of the function may provide an
- ast_bridge_features struct. Upon passing this to
- ast_bridge_impart, the caller must assume that ownership has
- passed to the function, as in all paths the function destroys the
- struct prior to returning (as its purpose is to configure the
- behavior of the channel while in the bridge). On one off nominal
- path - where the channel already has a PBX thread - the struct
- was not being destroyed. This patch fixes that glitch.
- ASTERISK-24437 #close Reported by: Scott Griepentrog ........
- Merged revisions 426431 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: main/manager: Fix typo in AMI event
- documentation of "OriginateResponse" The parameter name is
- "Response", not "Resonse". ASTERISK-24430 #close Reported by:
- Dafi Ni ........ Merged revisions 426366 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426367 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-28 14:56 +0000 [r426294-426362] mdavenport <mdavenport@localhost>:
-
- * res/res_agi.c: ASTERISK-24323, fix bug in documentation of AGI
- STREAM FILE CONTROL
-
- * configs/samples/extensions.conf.sample: ASTERISK-24419, fix
- incorrect syntax for setting language in extensions.conf.sample
-
-2014-10-28 11:20 +0000 [r426252-426266] Corey Farrell <git@cfware.com>
-
- * apps/app_queue.c, /: app_queue: Cleanup ao2_iterator Clean
- ao2_iterator, resolving reference leak to queue members.
- ASTERISK-24454 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4111/ ........ Merged
- revisions 426255 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426260 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * funcs/func_cdr.c: func_cdr: Fix CDR_PROP payload leak Remove
- duplicate allocation of payload, preventing leak. ASTERISK-24455
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4113/
-
-2014-10-27 17:54 +0000 [r426234] Sean Bright <sean@malleable.com>
-
- * build_tools/menuselect-deps.in, configure,
- include/asterisk/autoconfig.h.in, configure.ac, makeopts.in:
- configure: Add autoconf check for libopus. Because opus
- transcoding support cannot be included in the standard Asterisk
- distribution, a few codec_opus implementations have popped up. To
- make it easier for people to drop in opus support in their own
- installations, this patch adds configure checks for libopus.
- Review: https://reviewboard.asterisk.org/r/4106/
-
-2014-10-27 02:46 +0000 [r426143-426211] Matthew Jordan <mjordan@digium.com>
-
- * res/res_http_websocket.c, /: res/res_http_websocket: Fix minor
- nits found by wdoekes on r409681 When Moises committed the fixes
- for WSS (which was a great patch), wdoekes had a few style nits
- that were on the review that got missed. This patch resolves what
- I *think* were all of the ones that were still on the review.
- Thanks to both moy for the patch, and wdoekes for the reviews.
- Review: https://reviewboard.asterisk.org/r/3248/ ........ Merged
- revisions 426209 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426210 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_phoneprov.c: res/res_phoneprov: Fix crash on shutdown
- caused by container cleanup In res_phoneprov, unloading the
- module first destroys the http_routes container, followed by the
- users. However, users may have a route in the http_routes
- container; the validity of this container is not checked in the
- users destructor. Hence, we hit an assert as the container has
- already been set to NULL. This patch does two things: (1) It adds
- a sanity check in the user destructor (because why not) (2) It
- switches the order of destruction, so that users are disposed of
- prior to the HTTP routes they may hold a reference to. Note that
- this crash was caught by the Test Suite (go go testing!) ........
- Merged revisions 426174 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_srtp.c, /: res/res_srtp: Fix include issue for libsrtp
- 1.5.0 In libsrtp 1.5.0, crypto_get_random is no longer resolved
- simply by including srtp.h. Now, one must include crypto_kernel.h
- as well. As it turns out, this header file has been provided by
- the library since 2006, so this is a relatively benign change.
- ASTERISK-24436 #close Reported by: Patrick Laimbock ........
- Merged revisions 426140 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 426141 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 426142 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-24 15:17 +0000 [r426120] Jonathan Rose <jrose@digium.com>
-
- * main/manager.c: Documentation: Improve documentation for
- ExtensionStatus AMI events Review:
- https://reviewboard.asterisk.org/r/4085/
-
-2014-10-24 Asterisk Development Team <asteriskteam@digium.com>
-
- * Asterisk 13.0.0 Released.
-
-2014-10-22 21:27 +0000 [r426097] Shaun Ruffell <sruffell@digium.com>
-
- * codecs/codec_dahdi.c: codec_dahdi: Cannot use struct
- ast_translator.core_{src,src}_codec. This fixes a Segmentation
- fault introduced in r419044 "media formats: re-architect handling
- of media for performance improvements". The problem is that
- codec_dahdi was using core_src_codec and core_dst_codec in the
- ast_translator structure when these fields were never set. Now
- instead of trying to map the new core codec descriptions to the
- way DAHDI defines different codecs, we will store the DAHDI
- specific formats in 'struct translator' directly so we can refer
- to them without mapping. This also allows us to remove the
- "global_format_map" structure, since we can now query the list of
- translators directly to make sure we do not ever register a DAHDI
- based translator for a specific path more than once and eliminate
- the need to keep the list and the map in sync. ASTERISK-24435
- #close Reported by: Marian Koniuszko Review:
- https://reviewboard.asterisk.org/r/4105/
-
-2014-10-21 17:47 +0000 [r426079] Richard Mudgett <rmudgett@digium.com>
-
- * main/translate.c: translage.c: Fix regression when generating
- translation path strings. Fix the AMI Status action read and
- write translation path strings from growing for each channel in
- the status event list by reseting the ast string given to
- ast_translate_path_to_str() to fill in the given translation
- path.
-
-2014-10-20 14:15 +0000 [r425991] Matthew Jordan <mjordan@digium.com>
-
- * res/res_xmpp.c, main/tcptls.c, /: AST-2014-011: Fix POODLE
- security issues There are two aspects to the vulnerability: (1)
- res_jabber/res_xmpp use SSLv3 only. This patch updates the module
- to use TLSv1+. At this time, it does not refactor
- res_jabber/res_xmpp to use the TCP/TLS core, which should be done
- as an improvement at a latter date. (2) The TCP/TLS core, when
- tlsclientmethod/sslclientmethod is left unspecified, will default
- to the OpenSSL SSLv23_method. This method allows for all
- encryption methods, including SSLv2/SSLv3. A MITM can exploit
- this by forcing a fallback to SSLv3, which leaves the server
- vulnerable to POODLE. This patch adds WARNINGS if a user uses
- SSLv2/SSLv3 in their configuration, and explicitly disables
- SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk
- will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly
- chosen. For TLS servers, Asterisk will no longer support SSLv2 or
- SSLv3. Much thanks to abelbeck for reporting the vulnerability
- and providing a patch for the res_jabber/res_xmpp modules.
- Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425
- #close Reported by: abelbeck Tested by: abelbeck, opsmonitor,
- gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by
- abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch
- uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff
- uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded
- by mjordan (License 6283) ........ Merged revisions 425987 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-19 17:07 +0000 [r425965] George Joseph <george.joseph@fairview5.com>
-
- * Makefile, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac, makeopts.in: build: Force -fsigned-char on
- platforms where the default for char is unsigned gcc on the ARM
- platform defaults 'char' to 'unsigned char' whereas Intel and
- SPARC default to 'signed char'. This is only an issue in the rare
- cases where negative values are assigned to a 'char' but this
- this patch insures compatibility by detecting platforms that
- default to 'unsigned' and adding an '-fsigned-char' flag to
- _ASTCFLAGS. If compiling for ARM (native or cross-compile) be
- sure to run ./bootstrap.sh and ./configure to regenerate the
- build files. You shouldn't have to do this for Intel or SPARC.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4091/ ........ Merged
- revisions 425964 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-19 04:01 +0000 [r425923-425944] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Revert 425922
- This patch for r425922 introduced a bug, wherein sending an
- INVITE request with no SDP would cause Asterisk to not send an
- SDP Offer in the 200 OK. The current structure of
- res_pjsip_sdp_rtp is a bit hard to deal with to fix this, as
- create_outgoing_sdp has no knowledge of whether or not it is
- creating an SDP as a new Offer or an Answer. This is something of
- an oversight in the callback definition, as the caller of it does
- have this information.
-
- * res/res_pjsip_sdp_rtp.c: res/res_pjsip_sdp_rtp: Remove left over
- reference to override_prefs The usage of the local override_prefs
- variable in create_outgoing_sdp_stream was previously to track an
- override format preference set by PJSIP_MEDIA_OFFER. Now,
- however, that function simply sets the joint capabilities
- structure, session->req_caps. During the media format rework, the
- override_prefs was instead used to check if there were any
- formats in session->req_caps. However, this usage isn't useful in
- create_outgoing_sdp_stream. session->req_caps contains the
- negotiated formats for *all* streams, not just the current one
- being created. Thus, so long as any stream of any type has
- provided a format, override_prefs will be non-zero. Hence, its
- usage in checking whether or not we should look at the formats on
- the endpoint or the joint capabilities is generally useless.
- There's only two things useful to check: (1) Does the endpoint
- have a format for the media type? (2) Did we negotiate a format
- for the media type? If either of those is a 'no', then we must
- kill the media stream.
-
-2014-10-17 22:43 +0000 [r425905] Jonathan Rose <jrose@digium.com>
-
- * configs/samples/cli_aliases.conf.sample: Sample Configurations:
- make 'pjsip reload' reload all reloadable pjsip modules AST-1432
- #close Reported by: John Bigelow
-
-2014-10-17 13:35 +0000 [r425821-425879] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, res/res_pjsip.c,
- res/res_pjsip_session.c, /: res_pjsip_session/res_pjsip_sdp_rtp:
- Be more tolerant of offers When an inbound SDP offer is received,
- Asterisk currently makes a few incorrection assumptions: (1) If
- the offer contains more than a single audio/video stream,
- Asterisk will reject the entire stream with a 488. This is an
- overly strict response; generally, Asterisk should accept the
- media streams that it can accept and decline the others. (2) If
- the offer contains a declined media stream, Asterisk will attempt
- to process it anyway. This can result in attempting to match
- format capabilities on a declined media stream, leading to a 488.
- Asterisk should simply ignore declined media streams. (3)
- Asterisk will currently attempt to handle offers with AVPF with
- use_avpf=No/AVP with use_avpf=Yes. This mismatch results in
- invalid SDP answers being sent in response. If there is a
- mismatch between the media type being offered and the
- configuration, Asterisk must reject the offer with a 488. This
- patch does the following: * Asterisk will accept SDP offers with
- at least one media stream that it can use. Some WARNING messages
- have been dropped to NOTICEs as a result. * Asterisk will not
- accept an offer with a media type that doesn't match its
- configuration. * Asterisk will ignore declined media streams
- properly. #SIPit31 Review:
- https://reviewboard.asterisk.org/r/4063/ ASTERISK-24122 #close
- Reported by: James Van Vleet ASTERISK-24381 #close Reported by:
- Matt Jordan ........ Merged revisions 425868 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
- setting when sending qualify requests The outboundproxy setting
- is currently ignored when sending OPTIONS requests as a result of
- the qualify setting. This means that if an Asterisk server is
- unable to send the packet directly to a peer, it is unable to
- qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
- This patch grabs the outboundproxy information for a peer when a
- qualify attempt is being constructed and, if it finds the
- information, uses it when sending the OPTIONS request. Review:
- https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
- Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
- uploaded by Damian Ivereigh (License 6632) ........ Merged
- revisions 425818 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425819 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425820 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-17 02:41 +0000 [r425783] Richard Mudgett <rmudgett@digium.com>
-
- * main/core_unreal.c, main/channel.c, /: AMI: Add missing VarSet
- events when a channel inherits variables. There should be AMI
- VarSet events when channel variables are inherited by an outgoing
- channel. Also local;2 should generate VarSet events when it gets
- all of its channel variables from channel local;1. ASTERISK-24415
- #close Reported by: Richard Mudgett Patches:
- jira_asterisk_24415_v12.patch (license #5621) patch uploaded by
- Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/
- ........ Merged revisions 425782 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-17 01:57 +0000 [r425736-425761] Matthew Jordan <mjordan@digium.com>
-
- * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Fix audio
- issues when moving from remote bridge to softmix When a native
- RTP bridge that is remotely bridging its participants switches to
- a softmix bridge, it may not properly re-INVITE the media for one
- or both participants back to Asterisk. This is due to the current
- bridge_native_rtp code only re-INVITEs if it believes the channel
- will survive the bridge operation. Currently, that code is
- failing, as it expects the channels to have a soft hangup flag
- set on it indicating that a redirect has occurred or that the
- channel is going to leave the bridge. (The code did not take into
- account a smart bridge operation). This patch also renames a few
- things to be more reflective of the underlying types. Review:
- https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close
- ........ Merged revisions 425760 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, tests/test_cel.c: test_cel: Update pickup test to expect
- CANCEL instead of ANSWSER The CEL pickup test previously looked
- for a disposition of ANSWER between the original caller/peer when
- the call is picked up. This is actually incorrect: the
- disposition should, at the very least, not be ANSWER as the call
- was never ANSWERed. The disposition is now CANCEL; this patch
- updates the test accordingly. ........ Merged revisions 425757
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: main/cdr: Use 'time' when rescheduling batched
- CDRs as opposed to 'size' When refactoring CDRs to use the
- configuration framework, a 'whoops' was introduced where the CDR
- batch size was used when rescheduling a batch, as opposed to the
- time duration. This patch corrects that obvious mistake.
- ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged
- revisions 425735 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-16 17:30 +0000 [r425714] George Joseph <george.joseph@fairview5.com>
-
- * include/asterisk/config.h, tests/test_config.c, main/config.c, /:
- config: Fix inf loop using ast_category_browse and
- ast_variable_retrieve Fix infinite loop when calling
- ast_variable_retrieve inside an ast_category_browse loop when
- there is more than 1 category with the same name. Tested-by:
- George Joseph Review: https://reviewboard.asterisk.org/r/4089/
- ........ Merged revisions 425713 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-16 14:35 +0000 [r425691] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip_t38.c, res/res_pjsip_registrar_expire.c,
- res/res_pjsip_mwi_body_generator.c,
- res/res_pjsip_endpoint_identifier_user.c,
- res/res_pjsip_send_to_voicemail.c,
- include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_outbound_authenticator_digest.c,
- res/res_pjsip_outbound_registration.c,
- res/res_pjsip_endpoint_identifier_anonymous.c,
- res/res_pjsip_path.c, res/res_pjsip_one_touch_record_info.c,
- res/res_pjsip_acl.c, res/res_pjsip_pubsub.c,
- res/res_pjsip_diversion.c, res/res_pjsip_refer.c,
- include/asterisk/res_pjsip.h,
- res/res_pjsip_pidf_body_generator.c, res/res_pjsip_dtmf_info.c,
- res/res_pjsip_multihomed.c, res/res_pjsip_authenticator_digest.c,
- res/res_pjsip_sdp_rtp.c, res/res_hep_pjsip.c,
- res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
- res/res_pjsip_logger.c, res/res_pjsip_nat.c,
- res/res_pjsip_session.c, res/res_pjsip_exten_state.c,
- res/res_pjsip_header_funcs.c, res/res_pjsip_rfc3326.c,
- res/res_pjsip_phoneprov_provider.c, res/res_pjsip_mwi.c,
- res/res_pjsip_dialog_info_body_generator.c,
- res/res_pjsip_xpidf_body_generator.c, res/res_pjsip_registrar.c,
- channels/chan_pjsip.c, res/res_pjsip_transport_websocket.c,
- res/res_pjsip_pidf_eyebeam_body_supplement.c,
- include/asterisk/res_pjsip_session.h, /, res/res_pjsip_notify.c,
- res/res_pjsip_pidf_digium_body_supplement.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- res/res_pjsip_publish_asterisk.c: PJSIP: Enforce module load
- dependencies This enforces that res_pjsip, res_pjsip_session, and
- res_pjsip_pubsub have loaded properly before attempting to load
- any modules that depend on them since the module loader system is
- not currently capable of resolving module dependencies on its
- own. ASTERISK-24312 #close Reported by: Dafi Ni Review:
- https://reviewboard.asterisk.org/r/4062/ ........ Merged
- revisions 425690 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-16 06:11 +0000 [r425669] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * channels/chan_unistim.c, /: Fix loss of voice after second call
- drops (on a second line) in case using multiple lines on unistim
- phones. There is regression was introduced in r391379. Reported
- by: Rustam Khankishyiev (closes issue ASTERISK-23846) ........
- Merged revisions 425667 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425668 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-16 01:25 +0000 [r425646] Joshua Colp <jcolp@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix a bug where ICE
- state would get reset when it shouldn't. In the case where the
- ICE negotiation had not yet started current state would get wiped
- when it shouldn't. This also removes channel binding as in
- practice this does not work well with other implementations.
- ........ Merged revisions 425644 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425645 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-15 19:31 +0000 [r425627] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_motif.c: chan_motif: Cleanup
- jingle_tech.capabilities only once.
-
-2014-10-15 19:05 +0000 [r425611] Jonathan Rose <jrose@digium.com>
-
- * res/parking/parking_tests.c: parking_tests: Fix assertions and
- possibly crashes in res_parking unit tests Assertions were caused
- by attempting to play music on hold to a channel with no formats.
- Parking unit test channels were given formats and a technology so
- that they would be able to pretend to read/write frames.
- ASTERISK-24413 #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/4075/
-
-2014-10-15 09:59 +0000 [r425590] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c, /: chan_ooh323: fix rtptimeout general
- value checking correct condition to check rtptimeout in [general]
- config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
- Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch ........
- Merged revisions 425547 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425548 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425589 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 20:46 +0000 [r425526] George Joseph <george.joseph@fairview5.com>
-
- * /, include/asterisk/config.h, tests/test_config.c, main/config.c:
- config: Fix SEGV in unit test with MALLOC_DEBUG With MALLOC_DEBUG
- the /main/config config_basic_ops test was causing a SEGV while
- doing an ast_category_delete in an ast_category_browse loop.
- Apparently this never worked but was also never tested. I removed
- the test, added 2 notes to config.h indicating that it's not
- supported and added a few lines of code to ast_category_delete to
- prevent the SEGV should someone attempt it in the future.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4078/ ........ Merged
- revisions 425525 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 19:00 +0000 [r425504] Jonathan Rose <jrose@digium.com>
-
- * main/sched.c, /: Scheduler: Fix a nasty scheduler caching bug
- which makes new tasks not execute Tasks that were marked for
- pending deletion in the scheduler would be moved to the cache for
- later reuse, but after being recycled the deleted mark wouldn't
- be removed resulting in fresh tasks being deleted without
- reason... and immediately moved back into the cache where they
- could be reused again. This could cause horrendous things to
- happen in just about anything that used a scheduler.
- ASTERISK-24321 #close Reported by: Steve Pitts Review:
- https://reviewboard.asterisk.org/r/4071/ ........ Merged
- revisions 425503 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 18:12 +0000 [r425481] George Joseph <george.joseph@fairview5.com>
-
- * res/res_phoneprov.c, include/asterisk/phoneprov.h, /,
- res/res_pjsip_phoneprov_provider.c: res_phoneprov: Create
- accessor for ast_phoneprov_std_variable_lookup Based on feedback
- from Richard, I created an accessor for
- res_phoneprov/ast_phoneprov_std_variable_lookup and added load
- priority to AST_MODULE_INFO. Tested-by: George Joseph Tested-by:
- Richard Mudgett Review: https://reviewboard.asterisk.org/r/4076/
- ........ Merged revisions 425480 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 16:46 +0000 [r425459] Corey Farrell <git@cfware.com>
-
- * /, res/res_fax.c: res_fax: Fix reference leak caused by gateway
- sessions Fax gateway session objects can be re-used, causing the
- same gateway session to be added to faxregistry.container more
- than once. This change causes fax_session_new to remove the
- reserved session from the container before it's id is changed,
- ensuring it's possible for the session to be freed.
- ASTERISK-24392 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4049/ ........ Merged
- revisions 425457 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 16:35 +0000 [r425455] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/stasis_channels.c: stasis_channels.c: Resolve unfinished
- Dials when doing masquerades (Part 2) Masquerades into and out of
- channels that are involved in a dial operation don't create the
- expected dial end event. The missing dial end event goes against
- the model for things like CDRs and generating Dial end manager
- actions and such. There are four cases: 1) A channel masquerades
- into the caller channel. The case happens when performing a
- blonde transfer using the channel driver's protocol. 2) A channel
- masquerades into a callee channel. The case happens when
- performing a directed call pickup. 3) The caller channel
- masquerades out of dial. The case happens when using the Bridge
- application on the caller channel. 4) A callee channel
- masquerades out of dial. The case happens when using the Bridge
- application on a peer channel. As it turned out, all four cases
- need to be handled instead of just the first one. ASTERISK-24237
- Reported by: Richard Mudgett ASTERISK-24394 #close Reported by:
- Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/
- ........ Merged revisions 425430 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-14 16:19 +0000 [r425415] Corey Farrell <git@cfware.com>
-
- * /, res/res_fax.c: res_fax: Resolve module reference leak caused
- by reserved sessions Remove reference to module providing
- reserved session after adding a reference to the final module.
- This re-reference is done to ensure that module references are
- correct even if the final session selects a different module than
- the reserved session. ASTERISK-18923 #close Reported by: Grigoriy
- Puzankin Review: https://reviewboard.asterisk.org/r/4048/
- ........ Merged revisions 425405 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425407 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425411 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-13 16:10 +0000 [r425384] George Joseph <george.joseph@fairview5.com>
-
- * apps/app_directory.c, tests/test_sorcery.c, main/config.c,
- tests/test_sorcery_realtime.c, res/res_sorcery_realtime.c,
- apps/app_voicemail.c, res/res_sorcery_config.c, main/manager.c,
- /, include/asterisk/config.h, pbx/pbx_realtime.c,
- tests/test_config.c: manager/config: Support templates and
- non-unique category names via AMI This patch provides the
- capability to manipulate templates and categories with non-unique
- names via AMI. Summary of changes: GetConfig and GetConfigJSON:
- Added "Filter" parameter: A comma separated list of
- name_regex=value_regex expressions which will cause only
- categories whose variables match all expressions to be
- considered. The special variable name TEMPLATES can be used to
- control whether templates are included. Passing 'include' as the
- value will include templates along with normal categories.
- Passing 'restrict' as the value will restrict the operation to
- ONLY templates. Not specifying a TEMPLATES expression results in
- the current default behavior which is to not include templates.
- UpdateConfig: NewCat now includes options for allowing duplicate
- category names, indicating if the category should be created as a
- template, and specifying templates the category should inherit
- from. The rest of the actions now accept a filter string as
- defined above. If there are non-unique category names, you can
- now update specific ones based on variable values. To facilitate
- the new capabilities in manager, corresponding changes had to be
- made to config, most notably the addition of filter criteria to
- many of the APIs. In some cases it was easy to change the
- references to use the new prototype but others would have
- required touching too many files for this patch so a wrapper with
- the original prototype was created. Macros couldn't be used in
- this case because it would break binary compatibility with
- modules such as res_digium_phone that are linked to real symbols.
- Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4033/ ........ Merged
- revisions 425383 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-12 21:09 +0000 [r425362] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE
- transport check case insensitive as some implementations use
- 'udp'. ........ Merged revisions 425360 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425361 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-12 08:15 +0000 [r425289-425299] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_sip.c: chan_sip: Fix so asterisk won't send
- reINVITE after a BYE. After a reINVITE glare situation, Asterisk
- would re-send the reINVITE even though the call had been hung up
- in the mean time. This patch unschedules the reinvite when
- handling the BYE. ASTERISK-22791 #close Reported by: Paolo
- Compagnini Tested by: Paolo Compagnini Review:
- https://reviewboard.asterisk.org/r/4056/ (testcase is in review
- r4055) ........ Merged revisions 425296 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425297 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425298 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, Makefile: build: Relax badshell tilde test to allow for ~ in
- middle of DESTDIR. The main Makefile has a target test called
- 'badshell' that tests if DESTDIR does not happen to have an
- an-expanded tilde (~). This might be the case if you run: make
- install DESTDIR=~/somewhere/ That test also disallowed valid
- tildes in directory names. The test is now changed to only
- trigger on a tilde at the start of the path. ASTERISK-13797
- #close Reported by: Tzafrir Cohen Review:
- https://reviewboard.asterisk.org/r/4064/ ........ Merged
- revisions 425291 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425292 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425293 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_calendar_ews.c: res_calendar_ews: Relax neon version
- check to work with 0.30 too. Allow res_calendar_ews to work not
- only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close
- Reported by: Tzafrir Cohen Review:
- https://reviewboard.asterisk.org/r/4068/ ........ Merged
- revisions 425286 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425287 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425288 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-11 21:08 +0000 [r425265] George Joseph <george.joseph@fairview5.com>
-
- * /, res/res_phoneprov.c: res_phoneprov: Cleanup module load error
- handling Tested module load/reload interaction between
- res_phoneprov and res_pjsip_phoneprov_provider in cases where
- res_phoneprov didn't load correctly (usually misconfiguration or
- missing phoneprov.conf) Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/4069/ ........ Merged
- revisions 425264 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 20:48 +0000 [r425243] Joshua Colp <jcolp@digium.com>
-
- * /, main/bridge.c, bridges/bridge_native_rtp.c: bridge: During a
- smart bridge operation provide a more complete bridge to the old
- technology. When a smart bridge operation occurs and a bridge
- transitions from one technology to another the old technology is
- provided the channels formerly in it and told that they are
- leaving. Unfortunately the bridge provided along with them is
- incomplete. The bridge, despite there being channels in it,
- contains none. This forces technology implementations to have
- additional logic when channels are leaving or to store their own
- duplicated state. This change makes the bridge more complete so
- it contains the expected channels. Now that the bridge is
- complete special logic within bridge_native_rtp is no longer
- needed and has been removed. Review:
- https://reviewboard.asterisk.org/r/4057/ ........ Merged
- revisions 425242 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 14:31 +0000 [r425221] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_phoneprov.c: res/res_phoneprov: Bail on registration
- if res_phoneprov didn't load If res_phoneprov failed to fully
- load (due to not being configured), the providers container will
- be NULL. If a module attempts to register a phone provisioning
- provider, it should check for the presence of the container. If
- there is no providers container, it should return an error. This
- patch makes the ast_phoneprov_provider_register function do
- that... otherwise this would be a silly commit message. ........
- Merged revisions 425220 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 14:23 +0000 [r425217] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_phoneprov_provider.c:
- res_pjsip_phoneprov_provider: Add missing dependency on
- pjproject. ........ Merged revisions 425216 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 13:01 +0000 [r425155] Kinsey Moore <kmoore@digium.com>
-
- * /, tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing
- regression This fixes a regression in callerid parsing introduced
- when another bug was fixed. This bug occurred when the name was
- composed entirely of DTMF keys and quoted without a number
- section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard
- Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by
- Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/
- ........ Merged revisions 425152 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425153 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425154 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 12:10 +0000 [r425132] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_nat.c, /: res_pjsip_nat: Place source port into
- rport of responses if 'force_rport' is on. When the 'force_rport'
- option is enabled the behavior should be the same as if the
- remote side placed rport into the message themselves. Therefore
- any responses we send should include the source port of the
- request in the rport of the Via header. #SIPit31 ASTERISK-24387
- #close Reported by: Matt Jordan ........ Merged revisions 425131
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-10 07:32 +0000 [r425071] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_sip.c: chan_sip: Fix dialog leak resulting from
- missing ACK to re-INVITE. If a device re-INVITEs at the same time
- as the dialog is hung up, and if then the ACK to the re-INVITE
- never reaches Asterisk, chan_sip would fail to destroy the dialog
- after a while. This resulted in (most prominently) file handle
- leaks. (Patch reindented by me.) ASTERISK-20784 #close
- ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal
- Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle
- (License #5334) patch_asterisk_20784.txt uploaded by Nitesh
- Bansal (License #6418) Reviewboard:
- https://reviewboard.asterisk.org/r/4052/ (testcase can be found
- at r4051) ........ Merged revisions 425068 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 425069 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 425070 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 23:35 +0000 [r425052] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip_phoneprov_provider.c: res_pjsip_phoneprov_provider:
- fix compile breakage on AST_VECTOR endpoint->inbound_auths was
- changed to a vector in 13 and I committed the 12 patch instead of
- the 13 patch. Tested-by: George Joseph
-
-2014-10-09 21:38 +0000 [r425031] Kevin Harwell <kharwell@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Crash if no
- candidates received for component When starting ice if there is
- not at least one remote ice candidate with an RTP component
- asterisk will crash. This is due to an assertion in pjnath as it
- expects at least one candidate with an RTP component. Added a
- check to make sure at least one candidate contains an RTP
- component and at least one candidate has an RTCP component.
- ASTERISK-24383 #close Review:
- https://reviewboard.asterisk.org/r/4039/ ........ Merged
- revisions 425030 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 20:54 +0000 [r425008] George Joseph <george.joseph@fairview5.com>
-
- * /, res/res_pjsip_phoneprov_provider.c (added),
- configs/samples/pjsip.conf.sample: res_pjsip_phoneprov_provider:
- Provides pjsip integration with res_phoneprov This module allows
- res_pjsip to integrate with res_phoneprov. It handles the pjsip
- 'phoneprov' object type. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3976/ ........ Merged
- revisions 425007 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 18:37 +0000 [r424986] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_phoneprov.c: res/res_phoneprov: Don't cancel Asterisk
- load on module load failure ........ Merged revisions 424985 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 17:45 +0000 [r424964] George Joseph <george.joseph@fairview5.com>
-
- * include/asterisk/phoneprov.h (added), /,
- configs/samples/phoneprov.conf.sample,
- include/asterisk/chanvars.h, res/res_phoneprov.c,
- res/res_phoneprov.exports.in (added), main/chanvars.c:
- res_phoneprov: Refactor phoneprov to allow pluggable config
- providers This patch makes res_phoneprov more modular so other
- modules (like pjsip) can provide configuration information
- instead of res_phoneprov relying solely on users.conf and
- sip.conf. To accomplish this a new ast_phoneprov public API is
- now exposed which allows config providers to register themselves,
- set defaults (server profile, etc) and add user extensions. *
- ast_phoneprov_provider_register registers the provider and
- provides callbacks for loading default settings and loading
- users. * ast_phoneprov_provider_unregister clears the defaults
- and users. * ast_phoneprov_add_extension should be called once
- for each user/extension by the provider's load_users callback to
- add them. * ast_phoneprov_delete_extension deletes one extension.
- * ast_phoneprov_delete_extensions deletes all extensions for the
- provider. Tested-by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3970/ ........ Merged
- revisions 424963 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 16:36 +0000 [r424942] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/cdr.c: cdr.c: Make turning on CDR debug a one step
- process instead of two. Now "cdr set debug on" doesn't also
- require "core set verbose 1" to see CDR debug output. ........
- Merged revisions 424941 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-09 08:08 +0000 [r424880] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, contrib/scripts/safe_asterisk: safe_asterisk: Don't
- automatically exceed MAXFILES value of 2^20. On systems with lots
- of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can
- exceed the per-process file limit of 2^20. This patch ensures the
- value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close
- Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff
- uploaded by Michael Myles (License #6626) ........ Merged
- revisions 424875 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 424878 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424879 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-08 18:46 +0000 [r424854] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE
- candidates. The underlying library, pjnath, that res_rtp_asterisk
- uses for ICE support does not have support for ICE-TCP. As
- candidates are passed through directly to it this can cause error
- messages to occur when it receives something unexpected (such as
- a TCP candidate). This change merely ignores all non-UDP
- candidates so they never reach pjnath. ASTERISK-24326 #close
- Reported by: Joshua Colp ........ Merged revisions 424852 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424853 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-08 18:24 +0000 [r424769-424850] Kinsey Moore <kmoore@digium.com>
-
- * main/stasis.c: Stasis: Relegate log message to dev-mode This
- error message primarily applies to development tasks and will now
- only show up when dev-mode is enabled via configure.
-
- * main/sounds_index.c: Indexer: Format message types may not exist
- In Asterisk 13+, any given message type is not guaranteed to
- exist even if Asterisk comes up correctly since creation of the
- message type could be declined. The indexer should not prevent
- Asterisk from starting under these conditions.
-
- * main/stasis.c: Stasis: Only log errors for non-declined types
- When message type creation is declined via stasis.conf, certain
- operations log errors assuming that the declined type is being
- used before initialization or after destruction. These error
- messages get quite spammy for oft used message types and should
- not be logged in the first place since the message type is
- validly NULL. Reported by: Matt DiMeo
-
-2014-10-07 18:33 +0000 [r424752] Joshua Colp <jcolp@digium.com>
-
- * main/data.c: data: Properly access formats in capabilities
- structure when adding codecs. Formats within a capabilities
- structure are addressed starting at 0, not 1. Assuming 1 causes
- it to exceed an array. ASTERISK-24389 #close Reported by: Kevin
- Harwell
-
-2014-10-07 17:41 +0000 [r424692-424731] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c:
- res/res_pjsip_outbound_registration: Initialize
- auth_reject_permanent parameter Prior to this patch, the
- auth_reject_permanent parameter was not initialized on the
- registration client state, leading to the parameter being
- disabled regardless of the value specified in pjsip.conf. This
- patch initialized the setting on the registration client state to
- the provided configuration value. ASTERISK-24398 #close ........
- Merged revisions 424730 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Fix typo in WARNING
- message
-
- * main/message.c, /: message: Don't close an AMI connection on
- SendMessage action error If SendMessage encounters an error (such
- as incorrect input provided to the action), it will currently
- return -1. Actions should only return -1 if the connection to the
- AMI client should be closed. In this case, SendMessage causing
- the client to disconnect is inappropriate. This patch causes the
- action to return 0, which simply causes the action to fail.
- Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354
- #close Reported by: Peter Katzmann patches: sendMessage.patch
- uploaded by Peter Katzmann (License 5968) ........ Merged
- revisions 424690 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424691 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-06 15:38 +0000 [r424669] Richard Mudgett <rmudgett@digium.com>
-
- * main/features.c, /: features.c: Fix lingering channel ref while
- Bridge() application is active. Using the Bridge application to
- bridge a channel that is executing an applicaiton such as Wait
- results in a lingering Surrogate channel in the CLI "core show
- channels" output even though it has already hungup. * Fix
- bridge_exec() to not hold onto the current_dest_chan ref once it
- has been put into the bridge. * Eliminated bridge_exec()'s use of
- RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson
- Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged
- revisions 424668 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-06 12:38 +0000 [r424601-424647] Matthew Jordan <mjordan@digium.com>
-
- * /, main/sdp_srtp.c: sdp_srtp: Add new lines to some WARNING
- messages ........ Merged revisions 424646 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_options.c: res_pjsip/pjsip_options: Do not
- 404 an OPTIONS request not sent to an endpoint An OPTIONS request
- that is sent to Asterisk but not to a specific endpoint is
- currently sent a 404 in response. This is because, not
- surprisingly, an empty extension is never going to be found in
- the dialplan. This patch makes it so that we only attempt to look
- up the endpoint in the dialplan if it is specified in the OPTIONS
- request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt
- Jordan ........ Merged revisions 424624 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions:
- Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels Calling
- PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your
- health. It will treat the channels as a PJSIP channel, eventually
- hitting an ao2 error, FRACKing on assertion error, and quite
- likely crashing. This patch adds checks to the read/write
- callbacks that ensure that the channel technology is of type
- 'PJSIP' before attempting to operate on the channel. #SIPit31
- ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged
- revisions 424621 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_hep_pjsip.c, res/res_pjsip/pjsip_distributor.c,
- res/res_pjsip_logger.c: res_pjsip: Prevent crashes when PJPROJECT
- presents an rdata with no message When a message that exceeds the
- PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible
- (although it shouldn't occur) for pjproject to pass up an rdata
- object with a NULL msg in the msg_info. Needless to say, things
- that attempt to dereference this are in for a rough ride. In
- particular, this caused crashes in three different locations, all
- of which are 'low level' enough to intercept an rdata object
- early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3)
- res_pjsip/distributor Anything that can intercept an rdata object
- before res_pjsip/distributor should be defensive when looking at
- the received packet. #SIPit31 ASTERISK-24369 #close Reported by:
- Matt Jordan ........ Merged revisions 424618 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pubsub.c: res/res_pjsip_pubsub: Gracefully handle
- errors when re-creating subscriptions A subscription that has
- been persisted can - for various reasons - fail to be re-created
- on startup. This patch resolves a number of crashes that occurred
- when a subscription cannot be re-created on several off-nominal
- paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan
-
-2014-10-05 00:48 +0000 [r424552-424580] Corey Farrell <git@cfware.com>
-
- * main/manager.c, /: Release AMI connections on shutdown.
- ASTERISK-24378 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4037/ ........ Merged
- revisions 424578 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424579 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_motif.c: chan_motif: Correct last commit to use
- ao2_cleanup to free format cap This fix applies to 13 and trunk.
- ASTERISK-24384 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4043/
-
- * /, channels/chan_motif.c: chan_motif: Release format capabilities
- and config on module load error ASTERISK-24384 #close Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/4043/ ........ Merged
- revisions 424550 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424551 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-03 21:56 +0000 [r424472-424529] Richard Mudgett <rmudgett@digium.com>
-
- * /, CHANGES, res/res_pjsip.c: res_pjsip: Fix XML typo and update
- CHANGES. ASTERISK-24199 ........ Merged revisions 424528 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, /,
- main/framehook.c: audiohooks: Reevaluate the bridge technology
- when an audiohook is added or removed. Adding a mixmonitor to a
- channel causes the bridge to change technologies from native to
- simple_bridge so the call can be recorded. However, when the
- mixmonitor is stopped the bridge does not switch back to the
- native technology. * Added unbridge requests to reevaluate the
- bridge when a channel audiohook is removed. * Moved the unbridge
- request into ast_audiohook_attach() ensure that the bridge
- reevaluates whenever an audiohook is attached. This simplified
- the mixmonitor and chan_spy start code as well. * Added defensive
- code to stop_mixmonitor_full() in case additional arguments are
- ever added to the StopMixMonitor application. * Made
- ast_framehook_detach() not do an unbridge request if the
- framehook does not exist. * Made ast_framehook_list_fixup() do an
- unbridge request if there are any framehooks. Also simplified the
- loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/4046/ ........ Merged
- revisions 424506 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/core_unreal.c, main/taskprocessor.c, channels/chan_iax2.c,
- res/res_pjsip_session.c, main/channel.c, channels/chan_misdn.c,
- channels/chan_skinny.c, funcs/func_frame_trace.c,
- channels/chan_motif.c, include/asterisk/frame.h,
- main/bridge_channel.c, channels/chan_pjsip.c,
- channels/chan_unistim.c, include/asterisk/res_pjsip_session.h,
- addons/chan_ooh323.c, /, include/asterisk/taskprocessor.h,
- channels/chan_sip.c, res/res_pjsip_session.exports.in:
- chan_pjsip: Fix deadlock when masquerading PJSIP channels.
- Performing a directed call pickup resulted in a deadlock when
- PJSIP channels were involved. A masquerade needs to hold onto the
- channel locks while it swaps channel information between the two
- channels involved in the masquerade. With PJSIP channels, the
- fixup routine needed to push a fixup task onto the PJSIP
- channel's serializer. Unfortunately, if the serializer was also
- processing a task that needed to lock the channel, you get
- deadlock. * Added a new control frame that is used to notify the
- channels that a masquerade is about to start and when it has
- completed. * Added the ability to query taskprocessors if the
- current thread is the taskprocessor thread. * Added the ability
- to suspend/unsuspend the PJSIP serializer thread so a masquerade
- could fixup the PJSIP channel without using the serializer.
- ASTERISK-24356 #close Reported by: rmudgett Review:
- https://reviewboard.asterisk.org/r/4034/ ........ Merged
- revisions 424471 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-03 15:54 +0000 [r424448] George Joseph <george.joseph@fairview5.com>
-
- * /, main/sorcery.c: sorcery: Prevent SEGV in sorcery_wizard_create
- when there's no create function When you call
- ast_sorcery_create() you don't necessarily know which wizard is
- going to be invoked. If it happens to be a wizard like 'config'
- that doesn't have a 'create' virtual function you get a segfault
- in the sorcery_wizard_create callback. This patch catches the
- null function pointer, does an ast_assert, and logs an error.
- Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged
- revisions 424447 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-03 13:58 +0000 [r424424-424427] Kinsey Moore <kmoore@digium.com>
-
- * configs/samples/pjsip.conf.sample, /,
- res/res_pjsip/pjsip_configuration.c: PJSIP: Restore functional
- default for callerid_privacy The pjsip config option default
- fixups from r424263 altered the functional default from
- "allowed_not_screened" to "allowed". This change restores the
- functional default value when none is provided. ........ Merged
- revisions 424426 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: Manager: Add missing fields and documentation
- for CoreShowChannels This corrects some issues introduced in the
- responses to the CoreShowChannels AMI command as well as adding
- documentation for the responses. The command in Asterisk 12 was
- missing the following fields: Duration, Application,
- ApplicationData, and BridgedChannel and BridgedUniqueID (replaced
- with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn
- Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged
- revisions 424423 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-03 07:54 +0000 [r424415] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Reduce SDP size by
- removing duplicate connection lines. Due to the architecture of
- how media streams are handled each individual handler adds
- connection details (IP address) for it. The first media stream is
- then used as the top level SDP connection line. In practice each
- line ends up being the same so to reduce the SDP size
- stream-level connection information is also added to the SDP if
- it differs from the top level SDP connection line. ........
- Merged revisions 424414 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-02 21:52 +0000 [r424394] Richard Mudgett <rmudgett@digium.com>
-
- * /, configs/samples/pjsip.conf.sample, res/res_pjsip.c,
- res/res_pjsip/config_transport.c: res_pjsip: Make transport
- cipher option accept a comma separated list of cipher names.
- Improvements to the res_pjsip transport cipher option. * Made the
- cipher option accept a comma separated list of OpenSSL cipher
- names. Users of realtime will be glad if they have more than one
- name to list. * Added the CLI command 'pjsip list ciphers' so a
- user can know what OpenSSL names are available for the cipher
- option. * Updated the cipher option online XML documentation to
- specify what is expected for the value. * Updated
- pjsip.conf.sample to not indicate that ALL is acceptable since
- ALL does not imply a preference order for the ciphers and PJSIP
- does not simply pass the string to OpenSSL for interpretation.
- ASTERISK-24199 #close Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/4018/ ........ Merged
- revisions 424393 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-02 20:15 +0000 [r424373] Jonathan Rose <jrose@digium.com>
-
- * /,
- contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py
- (added): Alembic: Add enumerator value to sippeers -> directmedia
- - 'outgoing' The 'outgoing' value was left off of the enumerator
- when first creating the column. This patch adds it, and should
- gracefully upgrade keeping the existing data in tact.
- ASTERISK-23781 #close Reported by: Stephen More Review:
- https://reviewboard.asterisk.org/r/4013/ ........ Merged
- revisions 424372 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-02 13:35 +0000 [r424338] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, configs/samples/pjsip.conf.sample: res_pjsip: document use of
- rewrite_contact in sample conf Without setting rewrite_contact,
- an invite to an endpoint behind NAT will not reach it - unless
- the endpoint itself uses STUN or TURN to discover it's public
- URI. Thus, the use of this should be in the sample documentation.
- Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged
- revisions 424337 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-01 22:52 +0000 [r424333] Jonathan Rose <jrose@digium.com>
-
- * channels/chan_pjsip.c: chan_pjsip: Fix an assertion for channels
- that lack formats on creation ASTERISK-24222 #close Reported by:
- Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/
-
-2014-10-01 20:36 +0000 [r424313] Corey Farrell <git@cfware.com>
-
- * res/res_hep.c, /: res_hep: Release allocation reference to
- configuration. ASTERISK-24362 #close Reported by: Corey Farrell
- Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged
- revisions 424312 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-01 16:37 +0000 [r424288-424291] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/pjsip_configuration.c,
- configs/samples/pjsip.conf.sample, res/res_pjsip.c: res_pjsip:
- Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
- During the latest update to DTLS-SRTP support the ability to
- configure the hash used for fingerprints was added. This gave us
- two supported ones: SHA-1 and SHA-256. The default was
- accordingly updated to SHA-256. Unfortunately this configuration
- ability was not exposed within res_pjsip. This change adds a
- dtls_fingerprint option that controls it. #SIPit31 ........
- Merged revisions 424290 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Accept DTLS
- attributes in top level, not just media session. #SIPit31
- ........ Merged revisions 424287 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-01 12:27 +0000 [r424245-424266] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip/config_transport.c, /, res/res_pjsip/location.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- res/res_pjsip/pjsip_configuration.c,
- configs/samples/pjsip.conf.sample: PJSIP: Handle defaults
- properly This updates the code behind PJSIP configuration options
- with custom handlers to deal with the assigned default values
- properly where it makes sense and adjusting the default value
- where it doesn't. Before applying this patch, there were several
- cases where the default value for an option would prevent that
- config section from loading properly. Reported by: Thomas
- Thompson Review: https://reviewboard.asterisk.org/r/4019/
- ........ Merged revisions 424263 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_nat.c: PJSIP: Force transport on contact rewrite
- If contact rewriting is enabled but the contact differs in
- transport from what is actually being used, messages after the
- initial INVITE transaction can be sent to an incorrect
- transport/port combination. In the case where this bug occurred
- the remote party never received a BYE since it was sent to the
- remote party's TCP port over UDP. Review:
- https://reviewboard.asterisk.org/r/4032/ ........ Merged
- revisions 424244 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-10-01 10:09 +0000 [r424179-424184] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_sip.c: chan_sip: Simplify some unref code by
- removing unlink_peer_from_tables. ASTERISK-22945 #related
- Reported by: ibercom Patches:
- asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License
- #6599) ........ Merged revisions 424181 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 424182 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424183 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip: Remove excess ref of realtime
- peer before sip_poke_peer. The peer is referenced at the end of
- sip_poke_peer, it should not get an extra ref before the call to
- sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close
- Reported by: ibercom Tested by: Yuriy Gorlichenko Patches:
- asterisk11.patch uploaded by ibercom (License #6599) Review:
- https://reviewboard.asterisk.org/r/4031/ ........ Merged
- revisions 424176 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 424177 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424178 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-30 11:40 +0000 [r424153-424156] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't place an
- extra whitespace before 'rport' and don't put IPv6 addresses in
- brackets. #SIPit31 ........ Merged revisions 424155 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the base
- and mapped address for candidates is present in SDP. This change
- fixes an issue where ICE candidates put into the SDP did not
- contain the 'raddr' and 'rport' information for server reflexive
- and relay candidates. #SIPit31 ........ Merged revisions 424151
- from http://svn.asterisk.org/svn/asterisk/branches/11 ........
- Merged revisions 424152 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-29 21:59 +0000 [r424129] George Joseph <george.joseph@fairview5.com>
-
- * /, res/res_pjsip/pjsip_cli.c: pjsip_cli: Suppress header print on
- error or no objects If there's an error on the pjsip command line
- or there are no objects, don't print the column headers.
- ASTERISK-24350 #close Reported-by: Brad Latus Tested-by: George
- Joseph Tested-by: Brad Latus Review:
- https://reviewboard.asterisk.org/r/4025/ ........ Merged
- revisions 424128 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-29 21:26 +0000 [r424126] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, contrib/scripts/autosupport: autosupport: Fix bashism. '==' is
- bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
- 'case' works better there. Originally committed in r375059 and
- r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported
- by: Tzafrir Cohen ........ Merged revisions 424117 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 424125 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-29 21:17 +0000 [r424097-424105] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
- /, res/res_pjsip_authenticator_digest.c: Simplify UUID generation
- in several places. Replace code using ast_uuid_generate() with
- simpler and faster code using ast_uuid_generate_str(). The new
- code avoids a malloc(), free(), and copy. ........ Merged
- revisions 424103 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/threadpool.c: threadpool.c: Minor cleanup fixes. * Fix
- threadpool_alloc() prototype. * Add missing off-nominal NULL
- check of pool in threadpool_alloc(). * searializer_create() does
- not need to create the object with a lock as the lock is not
- used. ........ Merged revisions 424096 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-27 12:43 +0000 [r424057] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_pjsip.c, res/res_pjsip_session.c, /:
- res_pjsip_session: Add additional checks for delaying session
- refreshes. There are certain situations which no checks existed
- for which need to prevent session refreshes. This includes
- sending a session refresh with SDP before SDP negotiation has
- completed and sending a session refresh before the dialog itself
- has been established. Checks for these have been added.
- Additionally COLP related UPDATEs were including SDP when it is
- not needed. Review: https://reviewboard.asterisk.org/r/4008/
- ........ Merged revisions 424056 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-26 15:21 +0000 [r423992] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_fax.c: res_fax: Fix out of bounds error in
- update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy
- Laine Patches: res_fax_bounds.patch (license #6561) patch
- uploaded by Jeremy Laine Modified patch to not use magic numbers.
- ........ Merged revisions 423979 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423983 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423987 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-26 08:25 +0000 [r423918] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, doc/asterisk.8: docs: Escape unescaped minus sign in
- asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy
- Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy
- Lainé (License #6561) ........ Merged revisions 423915 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423916 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423917 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-25 21:01 +0000 [r423895] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip.c, /: res_pjsip.c: Add missing off nominal cleanup
- in ast_sip_push_task_synchronous(). * Made memset the std struct
- in ast_sip_push_task_synchronous() because if DEBUG_THREADS is
- enabled then uninitialized lock tracking data is used. ........
- Merged revisions 423894 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-24 18:32 +0000 [r423867] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c:
- pjsip_options.c: Fix race condition stopping periodic out of
- dialog OPTIONS request. The crash on the issues is a result of an
- invalid transport configuration change when asterisk is
- restarted. The attempt to send the qualify request fails and we
- cleaned up. However, the callback is also called which results in
- a double unref of the objects involved. * Put a wrapper around
- pjsip_endpt_send_request() to detect when the passed in callback
- is called because of an error so callers can know to not cleanup.
- * Made send_request_cb() able to handle repeated challenges (Up
- to 10). * Fix periodic endpoint qualify OPTIONS sched deletion
- race by avoiding it. The sched entry will no longer self stop and
- must be externally stopped. * Added REF_DEBUG description tags to
- struct sched_data in pjsip_options.c. * Fix some off-nominal ref
- leaks in schedule_qualify(), qualify_and_schedule(). * Reordered
- pjsip_options.c module start/stop code to cleanup better on
- error. ASTERISK-24295 #close Reported by: Rogger Padilla Review:
- https://reviewboard.asterisk.org/r/3954/ ........ Merged
- revisions 423866 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-24 08:53 +0000 [r423803] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_sip.c: chan_sip: Unref outbound proxy structure
- on dialog/pvt destruction. Make sure outbound proxy refs are
- always unreffed on dialog destruction. Review:
- https://reviewboard.asterisk.org/r/4016/ ........ Merged
- revisions 423800 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423801 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423802 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-23 14:29 +0000 [r423783] Mark Michelson <mmichelson@digium.com>
-
- * tests/test_cel.c, tests/test_cdr.c: Make CDR and CEL unit tests
- less FRACKy. Prior to this commit, CDR and CEL tests were
- expected to trigger FRACKs (i.e. assertions) due to the fact that
- the channels they create have no formats on them. Some code was
- independently added recently that attempts to prevent FRACKs from
- occurring by failing early when attempting to set up translation
- paths if one or both channels support no formats. Unfortunately,
- this attempt to be helpful made the CDR and CEL tests go from
- simply FRACKing to outright failing and in some cases, failing so
- badly as to crash Asterisk. This commit seeks to correct past
- mistakes by adding the ulaw format to channels created by the CDR
- and CEL unit tests. This makes setting up translation paths
- succeed, eliminates previously-seen FRACKs, and ultimately causes
- the unit tests to succeed again. Review:
- https://reviewboard.asterisk.org/r/4014
-
-2014-09-22 19:48 +0000 [r423660-423723] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_sip.c: chan_sip: On INVITE retransmission, don't
- add an extra 503 response. INVITE arrives to asterisk, asterisk
- responds Busy(). If the INVITE is retransmitted, asterisk would
- generate a 503 in addition to the 486. Thanks Torrey Searle for
- providing a working regression test. ASTERISK-24335 #close
- Review: https://reviewboard.asterisk.org/r/4003/ Patches:
- retrans_486_invite.patch uploaded by Torrey Searle (License
- #5334) ........ Merged revisions 423720 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423721 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423722 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/editline/readline.c: cli.c: Fix tab completion "module
- load" when MALLOC_DEBUG is enabled. r421600 conflicted with
- r155763. ASTERISK-24348 #close ........ Merged revisions 423657
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 423658 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423659 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-21 01:15 +0000 [r423618-423641] Matthew Jordan <mjordan@digium.com>
-
- * main/channel.c: main/channel: Unlock channel in off-nominal path
- In r423414 (13) / r423415 (trunk), an API call that determines if
- a format capability structure is empty was added. This returns
- true if the format capability structure is completely empty or
- "none". A check for this was added in channel.c's set_format
- call. Unfortunately, when this check was true, it returned from
- the function while still holding the channel lock. This caused
- the CDR unit tests - which have a tendency to create channels
- with no formats - to deadlock. Whoops. This patch unlocks the
- channel on the off-nominal path.
-
- * rest-api/api-docs/events.json, /: rest-api/api-docs/events.json:
- Remove non-compliant 'extends' attribute Prior to the release of
- Swagger 1.2, the attribute 'extends' was being promoted as a
- possible way to show that a particular object extends an existing
- object. Instead, the Swagger specification went with the
- 'subTypes' attribute in the base object. This patch removes the
- unsupported attribute; the object that the offending objects
- proposed to extend already lists them in its 'subTypes'
- attribute. ASTERISK-24300 #close Reported by: Bradley Watkins
- ........ Merged revisions 423620 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json,
- rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json,
- rest-api/api-docs/playbacks.json: rest-api/api-docs: Correct
- basePath in resources to match top resources file The
- resources.json file that defines the resource JSON files used
- with ARI references a basePath of 'http://localhost:8088/ari'.
- This does not match what is defined in the resource files
- themselves, 'http://localhost:8088/stasis'. The correct base path
- is the one that includes 'ari' in the URL; this patch updates the
- various resource JSON files to have the correct basePath.
- ASTERISK-24339 #close Reported by: Bradley Watkins ........
- Merged revisions 423617 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-19 19:51 +0000 [r423580] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on
- unload/load and don't say the module doesn't exist on reload.
- When unloading the module did not unregister the CLI commands
- causing a crash upon load when they were registered again. When
- reloading the module the return value from the config options
- framework was not checked to determine if an error occurred or
- not. This caused a message to be output saying the module did not
- exist when reloading if no changes were present. AST-1433 #close
- AST-1434 #close ........ Merged revisions 423579 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-19 17:08 +0000 [r423561] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c:
- res_pjsip_sdp_rtp.c: Fix native formats containing formats that
- were not negotiated. Outgoing PJSIP calls can result in
- non-negotiated formats listed in the channel's native formats if
- video formats are listed in the endpoint's configuration. The
- resulting call could then use a non-negotiated format resulting
- in one way audio. * Simplified the update of session->req_caps in
- set_caps(). Why do something in five steps when only one is
- needed? AFS-162 #close Review:
- https://reviewboard.asterisk.org/r/4000/
-
-2014-09-19 15:18 +0000 [r423524-423530] Jonathan Rose <jrose@digium.com>
-
- * /, main/stasis_channels.c: Stasis_channels: Resolve unfinished
- Dials when doing masquerades Masquerades into channels that are
- in the dialing state don't end their dial and this goes against
- the model for things like CDRs and generating Dial end manager
- actions and such. ASTERISK-24237 #close Reported by: Richard
- Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........
- Merged revisions 423525 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2
- jitterbuffer settings Caused by format changes in Asterisk 13
- ASTERISK-24265 #close Reported by: Dafi Ni Review:
- https://reviewboard.asterisk.org/r/3999/
-
-2014-09-19 12:45 +0000 [r423504] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/framehook.h, /, main/framehook.c,
- res/res_pjsip_t38.c: PJSIP: Prevent T38 framehook being put on
- wrong channel This change gives framehooks a reverse-direction
- masquerade callback in addition to chan_fixup_cb similar to the
- callback added to datastores to handle the same situation. The
- new callback provides the same parameters as the fixup callback,
- but is called on the new channel's framehooks before moving
- framehooks from the old channel to the new channel. This gives
- the framehooks an oppurtunity to decide whether they should
- remain on the new channel or be removed. This new callback is
- used to prevent the PJSIP T.38 framehook from remaining on a
- masqueraded channel if the new channel is not also a PJSIP
- channel. This was causing a crash when a local channel was
- masqueraded into a PJSIP channel and the framehook was executed
- on the local channel since the channel's tech private data was
- not structured as expected. Review:
- https://reviewboard.asterisk.org/r/4001/ ........ Merged
- revisions 423503 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 19:30 +0000 [r423482] Sean Bright <sean@malleable.com>
-
- * res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a
- password when doing userpass authentication. An empty password is
- valid for username/password authentication so we should allow
- password to be empty/not supplied. Review:
- https://reviewboard.asterisk.org/r/3988 ........ Merged revisions
- 423481 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 19:22 +0000 [r423478] George Joseph <george.joseph@fairview5.com>
-
- * tests/test_strings.c, /, main/utils.c,
- include/asterisk/strings.h: utils: Create ast_strsep function
- that ignores separators inside quotes This function acts like
- strsep with three exceptions... * The separator is a single
- character instead of a string. * Separators inside quotes are
- treated literally instead of like separators. * You can elect to
- have leading and trailing whitespace and quotes stripped from the
- result and have '\' sequences unescaped. Like strsep, ast_strsep
- maintains no internal state and you can call it recursively using
- different separators on the same storage. Also like strsep, for
- consistent results, consecutive separators are not collapsed so
- you may get an empty string as a valid result. Tested by: George
- Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........
- Merged revisions 423476 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 18:31 +0000 [r423462] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c: Add subscription state test events. These
- are needed for a set of batched notification RLS tests that are
- about to be committed to the testsuite. Review:
- https://reviewboard.asterisk.org/r/3967
-
-2014-09-18 17:11 +0000 [r423425] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_endpoint_identifier_ip.c, /:
- res_pjsip_endpoint_identifier_ip: Fix parsing of match value with
- CIDR Also fixes comma separates match lists ASTERISK-24290 #close
- Reported by: Ray Crumrine Review:
- https://reviewboard.asterisk.org/r/3995/ ........ Merged
- revisions 423417 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 17:09 +0000 [r423418-423423] Richard Mudgett <rmudgett@digium.com>
-
- * bridges/bridge_softmix.c: bridge_softmix.c: Made use
- ao2_replace() instead of the inline equivalent. * Clarified some
- read/write format comments. * Fixed a doxygen tag typo.
-
- * main/astobj2.c, contrib/scripts/refcounter.py, /:
- astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
- Make astob2 REF_DEBUG output an invalid object line when an
- invalid ao2 object ref/unref is attempted. This is similar to the
- constructor/destructor lines. * Fixed refcounter.py to handle
- skewed objects that have constructor/destructor states. * Made
- refcounter.py highlight the invalid ao2 object refs by putting
- them in their own section of the processed output file. * Made
- refcounter.py highlight unreffing an object by more than one that
- results in a negative ref count and the object being destroyed.
- The abnormally destroyed object is reported in the invalid and
- finalized object sections of the output. Review:
- https://reviewboard.asterisk.org/r/3971/ ........ Merged
- revisions 423349 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423400 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423416 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 16:37 +0000 [r423348-423414] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/format_cap.h, main/channel.c, main/format_cap.c,
- main/translate.c: Add API call to determine if format capability
- structure is "empty". Empty here means that there are no formats
- in the format_cap structure or the only format in it is the
- "none" format. I've added calls to check the emptiness of a
- format_cap in a few places in order to short-circuit operations
- that would otherwise be pointless as well as to prevent some
- assertions from being triggered in cases where channels with no
- formats are used.
-
- * /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle
- cleanup before starting FAXes. If faxing fails at a very early
- stage, then it is possible for us to pass a NULL t30 state
- pointer to spandsp, which spandsp is none too pleased with. This
- patch ensures that we pass the correct pointer to spandsp in the
- situation where we have not yet set our local t30 state pointer.
- ASTERISK-24301 #close Reported by Matt Jordan Patches:
- ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License
- #5049) ........ Merged revisions 423360 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423365 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_mwi.c,
- res/res_pjsip_dialog_info_body_generator.c,
- res/res_pjsip_xpidf_body_generator.c,
- res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c,
- res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_pidf_body_generator.c: res_pjsip_pubsub: Add some
- type safety when generating NOTIFY bodies. res_pjsip_pubsub has
- two separate checks that it makes when a SUBSCRIBE arrives. * It
- checks that there is a subscription handler for the Event * It
- checks that there are body generators for the types in the Accept
- header The problem is, there's nothing that ensures that these
- two things will actually mesh with each other. For instance,
- Asterisk will accept a subscription to MWI that accepts pidf+xml
- bodies. That doesn't make sense. With this commit, we add some
- type information to the mix. Subscription handlers state they
- generate data of type X, and body generators state that they
- consume data of type X. This way, Asterisk doesn't end up in some
- hilariously mismatched situation like the one in the previous
- paragraph. ASTERISK-24136 #close Reported by Mark Michelson
- Review: https://reviewboard.asterisk.org/r/3877 Review:
- https://reviewboard.asterisk.org/r/3878 ........ Merged revisions
- 423344 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 15:13 +0000 [r423284] George Joseph <george.joseph@fairview5.com>
-
- * /, res/res_pjsip/location.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
- include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c:
- res_pjsip: ami: Fix error in AMI output when an endpoint has no
- transport When no transport is associated to an endpoint, the AMI
- output for PJSIPShowEndpoint indicates an error instead of
- silently ignoring the missing transport. This patch causes the
- error to appear only if a transport was specified on the endpoint
- and the transport doesn't exist. It also fixes an issue with
- counting the objects that were actually found. ASTERISK-24161
- #close ASTERISK-24331 #close Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3998/ ........ Merged
- revisions 423282 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-18 15:00 +0000 [r423281] David M. Lee <dlee@digium.com>
-
- * makeopts.in, Makefile: Only install dahdi_span_config_hook if
- DAHDI is enabled This patch changes the install to only install
- the hook script if DAHDI is enabled. It also adds the script to
- the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so
- that it's not between the _MAKEOPTS variables and their comment.
- This allows installs which specify a --prefix to work normally,
- as long as they don't enable DAHDI. Review:
- https://reviewboard.asterisk.org/r/3972/
-
-2014-09-18 14:45 +0000 [r423279] George Joseph <george.joseph@fairview5.com>
-
- * main/manager.c, /, include/asterisk/config.h, main/config.c:
- config: bug: Fix SEGV in ast_category_insert when matching
- category isn't found If you call ast_category_insert with a match
- category that doesn't exist, the list traverse runs out of 'next'
- categories and you get a SEGV. This patch adds check for the
- end-of-list condition and changes the signature to return an int
- for success/failure indication instead of a void. The only
- consumer of this function is manager and it was also changed to
- use the return value. Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3993/ ........ Merged
- revisions 423276 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423277 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423278 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-17 18:05 +0000 [r423209-423255] Joshua Colp <jcolp@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Ensure that the
- thread terminating pj stuff is registered. ........ Merged
- revisions 423253 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423254 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage
- due to timer heap thread spinning. Side note: I need a vacation.
- ........ Merged revisions 423210 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423211 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix building when
- pjproject is not used. ........ Merged revisions 423207 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423208 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-16 16:32 +0000 [r423192] Scott Griepentrog <sgriepentrog@digium.com>
-
- * apps/app_voicemail.c, include/asterisk/file.h, main/file.c:
- Voicemail: get correct duration when copying file to vm Changes
- made during format improvements resulted in the recording to
- voicemail option 'm' of the MixMonitor app writing a zero length
- duration in the msgXXXX.txt file. This change introduces a new
- function ast_ratestream(), which provides the sample rate of the
- format associated with the stream, and updates the app_voicemail
- function for ast_app_copy_recording_to_vm to calculate the right
- duration. Review: https://reviewboard.asterisk.org/r/3996/
- ASTERISK-24328 #close
-
-2014-09-16 12:12 +0000 [r423152-423173] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong
- memory pool when creating local SDP. ........ Merged revisions
- 423172 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /:
- res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The
- number of file descriptors an ioqueue instance can handle is
- fixed, so we now spawn the required number to handle the load. 2.
- Our transport identifiers were exceeding the range supported by
- pjnath. 3. The TURN client did not set up client binding causing
- needless bandwidth usage. 4. The code no longer updates address
- information on each packet. 5. STUN traffic was getting looped
- back to Asterisk instead of going through the TURN server. 6.
- Synchronization now ensures things are completely setup or
- destroyed. 7. Logging now reflects the target the TURN server is
- sending to/receiving from on our behalf. ASTERISK-23577 #close
- Reported by: Jay Jideliov ASTERISK-23634 #close Reported by:
- Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/
- ........ Merged revisions 423150 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423151 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-15 10:49 +0000 [r423069-423129] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /,
- contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py
- (added): contrib: Fix verifyi typo in alembic DB script
- ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff
- uploaded by Zogot, cleaned up by me. ........ Merged revisions
- 423128 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * configs/samples/sip.conf.sample, /: chan_sip: Clarify that
- sipdebug=yes cannot be undone by the CLI. Document it in
- sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod
- Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged
- revisions 423066 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 423067 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 423068 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-12 16:09 +0000 [r422985] Jonathan Rose <jrose@digium.com>
-
- * main/config.c, /: Realtime: Fix a bug that caused realtime
- destroy command to crash Also has could affect with anything that
- goes through ast_destroy_realtime. If a CLI user used the command
- 'realtime destroy <family>' with only a single column/value pair,
- Asterisk would crash when trying to create a variable list from a
- NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson
- Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged
- revisions 422984 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-11 22:16 +0000 [r422965] Mark Michelson <mmichelson@digium.com>
-
- * /, main/app.c: Remove undocumented default behavior of
- ast_play_and_record_full acceptdtmf. ast_play_and_record_full()
- has a parameter called "acceptdtmf" that is a string of
- acceptable DTMF digits that may be pressed by a caller to end and
- accept the recording. ARI uses this function in order to perform
- recording, and it provides options for what is passed as
- acceptdtmf to ast_play_and_record_full(). By default, ARI passes
- an empty string, with the intention that no DTMF can be used to
- end the recording. The problem is that ast_play_and_record_full()
- attempts to be "helpful" by setting "#" as the acceptdtmf if an
- empty string or NULL pointer has been passed in. With ARI, this
- results in unexpected behavior occurring if you have attempted to
- intercept "#" yourself in order to perform some other
- manipulation of the live recording. This change removes the
- "helpful" behavior by no longer accepting "#" as a default
- acceptdtmf if none is specified by the caller of
- ast_play_and_record_full(). This makes the ARI scenario work as
- expected. The other callers of ast_play_and_record_full() are
- app_voicemail and app_minivm, and in both cases, they pass an
- explicit "#" to ast_play_and_record_full() as acceptdtmf, so they
- are unaffected by this change. ........ Merged revisions 422964
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-10 16:04 +0000 [r422905] George Joseph <george.joseph@fairview5.com>
-
- * /, main/config.c: config: bug: fix truncation of included config
- files on permissions error ast_config_text_file_save() currently
- truncates include files as they are processed. If a subsequent
- include file or the main config file has a permissions error that
- prevents writing, earlier include files are left truncated
- resulting in a frantic search for backups. This patch causes
- ast_config_text_file_save to check for write access on all files
- before it truncates any of them. Will be applied 1.8 > trunk.
- Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3986/ ........ Merged
- revisions 422900 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422903 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422904 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-10 15:59 +0000 [r422901] Sean Bright <sean@malleable.com>
-
- * res/res_pjsip/config_auth.c, /: pjsip/config_auth.c: Add missing
- whitespace to log messages. The errors generated when validating
- 'auth' settings are missing a space which makes the messages a
- little confusing. ........ Merged revisions 422899 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-09 20:01 +0000 [r422883] Rusty Newton <rnewton@digium.com>
-
- * /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem:
- Modifications to include new releases and Japanese language.
- Modifying Makefile and sounds.xml to include new core 1.4.26 and
- extra 1.4.15 sound prompt releases, plus the new Japanese core
- sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
- Kevin McCoy Tested by: Rusty Newton ........ Merged revisions
- 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 422790 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422791 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-08 18:03 +0000 [r422851-422855] Mark Michelson <mmichelson@digium.com>
-
- * configs/samples/pjsip.conf.sample: Add note about configuring
- list_items on a single line.
-
- * configs/samples/pjsip.conf.sample: Add sample configuration for
- resource lists. On review /r/3977, it was recommended to note in
- the sample configuration about the size limitation for resource
- lists. However, since there was no section in the sample
- configuration at all for resource list subscriptions, I decided
- to make a separate commit where I have added the necessary sample
- configuration as well as the size limitation warning.
-
- * res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for
- RLS NOTIFY requests. PJSIP, unless a constant is modified at
- compilation time, limits SIP requests to 4000 bytes. Full-state
- RLS notifications can easily exceed this limit with moderately
- small lists. This changeset allows for Asterisk to work around
- this size limit by performing its own allocation of the
- transmission data buffer. This way, Asterisk can allocate a
- buffer that exceeds the built-in maximum. We still impose our own
- limit of 64000 bytes, mainly because making allocations larger
- than that is a bit absurd. ASTERISK-24181 #close Reported by Mark
- Michelson Review: https://reviewboard.asterisk.org/r/3977
-
-2014-09-08 15:41 +0000 [r422836] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers
- for eventlist when subscribing to resource list
- https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
- According to the off-nominal plan, if evenlist support is not
- specified in a SUBSCRIBE's supported header(s), that subscription
- should be rejected with an error. ASTERISK-23871 Reported by:
- Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3960/diff/#index_header
-
-2014-09-06 22:49 +0000 [r422767-422770] Matthew Jordan <mjordan@digium.com>
-
- * /, main/cdr.c: main/cdr: Copy over location information during a
- fork When a CDR is forked, a new CDR is created and appended to
- the CDR chain for the Party A. The forked CDR starts life off as
- a clone of the last non-finalized for the particular Party A. In
- the past, merely copying over the snapshots for Party A/Party B
- would be sufficient. However, as the CDRs now contain cached
- information from Party A - specifically application/data,
- context, and extension - we need to copy that over during a fork
- as well. Huzzah for unit tests catching this when the
- context/extension were derived from a cached value on the CDR
- instead of on Party A. ........ Merged revisions 422769 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/rtp_engine.c, /: main/rtp_engine: Format NTP timestamps as
- unsigned ints On some systems, a timeval's tv_sec/tv_usec will be
- unsigned lont ints, as opposed to long ints. When the RTP engine
- formats these as strings, it was previously formatting them as
- signed integers, which can result in some odd negative timestamp
- values (particularly on 32-bit systems). This patch formats the
- values as unsigned long integers. ........ Merged revisions
- 422766 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-06 19:12 +0000 [r422747] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix retrieval of
- "ice-pwd" attribute if in session and not media stream. ........
- Merged revisions 422746 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-05 22:03 +0000 [r422716-422719] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /, apps/app_macro.c, include/asterisk/channel.h,
- apps/app_stack.c: main/cdrs: Preserve context/extension when
- executing a Macro or GoSub The context/extension in a CDR is
- generally considered the destination of a call. When looking at a
- 2-party call CDR, users will typically be presented with the
- following: context exten channel dest_channel app data default
- 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial
- actually takes place in a Macro, the current behaviour in 12 will
- result in the following CDR: context exten channel dest_channel
- app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The
- same is true of a GoSub: context exten channel dest_channel app
- data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This
- generally makes the context/exten fields less than useful. It
- isn't hard to preserve these values in the CDR state machine;
- however, we need to have something that informs us when a channel
- is executing a subroutine. Prior to this patch, there isn't
- anything that does this. This patch solves this problem by adding
- a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on
- a channel when it executes a Macro or a GoSub. The CDR engine
- looks for this value when updating a Party A snapshot; if the
- flag is present, we don't override the context/exten on the main
- CDR object. In a funny quirk, executing a hangup handler must
- *not* abide by this logic, as the endbeforehexten logic assumes
- that the user wants to see data that occurs in hangup logic,
- which includes those subroutines. Since those execute outside of
- a typical Dial operation (and will typically have their own
- dedicated CDR anyway), this is unlikely to cause any heartburn.
- Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254
- #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis
- ........ Merged revisions 422718 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: main/cdr: Fix crash/memory consumption in CDRs in
- multi-party bridge scenarios This patch fixes an issue where CDRs
- would get stuck generating an infinite number of CDRs, eventually
- crashing Asterisk (and consuming a lot of memory along the way).
- When a channel enters into a multi-party bridge, the CDR engine
- creates mappings of each participant to each other participant,
- picking the 'A' party as it goes. So, if we have four channels in
- a multi-party bridge (Alice, Bob, Charlie, Denise), we would have
- something like: Alice => Bob Alice => Charlie Alice => Denise Bob
- => Charlie Bob => Denise Charlie => Denise This works fine when
- participants enter the bridge a single time. When a participant
- leaves a bridge, the CDRs for that channel are transitioned to a
- finalized state. The bug occurs if Bob rejoins. When the CDR
- engine creates mappings between the channels, it walks through
- all the participants currently in the bridge, and realizes that
- no one in the bridge can create a CDR with the channel (Bob). As
- such it creates a new CDR for the candidate and appends it to
- that candidate's chain. Unfortunately, on this particular code
- path, it doesn't stop traversing the candidate's chain. Since we
- just added ourselves to the chain, this causes the loop to keep
- going, constantly adding new CDRs. This patch makes it so the
- engine bails when it creates a CDR match in this case. Review:
- https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close
- Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat
- ASTERISK-24208 Reported by: Frankie Chin ........ Merged
- revisions 422715 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-05 20:35 +0000 [r422700] Richard Mudgett <rmudgett@digium.com>
-
- * funcs/func_channel.c: func_channel.c: Add missing locking to some
- CHANNEL() requests. * The CHANNEL() audionativeformat,
- videonativeformat, audioreadformat, and audiowriteformat now need
- locking since the media format rework when accessing the
- channel's format pointers. * Increased the buffer size for
- CHANNEL() audionativeformat and videonativeformat output strings
- since the allow=all can be a lengthy list. * Tweaked the
- CHANNEL() XML documentation for secure_bridge_signaling,
- secure_bridge_media, and state. * Ensured the output buffer is
- initialized for secure_bridge_signaling and secure_bridge_media.
- * Made use the locked_copy_string() macro instead of inlining it
- for trace and checkhangup.
-
-2014-09-05 20:11 +0000 [r422665-422684] Jonathan Rose <jrose@digium.com>
-
- * main/dial.c, include/asterisk/dial.h: Dial API: Add a dial option
- to indicate the dialed channel will replace dialer Adds an option
- to the dial API that marks an outgoing dial as replacing the
- dialing channel for the purpose of propagating accountcode. When
- it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of
- AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on
- the involved channels with ast_channel_req_accountcodes. Review:
- https://reviewboard.asterisk.org/r/3968/
-
- * main/cli.c, /: Call IDs: Fix appearance of call ID in core show
- channels when NULL NULL call IDs were meant to appear as '(none)'
- but instead were showing the contents of an uninitialized
- character buffer. ASTERISK-24223 Review:
- https://reviewboard.asterisk.org/r/3979/ ........ Merged
- revisions 422664 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-05 17:36 +0000 [r422661] Richard Mudgett <rmudgett@digium.com>
-
- * main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor
- tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a
- sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c.
-
-2014-09-05 13:28 +0000 [r422646] Kinsey Moore <kmoore@digium.com>
-
- * menuselect/menuselect.c: Menuselect: Fix incorrect enabling on
- failed deps This corrects a situation where menuselect can
- incorrectly enable a module by default that has defaultenabled
- set to "no" and has failed/non-selected dependencies. The bug is
- due to an inverted test when checking for whether the given
- module should be set to enabled by default on load. Review:
- https://reviewboard.asterisk.org/r/3975/ Reported by: John
- Bigelow
-
-2014-09-04 21:23 +0000 [r422631] Jonathan Rose <jrose@digium.com>
-
- * main/manager.c, /: Manager: Require read permission for SYSTEM in
- order to send FullyBooted Review:
- https://reviewboard.asterisk.org/r/3969/ ........ Merged
- revisions 422584 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422625 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422626 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-03 14:05 +0000 [r422558] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_transport_websocket.c, /:
- res_pjsip_transport_websocket: Fix crash when the Contact header
- is not a URI. The code for changing the Contact header wrongly
- assumed that the Contact would always contain a URI. This is
- incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged
- revisions 422557 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-02 20:29 +0000 [r422542] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_pjsip.c, res/res_pjsip_diversion.c,
- res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h:
- Resolve race condition where channels enter dialplan application
- before media has been negotiated. Testsuite tests will
- occasionally fail because on reception of a 200 OK SIP response,
- an AST_CONTROL_ANSWER frame is queued prior to when media has
- finished being negotiated. This is because session supplements
- are called into before PJSIP's inv_session code has told us that
- media has been updated. Sometimes the queued answer frame is
- handled by the PBX thread before the ensuing media negotiations
- occur, causing a test failure. As it turns out, there is another
- place that session supplements could be called into, which is
- after media has finished getting negotiated. What this commit
- introduces is a means for session supplements to indicate when
- they wish to be called into when handling an incoming SIP
- response. By default, all session supplements will be run at the
- same point that they were prior to this commit. However, session
- supplements may indicate that they wish to be handled earlier
- than normal on redirects, or they may indicate they wish to be
- handled after media has been negotiated. In this changeset, two
- session supplements have been updated to indicate a preference
- for when they should be run: res_pjsip_diversion executes before
- handling redirection in order to get information from the
- Diversion header, and chan_pjsip now handles responses to INVITEs
- after media negotiation to fix the race condition mentioned
- previously. ASTERISK-24212 #close Reported by Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3930 ........ Merged revisions
- 422536 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-09-01 14:16 +0000 [r422504-422507] Matthew Jordan <mjordan@digium.com>
-
- * main/cli.c, /: main/cli: Do not attempt to show CDR data for
- internal channels Internal channels don't have CDRs. Querying the
- CDR engine for their variables will make it cranky. ........
- Merged revisions 422506 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis.c, /, res/stasis/stasis_bridge.c: res_stasis:
- Don't play MoH to channels by default when added to holding
- bridges When ARI manipulates a bridge, it generally doesn't care
- what the mixing technology is. Operations on a bridge initiated
- through ARI should perform their action in generally the same
- way, regardless of the bridge's mixing technology. While the
- mixing technology may determine how media flows to channels, the
- actual operations on a bridge themselves should be the same.
- Currently, this isn't the case with holding bridges. When a
- channel joins without a role, MoH is started on that channel
- automatically. Subsequent bridge operations that would stop MoH
- would fail (as there is no Announcer channel playing MoH to the
- bridge). Starting MoH on the bridge will also create two MoH
- streams: one from the MoH being played on the participant
- channel, and one from the announcer channel. From the perspective
- of ARI users, this is counter-intuitive - I would not expect MoH
- to be started for me. The mixing technology determines how media
- is shared between participants, not the application experience.
- This patch does the following: * The Stasis bridge class now
- inspects channels as they are going into a bridge. If the bridge
- has a holding capability, and the channel has no roles, we give
- it a participant role and mark the default behaviour to have no
- entertainment. This allows addChannel operations to continue to
- set a participant role with an entertainment option if it felt
- like it (or could do it). * The music on hold channel is now
- Stasis approved (tm) Review:
- https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close
- Reported by: Samuel Galarneau Tested by: Samuel Galarneau
- ........ Merged revisions 422503 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-30 17:32 +0000 [r422442-422445] George Joseph <george.joseph@fairview5.com>
-
- * apps/app_confbridge.c, /: confbridge: Add Duration to
- ConfbridgeList event The ConfbridgeList event doesn't include how
- long the user has been a member of the conference. This patch
- adds Duration (seconds) which is based on user->chan->answertime.
- Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3955/ ........ Merged
- revisions 422444 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: manager: Make WaitEvent action respect
- eventfilters A WaitEvent issued via an http session isn't
- respecting eventfilters defined for the user. I just added a
- match_filter to the predicate that controls astman_append. Tested
- by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3958/ ........ Merged
- revisions 422439 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422440 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422441 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-29 19:40 +0000 [r422374-422379] Matthew Jordan <mjordan@digium.com>
-
- * doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility
- This patch adds a manpage for the smsq utility. Note that this is
- one of the patches the Debian distro applies for the Asterisk
- project, as per ASTERISK-24191. Review:
- https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close
- Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy
- Laine (License 6561) ........ Merged revisions 422376 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422377 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422378 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * doc/aelparse.8 (added), /: doc: Add a manpage for the aelparse
- utility This patch adds a manpage for the aelparse utility. Note
- that this is one of the patches the Debian distro applies for the
- Asterisk project, as per ASTERISK-24191. Review:
- https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close
- Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy
- Laine (License 6561) ........ Merged revisions 422371 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422372 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422373 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-29 19:05 +0000 [r422359] Scott Griepentrog <sgriepentrog@digium.com>
-
- * channels/chan_sip.c: The assertion that peer was not found on
- final event message was being triggered on configuration reload.
- This patch changes that case to just return instead. Review:
- https://reviewboard.asterisk.org/r/3953/ Commited in trunk
- revision 422358
-
-2014-08-28 21:54 +0000 [r422296] Matthew Jordan <mjordan@digium.com>
-
- * LICENSE, /: LICENSE: Clarify language in Asterisk's LICENSE to
- allow for linking to UniMRCP The UniMRCP project distributes
- Asterisk modules that integrate Asterisk with UniMRCP, and other
- Asterisk users use the UniMRCP library as well. Unfortunately,
- the UniMRCP license is Apache 2.0, which per the Free Software
- Foundation, is not a compatible license with the GPLv2. "Please
- note that this license is not compatible with GPL version 2,
- because it has some requirements that are not in that GPL
- version. These include certain patent termination and
- indemnification provisions. The patent termination provision is a
- good thing, which is why we recommend the Apache 2.0 license for
- substantial programs over other lax permissive licenses." On the
- other hand, UniMRCP is a great project and we'd like to let
- people use it with Asterisk. This patch updates the LICENSE text
- to allow users to link Asterisk with UniMRCP and distribute the
- resulting binaries. ........ Merged revisions 422293 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422294 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422295 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-28 20:30 +0000 [r422276] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, channels/chan_iax2.c: chan_iax2: Fix Dynamic IAX2
- Registrations After Temporary DNS Failure The reporter on the
- issue found some issues when upgrading from version 10 to 11 on
- 55 hosts. Two situations that can occur with dynamic
- registrations. 1. With dnsmgr disabled, if the host is not
- resolvable we are not trying to resolve the host again when it is
- time to attempt to register again. This results in never
- registering to the host. 2. With dnsmgr enabled, when the host is
- temporarily not resolvable the address is set to 0.0.0.0:0 and
- then when the host is resolvable the port is not being restored
- and stays set to 0. This patch resolves these two issues by: *
- Storing the hostname so that it can be used for resolving with
- DNS. * Resolve the hostname on the next scheduled attempt to
- register. * Storing the port used to reach the host so that when
- the hostname is resolvable again, we can set the port again if
- the port is still unset after looking up the host. ASTERISK-23767
- #close Reported by: David Herselman Tested by: David Herselman,
- Michael L. Young Patches:
- asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by
- Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/3856/ ........ Merged
- revisions 422274 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422275 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-28 17:25 +0000 [r422256] Richard Mudgett <rmudgett@digium.com>
-
- * /, UPGRADE.txt: Added ConfBridge AMI event note to UPGRADE.txt.
- ........ Merged revisions 422255 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-28 15:49 +0000 [r422239] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c: Fix bug that did not allow for multiple
- batched RLS notifications to be sent. A misunderstanding of how
- the scheduler worked caused further batched notifications beyond
- the first not to get scheduled. Now we reset our scheduler ID to
- -1 after the batched notification is sent. This way, further
- notifications can be scheduled when they arise.
-
-2014-08-28 00:36 +0000 [r422200-422215] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip/pjsip_options.c, /: res/res_pjsip/pjsip_options.c:
- Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in
- find_or_create_contact_status(). * Add missing NULL check of
- status in update_contact_status() and init_start_time(). ........
- Merged revisions 422214 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/sched.c, include/asterisk/sched.h: sched: Fix typo and
- whitespace change.
-
-2014-08-27 17:29 +0000 [r422177] George Joseph <george.joseph@fairview5.com>
-
- * /, apps/confbridge/confbridge_manager.c, apps/app_confbridge.c:
- confbridge: Add 'Admin' param to join, leave, mute, unmute and
- talking events Currently there's no way to tell if a user is an
- admin or not when receiving the join, leave, mute, unmute and
- talking events. This patch adds that capability. Tested by:
- George Joseph Review: https://reviewboard.asterisk.org/r/3950/
- ........ Merged revisions 422176 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-27 15:31 +0000 [r422154] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/utils.h, /, channels/chan_sip.c,
- tests/test_callerid.c (added), tests/test_utils.c,
- main/callerid.c, main/utils.c, res/res_pjsip_caller_id.c:
- CallerID: Fix parsing of malformed callerid This allows the
- callerid parsing function to handle malformed input strings and
- strings containing escaped and unescaped double quotes. This also
- adds a unittest to cover many of the cases where the parsing
- algorithm previously failed. Review:
- https://reviewboard.asterisk.org/r/3923/ Review:
- https://reviewboard.asterisk.org/r/3933/ ........ Merged
- revisions 422112 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 422113 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 422114 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-26 23:28 +0000 [r422091] George Joseph <george.joseph@fairview5.com>
-
- * apps/app_confbridge.c, /: confbridge: Make kick, mute and unmute
- handle channel targets consistently. Kick, mute and unmute were a
- little inconsistent in their handling of channel targets. This
- patch cleans that up by insuring they all handle the 'all' target
- consistently and adds the 'participants' target which acts on
- non-admins. Documentation for kick was also cleaned up as it
- never supported partial channel names. Tested by: George Joseph
- Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged
- revisions 422090 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-26 22:13 +0000 [r422071] Mark Michelson <mmichelson@digium.com>
-
- * main/sched.c, /: Fix race condition in the scheduler when
- deleting a running entry. When scheduled tasks run, they are
- removed from the heap (or hashtab). When a scheduled task is
- deleted, if the task can't be found in the heap (or hashtab), an
- assertion is triggered. If DO_CRASH is enabled, this assertion
- causes a crash. The problem is, sometimes it just so happens that
- someone attempts to delete a scheduled task at the time that it
- is running, leading to a crash. This change corrects the issue by
- tracking which task is currently running. If that task is
- attempted to be deleted, then we mark the task, and then wait for
- the task to complete. This way, we can be sure to coordinate task
- deletion and memory freeing. ASTERISK-24212 Reported by Matt
- Jordan Review: https://reviewboard.asterisk.org/r/3927 ........
- Merged revisions 422070 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-25 16:44 +0000 [r421979-422037] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_musiconhold.c: res_musiconhold.c: Release any format refs
- before memset(). * Clear the channel music_state pointer before
- destroying the music_state object for safety.
-
- * res/res_musiconhold.c, /: res_musiconhold: Fix MOH restarting
- where it left off from the last hold. Restore code removed by
- https://reviewboard.asterisk.org/r/3536/ that introduced a
- regression that prevents MOH from restarting were it left off the
- last time. ASTERISK-24019 #close Reported by: Jason Richards
- Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch
- uploaded by rmudgett Review:
- https://reviewboard.asterisk.org/r/3928/ ........ Merged
- revisions 421976 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421977 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421978 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-24 19:36 +0000 [r421911-421956] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_transport_websocket.c, /:
- res_pjsip_transport_websocket: Attach the Websocket module on
- outgoing INVITEs. In order to alter the Contact header on
- in-dialog requests and responses the Websocket module must be
- attached on outgoing INVITEs. The Contact header is modified so
- that the PJSIP transport layer can find and use the existing
- Websocket connection based on the source IP address, port, and
- transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov
- ........ Merged revisions 421955 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_transport_websocket.c:
- res_pjsip_transport_websocket: Fix a progressive memory growth.
- The packet structure used to receive messages was using the
- transport pool. This meant that for each parsing the pool would
- grow accordingly. Since memory can not be reclaimed without
- resetting it this would cause the memory pool to grow and grow.
- This change uses a specific memory pool for the packet structure
- and resets it to a fresh state after the message has been
- received and handled. ........ Merged revisions 421939 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_transport_websocket.c:
- res_pjsip_transport_websocket: Ensure secure Websocket clients
- can be called. This change enforces the transport in the Contact
- header for Websocket clients. Previously a client may provide a
- transport of 'ws' when it is actually using a transport of 'wss'.
- This would cause outgoing calls to fail as the existing
- connection could not be found. ........ Merged revisions 421931
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip: Use the server reflexive ICE
- candidate RTCP port as provided. This code originally worked
- around an issue within res_rtp_asterisk itself. The wrong socket
- was being used for the STUN check for RTCP, causing the port to
- be the same as RTP. This was subsequently fixed and the RTCP port
- provided for the ICE candidate is correct and does not need to be
- incremented. ASTERISK-23997 #close Reported by: Badalian
- Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav
- (license 5249) ........ Merged revisions 421909 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421910 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-22 16:56 +0000 [r421882] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_mixmonitor.c: Fix a locking inversion in MixMonitor. We
- need to unlock the audiohook before trying to lock the channel,
- since the correct locking order is channel then audiohook.
-
-2014-08-22 16:44 +0000 [r421880] Jonathan Rose <jrose@digium.com>
-
- * res/res_stasis_answer.c, res/res_stasis.c, res/stasis/command.c,
- res/res_stasis_playback.c, /, res/stasis/control.c,
- res/stasis/stasis_bridge.c, res/stasis/command.h,
- include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c:
- ARI: Fix a crash caused by hanging during playback to a channel
- in a bridge ASTERISK-24147 #close Reported by: Edvin Vidmar
- Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged
- revisions 421879 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-22 14:08 +0000 [r421860] Matthew Jordan <mjordan@digium.com>
-
- * main/message.c, /: main/message: Add a new-line to a DEBUG
- message ........ Merged revisions 421859 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-21 22:07 +0000 [r421802] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_musiconhold.c: res_musiconhold.c: Remove obsolete
- REF_DEBUG code. Remove unneeded code that writes to the wrong
- file location in an obsolete format. ........ Merged revisions
- 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 421800 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421801 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-21 21:42 +0000 [r421790-421797] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_session.c, /: Switch from hostname to an IP address
- in the SDP origin line. Using the hostname in the SDP origin line
- may not satisfy the requirement of RFC 4566 that we use a FQDN or
- IP address. This change has us use the same information from the
- SDP connection line if possible. If not possible, we'll use the
- configured media address. And if that's not possible, we use the
- result of a PJLIB call to get the IP address of ourself.
- ASTERISK-23994 #close Reported by Private Name Review:
- https://reviewboard.asterisk.org/r/3925 ........ Merged revisions
- 421796 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/stasis/control.c: Ensure after-bridge behavior is correct
- when moving from Stasis to a non-Stasis bridge. Because of the
- departable state of channels that enter Stasis bridges, Stasis
- has to take responsibility for directing the channel to its
- intended after-bridge destination if the channel moves from a
- Stasis bridge to a non-Stasis bridge. This change ensures that
- when such a move occurs, when the channel leaves the bridging
- system, any after bridge gotos are honored. Review:
- https://reviewboard.asterisk.org/r/3920 ........ Merged revisions
- 421792 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_caller_id.c, /: Let's try checking the name and
- number, instead of the name twice. ........ Merged revisions
- 421789 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-21 21:25 +0000 [r421788] Jonathan Rose <jrose@digium.com>
-
- * /, res/res_musiconhold.c: res_musiconhold: Fix reference leaks
- caused when reloading with REF_DEBUG set Due to a faulty function
- for debugging reference decrementing, it was possible to reduce
- the refcount on the wrong object if two moh classes of the same
- name were in the moh class container. (closes issue
- ASTERISK-22252) Reported by: Walter Doekes Patches:
- 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
- 6182) ........ Merged revisions 398937 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421777 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421779 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-21 21:18 +0000 [r421783] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_caller_id.c: Improve consistency of party ID
- privacy usage. Prior to this change, the Remote-Party-ID header
- took the position of "If caller name and number are not
- explicitly allowed, then they are private" and
- P-Asserted-Identity took the position of "Caller name and number
- are only private if marked explicitly so" Now both mechanisms of
- conveying party identification use the former approach. ........
- Merged revisions 421778 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-21 17:34 +0000 [r421675-421720] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Don't use port derived from
- fromdomain if it isn't set If a user does not provide a port in
- the fromdomain setting, chan_sip will set the fromdomainport to
- STANDARD_SIP_PORT (5060). The fromdomainport value will then get
- used unilaterally in certain places. This causes issues with TLS,
- where the default port is expected to be 5061. This patch
- modifies chan_sip such that fromdomainport is only used if it is
- not the standard SIP port; otherwise, the port from the SIP pvt's
- recorded self IP address is used. Review:
- https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close
- Reported by: Elazar Broad patches: fromdomainport_fix.diff
- uploaded by Elazar Broad (License 5835) ........ Merged revisions
- 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 421718 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421719 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, UPGRADE.txt, main/app.c: ARI: Fix implicit answer when
- playback is initiated on unanswered channel When issuing a POST
- /channels/{channel_id}/play on a channel that is not yet
- answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS
- on the channel * Start up the playback of the media Instead, we
- sneak an answer on the channel right before starting playing
- media. This is due to ARI's usage of control_streamfile. This
- function implicitly answers the channel (and doesn't give ARI the
- option to stop it). The answering of the channel here is probably
- unnecessary: * app_voicemail, by far the biggest consumer of this
- function, always answers the channels anyway * control stream
- file (in res_agi) and ControlPlayback probably shouldn't be
- implicitly answering the channel. Answering should not be tied
- directly to playing back media. As it turns out, the answering of
- the channel here is pretty old: 356042 twilson if
- (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res =
- ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that
- others ran into this problem and commented about it on various
- mailing lists. Review: https://reviewboard.asterisk.org/r/3907/
- ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged
- revisions 421695 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/stasis/messaging.h, main/dns.c, /, main/format_cache.c: Clean
- up files that do not end with newlines Trivial patch to add new
- lines to several files missing them. This fixes warnings when
- compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close
- Reported by: Shaun Ruffell patches:
- 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch
- uploaded by Shaun Ruffell (License 5417) ........ Merged
- revisions 421677 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/uri.h, main/uri.c: uri: Quiet warning about type
- qualifiers ignored on function return type This patch fixes gcc
- warnings that occur due to the type qualifier 'const' being
- ignored on a return type of int. ASTERISK-24246 #close Reported
- by: Shaun Ruffell patches:
- 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch
- uploaded by Shaun Ruffell (License 5417)
-
-2014-08-20 22:49 +0000 [r421616-421645] Richard Mudgett <rmudgett@digium.com>
-
- * main/bridge.c, res/res_pjsip_sdp_rtp.c, main/file.c,
- main/bridge_channel.c, channels/chan_pjsip.c, main/channel.c:
- chan_pjsip: Update media translation paths when new SDP
- negotiated. On a SIP reinvite that changes media strams, the
- PJSIP channel driver was flooding the log with "Asked to transmit
- frame type %s, while native formats is %s" warnings. * Fixes
- PJSIP not setting up translation paths when the formats change on
- a reinvite. AFS-63 was effectively reintroduced because of the
- media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the
- unexpected frame format WARNING message to include more
- information. * Added protective locking while altering formats on
- a channel. Reworked set_format() to simplify and protect the
- formats under manipulation. * Restored some code that got lost in
- the media_formats work. (channel.c:set_format() and
- res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark
- Michelson Review: https://reviewboard.asterisk.org/r/3906/
-
- * /, main/cli.c: cli.c: Fix tab completion of "module load" when
- MALLOC_DEBUG is enabled. filename_completion_function() returns
- memory that was not allocated by the MALLOC_DEBUG allocation
- tracker so the memory must be freed by ast_std_free(). ........
- Merged revisions 421600 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421602 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421608 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-20 20:40 +0000 [r421566-421585] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c: Set the role for inbound subscriptions
- correctly. This was causing the AMI show_subscriptions test in
- the testsuite to fail since all subscriptions were being seen as
- subscribers instead of notifiers.
-
- * /, channels/chan_pjsip.c: Move evaluation of set_var options in
- pjsip to the end of channel initialization. This allows for
- set_var to override certain defaults such as caller ID and codec
- values. This also fixes a test suite regression. The "set_var"
- test suite test attempted to use set_var to override caller ID,
- but a recent change caused that to no longer work. ........
- Merged revisions 421565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-20 13:04 +0000 [r421538] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/stasis_bridges.h, tests/test_cel.c,
- res/ari/ari_model_validators.c, main/stasis_bridges.c,
- res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
- res/stasis/app.c, main/bridge.c: Stasis: Add information to blind
- transfer event When a blind transfer occurs that is forced to
- create a local channel pair to satisfy the transfer request,
- information about the local channel pair is not published. This
- adds a field to describe that channel to the blind transfer
- message struct so that this information is conveyed properly to
- consumers of the blind transfer message. This also fixes a bug in
- which Stasis() was unable to properly identify the channel that
- was replacing an existing Stasis-controlled channel due to a
- blind transfer. Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3921/ ........ Merged
- revisions 421537 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-19 20:28 +0000 [r421448-421488] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip.c: Alter documentation for callerid_privacy to
- use correct values. ........ Merged revisions 421485 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis.c, /: Fix compilation error on certain versions of
- GCC. ........ Merged revisions 421447 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-19 19:42 +0000 [r421445] Kinsey Moore <kmoore@digium.com>
-
- * main/manager.c, /: AMI Docs: Fix Status channel parameter
- optionality ........ Merged revisions 421442 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421443 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421444 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-19 16:28 +0000 [r421423] Jonathan Rose <jrose@digium.com>
-
- * res/res_stasis.c, /: ARI: Fix a bug where
- /channels/{channelID}/continue doesn't execute PBX If
- /channels/{channelID}/continue is called on a channel that was
- originated without a PBX (such as the ARI command POST channel
- with a stasis application argument), the channel will not start
- dialplan execution. This patch will now run the PBX out of the
- stasis execution if the channel doesn't currently have an active
- PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon
- Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches:
- stasis-continue.diff submitted by Krandon Bruse (license 6631)
- ........ Merged revisions 421416 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-19 16:11 +0000 [r421403] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_pjsip_caller_id.c, channels/chan_pjsip.c,
- res/res_pjsip_session.c: chan_pjsip: Fix attended transfer
- connected line name update. A calls B B answers B SIP attended
- transfers to C C answers, B and C can see each other's connected
- line information B completes the transfer A has number but no
- name connected line information about C while C has the full
- information about A I examined the incoming and outgoing party id
- information handling of chan_pjsip and found several issues: *
- Fixed ast_sip_session_create_outgoing() not setting up the
- configured endpoint id as the new channel's caller id. This is
- why party A got default connected line information. * Made
- update_initial_connected_line() use the channel's CALLERID(id)
- information. The core, app_dial, or predial routine may have
- filled in or changed the endpoint caller id information. * Fixed
- chan_pjsip_new() not setting the full party id information
- available on the caller id and ANI party id. This includes the
- configured callerid_tag string and other party id fields. * Fixed
- accessing channel party id information without the channel lock
- held. * Fixed using the effective connected line id without doing
- a deep copy outside of holding the channel lock. Shallow copy
- string pointers can become stale if the channel lock is not held.
- * Made queue_connected_line_update() also update the channel's
- CALLERID(id) information. Moving the channel to another bridge
- would need the information there for the new bridge peer. * Fixed
- off nominal memory leak in update_incoming_connected_line(). *
- Added pjsip.conf callerid_tag string to party id information from
- enabled trust_inbound endpoint in caller_id_incoming_request().
- AFS-98 #close Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3913/ ........ Merged
- revisions 421400 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-18 21:10 +0000 [r421376] Damien Wedhorn <voip@facts.com.au>
-
- * channels/chan_skinny.c: Skinny: Fixup compile warning for non
- dev-mode.
-
-2014-08-18 20:19 +0000 [r421337] George Joseph <george.joseph@fairview5.com>
-
- * funcs/func_config.c, /: func_config: Change 'Not Found' message
- from ERROR to DEBUG When you call the CONFIG dialplan function
- with the name of a variable that doesn't exist in the target
- context you get an ERROR. This does nothing but clutter up the
- logs with messages that may be perfectly acceptable. Just because
- a variable wasn't in the context doesn't mean it's an error.
- Maybei t's optional or just needs to be defaulted or ignored.
- This patch changes the log level from ERROR to DEBUG. If a
- dialplan developer wants to debug their dialplan they still canby
- setting the console debug level as needed. Tested by: George
- Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........
- Merged revisions 421327 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421328 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421329 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-18 01:13 +0000 [r421230-421312] Matthew Jordan <mjordan@digium.com>
-
- * res/ari/resource_channels.c: res/ari/resource_channels: Fix
- compilation issue Forgot a parameter. Whoops.
-
- * res/ari/resource_channels.c: res/ari/resource_channels: Don't
- return allocation failure on failed function If a function fails
- to execute, it is most likely due to one of two reasons: (1) The
- function doesn't exist or can't be read from (2) The function is
- dangerous and is restricted based on the user's permissions
- Currently we return allocation failure, which is incorrect. This
- updates the reason code to more accurately reflect why the
- request failed. ASTERISK-24215
-
- * /, apps/app_meetme.c: apps/app_meetme: Fix crash when publishing
- MeetMe messages with no channel The same function,
- meetme_stasis_generate_msg, handles creating and publishing
- Stasis message both when there are channels in the MeetMe
- conference and when there are no channels in the conference. When
- the performance improvement was made to use cached snapshots,
- this created a situation where Asterisk would crash: obtaining a
- cached snapshot is not NULL tolerant. This patch restores the
- previous implementation, which used a NULL safe set of routines
- to produce a blob containing the channel snapshot (if available)
- and information about the MeetMe conference. ASTERISK-24234
- #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell
- ........ Merged revisions 421270 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_dial.c, /: apps/app_dial: Fix Dial 'z' option The 'z'
- option is supposed to disable the dial timeout in the case of a
- call forward. Unfortunately, the wrong timeout timer was passed
- to the do_forward function, resulting in the option not working.
- ASTERISK-24225 #close Reported by: dimitripietro Tested by:
- dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by
- rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by
- rmudgett (License 5621) ........ Merged revisions 421232 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421233 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421234 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, configure, configure.ac: configure: Undefine FORTIFY_SOURCE
- prior to defining it for patched gcc Some distributions of Linux
- patch gcc to define FORTIFY_SOURCE when gcc is executed with
- optimization. This "help" unfortunately results in re-definition
- warnings when FORTIFY_SOURCE is later defined in Asterisk's build
- system. This patch undefines FORTIFY_SOURCE prior to defining it
- to prevent this warning. Review:
- https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close
- Reported by: Kilburn Tested by: Kilburn, wdoekes patches:
- 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by
- cloos (License 5956) 11.diff uploaded by cloos (License 5956)
- 12.diff uploaded by cloos (License 5956) 13.diff uploaded by
- cloos (License 5956) ........ Merged revisions 421227 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421228 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421229 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-17 16:10 +0000 [r421210] Joshua Colp <jcolp@digium.com>
-
- * res/res_http_websocket.c: res_http_websocket: Include query
- parameters in client connection requests. Review:
- https://reviewboard.asterisk.org/r/3914/
-
-2014-08-15 17:08 +0000 [r421187] Jonathan Rose <jrose@digium.com>
-
- * main/channel.c, /: Bridging: Fix a behavioral change when
- checking if a channel is leaving a bridge r420934 introduced some
- failures in the test suite. Upon investigating, it was discovered
- that differences in the way we were evaluating whether a channel
- was in the process of leaving a bridge were causing some
- reinvites not to occur (mostly reinvites back to Asterisk when
- ending a call). This patch fixes that behavioral change.
- ASTERISK-24027 #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3910/ ........ Merged
- revisions 421186 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-15 15:45 +0000 [r421042-421166] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_voicemail.c, /, main/app.c: app_voicemail/app: Remove
- test events that were duplicated by r421059 Moving the test event
- raised when a file is played back (which occurred in r421059)
- broke the ever loving snot out of the voicemail tests. This
- caused duplicate test events to get raised, as app_voicemail and
- main/app were raising events prior to call ast_streamfile. The
- voicemail tests did not enjoy getting multiple events. Since
- raising the playback event in ast_streamfile is far more useful
- to the vast majority of tests, this patch keeps the call there
- and simply removes the extraneous calls that duplicated the
- event. ........ Merged revisions 421125 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421164 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421165 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_hep_rtcp.c, /: res/res_hep_rtcp: Remove dependency on
- PJSIP The res_hep_rtcp module was incorrectly including
- <pjsip.h>. This didn't need to be included, as the module does
- not using PJPROJECT any fashion. Unfortunately, because
- res_hep_rtcp did not include pjsip in its MODULEINFO as a
- dependency, this also meant that res_hep_rtcp will fail to
- compile on a system without PJPROJECT. This patch removes the
- include. Thanks to Damien Wedhorn for pointing this out in
- #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn,
- Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions
- 421064 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/file.c, main/app.c: main/file: Move test event to emit
- PLAYBACK event more consistently This is being done in advance of
- the test for ASTERISK-23953 ........ Merged revisions 421059 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 421060 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 421061 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_cel.c, main/cel.c, /: cel: Make sure channels in extra
- fields include their unique IDs as well CEL typically tracks a
- lot of information using the unique ID of the channel. This is
- typically needed due to tying events together using the linked ID
- of the various channels involved in a "call", which is derived
- from the channel ID of the oldest channel involved in a bridge
- (or in the case of a Dial, the parent channel). Previously, we
- had updated the extra fields to include the involved channel
- names, but forgot to put in the unique ID. This patch corrects
- that error. ........ Merged revisions 421037 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-14 16:32 +0000 [r420957-421010] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/ari/resource_channels.c: ARI: Originate to app local
- channel subscription code optimization. Reduce the scope of
- local_peer and only get it if the ARI originate is subscribing to
- the channels. Review: https://reviewboard.asterisk.org/r/3905/
- ........ Merged revisions 421009 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/channel_internal_api.c, main/channel.c:
- channel_internal_api.c: Replace some code with ao2_replace(). Use
- ao2_replace() instead of ao2_cleanup(); ao2_bump(). ao2_replace()
- has the advantange of not altering the ref count if the replaced
- pointer is the same. Review:
- https://reviewboard.asterisk.org/r/3904/
-
- * /, res/res_pjsip_send_to_voicemail.c:
- res_pjsip_send_to_voicemail.c: Fix svn file properties. ........
- Merged revisions 420956 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-13 16:53 +0000 [r420950] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip.c, /: PJSIP: Prevent crash no-URI contacts This
- prevents a crash from occurring when a contact with no URI is
- used for the creation of an outbound out-of-dialog request with
- no associated endpoint. ........ Merged revisions 420949 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-13 16:07 +0000 [r420940] Jonathan Rose <jrose@digium.com>
-
- * main/bridge_after.c, main/channel_internal_api.c,
- include/asterisk/channel.h, apps/app_chanspy.c,
- apps/app_mixmonitor.c, apps/app_stack.c, main/bridge_channel.c,
- main/channel.c, main/pbx.c, /, main/framehook.c: Bridges: Fix
- feature interruption/unintended kick caused by external actions
- If a manager or CLI user attached a mixmonitor to a call running
- a dynamic bridge feature while in a bridge, the feature would be
- interrupted and the channel would be forcibly kicked out of the
- bridge (usually ending the call during a simple 1 to 1 call).
- This would also occur during any similar action that could set
- the unbridge soft hangup flag, so the fix for this was to remove
- unbridge from the soft hangup flags and make it a separate thing
- all together. ASTERISK-24027 #close Reported by: mjordan Review:
- https://reviewboard.asterisk.org/r/3900/ ........ Merged
- revisions 420934 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-13 14:24 +0000 [r420919] Kinsey Moore <kmoore@digium.com>
-
- * main/manager.c: AMI: Improve documentation for Status action
-
-2014-08-13 07:52 +0000 [r420899] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, main/logger.c: logger: Don't store verbose-magic in the log
- files. In r399267, the verbose2magic stuff was edited. This time
- it results in magic characters in the log files for multiline
- messages. In trunk (and 13) this was fixed by the "stripping" of
- those characters from multiline messages (in r414798). This fix
- is altered to actually strip the characters and not replace them
- with blanks. Review: https://reviewboard.asterisk.org/r/3901/
- Review: https://reviewboard.asterisk.org/r/3902/ ........ Merged
- revisions 420897 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 420898 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-12 23:43 +0000 [r420879-420881] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_sip.c: chan_sip: Fix type mismatch when the format
- is changed. Symptom is most likely an invalid ao2 object bad
- magic number message or a less likely crash.
-
- * res/res_stasis_snoop.c: res_stasis_snoop.c: Fix off nominial exit
- path leaving Snoop channel locked and not hungup. * Made use
- ast_copy_string() instead of strcpy() for snoop uniqueid for
- safety. There is no guarantee that the max channel uniqueid
- length will remain the same as the snoop uniqueid space.
-
-2014-08-12 11:17 +0000 [r420856] Joshua Colp <jcolp@digium.com>
-
- * apps/app_voicemail.c: app_voicemail: Fix the
- "test_voicemail_vm_info" unit test.
-
-2014-08-11 20:53 +0000 [r420837] Richard Mudgett <rmudgett@digium.com>
-
- * res/stasis/command.c, /: res/stasis/command.c: Fix recent commit
- using spaces instead of tabs. ........ Merged revisions 420836
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-11 18:50 +0000 [r420808] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/playbacks.json,
- rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/resources.json, include/asterisk/manager.h,
- rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json,
- rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json: AMI/ARI: Update version to
- 2.5.0/1.5.0 respectively This is to support the backwards
- compatible changes made in the next version of Asterisk. ........
- Merged revisions 420805 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-11 18:46 +0000 [r420796-420803] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_stasis.c: Stasis: Use the correct return value Return
- the correct value instead of always returning 0 when setting
- internal status on unreal channels. Reported by: Richard Mudgett
- ........ Merged revisions 420802 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis.c, res/ari/resource_bridges.c, /,
- res/stasis/stasis_bridge.c, include/asterisk/stasis_app.h:
- Stasis: Allow internal channels directly into bridges The patch
- to catch channels being shoehorned into Stasis() via external
- mechanisms also happens to catch Announcer and Recorder channels
- because they aren't known to be stasis-controlled channels in the
- usual sense. This marks those channels as Stasis()-internal
- channels and allows them directly into bridges. Review:
- https://reviewboard.asterisk.org/r/3903/ ........ Merged
- revisions 420795 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-11 18:32 +0000 [r420758-420794] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/stasis_app.h, main/stasis_channels.c,
- res/ari/resource_channels.c, CHANGES, res/res_pjsip_pubsub.c,
- main/manager_channels.c, apps/app_dial.c, res/stasis/app.c,
- res/stasis/control.c: Improve call forwarding reporting,
- especially with regards to ARI. This patch addresses a few
- issues: 1) The order of Dial events have been changed when
- performing a call forward. The order has now been altered to 1)
- Dial begins dialing channel A. 2) When A forwards the call to B,
- we issue the dial end event to channel A, indicating the dial is
- being canceled due to a forward to B. 3) When the call to channel
- B occurs, we then issue a new dial begin to channel B. 2) Call
- forwards are now reported on the calling channel, not the peer
- channel. 3) AMI DialEnd events have been altered to display the
- extension the call is being forwarded to when relevant. 4) You
- can now get the values of channel variables for channels that are
- not currently in the Stasis application. This brings the
- retrieval of channel variables more in line with the rest of
- channel read operations since they may be performed on channels
- not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan
- ASTERISK-24138 #close Reported by Matt Jordan Patches:
- forward-shenanigans.diff uploaded by Matt Jordan (License #6283)
- Review: https://reviewboard.asterisk.org/r/3899
-
- * res/res_pjsip_pubsub.c: Fix crashing unit tests with regards to
- RLS. The unit tests require a sorcery.conf file that has been set
- up to store resource lists in memory rather than retrieving from
- configuration. With a setup that is not conducive to running the
- tests, a fault in sorcery currently causes Asterisk to crash when
- attempting to run any of the tests. To get around the crash, this
- adds a function that verifies the current environment and marks
- the tests as "not run" if the setup is not correct.
-
- * res/res_pjsip_pubsub.c: Fix crash encountered by the testsuite.
- Running testsuite tests locally produced no errors, but when run
- using the continuous integration framework, crashes occurred. The
- crashes occurred due to a refcounting error that had been fixed
- for a similar situation.
-
-2014-08-11 13:57 +0000 [r420742] Matthew Jordan <mjordan@digium.com>
-
- * res/res_hep.c, res/res_hep_pjsip.c, res/res_hep_rtcp.c: res_hep:
- Remove disabling of modules These modules were originally
- specified as being disabled, as they were introduced midstream in
- Asterisk 12. That makes it nicer for folks who are upgrading to a
- new release in the middle of Asterisk 12. That's not the case for
- Asterisk 13: it's a brand new release. There's no reason to have
- the modules disabled by default in that case.
-
-2014-08-11 10:40 +0000 [r420657-420717] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, main/utils.c: general: Fix memory Corruption in
- __ast_string_field_ptr_build_va. If the space left in a
- stringfield is between 0 and
- (alignof(ast_string_field_allocation)-1) adding new data would
- cause memory corruption, because we would assume enough space
- (unsigned underrun). Thanks Arnd Schmitter for reporting and
- finding out the cause! ASTERISK-23508 #close Reported by: Arnd
- Schmitter Tested by: Arnd Schmitter, JoshE Review:
- https://reviewboard.asterisk.org/r/3898/ ........ Merged
- revisions 420680 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 420715 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 420716 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode.
- ........ Merged revisions 420654 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 420655 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 420656 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-11 01:31 +0000 [r420607-420639] Matthew Jordan <mjordan@digium.com>
-
- * funcs/func_jitterbuffer.c: funcs/func_jitterbuffer: Tweak
- documentation This patch merely reformats and cleans up a bit of
- the jitterbuffer documentation for the wiki.
-
- * UPGRADE.txt, configs/samples/extconfig.conf.sample, CHANGES,
- apps/app_queue.c,
- contrib/ast-db-manage/config/versions/d39508cb8d8_create_queue_rules.py
- (added), configs/samples/queuerules.conf.sample: app_queue: Add
- RealTime support for queue rules This patch gives the optional
- ability to keep queue rules in RealTime. It is important to note
- that with this patch: (a) Queue rules in RealTime are only
- examined on module load/reload (b) Queue rules are loaded both
- from the queuerules.conf file as well as the RealTime backend To
- inform app_queue to examine RealTime for queue rules, a new
- setting has been added to queuerules.conf's general section
- "realtime_rules". RealTime queue rules will only be used when
- this setting is set to "yes". The schema for the database table
- supports a rule_name, time, min_penalty, and max_penalty columns.
- min_penalty and max_penalty can be relative, if a '-' or '+'
- literal is provided. Otherwise, the penalties are treated as
- constants. For example: rule_name, time, min_penalty, max_penalty
- 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2',
- '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0',
- '4564', '46546' 'test_rule', '40', '15', '50' which would result
- in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY
- to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20
- seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
- QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust
- QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 -
- After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
- QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust
- QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564
- Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to
- 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the
- queue rules will be always reloaded on a module reload, even if
- the underlying file did not change. With the option disabled, the
- rules will only be reloaded if the file was modified. Review:
- https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close
- Reported by: Michael K patches: app_queue.c_realtime_trunk.patch
- uploaded by Michael K (License 6621)
-
- * CHANGES: Update CHANGES file
-
- * UPGRADE.txt: Update UPGRADE.txt file
-
-2014-08-08 20:08 +0000 [r420577-420592] Jason Parker <jparker@digium.com>
-
- * apps/app_voicemail.c: Fix build in devmode.
-
- * CHANGES, configs/samples/voicemail.conf.sample,
- apps/app_voicemail.c: app_voicemail: Add the ability to specify
- multiple email addresses. ASTERISK-24045 Reported by: Jacob
- Barber Review: https://reviewboard.asterisk.org/r/3833/
-
-2014-08-08 17:53 +0000 [r420534-420562] Matthew Jordan <mjordan@digium.com>
-
- * channels/chan_sip.c, channels/sip/security_events.c,
- channels/sip/dialplan_functions.c, channels/sip/reqresp_parser.c,
- channels/sip/route.c, channels/sip/utils.c,
- channels/sip/config_parser.c: chan_sip: Mark chan_sip and its
- files as extended support
-
- * rest-api-templates/make_ari_stubs.py: make_ari_stubs: Update wiki
- prefix to '13'
-
- * rest-api-templates/res_ari_resource.c.mustache:
- res_ari_resource.c.mustache: Update template to emit module
- support level
-
- * main/message.c, /: main/message: remove debug message ........
- Merged revisions 420533 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-08 03:03 +0000 [r420514] Kinsey Moore <kmoore@digium.com>
-
- * tests/test_cel.c, /: CEL: Update unit tests for additional
- information This updates the CEL unit tests for the new
- information contained in the attended transfer CEL extra field.
- ........ Merged revisions 420513 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-08 01:31 +0000 [r420494-420496] Matthew Jordan <mjordan@digium.com>
-
- * UPGRADE.txt: Update UPGRADE file for 13 branch
-
- * /: Remove old properties
-
- * / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\
- \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| /
- __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_|
- |_|___/\__\___|_| |_|___|_|\_\ \___\____/
-
-2014-08-07 21:58 +0000 [r420437] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
- resolve the large SDP poll issue. Replace sip_tls_read() and
- sip_tcp_read() with a single function and resolve the poll/wait
- issue with large SDP payloads. ASTERISK-18345 #close Reported by:
- Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
- patch uploaded by Elazar Broad Review:
- https://reviewboard.asterisk.org/r/3882/ ........ Merged
- revisions 420434 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 420435 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 420436 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-07 21:17 +0000 [r420389-420415] Kinsey Moore <kmoore@digium.com>
-
- * main/stasis_bridges.c, /: Stasis: Correct blind transfer message
- generation This fixes the json object creation format string and
- key name for the BridgeBlindTransfer Stasis event allowing it to
- be published properly. ........ Merged revisions 420414 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_bridges.c, /: Stasis: Ensure transfer messages follow
- validation rules This makes Stasis() event generation for
- transfer messages follow validation rules. Currently,
- ast_json_null() is being used in place of omitting a key entirely
- which falls afoul of these validation rules.
- https://reviewboard.asterisk.org/r/3892/ ........ Merged
- revisions 420408 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pubsub.c: Fix build in dev mode
-
-2014-08-07 19:44 +0000 [r420384-420388] Mark Michelson <mmichelson@digium.com>
-
- * /, main/bridge.c: Ensure bridges exist when trying to determine
- bridged parties when publishing transfer information. ........
- Merged revisions 420387 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/strings.c, include/asterisk/res_pjsip_presence_xml.h,
- res/res_pjsip_mwi.c, res/res_pjsip_dialog_info_body_generator.c,
- res/res_pjsip_xpidf_body_generator.c, include/asterisk/strings.h,
- res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c,
- include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_pidf_body_generator.c: Add support for RFC 4662
- resource list subscriptions. This commit adds the ability for a
- user to configure a resource list in pjsip.conf. Subscribing to
- this list simultaneously subscribes the subscriber to all
- resources listed. This has the potential to reduce the amount of
- SIP traffic when loads of subscribers on a system attempt to
- subscribe to each others' states.
-
-2014-08-07 18:51 +0000 [r420364] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/format_compatibility.h,
- channels/iax2/format_compatibility.c,
- channels/iax2/include/codec_pref.h, main/format_compatibility.c,
- channels/chan_iax2.c, channels/iax2/codec_pref.c,
- channels/iax2/include/format_compatibility.h: chan_iax2: Several
- media format fixes. * Fixed the iax.conf bandwidth option. This
- is the root cause of ASTERISK-24150. * Added checks in
- iax2_request() to ensure that there are actual formats requested
- for the new channel to prevent any more fracks from issues like
- ASTERISK-24150. This is a consequence of the iax.conf bandwidth
- option not working. * Fixed struct iax2_codec_pref.order member
- size mismatch issue when converting to and from the codec
- preference order list passed over the wire. In addition the
- values sent over the wire are now compatible with previous
- Asterisk versions. * Fixed several issues dealing with the struct
- iax2_codec_pref members. Off-by-one, array limit errors, and the
- order/framing members always need to be updated together. * Made
- iax2_request() setup the channel's native format preference order
- according to the user's wishes. The new media format strategy
- needs the order specified earler. * Fixed usage of
- ast_format_compatibility_bitfield2format(). The function can
- return NULL if the bitfield was not associated with a function. *
- Deleted dead code iax2_codec_pref_getsize() and
- iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and
- iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of
- inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH,
- IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants
- again as they were in Asterisk v1.8. * Renamed prefs to
- prefs_global so it won't get confused with the local pref
- versions. * Fixed too small buffer in
- handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in
- handle_cli_iax2_show_peer() to output complete lines. * Changed
- struct create_addr_info.prefs to be struct iax2_codec_pref as an
- optimization so iax2_request() and iax2_call() do less work. *
- Fixed a potential deadlock in ast_iax2_new() on an off-nominal
- path when the pbx could not get started. * Made set_config()
- setup a local prefs list along side the local capability format
- bitfield. Once the config is loaded, then the local copies are
- put into the global versions. * Fix unininialized codec_buf in
- function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott
- Griepentrog Review: https://reviewboard.asterisk.org/r/3890/
-
-2014-08-07 15:30 +0000 [r420338] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/bridge_features.h, res/res_stasis.c,
- res/stasis/command.c, rest-api/api-docs/events.json, /,
- res/stasis/app.c, res/stasis/control.c, main/bridge.c,
- main/bridge_basic.c, res/stasis/stasis_bridge.c,
- include/asterisk/stasis_bridges.h, res/stasis/command.h,
- include/asterisk/stasis_app.h, res/stasis/app.h,
- res/stasis/control.h, apps/app_queue.c,
- res/ari/ari_model_validators.c, main/cel.c,
- main/stasis_bridges.c, res/ari/ari_model_validators.h,
- main/channel.c, include/asterisk/datastore.h, tests/test_cel.c:
- Stasis: Convey transfer information to applications This fixes a
- class of issues where Stasis applications were not made aware
- that their channels were being manipulated or replaced by
- external entitiessuch as transfers, AMI commands, or dialplan
- applications such as Bridge(). Inconsistent information such as
- StasisEnd events with unknown channels as a result of masquerades
- has also been corrected. To accomplish these fixes, several new
- fields were added to blind and attended transfer messages as well
- as StasisStart and BridgeAttendedTransfer Stasis events.
- ASTERISK-23941 #close Review:
- https://reviewboard.asterisk.org/r/3865/ Review:
- https://reviewboard.asterisk.org/r/3857/ Review:
- https://reviewboard.asterisk.org/r/3852/ Review:
- https://reviewboard.asterisk.org/r/3816/ Review:
- https://reviewboard.asterisk.org/r/3731/ Review:
- https://reviewboard.asterisk.org/r/3729/ Review:
- https://reviewboard.asterisk.org/r/3728/ ........ Merged
- revisions 420325 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-07 14:37 +0000 [r420314-420315] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_pubsub.exports.in, res/res_pjsip_publish_asterisk.c
- (added), res/res_pjsip_pubsub.c: res_pjsip_publish_asterisk: Add
- support for exchanging device and mailbox state using SIP. This
- module uses the inbound and outbound PUBLISH support to exchange
- device and mailbox state between Asterisk instances. Each
- instance is configured to publish to the other and requires no
- intermediary server. The functionality provided is similar to the
- XMPP and Corosync support. Review:
- https://reviewboard.asterisk.org/r/3780/
-
- * include/asterisk/res_pjsip_outbound_publish.h (added),
- res/res_pjsip_outbound_publish.exports.in (added),
- res/res_pjsip_outbound_publish.c (added):
- res_pjsip_outbound_publish: Add module which provides outbound
- PUBLISH support. This module implements the core parts required
- for doing outbound PUBLISH. It takes care of configuration,
- lifetime management, and authentication. Additional modules
- implement the specific events that are published. Review:
- https://reviewboard.asterisk.org/r/3780/
-
-2014-08-07 14:17 +0000 [r420289-420309] Matthew Jordan <mjordan@digium.com>
-
- * main/pbx.c: pbx: Filter out pattern matching hints in responses
- sent to ExtensionStateList Hints that are a pattern match are
- technically stored in the hint container in the same fashion as
- concrete implementations of hints. The pattern matching hints,
- however, are not "real" in the sense that things can subscribe to
- them: rather, they are stored in the hints container so that when
- a subscription is made a "real" hint can be generated for the
- subscription if one does not yet exist. The extension state core
- takes care of this correctly by matching against non-pattern
- matching extensions prior to pattern matching extensions. Because
- of this, however, the ExtensionStateList AMI action was returning
- pattern matching hints when executed. These hints are meaningless
- from the perspective of AMI clients: their state will never
- change, they cannot be subscribed to, and events would never
- normally be generated from them. As such, we now filter these out
- of the response.
-
- * build_tools/post_process_documentation.py: build_tools: Skip
- managerEvent combining for AMI action responses AMI action
- responses can (and will) reference AMI events that they return.
- These event references and definitions should not be combined
- with AMI events raised elsewhere in the code, as they are
- specifically tied to the AMI action that raised them.
- ASTERISK-24156 #close Reported by: Rusty Newton
-
-2014-08-06 18:12 +0000 [r420212-420237] Richard Mudgett <rmudgett@digium.com>
-
- * contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
- /: Fix alembic script to work properly in offline mode. When run
- in offline mode, this would attempt to check the database for the
- presence of a type it was going to try to create. I now check the
- context to see if we're running in offline mode and change a
- parameter accordingly. ........ Merged revisions 407567 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py
- (added), /: Add alembic script that adds contact user_agent and
- endpoint message_context. ........ Merged revisions 411514 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py
- (added), /,
- contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
- contrib/ast-db-manage/config.ini.sample,
- contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py
- (added),
- contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py
- (added), contrib/ast-db-manage/cdr.ini.sample,
- contrib/ast-db-manage/voicemail.ini.sample: alembic: Adjust
- sippeers, queue_members, and voicemail_messages tables. *
- Increased the sippeers useragent max string size to 255. *
- Changed the queue_members uniqueid to an auto incremented integer
- instead of a string. * Increased the voicemail_messages BLOB size
- to LONGBLOB on mysql. * Fixed the add_tables_for_pjsip config
- change version downgrade actions to drop a table it created. *
- Adjusted the sample alembic.ini files cdr.ini.sample,
- config.ini.sample, and voicemail.ini.sample to give a mysql and
- postgres sqlalchemy.url lines. ASTERISK-23847 #close Reported by:
- Stephen More ASTERISK-23825 #close Reported by: Stephen More
- ASTERISK-23909 #close Reported by: Stephen More Review:
- https://reviewboard.asterisk.org/r/3870/ ........ Merged
- revisions 420211 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-06 16:12 +0000 [r420149] George Joseph <george.joseph@fairview5.com>
-
- * /, pbx/pbx_lua.c, main/pbx.c: pbx_lua: fix regression with global
- sym export and context clash by pbx_config. ASTERISK-23818 (lua
- contexts being overwritten by contexts of the same name in
- pbx_config) surfaced because pbx_lua, having the
- AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before
- pbx_config. Since I couldn't find any reason for pbx_lua to
- export it's symbols to the rest of Asterisk, I simply changed the
- flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
- realize was that the symbols need to be exported not because
- Asterisk needs them but because any external Lua modules like
- luasql.mysql need the base Lua language APIs exported
- (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's
- an issue in pbx.c where context_merge was only merging includes,
- switches and ignore patterns if the context was already existing
- AND has extensions, or if the context was brand new. If pbx_lua
- is loaded before pbx_config, the context will exist BUT pbx_lua,
- being implemented as a switch, will never place extensions in it,
- just the switch statement. The result is that when pbx_config
- loads, it never merges the switch statement created by pbx_lua
- into the final context. This patch sets pbx_lua's modflag back to
- AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge
- that catches the case where an existing context has includes,
- switchs or ingore patterns but no actual extensions.
- ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo
- Teräs Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3891/ ........ Merged
- revisions 420146 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 420147 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 420148 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-06 15:32 +0000 [r420144] Walter Doekes <walter+asterisk@wjd.nu>
-
- * funcs/func_channel.c: Add documentation to the ability to
- retrieve the source port of a SIP call. (belongs with r419970)
- ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by
- dtryba Review: https://reviewboard.asterisk.org/r/3781/
-
-2014-08-06 12:55 +0000 [r420124] Kinsey Moore <kmoore@digium.com>
-
- * configs/samples/stasis.conf.sample (added), main/named_acl.c,
- apps/app_queue.c, main/stasis_bridges.c, main/loader.c,
- main/stasis.c, apps/app_forkcdr.c, main/stasis_message.c,
- funcs/func_cdr.c, res/res_corosync.c, res/res_stun_monitor.c,
- res/res_stasis_test.c, res/res_stasis.c, apps/app_chanspy.c,
- main/stasis_cache.c, main/pickup.c, main/security_events.c,
- include/asterisk/stasis.h, main/devicestate.c, main/core_local.c,
- res/res_stasis_snoop.c, main/endpoints.c, main/presencestate.c,
- main/cdr.c, main/channel.c, main/stasis_system.c, main/manager.c,
- main/test.c, main/file.c, main/app.c, pbx/pbx_realtime.c,
- main/stasis_channels.c, tests/test_stasis.c,
- res/parking/parking_manager.c, main/stasis_endpoints.c,
- main/rtp_engine.c, main/ccss.c, main/bridge.c,
- tests/test_stasis_channels.c: Stasis: Allow message types to be
- blocked This introduces stasis.conf and a mechanism to prevent
- certain message types from being published. Internally, this
- works by preventing the chosen message types from being created
- which ensures that those message types can never be published.
- This patch also adjusts message publishers such that message
- payloads are not created if the related message type is not
- available. ASTERISK-23943 #close Review:
- https://reviewboard.asterisk.org/r/3823/
-
-2014-08-05 21:48 +0000 [r420098-420100] Matthew Jordan <mjordan@digium.com>
-
- * res/stasis/messaging.c, /: stasis: Fix compilation issue with ao2
- tagged objects ........ Merged revisions 420099 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_endpoints.c, rest-api/api-docs/events.json, /,
- channels/chan_sip.c, res/stasis/app.c, res/stasis/messaging.h
- (added), res/ari/resource_endpoints.h, res/res_pjsip_messaging.c,
- tests/test_message.c (added), res/res_xmpp.c,
- include/asterisk/json.h, CHANGES, include/asterisk/manager.h,
- res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
- main/json.c, res/res_ari_endpoints.c, include/asterisk/message.h,
- res/ari/resource_channels.c, main/message.c, res/res_stasis.c,
- res/stasis/messaging.c (added), rest-api/api-docs/endpoints.json:
- Multiple revisions 420089-420090,420097 ........ r420089 |
- mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
- ARI: Add channel technology agnostic out of call text messaging
- This patch adds the ability to send and receive text messages
- from various technology stacks in Asterisk through ARI. This
- includes chan_sip (sip), res_pjsip_messaging (pjsip), and
- res_xmpp (xmpp). Messages are sent using the endpoints resource,
- and can be sent directly through that resource, or to a
- particular endpoint. For example, the following would send the
- message "Hello there" to PJSIP endpoint alice with a display URI
- of sip:asterisk@mycooldomain.org:
- ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
- This is equivalent to the following as well:
- ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
- Both forms are available for message technologies that allow for
- arbitrary destinations, such as chan_sip. Inbound messages can
- now be received over ARI as well. An ARI application that
- subscribes to endpoints will receive messages from those
- endpoints: { "type": "TextMessageReceived", "timestamp":
- "2014-07-12T22:53:13.494-0500", "endpoint": { "technology":
- "PJSIP", "resource": "alice", "state": "online", "channel_ids":
- [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>",
- "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.",
- "variables": [] }, "application": "testsuite" } The above was
- made possible due to some rather major changes in the message
- core. This includes (but is not limited to): - Users of the
- message API can now register message handlers. A handler has two
- callbacks: one to determine if the handler has a destination for
- the message, and another to handle it. - All dialplan
- functionality of handling a message was moved into a message
- handler provided by the message API. - Messages can now have the
- technology/endpoint associated with them. Various other
- properties are also now more easily accessible. - A number of ao2
- containers that weren't really needed were replaced with vectors.
- Iteration over ao2_containers is expensive and pointless when the
- lifetime of things is well defined and the number of things is
- very small. res_stasis now has a new file that makes up its
- structure, messaging. The messaging functionality implements a
- message handler, and passes received messages that match an
- interested endpoint over to the app for processing. Note that
- inadvertently while testing this, I reproduced ASTERISK-23969.
- res_pjsip_messaging was incorrectly parsing out the 'to' field,
- such that arbitrary SIP URIs mangled the endpoint lookup. This
- patch includes the fix for that as well. Review:
- https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close
- Reported by: Matt Jordan ASTERISK-23969 #close Reported by:
- Andrew Nagy ........ r420090 | mjordan | 2014-08-05 15:16:37
- -0500 (Tue, 05 Aug 2014) | 2 lines Remove automerge properties
- :-( ........ r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue,
- 05 Aug 2014) | 2 lines test_message: Fix strict-aliasing
- compilation issue ........ Merged revisions 420089-420090,420097
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-05 13:59 +0000 [r420028] Jonathan Rose <jrose@digium.com>
-
- * main/format.c: chan_iax2: Fix a crash that occurs when using
- allow=all for an IAX2 peer Or any combination of codecs that
- includes Opus. ASTERISK-24107 #close Review:
- https://reviewboard.asterisk.org/r/3885/
-
-2014-08-04 21:00 +0000 [r420007] Richard Mudgett <rmudgett@digium.com>
-
- * main/format_cache.c, include/asterisk/format_cache.h: Remove
- duplicate definitions of ast_format_vp8.
-
-2014-08-04 20:25 +0000 [r419970] Mark Michelson <mmichelson@digium.com>
-
- * channels/sip/dialplan_functions.c: Add the ability to retrieve
- the source port of a SIP call. This adds the ability to call
- CHANNEL(recvport) on chan_sip channels to see the port on which
- an INVITE was received. ASTERISK-24040 #close Reported by dtryba
- Patches: dialplan_functions.patch uploaded by dtryba (License
- #6628) Review: https://reviewboard.asterisk.org/r/3781
-
-2014-08-04 19:47 +0000 [r419945] Rusty Newton <rnewton@digium.com>
-
- * main/manager.c, /: Manager - Improve documentation for manager
- commands Getvar and Setvar. The documentation for these commands
- did not make it clear that they could accept expressions and
- functions. Modified to make this clear, but tried not to be
- overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton
- Tested by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/3854 ........ Merged revisions
- 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 419943 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 419944 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-08-02 03:37 +0000 [r419914] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip.c: Manager: Add PJSIPShowEndpoint[s] documentation
- This adds a large swath of response documentation for
- PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies
- heavily on the existing text in the configInfo documentation via
- xi:include tags to avoid documentation duplication. Review:
- https://reviewboard.asterisk.org/r/3888/
-
-2014-08-01 14:48 +0000 [r419888] Mark Michelson <mmichelson@digium.com>
-
- * CHANGES, res/res_pjsip/pjsip_options.c: Add ContactStatusDetail
- to PJSIPShowEndpoint AMI output. Now when running
- PJSIPShowEndpoint, you will receive a ContactStatusDetail for
- each bound contact that Asterisk is qualifying. This information
- includes the URI of the contact, current reachability, and
- roundtrip time. AFS-91 #close Reported by Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3797
-
-2014-07-31 16:19 +0000 [r419851] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, res/res_pjsip_notify.c: PJSIP: Send Notify AMI and CLI
- commands can now send to URI instead of endpoint Review:
- https://reviewboard.asterisk.org/r/3817/
-
-2014-07-31 11:57 +0000 [r419822-419825] Matthew Jordan <mjordan@digium.com>
-
- * main/rtp_engine.c, /, res/res_hep_rtcp.c (added), CHANGES,
- channels/chan_pjsip.c, res/res_rtp_asterisk.c: res_hep_rtcp: Add
- module that sends RTCP information to a Homer Server This patch
- adds a new module to Asterisk, res_hep_rtcp. The module
- subscribes to the RTCP topics in Stasis and receives RTCP
- information back from the message bus. It encodes into HEPv3
- packets and sends the information to the res_hep module for
- transmission. Using this, someone with a Homer server can get
- live call quality monitoring for all RTP-based channels in their
- Asterisk 12+ systems. In addition, there were a few bugs in the
- RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered
- by the tests written for the Asterisk Test Suite. This patch
- fixes the following: 1) chan_pjsip failed to set its channel
- unique ids on its RTP instance on outbound calls. It now does
- this in the appropriate location, in the serialized call
- callback. 2) The rtp_engine was overflowing some values when
- packed into JSON. Specifically, some longs and unsigned ints
- can't be be packed into integer values, for obvious reasons.
- Since libjansson only supports integers, floats, strings,
- booleans, and objects, we print these values into strings. 3)
- res_rtp_asterisk had a few problems: (a) it would emit a source
- IP address of 0.0.0.0 if bound to that IP address. We now use
- ast_find_ourip to get a better IP address, and properly marshal
- the result into an ast_strdupa'd string. (b) Reports can be
- generated with no report bodies. In particular, this occurs when
- a sender is transmitting information to a receiver (who will send
- no RTP back to the sender). As such, the sender has no report
- body for what it received. We now properly handle this case, and
- the sender will emit SR reports with no body. Likewise, if we
- receive an RTCP packet with no report body, we will still
- generate the appropriate events. ASTERISK-24119 #close ........
- Merged revisions 419823 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * funcs/func_jitterbuffer.c, doc/appdocsxml.dtd, main/xmldoc.c:
- xmldocs: Add support for an <example> tag in the Asterisk XML
- Documentation This patch adds support for an <example /> tag in
- the XML documentation schema. For CLI help, this doesn't change
- the formatting too much: - Preceeding white space is removed -
- Unlike with para elements, new lines are preserved However,
- having an <example /> tag in the XML schema allows for the wiki
- documentation generation script to surround the documentation
- with {code} or {noformat} tags, generating much better content
- for the wiki - and allowing us to put dialplan examples (and
- other code snippets, if desired) into the documentation for an
- application/function/AMI command/etc. Review:
- https://reviewboard.asterisk.org/r/3807/
-
-2014-07-30 18:32 +0000 [r419806] Kinsey Moore <kmoore@digium.com>
-
- * main/manager.c, res/res_manager_presencestate.c,
- res/res_manager_devicestate.c, main/pbx.c: manager: Add state
- list commands This patch adds three new AMI commands: *
- ExtensionStateList (pbx.c) - list all known extension state hints
- and their current statuses. Events emitted by the list action are
- equivalent to the ExtensionStatus events. * PresenceStateList
- (res_manager_presencestate) - list all known presence state
- values. Events emitted are generated by the stasis message type,
- and hence are PresenceStateChange events. * DeviceStateList
- (res_manager_devicestate) - list all known device state values.
- Events emitted are generated by the stasis message type, and
- hence are DeviceStateChange events. Patch-by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3799/
-
-2014-07-29 19:41 +0000 [r419789] Mark Michelson <mmichelson@digium.com>
-
- * main/manager.c: Do not omit the first header of a UserEvent AMI
- action from the corresponding emitted UserEvent. ASTERISK-24124
- #close Reported by Matt Jordan AFS-131 #close Reported by Matt
- Jordan Patches: userevent.patch uploaded by Matt Jordan (License
- #6283)
-
-2014-07-29 10:56 +0000 [r419751-419766] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Fix race condition
- where redirecting information may not be set. Since the PJSIP
- INVITE session module is invoked before any session supplements
- it was possible for it to handle a redirect before the
- res_pjsip_diversion module interpreted and set redirecting
- information on the channel. This would cause the redirecting
- information to get lost. This patch ensures that session
- supplements are *always* invoked before a redirect occurs by
- explicitly calling them in the redirect handler. Review:
- https://reviewboard.asterisk.org/r/3850/ ........ Merged
- revisions 419764 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_xpidf_body_generator.c,
- res/res_pjsip_pidf_body_generator.c:
- res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator:
- Ensure local entity is unquoted. The local entity as provided by
- PJSIP is quoted within '<' and '>'. As a result placing this
- value into XML will result in malformed XML being produced. This
- patch now unquotes the local entity so it can go safely into the
- XML. Review: https://reviewboard.asterisk.org/r/3851/ ........
- Merged revisions 419750 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-28 18:58 +0000 [r419688] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_speech_utils.c, main/channel.c, /,
- funcs/func_frame_trace.c, main/abstract_jb.c: datastores: Audit
- ast_channel_datastore_remove usage. Audit of v1.8 usage of
- ast_channel_datastore_remove() for datastore memory leaks. *
- Fixed leaks in app_speech_utils and func_frame_trace. * Fixed
- app_speech_utils not locking the channel when accessing the
- channel datastore list. Review:
- https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of
- ast_channel_datastore_remove() for datastore memory leaks. *
- Fixed leak in func_jitterbuffer. (Was not in v12) Review:
- https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of
- ast_channel_datastore_remove() for datastore memory leaks. *
- Fixed leaks in abstract_jb. * Fixed leak in
- ast_channel_unsuppress(). Used by ARI mute control and
- res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used
- by ARI mute control and res_mutestream. Review:
- https://reviewboard.asterisk.org/r/3861/ ........ Merged
- revisions 419684 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 419685 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 419686 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-25 18:09 +0000 [r419612] Joshua Colp <jcolp@digium.com>
-
- * main/loader.c: loader: Fix an infinite loop when printing modules
- using "module show". When creating the alphabetical sorted list
- each module is added to a list temporarily. On the second
- iteration each module already has a pointer to another module,
- causing stuff to go into a loop. ASTERISK-24123 #close Reported
- by: Malcolm Davenport
-
-2014-07-25 16:47 +0000 [r419592] Mark Michelson <mmichelson@digium.com>
-
- * res/res_ari_sounds.c, res/res_stasis.c, res/res_fax_spandsp.c,
- res/res_timing_kqueue.c, res/res_odbc.c,
- res/res_pjsip_transport_websocket.c, apps/app_voicemail.c,
- res/res_calendar.c, channels/chan_unistim.c, cel/cel_radius.c,
- channels/chan_multicast_rtp.c, res/res_pjsip_notify.c,
- res/res_snmp.c, formats/format_sln.c, apps/app_meetme.c,
- apps/app_dictate.c, codecs/codec_gsm.c, res/res_stasis_snoop.c,
- res/res_musiconhold.c, res/res_format_attr_h264.c,
- res/res_http_websocket.c, apps/app_followme.c,
- res/res_config_sqlite.c, formats/format_siren7.c, cdr/cdr_csv.c,
- formats/format_ilbc.c, channels/chan_phone.c,
- apps/app_setcallerid.c, apps/app_osplookup.c, cel/cel_custom.c,
- apps/app_mp3.c, res/res_agi.c, channels/chan_motif.c,
- res/res_timing_timerfd.c, apps/app_confbridge.c,
- res/res_format_attr_silk.c, formats/format_siren14.c,
- res/res_sorcery_realtime.c, channels/chan_mgcp.c,
- apps/app_jack.c, codecs/codec_lpc10.c,
- res/res_pjsip_pidf_body_generator.c, res/res_config_pgsql.c,
- funcs/func_dialplan.c, apps/app_nbscat.c, cdr/cdr_syslog.c,
- res/res_pjsip_authenticator_digest.c, apps/app_festival.c,
- res/res_fax.c, apps/app_waitforsilence.c, res/res_adsi.c,
- res/res_crypto.c, res/res_ari_applications.c,
- res/res_hep_pjsip.c, pbx/pbx_lua.c, res/res_pjsip_messaging.c,
- res/res_pjsip_caller_id.c, channels/chan_console.c,
- apps/app_getcpeid.c, res/res_stasis_answer.c,
- channels/chan_oss.c, res/res_pjsip_nat.c,
- res/res_pjsip_session.c, cdr/cdr_tds.c,
- res/res_pjsip_header_funcs.c, res/res_parking.c,
- formats/format_vox.c, res/res_pjsip_rfc3326.c,
- res/res_ari_endpoints.c, res/res_stun_monitor.c,
- res/res_pjsip_mwi.c, res/res_stasis_recording.c,
- res/res_pjsip_xpidf_body_generator.c, apps/app_sms.c,
- codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_stack.c,
- channels/chan_pjsip.c, formats/format_g729.c, cel/cel_pgsql.c,
- res/res_sorcery_config.c, res/res_ari.c, addons/chan_ooh323.c,
- cdr/cdr_sqlite3_custom.c, codecs/codec_adpcm.c,
- res/res_ari_asterisk.c, res/res_calendar_caldav.c,
- apps/app_image.c, apps/app_ices.c, formats/format_wav_gsm.c,
- main/cli.c, res/res_format_attr_celt.c, res/res_rtp_multicast.c,
- channels/chan_dahdi.c, funcs/func_pitchshift.c, res/res_smdi.c,
- res/res_pjsip_one_touch_record_info.c, pbx/pbx_ael.c,
- pbx/pbx_realtime.c, apps/app_amd.c, channels/chan_alsa.c,
- formats/format_h263.c, apps/app_url.c, res/res_pjsip_acl.c,
- apps/app_externalivr.c, res/res_curl.c, formats/format_gsm.c,
- res/res_speech.c, cdr/cdr_manager.c, res/res_calendar_exchange.c,
- codecs/codec_g722.c, res/res_pjsip_multihomed.c,
- res/res_ari_mailboxes.c, cel/cel_tds.c, res/res_sorcery_memory.c,
- apps/app_fax.c, codecs/codec_speex.c, res/res_pjsip_sdp_rtp.c,
- codecs/codec_g726.c, formats/format_ogg_vorbis.c,
- apps/app_talkdetect.c, res/res_ari_channels.c,
- res/res_pjsip_exten_state.c, apps/app_speech_utils.c,
- apps/app_agent_pool.c, apps/app_waitforring.c, res/res_statsd.c,
- addons/cdr_mysql.c, formats/format_g726.c, res/res_ari_bridges.c,
- addons/app_mysql.c, res/res_stasis_playback.c,
- addons/format_mp3.c, res/res_pjsip_endpoint_identifier_ip.c,
- res/res_phoneprov.c, res/res_pjsip_t38.c,
- res/res_pjsip_registrar_expire.c, cdr/cdr_pgsql.c,
- cdr/cdr_radius.c, res/res_chan_stats.c,
- res/res_format_attr_opus.c, res/res_config_odbc.c,
- funcs/func_audiohookinherit.c,
- res/res_pjsip_outbound_registration.c, cel/cel_manager.c,
- funcs/func_odbc.c, res/res_pjsip_endpoint_identifier_anonymous.c,
- funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c,
- apps/app_minivm.c, res/res_pjsip_log_forwarder.c,
- formats/format_h264.c, res/res_config_ldap.c, apps/app_ivrdemo.c,
- addons/chan_mobile.c, apps/app_stasis.c,
- res/res_pjsip_diversion.c, cdr/cdr_custom.c, apps/app_adsiprog.c,
- res/res_pjsip_dtmf_info.c, res/res_rtp_asterisk.c,
- res/res_calendar_icalendar.c, res/res_hep.c, channels/chan_sip.c,
- channels/chan_bridge_media.c, codecs/codec_alaw.c,
- apps/app_queue.c, res/res_srtp.c, funcs/func_presencestate.c,
- res/res_timing_pthread.c, res/res_manager_presencestate.c,
- res/res_corosync.c, apps/app_celgenuserevent.c,
- cel/cel_sqlite3_custom.c, res/snmp/agent.c, pbx/pbx_dundi.c,
- formats/format_g723.c, funcs/func_devstate.c,
- res/res_pjsip_registrar.c,
- res/res_pjsip_pidf_eyebeam_body_supplement.c,
- addons/res_config_mysql.c,
- res/res_pjsip_pidf_digium_body_supplement.c, apps/app_test.c,
- res/res_timing_dahdi.c, cdr/cdr_adaptive_odbc.c,
- apps/app_alarmreceiver.c, apps/app_chanisavail.c,
- res/res_format_attr_h263.c, res/res_pjsip_mwi_body_generator.c,
- res/res_xmpp.c, res/res_http_post.c, channels/chan_iax2.c,
- res/res_pjsip_endpoint_identifier_user.c, res/res_pjsip.c,
- res/res_pktccops.c, res/res_pjsip_send_to_voicemail.c,
- main/loader.c, cel/cel_odbc.c, res/res_ari_model.c,
- channels/chan_skinny.c,
- res/res_pjsip_outbound_authenticator_digest.c,
- res/res_mwi_external.c, apps/app_skel.c, formats/format_pcm.c,
- include/asterisk/module.h, res/res_pjsip_path.c,
- res/res_ari_playbacks.c, res/res_pjsip_pubsub.c, cdr/cdr_odbc.c,
- funcs/func_periodic_hook.c, res/res_stasis_test.c,
- formats/format_jpeg.c, res/res_pjsip_refer.c,
- formats/format_g719.c, res/res_clialiases.c,
- res/res_config_sqlite3.c, res/res_ari_device_states.c,
- formats/format_wav.c, apps/app_saycounted.c, apps/app_dahdiras.c,
- apps/app_morsecode.c, res/res_stasis_mailbox.c,
- res/res_ael_share.c, res/res_mwi_external_ami.c,
- res/res_pjsip_logger.c, res/res_stasis_device_state.c,
- res/res_calendar_ews.c, res/res_monitor.c, apps/app_playback.c,
- res/res_ari_recordings.c, res/res_manager_devicestate.c,
- res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c,
- res/res_ari_events.c, res/res_pjsip_dialog_info_body_generator.c,
- res/res_sorcery_astdb.c, codecs/codec_dahdi.c,
- apps/app_zapateller.c, pbx/pbx_config.c: Add module support level
- to ast_module_info structure. Print it in CLI "module show" .
- ASTERISK-23919 #close Reported by Malcolm Davenport Review:
- https://reviewboard.asterisk.org/r/3802
-
-2014-07-25 14:47 +0000 [r419563-419567] Matthew Jordan <mjordan@digium.com>
-
- * CHANGES, res/ari/ari_model_validators.c,
- rest-api/api-docs/recordings.json,
- res/ari/ari_model_validators.h, /, res/res_stasis_recording.c:
- Multiple revisions 419565-419566 ........ r419565 | mjordan |
- 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) | 21 lines ARI:
- report duration values in LiveRecording objects This patch adds
- three new fields to the LiveRecording model: - total_duration:
- the total length of the live recording - talking_duration:
- optional. The duration of talking energy that was detected while
- the recording was made. - silence_duration: optional. The
- duration of silence that was detected while the recording was
- made. These values are reported in the RecordingFinished ARI
- event. When a DSP is enabled on the channel during the recording
- - which occurs when the recording is created with
- max_silence_seconds (indicating that the user actually cares
- about how much silence is in the file), we will report the
- talking_duration and silence_duration in addition to the
- total_duration. Review: https://reviewboard.asterisk.org/r/3770/
- ASTERISK-24037 #close Reported by: Samuel Galarneau ........
- r419566 | mjordan | 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014)
- | 1 line Update CHANGES for r419565 ........ Merged revisions
- 419565-419566 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/loader.c, res/res_calendar.c: module loader: Unload modules
- in reverse order of their start order When Asterisk starts a
- module (calling its load_module function), it re-orders the
- module list, sorting it alphabetically. Ostensibly, this was done
- so that the output of 'module show' listed modules in alphabetic
- order. This had the unfortunate side effect of making modules
- with complex usage patterns unloadable. A module that has a large
- number of modules that depend on it is typically abandoned during
- the unloading process. This results in its memory not being
- reclaimed during exit. Generally, this isn't harmful - when the
- process is destroyed, the operating system will reclaim all
- memory allocated by the process. Prior to Asterisk 12, we also
- didn't have many modules with complex dependencies. However, with
- the advent of ARI and PJSIP, this can make make unloading those
- modules successfully nearly impossible, and thus tracking memory
- leaks or ref debug leaks a real pain. While this patch is not a
- complete overhaul of the module loader - such an effort would be
- beyond the scope of what could be done for Asterisk 13 - this
- does make some marginal improvements to the loader such that
- modules like res_pjsip or res_stasis *may* be made properly
- un-loadable in the future. 1. The linked list of modules has been
- replaced with a doubly linked list. This allows traversal of the
- module list to occur backwards. The module shutdown routine now
- walks the global list backwards when it attempts to unload
- modules. 2. The alphabetic reorganization of the module list on
- startup has been removed. Instead, a started module is placed at
- the end of the module list. 3. The ast_update_module_list
- function - which is used by the CLI to display the modules - now
- does the sorting alphabetically itself. It creates its own linked
- list and inserts the modules into it in alphabetic order. This
- allows for the intent of the previous code to be maintained. This
- patch also contains a fix for res_calendar. Without
- calendar.conf, the calendar modules were improperly bumping the
- use count of res_calendar, then failing to load themselves. This
- patch makes it so that we detect whether or not calendaring is
- enabled before altering the use count. Review:
- https://reviewboard.asterisk.org/r/3777/
-
-2014-07-25 10:54 +0000 [r419537-419539] Joshua Colp <jcolp@digium.com>
-
- * apps/app_bridgewait.c, /: app_bridgewait: Remove possibility of
- race condition between channels leaving/joining. Bridges created
- by app_bridgewait previously had the "dissolve when empty" flag
- set. This caused the bridge core to destroy them when the last
- channel had left. This introduced a race condition where we may
- have a reference to the bridge but it is not actually joinable
- when we try to join it. This flag has now been removed and the
- bridge is guaranteed to be joinable at all times. ASTERISK-23987
- #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3836/ ........ Merged
- revisions 419538 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/bridge.c: bridge: Make "bridge destroy" only available in
- developer mode and add "all" to "bridge kick". The "bridge
- destroy" CLI command is invasive to bridges and can leave them in
- an unexpected state for the users of them. Since this command may
- be useful for developers it is now only available when developer
- mode is available. To take its place "all" has been added as a
- valid option to the "bridge kick" CLI command. It will kick all
- of the channels in the bridge out. ASTERISK-23987 Reported by:
- Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/
- ........ Merged revisions 419536 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-24 22:48 +0000 [r419520] Richard Mudgett <rmudgett@digium.com>
-
- * main/bridge.c, main/bridge_basic.c, main/core_unreal.c,
- UPGRADE.txt, include/asterisk/channel.h, CHANGES,
- apps/app_followme.c, apps/app_queue.c, main/cel.c,
- res/parking/parking_bridge_features.c, apps/app_dial.c,
- main/channel.c, main/dial.c, main/pbx.c: accountcode: Slightly
- change accountcode propagation. The previous behavior was to
- simply set the accountcode of an outgoing channel to the
- accountcode of the channel initiating the call. It was done this
- way a long time ago to allow the accountcode set on the SIP/100
- channel to be propagated to a local channel so the dialplan
- execution on the Local;2 channel would have the SIP/100
- accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200
- Propagating the SIP/100 accountcode to the local channels is very
- useful. Without any dialplan manipulation, all channels in this
- call would have the same accountcode. Using dialplan, you can set
- a different accountcode on the SIP/200 channel either by setting
- the accountcode on the Local;2 channel or by the Dial
- application's b(pre-dial), M(macro) or U(gosub) options, or by
- the FollowMe application's b(pre-dial) option, or by the Queue
- application's macro or gosub options. Before Asterisk v12, the
- altered accountcode on SIP/200 will remain until the local
- channels optimize out and the accountcode would change to the
- SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount
- support but ultimately had to punt on the support. The
- peeraccount support was rendered useless because of how the CDR
- code needed to unconditionally force the caller's accountcode
- onto the peer channel's accountcode. The CEL events were thus
- intentionally made to always use the channel's accountcode as the
- peeraccount value. With the arrival of Asterisk v12, the
- situation has improved somewhat so peeraccount support can be
- made to work. Using the indicated example, the the accountcode
- values become as follows when the peeraccount is set on SIP/100
- before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 --->
- SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer:
- 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already
- has an accountcode it can only change by the following explicit
- user actions: 1) A channel originate method that can specify an
- accountcode to use. 2) The calling channel propagating its
- non-empty peeraccount or its non-empty accountcode if the
- peeraccount was empty to the outgoing channel's accountcode
- before initiating the dial. e.g., Dial and FollowMe. The
- exception to this propagation method is Queue. Queue will only
- propagate peeraccounts this way only if the outgoing channel does
- not have an accountcode. 3) Dialplan using CHANNEL(accountcode).
- 4) Dialplan using CHANNEL(peeraccount) on the other end of a
- local channel pair. If a channel does not have an accountcode it
- can get one from the following places: 1) The channel driver's
- configuration at channel creation. 2) Explicit user action as
- already indicated. 3) Entering a basic or stasis-mixing bridge
- from a peer channel's peeraccount value. You can specify the
- accountcode for an outgoing channel by setting the
- CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
- applications. Queue adds the wrinkle that it will not overwrite
- an existing accountcode on the outgoing channel with the calling
- channels values. Accountcode and peeraccount values propagate to
- an outgoing channel before dialing. Accountcodes also propagate
- when channels enter or leave a basic or stasis-mixing bridge. The
- peeraccount value only makes sense for mixing bridges with two
- channels; it is meaningless otherwise. * Made peeraccount
- functional by changing accountcode propagation as described
- above. * Fixed CEL extracting the wrong ie value for the
- peeraccount. This was done intentionally in Asterisk v1.8 when
- that version had to punt on peeraccount. * Fixed a few places
- dealing with accountcodes that were reading from channels without
- the lock held. AFS-65 #close Review:
- https://reviewboard.asterisk.org/r/3601/
-
-2014-07-24 21:01 +0000 [r419504] Michael L. Young <elgueromexicano@gmail.com>
-
- * main/db.c, include/asterisk/astdb.h: core/db: Revert Patch Added
- In Attempt To Improve I/O Performance Reverting the patch since
- it was causing a regression and after fixing the regression,
- there were no performance gains. At least based on my method for
- measurement. ASTERISK-24050 Review:
- https://reviewboard.asterisk.org/r/3841/
-
-2014-07-24 17:50 +0000 [r419438-419439] Corey Farrell <git@cfware.com>
-
- * include/asterisk/astobj.h: Deprecate astobj.h This flags astobj.h
- as deprecated, warns people to use astobj2.h instead. Only
- netsock.c (also deprecated) still uses astobj.h. ASTERISK-24069
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3818/
-
- * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip:
- complete upgrade to ao2 This change upgrades sip_registry and
- sip_subscription_mwi to astobj2. ASTERISK-24067 #close Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3759/
-
-2014-07-24 16:52 +0000 [r419377] Jason Parker <jparker@digium.com>
-
- * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if
- ooh323.conf not found. (closes issue ASTERISK-23814) ........
- Merged revisions 419374 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 419375 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 419376 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-24 15:20 +0000 [r419358] Matthew Jordan <mjordan@digium.com>
-
- * main/devicestate.c, channels/chan_pjsip.c: device state: Update
- the core to report ONHOLD if a channel is on hold In Asterisk, it
- is possible for a device to have a status of ONHOLD. This is not
- typically an easy thing to determine, as a channel being on hold
- is not a direct channel state. Typically, this has to be
- calculated outside of the core independently in channel drivers,
- notably, chan_sip and chan_pjsip. Both of these channel drivers
- already have to calculate device state in a fashion more complex
- than the core can handle, as they aggregate all state of all
- channels associated with a peer/endpoint; they also independently
- track whether or not one of those channels is currently on hold
- and mark the device state appropriately. In 12+, we now have the
- ability to report an AST_DEVICE_ONHOLD state for all channels
- that defer their device state to the core. This is due to channel
- hold state actually now being tracked on the channel itself. If a
- channel driver defers its device state to the core (which many,
- such as DAHDI, IAX2, and others do in most situations), the
- device state core already goes out to get a channel associated
- with the device. As such, it can now also factor the channel hold
- state in its calculation. This patch adds this logic to the
- device state core. It also uses an existing mapping between
- device state and channel state to handle more channel states.
- chan_pjsip has been updated slightly as well to make use of this
- (as it was, for some reason, reporting a channel state of BUSY as
- a device state of INUSE, which feels slightly wrong). Review:
- https://reviewboard.asterisk.org/r/3771/ ASTERISK-24038 #close
-
-2014-07-24 13:00 +0000 [r419342] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/manager.h, doc/appdocsxml.dtd, main/xmldoc.c,
- main/manager_bridges.c, main/manager.c,
- include/asterisk/xmldoc.h, main/config_options.c: AMI: Allow for
- command response documentation Allow for responses to AMI
- actions/commands to be documented properly in XML and displayed
- via the CLI. Response events are documented exactly as standard
- AMI events are documented. Review:
- https://reviewboard.asterisk.org/r/3812/
-
-2014-07-23 16:46 +0000 [r419319] Matthew Jordan <mjordan@digium.com>
-
- * main/endpoints.c, tests/test_stasis_endpoints.c, /: endpoints:
- Fix failing unit tests from r419196 This patch does two things:
- (1) It updates the unit tests to expect additional stasis
- messages. More messages are now sent to the endpoint topic, due
- to forwarding all channel messages and the forwarding
- relationship set up between endpoints themselves. (2) Remove the
- technology forwarding subscription during ast_endpoint_shutdown.
- This prevents an improper double shutdown of an endpoint from
- occurring. ........ Merged revisions 419318 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-23 14:00 +0000 [r419286] Scott Griepentrog <sgriepentrog@digium.com>
-
- * apps/app_voicemail.c, /: app_voicemail: use a consistent
- generator string When updating voicemail.conf when a user changes
- their pin, change the generator string to be the same as the
- module name when reading so that the same config_hook will be
- called. Review: https://reviewboard.asterisk.org/r/3837/ ........
- Merged revisions 419284 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 419285 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-23 01:28 +0000 [r419268] Corey Farrell <git@cfware.com>
-
- * main/manager.c, res/res_fax.c: res_fax: unregister manager
- actions on unload * Unregister manager actions FAXSessions,
- FAXSession and FAXStats at unload. * Update ast_manager_register2
- use ao2_t_alloc tagged with the action name. ASTERISK-24058
- #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3831/
-
-2014-07-22 20:22 +0000 [r419222-419252] Michael L. Young <elgueromexicano@gmail.com>
-
- * CHANGES, main/bridge_channel.c: core/bridge_channel: Substitute
- Variables In Features Application Map Say you wanted to include
- variables in an application map and have those variables
- substituted and passed along to the application being executed;
- currently this does not happen. This patch adds this ability to
- pass channel variable values to an application before being
- executed. ASTERISK-22608 #close Reported by: Michael L. Young
- patches: features_substitute_arguments_v2.diff uploaded by
- Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/3819/
-
- * CHANGES, apps/app_mixmonitor.c: apps/app_mixmonitor: Add Options
- To Play Beep At Start Or Stop We have a new periodic beep feature
- but sometimes a user needs some sort of feedback, without the
- need to have a periodic beep during the recording, to let them
- know that MixMonitor started recording or ended the recording.
- The use case where this patch is being used is when using Dynamic
- Features to start and end MixMonitor. This patch adds an option
- to play a beep when MixMonitor starts and an option to play a
- beep when MixMonitor ends. ASTERISK-24051 #close Reported by:
- Michael L. Young patches: mixmonitor-play-beep-start-stop.diff
- uploaded by Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/3820/
-
- * main/db.c, include/asterisk/astdb.h: core/db: Improve I/O When
- Updating Rows When updating a row, we are currently doing an
- INSERT OR REPLACE INTO. The downside to this is that the row is
- deleted if it exists and then a new row is inserted. So, we are
- hitting the disk twice. One for the deletion and one for the
- insertion. This patch changes this statement to an INSERT INTO
- and if the insert fails because a row with that key exists, we
- will IGNORE the failure. Then we will attempt to perform an
- UPDATE on the existing row if that row wasn't just INSERTed.
- ASTERISK-24050 #close Reported by: Michael L. Young patches:
- astdb-insert-update-io-help_trunk_v2.diff uploaded by Michael L.
- Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/3815/
-
-2014-07-22 17:10 +0000 [r419206] Richard Mudgett <rmudgett@digium.com>
-
- * codecs/codec_speex.c: codec_speex: Fix trashing normal static
- frame for AST_FRAME_CNG. Made use a local static frame to
- generate the AST_FRAME_CNG frame when silence starts. I don't
- think the handling of the AST_FRAME_CNG has ever really worked
- because there doesn't seem to be any consumers of it. Review:
- https://reviewboard.asterisk.org/r/3813/
-
-2014-07-22 16:20 +0000 [r419203] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/endpoints.h,
- rest-api/api-docs/applications.json, include/asterisk/xmpp.h,
- main/channel_internal_api.c, channels/chan_motif.c,
- include/asterisk/channel.h, res/ari/resource_applications.h,
- res/res_xmpp.c, channels/chan_iax2.c, main/endpoints.c,
- channels/chan_pjsip.c, main/channel.c,
- res/ari/resource_endpoints.c, /, channels/chan_sip.c: ARI: Fix
- endpoint/channel subscription issues; allow for subscriptions to
- tech This patch serves two purposes: (1) It fixes some bugs with
- endpoint subscriptions not reporting all of the channel events
- (2) It serves as the preliminary work needed for ASTERISK-23692,
- which allows for sending/receiving arbitrary out of call text
- messages through ARI in a technology agnostic fashion. The
- messaging functionality described on ASTERISK-23692 requires two
- things: (1) The ability to send/receive messages associated with
- an endpoint. This is relatively straight forwards with the
- endpoint core in Asterisk now. (2) The ability to send/receive
- messages associated with a technology and an arbitrary technology
- defined URI. This is less straight forward, as endpoints are
- formed from a tech + resource pair. We don't have a mechanism to
- note that a technology that *may* have endpoints exists. This
- patch provides such a mechanism, and fixes a few bugs along the
- way. The first major bug this patch fixes is the forwarding of
- channel messages to their respective endpoints. Prior to this
- patch, there were two problems: (1) Channel caching messages
- weren't forwarded. Thus, the endpoints missed most of the
- interesting bits (such as channel creation, destruction, state
- changes, etc.) (2) Channels weren't associated with their
- endpoint until after creation. This resulted in endpoints missing
- the channel creation message, which limited the usefulness of the
- subscription in the first place (a major use case being 'tell me
- when this endpoint has a channel'). Unfortunately, this meant
- another parameter to ast_channel_alloc. Since not all channel
- technologies support an ast_endpoint, this patch makes such a
- call optional and opts for a new function,
- ast_channel_alloc_with_endpoint. When endpoints are created, they
- will implicitly create a technology endpoint for their technology
- (if one does not already exist). A technology endpoint is special
- in that it has no state, cannot have channels created for it,
- cannot be created explicitly, and cannot be destroyed except on
- shutdown. It does, however, have all messages from other
- endpoints in its technology forwarded to it. Combined with the
- bug fixes, we now have Stasis messages being properly forwarded.
- Consider the following scenario: two PJSIP endpoints (foo and
- bar), where bar has a single channel associated with it and foo
- has two channels associated with it. The messages would be
- forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint
- PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP /
- channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the
- applications resource, can: - subscribe to endpoint:PJSIP/foo and
- get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and
- endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get
- notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar -
- subscribe to endpoint:PJSIP and get notifications for channels
- PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints
- PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes,
- it never has events itself. It merely provides an aggregation
- point for all other endpoints in its technology (which in turn
- aggregate all channel messages associated with that endpoint).
- This patch also adds endpoints to res_xmpp and chan_motif,
- because the actual messaging work will need it (messaging without
- XMPP is just sad). Review:
- https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........
- Merged revisions 419196 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-22 14:36 +0000 [r419180] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: chan_iax2: Restore previous behavior of
- iax2_best_codec. The iax2_best_codec function was changed to
- convert the formats into a format compatibilities structure and
- grab the first format from it. The resulting order differs from
- the previous order of iax2_best_codec which causes unexpected
- formats to get chosen (such as g723). This commit brings back the
- old behavior of iax2_best_codec by having a specified preference
- list. Review: https://reviewboard.asterisk.org/r/3835/
-
-2014-07-22 14:22 +0000 [r419110-419175] Kinsey Moore <kmoore@digium.com>
-
- * addons/ooh323c/src/printHandler.c, tests/test_sorcery_realtime.c,
- tests/test_json.c, addons/ooh323c/src/ooq931.c,
- tests/test_astobj2_thrash.c, addons/chan_ooh323.c, /,
- tests/test_optional_api.c, tests/test_abstract_jb.c,
- apps/app_meetme.c, tests/test_logger.c, tests/test_event.c,
- tests/test_hashtab_thrash.c, res/res_mwi_external_ami.c,
- tests/test_sorcery.c, res/res_corosync.c,
- tests/test_voicemail_api.c, tests/test_aoc.c,
- tests/test_astobj2.c, tests/test_config.c: Fix more dev-mode
- build issues ........ Merged revisions 419129 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 419162 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 419163 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/dial.c: Dial API: Prevent crash on NULL cap This prevents a
- crash in the Dial API triggered by use of the Page() application
- where a format capability struct was used before checking whether
- it was NULL. ASTERISK-24074 #close
-
- * channels/chan_skinny.c, tests/test_core_format.c: Fix build in
- dev-mode
-
-2014-07-21 16:26 +0000 [r419109] Jonathan Rose <jrose@digium.com>
-
- * channels/chan_iax2.c: chan_iax2: Restore codec choice behavior
- from media formats branch After merging the media formats branch,
- chan_iax2 was discarding codec preferences for the purpose of
- choosing which codec a channel would use once a call started.
- This patch restores the Asterisk 1.8-12 codec choice behaviors.
- ASTERISK-23958 #close Review:
- https://reviewboard.asterisk.org/r/3800/
-
-2014-07-21 16:09 +0000 [r419093] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_iax2.c: chan_iax2: Only send mini frames if the
- underlying format has not changed, not if it has. ASTERISK-24072
- #close Reported by: Matt Jordan
-
-2014-07-21 14:49 +0000 [r419077] Sean Bright <sean@malleable.com>
-
- * configure, configure.ac: Fix build when pjproject is installed in
- a non-standard location. When configuring Asterisk to build
- against a version of pjproject installed in a non-standard
- location, the checks for "PJSIP Transaction Group Lock Support"
- and "PJSIP Media Stream Replacement Support" fail. This is
- because these secondary checks are not taking the CFLAGS and LIBS
- returned by the pkg-config check into account. Review:
- https://reviewboard.asterisk.org/r/3830
-
-2014-07-21 08:41 +0000 [r419060] Corey Farrell <git@cfware.com>
-
- * channels/sig_analog.c, res/res_smdi.c, channels/chan_motif.c,
- include/asterisk/smdi.h, apps/app_voicemail.c,
- channels/chan_dahdi.c: res_smdi: convert to astobj2 Remove
- functions: ast_smdi_interface_unref ast_smdi_md_message_putback
- ast_smdi_mwi_message_putback ast_smdi_md_message destructor
- ast_smdi_mwi_message destructor Includes for astobj.h are removed
- everywhere it's possible. ASTERISK-24066 #close Review:
- https://reviewboard.asterisk.org/r/3758/
-
-2014-07-20 22:06 +0000 [r419044] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_confbridge.c, res/ari/resource_channels.c,
- include/asterisk/rtp_engine.h, include/asterisk/slinfactory.h,
- res/res_calendar.c, codecs/codec_g722.c,
- include/asterisk/res_pjsip_session.h, main/frame.c,
- codecs/ex_lpc10.h, apps/app_dictate.c, res/res_fax.c,
- apps/app_echo.c, include/asterisk/slin.h, codecs/codec_g726.c,
- formats/format_ogg_vorbis.c, codecs/codec_gsm.c,
- codecs/ex_alaw.h, formats/format_wav_gsm.c,
- channels/iax2/provision.c, channels/chan_iax2.c,
- res/res_format_attr_h264.c, main/data.c, main/manager.c,
- include/asterisk/audiohook.h, formats/format_pcm.c,
- main/config_options.c, res/res_format_attr_silk.c,
- main/bridge_channel.c, res/res_speech.c, channels/chan_pjsip.c,
- res/res_clioriginate.c, formats/format_g729.c,
- channels/chan_unistim.c, res/res_rtp_asterisk.c,
- include/asterisk/smoother.h (added), main/rtp_engine.c,
- addons/format_mp3.c, formats/format_wav.c,
- apps/confbridge/conf_chan_record.c, include/asterisk/speech.h,
- codecs/ex_adpcm.h, channels/iax2/codec_pref.c (added),
- include/asterisk/codec.h (added), formats/format_siren7.c,
- include/asterisk/file.h, channels/chan_dahdi.c,
- include/asterisk/image.h, funcs/func_channel.c,
- main/abstract_jb.c, formats/format_h263.c, codecs/codec_dahdi.c,
- main/dsp.c, apps/app_voicemail.c, apps/app_jack.c,
- funcs/func_talkdetect.c, channels/chan_vpb.cc,
- channels/chan_sip.c, formats/format_sln.c,
- tests/test_abstract_jb.c, codecs/codec_alaw.c, UPGRADE.txt,
- main/smoother.c (added), codecs/ex_speex.h,
- channels/chan_console.c, apps/app_talkdetect.c,
- main/format_pref.c (removed), main/indications.c,
- include/asterisk/format_cap.h, main/media_index.c,
- apps/app_agent_pool.c, res/res_pjsip_session.c, main/cli.c,
- res/res_format_attr_celt.c, channels/chan_skinny.c,
- tests/test_core_format.c (added), funcs/func_frame_trace.c,
- res/res_pjsip/pjsip_configuration.c, main/file.c,
- include/asterisk/frame.h, formats/format_g726.c,
- apps/app_mixmonitor.c, channels/chan_mgcp.c, main/sorcery.c,
- codecs/ex_ilbc.h, codecs/codec_lpc10.c, tests/test_format_cache.c
- (added), apps/app_meetme.c, main/translate.c,
- apps/app_originate.c, res/parking/parking_applications.c,
- apps/app_ices.c, channels/iax2/parser.c, res/res_rtp_multicast.c,
- pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_vox.c,
- main/format_cap.c, tests/test_cel.c, include/asterisk/format.h,
- formats/format_h264.c, apps/app_chanspy.c, apps/app_nbscat.c,
- addons/chan_ooh323.c, bridges/bridge_holding.c,
- channels/iax2/include/codec_pref.h (added), codecs/codec_adpcm.c,
- apps/app_waitforsilence.c, res/res_pjsip_sdp_rtp.c,
- addons/chan_ooh323.h, bridges/bridge_simple.c,
- apps/app_alarmreceiver.c, bridges/bridge_softmix.c,
- res/res_stasis_snoop.c, main/sounds_index.c, main/core_local.c,
- main/codec_builtin.c (added), include/asterisk/format_cache.h
- (added), apps/app_speech_utils.c, res/res_format_attr_opus.c,
- include/asterisk/abstract_jb.h, main/channel.c,
- include/asterisk/format_compatibility.h (added), apps/app_mp3.c,
- tests/test_voicemail_api.c, channels/chan_alsa.c, main/app.c,
- formats/format_g723.c, codecs/codec_ilbc.c, tests/test_config.c,
- formats/format_gsm.c, apps/app_milliwatt.c, codecs/ex_ulaw.h,
- main/asterisk.c, include/asterisk/res_pjsip.h, main/format.c,
- main/ccss.c, main/bridge.c, codecs/codec_speex.c,
- include/asterisk/format_pref.h (removed), apps/app_record.c,
- main/slinfactory.c, res/res_adsi.c, main/core_unreal.c,
- res/ari/resource_bridges.c, include/asterisk/callerid.h,
- channels/pjsip/dialplan_functions.c, main/dial.c,
- channels/dahdi/bridge_native_dahdi.c, main/format_cache.c
- (added), include/asterisk/mod_format.h, apps/app_sms.c,
- codecs/codec_resample.c, main/format_compatibility.c (added),
- main/audiohook.c, formats/format_jpeg.c, res/res_stasis.c,
- formats/format_g719.c, include/asterisk/translate.h,
- funcs/func_speex.c, codecs/codec_a_mu.c,
- channels/iax2/format_compatibility.c (added),
- apps/app_festival.c, main/channel_internal_api.c,
- tests/test_format_api.c (removed), codecs/ex_g722.h,
- main/utils.c, res/ari/resource_sounds.c,
- res/res_format_attr_h263.c, codecs/ex_g726.h,
- include/asterisk/_private.h, channels/chan_oss.c,
- channels/chan_misdn.c, main/codec.c (added), main/callerid.c,
- addons/ooh323cDriver.c, apps/app_amd.c, codecs/codec_ulaw.c,
- main/image.c, channels/chan_nbs.c, bridges/bridge_native_rtp.c,
- channels/iax2/include/format_compatibility.h (added),
- formats/format_siren14.c, res/res_fax_spandsp.c,
- addons/chan_mobile.c, addons/ooh323cDriver.h,
- channels/sip/include/sip.h, tests/test_format_cap.c (added),
- channels/chan_multicast_rtp.c, include/asterisk/vector.h,
- channels/chan_bridge_media.c, apps/app_fax.c,
- main/bridge_basic.c, apps/app_test.c, include/asterisk/channel.h,
- include/asterisk/data.h, tests/test_core_codec.c (added),
- res/res_musiconhold.c, codecs/ex_gsm.h, formats/format_ilbc.c,
- include/asterisk/config_options.h, channels/chan_phone.c,
- include/asterisk/bridge_channel.h, apps/app_dumpchan.c,
- channels/chan_motif.c, res/res_agi.c: media formats: re-architect
- handling of media for performance improvements In the old times
- media formats were represented using a bit field. This was fast
- but had a few limitations. 1. Asterisk was limited in how many
- formats it could handle. 2. Formats, being a bit field, could not
- include any attribute information. A format was strictly its
- type, e.g., "this is ulaw". This was changed in Asterisk 10 (see
- https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
- for notes on that work) which led to the creation of the
- ast_format structure. This structure allowed Asterisk to handle
- attributes and bundle information with a format. Additionally,
- ast_format_cap was created to act as a container for multiple
- formats that, together, formed the capability of some entity.
- Another mechanism was added to allow logic to be registered which
- performed format attribute negotiation. Everywhere throughout the
- codebase Asterisk was changed to use this strategy.
- Unfortunately, in software, there is no free lunch. These new
- capabilities came at a cost. Performance analysis and profiling
- showed that we spend an inordinate amount of time comparing,
- copying, and generally manipulating formats and their related
- structures. Basic prototyping has shown that a reasonably large
- performance improvement could be made in this area. This patch is
- the result of that project, which overhauled the media format
- architecture and its usage in Asterisk to improve performance.
- Generally, the new philosophy for handling formats is as follows:
- * The ast_format structure is reference counted. This removed a
- large amount of the memory allocations and copying that was done
- in prior versions. * In order to prevent race conditions while
- keeping things performant, the ast_format structure is immutable
- by convention and lock-free. Violate this tenet at your peril! *
- Because formats are reference counted, codecs are also reference
- counted. The Asterisk core generally provides built-in codecs and
- caches the ast_format structures created to represent them.
- Generally, to prevent inordinate amounts of module reference
- bumping, codecs and formats can be added at run-time but cannot
- be removed. * All compatibility with the bit field representation
- of codecs/formats has been moved to a compatibility API. The
- primary user of this representation is chan_iax2, which must
- continue to maintain its bit-field usage of formats for
- interoperability concerns. * When a format is negotiated with
- attributes, or when a format cannot be represented by one of the
- cached formats, a new format object is created or cloned from an
- existing format. That format may have the same codec underlying
- it, but is a different format than a version of the format with
- different attributes or without attributes. * While formats are
- reference counted objects, the reference count maintained on the
- format should be manipulated with care. Formats are generally
- cached and will persist for the lifetime of Asterisk and do not
- explicitly need to have their lifetime modified. An exception to
- this is when the user of a format does not know where the format
- came from *and* the user may outlive the provider of the format.
- This occurs, for example, when a format is read from a channel:
- the channel may have a format with attributes (hence, non-cached)
- and the user of the format may last longer than the channel (if
- the reference to the channel is released prior to the format's
- reference). For more information on this work, see the API design
- notes:
- https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
- Finally, this work was the culmination of a large number of
- developer's efforts. Extra thanks goes to Corey Farrell, who took
- on a large amount of the work in the Asterisk core, chan_sip, and
- was an invaluable resource in peer reviews throughout this
- project. There were a substantial number of patches contributed
- during this work; the following issues/patch names simply reflect
- some of the work (and will cause the release scripts to give
- attribution to the individuals who work on them). Reviews:
- https://reviewboard.asterisk.org/r/3814
- https://reviewboard.asterisk.org/r/3808
- https://reviewboard.asterisk.org/r/3805
- https://reviewboard.asterisk.org/r/3803
- https://reviewboard.asterisk.org/r/3801
- https://reviewboard.asterisk.org/r/3798
- https://reviewboard.asterisk.org/r/3800
- https://reviewboard.asterisk.org/r/3794
- https://reviewboard.asterisk.org/r/3793
- https://reviewboard.asterisk.org/r/3792
- https://reviewboard.asterisk.org/r/3791
- https://reviewboard.asterisk.org/r/3790
- https://reviewboard.asterisk.org/r/3789
- https://reviewboard.asterisk.org/r/3788
- https://reviewboard.asterisk.org/r/3787
- https://reviewboard.asterisk.org/r/3786
- https://reviewboard.asterisk.org/r/3784
- https://reviewboard.asterisk.org/r/3783
- https://reviewboard.asterisk.org/r/3778
- https://reviewboard.asterisk.org/r/3774
- https://reviewboard.asterisk.org/r/3775
- https://reviewboard.asterisk.org/r/3772
- https://reviewboard.asterisk.org/r/3761
- https://reviewboard.asterisk.org/r/3754
- https://reviewboard.asterisk.org/r/3753
- https://reviewboard.asterisk.org/r/3751
- https://reviewboard.asterisk.org/r/3750
- https://reviewboard.asterisk.org/r/3748
- https://reviewboard.asterisk.org/r/3747
- https://reviewboard.asterisk.org/r/3746
- https://reviewboard.asterisk.org/r/3742
- https://reviewboard.asterisk.org/r/3740
- https://reviewboard.asterisk.org/r/3739
- https://reviewboard.asterisk.org/r/3738
- https://reviewboard.asterisk.org/r/3737
- https://reviewboard.asterisk.org/r/3736
- https://reviewboard.asterisk.org/r/3734
- https://reviewboard.asterisk.org/r/3722
- https://reviewboard.asterisk.org/r/3713
- https://reviewboard.asterisk.org/r/3703
- https://reviewboard.asterisk.org/r/3689
- https://reviewboard.asterisk.org/r/3687
- https://reviewboard.asterisk.org/r/3674
- https://reviewboard.asterisk.org/r/3671
- https://reviewboard.asterisk.org/r/3667
- https://reviewboard.asterisk.org/r/3665
- https://reviewboard.asterisk.org/r/3625
- https://reviewboard.asterisk.org/r/3602
- https://reviewboard.asterisk.org/r/3519
- https://reviewboard.asterisk.org/r/3518
- https://reviewboard.asterisk.org/r/3516
- https://reviewboard.asterisk.org/r/3515
- https://reviewboard.asterisk.org/r/3512
- https://reviewboard.asterisk.org/r/3506
- https://reviewboard.asterisk.org/r/3413
- https://reviewboard.asterisk.org/r/3410
- https://reviewboard.asterisk.org/r/3387
- https://reviewboard.asterisk.org/r/3388
- https://reviewboard.asterisk.org/r/3389
- https://reviewboard.asterisk.org/r/3390
- https://reviewboard.asterisk.org/r/3321
- https://reviewboard.asterisk.org/r/3320
- https://reviewboard.asterisk.org/r/3319
- https://reviewboard.asterisk.org/r/3318
- https://reviewboard.asterisk.org/r/3266
- https://reviewboard.asterisk.org/r/3265
- https://reviewboard.asterisk.org/r/3234
- https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close
- Reported by: mjordan media_formats_translation_core.diff uploaded
- by kharwell (License 6464) rb3506.diff uploaded by mjordan
- (License 6283) media_format_app_file.diff uploaded by kharwell
- (License 6464) misc-2.diff uploaded by file (License 5000)
- chan_mild-3.diff uploaded by file (License 5000)
- chan_obscure.diff uploaded by file (License 5000) jingle.diff
- uploaded by file (License 5000) funcs.diff uploaded by file
- (License 5000) formats.diff uploaded by file (License 5000)
- core.diff uploaded by file (License 5000) bridges.diff uploaded
- by file (License 5000) mf-codecs-2.diff uploaded by file (License
- 5000) mf-app_fax.diff uploaded by file (License 5000)
- mf-apps-3.diff uploaded by file (License 5000)
- media-formats-3.diff uploaded by file (License 5000)
- ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License
- 5909) rb3689.patch uploaded by mjordan (License 6283)
- ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283)
- mf-attributes-3.diff uploaded by file (License 5000)
- ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by
- coreyfarrell (License 5909) rb3800.patch uploaded by jrose
- (License 6182) chan_sip.diff uploaded by mjordan (License 6283)
- rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959
- #close Tested by: sgriepentrog, mjordan, coreyfarrell
- sip_cleanup.diff uploaded by opticron (License 6273)
- chan_sip_caps.diff uploaded by mjordan (License 6283)
- rb3751.patch uploaded by coreyfarrell (License 5909)
- chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960
- #close Tested by: opticron direct_media.diff uploaded by opticron
- (License 6273) pjsip-direct-media.diff uploaded by file (License
- 5000) format_cap_remove.diff uploaded by opticron (License 6273)
- media_format_fixes.diff uploaded by opticron (License 6273)
- chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966
- #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti
- (License 5621) chan_dahdi.diff uploaded by file (License 5000)
- ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron,
- file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by
- rmudgett (License 5621) moh_cleanup.diff uploaded by opticron
- (License 6273) bridge_leak.diff uploaded by opticron (License
- 6273) translate.diff uploaded by file (License 5000) rb3795.patch
- uploaded by rmudgett (License 5621) tls_fix.diff uploaded by
- mjordan (License 6283) fax-mf-fix-2.diff uploaded by file
- (License 5000) rtp_transfer_stuff uploaded by mjordan (License
- 6283) rb3787.patch uploaded by rmudgett (License 5621)
- media-formats-explicit-translate-format-3.diff uploaded by file
- (License 5000) format_cache_case_fix.diff uploaded by opticron
- (License 6273) rb3774.patch uploaded by rmudgett (License 5621)
- rb3775.patch uploaded by rmudgett (License 5621)
- rtp_engine_fix.diff uploaded by opticron (License 6273)
- rtp_crash_fix.diff uploaded by opticron (License 6273)
- rb3753.patch uploaded by mjordan (License 6283) rb3750.patch
- uploaded by mjordan (License 6283) rb3748.patch uploaded by
- rmudgett (License 5621) media_format_fixes.diff uploaded by
- opticron (License 6273) rb3740.patch uploaded by mjordan (License
- 6283) rb3739.patch uploaded by mjordan (License 6283)
- rb3734.patch uploaded by mjordan (License 6283) rb3689.patch
- uploaded by mjordan (License 6283) rb3674.patch uploaded by
- coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell
- (License 5909) rb3667.patch uploaded by coreyfarrell (License
- 5909) rb3665.patch uploaded by mjordan (License 6283)
- rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch
- uploaded by coreyfarrell (License 5909)
- format_compatibility-2.diff uploaded by file (License 5000)
- core.diff uploaded by file (License 5000)
-
-2014-07-18 21:48 +0000 [r419022] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
- res/stasis_recording/stored.c, res/res_ari_recordings.c, /,
- include/asterisk/stasis_app_recording.h,
- res/ari/resource_recordings.h, CHANGES: ari: Add a copy operation
- for stored recordings This patch adds a new operation for stored
- recordings, copy. It takes an existing stored recording and makes
- a copy of it in the same directory or a relative directory under
- the stored recording directory.
- /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name}
- This is particularly useful for voicemail-esque applications,
- which may need to copy or move recordings around a directory
- structure. Review: https://reviewboard.asterisk.org/r/3768/
- ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam
- Galarneau ........ Merged revisions 419021 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-18 21:25 +0000 [r418997-419020] Corey Farrell <git@cfware.com>
-
- * main/stasis_message_router.c, /: stasis: fix call to ao2_t_alloc
- for stasis_message_router_create This fixes a build failure
- introduced by r3821. struct stasis_topic is opaque, so
- topic->name is unavailable. Switch to using stasis_topic_name().
- ........ Merged revisions 419019 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis.c, main/stasis_cache_pattern.c,
- main/stasis_message.c, main/stasis_message_router.c, /: stasis:
- use ao2_t_alloc for certain object allocators Add tags to stasis
- objects using the name. This makes it easier to track the source
- of certain stasis ref leaks. Review:
- https://reviewboard.asterisk.org/r/3821/ ........ Merged
- revisions 418996 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-18 19:07 +0000 [r418980] Kinsey Moore <kmoore@digium.com>
-
- * res/res_fax_spandsp.c: Fix build in dev-mode
-
-2014-07-18 17:55 +0000 [r418961-418963] Scott Griepentrog <sgriepentrog@digium.com>
-
- * res/res_pjsip_pubsub.c, main/astobj2.c,
- include/asterisk/astobj2.h, main/logger.c, main/utils.c: astobj2:
- assert on invalid ref and backtrace cleanup If a reference count
- goes negative, instead of just logging that fact, be more helpful
- with a backtrace and an assert that will DO_CRASH. This patch
- also removes the duplicate ao2_bt() function and cleans up
- extraneous usage of the ast_log_backtrace() call. Review:
- https://reviewboard.asterisk.org/r/3765/
-
- * /, channels/chan_sip.c: media formats: fix ref leak of peer for
- mwi subscription Holding a reference to the peer during mwi
- subscriptions resulted in a circular reference because the final
- event message would not be sent until destruction of the peer.
- Instead, pass the name of the peer to the event callback so that
- it can fail gracefully after the peer has gone. ASTERISK-23959
- Review: https://reviewboard.asterisk.org/r/3754/ ........ Merged
- revisions 418636 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/features_config.c: feature_config: insure featuregroups
- and applicationmaps are initialized If the features.conf is
- missing, the cfg->featurgroups and cfg->applicationmaps is not
- initialized, resulting in assert on ao2_find of a null container.
- This patch changes the initialization call and adds asserts for a
- safeguard. Review: https://reviewboard.asterisk.org/r/3809/
- ........ Merged revisions 418886 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-18 16:47 +0000 [r418938] Richard Mudgett <rmudgett@digium.com>
-
- * funcs/func_audiohookinherit.c, /: func_audiohookinherit.c: Fixup
- some XML documentation wording. ........ Merged revisions 418937
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-18 16:28 +0000 [r418911-418936] Jonathan Rose <jrose@digium.com>
-
- * main/channel.c, funcs/func_audiohookinherit.c, /,
- include/asterisk/audiohook.h, main/framehook.c, res/res_fax.c,
- main/bridge_basic.c, include/asterisk/res_fax.h,
- bridges/bridge_native_rtp.c, main/audiohook.c, CHANGES,
- include/asterisk/framehook.h, res/res_pjsip_refer.c: Channels:
- Masquerades to automatically move frame/audio hooks Whenever
- possible, audiohooks and framehooks will now be copied over to
- the channel that the masquerading channel gets cloned into. This
- should occur for all audiohooks and most framehooks. As a result,
- in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
- deprecated and its behavior is essentially the new default for
- all audiohooks, plus some additional audiohooks/framehooks.
- Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged
- revisions 418914 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_fax.c, include/asterisk/res_fax.h, CHANGES,
- res/res_fax.exports.in, res/res_fax_spandsp.c: res_fax: Provide
- AMI equivalents for fax CLI commands Specifically the following
- equivalents were created: fax show session -> FAXSession fax show
- sessions -> FAXSessions fax show stats -> FAXStats Review:
- https://reviewboard.asterisk.org/r/3666/
-
-2014-07-18 00:11 +0000 [r418893-418895] Sean Bright <sean@malleable.com>
-
- * config.sub, menuselect/config.guess, menuselect/config.sub,
- config.guess: Update config.guess and config.sub
-
- * autoconf/ast_ext_tool_check.m4: Add missing file from previous
- commit.
-
- * menuselect/aclocal.m4, menuselect/configure,
- menuselect/acinclude.m4 (removed), menuselect/bootstrap.sh,
- menuselect/autoconfig.h.in: Import Asterisk's autoconf magic
- instead of using our own.
-
-2014-07-17 21:17 +0000 [r418832-418870] Matthew Jordan <mjordan@digium.com>
-
- * configs/samples/acl.conf.sample (added),
- configs/samples/extensions.conf.sample (added),
- configs/res_parking.conf.sample (removed),
- configs/samples/cel_sqlite3_custom.conf.sample (added),
- configs/cdr_sqlite3_custom.conf.sample (removed),
- configs/modules.conf.sample (removed),
- configs/samples/cli_aliases.conf.sample (added),
- configs/meetme.conf.sample (removed),
- configs/cdr_pgsql.conf.sample (removed),
- configs/samples/extensions.ael.sample (added),
- configs/samples/cdr_adaptive_odbc.conf.sample (added),
- configs/samples/motif.conf.sample (added),
- configs/samples/extensions_minivm.conf.sample (added),
- configs/samples/res_curl.conf.sample (added),
- configs/res_config_sqlite3.conf.sample (removed),
- configs/mgcp.conf.sample (removed), configs/dsp.conf.sample
- (removed), configs/udptl.conf.sample (removed),
- configs/sip.conf.sample (removed), configs/dbsep.conf.sample
- (removed), configs/queuerules.conf.sample (removed),
- configs/samples/cdr_mysql.conf.sample (added),
- configs/confbridge.conf.sample (removed),
- configs/samples/cdr_odbc.conf.sample (added),
- configs/samples/minivm.conf.sample (added),
- configs/enum.conf.sample (removed),
- configs/samples/codecs.conf.sample (added),
- configs/samples/chan_dahdi.conf.sample (added),
- configs/samples/cdr_custom.conf.sample (added),
- configs/samples/res_config_mysql.conf.sample (added),
- configs/samples/dundi.conf.sample (added),
- configs/samples/oss.conf.sample (added),
- configs/samples/app_mysql.conf.sample (added),
- configs/samples/queues.conf.sample (added),
- configs/samples/cdr.conf.sample (added),
- configs/samples/cdr_syslog.conf.sample (added),
- configs/festival.conf.sample (removed),
- configs/samples/cel_pgsql.conf.sample (added),
- configs/http.conf.sample (removed), configs/phoneprov.conf.sample
- (removed), configs/alarmreceiver.conf.sample (removed),
- configs/samples/features.conf.sample (added),
- configs/cdr_tds.conf.sample (removed),
- configs/func_odbc.conf.sample (removed),
- configs/samples/logger.conf.sample (added),
- configs/samples/res_odbc.conf.sample (added),
- configs/samples/agents.conf.sample (added),
- configs/res_fax.conf.sample (removed),
- configs/samples/xmpp.conf.sample (added),
- configs/iaxprov.conf.sample (removed),
- configs/res_pgsql.conf.sample (removed),
- configs/extensions.conf.sample (removed),
- configs/chan_mobile.conf.sample (removed), configs/asterisk.adsi
- (removed), configs/cel_sqlite3_custom.conf.sample (removed),
- configs/users.conf.sample (removed),
- configs/samples/res_pktccops.conf.sample (added),
- configs/samples/amd.conf.sample (added), configs/rtp.conf.sample
- (removed), configs/samples/res_parking.conf.sample (added),
- configs/hep.conf.sample (removed),
- configs/samples/modules.conf.sample (added),
- configs/cel_tds.conf.sample (removed),
- configs/res_curl.conf.sample (removed),
- configs/samples/skinny.conf.sample (added),
- configs/samples/cdr_pgsql.conf.sample (added),
- configs/samples/sip_notify.conf.sample (added),
- configs/samples/test_sorcery.conf.sample (added),
- configs/samples/dsp.conf.sample (added),
- configs/ss7.timers.sample (removed),
- configs/samples/udptl.conf.sample (added),
- configs/cdr_odbc.conf.sample (removed),
- configs/samples/sip.conf.sample (added),
- configs/minivm.conf.sample (removed),
- configs/res_config_sqlite.conf.sample (removed),
- configs/codecs.conf.sample (removed), configs/osp.conf.sample
- (removed), configs/samples/cel_custom.conf.sample (added),
- configs/samples/dbsep.conf.sample (added),
- configs/samples/app_skel.conf.sample (added),
- configs/console.conf.sample (removed),
- configs/cdr_manager.conf.sample (removed),
- configs/cdr_custom.conf.sample (removed),
- configs/chan_dahdi.conf.sample (removed),
- configs/res_config_mysql.conf.sample (removed),
- configs/samples/statsd.conf.sample (added),
- configs/cli.conf.sample (removed), configs/queues.conf.sample
- (removed), configs/cdr_syslog.conf.sample (removed), UPGRADE.txt,
- configs/manager.conf.sample (removed),
- configs/samples/res_corosync.conf.sample (added),
- configs/features.conf.sample (removed), configs/sla.conf.sample
- (removed), configs/logger.conf.sample (removed),
- configs/res_odbc.conf.sample (removed),
- configs/agents.conf.sample (removed),
- configs/samples/ooh323.conf.sample (added), Makefile,
- configs/xmpp.conf.sample (removed),
- configs/samples/phoneprov.conf.sample (added),
- configs/samples/alarmreceiver.conf.sample (added),
- configs/samples/cdr_tds.conf.sample (added),
- configs/extconfig.conf.sample (removed),
- configs/samples/func_odbc.conf.sample (added),
- configs/samples/res_fax.conf.sample (added),
- configs/samples/iaxprov.conf.sample (added),
- configs/samples/res_ldap.conf.sample (added),
- configs/samples/dnsmgr.conf.sample (added),
- configs/res_pktccops.conf.sample (removed),
- configs/cel.conf.sample (removed),
- configs/samples/res_pgsql.conf.sample (added),
- configs/samples/chan_mobile.conf.sample (added),
- configs/samples/asterisk.adsi (added),
- configs/samples/users.conf.sample (added),
- configs/samples/rtp.conf.sample (added),
- configs/phone.conf.sample (removed), configs/skinny.conf.sample
- (removed), configs/muted.conf.sample (removed),
- configs/samples/hep.conf.sample (added), configs/iax.conf.sample
- (removed), configs/samples/cel_tds.conf.sample (added),
- configs/sip_notify.conf.sample (removed),
- configs/samples/telcordia-1.adsi (added),
- configs/samples/alsa.conf.sample (added),
- configs/samples/adsi.conf.sample (added),
- configs/test_sorcery.conf.sample (removed),
- configs/samples/followme.conf.sample (added),
- configs/samples/asterisk.conf.sample (added),
- configs/extensions.lua.sample (removed), configs/say.conf.sample
- (removed), configs/cel_custom.conf.sample (removed),
- configs/samples/ss7.timers.sample (added),
- configs/samples/cel_odbc.conf.sample (added),
- configs/app_skel.conf.sample (removed),
- configs/samples/ccss.conf.sample (added),
- configs/cli_permissions.conf.sample (removed),
- configs/statsd.conf.sample (removed),
- configs/samples/res_config_sqlite.conf.sample (added),
- configs/config_test.conf.sample (removed),
- configs/indications.conf.sample (removed),
- configs/samples/osp.conf.sample (added),
- configs/samples/cdr_manager.conf.sample (added),
- configs/samples/console.conf.sample (added),
- configs/voicemail.conf.sample (removed),
- configs/res_corosync.conf.sample (removed),
- configs/misdn.conf.sample (removed),
- configs/samples/cli.conf.sample (added), configs/ari.conf.sample
- (removed), configs/ooh323.conf.sample (removed),
- configs/samples/calendar.conf.sample (added),
- configs/samples/res_stun_monitor.conf.sample (added),
- configs/samples/manager.conf.sample (added),
- configs/samples/pjsip_notify.conf.sample (added),
- configs/samples/sla.conf.sample (added),
- configs/musiconhold.conf.sample (removed),
- configs/pjsip.conf.sample (removed), configs/sorcery.conf.sample
- (removed), configs/vpb.conf.sample (removed),
- configs/unistim.conf.sample (removed),
- configs/res_ldap.conf.sample (removed),
- configs/dnsmgr.conf.sample (removed),
- configs/samples/extconfig.conf.sample (added),
- configs/samples/res_snmp.conf.sample (added),
- configs/acl.conf.sample (removed),
- configs/samples/smdi.conf.sample (added),
- configs/samples/cel.conf.sample (added),
- configs/cli_aliases.conf.sample (removed),
- configs/samples/cdr_sqlite3_custom.conf.sample (added),
- configs/extensions.ael.sample (removed),
- configs/cdr_adaptive_odbc.conf.sample (removed),
- configs/samples/phone.conf.sample (added),
- configs/extensions_minivm.conf.sample (removed),
- configs/motif.conf.sample (removed), configs/telcordia-1.adsi
- (removed), configs/samples/meetme.conf.sample (added),
- configs/adsi.conf.sample (removed), configs/alsa.conf.sample
- (removed), configs/samples/muted.conf.sample (added),
- configs/followme.conf.sample (removed),
- configs/asterisk.conf.sample (removed),
- configs/samples/iax.conf.sample (added),
- configs/samples/res_config_sqlite3.conf.sample (added),
- configs/samples/mgcp.conf.sample (added),
- configs/cel_odbc.conf.sample (removed), configs/ccss.conf.sample
- (removed), configs/cdr_mysql.conf.sample (removed),
- configs/samples/extensions.lua.sample (added),
- configs/samples/say.conf.sample (added),
- configs/dundi.conf.sample (removed),
- configs/samples/queuerules.conf.sample (added),
- configs/oss.conf.sample (removed), configs/app_mysql.conf.sample
- (removed), configs/samples/confbridge.conf.sample (added),
- configs/samples/cli_permissions.conf.sample (added),
- configs/samples/enum.conf.sample (added),
- configs/samples/config_test.conf.sample (added),
- configs/cdr.conf.sample (removed),
- configs/samples/indications.conf.sample (added),
- configs/cel_pgsql.conf.sample (removed),
- configs/res_stun_monitor.conf.sample (removed),
- configs/calendar.conf.sample (removed),
- configs/samples/voicemail.conf.sample (added),
- configs/pjsip_notify.conf.sample (removed),
- configs/samples/misdn.conf.sample (added),
- configs/samples/ari.conf.sample (added),
- configs/samples/festival.conf.sample (added),
- configs/samples/http.conf.sample (added),
- configs/res_snmp.conf.sample (removed),
- configs/samples/musiconhold.conf.sample (added),
- configs/samples/pjsip.conf.sample (added),
- configs/samples/sorcery.conf.sample (added),
- configs/samples/vpb.conf.sample (added), configs/smdi.conf.sample
- (removed), configs/samples/unistim.conf.sample (added),
- configs/samples (added), configs/amd.conf.sample (removed):
- configs: Move sample config files into a subdirectory of configs
- This moves all samples configs from configs/ to configs/samples.
- This allows for additional sets of sample configuration files to
- be added in the future. Review:
- https://reviewboard.asterisk.org/r/3804/
-
- * channels/chan_sip.c, UPGRADE.txt: chan_sip: Make
- progressinband=never really mean 'never' progressinband=never in
- sip.conf is easily defeated if an onward trunk sends a progress
- indication of its own. This is almost certain to happen if the
- onward trunk is ISDN or IAX as these technologies send a progress
- indication even if early media is not required. This progress
- message is passed to the caller, and causes the "never" option to
- be rather badly named. This patch changes the behaviour of this
- setting in the following ways: 1) In sip_write(), do not pass the
- media unless we have either progressed beyond INV_EARLY_MEDIA, or
- we are in INV_EARLY_MEDIA state, and early media is both set-up
- and wanted. This helps resolve double-ringing on some buggy
- handsets. 2) In sip_indicate(), if we see AST_CONTROL_PROGRESS,
- but SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to
- avoid implicitly enabling early media. Avoid sending double ring
- indications. NOTE: the meaning of the SIP_PROGRESS_SENT flag
- changes slightly in this patch to also encapsulate the fact that
- a channel has *sent or received* a 183 Progress indication. This
- makes the updated code in sip_write() much more simple. Review:
- https://reviewboard.asterisk.org/r/3700 ASTERISK-23972 #close
- Reported by: Steve Davies patches:
- inband_never_present_early_media2 uploaded by Steve Davies
- (License 5012)
-
- * menuselect: Add svn:ignore property
-
- * UPGRADE.txt, menuselect/configure, menuselect/configure.ac,
- configure, configure.ac: configure: Fix libxml2 development
- library dependency checking The commit that added libxml2 support
- didn't fully check for the libxml2 development script in the
- Asterisk configure file. As a result, Asterisk could be
- configured, then fail on menuselect. This patch fixes it so that
- Asterisk should detect the libxml2 dependency failure first.
-
- * menuselect/makeopts.in, menuselect/autoconfig.h.in,
- menuselect/menuselect.h, menuselect/example_menuselect-tree,
- configure, include/asterisk/autoconfig.h.in, menuselect/Makefile,
- menuselect/README, menuselect/aclocal.m4, configure.ac,
- UPGRADE.txt, menuselect/configure, menuselect/configure.ac,
- menuselect/menuselect.c, menuselect/acinclude.m4: menuselect: Add
- libxml2 support (Patch 3) This is the final patch in adding
- menuselect to Asterisk. - The first patch (r418832) added
- menuselect along with mxml - The second patch (r418833) removed
- mxml from menuselect This patch adds support for libxml2 to
- menuselect, and makes libxml2 a required library for Asterisk.
- Note that the libxml2 portion of this patch was written by Sean
- Bright, and was made available on a team branch:
- http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/
- Review: https://reviewboard.asterisk.org/r/3773/ ASTERISK-20703
- #close patches: some_mysterious_team_branch uploaded by
- seanbright (License 5060)
-
- * menuselect/mxml (removed): menuselect: Remove mxml from
- menuselect (Patch 2) This is the second patch that adds
- menuselect to Asterisk trunk. The previous commit (r418832) added
- menuselect along with mxml; this patch removes mxml completely
- from Menuselect. A subsequent patch will switch menuselect over
- to using libxml2, and make libxml2 a required dependency for
- Asterisk. ASTERISK-20703
-
- * menuselect/mxml/configure.in (added), menuselect/acinclude.m4
- (added), menuselect/mxml/mxml.list.in (added),
- menuselect/mxml/README (added), menuselect/linkedlists.h (added),
- menuselect/mxml (added), menuselect/mxml/config.h.in (added),
- menuselect/aclocal.m4 (added), menuselect/install-sh (added),
- menuselect/mxml/mxml-string.c (added),
- menuselect/menuselect_stub.c (added), menuselect/make_version
- (added), menuselect/mxml/mxml-entity.c (added),
- menuselect/bootstrap.sh (added), menuselect/makeopts.in (added),
- menuselect/autoconfig.h.in (added), menuselect/config.guess
- (added), menuselect/mxml/install-sh (added),
- menuselect/test/build_tools/menuselect-deps (added), /,
- menuselect/contrib/menuselect-dummy (added),
- menuselect/config.sub (added), menuselect/mxml/configure (added),
- menuselect/mxml/Makefile.in (added), menuselect (added),
- menuselect/contrib (added), menuselect/mxml/mxml.pc.in (added),
- menuselect/configure.ac (added), menuselect/mxml/mxml-set.c
- (added), menuselect/contrib/Makefile-dummy (added),
- menuselect/mxml/ANNOUNCEMENT (added), menuselect/missing (added),
- menuselect/menuselect_curses.c (added),
- menuselect/example_menuselect-tree (added), menuselect/Makefile
- (added), menuselect/mxml/mxml-search.c (added), menuselect/test
- (added), menuselect/test/menuselect-tree (added),
- menuselect/mxml/mxml.h (added), menuselect/mxml/mxml-index.c
- (added), menuselect/configure (added),
- menuselect/menuselect_newt.c (added), menuselect/mxml/mxml-attr.c
- (added), menuselect/mxml/mxml-private.c (added),
- menuselect/menuselect.c (added), menuselect/mxml/CHANGES (added),
- menuselect/mxml/COPYING (added), menuselect/mxml/mxml-file.c
- (added), menuselect/menuselect.h (added),
- menuselect/menuselect_gtk.c (added), menuselect/README (added),
- menuselect/strcompat.c (added), menuselect/mxml/mxml-node.c
- (added), menuselect/test/build_tools (added): menuselect: Add
- menuselect to Asterisk trunk (Patch 1) This is the first patch
- that adds menuselect to Asterisk trunk, and removes the
- svn:externals property. This is being done for two reasons: (1)
- The removal of external repositories eases a future migration to
- git (2) Asterisk is now the only thing that uses menuselect; as a
- result, there's little need to keep it in an external repository
- Subsequent patches will remove the mxml dependency from
- menuselect and tidy up the build system. ASTERISK-20703
-
-2014-07-17 14:28 +0000 [r418811] Kinsey Moore <kmoore@digium.com>
-
- * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix threewaytransfer
- reporting Ensure that three-way transfers can be reported even if
- featuremap is non-NULL. ........ Merged revisions 418810 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-16 23:08 +0000 [r418788] Corey Farrell <git@cfware.com>
-
- * /, channels/dahdi/bridge_native_dahdi.c: Remove include of
- astobj.h from channels/dahdi/bridge_native_dahdi.c. The include
- was unneeded, this is split off from r3758 as it applies to 12.
- ........ Merged revisions 418787 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-16 14:03 +0000 [r418717-418757] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c, CHANGES, res/res_pjsip.c,
- channels/chan_pjsip.c, include/asterisk/res_pjsip.h,
- contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py
- (added), /, configs/pjsip.conf.sample: res_pjsip: Support setting
- a default accountcode on endpoints Most channel drivers let you
- specify a default accountcode to be set on channels associated
- with a particular peer/endpoint/object. Prior to this patch,
- chan_pjsip/res_pjsip did not support such a setting. This patch
- adds a new setting to the res_pjsip endpoint object,
- 'accountcode'. When a channel is created that is associated with
- an endpoint with this value set, the channel will automatically
- have its accountcode property set to the value configured for the
- endpoint. Review: https://reviewboard.asterisk.org/r/3724/
- ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged
- revisions 418756 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * cdr/cdr_pgsql.c, CHANGES, configs/cdr_pgsql.conf.sample,
- configs/res_pgsql.conf.sample, cel/cel_pgsql.c,
- res/res_config_pgsql.c, configs/cel_pgsql.conf.sample: cel_pgsql,
- cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name
- support This patch adds support for the PostgreSQL
- application_name connection setting. When the appropriate
- PostgreSQL module's configuration is set with an application
- name, the name will be passed to PostgreSQL on connection and
- displayed in the database's pg_stat_activity view, as well as in
- CSV logs. This aids in managing which applications/servers are
- connected to a PostgreSQL database, as well as tracing the
- activity of those connections. Review:
- https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close
- Reported by: Gergely Domodi patches: pgsql_application_name.patch
- uploaded by Gergely Domodi (License 6610)
-
- * codecs/codec_adpcm.c, main/format.c: codec_adpcm: Change
- description of codec "ADPCM" to "Dialogic ADPCM" Technically,
- ADPCM is a method that can be applied to several codecs.
- Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec. See
- http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information
- about said codec. Review: https://reviewboard.asterisk.org/r/3744
- patches: rb3744.patch uploaded by dennis.guse (License 6513)
-
- * UPGRADE.txt, main/manager.c, /: manager: Return ActionID on
- nominal responses to PresenceState action When the PresenceState
- action is executed, the nominal path fails to include the
- ActionID in the successful response. This patch adds a call to
- astman_start_ack, which guarantees that an ActionID (if provided)
- will be sent back to the AMI client. Unlike the Asterisk 11 and
- 12 patches, this patch also deprecates the duplicate Message key
- in the response to the action, replacing it with the key
- 'PresenceMessage'. Review:
- https://reviewboard.asterisk.org/r/3776/ ASTERISK-23985 #close
- ........ Merged revisions 418713 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418714 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-15 23:03 +0000 [r418716] Kinsey Moore <kmoore@digium.com>
-
- * /, main/bridge_channel.c: TEST_FRAMEWORK: Fix ref leak in feature
- activation This fixes two reference leaks that would occur when
- TEST_FRAMEWORK was enabled and features were successfully
- executed. ........ Merged revisions 418715 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-15 17:57 +0000 [r418654] Jonathan Rose <jrose@digium.com>
-
- * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty
- strings as argument Previously these two dialplan functions would
- issue warnings and return failure when an empty string is used as
- the argument. Now they will not issue a warning and will
- successfully return an empty string. ASTERISK-23911 #close
- Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3745/ ........ Merged
- revisions 418641 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 418649 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418650 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-15 12:11 +0000 [r418616] Sean Bright <sean@malleable.com>
-
- * main/asterisk.c: Update Asterisk copyright year in
- main/asterisk.c It's been 2014 for like... 6 months.
-
-2014-07-14 14:55 +0000 [r418566-418587] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/logger.h, /: logger.h: Extract DEBUG_ATLEAST()
- to complement VERBOSITY_ATLEAST(). ........ Merged revisions
- 418586 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/jabber.h (removed), include/asterisk/jingle.h
- (removed), include/asterisk/frame_defs.h (removed),
- configs/h323.conf.sample (removed): Actually delete the removed
- files.
-
-2014-07-13 21:57 +0000 [r418507] Corey Farrell <git@cfware.com>
-
- * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work
- around REF_DEBUG race which causes out of order log entries *
- Update refcounter.py to use delta's to track the current
- reference count. * Use result from internal_ao2_ref to write
- old_refcount to refs_log. Review:
- https://reviewboard.asterisk.org/r/3756/ ........ Merged
- revisions 418504 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 418505 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418506 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-13 20:08 +0000 [r418488] Scott Griepentrog <sgriepentrog@digium.com>
-
- * include/asterisk/astobj2.h: astobj2: correct define for
- ao2_t_cleanup This change maps the ao2_t_cleanup() function to
- the correct debug function so that it can be used. Review:
- https://reviewboard.asterisk.org/r/3764/
-
-2014-07-13 16:48 +0000 [r418448-418467] Corey Farrell <git@cfware.com>
-
- * main/manager.c, /, apps/app_skel.c: Fix minor reference leaks in
- app_skel and TEST_FRAMEWORK * Cleanup games object in app_skel. *
- Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+).
- Review: https://reviewboard.asterisk.org/r/3757/ ........ Merged
- revisions 418465 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418466 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/jabber.h, include/asterisk/jingle.h,
- configs/h323.conf.sample: Remove files left behind on removal of
- h323, jingle and jabber. This change removes h323.conf.sample,
- jingle.h, jabber.h left behind by r3698. Review:
- https://reviewboard.asterisk.org/r/3755/
-
-2014-07-11 23:00 +0000 [r418419] Matthew Jordan <mjordan@digium.com>
-
- * main/astobj2.c, include/asterisk/astobj2.h: astobj2: Add tag
- variants for ao2_bump, ao2_cleanup, and ao2_replace Tags are
- useful in hunting down ref imbalances; this patch adds tag
- variants for these commonly used macros/functions. Review:
- https://reviewboard.asterisk.org/r/3750/
-
-2014-07-11 21:10 +0000 [r418397] Corey Farrell <git@cfware.com>
-
- * /, include/asterisk/astobj2.h: astobj2: tweak ao2_replace to do
- nothing when it would be a NoOp This change causes ao2_replace to
- do nothing when src == dst. This avoids REF_DEBUG logging when
- we're not actually doing anything. Review:
- https://reviewboard.asterisk.org/r/3743/ ........ Merged
- revisions 418396 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-11 16:42 +0000 [r418370] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/config.c: config: inform config hook of change when
- writing file When updated configuration is written back to the
- conf file - for example when a user changes their voicemail pin,
- make sure that any config hook that wants to know of changes is
- informed. Review: https://reviewboard.asterisk.org/r/3708/
- ........ Merged revisions 418366 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418369 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-10 15:36 +0000 [r418325] Matthew Jordan <mjordan@digium.com>
-
- * /, include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert
- indentation to tabs This is a whitespace only change. ........
- Merged revisions 418323 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418324 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-10 01:59 +0000 [r418226-418264] Richard Mudgett <rmudgett@digium.com>
-
- * channels/sig_pri.c, /: chan_dahdi/sig_pri: Fix type mismatch in
- the idledial feature's channel creation. Square pegs in round
- holes don't work very well. ........ Merged revisions 418261 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 418262 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 418263 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/stasis/stasis_bridge.h (added), main/bridge_channel.c,
- res/res_stasis.c, /, res/stasis/stasis_bridge.c (added),
- include/asterisk/bridge_channel.h, main/bridge_basic.c: ARI: Make
- mixing bridges propagate linkedids and accountcodes. * Create a
- Stasis bridge sub-class to propagate linkedids and accountcodes.
- * Fixed the basic bridge sub-class to update peeraccount codes
- when the number of channels in the bridge drops back down to two
- parties. * Refactored ast_bridge_channel_update_accountcodes() to
- handle channels joining/leaving the bridge. * Fixed the basic
- bridge sub-class to not call the base bridge class pull method
- twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard
- Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........
- Merged revisions 418225 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-08 14:48 +0000 [r418174-418183] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json,
- rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json,
- rest-api/api-docs/playbacks.json,
- rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/resources.json, include/asterisk/manager.h,
- rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json: manager/ARI: Update version to
- 2.4.0/1.4.0; Update UPGRADE.txt ........ Merged revisions 418182
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix undefined
- function when PJPROJECT is not installed The
- dtls_perform_handshake function was mistakenly placed under the
- guards for USE_PJPROJECT. If PJPROJECT was not installed, the
- function would not be defined, while other functions would
- attempt to still use it. This prevented res_rtp_asterisk from
- being loaded. ASTERISK-24001 #close Reported by: Don Fanning
- ........ Merged revisions 418172 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-07 16:08 +0000 [r418117] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/res_pjsip_body_generator_types.h,
- res/res_pjsip_dialog_info_body_generator.c (added),
- res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c, /,
- include/asterisk/res_pjsip_presence_xml.h:
- res_pjsip_dialog_info_body_generator: Add dialog-info+xml support
- for presence. This module implements dialog-info+xml for the
- purposes of presence. This means that phones such as Grandstreams
- can now subscribe to receive presence information for an
- extension. ASTERISK-21443 #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3705/ ........ Merged
- revisions 418116 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-07 02:15 +0000 [r418090] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/stasis_app.h, res/ari/resource_channels.c,
- res/res_stasis.c, /, res/stasis/app.c: ARI/res_stasis: Subscribe
- to both Local channel halves when originating to app This patch
- fixes two bugs: 1. When originating a channel into a Stasis
- application, we already create a subscription for the channel
- that is going into our Stasis app. Unfortunately, when you create
- a Local channel and pass it off to a Stasis app, you really
- aren't creating just one channel: you're creating two. This patch
- snags the second half of the Local channel pair (assuming it is a
- Local channel pair, but luckily core_local is kind about such
- assumptions) and subscribes to it as well. 2. Subscriptions are a
- bit sticky right now. If a subscription is made, the 'interest'
- count gets bumped on the Stasis subscription - but unless
- something explicitly unsubscribes the channel, said subscription
- sticks around. This is not much of a problem is a user is
- creating the subscription - if they made it, they must want it.
- However, when we are creating implicit subscriptions, we need to
- make sure something clears them out. This patch takes a
- pessimistic approach: it watches the cache updates coming from
- Stasis and, if we notice that the cache just cleared out an
- object, we delete our subscription object. This keeps our ao2
- container of Stasis forwards in an application from growing out
- of hand; it also is a bit more forgiving for end users who may
- not realize they were supposed to unsubscribe from that channel
- that just hung up. Review:
- https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close
- ........ Merged revisions 418089 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-07 01:22 +0000 [r418067-418084] Kinsey Moore <kmoore@digium.com>
-
- * tests/test_cel.c, main/cel.c, channels/chan_pjsip.c,
- res/res_pjsip_session.c, /: CEL: Fix incorrect/missing extra
- field information This corrects two issues with the extra field
- information in Asterisk 12+ in channel event logs. It is possible
- to inject custom values into the dialstatus provided by
- ast_channel_dial_type() Stasis messages that fall outside the
- enumeration allowed for the DIALSTATUS channel variable. CEL now
- filters for the allowed values and ignores other values. The
- "hangupsource" extra field key is always blank if the far end
- channel is a chan_pjsip channel. This is because the hangupsource
- is never set for the pjsip channel driver. This change sets the
- hangupsource whenever a hangup is queued for chan_pjsip channels.
- This corrects an issue with the pjsip channel driver where the
- hangupcause information was not being set properly. Review:
- https://reviewboard.asterisk.org/r/3690/ ........ Merged
- revisions 418071 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/http.c: HTTP: Fix build for gcc 4.10 ........ Merged
- revisions 418066 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-04 15:26 +0000 [r418019-418050] Matthew Jordan <mjordan@digium.com>
-
- * main/Makefile: main/Makefile: fix compilation error of buildinfo
- occurring on 'make install' Egads. Another bad deletion of too
- much when attempting to remove h323 stuff.
-
- * configure.ac, build_tools/menuselect-deps.in, configure,
- main/Makefile: configure: Remove last vestiges of h323; DO create
- menuselect-deps The previous patch (r418034) fixed the 'glitch'
- that the channels/h323 Makefile no longer existed. Unfortunately,
- removing the entire line was a bit of a blunder, as it meant that
- build_tools/menuselect-deps was never generated. Hilarity ensued
- when actually trying to compile. But hey! At least configure
- worked. This patch fixes *that* glitch, and removes some more of
- the vestiges of h323. (It had tendrils in the main Makefile?
- Crazy.)
-
- * configure.ac, configure: configure: Update script to pass if
- channels/h323/Makefile.in does not exist This simply removes that
- check from the configure script, as r418019 removed chan_h323.
-
- * apps/app_dahdibarge.c (removed), configs/gtalk.conf.sample
- (removed), main/pbx.c, apps/app_readfile.c (removed),
- channels/chan_sip.c, configs/jingle.conf.sample (removed),
- UPGRADE.txt, res/res_musiconhold.c, channels/chan_gtalk.c
- (removed), channels/Makefile, CHANGES, res/res_jabber.c
- (removed), channels/h323 (removed), utils/conf2ael.c,
- channels/chan_jingle.c (removed), res/ael/pval.c,
- configs/jabber.conf.sample (removed),
- configs/asterisk.conf.sample, res/res_agi.c, channels/chan_h323.c
- (removed), addons/Makefile, pbx/pbx_realtime.c, utils/ael_main.c,
- include/asterisk/options.h, main/asterisk.c,
- addons/app_saycountpl.c (removed): Remove many deprecated modules
- Billing records are fair, To get paid is quite bright, You should
- really use ODBC; Good-bye cdr_sqlite. Microsoft did once push
- H.323, Hell, we all remember NetMeeting. But try to compile
- chan_h323 now And you will take quite a beating. The XMPP and SIP
- war was fierce, And in the distant fray Was birthed
- res_jabber/chan_jingle; But neither to stay. For everyone did
- care and chase what Google professed. "Free Internet Calling" was
- what devotees cried, But Google did change the specs so often
- That the developers were happy the day chan_gtalk died. And then
- there was that odd application Dedicated to the Polish tongue.
- app_saycountpl was subsumed by Say; One could say its bell was
- rung. To read and parse a file from the dialplan You could (I
- guess) use an application. app_readfile did fill that purpose,
- but I think A function is perhaps better in its creation. Barging
- is rude, I'm not sure why we do it. Inwardly, the caller will
- probably sigh. But if you really must do it, Don't use
- app_dahdibarge, use ChanSpy. We all despise the sound of tinny
- robots It makes our queues so cold. To control such an
- abomination It's better to not use Wait/SetMusicOnHold. It's
- often nice to know properties of a channel It makes our calls
- right We have a nice function called CHANNEL And so SIPCHANINFO
- is sent off into the night. And now things get odd; Apparently
- one could delimit with a colon Properties from the SIPPEER
- function! Commas are in; all others are done. Finally, a word on
- pipes and commas. We're sorry. We can't say it enough. But those
- compatibility options in asterisk.conf; To maintain them forever
- was just too tough. This patch removes: * cdr_sqlite * chan_gtalk
- * chan_jingle * chan_h323 * res_jabber * app_saycountpl *
- app_readfile * app_dahdibarge It removes the following
- applications/functions: * WaitMusicOnHold * SetMusicOnHold *
- SIPCHANINFO It removes the colon delimiter from the SIPPEER
- function. Finally, it also removes all compatibility options that
- were configurable from asterisk.conf, as these all applied to
- compatibility with Asterisk 1.4 systems. Review:
- https://reviewboard.asterisk.org/r/3698/
-
-2014-07-03 22:22 +0000 [r417933-417976] Richard Mudgett <rmudgett@digium.com>
-
- * channels/sig_pri.h, channels/chan_dahdi.c,
- configs/chan_dahdi.conf.sample, /, UPGRADE.txt,
- channels/sig_pri.c: chan_dahdi: Add inband_on_setup_ack
- compatibility option. The new inband_on_setup_ack option causes
- Asterisk to assume inband audio may be present when a
- SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says
- that in scenarios with overlap dialing, when a dialtone is sent
- from the network side, progress indicator 8 "Inband info now
- available" MAY be sent to the CPE if no digits were received with
- the SETUP. It is thus implied that the ie is mandatory if digits
- came with the SETUP and dialtone is needed. This option should be
- enabled, when the network sends dialtone and you want to hear it,
- but the network doesn't send the progress indicator when needed.
- NOTE: For Q.SIG setups this option should be enabled when
- outgoing overlap dialing is also enabled because Q.SIG does not
- send the progress indicator with the SETUP ACK. The commit
- -r413714 (AST-1338) which causes this issue was dealing with a
- SIP-to-ISDN interoperability issue. This commit is a merge of the
- two patches indicated below. ASTERISK-23897 #close Reported by:
- Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded
- by Pavel Troller jira_asterisk_23897_v11.patch (license #5621)
- patch uploaded by rmudgett Review:
- https://reviewboard.asterisk.org/r/3633/ ........ Merged
- revisions 417956 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417957 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417958 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_channels.c, res/res_ari.c, main/manager.c, /:
- res_ari: Fix some off-nominal paths just dropping the HTTP
- connection. * Removed some incorrect newlines on ast_http_error()
- messages in manager.c. * Removed an incorrect newline in
- res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged
- revisions 417932 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-03 17:34 +0000 [r417910-417916] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, channels/chan_dahdi.c: chan_dahdi: Add AMI commands for
- controlling PRI debugging output Adds the following AMI commands:
- PRIDebugSet - Set PRI debug levels for a specific span
- PRIDebugFileSet - Set the file used for PRI debug message output
- PRIDebugFileUnset - Disables file output for PRI debug messages
- Review: https://reviewboard.asterisk.org/r/3681/
-
- * CHANGES, pbx/pbx_config.c, main/pbx.c: pbx_config: Add manager
- actions to add/remove extensions Adds two new manager commands to
- pbx_config - DialplanExtensionAdd and DialplanExtensionRemove
- which allow manager users to create and delete extensions
- respectively. Review: https://reviewboard.asterisk.org/r/3650/
-
-2014-07-03 17:16 +0000 [r417901] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_phoneprov.c, main/http.c, UPGRADE.txt,
- include/asterisk/tcptls.h, res/res_http_post.c,
- res/res_http_websocket.c, configs/http.conf.sample,
- include/asterisk/http.h, main/tcptls.c, res/res_ari.c,
- main/manager.c, /: HTTP: Add persistent connection support.
- Persistent HTTP connection support is needed due to the increased
- usage of the Asterisk core HTTP transport and the frequency at
- which REST API calls are going to be issued. * Add http.conf
- session_keep_alive option to enable persistent connections. *
- Parse and discard optional chunked body extension information and
- trailing request headers. * Increased the maximum
- application/json and application/x-www-form-urlencoded body size
- allowed to 4k. The previous 1k was kind of small. * Removed a
- couple inlined versions of ast_http_manid_from_vars() by calling
- the function. manager.c:generic_http_callback() and
- res_http_post.c:http_post_callback() * Add missing va_end() in
- ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use
- in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott
- Griepentrog Review: https://reviewboard.asterisk.org/r/3691/
- ........ Merged revisions 417880 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-03 16:55 +0000 [r417900] Matthew Jordan <mjordan@digium.com>
-
- * main/tcptls.c, configure, include/asterisk/autoconfig.h.in,
- configure.ac: main/tcptls: Add checks for OpenSSL Elliptic Curve
- support The patch for ASTERISK-23905 that added PFS support in
- Asterisk depends on the elliptic curve library support being
- present in OpenSSL. As it turns out, some versions of OpenSSL
- don't have this library - notably the version running on our
- build agents. This patch fixes the build by providing a configure
- check for the specific library calls that the PFS patch relies
- on. Review: https://reviewboard.asterisk.org/r/3709/
-
-2014-07-03 16:14 +0000 [r417877-417879] sgalarneau <sgalarneau@localhost>:
-
- * res/ari/resource_events.h, rest-api/api-docs/channels.json,
- res/ari/resource_channels.h, rest-api/api-docs/events.json, /:
- ARI: Improvements to body parameters documentation The variables
- body parameter under the originate and originate with id
- operations of the channel resource showed invalid JSON in its
- description. The variables body parameter under the userEvent
- operation of the event resource made no mention that the custom
- key/value pairs should be wrapped in a variables key in order to
- be added to the custom user event. ASTERISK-23975 #close Review:
- https://reviewboard.asterisk.org/r/3692/ ........ Merged
- revisions 417878 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * rest-api-templates/api.wiki.mustache,
- rest-api-templates/swagger_model.py, /: api.wiki.mustache: Update
- wiki template to support body parameters This patch updates the
- api.wiki.mustache template and the swagger_model python script to
- understand if an operation has a body parameter. If an operation
- does have a body parameter, it will now be displayed in the
- corresponding wiki entry. ........ Merged revisions 407389 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-03 14:08 +0000 [r417863] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * Makefile, contrib/scripts/dahdi_span_config_hook (added):
- dahdi_span_config_hook: automatically register new dahdi channels
- Install a hook script for DAHDI to register new spans with
- Asterisk automatically by running: asterisk -rx 'dahdi create
- channel FIRST LAST' Review:
- https://reviewboard.asterisk.org/r/3157/
-
-2014-07-03 12:10 +0000 [r417800-417803] Matthew Jordan <mjordan@digium.com>
-
- * main/tcptls.c, CHANGES: main/tcptls: Add support for Perfect
- Forward Secrecy This patch enables Perfect Forward Secrecy (PFS)
- in Asterisk's core TLS API. Modules that wish to enable PFS
- should consider the following: - Ephemeral ECDH (ECDHE) is
- enabled by default. To disable it, do not specify a ECDHE cipher
- suite in a module's configuration, for example:
- tlscipher=AES128-SHA:DES-CBC3-SHA - Ephemeral DH (DHE) is
- disabled by default. To enable it, add DH parameters into the
- private key file, i.e., tlsprivatekey. For an example, see the
- default dh2048.pem at
- http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
- - Because clients expect the server to prefer PFS, and because
- OpenSSL sorts its cipher suites by bit strength, (see "openssl
- ciphers -v DEFAULT") consider re-ordering your cipher suites in
- the conf file. For example:
- tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
- will use PFS when offered by the client. Clients which do not
- offer PFS fall-back to AES-128 (or even 3DES as recommend by RFC
- 3261). Review: https://reviewboard.asterisk.org/r/3647/
- ASTERISK-23905 #close Reported by: Alexander Traud patches:
- tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520)
- tlsPFS.patch uploaded by Alexander Traud (License 6520)
-
- * /, main/utils.c: main/untils: Prevent potential infinite loop in
- ast_careful_fwrite A loop in ast_careful_fwrite exists that will
- continually attempt to write to a file stream, even in the
- presence of EAGAIN/EINTR errors. However, if a connection that
- uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
- call to fflush may return EAGAIN/EINTER along with EOF. A
- subsequent call to fflush will return EOF but not clear errno,
- resulting in an infinite loop. This patch clears errno after it
- is detected and handled the loop, such that any subsequent call
- to fflush will not get erroneously stuck. Review:
- https://reviewboard.asterisk.org/r/3704 #ASTERISK-23984 #close
- Reported by: Steve Davies patches: fflush_loop_fix uploaded by
- one47 (License 5012) ........ Merged revisions 417797 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417798 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417799 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-02 21:13 +0000 [r417770] Jonathan Rose <jrose@digium.com>
-
- * res/ari/resource_events.h, res/ari/resource_asterisk.h,
- res/ari/resource_applications.h, res/ari/resource_playbacks.h,
- res/ari/resource_channels.h, res/ari/resource_sounds.h, /,
- res/ari/resource_bridges.h, res/ari/resource_recordings.h,
- rest-api-templates/ari_resource.h.mustache,
- res/ari/resource_device_states.h, res/ari/resource_endpoints.h,
- res/ari/resource_mailboxes.h: ARI: Remove unnecessary \briefs
- from automatically generated documentation Review:
- https://reviewboard.asterisk.org/r/3440/ ........ Merged
- revisions 412653 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-07-01 14:42 +0000 [r417679-417706] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Don't leak memory or
- reset state if DTLS configuration is set multiple times. ........
- Merged revisions 417705 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c,
- contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py
- (added), include/asterisk/res_pjsip_session.h, main/rtp_engine.c,
- /, channels/chan_sip.c, main/sdp_srtp.c, res/res_pjsip_sdp_rtp.c,
- res/res_pjsip/pjsip_configuration.c, configs/sip.conf.sample,
- include/asterisk/rtp_engine.h, res/res_pjsip.c,
- channels/sip/include/sip.h, include/asterisk/res_pjsip.h,
- include/asterisk/sdp_srtp.h: Recorded merge of revisions 417677
- from http://svn.asterisk.org/svn/asterisk/branches/11 ........
- res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS
- negotiation on RTCP. This change fixes up DTLS support in
- res_rtp_asterisk so it can accept and provide a SHA-256
- fingerprint, so it occurs on RTCP, and so it occurs after ICE
- negotiation completes. Configuration options to chan_sip and
- chan_pjsip have also been added to allow behavior to be tweaked
- (such as forcing the AVP type media transports in SDP).
- ASTERISK-22961 #close Reported by: Jay Jideliov Review:
- https://reviewboard.asterisk.org/r/3679/ Review:
- https://reviewboard.asterisk.org/r/3686/ ........ Merged
- revisions 417678 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-30 18:39 +0000 [r417663] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c: Reverse logic during subscription
- persistence recreation. In the abstraction effort, this bit of
- logic got messed up. We want to recreate the persistence if
- things go well, not if things fail.
-
-2014-06-30 13:02 +0000 [r417590-417649] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_voicemail.c: apps/app_voicemail: Fix compilation error
- introduced in r417591 Not sure why that change to
- ast_channel_alloc was made but ... okay.
-
- * apps/app_voicemail.c, main/say.c, CHANGES: app_voicemail, say:
- Add support for Japanese Language This patch adds support for the
- Japanese language to both the say family of applications, as well
- as for VoiceMail and VoiceMailMain. A new pack of language sounds
- will be released at the same time as the next major version of
- Asterisk to support the new language features. The language
- features can be enabled using a language code of 'ja'. Review:
- https://reviewboard.asterisk.org/r/3477 ASTERISK-23324 #close
- Reported by: Kevin McCoy patches:
- app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy
- (License 6586) say.c.20140226.jb.patch uploaded by Kevin McCoy
- (License 6586)
-
- * /, channels/chan_sip.c: chan_sip: be more tolerant of whitespace
- between attributes in SDP fmtp line This patch is essentially a
- backport of a small portion of r397526 from ASTERISK-21981. In
- that patch, pass through support and format attribute negotiation
- was added for Opus. Part of that included being more tolerant to
- whitespace in the fmtp line of an SDP; that part of the patch is
- being applied here. As the author of the backport pointed out, in
- SDP, the fmtp line is allowed to include whitespace between
- attributes. RFC 3267 chapter 8.3 (from 2001) includes an example
- for this. This was not removed in the updated RFC 4867 in 2007.
- Review: https://reviewboard.asterisk.org/r/3658 #ASTERISK-23916
- #close Reported by: Alexander Traud patches:
- sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud
- (License 6520) ........ Merged revisions 417587 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417588 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417589 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-27 23:21 +0000 [r417571] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/event.c: event.c: Fix type mismatch errors in ie_maps[].
- In v12+ the type values from the table are only used by the CEL
- unit tests. Since the unit tests were only comparing a generated
- expected event with a real event to see if the ie contents
- matched and using the same table IE_PLTYPE values to read the
- event contents, the type mismatches were not detected. ........
- Merged revisions 417565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-27 19:27 +0000 [r417485-417511] Corey Farrell <git@cfware.com>
-
- * /, main/astobj2.c: Ensure REF_DEBUG records entrys for attempts
- to ao2_ref an invalid object This change ensures that
- __ao2_ref_debug writes to ref_log when given a non-NULL pointer
- to an invalid ao2 object. This is to ensure that we record any
- attempt manipulate references of already freed objects.
- ASTERISK-23948 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3677/ ........ Merged
- revisions 417500 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417505 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417509 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, contrib/scripts/refcounter.py: refcounter.py: prevent use of
- excessive RAM with large refs logs When processing a 212MB refs
- file, refcounter.py used over 3GB of RAM. This change greatly
- reduces memory usage in two ways: * Saving object history in
- whole lines instead of separated values. * Not saving
- normal/skewed/leaked object lists unless they are requested.
- ASTERISK-23921 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3668/ ........ Merged
- revisions 417480 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417481 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417483 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-27 13:50 +0000 [r417461] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c, res/res_pjsip_pubsub.c,
- res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h, /,
- res/res_pjsip_outbound_registration.c: res_pjsip: Add ActionID to
- events created as a result of PJSIP AMI actions A number of
- various PJSIP AMI actions were failing to parse out and place the
- ActionID into their responses. This patch updates the various
- PJSIP actions such that the passed in ActionID is emitted on any
- event list complete events, as well as any intermediate events
- created as a result of the action. #ASTERISK-23947 #close
- Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3675/ ........ Merged
- revisions 417460 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-27 02:04 +0000 [r417423-417447] Kinsey Moore <kmoore@digium.com>
-
- * tests/test_cel.c: CEL: Update unit tests for bridge tech field
- Update the CEL unit tests that handle BRIDGE_ENTER and
- BRIDGE_EXIT events to expect the "bridge_technology" extra field
- key.
-
- * CHANGES: CHANGES: Add missing changes Add missing CHANGES changes
- from r417361 and r417383.
-
-2014-06-26 18:27 +0000 [r417400-417421] Matthew Jordan <mjordan@digium.com>
-
- * res/res_http_websocket.exports.in, /: res_http_websocket: Export
- symbol for ast_websocket_set_timeout Thanks to Sean Bright for
- pointing out that this was missed in #asterisk-dev. ........
- Merged revisions 417419 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417420 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_pjsip.c, /: chan_pjsip: Add a test event for fast
- picture updates This will drive the test on review r3419. Note
- that the patch for this was done by Ben Ford, although it was
- slightly modified for this commit. ASTERISK-23562 Reported by:
- Matt Jordan ........ Merged revisions 417399 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-26 14:48 +0000 [r417361-417383] Kinsey Moore <kmoore@digium.com>
-
- * main/cel.c: CEL: Add bridge tech to relevant CEL records Add the
- "bridge_technology" extra field key to BRIDGE_ENTER and
- BRIDGE_EXIT CEL events to convey the bridge technology in use at
- the time the record was generated.
-
- * main/bridge.c, include/asterisk/channel.h,
- include/asterisk/bridge_features.h,
- tests/test_channel_feature_hooks.c (added),
- main/bridge_channel.c, main/channel.c: Bridging: Allow channels
- to define bridging hooks This patch allows the current owner of a
- channel to define various feature hooks to be made available once
- the channel has entered a bridge. This includes any hooks that
- are setup on the ast_bridge_features struct such as DTMF hooks,
- bridge event hooks (join, leave, etc.), and interval hooks.
- Review: https://reviewboard.asterisk.org/r/3649/
-
-2014-06-26 12:43 +0000 [r417317-417360] Matthew Jordan <mjordan@digium.com>
-
- * CHANGES, apps/app_jack.c: app_jack: Support audio with a sampling
- rate higher than 8kHz This patch enables the jack-audiohook to
- cope with dynamic sampling rates from and to Asterisk.
- Information from the channel is taken to derive the channel's
- sampling rate, suiting SLINxx format and frame->datalen. There
- are stil a few limitations after this patch: * Required
- information is taken from the channel during initialization as
- the audiohook does not provide this information.
- Audiohook.internal_sampl_rate(...) is set later, but no callback
- is available to inform app_jack. * Frame.datalen is computed
- using "rate / 50" assuming a ptime of 20ms. There is no internal
- API available to determine datalen for a SLINxx. * Ringbuffer
- size is now dynamic depending on the value of frame.datalen (see
- above) and the number of frames, which are in
- RINGBUFFER_FRAME_CAPACITY, that need to fit. Review:
- https://reviewboard.asterisk.org/r/3618 Note that the patch being
- committed here is based on the patch posted on ASTERISK-23836.
- However, Matthis Schmieder also provided a patch to enable this
- functionality, and that patch is noted below. ASTERISK-20696
- #close Reported by: Matthis Schmieder patches: app_jack.patch
- uploaded by Matthis Schmieder (License 6445) ASTERISK-23836
- #close Reported by: Dennis Guse patches: patch-app_jack.c
- uploaded by Dennis Guse (License 6513)
-
- * main/udptl.c, /: udptl: Correct FEC to not consider negative
- sequence numbers as missing When using FEC, with span=3 and
- entries=4 Asterisk will attempt to repair the packet with
- sequence number 5, as it will see that packet -4 is missing. The
- result is Asterisk sending garbage packets that can kill a fax.
- This patch adds a check to see if the sequence number is valid
- before checking if the packet is missing. Review:
- https://reviewboard.asterisk.org/r/3657/ #ASTERISK-23908 #close
- Reported by: Torrey Searle patches: udptl_fec.patch uploaded by
- Torrey Searle (License 5334) ........ Merged revisions 417318
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 417320 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417324 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/internal.h, configs/ari.conf.sample,
- res/res_http_websocket.c, res/res_pjsip.c,
- configs/pjsip.conf.sample, include/asterisk/http_websocket.h,
- configs/sip.conf.sample, res/res_pjsip/config_transport.c,
- res/ari/ari_websockets.c, res/res_pjsip_transport_websocket.c,
- res/ari/config.c, channels/sip/include/sip.h,
- include/asterisk/res_pjsip.h, res/res_ari.c, /,
- channels/chan_sip.c, UPGRADE.txt: res_http_websocket: Close
- websocket correctly and use careful fwrite When a client takes a
- long time to process information received from Asterisk, a write
- operation using fwrite may fail to write all information. This
- causes the underlying file stream to be in an unknown state, such
- that the socket must be disconnected. Unfortunately, there are
- two problems with this in Asterisk's existing websocket code: 1.
- Periodically, during the read loop, Asterisk must write to the
- connected websocket to respond to pings. As such, Asterisk
- maintains a reference to the session during the loop. When
- ast_http_websocket_write fails, it may cause the session to
- decrement its ref count, but this in and of itself does not break
- the read loop. The read loop's write, on the other hand, does not
- break the loop if it fails. This causes the socket to get in a
- 'stuck' state, preventing the client from reconnecting to the
- server. 2. More importantly, however, is that the fwrite in
- ast_http_websocket_write fails with a large volume of data when
- the client takes awhile to process the information. When it does
- fail, it fails writing only a portion of the bytes. With some
- debugging, it was shown that this was failing in a similar
- fashion to ASTERISK-12767. Switching this over to
- ast_careful_fwrite with a long enough timeout solved the problem.
- Note that this version of the patch, unlike r417310 in Asterisk
- 11, exposes configuration options beyond just chan_sip's
- sip.conf. Configuration options to configure the write timeout
- have also been added to pjsip.conf and ari.conf. #ASTERISK-23917
- #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3624/ ........ Merged
- revisions 417310 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417311 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-26 10:06 +0000 [r417251] Corey Farrell <git@cfware.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix handling of "From" headers
- longer than 256 characters From headers were processed using a
- 256 character buffer on the stack. This change replaces that with
- a heap allocation by ast_strdup. ASTERISK-23790 #close Reported
- by: uniken1 Tested by: uniken1 Review:
- https://reviewboard.asterisk.org/r/3669/ Patches:
- chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes
- (license 5674) ........ Merged revisions 417248 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 417249 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 417250 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-25 20:57 +0000 [r417233] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c, res/res_pjsip_exten_state.c,
- include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_pidf_body_generator.c,
- res/res_pjsip_pubsub.exports.in, res/res_pjsip_mwi.c,
- res/res_pjsip_xpidf_body_generator.c: Abstract PJSIP-specific
- elements from the pubsub API. This helps to pave the way for RLS
- work that is to come. Since this is a self-contained change and
- subscription tests still pass, this work is being committed
- directly to trunk instead of a working branch. ASTERISK-23865
- #close Review: https://reviewboard.asterisk.org/r/3628
-
-2014-06-25 18:57 +0000 [r417213] Corey Farrell <git@cfware.com>
-
- * main/astobj2_container.c, /: ao2_container node object ignores
- REF_DEBUG in all places except one Almost every reference
- operation against container node's uses __ao2_alloc or __ao2_ref,
- thereby preventing ref logging for the nodes. One node reference
- is released with ao2_t_ref, causing refcounter.py to falsely
- report skews and leaks for many nodes. ASTERISK-23922 #close
- Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3670/ ........ Merged
- revisions 417212 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-25 00:45 +0000 [r417193] Damien Wedhorn <voip@facts.com.au>
-
- * channels/chan_skinny.c: Skinny: cleanup some log messages around
- sessions.
-
-2014-06-24 02:50 +0000 [r417167] Corey Farrell <git@cfware.com>
-
- * include/asterisk/netsock.h, main/utils.c, main/netsock.c,
- include/asterisk/res_pjsip_session.h: Move eid functions to
- utils.c, mark netsock.h deprecated Move eid functions from
- netsock.c to utils.c. These functions were already published by
- utils.h. Flag netsock.h as deprecated and switch
- res_pjsip_session.h to use netsock2.h. The only code that still
- uses netsock.h is chan_iax2. ASTERISK-23920 #close Reported by:
- Corey Farrell Review: https://reviewboard.asterisk.org/r/3661/
-
-2014-06-23 18:50 +0000 [r417143] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of
- data written when sending via ICE instead of 0. ASTERISK-23834
- #close Reported by: Richard Kenner ........ Merged revisions
- 417141 from http://svn.asterisk.org/svn/asterisk/branches/11
- ........ Merged revisions 417142 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-23 16:04 +0000 [r417120] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/core_unreal.c: core_unreal: Fix off by one buffer
- overwrite error. Appending the ;2 to the user supplied ;1
- uniqueid to create the ;2 version if the user did not also supply
- an extra uniqueid for the ;2 channel resulted in allocating a
- buffer that was one byte too small. * Fix off by one error in
- ast_unreal_new_channels() when generating the ;2 uniqueid from
- the user suppled ;1 version. * Pulled some long assignment lines
- from if tests to improve line break readability in
- ast_unreal_new_channels(). ........ Merged revisions 417119 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-23 07:44 +0000 [r417059] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c:
- suspended destructions of pri spans on events If a DAHDI span
- disappears, we wish for its representation in Asterisk to be
- destroyed as well. The information about the span's removal may
- come from several paths: 1. DAHDI sends DAHDI_EVENT_REMOVE on
- every channel. 2. An extra DAHDI_EVENT_REMOVED is sent on every
- subsequent call to DAHDI_GET_EVENT. 3. Every read (including the
- internal one by libpri on the D-channel) returns -ENODEV.
- Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by
- destroying it. Destroying a channel requires holding the channel
- list lock (iflock). Destroying a channel that is part of a span
- requires holding the span's lock. Destroying a channel from a
- context that holds the span lock, while at the same time another
- channel is destroyed directly, leads to a deadlock. Solution:
- don't destroy span while holding the channels list lock. Thus
- changes in this patch: * Deferring removal of PRI spans in
- response to events: doomed spans are collected on a list. *
- Doomed spans are removed periodically by the monitor thread. *
- ENODEV reads from the D-channel will warant the same deferred
- removal. Review: https://reviewboard.asterisk.org/r/3548/
-
-2014-06-22 18:53 +0000 [r416996] George Joseph <george.joseph@fairview5.com>
-
- * include/asterisk/astobj2.h, Makefile.rules, Makefile, /: astobj2:
- Add an ao2_replace macro to astobj2.h This macro replaces one
- object reference with another cleaning up the original. param dst
- Pointer to the object that will be cleaned up. param src Pointer
- to the object replacing it. src's ref count is bumped if it's
- non-NULL. dst's ref count is decremented if it's non-NULL. src is
- assigned to dst, This patch was reviewed on IRC by coreyfarrell
- and mjordan. Tested by: George Joseph ........ Merged revisions
- 416995 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-20 23:18 +0000 [r416872-416935] George Joseph <george.joseph@fairview5.com>
-
- * /, configure, include/asterisk/autoconfig.h.in: build: Allow
- autoconf/ast_ext_tool_check to handle cross-compiling better.
- ast_ext_tool_check.m4 isn't handling cases where a path to a
- package is provided (E.G. --with-mysqlclient=/some/sysroot) and
- the package has a config tool (E.G. mysql_config) and the package
- has its own subdirectories in include or lib. For example,
- mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
- ast_ext_tool_check sets MYSQLCLIENT_LIB to
- ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
- includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
- directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
- fail and there are others in the same boat. The problem is caused
- by logic in ast_ext_tool_check that overrides the result of the
- config tool's --cflags and --libs options if package_DIR is set.
- This patch prepends package_DIR (if specified) to the -L and -I
- results from the package's config tool instead of overriding
- them. A regenerated ./configure and
- include/asterisk/autoconfig.h.in are included but can be
- regenerated by running ./bootstrap.sh at any time. Tested by:
- George Joseph Tested by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3550/ ........ Merged
- revisions 416929 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416930 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416931 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * autoconf/ast_ext_tool_check.m4, /: build: Allow
- autoconf/ast_ext_tool_check to handle cross-compiling better.
- ast_ext_tool_check.m4 isn't handling cases where a path to a
- package is provided (E.G. --with-mysqlclient=/some/sysroot) and
- the package has a config tool (E.G. mysql_config) and the package
- has its own subdirectories in include or lib. For example,
- mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
- ast_ext_tool_check sets MYSQLCLIENT_LIB to
- ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
- includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
- directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
- fail and there are others in the same boat. The problem is caused
- by logic in ast_ext_tool_check that overrides the result of the
- config tool's --cflags and --libs options if package_DIR is set.
- This patch prepends package_DIR (if specified) to the -L and -I
- results from the package's config tool instead of overriding
- them. Tested by: George Joseph Tested by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3550/ ........ Merged
- revisions 416870 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416871 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-20 20:57 +0000 [r416848-416850] Jonathan Rose <jrose@digium.com>
-
- * res/parking/parking_manager.c, /: res_parking: Make manager
- commands register with module information Previously module
- information was not included due to an oversight. Review:
- https://reviewboard.asterisk.org/r/3626/ ........ Merged
- revisions 416849 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/logger.c, CHANGES, include/asterisk/logger.h,
- main/manager.c: Logger: Add manager command 'LoggerRotate' to
- rotate logger Part of a series of AMI command equivalents to
- existing CLI commands Review:
- https://reviewboard.asterisk.org/r/3651/
-
-2014-06-20 17:06 +0000 [r416830] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_voicemail.c, include/asterisk/app.h, main/app.c,
- apps/app_directory.c, apps/app_chanspy.c: voicemail API
- callbacks: Extract the sayname API call to its own registerd
- callback. * Extract the sayname API call to its own registerd
- callback. This allows the app_directory and app_chanspy
- applications to say a mailbox owner's name using an alternate
- provider when app_voicemail is not available because you are
- using res_mwi_external. app_directory still uses the
- voicemail.conf file. AFS-64 #close Reported by: Mark Michelson
-
-2014-06-20 15:27 +0000 [r416738-416807] George Joseph <george.joseph@fairview5.com>
-
- * main/astobj2_private.h, main/astobj2_container_private.h,
- main/astobj2_container.c, main/astobj2_hash.c,
- main/astobj2_rbtree.c, build_tools/cflags.xml, /,
- tests/test_astobj2.c: astobj2: Additional refactoring to push
- impl specific code down into the impls. Move some implementation
- specific code from astobj2_container.c into astobj2_hash.c and
- astobj2_rbtree.c. This completely removes the need for
- astobj2_container to switch on RTTI and it no longer has any
- knowledge of the implementation details. Also adds AO2_DEBUG as a
- new compile option in menuselect which controls astobj2 debugging
- independently of AST_DEVMODE and REF_DEBUG. Tested by: George
- Joseph Review: https://reviewboard.asterisk.org/r/3593/ ........
- Merged revisions 416806 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_endpoint_identifier_ip.c, main/acl.c,
- include/asterisk/netsock2.h, include/asterisk/acl.h,
- main/netsock2.c: pjsip cli: Change Identify to show CIDR notation
- instead of netmasks. * Added ast_sockaddr_cidr_bits() to count
- the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which
- uses ast_sockaddr_cidr_bits() for the netmask instead of
- ast_sockaddr_stringify_addr. * Changed
- res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr()
- instead of ast_ha_join() for the CLI output. This is a CLI change
- only. AMI was not affected. Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3652/ ........ Merged
- revisions 416737 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-19 19:40 +0000 [r416736] Kinsey Moore <kmoore@digium.com>
-
- * /, main/bridge.c, res/parking/parking_tests.c,
- channels/sip/reqresp_parser.c, main/logger.c, main/test.c: Fix
- build warnings with TEST_FRAMEWORK enabled ........ Merged
- revisions 416732 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416733 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416734 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-19 16:04 +0000 [r416589-416670] George Joseph <george.joseph@fairview5.com>
-
- * pbx/pbx_lua.c, /: Remove the problematic and unneeded
- AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
- AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be
- incorrectly loaded before pbx_config. pbx_config was therefore
- blowing away contexts that were created by pbx_lua. With
- AST_MODFLAG_DEFAULT the load order is now correct and contexs are
- being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed
- anyway since no other modules needed its global symbols that
- early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by:
- Dennis Guse Tested by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3629/ ........ Merged
- revisions 416668 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416669 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * configs/extensions.lua.sample, /: Update extensions.lua.sample
- with naming conflict guidance. The sample extensions.lua was
- causing pbx_lua to fail to load when parsing 'app.goto("default",
- "s", 1)' because in Lua 5.2, 'goto' is now a reserved word. This
- patch adds guidance to extensions.lua.sample and changed
- 'app.goto("default", "s", 1)' to 'app.['goto']("default", "s",
- 1)'. ASTERISK-23844 #close Reported by: rnewton Tested by:
- gtjoseph Review: https://reviewboard.asterisk.org/r/3627/
- ........ Merged revisions 416581 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416582 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-18 04:22 +0000 [r416561] Matthew Jordan <mjordan@digium.com>
-
- * /, main/stasis_channels.c: stasis_channels: Update the stasis
- cache if manager variables are needed In r416211, the publishing
- of variable changes was modified such that a cached channel
- snapshot was used if manager variables were not requested with
- each AMI event. This was done to reduce the amount of channel
- snapshots created. However, an assumption was made that
- generating a channel snapshot and publishing the snapshot to the
- channel topic was sufficient to ensure that the cache would be
- updated; this is not the case. The channel snapshot type must be
- used to force a snapshot update. This patch updates the
- publication of channel variables such that the cache is updated
- prior to publication of the channel variable message if manager
- variables are in use. This ensures that all AMI events receive
- the variable update when they are supposed to. Note that this
- issue was caught by the Asterisk Test Suite (go go testing)
- ........ Merged revisions 416557 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-17 18:45 +0000 [r416444-416503] Mark Michelson <mmichelson@digium.com>
-
- * /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to
- set inheritable channel variables. ........ Merged revisions
- 416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 416501 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416502 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pidf_body_generator.c, /,
- res/res_pjsip_xpidf_body_generator.c: Fix string growth algorithm
- for XML presence bodies. pjpidf_print() does not return < 0 if
- there is not enough room for the document to be printed. Rather,
- it returns 39, the length of the XML prolog. The algorithm also
- had a bug in that it would return if it attempted to grow the
- string larger. ........ Merged revisions 416442 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-17 16:33 +0000 [r416443] Kinsey Moore <kmoore@digium.com>
-
- * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated
- start calls Currently, music on hold will stop and then start
- again from the beginning if ast_moh_start() is called multiple
- times. This can happen if a call is put on hold repeatedly (the
- channel receives multiple HOLD control frames) and can be
- triggered from ARI by starting MoH on a channel multiple times.
- This is fairly jarring/annoying to users. This change prevents
- MoH from being restarted if the requested music class is the same
- as the one currently playing. This includes an extra check to
- prevent the errors previously experienced in the testsuite and
- has 100+ test runs behind it. Review:
- https://reviewboard.asterisk.org/r/3615/ ........ Merged
- revisions 416439 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416440 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416441 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-16 18:27 +0000 [r416416] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
- channels/sig_ss7.h, configure, channels/chan_dahdi.h,
- configure.ac, UPGRADE.txt, configs/ss7.timers.sample (added),
- CHANGES, channels/sig_ss7.c: chan_dahdi: Adds support for major
- update to libss7. * SS7 support now requires libss7 v2.0 or
- later. The new libss7 is not backwards compatible. * Added SS7
- support for connected line and redirecting. * Most SS7 CLI
- commands are reworked as well as new SS7 commands added. See
- online CLI help. * Added several SS7 config option parameters
- described in chan_dahdi.conf.sample. * ISUP timer support
- reworked and now requires explicit configuration. See
- ss7.timers.sample. Special thanks to Kaloyan Kovachev for his
- support and persistence in getting the original patch by adomjan
- updated and ready for release. SS7-27 #close Reported by: adomjan
-
-2014-06-16 16:22 +0000 [r416394] Kevin Harwell <kharwell@digium.com>
-
- * include/asterisk/http_websocket.h, tests/test_websocket_client.c,
- res/res_http_websocket.c: res_http_websocket: read/write string
- fixup There was a problem when reading a string from the
- websocket. It assumed the received data had a null terminator and
- tried to write the data to an ast_str. This of course could/would
- read past the end of the given buffer while writing the data to
- the internal buffer of ast_str. Modified the the code to
- correctly place a null terminator on the result string.
-
-2014-06-16 09:04 +0000 [r416339] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c,
- cdr/cdr_sqlite3_custom.c, /: We have faced situation when using
- CDR and CEL by sqlite3 modules. With system having high load
- (~100 concurrent calls created by sipp) we found many cdr and cel
- records missed. There is special finction in sqlite3, that make
- able to fix this situation - sqlite3_wait_timeout, that also can
- replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this
- function can be used for aastdb and res_config_sqlite3 to avoid
- missed writes to sqlite db. #ASTERISK-23766 #close Reported by:
- Igor Goncharovsky Review:
- https://reviewboard.asterisk.org/r/3559/ ........ Merged
- revisions 416336 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416337 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416338 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-16 02:40 +0000 [r416267-416319] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: channels/chan_sip: Forbid remote bridging
- if T.38 is negotiated When a framehook is removed - such as the
- fax gateway framehook - the bridge framework will re-evaluate the
- bridge mixing technologies to see if it can improve the bridging.
- When this occurs, get_rtp_info will be called to determine if
- local or remote bridging can be used. Using remote bridging will
- cause a fax to fail, as direct media negotiation will cause some
- small number of packets to not arrive at the remote endpoint.
- This patch forces local native bridging if T.38 negotiation is in
- progress or has been established. ........ Merged revisions
- 416318 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/channel_internal_api.c: channel_internal_api: Publish a
- snapshot change when linkedids change Snapshots are now not
- published *quite* as much as they used to. One instance where
- they are not published any longer is during bridge enter and exit
- - the state of the channel doesn't change, the bridge does.
- However, channels are changed when a linkedid is propagated;
- previously, the channel's state would be updated and published
- during the bridge enter event. Now this must be explicitly done.
- ........ Merged revisions 416300 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, tests/test_stasis_endpoints.c: test_stasis_endpoints: Remove
- expected channel snapshot We no longer publish a channel snapshot
- when it is associated with an endpoint; after all, the channel
- itself hasn't changed - the endpoint state has changed. This
- updates the channel_messages unit test accordingly. ........
- Merged revisions 416298 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This
- patch reverts r416150. When the comparison between mohclass->name
- and state->class->name is made, you are not guaranteed that (a)
- state->class is non-NULL or that state or state->class are in a
- safe state. Crashes caught by the bridges/transfer_capabilities
- test. ........ Merged revisions 416251 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416252 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416255 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-14 19:26 +0000 [r416237] Corey Farrell <git@cfware.com>
-
- * res/res_manager_devicestate.c, res/res_manager_presencestate.c:
- res_manager_devicestate and res_manager_presencestate missing
- support level Add MODULEINFO comment block to define support
- level core for these new modules. Review:
- https://reviewboard.asterisk.org/r/3620/
-
-2014-06-13 18:24 +0000 [r416216] Matthew Jordan <mjordan@digium.com>
-
- * res/res_agi.c, res/res_pjsip/pjsip_configuration.c,
- main/stasis_channels.c, res/ari/resource_channels.c,
- main/bridge_channel.c, main/pbx.c, main/stasis_cache.c, /,
- apps/app_meetme.c, main/pickup.c, main/channel_internal_api.c,
- include/asterisk/channel.h, main/core_local.c, main/aoc.c,
- main/endpoints.c, main/cel.c, apps/app_queue.c,
- main/stasis_bridges.c, apps/app_agent_pool.c, main/cli.c,
- main/channel.c, main/dial.c, main/manager.c,
- include/asterisk/stasis_channels.h: stasis: Reduce creation of
- channel snapshots to improve performance During some performance
- testing of Asterisk with AGI, ARI, and lots of Local channels, we
- noticed that there's quite a hit in performance during channel
- creation and releasing to the dialplan (ARI continue). After
- investigating the performance spike that occurs during channel
- creation, we discovered that we create a lot of channel snapshots
- that are technically unnecessary. This includes creating
- snapshots during: * AGI execution * Returning objects for ARI
- commands * During some Local channel operations * During some
- dialling operations * During variable setting * During some
- bridging operations And more. This patch does the following: - It
- removes a number of fields from channel snapshots. These fields
- were rarely used, were expensive to have on the snapshot, and
- hurt performance. This included formats, translation paths, Log
- Call ID, callgroup, pickup group, and all channel variables. As a
- result, AMI Status, "core show channel", "core show channelvar",
- and "pjsip show channel" were modified to either hit the live
- channel or not show certain pieces of data. While this is
- unfortunate, the performance gain from this patch is worth the
- loss in behaviour. - It adds a mechanism to publish a cached
- snapshot + blob. A large number of publications were changed to
- use this, including: - During Dial begin - During Variable
- assignment (if no AMI variables are emitted - if AMI variables
- are set, we have to make snapshots when a variable is changed) -
- During channel pickup - When a channel is put on hold/unhold -
- When a DTMF digit is begun/ended - When creating a bridge
- snapshot - When an AOC event is raised - During Local channel
- optimization/Local bridging - When endpoint snapshots are
- generated - All AGI events - All ARI responses that return a
- channel - Events in the AgentPool, MeetMe, and some in Queue -
- Additionally, some extraneous channel snapshots were being made
- that were unnecessary. These were removed. - The result of
- ast_hashtab_hash_string is now cached in stasis_cache. This
- reduces a large number of calls to ast_hashtab_hash_string, which
- reduced the amount of time spent in this function in gprof by
- around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged
- revisions 416211 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-13 13:11 +0000 [r416149-416153] Kinsey Moore <kmoore@digium.com>
-
- * res/res_musiconhold.c, /: MoH: Don't restart stream on repeated
- start calls Currently, music on hold will stop and then start
- again from the beginning if ast_moh_start() is called multiple
- times. This can happen if a call is put on hold repeatedly (the
- channel receives multiple HOLD control frames) and can be
- triggered from ARI by starting MoH on a channel multiple times.
- This is fairly jarring/annoying to users. This change prevents
- MoH from being restarted if the requested music class is the same
- as the one currently playing. Review:
- https://reviewboard.asterisk.org/r/3615/ ........ Merged
- revisions 416150 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416151 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416152 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cel.c, /: CEL: Expose parking retreiver in extra field This
- exposes the retreiver of a parked call under the "retreiver" key
- of the extra field when this information is available. Review:
- https://reviewboard.asterisk.org/r/3608/ ........ Merged
- revisions 416148 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-13 05:16 +0000 [r416071] Richard Mudgett <rmudgett@digium.com>
-
- * main/http.c, include/asterisk/tcptls.h, main/tcptls.c,
- main/manager.c, /, channels/chan_sip.c: AST-2014-007: Fix of fix
- to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close
- Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/3617/ ........ Merged
- revisions 416066 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 416067 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 416070 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 21:27 +0000 [r416024] Rusty Newton <rnewton@digium.com>
-
- * main/pbx.c: main/pbx - documentation - enhance 'core show hints'
- and 'core show hint' help text Adds descriptive help text to
- 'core show hints' and 'core show hint'. The text describes the
- various columns for the sake of clarity. It takes into account
- recent changes to the content displayed by the commands
- https://reviewboard.asterisk.org/r/3604/ and
- https://reviewboard.asterisk.org/r/3611/. ASTERISK-23764 Review:
- https://reviewboard.asterisk.org/r/3610/
-
-2014-06-12 20:17 +0000 [r415982] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip_pubsub.c, /: Fix build in devmode for GCC 4.10
- ........ Merged revisions 415980 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 17:00 +0000 [r415907] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/utils.h, main/tcptls.c, main/manager.c, /,
- channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c,
- include/asterisk/tcptls.h, res/res_http_websocket.c,
- configs/http.conf.sample: AST-2014-007: Fix DOS by consuming the
- number of allowed HTTP connections. Simply establishing a TCP
- connection and never sending anything to the configured HTTP port
- in http.conf will tie up a HTTP connection. Since there is a
- maximum number of open HTTP sessions allowed at a time you can
- block legitimate connections. A similar problem exists if a HTTP
- request is started but never finished. * Added http.conf
- session_inactivity timer option to close HTTP connections that
- aren't doing anything. Defaults to 30000 ms. * Removed the
- undocumented manager.conf block-sockets option. It interferes
- with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections
- now have better authentication timeout protection. Though I
- didn't remove the bizzare TLS timeout polling code from chan_sip.
- * chan_sip can now handle SSL certificate renegotiations in the
- middle of a session. It couldn't do that before because the
- socket was non-blocking and the SSL calls were not restarted as
- documented by the OpenSSL documentation. * Fixed an off nominal
- leak of the ssl struct in handle_tcptls_connection() if the FILE
- stream failed to open and the SSL certificate negotiations
- failed. The patch creates a custom FILE stream handler to give
- the created FILE streams inactivity timeout and timeout after a
- specific moment in time capability. This approach eliminates the
- need for code using the FILE stream to be redesigned to deal with
- the timeouts. This patch indirectly fixes most of ASTERISK-18345
- by fixing the usage of the SSL_read/SSL_write operations.
- ASTERISK-23673 #close Reported by: Richard Mudgett ........
- Merged revisions 415841 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415854 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415896 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 15:50 +0000 [r415839] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, apps/app_queue.c: app_queue: delayed state can cause early
- leavewhenempty ringing In app_queue, device state changes arrive
- in event messages and update the queue member status value. That
- value is checked in get_member_status() to decide that the caller
- should leave when there are no available members. Although event
- messages can be delayed by other activity, there is no adverse
- affect by lagged status except in one specific case: there is
- only one available member, it was just rung, and leavewhenempty
- is enabled set for ringing members. This change adds a direct
- check of the device state only under this condition where the
- caller may be dropped incorrectly, resolving this issue without
- affecting performance of app_queue normally. AST-1248 #close
- Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
- Thomas Arimont ........ Merged revisions 415833 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415835 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415836 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 15:39 +0000 [r415834] Jonathan Rose <jrose@digium.com>
-
- * apps/app_mixmonitor.c, /, UPGRADE.txt: MixMontior: Add class
- authorization requirements to MixMonitor AMI commands MixMonitor
- AMI commands StartMixMonitor and StopMixMonitor lacked class
- authorization. StopMixMonitor now requires that the manager user
- either have the call or system class authorization.
- StartMixMonitor is a slightly larger issue since it can execute
- shell commands if the right arguments are passed into it, and we
- consider this a permission escalation. A security release will be
- issued for problem this shortly. ASTERISK-23609 #close Reported
- by: Corey Farrell ........ Merged revisions 415825 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415832 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 14:39 +0000 [r415813] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: unauthenticated
- remote crash in PJSIP pub/sub framework A remotely exploitable
- crash vulnerability exists in the PJSIP channel driver's pub/sub
- framework. If an attempt is made to unsubscribe when not
- currently subscribed and the endpoint's "sub_min_expiry" is set
- to zero, Asterisk tries to create an expiration timer with zero
- seconds, which is not allowed, so an assertion raised. The fix
- was to reject a subscription that is attempting to unsubscribe
- when not being already subscribed. Asterisk now checks for this
- situation appropriately and responds with a 400 instead of
- crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged
- revisions 415812 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 14:15 +0000 [r415795] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip.c, /: Fix potential deadlock situation in
- res_pjsip. SIP transaction timeouts are handled in the PJSIP
- monitor thread. When this happens on a subscription, and the
- subscription is destroyed, the subscription destruction is
- dispatched synchronously to the threadpool. The issue is that the
- PJSIP dialog is locked by the monitor thread, and then the
- dispatched task attempts to lock the dialog. This leads to a
- deadlock that causes SIP traffic to no longer be accepted on the
- Asterisk server. The fix here is to treat the monitor thread as
- if it were a threadpool thread when it attempts to dispatch
- synchronous tasks. This way, the dispatched task turns into a
- simple function call within the same thread, and the locking
- issue is averted. AST-2014-008 ASTERISK-23802 #close ........
- Merged revisions 415794 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 11:34 +0000 [r415767] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_pubsub.c,
- res/res_pjsip_exten_state.c, include/asterisk/res_pjsip.h,
- include/asterisk/res_pjsip_pubsub.h,
- res/res_pjsip_pubsub.exports.in, /,
- contrib/ast-db-manage/config/versions/c6d929b23a8_create_pjsip_subscription_persistence_.py
- (added), res/res_pjsip_mwi.c: res_pjsip_pubsub: Persist
- subscriptions in sorcery so they are recreated on startup. This
- change makes res_pjsip_pubsub persist inbound subscriptions in
- sorcery. By default this uses the local astdb but it can also be
- configured to store within an outside database. When Asterisk is
- started these subscriptions are recreated if they have not
- expired. Notifications are sent to the devices which have
- subscribed and they are none the wiser that the system has
- restarted. Review: https://reviewboard.asterisk.org/r/3598/
- ........ Merged revisions 415766 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-12 07:52 +0000 [r415749] Walter Doekes <walter+asterisk@wjd.nu>
-
- * UPGRADE.txt, contrib/scripts/safe_asterisk, Makefile, /:
- safe_asterisk: Overwrite old safe_asterisk on make install. From
- now on, make install will overwrite safe_asterisk with the latest
- version. You need to move any local modifications to files inside
- /etc/asterisk/startup.d, if you have any. See also commits
- r394939 and r397938. ASTERISK-21965 #close Patches:
- safe_asterisk.patch uploaded by jkister (License 6232, modified
- by me) ........ Merged revisions 415748 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-11 23:01 +0000 [r415730] Richard Mudgett <rmudgett@digium.com>
-
- * main/format.c, /: format.c: Fix misuse of hash container
- function. The supplied hash function to a container must be
- idempotent given the object's key value to figure out which
- container bucket the object belongs in. Returning a random number
- or the current container count is not idempotent. The "computed
- hash" value doesn't help find the object later in those cases. *
- Fixed the format_list container to actually be a list since that
- is how the container is used. Conceptually, if more than 283
- formats were added to the format_list then odd things may have
- happened before the fix. ........ Merged revisions 415728 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415729 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-11 20:22 +0000 [r415698-415715] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/pbx.c: CLI: correct presence information on core show hints
- Adds presence to core show hint and changes presence string
- conversion to use the correct function. ASTERISK-23858 #close
- Review: https://reviewboard.asterisk.org/r/3611/
-
- * main/pbx.c: CLI: add presence information to core show hints Adds
- presence state value to output of core show hints. Also reformats
- the output slightly so it doesn't use as much space as it would
- otherwise. Was: 1000@demo : SIP/1000 State:Unavailable Watchers 0
- Now: 1000@demo : SIP/1000 State:Unavailable Presence:Idle
- Watchers 0 AFS-53 #close Review:
- https://reviewboard.asterisk.org/r/3604/
-
-2014-06-10 18:32 +0000 [r415679] Kinsey Moore <kmoore@digium.com>
-
- * main/channel.c, /: Fix build in dev mode due to signed/unsigned
- mismatch ........ Merged revisions 415678 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-10 16:06 +0000 [r415659] Jonathan Rose <jrose@digium.com>
-
- * main/message.c, /, res/res_pjsip_notify.c: PJSIP: PJSIPNotify -
- Strip content-length headers and add documentation Documentation
- for how to add custom headers/content to notifies created with
- the PJSIPNotify manager action was a little sparse and it also
- wasn't vetting application of Content-length headers like its
- chan_sip equivalent was (so two Content-length headers could be
- applied... and PJSIP determines the content length anyway, so it
- just opens people up for error). This patch also flips the
- variable order so that the variables are interpreted in the same
- order as they are put in the AMI action. Review:
- https://reviewboard.asterisk.org/r/3587/ ........ Merged
- revisions 415658 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-10 09:28 +0000 [r415630] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c, /: chan_ooh323: fix loading module failure
- if there no accessible h323_log or ooh323 config file change
- return 1 to return AST_MODULE_LOAD_FAILURE on module load routine
- few cosmetic changes ASTERISK-23814 #close (closes issue
- ASTERISK-23814) Reported by: Igor Goncharovsky Patches:
- ASTERISK-23814-ast11.patch ........ Merged revisions 415599 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415602 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-09 20:21 +0000 [r415580] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_header_funcs.c, /: chan_pjsip: Fix bug where custom
- SIP headers could be duplicated on outgoing INVITEs. When using
- PJSIP_HEADER() to add custom headers to outgoing INVITE requests,
- certain situations could result in the headers being duplicated.
- For instance, if the request were retransmitted, or if the INVITE
- were re-sent with authentication credentials, the custom headers
- would be re-added to the request. The fix here is to, after
- adding the custom headers to the outbound INVITE, remove the
- datastore that holds the custom headers to add. This way, there
- is no risk in accidentally adding them if the session supplement
- is called into a second or third time. ........ Merged revisions
- 415579 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-09 12:12 +0000 [r415524] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, UPGRADE.txt, contrib/scripts/safe_asterisk: safe_asterisk:
- Cleanup additions to r415132. * Replaced a stray echo that
- should've been a message call in safe_asterisk. This replaces a
- conditional log message by a slightly different message. Please
- update your log parsing scripts. * Made the $NOTIFY mail Subject
- more verbose by adding the machine name and exitstatus. (Note
- that a 'make install' still won't overwrite your old
- safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492
- #close ........ Merged revisions 415521 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415522 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415523 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-09 03:50 +0000 [r415466] Corey Farrell <git@cfware.com>
-
- * /, main/autoservice.c: autoservice: stop thread on graceful
- shutdown This change adds thread shutdown to autoservice for
- graceful shutdowns only. ast_register_cleanup is backported to
- 1.8 to allow this. The logger callid is also released on shutdown
- in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3594/ ........ Merged
- revisions 415463 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415464 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415465 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-08 18:12 +0000 [r415444] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- main/bridge_channel.c, main/channel.c, main/pbx.c, /,
- main/framehook.c, main/bridge_after.c: bridges/bridge_native_rtp:
- Reconfigure bridge on removal of framehook This patch is a re-do
- of r414122. When r414122 was merged, a major problem with it was
- uncovered. UNBRIDGE soft hangup flags have a catastrophic effect
- on the pbx core if they leak out from the bridge layer: the
- channel gets hung up. With the number of threads involved in a
- blind transfer, and with the initial patch, it was likely that
- this would occur. This caused a large number of test failures
- This patch is nearly identical with the one proposed in r414122,
- save for the following changes: - We explicitly clear the
- UNBRIDGE flag when setting an after goto on a channel in a bridge
- - Defensively, if we encounter an UNBRIDGE flag in the pbx core,
- we handle it https://reviewboard.asterisk.org/r/3585/ ........
- Merged revisions 415443 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-07 00:42 +0000 [r415428] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/bridge.h, /: bridge.h: Remove redundant struct
- ast_bridge_channel forward declaration. ........ Merged revisions
- 415427 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-06 21:44 +0000 [r415411] Jonathan Rose <jrose@digium.com>
-
- * include/asterisk/manager.h, main/config.c, main/manager.c, /,
- channels/chan_sip.c, include/asterisk/config.h: chan_sip: Fix
- order of variables specified in SIPNotify action Prior to this
- patch, sequential variables would be ordered in reverse from the
- order specified in the manager action. Review:
- https://reviewboard.asterisk.org/r/3588/ ........ Merged
- revisions 415359 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415390 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415410 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-06 20:45 +0000 [r415358] Kevin Harwell <kharwell@digium.com>
-
- * main/uri.c, tests/test_websocket_client.c: core uri: Custom uri
- parsing error when no query parameters If using the custom URI
- parsing code (not external uriparser lib) and there was no query
- parameters the resulting pointer would be NULL and then an
- attempt was made to subtract from it. The pointer is now set to a
- valid value if there is no query parameter(s). Also, in the
- 'ast_uri_make_host_with_port' function when setting the
- terminator on the resulting string it was writing it one past the
- end of allocated memory. It now writes the string terminator
- appropriately.
-
-2014-06-06 19:13 +0000 [r415343] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip_sdp_rtp.c: PJSIP: Remove premature write of raw
- formats Currently, there are situations that can occur when using
- chan_pjsip and certain dialplan applications (notably ChanSpy())
- that can cause the channel to get no audio with scrolling
- warnings about format mismatches. This is caused by a failure to
- update translation paths on a mid-call native format update since
- the raw formats have already been updated by res_pjsip_sdp_rtp.c
- in set_caps(). Removing the premature raw format updates allows
- the translation paths to be setup correctly and the raw read and
- write formats with them. AFS-63 #close ........ Merged revisions
- 415342 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-06 14:12 +0000 [r415319] George Joseph <george.joseph@fairview5.com>
-
- * tests/test_astobj2.c, main/astobj2_private.h (added),
- main/astobj2.c, main/astobj2_container_private.h (added),
- main/astobj2_container.c (added), main/astobj2_hash.c (added),
- main/astobj2_rbtree.c (added), /, include/asterisk/astobj2.h:
- Split astobj2.c into more maintainable components. Split
- astobj2.c into the following files to improve maintainability.
- astobj2.c - object primitives, object primitive misc and
- initialization code. astobj2_private.h - internal object
- declarations needed by the containers. astobj2_container.c -
- generic conainer and container misc code.
- astobj2_container_hash.c - hash container specific code.
- astobj2_container_rbtree.c - rbtree container specific code.
- astobj2_container_private.h - generic container definitions and
- rtti prototypes. https://reviewboard.asterisk.org/r/3576/
- ........ Merged revisions 415317 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-06 12:49 +0000 [r415302] Rusty Newton <rnewton@digium.com>
-
- * /, configs/cli_aliases.conf.sample: configs/cli_aliases.conf: Two
- new aliases, plus enhancements for context names. Changed naming
- of included alias templates to avoid confusion between version
- names. For example, asterisk12 was for asterisk 1.2, so I changed
- it to asterisk_1dot2, so that later we can use asterisk_12 for
- Asterisk 12. Added alias for "features reload" to the template
- for Asterisk 11 style syntax template, as features reload was
- removed in 12, but you can still do "module reload features"
- Added alias for "pjsip reload" to the friendly template. It is
- shorter than "module reload res_pjsip.so" and if some are like
- me; I constantly forget that reloading chan_pjsip doesn't parse
- config. Remembering "pjsip reload" is just easier. ASTERISK-23654
- #close ASTERISK-23654 #comment Fixed by adding two new aliases
- and enhancements for context names. Review:
- https://reviewboard.asterisk.org/r/3572/ ........ Merged
- revisions 415301 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-05 19:04 +0000 [r415231-415288] Richard Mudgett <rmudgett@digium.com>
-
- * main/config.c: config: Fix indentation and missing curlies in
- config_text_file_load().
-
- * main/config.c, /: config: Fix config files not reloading when
- only an included file changes. The twisted logic determining if a
- config file should be reloaded was mostly broken and disabled.
- The incorrect test that ASTERISK-23383 fixed actually reenabled
- the broken logic. The incorrect test was causing the timestamp to
- always be cleared which caused config files with includes to
- always be reloaded. * Made wildcard includes always cause a
- reload. Determining if a file was deleted cannot be determined
- without restructuring the cache to determine if any files are
- missing from the last files actually loaded. Also without
- refactoring config_text_file_load(), the glob loop couldn't check
- more than one file for changes anyway. * Made remove the cache
- entry if the file no longer exists when trying to get its
- timestamp or it is no longer a regular file. This fixes the
- corner case where the file was loaded, then deleted, then the
- config reloaded, then the file restored with the same timestamp,
- and then the config reloaded again. * Made remove the cache entry
- include list when actually loading the file. This gets rid of any
- stale includes the file had from the last time the file was
- loaded. ASTERISK-23683 #close Reported by: tootai Review:
- https://reviewboard.asterisk.org/r/3575/ ........ Merged
- revisions 415225 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415229 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415230 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-05 17:22 +0000 [r415223] Kevin Harwell <kharwell@digium.com>
-
- * tests/test_uri.c (added), include/asterisk/http_websocket.h,
- main/http.c, main/uri.c (added), tests/test_websocket_client.c
- (added), res/res_http_websocket.c, include/asterisk/http.h,
- include/asterisk/uri.h (added),
- res/res_http_websocket.exports.in: res_http_websocket: Create a
- websocket client Added a websocket server client in Asterisk.
- Asterisk has a websocket server, but not a client. The ability to
- have Asterisk be able to connect to a websocket server can
- potentially be useful for future work (for instance this could
- allow ARI to connect back to some external system, although more
- work would be needed in order to incorporate that). Also a couple
- of things to note - proxy connection support has not been
- implemented and there is limited http response code handling
- (basically, it is connect or not). Also added an initial new URI
- handling mechanism to core. Internet type URI's are parsed into a
- data structure that contains pointers to the various parts of the
- URI. (closes issue ASTERISK-23742) Reported by: Kevin Harwell
- Review: https://reviewboard.asterisk.org/r/3541/
-
-2014-06-05 14:49 +0000 [r415208] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/app_confbridge.c: app_confbridge: Allow muting of users
- waiting to enter a ConfBridge Prior to this patch, users waiting
- to enter a ConfBridge were not considered when muted via the CLI
- or via AMI. Instead, a confusing message would be emitted stating
- that the channel did not exist. This patch allows a user to be
- muted when waiting to enter a ConfBridge conference. This is
- equivalent to start when muted, only toggled via the CLI or AMI.
- Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824
- #close patches: rb3582.patch uploaded by tm1000 (License 6524)
- ........ Merged revisions 415206 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415207 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-05 11:59 +0000 [r415192] Kinsey Moore <kmoore@digium.com>
-
- * /, channels/chan_pjsip.c: PJSIP: Send initial connected line
- information This makes chan_pjsip send connected line information
- when it is called so that connected line information is available
- on the connected channel. (closes issue DPMA-442) Reported by:
- John Bigelow Review: https://reviewboard.asterisk.org/r/3584/
- ........ Merged revisions 415191 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-04 20:16 +0000 [r415173] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, contrib/scripts/safe_asterisk: safe_asterisk: Cleanup and
- debian compatibility. Cleans up the safe_asterisk script and adds
- the ASTSAFE_FOREGROUND option that allows the debian asterisk
- init script to capture the right pid. * Drop the vim #modeline
- which wasn't used. Use test consistently without the odd
- configure xno syntax. Double quote all paths. General cleanup. *
- Don't output message()s to the console but only to TTY if set. *
- Allow TTY to be "no" as well as empty (debian compatibility with
- debian/patches/safe_asterisk-config). * Add option to export
- ASTSAFE_FOREGROUND=1 from the init script that calls this to
- disable backgrounding. Debian uses a similar method in
- debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review:
- https://reviewboard.asterisk.org/r/3574/ ........ Merged
- revisions 415132 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415171 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415172 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-04 14:13 +0000 [r415116-415118] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_pjsip.c: chan_pjsip: Add debug in RTP Engine
- glue callback This patch adds some debug statements that aid with
- determining why a direct media request may or may not be
- initiated. ........ Merged revisions 415117 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Add debug
- statement for session refreshes This small patch adds a debug
- level 3 statement indicating how a session refresh is being sent
- - either as a re-INVITE or as an UPDATE - and where the session
- refresh is going. ........ Merged revisions 415115 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-04 07:27 +0000 [r415080] Corey Farrell <git@cfware.com>
-
- * /, apps/confbridge/include/confbridge.h, apps/app_confbridge.c:
- app_confbridge: Correct verification of conference name length
- Conference names were not checked for maximum length, allowing
- unexpected behaviour. This change adds checking to ensure the
- maximum length is not exceeded. The maximum length is also
- changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close
- Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches:
- confbridge-enforce_max-1.8.patch uploaded by coreyfarrell
- (license 5909) confbridge-enforce_max-11up.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 415060 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 415066 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 415078 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-03 07:36 +0000 [r415000] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, funcs/func_odbc.c: func_odbc: Fix fixed size buffers fix
- (r414968). The change that removed the fixed size buffers in
- odbc-related code -- removing arbitrary column width limits --
- was incomplete. This change adds: no segfault on writesql without
- insertsql and return value checks after strdup. While I was in
- the vicinity I cleaned up the linefeeds in the odbc function
- descriptions, moved some code for clarity, removed some blobs and
- noted (but didn't fix) that the 'odbc write ... exec' CLI command
- doesn't behave as the dialplan equivalent when insertsql= is
- used. ASTERISK-23582 #close Review:
- https://reviewboard.asterisk.org/r/3579/ ........ Merged
- revisions 414997 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414998 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414999 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-06-01 15:32 +0000 [r414976] Joshua Colp <jcolp@digium.com>
-
- * /, bridges/bridge_native_rtp.c: bridge_native_rtp: Take the
- bridge type choice of both channels into account. The
- bridge_native_rtp module currently uses the bridge result of the
- first channel that joins a bridge as the ultimate result. This
- means that if the first channel has direct media enabled but the
- second does not a direct media bridge will still occur. This
- change makes it so that both sides are taken into account. If
- either side forbids the bridge or responds with a local bridge
- result then either a generic or local bridge occurs.
- ASTERISK-23541 #close Reported by: Justin E Review:
- https://reviewboard.asterisk.org/r/3577/ ........ Merged
- revisions 414975 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-30 14:53 +0000 [r414949] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip_refer.c, /: PJSIP: Prevent crash on blind transfer
- Blind transfers don't go too well with NULL channels which can
- occur if the channel has already been transferred away. (closes
- issue ASTERISK-23718) Reported by: Jonathan Rose ........ Merged
- revisions 414948 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-30 12:42 +0000 [r414883-414935] Matthew Jordan <mjordan@digium.com>
-
- * main/audiohook.c, CHANGES, res/ari/ari_model_validators.c,
- res/ari/ari_model_validators.h, funcs/func_talkdetect.c (added),
- include/asterisk/stasis_channels.h,
- rest-api/api-docs/events.json, /, main/stasis_channels.c:
- TALK_DETECT: A channel function that raises events when talking
- is detected This patch adds a new channel function TALK_DETECT
- that, when set on a channel, causes events indicating the
- start/stop of talking on a channel to be emitted to both AMI and
- ARI clients. The function allows setting both the silence
- threshold (the length of silence after which we decide no one is
- talking) as well as the talking threshold (the amount of energy
- that counts as talking). Parameters can be updated on a channel
- after talk detection has been enabled, and talk detection can be
- removed at any time. The events raised by the function use a
- nomenclature similar to existing AMI/ARI events. For AMI:
- ChannelTalkingStart/ChannelTalkingStop For ARI:
- ChannelTalkingStarted/ChannelTalkingFinished Review:
- https://reviewboard.asterisk.org/r/3563/ #ASTERISK-23786 #close
- Reported by: Matt Jordan ........ Merged revisions 414934 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/config.c, /: main/config.c: AMI action UpdateConfig EmptyCat
- clears all categories When invoking UpdateConfig AMI action with
- Action set to EmptyCat, Asterisk will make all categories empty
- in the config but the one requested with a Cat variable. This is
- due to a bug in ast_category_empty (main/config.c) that makes an
- incorrect comparison for a category name. This patch corrects the
- comparison such that only the requested category is cleared.
- Review: https://reviewboard.asterisk.org/r/3573/ #ASTERISK-23803
- #close Reported by: zvision patches: manager.c.diff uploaded by
- zvision (License 5755) ........ Merged revisions 414880 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414881 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414882 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-29 18:51 +0000 [r414861] Kinsey Moore <kmoore@digium.com>
-
- * main/pbx.c, /: PBX: Prevent incorrect hint parsing Dynamic and
- pattern matching hints should not be checked for their last known
- state until they are instantiated by subscribers. (closes issue
- AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted
- by Matt Jordan (license 6283) ........ Merged revisions 414813
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 414859 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414860 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-28 22:54 +0000 [r414798] Matthew Jordan <mjordan@digium.com>
-
- * main/loader.c, include/asterisk/logger.h, res/res_config_curl.c,
- cel/cel_odbc.c, res/res_config_odbc.c,
- bridges/bridge_builtin_features.c, main/optional_api.c,
- main/logger.c, main/config_options.c, cdr/cdr_odbc.c,
- apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c,
- main/xmldoc.c, apps/app_voicemail.c, cel/cel_pgsql.c,
- channels/chan_unistim.c, res/res_config_pgsql.c, main/pbx.c,
- cdr/cdr_sqlite3_custom.c, res/res_fax.c, main/bridge.c,
- apps/app_waitforsilence.c, cdr/cdr_adaptive_odbc.c,
- res/parking/parking_applications.c, cdr/cdr_pgsql.c,
- res/res_jabber.c: Logger/CLI/etc.: Fix some aesthetic issues;
- reduce chatty verbose messages This patch addresses some
- aesthetic issues in Asterisk. These are all just minor tweaks to
- improve the look of the CLI when used in a variety of settings.
- Specifically: * A number of chatty verbose messages were removed
- or demoted to DEBUG messages. Verbose messages with a verbosity
- level of 5 or higher were - if kept as verbose messages - demoted
- to level 4. Several messages that were emitted at verbose level 3
- were demoted to 4, as announcement of dialplan applications being
- executed occur at level 3 (and so the effects of those
- applications should generally be less). * Some verbose messages
- that only appear when their respective 'debug' options are
- enabled were bumped up to always be displayed. *
- Prefix/timestamping of verbose messages were moved to the
- verboser handlers. This was done to prevent duplication of
- prefixes when the timestamp option (-T) is used with the CLI. *
- Verbose magic is removed from messages before being emitted to
- non-verboser handlers. This prevents the magic in multi-line
- verbose messages (such as SIP debug traces or the output of
- DumpChan) from being written to files. * _Slightly_ better
- support for the "light background" option (-W) was added. This
- includes using ast_term_quit in the output of XML documentation
- help, as well as changing the "Asterisk Ready" prompt to bright
- green on the default background (which stands a better chance of
- being displayed properly than bright white). Review:
- https://reviewboard.asterisk.org/r/3547/
-
-2014-05-28 20:53 +0000 [r414781] Rusty Newton <rnewton@digium.com>
-
- * /, configs/pjsip.conf.sample: pjsip.conf: privkey_file should be
- priv_key_file, mediaencryption=yes should be mediaencryption=sdes
- privkey_file was missed in the snake case update. An example
- included an invalid value for the mediaencryption option.
- ........ Merged revisions 414780 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-28 17:46 +0000 [r414764-414766] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json,
- rest-api/api-docs/mailboxes.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json,
- rest-api/api-docs/playbacks.json,
- rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/resources.json, include/asterisk/manager.h,
- rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json: AMI/ARI: Update version
- numbers Update the semantic versioning of ARI to 1.3.0 and AMI to
- 2.3.0 to account for backwards compatible changes going from
- 12.2.0 to 12.3.0. ........ Merged revisions 414765 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * contrib/ast-db-manage/cdr/env.py, /: ast-db-manage/cdr/env.py:
- Don't fail if a config file can't be loaded When generating SQL
- files via the repotools alembic_creator.py script, a
- configuration object is used programatically with SQLAlechemy, as
- opposed to a configuration file. This patch ignores failures to
- interpret a config file, as ... there isn't one in this case.
- ........ Merged revisions 414763 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-28 16:56 +0000 [r414748-414750] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h, /,
- res/res_pjsip_t38.c: res_pjsip_session: Fix leaked video RTP
- ports. Simply enabling PJSIP to negotiage a video codec (e.g.,
- h264) would leak video RTP ports if the codec were not negotiated
- by an incoming call. * Made add_sdp_streams() associate the
- handler with the media stream if the handler handled the media
- stream. Otherwise, when the ast_sip_session_media object was
- destroyed it didn't know how to clean up the RTP resources. *
- Fixed sdp_requires_deferral() associating the handler with the
- media stream when deciding if the SDP processing needs to be
- deferred for T.38. Like the leaked video RTP ports, the T.38
- handler needs to clean up allocated resources from deciding if
- SDP processing needs to be deffered. * Cleaned up some dead code
- in handle_incoming_sdp() and sdp_requires_deferral().
- ASTERISK-23721 #close Reported by: cervajs Review:
- https://reviewboard.asterisk.org/r/3571/ ........ Merged
- revisions 414749 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, CHANGES, apps/app_agent_pool.c: app_agent_pool: Return to
- dialplan if the agent fails to ack the call. Improvements to the
- agent pool functionality. * AgentRequest no longer hangs up the
- caller if the agent fails to connect with the caller. It now
- continues in the dialplan. * AgentRequest returns AGENT_STATUS
- set to NOT_CONNECTED if the agent failed to connect with the
- call. Most likely because the agent did not acknowledge the call
- in time or got disconnected. * The agent alerting play file
- configured by the agent.conf custom_beep option can now be
- disabled by setting the option to an empty string. The agent is
- effectively alerted to a call presence when MOH stops. * Fixed
- bridge reference leak when the agent connects with a caller.
- ASTERISK-23499 #close Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3551/ ........ Merged
- revisions 414747 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-28 11:37 +0000 [r414696] Joshua Colp <jcolp@digium.com>
-
- * res/res_config_odbc.c, /, funcs/func_odbc.c: res_config_odbc: Use
- dynamically sized buffers to store row data so values do not get
- truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported
- by: Walter Doekes Review:
- https://reviewboard.asterisk.org/r/3557/ ........ Merged
- revisions 414693 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414694 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414695 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-28 09:43 +0000 [r414567-414679] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, channels/chan_unistim.c: chan_unistim: Unlock mutex in rare
- OOM condition. #ASTERISK-23792 #close Reported by: Peter Whisker
- Review: https://reviewboard.asterisk.org/r/3567/ ........ Merged
- revisions 414677 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414678 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip: Start session timer at 200, not
- at INVITE. Asterisk started counting the session timer at INVITE
- while the other end correctly started at 200. This meant that for
- short session-expiries (90 seconds) combined with long ringing
- times (e.g. 30 seconds), asterisk would wrongly assume that the
- timer was hit before the other end thought it was time to send a
- session refresh. This resulted in prematurely ended calls. This
- changes the session timer to start counting first at 200 like RFC
- says it should. (Also removed a few excess NULL checks that would
- never hit, because if they did, asterisk would have crashed
- already.) ASTERISK-22551 #close Reported by: i2045 Review:
- https://reviewboard.asterisk.org/r/3562/ ........ Merged
- revisions 414620 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414628 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414636 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_config_odbc.c, /: res_config_odbc: Fix old and new
- ast_string_field memory leaks. The ODBC realtime driver uses ^NN
- parameter encoding to cope with the special meaning of the
- semi-colon. A semi-colon in a field is interpreted as if the key
- was supplied twice, something which isn't otherwise possible with
- fixed database columns. E.g. allow=alaw;ulaw is parsed as
- allow=alaw and allow=ulaw. A literal semi-colon is rewritten to
- ^3B when stored in the database. The module uses a stringfield to
- efficiently store the encoded parameters. However, this
- stringfield wasn't always freed in some off-nominal cases. Commit
- r413241 fixed initialization so the encoding for INSERT and
- DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked
- apparently.) But that commit forgot the frees. This change cleans
- that up. Review: https://reviewboard.asterisk.org/r/3555/
- ........ Merged revisions 414564 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414565 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414566 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-25 02:37 +0000 [r414543] Matthew Jordan <mjordan@digium.com>
-
- * /, main/core_unreal.c: core_unreal: Prevent double free of
- core_unreal pvt When a channel is destroyed (such as via
- ast_channel_release in off nominal paths in core_unreal), it will
- attempt to free (via ast_free) the channel tech pvt. This is
- problematic for a few reasons: 1. The channel tech pvt is an ao2
- object in core_unreal. Free'ing the pvt directly is no good. 2.
- The channel tech pvt's reference count is dropped just prior to
- calling ast_channel_release, resulting in the pvt's destruction.
- Hence, the channel destructor is free'ing an invalid pointer.
- This patch keeps the dropping of the reference count, but sets
- the pvt to NULL on the channel prior to releasing it. This models
- what would occur if the channel was hung up directly. ........
- Merged revisions 414542 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-23 17:36 +0000 [r414529] Matthew Jordan <mjordan@digium.com>
-
- * tests/test_cel.c, /: test_cel: Fix unit tests broken due to event
- def changes from res_corosync This patch instructs test_cel to
- skip any IE types it doesn't care about. The addition of the raw
- and bitfield types caused the tests to fail. ........ Merged
- revisions 414528 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-23 14:36 +0000 [r414475] Kinsey Moore <kmoore@digium.com>
-
- * main/event.c, /: Fix signed/unsigned build warnings ........
- Merged revisions 414474 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-22 16:19 +0000 [r414417] Richard Mudgett <rmudgett@digium.com>
-
- * /, apps/app_meetme.c: app_meetme: Don't interrupt MOH for
- waitmarked users. Occasionally, when the last marked user leaves
- the conference, waitmarked users don't get MOH if MOH is supposed
- to be played while a waitmarked user is waiting for another
- marked user. * Made not interrupt MOH when the user is a
- waitmarked user. The waitmarked user doesn't need to hear any
- leave announcements from the conference as the user would have
- already heard different leave announcements if they were enabled.
- Apparently DAHDI occasionally sends unending non-silent streams
- to these users or a normal user still in the conference has
- continuous high background noise. These non-silent streams cause
- MOH to be suspended while the never ending "announcement" is
- played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
- by: Tyler Stewart Review:
- https://reviewboard.asterisk.org/r/3543/ ........ Merged
- revisions 414401 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414402 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414404 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-22 16:09 +0000 [r414406] Scott Griepentrog <sgriepentrog@digium.com>
-
- * rest-api/api-docs/events.json, /, res/stasis/app.c,
- res/ari/resource_events.c, include/asterisk/stasis_app.h,
- include/asterisk/stasis.h, apps/app_userevent.c,
- res/ari/resource_events.h, res/ari/ari_model_validators.c,
- CHANGES, main/stasis.c, res/ari/ari_model_validators.h,
- include/asterisk/stasis_channels.h, res/res_ari_events.c,
- main/stasis_channels.c, res/res_stasis.c,
- main/manager_channels.c, main/stasis_endpoints.c: ARI: Add
- ability to raise arbitrary User Events User events can now be
- generated from ARI. Events can be signalled with arbitrary json
- variables, and include one or more of channel, bridge, or
- endpoint snapshots. An application must be specified which will
- receive the event message (other applications can subscribe to
- it). The message will also be delivered via AMI provided a
- channel is attached. Dialplan generated user event messages are
- still transmitted via the channel, and will only be received by a
- stasis application they are attached to or if the channel is
- subscribed to. This change also introduces the multi object blob
- mechanism used to send multiple snapshot types in a single
- message. The dialplan app UserEvent was also changed to use multi
- object blob, and a new stasis message type created to handle
- them. ASTERISK-22697 #close Review:
- https://reviewboard.asterisk.org/r/3494/ ........ Merged
- revisions 414405 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-22 15:52 +0000 [r414403] Jonathan Rose <jrose@digium.com>
-
- * include/asterisk/bridge.h, res/parking/parking_bridge_features.c,
- channels/chan_mgcp.c, res/res_pjsip_refer.c,
- channels/chan_dahdi.c, channels/sig_analog.c, /,
- channels/chan_sip.c, main/parking.c, main/bridge.c,
- main/bridge_basic.c, res/parking/parking_applications.c,
- include/asterisk/parking.h: res_pjsip_refer: Fix bugs involving
- Parking/PJSIP/transfers PJSIP would never send the final 200
- Notify for a blind transfer when transferring to parking. This
- patch fixes that. In addition, it fixes a reference leak when
- performing blind transfers to non-bridging extensions. Review:
- https://reviewboard.asterisk.org/r/3485/ ........ Merged
- revisions 414400 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-22 14:02 +0000 [r414331-414348] Matthew Jordan <mjordan@digium.com>
-
- * /, UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag ........
- Merged revisions 414345 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414346 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414347 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_corosync.c, include/asterisk/stasis.h, main/app.c,
- main/devicestate.c, main/event.c, main/stasis.c,
- include/asterisk/devicestate.h, include/asterisk/event.h,
- main/stasis_message.c, /, include/asterisk/event_defs.h:
- res_corosync: Update module to work with Stasis (and compile)
- This patch fixes res_corosync such that it works with Asterisk
- 12. This restores the functionality that was present in previous
- versions of Asterisk, and ensures compatibility with those
- versions by restoring the binary message format needed to pass
- information from/to them. The following changes were made in the
- core to support this: * The event system has been partially
- restored. All event definition and event types in this patch were
- pulled from Asterisk 11. Previously, we had hoped that this
- information would live in res_corosync; however, the approach in
- this patch seems to be better for a few reasons: (1)
- Theoretically, ast_events can be used by any module as a binary
- representation of a Stasis message. Given the structure of an
- ast_event object, that information has to live in the core to be
- used universally. For example, defining the payload of a device
- state ast_event in res_corosync could result in an incompatible
- device state representation in another module. (2) Much of this
- representation already lived in the core, and was not easily
- extensible. (3) The code already existed. :-) * Stasis message
- types now have a message formatter that converts their payload to
- an ast_event object. * Stasis message forwarders now handle
- forwarding to themselves. Previously this would result in an
- infinite recursive call. Now, this simply creates a new
- forwarding object with no forwards set up (as it is the thing it
- is forwarding to). This is advantageous for res_corosync, as
- returning NULL would also imply an unrecoverable error. Returning
- a subscription in this case allows for easier handling of message
- types that are published directly to an aggregate topic that has
- forwarders. Review: https://reviewboard.asterisk.org/r/3486/
- ASTERISK-22912 #close ASTERISK-22372 #close ........ Merged
- revisions 414330 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-21 22:24 +0000 [r414297] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/core_unreal.c: core_unreal: Only block media frames when
- a generator is on both ends of an unreal channel. The fix for
- ASTERISK-12292 was a bit too aggressive. You could have
- generators pointed at each other on local channels but need to
- get other kinds of frames such as DTMF or CONNECTED_LINE frames
- accross. ........ Merged revisions 414269 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414270 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414272 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-21 19:08 +0000 [r414217] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, funcs/func_strings.c: pbx.c: prevent potential crash from
- recursive replace() Recurisve usage of replace() resulted in
- corruption of the temporary string storage and potential crash.
- By changing the string to be allocated separtely per instance,
- this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
- Meer ASTERISK-23650 #close Review:
- https://reviewboard.asterisk.org/r/3539/ ........ Merged
- revisions 414214 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414215 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414216 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-19 19:52 +0000 [r414196] Paul Belanger <paul.belanger@polybeacon.com>
-
- * res/res_stasis_answer.c, /: Replace __ast_answer with
- ast_raw_answer in app_control_answer While load testing an ARI
- application, I noticed asterisk was returning HTTP 500 internal
- server errors on channels/:id/answer. After talking to
- #asterisk-dev, the issue appeared to be a lack of media flowing
- after __ast_answer() was called. So now, we call ast_raw_answer
- instead and no longer wait for media. ASTERISK-23758 #close
- Review: https://reviewboard.asterisk.org/r/3549/ ........ Merged
- revisions 414195 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-19 01:10 +0000 [r414123-414138] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- main/bridge_channel.c, res/res_pjsip_refer.c,
- res/res_pjsip_session.c, main/channel.c, /, main/framehook.c:
- Undo r414123 The Test Suite caught a few problems, undoing until
- those are resolved
-
- * include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- main/bridge_channel.c, res/res_pjsip_session.c, main/channel.c,
- /, main/framehook.c: bridge_native_rtp/bridge_channel: Fix direct
- media issues due to frame hook This patch fixes issues with
- direct media bridges that occur after a blind transfer. These
- issues were caught by the (currently failing)
- pjsip/transfers/blind_transfer/caller_direct_media test. The test
- currently fails primarily for two reasons: (1) When Bob and
- Charlie (the transfer target and the transfer destination) enter
- a bridge together, the framehook remains on the transfer target
- channel until both channels are in the bridge. As it consumes
- voice frames, the initial bridge type is a simple bridge. The
- framehook is removed when both channels are in the bridge;
- however, this does not currently cause the bridging framework to
- re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE
- poke to the transfer target channel when a framehook is removed
- so the bridge can re-evaluate itself. (2) When a channel leaves a
- native RTP bridge, it may be leaving due to being hung up.
- Sending a re-INVITE to a channel that is about to be hung up is
- not nice - in fact, there's a good chance we'll send the BYE
- request before the channel has had a chance to send back a 200
- OK. To be somewhat nicer, this patch adds a function to channel.h
- that allows the bridging framework to query for exactly why a
- channel is leaving a bridge via the channel's soft hangup flags.
- This allows it to only send the re-INVITE if there's a chance the
- channel will survive the native bridging experience. Review:
- https://reviewboard.asterisk.org/r/3535/ ........ Merged
- revisions 414122 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-16 20:06 +0000 [r413994-414070] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone
- detection. * Check if waitingfordt (waitfordialtone) is enabled
- in dahdi_read() to allow the DSP to operate early enough to
- detect dialtone. * Made use the correct variable in
- my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
- Davies Patches: dialtone_detect_fix (license #5012) patch
- uploaded by Steve Davies Review:
- https://reviewboard.asterisk.org/r/3534/ ........ Merged
- revisions 414067 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 414068 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414069 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/sig_pri.c, /: sig_pri.c: Pull the pri_dchannel()
- PRI_EVENT_RING case into its own function. * Populate the
- CALLERID(ani2) value (and the special CALLINGANI2 channel
- variable) with the ANI2 value in addition to the PRI specific
- ANI2 channel variable. * Made complete snapshot staging with the
- channel lock held. All channel snapshots need to be done while
- the channel lock is held. ........ Merged revisions 414050 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 414051 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI
- conference data structure. Starting a conference recording using
- the admin menu overwrites the DAHDI conference data structure
- used to modify the admin user's conference mute mode. * Made no
- longer pass the user's DAHDI conference data structure into the
- menu functions. The menu now uses its own DAHDI conference data
- structure to start the recording channel. * Moved the unlock
- conf->playlock to before playing the conf-full message. No sense
- keeping the lock while that prompt is playing. The user is never
- going to get into the conference at that point. ........ Merged
- revisions 413991 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413992 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413993 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-14 15:41 +0000 [r413897] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a
- few free()'s that should be ast_free()'s. Reverted an old
- workaround that isn't necessary. Reorder a tiny bit of code.
- Remove a bit of commented-out code. Review:
- https://reviewboard.asterisk.org/r/3536/ ........ Merged
- revisions 413894 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413895 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413896 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-13 18:09 +0000 [r413878] Jonathan Rose <jrose@digium.com>
-
- * main/netsock2.c, /, channels/chan_sip.c,
- include/asterisk/netsock2.h: chan_sip: Add TLS and SRTP status to
- CLI command 'sip show channel' ASTERISK-23564 #close Reported by:
- Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/
- ........ Merged revisions 413876 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413877 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-13 13:53 +0000 [r413790-413793] Walter Doekes <walter+asterisk@wjd.nu>
-
- * res/res_format_attr_h264.c, /: h264: Fix H264 SDP payload format.
- https://tools.ietf.org/html/rfc3984#section-8.1 says
- profile-level-id takes 3 bytes in base16 (6 hex digits). This
- fixes video setup in certain cases. ASTERISK-23664 #close
- ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume
- Maudoux. Review: https://reviewboard.asterisk.org/r/3530/
- ........ Merged revisions 413791 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413792 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/rtp_engine.c: rtp: Fix case typo in H263+ mime.
- http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
- canonical mime subtype is "H263-1998", not "h263-1998". Original
- code was added in r183101 on 2009-03-19 02:26:50 +0100. This
- fixes issues with Polycom phones. ASTERISK-23665 #close
- ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
- Maudoux, backported by me. Review:
- https://reviewboard.asterisk.org/r/3529/ ........ Merged
- revisions 413787 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413788 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413789 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-13 00:35 +0000 [r413770-413772] Richard Mudgett <rmudgett@digium.com>
-
- * configure.ac, channels/sig_pri.c, /, configure,
- include/asterisk/autoconfig.h.in: chan_dahdi/sig_pri: Prevent
- unnecessary PROGRESS events when overlap dialing is enabled. When
- overlap dialing is enabled, the lack of inband audio available
- information in the SETUP_ACKNOWLEDGE events causes an
- interoperability problem with SIP. sig_pri doesn't know if there
- is dialtone present when a SETUP_ACKNOWLEDGE is received so it
- assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
- SIP channel driver then sends out a 183 Session Progress and
- blocks the desired 180 Ringing message when the ALERTING message
- comes in. * Made the configure script detect if the installed
- version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
- Using the new API, made generate an AST_CONTROL_PROGRESS frame on
- an incoming SETUP_ACKNOWLEDGE message when the message indicates
- inband audio is present instead of assuming that dialtone is
- present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
- inband audio available indication only if dialtone is expected.
- The change also makes the fallback behaviour of sending the
- PROGRESS message better by sending it only if dialtone is
- expected. * Changed receiving a PROCEEDING message to not
- generate an AST_CONTROL_PROGRESS frame if the progress indication
- ie indicates non-end-to-end-ISDN. This helps interoperability
- with SIP. * Changed sending a PROCEEDING message in response to
- an AST_CONTROL_PROCEEDING frame to not indicate inband audio
- available. It was silly to do so anyway because the channel
- driver doesn't know if inband audio is even available. This helps
- interoperability with SIP. This patch and a corresponding change
- in libpri work together to allow Asterisk to control the inband
- audio available progress indication ie on the SETUP_ACKNOWLEDGE
- message when dialtone is present. AST-1338 #close Reported by:
- Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
- ........ Merged revisions 413714 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413765 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413771 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/sig_pri.c: Fix compiler warning from GCC 4.10 fixup.
- ........ Merged revisions 413766 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-12 22:33 +0000 [r413713] Jonathan Rose <jrose@digium.com>
-
- * apps/app_chanspy.c, /: app_chanspy: Fix a test that was failing
- on account of r413551 ASTERISK-23381 #close ASTERISK-23381
- #comment Reported by: Robert Moss Review:
- https://reviewboard.asterisk.org/r/3505/ ........ Merged
- revisions 413710 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413712 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-11 02:09 +0000 [r413651-413682] Joshua Colp <jcolp@digium.com>
-
- * main/bridge_basic.c, include/asterisk/channel.h,
- bridges/bridge_native_rtp.c, include/asterisk/framehook.h,
- main/channel.c, /, main/framehook.c: framehooks: Add callback for
- determining if a hook is consuming frames of a specific type. In
- the past framehooks have had no capability to determine what
- frame types a hook is actually interested in consuming. This has
- meant that code has had to assume they want all frames, thus
- preventing native bridging. This change adds a callback which
- allows a framehook to be queried for whether it is consuming a
- frame of a specific type. The native RTP bridging module has also
- been updated to take advantange of this, allowing native bridging
- to occur when previously it would not. ASTERISK-23497 #comment
- Reported by: Etienne Lessard ASTERISK-23497 #close Review:
- https://reviewboard.asterisk.org/r/3522/ ........ Merged
- revisions 413681 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- include/asterisk/framehook.h, main/channel.c, /,
- main/framehook.c, main/bridge_basic.c: Undoing framehook support.
- Issues were uncovered by Bamboo.
-
- * /, main/framehook.c, main/bridge_basic.c,
- include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- include/asterisk/framehook.h, main/channel.c: framehooks: Add
- callback for determining if a hook is consuming frames of a
- specific type. In the past framehooks have had no capability to
- determine what frame types a hook is actually interested in
- consuming. This has meant that code has had to assume they want
- all frames, thus preventing native bridging. This change adds a
- callback which allows a framehook to be queried for whether it is
- consuming a frame of a specific type. The native RTP bridging
- module has also been updated to take advantange of this, allowing
- native bridging to occur when previously it would not.
- ASTERISK-23497 #comment Reported by: Etienne Lessard
- ASTERISK-23497 #close Review:
- https://reviewboard.asterisk.org/r/3522/ ........ Merged
- revisions 413650 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-09 23:18 +0000 [r413589-413599] Kinsey Moore <kmoore@digium.com>
-
- * /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged
- revisions 413592 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413595 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413597 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_festival.c, pbx/dundi-parser.c, apps/app_getcpeid.c,
- main/netsock.c, funcs/func_channel.c, main/audiohook.c,
- pbx/pbx_config.c, res/res_pjsip_registrar.c, main/xmldoc.c,
- channels/iax2/firmware.c, apps/app_voicemail.c, main/format.c,
- cel/cel_pgsql.c, main/rtp_engine.c, main/parking.c,
- main/bridge.c, res/res_jabber.c, res/res_http_websocket.c,
- main/config.c, res/res_format_attr_opus.c, main/loader.c,
- res/parking/parking_bridge.c, main/cdr.c, main/manager.c,
- include/asterisk/astobj.h, main/bucket.c, apps/app_dumpchan.c,
- main/app.c, res/res_pjsip/config_transport.c,
- res/res_pjsip_refer.c, channels/chan_mgcp.c,
- res/res_rtp_asterisk.c, main/slinfactory.c, main/core_unreal.c,
- res/res_pjsip_sdp_rtp.c, res/res_crypto.c, main/acl.c,
- channels/sig_pri.c, res/res_monitor.c, res/res_srtp.c,
- main/data.c, res/res_corosync.c, channels/sip/config_parser.c,
- res/res_fax_spandsp.c, apps/app_stack.c, main/asterisk.c,
- main/udptl.c, res/res_sorcery_config.c, main/security_events.c,
- res/res_timing_dahdi.c, res/res_pjsip_t38.c,
- res/res_musiconhold.c, main/taskprocessor.c,
- res/res_format_attr_h263.c, res/res_xmpp.c, res/res_pktccops.c,
- funcs/func_hangupcause.c, channels/chan_phone.c,
- main/manager_bridges.c, cel/cel_odbc.c, channels/chan_skinny.c,
- channels/chan_motif.c, res/res_agi.c, main/logger.c,
- funcs/func_srv.c, channels/chan_alsa.c, apps/app_confbridge.c,
- res/res_pjsip_pubsub.c, channels/sip/include/sip.h, main/sched.c,
- apps/app_adsiprog.c, main/pbx.c, channels/chan_sip.c,
- res/res_fax.c, main/aoc.c, res/res_calendar_ews.c,
- res/parking/parking_bridge_features.c, channels/iax2/parser.c,
- main/callerid.c, main/file.c,
- res/res_pjsip/pjsip_configuration.c, main/adsi.c,
- main/config_options.c, pbx/pbx_dundi.c, funcs/func_iconv.c,
- main/bridge_channel.c, res/res_odbc.c, channels/chan_pjsip.c,
- res/parking/parking_manager.c, res/res_calendar.c, /,
- funcs/func_sysinfo.c, main/utils.c, cdr/cdr_adaptive_odbc.c,
- res/res_calendar_caldav.c, res/res_stasis_snoop.c,
- res/res_format_attr_h264.c, main/channel.c, res/ael/pval.c,
- res/res_ari_model.c, channels/chan_dahdi.c,
- channels/sig_analog.c, funcs/func_frame_trace.c,
- res/res_format_attr_silk.c, main/manager_channels.c,
- apps/app_dial.c, res/res_calendar_icalendar.c, main/translate.c,
- apps/app_queue.c, channels/chan_jingle.c, res/res_stun_monitor.c,
- main/abstract_jb.c, res/res_stasis_recording.c, apps/app_sms.c,
- main/event.c, apps/app_verbose.c, main/dsp.c,
- channels/chan_unistim.c, main/frame.c, res/res_stasis_playback.c,
- main/ccss.c, funcs/func_env.c, main/devicestate.c,
- bridges/bridge_softmix.c, channels/chan_gtalk.c,
- channels/chan_iax2.c, main/enum.c, main/cli.c,
- res/res_format_attr_celt.c, apps/confbridge/conf_config_parser.c,
- main/io.c, channels/pjsip/dialplan_functions.c,
- res/res_config_odbc.c, res/res_pjsip/location.c,
- res/res_pjsip_outbound_registration.c, formats/format_pcm.c,
- apps/app_minivm.c, main/stdtime/localtime.c, main/stun.c: Allow
- Asterisk to compile under GCC 4.10 This resolves a large number
- of compiler warnings from GCC 4.10 which cause the build to fail
- under dev mode. The vast majority are signed/unsigned mismatches
- in printf-style format strings. ........ Merged revisions 413586
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 413587 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413588 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-09 18:15 +0000 [r413572] Richard Mudgett <rmudgett@digium.com>
-
- * main/http.c: http.c: Remove dead code.
-
-2014-05-09 17:03 +0000 [r413557] Jonathan Rose <jrose@digium.com>
-
- * apps/app_chanspy.c, /: app_chanspy: Fix a bug where Barge mode
- could fail If the barge audiohook was attached prior to the spyee
- and its peer actually being bridged, the audiohook would not be
- applied and the connected peer would not be able to hear audio
- from the spy when the spy is in barge mode. (closes issue
- ASTERISK-23381) Reported by: Robert Moss Review:
- https://reviewboard.asterisk.org/r/3505/ ........ Merged
- revisions 413551 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413556 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-08 00:36 +0000 [r413488] Joshua Colp <jcolp@digium.com>
-
- * apps/app_queue.c, main/manager.c, /: app_queue: Extend
- documentation for various Manager actions and events. ........
- Merged revisions 413485 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413486 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413487 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-07 21:58 +0000 [r413469] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_presencestate.c: Ensure that presence state is decoded
- properly on Asterisk startup. The CustomPresence provider
- callback will automatically base64 decode stored data if the 'e'
- option was present when the state was set. However, since the
- provider callback was bypassed on Asterisk startup, encoded
- presence subtypes and messages were being sent instead. This fix
- makes it so the provider callback is always used when providing
- presence state updates.
-
-2014-05-07 20:59 +0000 [r413453-413455] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_confbridge.c, /: app_confbridge: Fixed "CBAnn" channels
- not going away. Fixed a ref leak in conf_handle_talker_cb()
- everytime the conference bridge was found to report a channel's
- talker status change. The resulting leak caused the "CBAnn"
- channels and the conference bridge to never be destroyed. Thanks
- to Richard Kenner on the asterisk-user's list for locating the
- problem. Reported by: Richard Kenner ........ Merged revisions
- 413454 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_confbridge.c, /: app_confbridge: Fix ref leak in CLI
- "confbridge kick" command. Fixed ref leak in the CLI "confbridge
- kick" command when the channel to be kicked was not in the
- conference. ........ Merged revisions 413451 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413452 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-07 17:56 +0000 [r413307-413399] Mark Michelson <mmichelson@digium.com>
-
- * res/res_config_odbc.c, /: Fix encoding of custom prepare extra
- data. Patches: res_config_odbc-take2.patch by John Hardin
- (License #6512) ........ Merged revisions 413396 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413397 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413398 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip/presence_xml.c, /,
- res/res_pjsip_pidf_digium_body_supplement.c: Improve XML
- sanitization in NOTIFYs, especially for presence subtypes and
- messages. Embedded carriage return line feed combinations may
- appear in presence subtypes and messages since they may be
- derived from user input in an instant messenger client. As such,
- they need to be properly escaped so that XML parsers do not vomit
- when the messages are received. ........ Merged revisions 413372
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_registrar.c, /: Check for an act on failures to
- update contacts during registration. There was an underlying
- issue in a realtime backend where database updates would fail.
- Since we were not checking for failure, we would end up in a
- strange state where the old database entry was still present but
- Asterisk thought that it had been updated. Now when an entry
- fails to update, we print a warning and delete the old contact
- from sorcery so there is no mismatch between foreground and
- backend state. Patches: res_pjsip_registrar.patch by John Hardin
- (License #6512) ........ Merged revisions 413358 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_config_odbc.c, /: Ensure that all parts of SQL UPDATEs
- and DELETEs are encoded. Patches: res_config_odbc.patch by John
- Hardin (License #6512) ........ Merged revisions 413304 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413305 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413306 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-02 20:28 +0000 [r413227-413263] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_config_odbc.c: Prevent crashes in res_config_odbc due
- to uninitialized string fields. Patches: odbc-crash.patch by John
- Hardin (License #6512) ........ Merged revisions 413241 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413251 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413258 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_config_pgsql.c, /: Return the number of rows affected by
- a SQL insert, rather than an object ID. The realtime API
- specifies that the store callback is supposed to return the
- number of rows affected. res_config_pgsql was instead returning
- an Oid cast as an int, which during any nominal execution would
- be cast to 0. Returning 0 when more than 0 rows were inserted
- causes problems to the function's callers. To give an idea of how
- strange code can be, this is the necessary code change to fix a
- device state issue reported against chan_pjsip in Asterisk 12+.
- The issue was that the registrar would attempt to insert contacts
- into the database. Because of the 0 return from res_config_pgsql,
- the registrar would think that the contact was not successfully
- inserted, even though it actually was. As such, even though the
- contact was query-able and it was possible to call the endpoint,
- Asterisk would "think" the endpoint was unregistered, meaning it
- would report the device state as UNAVAILABLE instead of
- NOT_INUSE. The necessary fix applies to all versions of Asterisk,
- so even though the bug reported only applies to Asterisk 12+, the
- code correction is being inserted into 1.8+. Closes issue
- ASTERISK-23707 Reported by Mark Michelson ........ Merged
- revisions 413224 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 413225 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413226 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-02 16:39 +0000 [r413211] Richard Mudgett <rmudgett@digium.com>
-
- * UPGRADE.txt, res/res_pjsip_refer.c, /, channels/chan_sip.c:
- res_pjsip_refer: Add Referred-By header on INVITE for blind
- transfers. Per rfc3892, the Referred-By header in a REFER must be
- copied into the referenced request (IE. The outgoing INVITE to
- the transfer target). * Automatically put the Referred-By header
- in the outgoing INVITE message if the SIPREFERREDBYHDR channel
- variable is defined with a value. * Made
- chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance
- so chan_pjsip has a better chance to interoperate. * Fixed
- refer_blind_callback() and refer_incoming_refer_request() to not
- modify the data in the pointer returned by
- pjsip_msg_find_hdr_by_name(). It seems wrong to modify that data
- since the calling routine doesn't own the buffer. ASTERISK-23501
- #close Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/3514/ ........ Merged
- revisions 413210 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-02 16:06 +0000 [r413197] Jonathan Rose <jrose@digium.com>
-
- * res/parking/res_parking.h, /, CHANGES,
- res/parking/parking_bridge_features.c,
- res/parking/parking_manager.c: Parking: Add 'AnnounceChannel'
- argument to manager action 'Park' (closes ASTERISK-23397)
- Reported by: Denis Review:
- https://reviewboard.asterisk.org/r/3446/ ........ Merged
- revisions 413196 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-01 16:21 +0000 [r413174-413183] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_presencestate.c: Make behavior of the PRESENCE_STATE
- 'e' option more consistent. When writing presence state, if 'e'
- is specified, then the presence state will be stored in the astdb
- encoded. However, consumers of presence state events or those
- that query for the presence state will be given decoded
- information. If base64 encoding is desired for consumers, then
- the information can be base64-encoded manually and the 'e' option
- can be omitted. closes issue ASTERISK-23671 Reported by Mark
- Michelson Review: https://reviewboard.asterisk.org/r/3482
-
- * res/res_pjsip_exten_state.c, /: Remove unnecessary repetition
- checks from res_pjsip_exten_state The PBX core already takes care
- of ensuring that repeated state changes are not communicated to
- exten state consumers. Because the check in res_pjsip_exten_state
- was incomplete, it was causing valid presence state changes not
- to be sent out. For instance, if the presence state did not
- change but the message or subtype did, then no presence-related
- NOTIFY request would be sent out. closes issue ASTERISK-23672
- Reported by Mark Michelson ........ Merged revisions 413173 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-05-01 12:31 +0000 [r413160] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip/config_transport.c, /: res_pjsip: Add the ability
- to configure ciphers based on name. Previously this code would
- only accept the OpenSSL identifier instead of the documented
- name. ASTERISK-23498 #close ASTERISK-23498 #comment Reported by:
- Anthony Messina Review: https://reviewboard.asterisk.org/r/3491/
- ........ Merged revisions 413159 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-30 21:03 +0000 [r413144] Richard Mudgett <rmudgett@digium.com>
-
- * main/message.c, /, channels/chan_sip.c,
- include/asterisk/message.h, res/res_pjsip_messaging.c:
- chan_sip.c: Fixed off-nominal message iterator ref count and
- alloc fail issues. * Fixed early exit in sip_msg_send() not
- destroying the message iterator. * Made
- ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
- tolerant of a NULL iter parameter in case
- ast_msg_var_iterator_init() fails. * Made
- ast_msg_var_iterator_destroy() clean up any current message data
- ref. * Made struct ast_msg_var_iterator,
- ast_msg_var_iterator_init(), ast_msg_var_iterator_next(),
- ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy()
- use iter instead of i. * Eliminated RAII_VAR usage in
- res_pjsip_messaging.c:vars_to_headers(). ........ Merged
- revisions 413139 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413142 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-30 20:39 +0000 [r413141] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_pjsip.c: chan_pjsip: Fix deadlock when
- retrieving call-id of channel. If a task was in-flight which
- required the channel or bridge lock it was possible for the
- synchronous task retrieving the call-id to deadlock as it holds
- those locks. After discussing with Mark Michelson the synchronous
- task was removed and the call-id accessed directly. This should
- be safe as each object involved is guaranteed to exist and the
- call-id will never change. ........ Merged revisions 413140 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-30 13:08 +0000 [r413125] Kinsey Moore <kmoore@digium.com>
-
- * res/res_http_websocket.c, /: Websocket: Add session locking and
- delay close This resolves a race condition where data could be
- written to a NULL FILE pointer causing a crash as a websocket
- connection was in the process of shutting down by adding locking
- to websocket session writes and by deferring session teardown
- until session destruction. (closes issue ASTERISK-23605) Review:
- https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan
- ........ Merged revisions 413123 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413124 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-30 12:42 +0000 [r413118-413122] Joshua Colp <jcolp@digium.com>
-
- * /, res/stasis/control.c: res_stasis: Add progress indications to
- operations which perform media. This change fixes operations
- which did not account for the fact that they may be executed on
- channels which have not been answered. These operations will now
- indicate progress when invoked. ASTERISK-23560 #close
- ASTERISk-23560 #comment Reported by: Jan Svoboda Review:
- https://reviewboard.asterisk.org/r/3495/ ........ Merged
- revisions 413121 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix issue where
- sending a hold SDP twice could cause an unhold. This change fixes
- a bug where if an SDP with media address and sendonly was
- received twice the underlying call would go off hold, instead of
- remaining on hold. This occured because the code did not properly
- take into account that the SDP may contain both a valid media
- address and the sendonly attribute. The code now examines the
- sendonly attribute and media address first, so if the SDP is
- received again no change will occur. ASTERISK-23558 #comment
- Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/3472/ ........ Merged
- revisions 413119 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip:
- Add support for picking up calls in the configured pickup group.
- AST-1363 Review: https://reviewboard.asterisk.org/r/3478/
- ........ Merged revisions 413117 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-29 15:10 +0000 [r413103] George Joseph <george.joseph@fairview5.com>
-
- * /, include/asterisk/spinlock.h: Add "destroy" implementation for
- spinlock. The original commit for spinlock was missing "destroy"
- implementations. Most of them are no-ops but phtread_spin and
- pthread_mutex do need their locks destroyed. ........ Merged
- revisions 413102 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-29 11:27 +0000 [r413089] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_pjsip.c, /: chan_pjsip: Implement core ability to
- get Call-ID of a channel. This changes implement the
- "get_pvt_uniqueid" which is used to return the technology
- specific unique identifier. In the case of SIP this is the
- Call-ID of the dialog. Review:
- https://reviewboard.asterisk.org/r/3480/ ........ Merged
- revisions 413088 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-28 20:07 +0000 [r413074] Kinsey Moore <kmoore@digium.com>
-
- * /, main/bridge.c, main/bridge_basic.c: Bridging: Don't lock NULL
- bridges When bridge locking was added for bridge snapshot
- creation, some locations where bridge locking was added were not
- guaranteed to actually have a bridge and locking NULL AO2 objects
- tends to cause segfaults. This ensures that NULL bridges aren't
- locked. ........ Merged revisions 413073 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-28 14:40 +0000 [r413060] Mark Michelson <mmichelson@digium.com>
-
- * res/res_manager_presencestate.c (added), main/devicestate.c,
- CHANGES, main/presencestate.c, res/res_manager_devicestate.c
- (added): Add DeviceStateChanged and PresenceStateChanged AMI
- events. These events are controlled by two new modules,
- res_manager_devicestate and res_manager_presencestate. Review:
- https://reviewboard.asterisk.org/r/3417
-
-2014-04-28 07:43 +0000 [r413048] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * UPGRADE.txt, CHANGES, channels/chan_unistim.c,
- configs/unistim.conf.sample: Introducing changes proposed to
- chan_unistim driver: 1) Added the unistim.conf variable
- dtmf_duration which can select the DTMF playback duration from
- 0ms to 150ms (0 is off and is the new default) 2) Enabled the
- transmission of month names, which are sent with the date and
- changed the dateformat variable to accept the values 0-3 as per
- the UNISTIM standard (2 & 3 match the previous 1 & 2 formats). 3)
- Enabled the "Mute" packet so muting microphone works as expected
- and microphone muted for all calls while LED light on 4) Changed
- Duree to Timer on i2004 display (closes issue ASTERISK-23592)
-
-2014-04-27 19:29 +0000 [r413036] Olle Johansson <oej@edvina.net>
-
- * main/tcptls.c: tcptls.c : Log errors as ERROR, not warning or
- something else.
-
-2014-04-25 19:26 +0000 [r413012] Matthew Jordan <mjordan@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Add support for DTLS
- handshake retransmissions On congested networks, it is possible
- for the DTLS handshake messages to get lost. This patch adds a
- timer to res_rtp_asterisk that will periodically check to see if
- the handshake has succeeded. If not, it will retransmit the DTLS
- handshake. Review: https://reviewboard.asterisk.org/r/3337
- ASTERISK-23649 #close Reported by: Nitesh Bansal patches:
- dtls_retransmission.patch uploaded by Nitesh Bansal (License
- 6418) ........ Merged revisions 413008 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 413009 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-24 14:37 +0000 [r412993] Kevin Harwell <kharwell@digium.com>
-
- * /,
- contrib/ast-db-manage/config/versions/e96a0b8071c_increase_pjsip_column_size.py
- (added): pjsip realtime: increase the size of some columns The
- string lengths on certain columns created through alembic for
- PJSIP were too short. For instance, columns containing URIs are
- currently set to 40 characters, but this can be too small and
- result in truncated values. Added an alembic migration script
- that increases the size of these columns and a few others to 255.
- ASTERISK-23639 #close Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3475/ ........ Merged
- revisions 412992 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-23 20:13 +0000 [r412977] George Joseph <george.joseph@fairview5.com>
-
- * include/asterisk/spinlock.h (added), /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: This patch adds
- support for spinlocks in Asterisk. There are cases in Asterisk
- where it might be desirable to lock a short critical code section
- but not incur the context switch and yield penalty of a mutex or
- rwlock. The primary spinlock implementations execute exclusively
- in userspace and therefore don't incur those penalties. Spinlocks
- are NOT meant to be a general replacement for mutexes. They
- should be used only for protecting short blocks of critical code
- such as simple compares and assignments. Operations that may
- block, hold a lock, or cause the thread to give up it's timeslice
- should NEVER be attempted in a spinlock. The first use case for
- spinlocks is in astobj2 - internal_ao2_ref. Currently the
- manipulation of the reference counter is done with an
- ast_atomic_fetchadd_int which works fine. When weak reference
- containers are introduced however, there's an additional
- comparison and assignment that'll need to be done while the lock
- is held. A mutex would be way too expensive here, hence the
- spinlock. Given that lock contention in this situation would be
- infrequent, the overhead of the spinlock is only a few more
- machine instructions than the current ast_atomic_fetchadd_int
- call. ASTERISK-23553 #close Review:
- https://reviewboard.asterisk.org/r/3405/ ........ Merged
- revisions 412976 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-23 18:03 +0000 [r412925] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/http.c: http: Fix spurious ERROR message in responses
- with no content. Backport -r411687 and fix the fix because
- content_length is the length of out plus the length of the file
- controlled by fd. When a response has an out content length of 0,
- fwrite would be called to write a buffer with no data in it. This
- resulted in the following classic error message: [Apr 3 11:49:17]
- ERROR[26421] http.c: fwrite() failed: Success This patch makes it
- so that we only attempt to write the content of out if the out
- string is non-zero. ........ Merged revisions 412922 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412923 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412924 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-23 15:02 +0000 [r412910] Russell Bryant <russell@russellbryant.com>
-
- * res/res_monitor.c, funcs/func_periodic_hook.exports.in (added),
- main/asterisk.dynamics, funcs/func_periodic_hook.c: Fix error
- loading res_monitor. For some odd reason, loading app_mixmonitor
- was fine, but res_monitor was not. This patch fixes a set of
- issues related to func_periodic_hook exporting the beep functions
- that gets res_monitor working again.
-
-2014-04-22 10:09 +0000 [r412883] Joshua Colp <jcolp@digium.com>
-
- * /, res/stasis/app.c: res_stasis: Fix crash when handling a failed
- blind transfer message. This changes fixes a crash that occurs
- when stasis determines if it should send a message out to an
- application or not. The code incorrectly assumed that a bridge
- snapshot would always be present when in reality for failure
- cases it may not be. ASTERISK-23573 #close ........ Merged
- revisions 412882 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-21 17:56 +0000 [r412759-412824] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, /: chan_sip: trust_id_outbound CHANGES message
- improvement (closes issue AST-1301) (closes issue ASTERISK-19465)
- Reported by: Krzysztof Chmielewski ........ Merged revisions
- 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 412822 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412823 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
- channels/sip/include/sip.h: chan_sip: Add sendrpid trust options
- In r411189, some behavior was changed which made sendrpid
- behavior act in a more trusting manner by sending full user data
- for peers set with private caller presence in P-Asserted-Identity
- headers. Since this changed long time expected behaviors, we
- decided to pull that patch when that was pointed out by the
- community. Instead, this patch provides a trust_id_outbound
- setting which will expose the data per RFC-3325 if set to 'yes'
- and simply not send the PAI/RPID headers at all if set to 'no'.
- By default trust_id_outbound will be set to 'legacy' which will
- preserve the behavior prior to these patches. Extra special
- thanks to Walter Doekes for providing advice and feedback.
- (closes issue AST-1301) (closes issue ASTERISK-19465) Reported
- by: Krzysztof Chmielewski Review:
- https://reviewboard.asterisk.org/r/3447/ ........ Merged
- revisions 412744 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412746 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412747 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-21 16:16 +0000 [r412729-412750] Kinsey Moore <kmoore@digium.com>
-
- * main/http.c, main/manager.c, /: HTTP: Add TCP_NODELAY to accepted
- connections This adds the TCP_NODELAY option to accepted
- connections on the HTTP server built into Asterisk. This option
- disables the Nagle algorithm which controls queueing of outbound
- data and in some cases can cause delays on receipt of response by
- the client due to how the Nagle algorithm interacts with TCP
- delayed ACK. This option is already set on all non-HTTP AMI
- connections and this change would cover standard HTTP requests,
- manager HTTP connections, and ARI HTTP requests and websockets in
- Asterisk 12+ along with any future use of the HTTP server.
- Review: https://reviewboard.asterisk.org/r/3466/ ........ Merged
- revisions 412745 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412748 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412749 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_confbridge.c, /: Confbridge: Fix ConfbridgeKick AMI
- documentation This adds documentation for the "all" channel
- option for the ConfbridgeKick AMI action and adjusts AMI
- responses accordingly. (issue ASTERISK-23282) Reported by: Dorian
- Logan ........ Merged revisions 412730 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_confbridge.c: Confbridge: Add references for kick all
- option After the ability to kick all attendees from a conference
- was added, a rework removed the comment about that feature from
- the CLI documentation. This adds that documentation and adds
- "all" to the participant tab completion list for the confbridge
- kick command. (closes issue ASTERISK-23282) Reported by: Dorian
- Logan ........ Merged revisions 412728 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-21 08:36 +0000 [r412714] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * /, channels/chan_unistim.c: Fix wrong dialtone. The "modulation"
- should not be referenced for tone+tone as it refers to the on-off
- characteristic - this often resulted in a single tone rather than
- the multitone as in the UK. ........ Merged revisions 412712 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412713 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-19 02:14 +0000 [r412697-412699] Matthew Jordan <mjordan@digium.com>
-
- * /, main/asterisk.c: main/asterisk: Fix startup sequence for
- realtime features When ASTERISK-23265/ASTERISK-23320 was fixed,
- it inadvertently led to realtime features breaking. This was due
- to features loading prior to realtime. This patch fixes this by
- loading features after loading dynamic modules. ASTERISK-23487
- #close Reported by: Denis Tested by: Denis ........ Merged
- revisions 412698 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_sms.c: app_sms: Fix uninitialized values; hangup
- channel when REL is sent successfully This patch fixes two issues
- in app_sms: (1) Firstly, the 'flags' field on the stack in
- sms_exec() is uninitialised, causing it to use the wrong protocol
- in some cases. This patch correctly initializes the flags fields.
- (2) Secondly, when disconnect supervision is not working or
- inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was
- failing to terminate the call after it sent the REL(ease) message
- and the peer stopped talking to it. This patch fixes the code to
- handle the 'bad stop bit' message more gracefully in that case,
- and hang up the call. Review:
- https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close
- Reported by: David Woodhouse patches: asterisk-fix-sms.patch
- uploaded by David Woodhouse (License 5754) ........ Merged
- revisions 412655 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412656 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412657 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-18 20:09 +0000 [r412641] Jonathan Rose <jrose@digium.com>
-
- * /, res/ari/resource_bridges.h, res/stasis/control.c,
- include/asterisk/stasis_app.h, res/stasis/control.h,
- res/ari/resource_channels.c, CHANGES, res/res_stasis.c,
- rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
- res/res_ari_bridges.c, res/res_stasis_playback.c: ARI: Make
- bridges/{bridgeID}/play queue sound files Previously multiple
- play actions against a bridge at one time would cause the sounds
- to play simultaneously on the bridge. Now if a sound is already
- playing, the play action will queue playback to occur after the
- completion of other sounds currently on the queue. (closes issue
- ASTERISK-22677) Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/3379/ ........ Merged
- revisions 412639 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-18 17:17 +0000 [r412589] Rusty Newton <rnewton@digium.com>
-
- * sounds/sounds.xml, sounds/Makefile, /: sounds: Fix Sounds
- Makefile and XML that didn't support new sound prompt sets In
- sounds/Makefile 1 Adds and moves some lines necessary for the
- en_GB core set. I'm just following how the other sets are defined
- here. 2 removes the ES extra sounds related lines as we don't
- have ES extra sound sets. In sounds/sounds.xml 3 Adds member
- definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in
- extra sound sets ASTERISK-23550 #close Review:
- https://reviewboard.asterisk.org/r/3464/ ........ Merged
- revisions 412586 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412587 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-18 17:02 +0000 [r412584] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip/location.c: Allow for multiple contacts to be
- configured in a single contact= line. This is useful for
- configuring multiple permanent contacts for an AOR when using
- realtime AORs. Review: https://reviewboard.asterisk.org/r/3462
- ........ Merged revisions 412582 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-18 16:44 +0000 [r412580-412583] Richard Mudgett <rmudgett@digium.com>
-
- * main/dial.c, main/pbx.c, /, apps/app_originate.c,
- include/asterisk/pbx.h: Originated calls: Fix several originate
- call problems. * Restore the reason value set by
- pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the
- consumers were expecting rather than cause codes. * Fixed the
- dial routines to set cause codes for more than just ast_request()
- so pbx_outgoing_attempt() reason codes will function. * Fix
- inconsistent locked_channel return status in
- pbx_outgoing_attempt(). The chanel may not have been locked or
- the channel may have been a stale pointer. * Fixed the
- OutgoingSpoolFailed channel to run dialplan whenever the dialing
- fails for an originate exten and 1 < synchronous. * Fix incorrect
- ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by
- issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the
- ao2 lock instead of its own lock for the cond wait mutex. No
- sense in having two locks associated with the same struct when
- only one is needed. Review:
- https://reviewboard.asterisk.org/r/3421/ ........ Merged
- revisions 412581 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_channels.c, apps/app_queue.c, apps/app_dial.c, /:
- app_dial and app_queue: Make lock the forwarding channel while
- taking the channel snapshot. * Fixed
- ast_channel_publish_dial_forward() not locking the forwarded
- channel when taking the channel snapshot. * Fixed
- app_dial.c:do_forward() using the wrong channel to get the
- original call forwarding string. * Removed unnecessary locking
- when calling ast_channel_publish_dial() and
- ast_channel_publish_dial_forward() in app_dial and app_queue.
- Holding channel locks when calling
- ast_channel_publish_dial_forward() with a forwarded channel could
- result in pausing the system while the stasis bus completes
- processsing a forwarded channel subscription. Review:
- https://reviewboard.asterisk.org/r/3451/ ........ Merged
- revisions 412579 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-18 14:25 +0000 [r412566] Kinsey Moore <kmoore@digium.com>
-
- * res/ari/ari_websockets.c, res/res_ari.c, main/manager.c, /: ARI:
- Add debug logging for events and responses This adds DEBUG level
- logging for ARI websocket events and HTTP responses similar to
- what is available for AMI. Logging for ARI HTTP requests is
- already adequate for debugging purposes. ........ Merged
- revisions 412565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-17 22:50 +0000 [r412552] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/location.c, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
- res/res_pjsip_registrar.c: res_pjsip: Handle reloading when
- permanent contacts exist and qualify is configured. This change
- fixes a problem where permanent contacts being qualified were not
- being updated. This was caused by the permanent contacts getting
- a uuid and not a known identifier, causing an inability to look
- them up when updating in the qualify code. A bug also existed
- where the new configuration may not be available immediately when
- updating qualifies. (closes issue ASTERISK-23514) Reported by:
- Richard Mudgett Review: https://reviewboard.asterisk.org/r/3448/
- ........ Merged revisions 412551 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-17 22:42 +0000 [r412536-412550] Jonathan Rose <jrose@digium.com>
-
- * /, main/app.c: Fix a silly shadowed variable mistake that was
- missed from play tones patch ........ Merged revisions 412549
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/ari/resource_bridges.h, main/app.c,
- rest-api/api-docs/channels.json, CHANGES,
- rest-api/api-docs/bridges.json, res/ari/resource_channels.h,
- include/asterisk/app.h, res/res_stasis_playback.c: ARI: Add tones
- playback resource Adds a tones URI type to the playback resource.
- The tone can be specified by name (from indications.conf) or by a
- tone pattern. In addition, tonezone can be specified in the URI
- (by appending ;tonezone=<zone>). Tones must be stopped manually
- in order for a stasis control to move on from playback of the
- tone. Tones may be paused, resumed, restarted, and stopped. They
- may not be rewound or fast forwarded (tones can't be controlled
- in a way that lets you skip around from note to note and pausing
- and resuming will also restart the tone from the beginning).
- Tests are currently in development for this feature
- (https://reviewboard.asterisk.org/r/3428/). (closes issue
- ASTERISK-23433) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3427/ ........ Merged
- revisions 412535 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-17 20:25 +0000 [r412467-412484] Matthew Jordan <mjordan@digium.com>
-
- * channels/chan_oss.c, /, main/Makefile: main/Makefile: Fix build
- failure on SmartOS/Illumos/SunOS This patch fixes two issues when
- building on SmartOS: - channels/chan_oss.c: it makes sure
- soundcard.h is found - main/Makefile: only use
- "-Wl,--version-script" when GNU LD is used as the Sun Linker
- doesn't support that. Similar checks are already used elswhere in
- the Makefile Review: https://reviewboard.asterisk.org/r/3426
- ASTERISK-23576 #close Reported by: Sebastian Wiedenroth patches:
- fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
- ........ Merged revisions 412468 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412483 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/sip/include/sip.h, channels/chan_sip.c, CHANGES:
- chan_sip: Add SIPURIPHONECONTEXT channel variable for Request TEL
- URIs This patch is a continuation of
- https://reviewboard.asterisk.org/r/3349/, committed in r412303.
- It resolves a finding oej had that the phone-context be available
- in a channel variable separate from SIPDOMAIN. This patch adds
- that variable as SIPURIPHONECONTEXT. It also allows a local
- number (or global number specified in the TEL URI) to be used to
- look up as a peer. (issue ASTERISK-17179) Review:
- https://reviewboard.asterisk.org/r/3349/
-
-2014-04-17 15:17 +0000 [r412454] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_refer.c, /: res_pjsip_refer: Channel variable
- SIPREFERTOHDR not being set during blind transfer The
- SIPREFERTOHDR channel variable is not being set on any channel
- when performing a blind transfer using PJSIP. The
- 'refer->refer_to' was not being set during a blind transfer.
- Updated so the 'refer_to' is set to the target uri on a blind
- transfer. (closes issue ASTERISK-23502) Reported by: John Bigelow
- Review: https://reviewboard.asterisk.org/r/3445/ ........ Merged
- revisions 412453 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-16 19:14 +0000 [r412440] Kinsey Moore <kmoore@digium.com>
-
- * /, include/asterisk/stasis_app.h: Stasis: Add a usage note on
- stasis_app_get_bridge This function returns an ast_bridge without
- a refcount bump and the caller must increment the count if it
- intends to hold the pointer. (closes issue ASTERISK-23588)
- Review: https://reviewboard.asterisk.org/r/3450/ Reported by:
- Matt Jordan ........ Merged revisions 412439 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-15 23:21 +0000 [r412427] Russell Bryant <russell@russellbryant.com>
-
- * bridges/bridge_builtin_features.c, include/asterisk/monitor.h,
- CHANGES, apps/app_queue.c, funcs/func_periodic_hook.c,
- apps/app_mixmonitor.c, include/asterisk/beep.h (added),
- res/res_monitor.c: (mix)monitor: Add options to enable a periodic
- beep Add an option to enable a periodic beep to be played into a
- call if it is being recorded. If enabled, it uses the
- PERIODIC_HOOK() function internally to play the 'beep' prompt
- into the call at a specified interval. This option is provided
- for both Monitor() and MixMonitor(). Review:
- https://reviewboard.asterisk.org/r/3424/
-
-2014-04-15 18:30 +0000 [r412384-412414] Richard Mudgett <rmudgett@digium.com>
-
- * main/stasis_channels.c, main/features_config.c,
- res/res_parking.c, main/rtp_engine.c, /: Eliminate some more
- unnecessary RAII_VAR() uses. RAII_VAR() is not a hammer
- appropriate to pound all nails. ........ Merged revisions 412413
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_playback.c, /, res/stasis/app.c, res/res_fax.c,
- res/res_pjsip/security_events.c,
- res/parking/parking_applications.c, channels/chan_oss.c,
- main/stasis_bridges.c, res/res_pjsip_session.c,
- res/stasis_recording/stored.c, main/cdr.c, res/res_parking.c,
- channels/chan_skinny.c, res/res_pjsip/location.c,
- res/res_stasis_recording.c, main/stasis_channels.c,
- res/ari/resource_channels.c, res/parking/parking_manager.c,
- res/ari/resource_recordings.c, res/res_pjsip_refer.c,
- res/res_ari.c, main/pbx.c: Remove unused RAII_VAR() declarations.
- * Remove unused RAII_VAR() declarations. The compiler cannot
- catch these because the cleanup function "references" the unused
- variable. Some actually allocated and released resources that
- were never used. * Fixed some whitespace issues in
- stasis_bridges.c. ........ Merged revisions 412399 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/rtp_engine.h, main/rtp_engine.c, /,
- channels/chan_sip.c: chan_sip.c: Fix channel staging assertion
- failure. The failing assertion ensures that the final snapshot
- gets generated so CDR records can get finalized. The only place
- where a channel staging snapshot flag could be left set is in
- chan_sip.c:handle_request_bye(). The function could return before
- clearing the flag because the channel could dissappear while the
- function had to have the channel unlocked. * Fixed
- handle_request_bye() channel snapshot staging coverage area to
- not have a return in the middle of it and be unable to clear the
- staging flag. * Pushed the channel snapshot staging coverage area
- into ast_rtp_instance_set_stats_vars() to ensure that the staging
- is not interrutped. * Made callers of
- ast_rtp_instance_set_stats_vars() not call it with any channels
- or channel driver private locks held to eliminate the deadlock
- potential. The callers must hold references to the passed in
- channel and rtp objects. * Eliminated sip_hangup() trying to get
- the bridge peer. It is futile at this point because the channel
- could never be in a bridge. Review:
- https://reviewboard.asterisk.org/r/3431/ ........ Merged
- revisions 412385 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip.c: Moved some sip_pvt unrefs
- after their last use. * Moved sip_pvt unref in ast_hangup() and
- handle_request_do() to the end of the function. The unref needs
- to happen after the last use of the pointer. ........ Merged
- revisions 412348 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412383 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-15 16:13 +0000 [r412331] Jonathan Rose <jrose@digium.com>
-
- * configs/sip.conf.sample, /, channels/chan_sip.c: Reverting
- r411189 so that it can be put up for public review --- r411189 |
- jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines
- chan_sip: Send real CallerID information with
- P-Assserted-Identity (RFC-3325) Prior to this patch, the
- P-Asserted-Identity header would include anonymous caller id
- information which seems to go against the point of the
- P-Asserted-Identity header. Now the real caller ID information
- will be included in this header. Also, no privacy header would be
- included. This patch adds 'Privacy: id' to outgoing SIP messages
- that include the P-Asserted-Identity header. (closes issue
- AST-1301) --- ........ Merged revisions 412328 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412329 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412330 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-14 15:54 +0000 [r412307] Corey Farrell <git@cfware.com>
-
- * main/autoservice.c, /: autoservice: fix reference leak of logger
- callid. autoservice acquires a local reference to the logger
- callid of each channel in a loop. This local reference was not
- released, causing the callid of every channel in autoservice to
- leak. This change moves the callid unref inside the loop.
- ASTERISK-23616 #close Reported by: ibercom ........ Merged
- revisions 412305 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412306 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-12 02:27 +0000 [r412292] Matthew Jordan <mjordan@digium.com>
-
- * channels/sip/reqresp_parser.c, CHANGES, channels/chan_sip.c:
- chan_sip: Support RFC-3966 TEL URIs in inbound INVITE requests
- This patch adds support for handling TEL URIs in inbound INVITE
- requests. This includes the Request URI and the From URI. The
- number specified in the Request URI will be the destination of
- the inbound channel in the dialplan. The phone-context specified
- in the Request URI will be stored in the TELPHONECONTEXT channel
- variable. Review: https://reviewboard.asterisk.org/r/3349
- ASTERISK-17179 #close Reported by: Geert Van Pamel Tested by:
- Geert Van Pamel patches:
- asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van
- Pamel (License 6140)
- asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by
- Geert Van Pamel (License 6140)
-
-2014-04-12 01:35 +0000 [r412279-412280] Russell Bryant <russell@russellbryant.com>
-
- * funcs/func_periodic_hook.c: func_periodic_hook: move module ref
- The previous code left one error path where the module would be
- unref'd twice instead of once. It was done once in the error
- handling block, and again inside of datastore destruction. Now
- the module ref is only released in the datastore destructor and
- only acquired when the datastore has been successfully allocated.
-
- * funcs/func_periodic_hook.c: func_periodic_hook: add module ref
- counting This module lacked necessary module ref count
- incrementing and decrementing when used. This patch adds it.
- There's already a datastore used, so doing the ref counting along
- with the lifetime of the datastore provides a convenient place to
- do it.
-
-2014-04-11 21:43 +0000 [r412213-412228] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_stack.c, /: app_stack: Add missing unlock in off-nominal
- path of STACK_PEEK function. ASTERISK-23620 #close Reported by:
- Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch
- (license #5021) patch uploaded by Bradley Watkins ........ Merged
- revisions 412225 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412226 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412227 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * utils/Makefile, utils: utils dir: Remove no longer needed traces
- of refcounter except in the clean make target. * Removed no
- longer needed files from the svn:ignore property to make them
- visible.
-
-2014-04-11 12:43 +0000 [r412194] Kinsey Moore <kmoore@digium.com>
-
- * /, main/bridge.c, main/bridge_basic.c,
- include/asterisk/stasis_bridges.h, tests/test_cel.c,
- apps/app_confbridge.c, res/ari/resource_bridges.c: bridging:
- Ensure locking during snapshot creation While the vast majority
- of bridge snapshot creation is locked properly, there are
- currently some instances that are not. This adds the missing
- locking to ensure bridge state is not malleable during snapshot
- creation. (closes issue ASTERISK-22904) Review:
- https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan
- ........ Merged revisions 412193 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-11 08:28 +0000 [r412168-412180] Olle Johansson <oej@edvina.net>
-
- * main/audiohook.c: Formatting: Remove invisible characters
-
- * main/audiohook.c: Formatting only.
-
-2014-04-11 02:59 +0000 [r412154] Matthew Jordan <mjordan@digium.com>
-
- * main/astobj2.c, contrib/scripts/refcounter.py (added),
- main/asterisk.c, utils/refcounter.c (removed),
- build_tools/cflags.xml, utils/utils.xml, /, channels/chan_sip.c,
- channels/sip/security_events.c, include/asterisk/astobj2.h,
- UPGRADE.txt: main/astobj2: Make REF_DEBUG a menuselect item;
- improve REF_DEBUG output This patch does the following: (1) It
- makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
- REF_DEBUG globally throughout Asterisk. (2) The ref debug log
- file is now created in the AST_LOG_DIR directory. Every run will
- now blow away the previous run (as large ref files sometimes
- caused issues). We now also no longer open/close the file on each
- write, instead relying on fflush to make sure data gets written
- to the file (in case the ao2 call being performed is about to
- cause a crash) (3) It goes with a comma delineated format for the
- ref debug file. This makes parsing much easier. This also now
- includes the thread ID of the thread that caused ref change. (4)
- A new python script instead for refcounting has been added in the
- contrib/scripts folder. (5) The old refcounter implementation in
- utils/ has been removed. Review:
- https://reviewboard.asterisk.org/r/3377/ ........ Merged
- revisions 412114 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 412115 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 412153 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-11 01:12 +0000 [r412102] Russell Bryant <russell@russellbryant.com>
-
- * res/res_monitor.c: monitor: use app options parsing helper code
- This app is pretty ancient, so it was never converted to use the
- option parsing helper code. I'd like to add an option to this app
- that takes an argument, and that's a pain to do when not using
- this helper, so start by doing this conversion. Review:
- https://reviewboard.asterisk.org/r/3429/
-
-2014-04-10 21:28 +0000 [r412089] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_hep_pjsip.c: res_hep_pjsip: Use the channel name
- instead of the call ID when it is available During discussions
- with Alexandr Dubovikov at Kamailio World, it became apparent
- that while the SIP call ID is a useful identifier prior to an
- Asterisk channel being created, it is far more preferable to use
- the channel name (or some channel based identifier) when the
- channel is available. Homer is smart enough to tie the various
- messages together. This patch opts to use the channel name when
- it is available, falling back to the call ID otherwise. ........
- Merged revisions 412088 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-10 21:10 +0000 [r412075] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Set the body
- generation result to 0 for a valid path The result of the
- "ast_sip_pubsub_generate_body_content" was not set/initialized.
- Consequently, the nominal path potentially returned an invalid
- value, thus not sending mwi notifications. ........ Merged
- revisions 412074 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-09 21:43 +0000 [r412050] Mark Michelson <mmichelson@digium.com>
-
- * /, CHANGES, apps/app_mixmonitor.c: Add a Command header to the
- AMI Mixmonitor action. This fixes a parsing error that occurred
- during the processing of the AMI action. The error did not result
- in MixMonitor itself misbehaving, but it could result in the AMI
- response not giving correct information back. The new header
- allows for one to specify a post-process command to run when
- recording finishes. Previously, in order to do this, the
- post-process command would have to be placed at the end of the
- Options: header. Patches: mixmonitor_command_2.patch by jhardin
- (License #6512) ........ Merged revisions 412048 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-09 18:17 +0000 [r412035] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_stasis_answer.c: res_stasis_answer: Add missing
- newlines ........ Merged revisions 412034 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-08 21:25 +0000 [r411946-411990] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/asterisk.c: Internal timing: Add notice that the -I and
- internal_timing option are no longer needed. Add notice messages
- during execution that the -I command line option and the
- astersik.conf internal_timing option are no longer needed. The
- internal timing functionality is now always enabled if there is a
- timing module loaded. NOTE: Since the command line options and
- the asterisk.conf config file are processed before the logging
- system is initialized, the messages are output to stderr. Change
- requested as a result of asterisk-dev list comments about the
- commit for ASTERISK-22846 that removed the -I and internal_timing
- options. Review: https://reviewboard.asterisk.org/r/3423/
- ........ Merged revisions 411964 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411974 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411985 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/config.c, /: config: Fix CB_ADD_LEN() to work as originally
- intended. Fix a long standing bug in CB_ADD_LEN() behaving like
- CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes
- ........ Merged revisions 411960 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411961 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411962 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix
- confbridge.conf dsp_talking_threshold option setting wrong
- parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported
- by: John Knott ........ Merged revisions 411944 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411945 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-08 14:49 +0000 [r411928] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip.c: res_pjsip: Ignore explicit transport
- configuration if a WebSocket transport is specified. This change
- makes it so if a transport is configured on an endpoint that is a
- WebSocket type the option will be ignored. In practice this is
- fine because the WebSocket transport can not create outgoing
- connections, it can only reuse existing ones. By ignoring the
- option the existing PJSIP logic for using the existing connection
- will be invoked and stuff will proceed. (closes issue
- ASTERISK-23584) Reported by: Rusty Newton ........ Merged
- revisions 411927 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-08 00:26 +0000 [r411897] Russell Bryant <russell@russellbryant.com>
-
- * funcs/func_periodic_hook.c: func_periodic_hook: List more modules
- as dependencies This module makes use of some existing Asterisk
- components. app_chanspy was already listed as a dependency. There
- are a few function modules used, as well, so list them.
-
-2014-04-07 20:41 +0000 [r411884] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip_pubsub.c: PJSIP: Ensure test event has new state
- The change that fixed the pubsub test event's use of a dangling
- pointer also changed when it was processed relative to the pjsip
- subscription state change processing. This change corrects the
- order of events while holding a reference to the pointer that was
- previously dangling. ........ Merged revisions 411883 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-07 16:15 +0000 [r411870] Jonathan Rose <jrose@digium.com>
-
- * main/manager_channels.c, /: AGI/Manager: Prevent multiple
- NewExten events during AGI application changes AGI applications
- would trigger NewExten events every time the state of the AGI
- application changed. This has historically not been the behavior
- and this behavior was introduced with a CDR patch. This patch
- corrects that. (closes issue ASTERISK-23390) Reported by:
- Benjamin Keith Ford Review:
- https://reviewboard.asterisk.org/r/3406/ ........ Merged
- revisions 411868 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-07 14:57 +0000 [r411812] Walter Doekes <walter+asterisk@wjd.nu>
-
- * apps/app_queue.c, /: app_queue: Re-add HoldTime to
- QueueCallerAbandon event (simple typo during ast12 refactor).
- Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........
- Merged revisions 411811 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-07 14:29 +0000 [r411791-411806] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_stasis.c: Stasis: Fix Stasis() bridge refcount issue
- The Stasis() dialplan application monitors what bridge a channel
- is in and so necessarily holds on to a bridge pointer. This
- change ensures that it also holds on to a reference for that
- bridge to prevent the bridge pointer from becoming a dangling
- pointer. ........ Merged revisions 411804 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pubsub.c, /: PJSIP: Fix crash introduced in r411671
- The test event introduced in revision 411671 uses a dangling
- pointer to access information about pubsub state changes. This
- moves the event to within the lifetime of the pointer. ........
- Merged revisions 411790 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-05 13:06 +0000 [r411768] Russell Bryant <russell@russellbryant.com>
-
- * CHANGES, funcs/func_periodic_hook.c (added): func_periodic_hook:
- New function for periodic hooks. This commit introduces a new
- dialplan function, PERIODIC_HOOK(). It allows you run to a
- dialplan hook on a channel periodically. The original use case
- that inspired this was the ability to play a beep periodically
- into a call being recorded. The implementation is much more
- generic though and could be used for many other things. The
- implementation makes heavy use of existing Asterisk components.
- It uses a combination of Local channels and ChanSpy() to run some
- custom dialplan and inject any audio it generates into an active
- call. The other important bit of the implementation is how it
- figures out when to trigger the beep playback. This
- implementation uses the audiohook API, even though it's not
- actually touching the audio in any way. It's a convenient way to
- get a callback and check if it's time to kick off another beep.
- It would be nice if this was timer event based instead of polling
- based, but unfortunately I don't see a way to do it that won't
- interfere with other things. Review:
- https://reviewboard.asterisk.org/r/3362/
-
-2014-04-04 19:19 +0000 [r411702-411724] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/options.h, main/asterisk.c, main/channel.c, /,
- channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt,
- include/asterisk/channel.h, utils/extconf.c: internal_timing:
- Remove the option and always make it enabled if a timing module
- is loaded. The masquerade supertest frequently fails because
- either the local channel chain doesn't completely optimize out or
- the DTMF handshake doesn't completely get accross. Local channel
- optimization requires frames flowing to trigger when optimization
- can happen. When optimization happens the media frame that
- triggered the optimization is dropped. Sending DTMF requires
- frames to flow in the other direction for timing purposes while
- sending nothing. If internal timing is not enabled when MOH is
- playing, Asterisk switches to received timing when an audio frame
- is received. With optimization dropping media frames and MOH not
- sending frames unless it receives frames, occasionaly there are
- no more frames being passed and the test fails. * The asterisk
- command line -I option and the asterisk.conf internal_timing
- option are removed. Asterisk now always uses internal timing when
- needed if any timing module is loaded. The issue ASTERISK-14861
- did this quite awhile ago in v1.4 but effectively is broken if
- other internal timing modules besides DAHDI are used. The
- ast_read_generator_actions() now only does received timing if it
- has no choice for frame generators like MOH, silence, and
- playback streaming. * Cleaned up some code dealing with frame
- generators in ast_deactivate_generator(),
- generator_write_format_change(), ast_activate_generator(), and
- ast_channel_stop_silence_generator(). * Removed
- ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
- ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........
- Merged revisions 411715 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411716 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411717 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/utils.c, res/res_musiconhold.c, main/channel.c,
- main/stasis_cache.c, /: Add some asserts that were handy when
- looking for a stasis cache problem. * Assert if a channel is
- destroyed but has the snapshot staging flag set. In this case the
- final channel destruction snapshot would never get taken. *
- Assert if what we just got out of the stasis cache is not what we
- were looking for. This assert would have saved several days
- searching for a bug and a lot of my hair. * Assert if the music
- on hold message posts could not find the associated channel. A
- crash will happen later when manager tries to send the MOH AMI
- message. This assert catches the problem when the stasis message
- is posted instead of by the thread processing the defective
- message. * Always generate a backtrace when an ast_assert()
- fails. Review: https://reviewboard.asterisk.org/r/3411/ ........
- Merged revisions 411701 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-04 15:13 +0000 [r411688] Matthew Jordan <mjordan@digium.com>
-
- * /, main/http.c: http: Fix spurious ERROR message in responses
- with no content When a response has a content length of 0, fwrite
- would be called to write a buffer with no data in it. This
- resulted in the following classic error message: [Apr 3 11:49:17]
- ERROR[26421] http.c: fwrite() failed: Success This patch makes it
- so that we only attempt to write out the content if the
- calculated content_length is non-zero. ........ Merged revisions
- 411687 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-03 12:06 +0000 [r411671] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Add test event for
- state change This adds a test event when subscription state
- changes so that integration tests may trigger new actions at the
- appropriate times. Review:
- https://reviewboard.asterisk.org/r/3383/ ........ Merged
- revisions 411670 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-03 11:47 +0000 [r411669] Matthew Jordan <mjordan@digium.com>
-
- * res/res_hep.c, /: res_hep: Fix crash when hep.conf not available
- Parts of res_hep properly checked for a valid configuration
- object before attempting to access the configuration. A check,
- however, was missed when a packet is sent. This patch fixes the
- crash caused by not checking if the configuration object is
- valid. ........ Merged revisions 411668 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-02 18:57 +0000 [r411656] Mark Michelson <mmichelson@digium.com>
-
- * main/sorcery.c, /, res/res_mwi_external.c,
- res/res_pjsip/config_system.c, configs/sorcery.conf.sample,
- main/bucket.c, include/asterisk/sorcery.h,
- res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
- tests/test_sorcery.c, tests/test_sorcery_realtime.c: Prevent
- duplicate sorcery wizards from being applied to sorcery object
- types. This commit contains several changes to sorcery: 1)
- Application of sorcery configuration based on module name is
- automatically performed when sorcery is opened for a module. 2)
- Sorcery will not attempt to apply the same wizard to an object
- type more than once. 3) Sorcery gives more exact results when
- attempting to apply a wizard, whether as the default or based on
- configuration. Sorcery unit tests still pass for me after making
- these changes. Review: https://reviewboard.asterisk.org/r/3326
- ........ Merged revisions 411159 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-01 22:42 +0000 [r411637-411639] Richard Mudgett <rmudgett@digium.com>
-
- * res/parking/parking_bridge.c, /: res_parking: Minor tweaks. * Use
- ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of
- ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables.
- * Use ast_copy_string() instead of inlining it. * Remove an
- already done TODO comment. * Some whitespace tweaks. ........
- Merged revisions 411638 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_channels.c, /: stasis_channels.c: Eliminate another
- overuse of RAII_VAR(). ........ Merged revisions 411636 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-04-01 16:52 +0000 [r411587] Joshua Colp <jcolp@digium.com>
-
- * /, apps/app_queue.c: app_queue: Fix a bug where realtime members
- would be deleted during reload causing waiting callers to get
- ejected. This patch causes realtime queue members to remain in
- queues during the reload process. Previously these members would
- be removed causing any waiting callers to be ejected from the
- queue with a reason of "EXITEMPTY". ASTERISK-23547 #close
- ASTERISK-23547 #comment Patch
- app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo
- Rossi (license 6409) Review:
- https://reviewboard.asterisk.org/r/3404/ ........ Merged
- revisions 411584 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411585 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411586 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-28 18:32 +0000 [r411556] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/res_hep.h (added), res/res_hep_pjsip.c (added),
- res/res_hep.exports.in (added), configs/hep.conf.sample (added),
- CHANGES, res/res_hep.c (added), /: res_hep/res_hep_pjsip: Add a
- HEPv3 capture agent module and a logger for PJSIP This patch adds
- the following: (1) A new module, res_hep, which implements a
- generic packet capture agent for the Homer Encapsulation Protocol
- (HEP) version 3. Note that this code is based on a patch provided
- by Alexandr Dubovikov; I basically just wrapped it up, added
- configuration via the configuration framework, and threw in a
- taskprocessor. (2) A new module, res_hep_pjsip, which forwards
- all SIP message traffic that passes through the res_pjsip stack
- over to res_hep for encapsulation and transmission to a HEPv3
- capture server. Much thanks to Alexandr for his Asterisk patch
- for this code and for a *lot* of patience waiting for me to port
- it to 12/trunk. Due to some dithering on my part, this has taken
- the better part of a year to port forward (I still blame CDRs for
- the delay). ASTERISK-23557 #close Review:
- https://reviewboard.asterisk.org/r/3207/ ........ Merged
- revisions 411534 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-28 18:00 +0000 [r411533] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
- addons/chan_ooh323.c, /, addons/ooh323c/src/oochannels.c,
- addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c:
- process stack command even if gatekeeper client isn't register
- don't destroy gatekeeper client if it is not started don't
- destroy gatekeeper client in some sort of gatekeeper errors
- signal rtp create condition when call cleared before rtp
- structure created (closes issue ASTERISK-23460) Reported by:
- Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry
- Melekhov ........ Merged revisions 411531 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411532 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-28 17:41 +0000 [r411515-411530] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/channels.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/playbacks.json, UPGRADE.txt,
- rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES,
- include/asterisk/manager.h, rest-api/api-docs/bridges.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/mailboxes.json,
- rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json: Update API versions and
- UPGRADE/CHANGES for 12.2.0 This patch does the following: * It
- updates the AMI version to 2.2.0 to indicate backwards compatible
- changes have been made since the last release * It updates the
- ARI version to 1.2.0 to indicate backwards compatible changes
- have been made since the last release * It updates the
- UPGRADE/CHANGES files with changes that were not mentioned
- ........ Merged revisions 411529 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * UPGRADE.txt, res/res_config_odbc.c: res_config_odbc: Fix for
- nullable integer columns and keyfield existence check in
- update_odbc. This patch fixes setting nullable integer columns to
- NULL instead of an empty string, which fails for PostgreSQL, for
- example. The current code is supposed to do so, but the check is
- broken. The patch also allows the first column in the list to be
- a nullable integer. Also, the check for existence of a mandatory
- column checked for the first column in the list instead of the
- key field lookup column. This patch fixes that issue as well.
- Finally, the compatibility option allow_empty_string_in_nontext,
- which was added to previous revisions to allow for some database
- backends with certain schemas to function, has been removed.
- Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459
- #close ASTERISK-23351 #close (closes issue ASTERISK-23459)
- Reported by: zvision patches: res_config_odbc.diff uploaded by
- zvision (License 5755)
-
-2014-03-28 16:18 +0000 [r411469] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/tcptls.c, main/manager.c, /, main/http.c: http: response
- body often missing after specific request This patch works around
- a problem with the HTTP body being dropped from the response to a
- specific client and under specific circumstances: a) Client
- request comes from node.js user agent "Shred" via use of
- swagger-client library. b) Asterisk and Client are *not* on the
- same host or TCP/IP stack In testing this problem, it has been
- determined that the write of the HTTP body is lost, even if the
- data is written using low level write function. The only solution
- found is to instruct the TCP stack with the shutdown function to
- flush the last write and finish the transmission. See review for
- more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
- Reported by: Sam Galarneau Review:
- https://reviewboard.asterisk.org/r/3402/ ........ Merged
- revisions 411462 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411463 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411465 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-28 15:48 +0000 [r411375-411460] Matthew Jordan <mjordan@digium.com>
-
- * UPGRADE.txt, /: UPGRADE: Note IAX2 compatibility issue between
- 1.4 and 1.8+ systems. ........ Merged revisions 411457 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411458 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411459 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * contrib/realtime/mysql/voicemail_messages.sql (removed),
- contrib/realtime/postgresql/realtime.sql (removed),
- contrib/realtime/mysql/voicemail_data.sql (removed),
- contrib/realtime/mysql/musiconhold.sql (removed),
- contrib/realtime/mysql/queue_log.sql (removed),
- contrib/realtime/mysql/voicemail.sql (removed),
- contrib/realtime/mysql/sippeers.sql (removed), /,
- contrib/realtime/mysql/iaxfriends.sql (removed),
- contrib/realtime/mysql/meetme.sql (removed): contrib/realtime:
- Remove empty SQL script files Since the relatime scripts are now
- managed by Alembic, the previous realtime scripts were previously
- removed. However, the removal process messed up, as the files
- were still in the repository. The contents were just empty. This
- removes the files from the tree. ........ Merged revisions 411442
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/sip/include/sip.h: chan_sip: Add MESSAGE request to
- allowed methods The allowed methods advertised by chan_sip did
- not previously note the MESSAGE request. Even in Asterisk 1.8, we
- do accept in-dialog MESSAGE requests; we should advertise that we
- support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
- #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
- Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
- Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged
- revisions 411372 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411373 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411374 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-27 19:21 +0000 [r411312-411328] Corey Farrell <git@cfware.com>
-
- * funcs/func_global.c, apps/app_speech_utils.c,
- apps/confbridge/conf_config_parser.c,
- funcs/func_callcompletion.c, funcs/func_frame_trace.c,
- funcs/func_callerid.c, main/message.c, /, res/res_mutestream.c,
- channels/pjsip/dialplan_functions.c,
- res/res_pjsip_header_funcs.c, funcs/func_pitchshift.c,
- funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c,
- funcs/func_channel.c, funcs/func_cdr.c, funcs/func_blacklist.c,
- apps/app_stack.c, apps/app_voicemail.c, res/res_calendar.c,
- apps/app_jack.c, funcs/func_dialplan.c, funcs/func_speex.c,
- channels/chan_sip.c, funcs/func_math.c, funcs/func_strings.c,
- funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c,
- main/features_config.c, res/res_jabber.c: Fix dialplan function
- NULL channel safety issues (closes issue ASTERISK-23391) Reported
- by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3386/ ........ Merged
- revisions 411313 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411314 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411315 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/format.c, include/asterisk.h, /: main/formats: Fix crash in
- ast_format_cmp during non-clean shutdown. * Update asterisk.h to
- reflect availability of ast_register_cleanup in 11.9. * Use
- ast_register_cleanup for format_attr_shutdown. (closes issue
- ASTERISK-23103) Reported by: JoshE ........ Merged revisions
- 411310 from http://svn.asterisk.org/svn/asterisk/branches/11
- ........ Merged revisions 411311 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-27 14:21 +0000 [r411296] Mark Michelson <mmichelson@digium.com>
-
- * main/sorcery.c, /: Give sorcery instances a reference to their
- wizards. On graceful shutdown, sorcery wizards are all killed
- off, but it is possible for sorcery instances to still have
- dangling pointers after this, possibly causing a crash. Giving
- the sorcery instances a reference to their wizards ensures that
- the wizard reference will remain valid for the lifetime of the
- sorcery instance. Review: https://reviewboard.asterisk.org/r/3401
- ........ Merged revisions 411295 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-26 22:45 +0000 [r411246] Joshua Colp <jcolp@digium.com>
-
- * /, main/say.c: say: Fix a bug where SayNumber in Polish tries to
- play incorrect sound. This change fixes a bug where calling
- SayNumber with a number divisible by 100 using the Polish
- language would cause the code to attempt to play a sound file
- with an empty name. (closes issue ASTERISK-23509) Reported by:
- zvision Review: https://reviewboard.asterisk.org/r/3378/ ........
- Merged revisions 411243 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411244 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411245 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-26 16:15 +0000 [r411194] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send
- real CallerID information with P-Assserted-Identity (RFC-3325)
- Prior too this patch, the P-Asserted-Identity header would
- include anonymous caller id information which seems to go against
- the point of the P-Asserted-Identity header. Now the real caller
- ID information will be included in this header. Also, no privacy
- header would be included. This patch adds 'Privacy: id' to
- outgoing SIP messages that include the P-Asserted-Identity
- header. (closes issue AST-1301) ........ Merged revisions 411189
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 411190 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411193 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-26 16:05 +0000 [r411192] Richard Mudgett <rmudgett@digium.com>
-
- * /,
- contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py:
- Fix 'alembic branches' merge conflict as described by the web
- page. ........ Merged revisions 411191 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 18:44 +0000 [r411174] Sean Bright <sean@malleable.com>
-
- * /, res/ari/config.c: ARI: Don't complain about missing ARI users
- when we aren't enabled Currently, if ARI is not enabled it will
- still complain that there are no configured users. This patch
- checks to see if ARI is enabled before logging and error or
- iterating the container to validate the users. Review:
- https://reviewboard.asterisk.org/r/3391/ ........ Merged
- revisions 411173 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 17:40 +0000 [r411158] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
- res/res_pjsip_messaging.c, res/res_pjsip.c,
- include/asterisk/res_pjsip.h: Add a "message_context" option for
- PJSIP endpoints. ........ Merged revisions 411157 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 16:57 +0000 [r411142] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip/pjsip_options.c, res/res_pjsip.c,
- include/asterisk/res_pjsip.h, /: res_pjsip: Fix contact
- authenticate_qualify endpoint lookup when qualifing a contact. *
- Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of
- find_endpoints() with find_an_endpoint() since only the first
- found endpoint is ever needed. * Fixed qualify_contact_cb() to
- update the contact with the aor authenticate_qualify setting.
- Otherwise, permanent contacts in the aor type sections would have
- a config line order dependancy. * Fixed off nominal path contact
- ref leak in qualify_contact(). The comment saying the unref is
- not needed was wrong. * Fixed off nominal path use of the
- endpoint parameter if it is NULL in send_out_of_dialog_request().
- * Added missing off nominal path unref of pjsip tdata in
- send_out_of_dialog_request(). * Fixed off nominal path failing to
- call the callback in send_request_cb() when the request is
- challenged for authentication. * Eliminated silly RAII_VAR() use
- in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen
- to better reflect reality. (closes issue ASTERISK-23254) Reported
- by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/
- ........ Merged revisions 411141 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 16:06 +0000 [r411092] Kinsey Moore <kmoore@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
- update_provisional_keepalive() is called while
- send_provisional_keepalive_full() is waiting on the PVT lock,
- then pvt->provisional_keepalive_sched_id will be changed to a new
- sched_id value by update_provisional_keepalive(), but that new
- sched_id then may be overwritten with -1 by
- send_provisional_keepalive_full(), killing the pvt's reference to
- a schedule and "leaking" the reference. (closes issue
- ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
- Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
- Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
- (license 5012) ........ Merged revisions 411088 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411089 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411091 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 15:56 +0000 [r411090] Jonathan Rose <jrose@digium.com>
-
- * /, res/res_stasis.c: ARI: Resolve a subscription leak against
- implicit bridge subscriptions When a channel in a stasis
- application is joined to a bridge, a subscription for that bridge
- is created implicitly for the stasis application serving the
- channel. Prior to this patch, subsequent removals of the channel
- from the bridge would leave the subscription open. Review:
- https://reviewboard.asterisk.org/r/3380/ ........ Merged
- revisions 411086 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-25 15:47 +0000 [r411073-411087] Richard Mudgett <rmudgett@digium.com>
-
- * utils/conf2ael.c, main/lock.c, utils/ael_main.c: Revert -r411073.
- It didn't help and blew up the system.
-
- * utils/ael_main.c, utils/conf2ael.c, main/lock.c: locking: Add
- temporary sanity checks. Add some temporary sanity checks to hunt
- for locking problems with the masquerade supertest.
-
-2014-03-24 21:39 +0000 [r411024] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Always use fromdomain if set
- for domain, even if callerid is set to restricted. (closes issue
- ASTERISK-20841) Reported by: Kelly Goedert ........ Merged
- revisions 411021 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 411022 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 411023 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-21 16:04 +0000 [r410996] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_pjsip_registrar.c: res_pjsip_registrar.c:
- Miscellaneous cleanup in rx_task(). * Fix variable shadowing of
- 'updated' by renaming it to 'contact_update'. * Checked
- 'contact_update' for ast_sorcery_copy() failure. * Removed silly
- use of RAII_VAR() for 'contact_update'. ........ Merged revisions
- 410995 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-21 15:50 +0000 [r410981-410994] Sean Bright <sean@malleable.com>
-
- * res/ael/ael.flex, utils/Makefile, pbx/pbx_ael.c,
- res/ael/ael_lex.c: Make the AEL load process less chatty.
- Switched a bunch of LOG_NOTICEs to ast_debug. This time without
- breaking the build.
-
- * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Revert
- r410981. aelparse blew up.
-
- * main/config.c: Remove a LOG_NOTICE from
- ast_config_engine_register. There is enough indication from the
- CLI that we are loading a realtime engine as it is.
-
- * pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.flex: Make the AEL
- load process less chatty. Switched a bunch of LOG_NOTICEs to
- ast_debug.
-
-2014-03-20 23:02 +0000 [r410967] Jonathan Rose <jrose@digium.com>
-
- * apps/app_confbridge.c, /: app_confbridge: Fix bug - users with
- startmuted set don't start muted (closes issue ASTERISK-23461)
- Reported by: Chico Manobela Review:
- https://reviewboard.asterisk.org/r/3373/ ........ Merged
- revisions 410965 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410966 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-20 16:35 +0000 [r410950] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/rtp_engine.h, main/dial.c, main/manager.c, /,
- main/channel_internal_api.c, main/core_unreal.c,
- include/asterisk/channel.h, res/ari/resource_channels.c,
- res/res_stasis_snoop.c: assigned-uniqueids: Miscellaneous cleanup
- and fixes. * Fix memory leak in ast_unreal_new_channels(). Made
- it generate the ;2 uniqueid on a stack variable instead of
- mallocing it. * Made send error response to ARI and AMI requests
- instead of just logging excessive uniqueid length and allowing
- truncation. action_originate() and
- ari_channels_handle_originate_with_id(). * Fixed minor truncating
- uniqueid hole when generating the ;2 uniqueid string length.
- Created public and internal lengths of uniqueid. The internal
- length can handle a max public uniqueid plus an appended ;2. *
- free() and ast_free() are NULL tolerant so they don't need a NULL
- test before calling. * Made use better struct initialization
- format instead of the position dependent initialization format.
- Also anything not explicitly initialized in the struct is
- initialized to zero by the compiler. * Made
- ast_channel_internal_set_fake_ids() use the safer
- ast_copy_string() instead of strncpy(). Review:
- https://reviewboard.asterisk.org/r/3371/ ........ Merged
- revisions 410949 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-19 17:27 +0000 [r410934] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_endpoint_identifier_ip.c: PJSIP: Allow for
- identify sections to be specified in sorcery.conf. "identify" is
- a special type of configuration object in PJSIP because unlike
- the other objects, it is not provided by the base res_pjsip
- module. Instead, it is provided by the
- res_pjsip_endpoint_identifier_ip module. If using the default
- sorcery wizard (config,criteria=type=identify) then things work
- because the module that applies the default wizard is the correct
- module. However, if attempting to use sorcery.conf to apply an
- alternate wizard, it was not possible. If you attempted to
- specify the identify object type in the res_pjsip section, then
- the object could not be registered since the object was
- undocumented for the res_pjsip module. There was no alternate
- configuration section defined for it, so you were out of luck if
- you wanted to override the default wizard. With this change, the
- identify section will properly have a sorcery.conf-based wizard
- applied when the identify definition is within the
- res_pjsip_endpoint_identifier_ip section. ........ Merged
- revisions 410933 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-19 14:25 +0000 [r410905-410919] Joshua Colp <jcolp@digium.com>
-
- * res/res_stasis.c, /: res_stasis: Fix a bug where the default
- bridge type was not set. ........ Merged revisions 410918 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * CHANGES, res/res_stasis.c, rest-api/api-docs/bridges.json, /,
- res/ari/resource_bridges.h: res_stasis: Extend bridge type to be
- a comma separated list of bridge attributes. This change turns
- the bridge type field into a comma separated list of attributes.
- These attributes include: mixing, holding, dtmf_events, and
- proxy_media. By setting the various attributes a user can control
- the type of bridge created with the behavior they need for their
- application. (closes issue ASTERISK-23437) Reported by: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/3359/ ........
- Merged revisions 410904 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-19 02:33 +0000 [r410891] Matthew Jordan <mjordan@digium.com>
-
- * res/res_ari.c, /: res_ari: Fix documentation schema error
- ........ Merged revisions 410890 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 23:32 +0000 [r410877] Rusty Newton <rnewton@digium.com>
-
- * res/res_ari.c, /: res_ari: Add notes about Asterisk HTTP server
- to the "enabled" config option for the res_ari general section
- Added note and see-also reminding user to enable the HTTP server.
- (closes issue ASTERISK-22499) Reported by: Rusty Newton ........
- Merged revisions 410876 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 15:45 +0000 [r410863] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/http.c: ARI: allow json content type with zero length
- body When a request was received with a Content-type of json, the
- body was sent for json parsing - even if it was zero length. This
- resulted in ARI requests failing that were valid, such as a
- channel DELETE with no parameters. The code has now been changed
- to skip json parsing with zero content length. (closes issue
- SWP-6748) Reported by: Samuel Galarneau Review:
- https://reviewboard.asterisk.org/r/3360/ ........ Merged
- revisions 410858 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 15:28 +0000 [r410862] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: cdr: Add asserts for when we don't know about a
- CDR for a channel In the CDR core, every channel should either be
- filtered out (due to being an 'internal' channel used as an
- implementation detail, such as playing media back into a bridge)
- or it should get a CDR. Even if that CDR ends up being discarded,
- we still give the channel a CDR in case we end up needing it. If
- we hit a situation where a channel does not have a CDR, we should
- blow up in -dev-mode. Asserts are appropriate for that. This
- patch adds those asserts, as they would have quickly caught the
- error fixed by r410814. ........ Merged revisions 410861 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 12:45 +0000 [r410845] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/config_system.c: res_pjsip: Fix memory leak of
- nameservers in off-nominal resolver creation failure. Thanks
- Walter Doekes! ........ Merged revisions 410844 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 11:52 +0000 [r410831] Sean Bright <sean@malleable.com>
-
- * res/res_fax_spandsp.c, /: res_fax_spandsp: Use g711_free() when
- available. Per Johann Steinwendtner on the asterisk-dev mailing
- list:
- http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
- g711_free() was introduced in spandsp 0.0.6pre4 and
- g711_release() became a noop. I opted not to remove the call to
- g711_release() since it is harmless and to call g711_free() if we
- have a sufficiently recent version of spandsp. (issue
- ASTERISK-20149) Reported by: Alexandr Gordeev ........ Merged
- revisions 410829 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410830 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-18 02:09 +0000 [r410814] Richard Mudgett <rmudgett@digium.com>
-
- * main/stasis_cache.c, /: stasis_cache: Use the right variable in
- the cache entry ao2 cmp function. ........ Merged revisions
- 410813 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-17 22:54 +0000 [r410794-410796] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/dns.h, CHANGES,
- res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
- main/dns.c, /, res/res_pjsip/config_system.c: res_pjsip: Enable
- PJSIP DNS client support. This change enables DNS client support
- within PJSIP. System nameservers are automatically discovered
- using res_init or res_ninit. If this fails then PJSIP will resort
- to using gethostbyname for resolution. By enabling this support
- we gain SRV support, failover, and weight support. (closes issue
- ASTERISK-23435) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3343/ ........ Merged
- revisions 410795 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Make address
- replacement less aggressive. This change makes the
- res_pjsip_multihomed module less aggressive when changing the
- address in messages. It will now only occur if the transport in
- use is bound to the any address OR if the system determined
- source address matches the bound address of the transport in use.
- Review: https://reviewboard.asterisk.org/r/3369/ ........ Merged
- revisions 410793 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-17 22:24 +0000 [r410775] Russ Meyerriecks <rmeyerreicks@digium.com>
-
- * /, main/callerid.c: callerid: Logic error in checksum processing
- Callerid checksum-ing was being handled incorrectly here. When
- the checksum is calculated to be 0x00, it will perform 0x100-0x00
- which results in 0x100. This value will then fail the otherwise
- correct callerid message. This patch changes the logic to simply
- add the calculated checksum to the transmitted 2's compliment
- checksum. Review: https://reviewboard.asterisk.org/r/3356/
- (closes issue ASTERISK-23488) ........ This is a merge of merged
- revisions 410750 410747 from
- http://svn.asterisk.org/svn/asterisk/branches/12 I didn't want a
- broken patch to be comitted to trunk so I pre-merge merged them.
-
-2014-03-17 19:35 +0000 [r410684-410699] Mark Michelson <mmichelson@digium.com>
-
- * res/res_mwi_external.c, res/res_pjsip/config_system.c,
- configs/sorcery.conf.sample, include/asterisk/sorcery.h,
- res/res_pjsip/pjsip_configuration.c, tests/test_sorcery_astdb.c,
- tests/test_sorcery.c, tests/test_sorcery_realtime.c,
- main/sorcery.c, /: Revert changes to sorcery that accidentally
- got committed. These changes were still up for review and have
- not been approved yet. I must have had the changes in my working
- copy when making a different change. ........ Merged revisions
- 410696 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * bridges/bridge_softmix.c, tests/test_sorcery.c, main/channel.c,
- res/res_pjsip/config_system.c, res/res_mwi_external.c,
- include/asterisk/bridge_channel.h, funcs/func_frame_trace.c,
- configs/sorcery.conf.sample, res/res_pjsip/pjsip_configuration.c,
- include/asterisk/sorcery.h, tests/test_sorcery_astdb.c,
- include/asterisk/frame.h, main/bridge_channel.c,
- tests/test_sorcery_realtime.c, main/sorcery.c,
- res/res_stasis_playback.c, main/frame.c, /: Fix stuck channel in
- ARI through the introduction of synchronous bridge actions.
- Playing back a file to a channel in an ARI bridge would attempt
- to wait until the playback concluded before returning. The method
- used involved signaling the waiting thread in the ARI custom
- playback function. The problem with this is that there were some
- corner cases that were not accounted for: * If a bridge channel
- could not be found, then we never would attempt the playback but
- would still attempt to wait for the playback to complete. * If
- the bridge playfile action failed to queue, we would still
- attempt to wait for the playback to complete. * If the bridge
- playfile action were queued but some circumstance caused the
- playback not to occur (the bridge dies, the channel is removed
- from the bridge), then we would never be notified. The solution
- to this is to move the waiting logic into the bridge code. A new
- bridge API function is added to queue a synchronous action on a
- bridge. The waiting thread is notified when the queued frame has
- been freed, either due to an error occurring or due to successful
- playback. As a failsafe, the waiting thread has a 10 minute
- timeout just in case there is a frame leak somewhere. Review:
- https://reviewboard.asterisk.org/r/3338 ........ Merged revisions
- 410673 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-17 16:48 +0000 [r410672] Richard Mudgett <rmudgett@digium.com>
-
- * /, apps/confbridge/conf_chan_announce.c: app_confbridge: Add
- missing destructor call to announcer channel destructor. ........
- Merged revisions 410671 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-16 20:27 +0000 [r410651] Matthew Jordan <mjordan@digium.com>
-
- * /, res/stasis/app.c: stasis/app.c: Add some extra debugging for
- subscription counts Events are sent to a connected ARI
- application based on the things that ARI application cares about.
- These subscriptions can be set up implicitly - such as when that
- ARI application creates a new object - or explicitly, via the
- application resource's subscription operations. Debugging *why*
- something was being sent to an application - or why something was
- not being sent to an application - was a bit tricky, as there was
- no debug information for the subscriptions. This patch adds some
- debug level 3 statements that show the subscription counts for
- applications. (Level 3 was chosen as it matches the verbose level
- 3 statements elsewhere) ........ Merged revisions 410650 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-15 15:24 +0000 [r410639] Russell Bryant <russell@russellbryant.com>
-
- * include/asterisk/framehook.h: framehook.h: Fix some doc typos.
- There were a number of instances in this header file where
- "function all" was intended to be "function call". This patch
- fixes that up.
-
-2014-03-14 21:56 +0000 [r410626] Mark Michelson <mmichelson@digium.com>
-
- * /, tests/test_sorcery_realtime.c: Fix failing realtime sorcery
- tests. The store realtime callback needs to return a positive
- value for sorcery to treat the store as a success. ........
- Merged revisions 410625 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 21:36 +0000 [r410624] Jonathan Rose <jrose@digium.com>
-
- * main/manager.c, /: manager: fix memory leak in manager_add_filter
- function (closes issue ASTERISK-23420) Reported by: Etienne
- Lessard Patches: manager_eventfilter_leak uploaded by Etienne
- Lessard (license 6394) ........ Merged revisions 410609 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410623 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 20:55 +0000 [r410591-410608] Mark Michelson <mmichelson@digium.com>
-
- * /, main/db.c: Remove an extra ast_cond_wait() that slipped
- through the patch. ........ Merged revisions 410606 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410607 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/config.c, res/res_sorcery_realtime.c: Handle the return
- values of realtime updates and stores more accurately. Realtime
- backends' update and store callbacks return the number of rows
- affected, or -1 if there was a failure. There were a couple of
- issues: * The config API was treating 0 as a successful return,
- and positive values as a failure. Now the config API treats
- anything >= 0 as a success. * res_sorcery_realtime was treating 0
- as a successful return from the store procedure, and any positive
- values as a failure. Now sorcery treats anything > 0 as a
- success. It still considers 0 a "failure" since there is no
- change to report to observers. Review:
- https://reviewboard.asterisk.org/r/3341 ........ Merged revisions
- 410592 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_mwi.c: Prevent conflicts regarding unsolicited
- and solicited MWI to an endpoint. If an endpoint is receiving
- unsolicited MWI for a mailbox and then attempts to subscribe to
- an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
- is rejected with a 500 response. Review:
- https://reviewboard.asterisk.org/r/3345 ........ Merged revisions
- 410590 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 17:56 +0000 [r410589] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, CHANGES: uniqueid: Update CHANGES to reflect new features Note
- the new features provided by uniqueid in the CHANGES file. (issue
- ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3316/
- ........ Merged revisions 410588 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 16:42 +0000 [r410575] Jonathan Rose <jrose@digium.com>
-
- * /, main/acl.c, res/res_pjsip/pjsip_configuration.c,
- contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py,
- CHANGES, res/res_pjsip/config_transport.c,
- include/asterisk/acl.h: PJSIP: TOS values should be represented
- as decimals in sorcery objects (closes issue ASTERISK-23235)
- Reported by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3324/ ........ Merged
- revisions 410574 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 16:19 +0000 [r410567] Mark Michelson <mmichelson@digium.com>
-
- * /, main/db.c: Prevent delayed astdb syncs. The syncing thread
- sleeps for a second before waiting to be told to attempt to sync
- again. If a signal were sent during this sleeping period, we
- would end up having to wait until the next sync signal occurred
- in order to sync up the astdb. This code rearrangement also
- ensures that any pending transactions will be synced prior to
- Asterisk shutting down. Patches: db_sync.patch by John Hardin
- (License #6512) ........ Merged revisions 410556 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410559 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 16:17 +0000 [r410560] Jonathan Rose <jrose@digium.com>
-
- * res/ari/resource_bridges.c, /: ARI/bridges: Forward
- Playback/Recording Started/Finished to bridge topic (closes issue
- ASTERISK-23444) Reported by: Ben Merrills Review:
- https://reviewboard.asterisk.org/r/3340/ ........ Merged
- revisions 410558 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-14 16:01 +0000 [r410542-410557] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/app.h, /, res/res_mwi_external.c, main/app.c:
- res_mwi_external: Clear the stasis cache entry when the external
- MWI is deleted. One of the things missing when external MWI
- support was added was the ability to clear the stasis cache entry
- of deleted external MWI mailboxes. Review:
- https://reviewboard.asterisk.org/r/3325/ ........ Merged
- revisions 410555 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/cdr.c: cdr.c: Add missing aow_unlock(cdr) in off nominal
- path of handle_dial_message(). * Trivial common code hoisting in
- handle_bridge_leave_message(). * Some whitespace fixing. ........
- Merged revisions 410541 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-13 19:33 +0000 [r410528] Kinsey Moore <kmoore@digium.com>
-
- * res/stasis/control.h, res/res_stasis.c, /, res/stasis/control.c:
- ARI: Ensure managing application receives ChannelEnteredBridge
- messages This fixes an issue where a Stasis application running
- over ARI and subscribed to ari/events could miss the
- ChannelEnteredBridge event because it did not subscribe to the
- new bridge fast enough. To accomplish this, it subscribes the
- application controlling the channel to the new bridge before
- adding it to that bridge which required the stasis_app_control
- structure to maintain a reference to the stasis_app. (closes
- issue ASTERISK-23295) Review:
- https://reviewboard.asterisk.org/r/3336/ ........ Merged
- revisions 410527 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-13 13:25 +0000 [r410511] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_multihomed.c, /: Multiple revisions 410509-410510
- ........ r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar
- 2014) | 2 lines res_pjsip_multihomed: Fix a bug where the 200 OK
- for a REGISTER would contain the wrong contact. ........ r410510
- | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines
- res_pjsip_multihomed: Remove change for testing fix. ........
- Merged revisions 410509-410510 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-12 19:06 +0000 [r410492-410494] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_musiconhold.c, main/channel.c, /: res_musiconhold.c:
- Generate MOH start/stop events whenever the MOH stream is
- started/stopped. * Made res_musiconhold.c always post the
- MusicOnHoldStart/MusicOnHoldStop events when it actually
- starts/stops the music streams. This allows the events to always
- happen when MOH starts/stops. The event posting code was moved to
- the MOH alloc/release routines. * Made channel_do_masquerade()
- stop any MOH on the original channel before masquerading so the
- original channel will get a stop event with correct information.
- * Cleaned up a couple odd codings in moh_files_alloc() and
- moh_alloc() dealing with the music state variable. (issue
- ASTERISK-23311) Reported by: Benjamin Keith Ford Review:
- https://reviewboard.asterisk.org/r/3306/ ........ Merged
- revisions 410493 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/confbridge/conf_state.c,
- apps/confbridge/conf_state_single.c,
- apps/confbridge/conf_state_inactive.c,
- apps/confbridge/conf_state_single_marked.c, /: app_confbridge:
- Make explicitly stop MOH if a user is kicked or hangs up while
- MOH is playing. When MOH is playing to a user in a conference and
- the user is kicked or hangs up from the conference then the AMI
- MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
- MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
- by: Benjamin Keith Ford Review:
- https://reviewboard.asterisk.org/r/3306/ ........ Merged
- revisions 410490 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410491 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-12 12:51 +0000 [r410452-410472] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_multihomed.c, /: res_pjsip_multihomed: Fix a bug
- where outgoing messages for TCP would go out using UDP. This
- change fixes a bug where the code which changes the transport did
- not check whether the message is going out over UDP or not before
- changing it. For TCP and TLS transports we don't need to change
- the transport as the correct one is already chosen. ........
- Merged revisions 410471 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_multihomed.c (added), /: res_pjsip_multihomed: Add
- module which places the correct address within messages. Due to
- how messages are handled within PJSIP it is not until a message
- is actually sent that the destination is reliably known. This
- means that the addresses placed within the message may not be of
- the interface the message is being sent out on. This module
- determines what interface a message is being sent on and updates
- the message to contain the correct address if applicable. This
- module was tested by myself in a virtualized environment with
- multiple interfaces and also by Kinsey Moore in the following
- configuration: Networks: * 10.24.16.0/21 ** hard phone ** default
- gateway * 10.24.64.0/21 ** softphone with pjsip-based stack
- Transport details: bind address: 0.0.0.0 protocol: UDP All
- endpoints were tested with explicitly configured transports and
- unconfigured transports. This was tested with inbound and
- outbound calls, both of which were experiencing detrimental
- effects from incorrect IP addresses in SIP messages. These
- effects were only experienced by the soft phone on the 10.24.64.0
- network since the messages to the hard phone on the 10.24.16.0
- network had the correct IP address. (closes issue ASTERISK-23020)
- Reported by: xrobau Review:
- https://reviewboard.asterisk.org/r/3102/ ........ Merged
- revisions 410451 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-10 17:21 +0000 [r410395] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
- of Cookie headers. Sending a HTTP request that is handled by
- Asterisk with a large number of Cookie headers could overflow the
- stack. Another vulnerability along similar lines is any HTTP
- request with a ridiculous number of headers in the request could
- exhaust system memory. (closes issue ASTERISK-23340) Reported by:
- Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
- Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
- 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 410381 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410383 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-10 16:33 +0000 [r410369] Scott Griepentrog <sgriepentrog@digium.com>
-
- * res/ari/resource_channels.c, main/manager.c, /: unqiueid: correct
- max uniqueid length test This patch adds null string test prior
- to checking for a max uniqueid value that was added in r410157.
- ........ Merged revisions 410368 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-10 13:30 +0000 [r410346] Kinsey Moore <kmoore@digium.com>
-
- * /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
- session timers request This change allows chan_sip to avoid
- creation of the channel and consumption of associated file
- descriptors altogether if the inbound request is going to be
- rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
- Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
- Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
- Corey Farrell (license 5909) ........ Merged revisions 410308
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 410311 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410329 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-10 12:53 +0000 [r410307] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/pjsip_options.c, res/res_pjsip.c: AST-2014-003:
- res_pjsip: When handling 401/407 responses don't assume a request
- will have an endpoint. This change removes the assumption that an
- outgoing request will always have an endpoint and makes the
- authenticate_qualify option work once again. (closes issue
- ASTERISK-23210) Reported by: Joshua Colp ........ Merged
- revisions 410306 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-08 16:50 +0000 [r410288] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c,
- res/res_pjsip_outbound_registration.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- include/asterisk/res_pjsip_cli.h, include/asterisk/sorcery.h,
- res/res_pjsip/pjsip_cli.c, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/config_transport.c, main/sorcery.c,
- include/asterisk/res_pjsip.h: pjsip_cli: Create pjsip show
- channel and contact, and general cli code cleanup. Created the
- 'pjsip show channel' and 'pjsip show contact' commands.
- Refactored out the hated ast_hashtab. Replaced with
- ao2_container. Cleaned up function naming. Internal only, no
- public name changes. Cleaned up whitespace and brace formatting
- in cli code. Changed some NULL checking from "if"s to
- ast_asserts. Fixed some register/unregister ordering to reduce
- deadlock potential. Fixed ast_sip_location_add_contact where the
- 'name' buffer was too short. Fixed some self-assignment issues in
- res_pjsip_outbound_registration. (closes issue ASTERISK-23276)
- Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged
- revisions 410287 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-08 15:45 +0000 [r410275] Matthew Jordan <mjordan@digium.com>
-
- * /, res/ari/resource_channels.c: resource_channels: Check if a
- passed in ID is NULL before checking its length Calling strlen on
- a NULL string is explosive. This patch checks whether or not the
- passed in string is NULL or zero length before checking to see if
- the string is too long. ........ Merged revisions 410274 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 22:56 +0000 [r410227] Corey Farrell <git@cfware.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
- unload_module and do_monitor Release monlock before calling
- pthread_join. This ensures do_monitor cannot freeze by locking
- monlock during module unload. (closes issue ASTERISK-21406)
- Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3284/ ........ Merged
- revisions 410224 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 410225 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410226 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 22:08 +0000 [r410212] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, include/asterisk/sorcery.h: sorcery: correct field register
- argument list This fixes mistakes I previously made in merging
- gtjoseph's changes with mine. ........ Merged revisions 410211
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 21:54 +0000 [r410208-410210] Matthew Jordan <mjordan@digium.com>
-
- * /, main/config_options.c: config_options: Display the see-also
- information for CLI config option help The config option help
- information has always parsed the <see-also> tags in the XML
- documentation. Unfortunately, it just never bothered displaying
- them on the CLI. With this patch, when you execute 'config show
- help [module] [obj] [option]', it will display what other options
- are useful to you. (closes issue ASTERISK-22008) Reported by:
- Richard Mudgett ........ Merged revisions 410209 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip.c, /: res_pjsip: Fix documentation for one touch
- recording see-also links The one touch recording options have
- several see-also links between the various configuration options.
- These were 'broken' by the snake casing of those options. This
- patch corrects the see-also links such that they reference the
- correct option names. ........ Merged revisions 410194 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 21:23 +0000 [r410207] Mark Michelson <mmichelson@digium.com>
-
- * main/sorcery.c, res/res_sorcery_realtime.c, /,
- include/asterisk/sorcery.h, tests/test_sorcery_realtime.c: Make
- res_sorcery_realtime filter unknown retrieved results. When
- retrieving data from a database or other realtime backend, it's
- quite possible to retrieve variables that Asterisk does not care
- about but that are legitimate to exist. Asterisk does not need to
- throw a hissy fit when these variables are encountered but rather
- just filter them out. Review:
- https://reviewboard.asterisk.org/r/3305 ........ Merged revisions
- 410187 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 21:11 +0000 [r410191] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/sorcery.c, /, include/asterisk/sorcery.h,
- res/res_pjsip/pjsip_configuration.c: pjsip: allow and disallow
- show same codecs In order to prevent confusion over the allow and
- disallow list of codecs being the same an option for registering
- a field as an alias is added. The alias field will be read from
- the configuration file, but afterwards is not listed as a known
- field. With disallow set as an alias, the CLI command pjsip show
- endpoint # will list the allow= field, but not the disallow
- field. (closes issue ASTERISK-23092) Review:
- https://reviewboard.asterisk.org/r/3193/ ........ Merged
- revisions 410190 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 20:41 +0000 [r410174-410185] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/devicestate.h, main/stasis_cache.c,
- main/stasis_message.c, /, tests/test_devicestate.c,
- include/asterisk/stasis.h, main/app.c, main/devicestate.c,
- tests/test_stasis.c: stasis cache: Enhance to keep track of an
- item from different entities. A stasis cache entry now contains
- more than a single message/snapshot. It contains
- messages/snapshots for the local entity as well as any remote
- entities that post to the cached item. In addition callbacks can
- be supplied when the cache is created to compute and post the
- aggregate message/snapshot representing all entities stored in
- the cache entry. * All stasis messages now have an eid to
- indicate what entity posted it. * The stasis cache enhancements
- allow device state to cache and aggregate the device states from
- local and remote entities in a single operation. The cached
- aggregate device state is available immediately after it is
- posted to the stasis bus. This improves performance by
- eliminating a cache dump and associated ao2 container traversals
- to calculate the aggregate state. (closes issue ASTERISK-23204)
- Reported by: Mark Michelson Review:
- https://reviewboard.asterisk.org/r/3281/ ........ Merged
- revisions 410184 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_cel.c, channels/sig_pri.c, channels/sig_ss7.c,
- include/asterisk/bridge.h, tests/test_cdr.c, channels/sig_pri.h,
- channels/chan_dahdi.c, channels/sig_ss7.h, /: uniqueid: Fix
- chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler
- errors. (issue ASTERISK-23120) ........ Merged revisions 410171
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 15:47 +0000 [r410158] Scott Griepentrog <sgriepentrog@digium.com>
-
- * tests/test_cdr.c, res/res_clioriginate.c, res/res_ari_bridges.c,
- tests/test_substitution.c, res/res_stasis_playback.c,
- channels/chan_multicast_rtp.c, apps/app_meetme.c, /,
- main/bridge_basic.c, include/asterisk/channel_internal.h,
- tests/test_app.c, apps/confbridge/conf_chan_record.c,
- main/core_unreal.c, channels/chan_gtalk.c,
- include/asterisk/stasis_app_playback.h,
- res/ari/resource_bridges.c, channels/chan_jingle.c,
- channels/chan_phone.c, pbx/pbx_spool.c,
- res/ari/resource_bridges.h, res/parking/parking_tests.c,
- channels/chan_motif.c, apps/app_confbridge.c,
- res/ari/resource_channels.c, include/asterisk/pbx.h,
- res/res_stasis.c, include/asterisk/bridge.h,
- apps/app_voicemail.c, res/ari/resource_channels.h,
- apps/app_dial.c, res/res_calendar_exchange.c,
- channels/chan_vpb.cc, apps/app_page.c, apps/app_chanisavail.c,
- include/asterisk/dial.h, main/core_local.c,
- res/parking/parking_bridge_features.c,
- tests/test_stasis_endpoints.c, res/parking/parking_bridge.c,
- channels/chan_skinny.c, include/asterisk/stasis_app_snoop.h,
- addons/chan_mobile.c, main/bridge_channel.c,
- channels/chan_pjsip.c, channels/chan_mgcp.c,
- channels/chan_unistim.c, main/pbx.c,
- res/res_calendar_icalendar.c, main/ccss.c,
- channels/chan_bridge_media.c, main/bridge.c,
- tests/test_stasis_channels.c, apps/app_bridgewait.c,
- apps/app_originate.c, res/res_calendar_caldav.c,
- include/asterisk/channel.h, res/parking/parking_applications.c,
- apps/app_followme.c, main/cel.c, apps/app_queue.c,
- res/res_ari_channels.c, res/res_calendar_ews.c,
- rest-api/api-docs/bridges.json, main/dial.c,
- channels/chan_dahdi.c, channels/chan_h323.c, tests/test_cel.c,
- rest-api/api-docs/channels.json,
- include/asterisk/bridge_internal.h,
- apps/confbridge/conf_chan_announce.c, res/res_calendar.c,
- include/asterisk/core_unreal.h, addons/chan_ooh323.c,
- res/stasis/control.c, channels/chan_sip.c,
- main/channel_internal_api.c, include/asterisk/stasis_app.h,
- res/res_stasis_snoop.c, channels/chan_console.c,
- channels/chan_iax2.c, channels/chan_oss.c, apps/app_agent_pool.c,
- main/channel.c, main/manager.c, channels/chan_misdn.c,
- tests/test_voicemail_api.c, channels/chan_alsa.c,
- channels/chan_nbs.c, main/message.c: uniqueid: channel linkedid,
- ami, ari object creation with id's Much needed was a way to
- assign id to objects on creation, and much change was necessary
- to accomplish it. Channel uniqueids and linkedids are split into
- separate string and creation time components without breaking
- linkedid propgation. This allowed the uniqueid to be specified by
- the user interface - and those values are now carried through to
- channel creation, adding the assignedids value to every function
- in the chain including the channel drivers. For local channels,
- the second channel can be specified or left to default to a ;2
- suffix of first. In ARI, bridge, playback, and snoop objects can
- also be created with a specified uniqueid. Along the way, the
- args order to allocating channels was fixed in chan_mgcp and
- chan_gtalk, and linkedid is no longer lost as masquerade occurs.
- (closes issue ASTERISK-23120) Review:
- https://reviewboard.asterisk.org/r/3191/ ........ Merged
- revisions 410157 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-07 05:04 +0000 [r410108] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Allow static realtime members
- to be qualified during module load. When a static realtime peer
- with qualify=yes is loaded, Asterisk will fail to send an OPTIONS
- request due to the lastms being equal to 0. This results in the
- peer being unable to receive calls from Asterisk because the
- status is permanently UNKNOWN. This patch allows an OPTIONS
- request to be sent during module load by ignoring the lastms
- value on startup only. Review:
- https://reviewboard.asterisk.org/r/3294/ (closes issue
- ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
- wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
- Peirce (license 6112) ........ Merged revisions 410105 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 410106 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410107 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 23:47 +0000 [r410092] Richard Mudgett <rmudgett@digium.com>
-
- * main/sorcery.c, /: sorcery.c: Fix off-nominal path ref and memory
- leak in ast_sorcery_objectset_json_create(). * Made exit a loop
- early on error in ast_sorcery_objectset_json_create(). * Removed
- some dead code in ast_sorcery_objectset_create2(). ........
- Merged revisions 410089 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 23:43 +0000 [r410091] Russell Bryant <russell@russellbryant.com>
-
- * /, res/res_musiconhold.c: moh: fix a refcount error with realtime
- MOH I observed a crash in res_musiconhold on an Asterisk 11
- system using realtime MOH. Investigation of the backtrace showed
- a corrupt mohclass, implying that it got destroyed before the
- code expected it to. I went looking for reference counting errors
- that could have caused this crash and this patch this result. It
- contains 2 changes. 1) Remove a usless block of code that was
- impossible to reach. There was even a comment indicating that it
- was impossible to reach. The conditional includes
- "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
- inside of an if block with the opposite check
- "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
- good reason to keep it around. 2) A similar block to #1 contained
- a reference counting error. It stores state->class in the local
- variable mohclass without increasing its reference count. The
- reference count on mohclass is decremented at the end of the
- function. This block of code probably very rarely runs, which
- would help explain why this system was working fine for many
- months before experiencing a crash. Review:
- https://reviewboard.asterisk.org/r/3282/ ........ Merged
- revisions 410043 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 410044 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 410090 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 22:39 +0000 [r410042] George Joseph <george.joseph@fairview5.com>
-
- * res/res_pjsip/config_auth.c, funcs/func_sorcery.c (added),
- res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c,
- main/bucket.c, res/res_pjsip_endpoint_identifier_ip.c,
- include/asterisk/config.h, include/asterisk/sorcery.h,
- res/res_pjsip/pjsip_configuration.c, res/res_pjsip_acl.c,
- CHANGES, tests/test_sorcery.c, res/res_pjsip/config_transport.c,
- main/config.c, main/sorcery.c: sorcery: Create AST_SORCERY
- dialplan function. This patch creates the AST_SORCERY dialplan
- function which allows someone to retrieve any value from a
- sorcery-based config file. It's similar to AST_CONFIG. The
- creation of the function itself was fairly straightforward but it
- required changes to the underlying sorcery infrastructure that
- rippled into individual sorcery objects. The changes stemmed from
- inconsistencies in how sorcery created ast_variable objectsets
- from sorcery objects and the inconsistency in how individual
- objects used that feature especially when it came to parameters
- that can be specified multiple times like contact in aor and
- match in identify. You can read more here...
- http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
- So, what this patch does, besides actually creating the
- AST_SORCERY function, is the following... * Creates
- ast_variable_list_append which is a helper to append one
- ast_variable list to another. * Modifies the
- ast_sorcery_object_field_register functions to accept the
- already-defined sorcery_fields_handler callback. * Modifies
- ast_sorcery_objectset_create to accept a parameter indicating
- return type preference...a single ast_variable with all values
- concatenated or an ast_variable list with multiple entries. Also
- fixed a few bugs. * Modifies individual sorcery object
- implementations to use the new function definition of the
- ast_sorcery_object_field_register functions. * Modifies
- location.c and res_pjsip_endpoint_identifier_ip.c to implement
- sorcery_fields_handler handlers so they return multiple
- occurrences as an ast_variable_list. * Added a whole bunch of
- tests to test_sorcery. (closes issue ASTERISK-22537) Review:
- http://reviewboard.asterisk.org/r/3254/
-
-2014-03-06 19:04 +0000 [r410029] Jonathan Rose <jrose@digium.com>
-
- * include/asterisk/acl.h, /, main/acl.c,
- res/res_pjsip/pjsip_configuration.c, UPGRADE.txt,
- contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py
- (added), res/res_pjsip/config_transport.c: pjsip configuration:
- Make transport TOS values consistent with endpoints Transport TOS
- values were interpreted as DSCP values without being documented
- as such. Endpoint TOS values (tos_audio/tos_video) behaved
- normally as TOS values have historically. This patch makes the
- transport TOS values behave as TOS values and makes all TOS
- values readable as string values (e.g. AF11). In addition,
- alembic scripts have been updated to use the proper field types
- for all TOS/COS values. (issue ASTERISK-23235) Reported by:
- George Joseph Review: https://reviewboard.asterisk.org/r/3304/
- ........ Merged revisions 410028 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 18:20 +0000 [r410027] Joshua Colp <jcolp@digium.com>
-
- * res/ari/resource_channels.c, CHANGES,
- res/ari/ari_model_validators.c,
- rest-api/api-docs/recordings.json, res/ari/resource_bridges.c,
- res/ari/ari_model_validators.h, /,
- include/asterisk/stasis_app_recording.h,
- res/res_stasis_recording.c: res_stasis_recording: Add a
- "target_uri" field to recording events. This change adds a
- target_uri field to the live recording object. It contains the
- URI of what is being recorded. (closes issue ASTERISK-23258)
- Reported by: Ben Merrills Review:
- https://reviewboard.asterisk.org/r/3299/ ........ Merged
- revisions 410025 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 15:58 +0000 [r410012] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_mwi.c, /: Don't attempt to link in an aggregate MWI
- subscription if an endpoint does not aggregate MWI. Attempting to
- link a NULL object into an ao2 container had been benign
- previously, but since enabling DO_CRASH in the testsuite, this is
- now causing a crash. It's better to be right here anyway.
- ........ Merged revisions 410011 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 02:22 +0000 [r409996] Matthew Jordan <mjordan@digium.com>
-
- * res/res_fax_spandsp.c, /: res_fax_spandsp: Fix crash when passing
- ulaw/alaw data to spandsp When acting as a T.38 fax gateway,
- res_fax_spandsp would at times cause a crash in libspandsp. This
- would occur when, during fax tone detection, a ulaw/alaw frame
- would be passed to modem_connect_tones_rx. That particular
- routine expects the data to be in slin format. This patch looks
- at the frame type and, if the data is ulaw/alaw, converts the
- format to slin before passing it to modem_connect_tones_rx.
- Review: https://reviewboard.asterisk.org/r/3296 (closes issue
- ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal
- Rybarik patches: spandsp_g711decode.diff uploaded by Michal
- Rybarik (license 6578) ........ Merged revisions 409990 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409991 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-06 00:33 +0000 [r409970-409977] Richard Mudgett <rmudgett@digium.com>
-
- * apps/confbridge/conf_state_multi.c,
- apps/confbridge/conf_state_inactive.c, /: app_confbridge: Remove
- some noop code. ........ Merged revisions 409976 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_musiconhold.c: res_musiconhold.c: Remove some
- unnecessary RAII_VAR() usage. * Made the moh_register() define
- use useful parameter names. ........ Merged revisions 409967 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 20:41 +0000 [r409904-409919] Kinsey Moore <kmoore@digium.com>
-
- * main/config.c, /: config: Fix inverted test The test of the
- result of the stat() call was inverted such that its output was
- only used if the call failed. This inverts the test so that the
- output of stat() is used correctly. This was causing full reloads
- on unchanged files. (closes issue ASTERISK-23383) Reported by:
- David Woolley ........ Merged revisions 409916 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409917 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409918 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Fix crash
- involving masquerade It is possible for a channel to be
- masqueraded out of a bridge which means it may no longer have RTP
- glue to check upon leaving said bridge. If this situation
- occurred (it's possible at least during dial and call pickup)
- then Asterisk would crash. This change makes sure the glue is
- checked before use. (closes issue AST-1290) Reported by: John
- Bigelow ........ Merged revisions 409900 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 18:51 +0000 [r409889] Richard Mudgett <rmudgett@digium.com>
-
- * contrib/ast-db-manage/cdr/versions,
- contrib/ast-db-manage/cdr/versions/210693f3123d_create_cdr_table.py,
- /,
- contrib/ast-db-manage/config/versions/28887f25a46f_create_queue_tables.py
- (added), contrib/ast-db-manage/cdr.ini.sample (added),
- contrib/ast-db-manage/cdr/env.py, contrib/ast-db-manage/cdr
- (added), contrib/ast-db-manage/cdr/script.py.mako: alembic: Add
- missing queue and CDR table creation scripts. * Added the queues
- and queue_members tables to the config alembic scripts. * Added
- the CDR table alembic creation script. The CDR table is more of
- an example for new setups since the actual table can be fully
- customized in cdr_adaptive_odbc.conf. (closes issue
- ASTERISK-23233) Reported by: jmls Review:
- https://reviewboard.asterisk.org/r/3227/ ........ Merged
- revisions 409885 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 18:47 +0000 [r409888] Mark Michelson <mmichelson@digium.com>
-
- * funcs/func_presencestate.c, /: Fix documentation for
- PRESENCE_STATE to properly illustrate how to create a presence
- hint. There was a missing comma. This was discovered by Dan
- Kaplan. ........ Merged revisions 409886 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409887 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 16:58 +0000 [r409836] David M. Lee <dlee@digium.com>
-
- * main/config.c, /, configure, include/asterisk/autoconfig.h.in,
- configure.ac: Corrected cross-platform stat nanosecond code When
- nanosecond time resolution was added for identifying config file
- changes, it didn't cover all of the myriad of ways that one might
- obtain nanosecond time resolution off of struct stat. Rather than
- complicate the #if even further figuring out one system from the
- next, this patch directly tests for the three struct members I
- know about today, and #ifdef's accordingly. Review:
- https://reviewboard.asterisk.org/r/3273/ ........ Merged
- revisions 409833 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409834 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409835 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 16:26 +0000 [r409831-409832] Moises Silva <moises.silva@gmail.com>
-
- * res/res_http_websocket.c: Fix res/res_http_websocket.c build
- failure in 32bit due to incorrect print format for uint64_t
-
- * res/res_http_websocket.c, /: Fix WebRTC over WSS not working
- Several fixes for the WebSockets implementation in
- res/res_http_websocket.c * Flush the websocket session FILE* as
- fwrite() may not actually guarantee sending the data to the
- network. If we do not flush, it seems that buffering on the SSL
- socket for outbound messages causes issues * Refactored
- ast_websocket_read to take into account that SSL file descriptors
- may be ready to read via fread() but poll() will not actually say
- so because the data was already read from the network buffers and
- is now in the libc buffers (closes issue ASTERISK-23099) (closes
- issue ASTERISK-21930) Review:
- https://reviewboard.asterisk.org/r/3248/ ........ Merged
- revisions 409681 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409697 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 12:06 +0000 [r409780] Sean Bright <sean@malleable.com>
-
- * contrib/scripts/astgenkey, contrib/scripts/astgenkey.8, /: Fix
- references to 'keys' CLI commands in astgenkey ........ Merged
- revisions 409777 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409778 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409779 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-05 06:17 +0000 [r409747] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * channels/chan_unistim.c: Add update_peer function to
- unistim_rtp_glue, improve other unistim_rtp_glue functions
- conforming to other channel drivers. Do not forget auto-detected
- and user-selected phone settings on 'unistim reload' ........
- Merged revisions 409705 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409745 from
- http://svn.asterisk.org/svn/asterisk/branches/11
-
-2014-03-05 01:05 +0000 [r409683] Richard Mudgett <rmudgett@digium.com>
-
- * /, include/asterisk/stasis_internal.h: stasis: Made
- internal_stasis_subscribe() prototype and definition match
- exactly. ........ Merged revisions 409682 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-04 19:34 +0000 [r409627] Michael L. Young <elgueromexicano@gmail.com>
-
- * funcs/func_audiohookinherit.c, /: func_audiohookinheritance:
- Check If A Channel Was Specified This patch prevents a crash when
- using the function audiohookinheritance without setting the
- channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal
- Tested by: Joel Vandal Patches:
- asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
- Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/3272/ ........ Merged
- revisions 409623 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409625 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409626 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-04 17:22 +0000 [r409587] Jonathan Rose <jrose@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio
- problems with hold/unhold when using ICE ICE sessions will now be
- restarted if sessions are changed to use new sets of remote
- candidates. (closes issue ASTERISK-22911) Reported by: Vytis
- Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/
- ........ Merged revisions 409565 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409570 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-04 16:55 +0000 [r409569] Kinsey Moore <kmoore@digium.com>
-
- * /, main/astobj2.c: AO2: Add an assert for bad objects This adds
- an assert that will only be active if Asterisk is compiled with
- DO_CRASH and allows the testsuite to fail tests that would
- otherwise require log file parsing. ........ Merged revisions
- 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 409567 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409568 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-04 14:55 +0000 [r409475] Sean Bright <sean@malleable.com>
-
- * /, channels/chan_sip.c: Minor whitespace change to 'sip show
- peers' output. (closes issue ASTERISK-23406) Reported by: ibercom
- Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom
- ........ Merged revisions 409472 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409473 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409474 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-03 19:44 +0000 [r409423] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_stasis_recording.c: res_stasis_recording: Fix memory
- leak of the absolute name. ........ Merged revisions 409422 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-03 02:08 +0000 [r409364] Matthew Jordan <mjordan@digium.com>
-
- * main/asterisk.c, /: doxygen: Tweak the link back to ye olde
- Digium website ........ Merged revisions 409361 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409362 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409363 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-02 17:03 +0000 [r409350] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a
- legal option of gcc. Unofficially gcc considers it to be
- equivalent of -O3. clang chalks on it, though. This commit sets
- the default optimization flag to be -O3, like gcc actually
- considered it. Review: https://reviewboard.asterisk.org/r/3280/
- ........ Merged revisions 409308 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409344 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409346 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-01 20:28 +0000 [r409288] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Set options
- (100rel, timers) on incoming sessions. This change passes options
- to the UAS creation function. This in turn sets up 100rel and
- session timer properties on the incoming session. Reported by
- Julian Russell on asterisk-users mailing list. ........ Merged
- revisions 409287 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-03-01 00:05 +0000 [r409257-409275] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/devicestate.c: devicestate.c: Simplified some logic in
- _ast_device_state(). ........ Merged revisions 409274 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_cache.c, /: stasis_cache.c: Remove some unnecessary
- RAII_VAR() usage. ........ Merged revisions 409272 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis.c, /: stasis.c: Misc code cleanups. * Remove some
- unnecessary RAII_VAR() usage. * Made the struct
- stasis_subscription ao2 object use the ao2 lock instead of a
- redundant join_lock in the struct for ast_cond_wait(). * Removed
- locks on some ao2 objects that don't need the lock. * Made the
- topic pool entries container use the ao2 template functions. *
- Add some missing allocation failure checks. * Add missing cleanup
- in off nominal path of dispatch_message(). ........ Merged
- revisions 409270 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip: Add precautionary p->owner
- checks. * Add precautionary p->owner checks in sip_hangup(),
- get_refer_info(), get_also_info(), and
- interpret_t38_parameters(). * Simplify some tangled logic in
- get_refer_info(), get_also_info(), and add_rpid(). * Removed some
- dead code in handle_request_invite(). (closes issue
- ASTERISK-23323) Reported by: Walter Doekes Patches:
- issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
- uploaded by wdoekes (modified)
- issueA23323-more_p_owner_checks-11.x.patch (license #5674)
- uploaded by wdoekes (modified)
- issueA23323-more_p_owner_checks-12.x.patch (license #5674)
- uploaded by wdoekes (modified)
- issueA23323-more_p_owner_checks-trunk.patch (license #5674)
- uploaded by wdoekes (modified) ........ Merged revisions 409207
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 409255 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409256 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-28 21:24 +0000 [r409237] Kinsey Moore <kmoore@digium.com>
-
- * apps/app_queue.c, /: app_queue: Fix documented AMI event name
- During the rewrite of AMI events to use the Stasis bus, the name
- of the QueueMemberPaused event was changed to QueueMemberPause.
- This corrects documentation to reflect that. ........ Merged
- revisions 409234 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-28 18:03 +0000 [r409159] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix crash in
- ast_channel_hangupcause_set(). * Fix crash in
- ast_channel_hangupcause_set() because p->owner not checked before
- calling. Regression introduced by the fix for ASTERISK-22621.
- (closes issue ASTERISK-23135) Reported by: OK (issue
- ASTERISK-23323) Reported by: Walter Doekes ........ Merged
- revisions 409156 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409157 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409158 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-27 19:54 +0000 [r409132] Jonathan Rose <jrose@digium.com>
-
- * res/res_rtp_asterisk.c, /: Multiple revisions 409129-409130
- ........ r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb
- 2014) | 15 lines res_rtp_asterisk: Fix checklist creating
- problems in ICE sessions Prior to this patch, local candidate
- lists including SRFLX would fail to start properly when building
- ICE candidate check lists. This patch fixes that problem by
- making sure that each SRFLX candidate is associated with the
- proper base address so that the check list can create matches
- properly. This patch was written by jcolp. The issue will be left
- open to await testing by the issue participants. (issue
- ASTERISK-23213) Reported by: Andrea Suisani Review:
- https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose
- | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines
- res_rtp_asterisk: correct build error from r409129 Accidentally
- placed a declaration below functional code (issue ASTERISK-23213)
- Reported by: Andrea Suisani Review:
- https://reviewboard.asterisk.org/r/3256/ ........ Merged
- revisions 409129-409130 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409131 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-27 16:26 +0000 [r409091] David M. Lee <dlee@digium.com>
-
- * utils/astman.c, /: Fix memory stomping bug in astman. This memset
- complained in dev mod on my Ubuntu box. The memset is both
- unnecessary and dangerous. At this point, m hasn't been
- initialized yet, so the memset will write off to whatever address
- happens to be on the stack at the time. ........ Merged revisions
- 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 409083 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409087 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-27 16:08 +0000 [r409055] Corey Farrell <git@cfware.com>
-
- * /, configs/res_fax.conf.sample: res_fax: Comment out default
- settings from res_fax.conf. Comment out many settings in
- res_fax.conf.sample. The defaults are set in res_fax.c, so
- setting the same value in sample config does nothing but make the
- sample config more fragile. (closes issue ASTERISK-23231)
- Reported by: David Brillert Review:
- https://reviewboard.asterisk.org/r/3261/ ........ Merged
- revisions 409052 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 409053 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 409054 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-27 12:29 +0000 [r409000] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Apply
- packetization rules on inbound SDP handling The setting
- 'use_ptime' is supposed to tell Asterisk to honour the ptime
- attribute in an offer, preferring it to whatever packetization
- preferences have been set internally. Currently, however,
- something rather quirky will happen: (1) The SDP answer will be
- constructed in create_outgoing_sdp_stream. This will use the
- preferences from the endpoint, such that the 200 OK response will
- add the packetization preferences from the endpoint, and not what
- was offered. (2) When the 200 response is issued,
- apply_negotiated_sdp_stream is called. This will call
- apply_packetization, which will use the ptime attribute from the
- offer internally. We end up telling the offerer to use the
- internal ptime attribute, but we end up using the offered ptime
- attribute. Hilarity ensues. This patch modifies the behaviour by
- calling apply_packetization from negotiate_incoming_sdp_stream,
- which is called prior to create_outgoing_sdp_stream. This causes
- the format preferences on the session's media object to be set to
- the inbound ptime value (if 'use_ptime' is enabled), such that
- the construction of the answer gets the right value immediately.
- Review: https://reviewboard.asterisk.org/r/3244/ ........ Merged
- revisions 408999 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-26 23:35 +0000 [r408984] Richard Mudgett <rmudgett@digium.com>
-
- * /, tests/test_stasis.c: test_stasis.c: Misc cleanups. * Make the
- consumer ao2 object use the ao2 lock instead of a redundant lock
- in the struct for ast_cond_wait(). * Fixed some curly brace
- placements. * Fixed use of malloc(0). malloc(0) has variant
- behavior. It is up to the implementation to determine if it
- returns NULL or a valid pointer that can be later passed to
- free(). ........ Merged revisions 408983 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-26 19:00 +0000 [r408971] Scott Griepentrog <sgriepentrog@digium.com>
-
- * channels/chan_pjsip.c, /: pjsip: avoid edge case potential crash
- in answer() When accidentally compiling against a wrong version
- of pjsip headers with a different pjsip_inv_session size, the
- invite_tsx structure could be null in the answer() function. This
- led to a crash because it attempted to send the session response
- with an uninitialized packet pointer. This patch presets packet
- to null and adds a diagnostic log message to explain why the call
- fails. Review: https://reviewboard.asterisk.org/r/3267/ ........
- Merged revisions 408970 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-26 17:04 +0000 [r408958] Joshua Colp <jcolp@digium.com>
-
- * res/res_ari.c, /: res_ari: Make some additional error responses
- consistent with the rest of the system. This change makes some
- error cases use ast_ari_response_error to construct their error
- responses instead of manually doing it. This ensures they are
- consistent with the other error responses. Based on the original
- patch as done by Paul Belanger on the associated review. Review:
- https://reviewboard.asterisk.org/r/2904/ ........ Merged
- revisions 408957 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-26 13:47 +0000 [r408942-408944] Kinsey Moore <kmoore@digium.com>
-
- * include/asterisk/res_pjsip_session.h, /: PJSIP: Fix some bad
- spacing ........ Merged revisions 408943 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_refer.c: PJSIP: Prevent crash if channel has
- gone away It is currently possible for an ast_sip_session to
- exist without an associated channel as is the case when a new
- invite is coming in or just after a hangup is issued on a
- chan_pjsip channel. Part of the attended transfer code assumed
- the channel would be non-NULL and used it as such causing a
- crash. This bug was exposed thanks to the attended transfer ARI
- test in the test suite. (closes issue ASTERISK-23287) Reported
- by: Matt Jordan ........ Merged revisions 408941 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-26 08:57 +0000 [r408932] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
-
- * channels/chan_unistim.c: Implement functions handling keypress,
- display icons and text for i2004 KEM support.
-
-2014-02-25 17:51 +0000 [r408881-408883] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_exten_state.c, /,
- res/res_pjsip_pidf_digium_body_supplement.c (added),
- include/asterisk/res_pjsip_body_generator_types.h:
- res_pjsip_exten_state: Presence for digium phones Added presence
- support for digium phones. Review:
- https://reviewboard.asterisk.org/r/3239/ ........ Merged
- revisions 408882 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_send_to_voicemail.c (added),
- res/res_pjsip_header_funcs.c: res_pjsip_send_to_voicemail:
- transferring to voicemail for digium phones Added the ability for
- transferring directly to voicemail on digium phones. Added a new
- module that checks for the presence of a custom header and/or
- diversion header within a sip REFER. If either is found and they
- specify a sending to voicemail action then variables are added to
- the channel allowing the user access to them in the dialplan.
- Dialplan can then be written that branches based upon these
- values allowing, for instace, for a single number to be used for
- dialing and/or accessing voicemail directly. Also fixed a problem
- where the PJSIP_HEADER function was allowing non pjsip channels
- through (checked to make sure it has the correct channel type
- before proceeding). Review:
- https://reviewboard.asterisk.org/r/3245/ ........ Merged
- revisions 408880 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-25 17:44 +0000 [r408879] Rusty Newton <rnewton@digium.com>
-
- * configs/voicemail.conf.sample, /: configs/voicemail.conf.sample -
- Make mailcmd sample text more explicit Made the wording a bit
- more explicit. Didn't really change the meaning. ........ Merged
- revisions 408876 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408877 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408878 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-22 23:31 +0000 [r408859] Matthew Jordan <mjordan@digium.com>
-
- * /, main/asterisk.c: main: Initialize dialplan providing core
- components prior to module pre-load It is possible to pre-load
- pbx_config. As a result, pbx_config - which will load and parse
- the dialplan - will attempt to use various dialplan components,
- such as device state providers and presence state providers,
- prior to them being initialized by the core. This would lead to a
- crash, as the components had not created their Stasis cache
- entries. This patch moves a number of core component
- initializations before the module pre-load. This guarantees that
- if someone does pre-load pbx_config - or other pbx modules - that
- the Stasis caches for the various core components are created.
- (closes issue ASTERISK-23320) Reported by: xrobau (closes issue
- ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy,
- Rusty Newton ........ Merged revisions 408855 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-22 18:01 +0000 [r408840] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c, /: ignore AST_CONTROL_PVT_CAUSE_CODE
- without any messages (closes issue ASTERISK-23336) Reported by:
- Alexander Semych ........ Merged revisions 408838 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408839 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-22 02:31 +0000 [r408788] Corey Farrell <git@cfware.com>
-
- * /, utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c:
- Remove extra defines of AST_PBX_MAX_STACK. * Ensure
- AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
- incorrect function parameters in utils/extconf.c. (closes issue
- ASTERISK-23141) Reported by: Maxim Review:
- https://reviewboard.asterisk.org/r/3241/ ........ Merged
- revisions 408785 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408786 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408787 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-21 18:37 +0000 [r408731] Kevin Harwell <kharwell@digium.com>
-
- * main/rtp_engine.c, /: rtp_engine: Dynamic payload change in rtp
- mapping not supported Asterisk didn't support the dynamic payload
- change in rtp mapping in the 200 OK response. Scenario: Asterisk
- sends the INVITE proposing alaw and telephone-event, it proposes
- rtpmap:101 for telephone-event. Peer responds with 2xx, it
- answers with alaw and telephone-event also, but it proposes a
- different rtpmap number (rtpmap:103) for telephone-event.
- Expected Behaviour: Asterisk should honour the rtpmapping in the
- response and send DTMF packets using 103 as payload type for
- DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload
- type 101. With this patch asterisk now supports changes that can
- occur in the rtp mapping in the response. (closes issue
- ASTERISK-23279) Reported by: NITESH BANSAL Review:
- https://reviewboard.asterisk.org/r/3225/ Patches:
- dynamic_payload_change.patch uploaded by nbansal (license 6418)
- ........ Merged revisions 408729 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408730 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-21 18:19 +0000 [r408712-408723] Richard Mudgett <rmudgett@digium.com>
-
- * main/manager.c, /: manager: Fix AMI Status action of a single
- channel. Fixed use of uninitialized ao2 container iterator in an
- off-nominal condition. Either a memory allocation error or the
- requested channel is an internal channel not exposed to the
- outside. ........ Merged revisions 408715 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/sorcery.c, res/ari/resource_endpoints.c, /,
- apps/app_meetme.c, res/res_fax.c, res/res_stasis_recording.c,
- main/stasis_channels.c, res/res_sorcery_astdb.c,
- include/asterisk/json.h: json: Fix off-nominal json ref counting
- issues. * Fixed off-nominal json ref counting issue with using
- the following API calls: ast_json_object_set() and
- ast_json_array_append(). * Fixed off-nominal error reporting in
- ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal
- json ref counting issues in report_receive_fax_status() and
- dial_to_json(). ........ Merged revisions 408713 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/json.c, /: json: Fix json API wrapper code for json library
- versions earlier than 2.3.0. * Fixed json ref counting issue with
- json API wrapper code for ast_json_object_update_existing() and
- ast_json_object_update_missing() when the json library is earlier
- than version 2.3.0. ........ Merged revisions 408711 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-21 16:49 +0000 [r408699] Corey Farrell <git@cfware.com>
-
- * channels/chan_sip.c: chan_sip: prevent add_route from adding
- empty header. Fix regression caused by ASTERISK-22582. Empty
- Route headers were added when the route had a single strict hop.
- (closes issue ASTERISK-23306) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3236/
-
-2014-02-21 16:27 +0000 [r408645-408652] Kevin Harwell <kharwell@digium.com>
-
- * main/rtp_engine.c, /: rtp_engine: Output mixup in
- ${CHANNEL(rtpqos,audio,all)} Fixed the output of
- CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
- (closes issue ASTERISK-23261) Reported by: rsw686 Patches:
- rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged
- revisions 408646 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408647 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408649 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/channel.c, /: channel.c: MOH is not working for transferee
- after attended transfer Updated the code to check to see if MOH
- is playing on the transferor and if so then start it on the
- channel that replaces it during a masquerade. Example scenario of
- the problem: Alice calls Bob and then Bob begins the attended
- transfer process into a queue. Upon going on hold Alice hears
- music and so does Bob once he is in the queue. Bob then transfers
- Alice into the queue and then music for Alice stops even though
- she should be hearing it since has now replaced Bob in the queue.
- The problem that was occurring is that once the channel was
- masqueraded the app (queues, confbridge, etc...) had no way of
- knowing that the channel had just been swapped out thus it did
- not start music for the present channel. Credit to Olle Johansson
- for pointing me in the right direction on this issue. (closes
- issue ASTERISK-19499) Reported by: Timo Teräs Review:
- https://reviewboard.asterisk.org/r/3226/ ........ Merged
- revisions 408642 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408643 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408644 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-21 10:45 +0000 [r408592] Alexandr Anikin <may@telecom-service.ru>
-
- * /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
- variables ........ Merged revisions 408589 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408590 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408591 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-21 00:50 +0000 [r408539] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, apps/app_chanspy.c: app_chanspy: Documentation Update To
- Clarify "x" Option When using the "x" option (specify a DTMF
- digit to exit the application), it is not obvious in the
- documentation that this only works when spying on a channel. If a
- channel being used to spy on other channels is waiting to connect
- to a channel or is no longer attached to a channel, the DTMF is
- ignored. As noted on the issue tracker, since there are
- workarounds available and this is a rarely used option we are
- opting for a documentation change here. (closes issue
- ASTERISK-22661) Reported by: Chris Hillman Patches:
- asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
- Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2990/ ........ Merged
- revisions 408536 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408537 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408538 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-20 21:12 +0000 [r408519-408523] George Joseph <george.joseph@fairview5.com>
-
- * /, res/res_pjsip/location.c,
- res/res_pjsip_outbound_registration.c: pjsip_cli: Add pjsip
- commands 'show registrations' and 'show contacts'. Added 'show
- registrations' and 'show contacts' to pjsip cli to make things a
- little more consistent. The output is exactly the same as the
- list command. Just needed to add entries to their respective
- ast_cli_entry structures. (closes issue ASTERISK-23275) Review:
- http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions
- 408522 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_cli.c, main/config.c: pjsip_cli: Fix
- memory leak in ast_sip_cli_print_sorcery_objectset. Fixed memory
- leaks in ast_sip_cli_print_sorcery_objectset and
- ast_variable_list_sort. (closes issue ASTERISK-23266) Review:
- http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions
- 408520 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/sorcery.h,
- res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c,
- tests/test_sorcery.c, main/sorcery.c, /,
- res/res_pjsip/config_system.c: sorcery: Create sorcery instance
- registry. In order to retrieve an arbitrary sorcery instance from
- a dialplan function (or any place else) there needs to be a
- registry of sorcery instances. ast_sorcery_init now creates a
- hashtab as a registry. ast_sorcery_open now checks the hashtab
- for an existing sorcery instance matching the caller's module
- name. If it finds one, it bumps the refcount and returns it. If
- not, it creates a new sorcery instance, adds it to the hashtab,
- then returns it. ast_sorcery_retrieve_by_module_name is a new
- function that does a hashtab lookup by module name. It can be
- called by the future dialplan function. res_pjsip/config_system
- needed a small change to share the main res_pjsip sorcery
- instance. tests/test_sorcery was updated to include a test for
- the registry. (closes issue ASTERISK-22537) Review:
- http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions
- 408518 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-20 19:02 +0000 [r408503] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip.c, /: res_pjsip: Update documentation for
- 'use_avpf' option When 'use_avpf' is set to True, inbound offers
- must use the AVPF/SAVPF RTP profile. However, when 'use_avpf' is
- set to False, Asterisk will accept both AVP/SAVP or AVPF/SAVPF
- RTP profiles in inbound offers. The documentation previously
- implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was
- set to False and a UA offered said profile in an INVITE request.
- ........ Merged revisions 408502 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-20 02:44 +0000 [r408450] Rusty Newton <rnewton@digium.com>
-
- * /, apps/app_queue.c: apps/app_queue - Fix incorrect Macro
- parameter documentation Macro is executed on the called channel,
- not the calling channel. (closes issue ASTERISK-23069) Reported
- By: Bryan Anderson ........ Merged revisions 408447 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408448 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408449 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-19 19:09 +0000 [r408386-408390] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/config.c: config: Add file size and nanosecond resolution
- fields to the cached modified config file information. Repeatedly
- modifying config files and reloading too fast sometimes fails to
- reload the configuration because the cached modification
- timestamp has one second resolution. * Added file size and
- nanosecond resolution fields to the cached config file
- modification timestamp information. Now if the file size changes
- or the file system supports nanosecond resolution the modified
- file has a better chance of being detected for reload. * Added a
- missing unlock in an off-nominal code path. (closes issue
- AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
- ........ Merged revisions 408387 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408388 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408389 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix regex
- handling and keep simple prefix matching performance. The sorcery
- astDB wizzard does not handle regex correctly if the pattern
- begins with an anchor character. This patch attempts to convert
- the anchored regex pattern to a prefix pattern supported by astDB
- for performance reasons. If it is not able to convert the pattern
- it falls back to getting all astDB members of the family and
- doing a normal regex pattern matching on the retrieved records.
- Review: https://reviewboard.asterisk.org/r/3161/ ........ Merged
- revisions 408385 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-19 12:04 +0000 [r408315-408332] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/ooh323c/src/ooCapability.c, /,
- addons/ooh323c/src/ooh245.c: process receiveAndTransmit user
- input remote caps instead of receive only send receiveAndTransmit
- user input our caps instead of receive only ........ Merged
- revisions 408328 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408330 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408331 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * addons/ooh323c/src/ooh323.c, /: Allow different socket and
- signalling ip on h.323 connection if gk mode is active Reported
- by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by:
- Gabriele Odone (closes issue ASTERISK-22738) ........ Merged
- revisions 408312 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408314 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-18 19:19 +0000 [r408299] Richard Mudgett <rmudgett@digium.com>
-
- * contrib/ast-db-manage/config/env.py,
- contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
- contrib/ast-db-manage/config,
- contrib/ast-db-manage/voicemail/env.py,
- contrib/ast-db-manage/voicemail,
- contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py,
- contrib/ast-db-manage/config/versions,
- contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py,
- contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py,
- contrib/ast-db-manage/voicemail/versions, contrib/ast-db-manage,
- /: alembic: Add svn:ignore *.pyc to directories and
- svn:executable to *.py files. ........ Merged revisions 408297
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-17 15:36 +0000 [r408272] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip/location.c, UPGRADE.txt, res/res_pjsip.c,
- res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h: Store
- SIP User-Agent information in contacts. When an endpoint sends a
- REGISTER request to Asterisk, we now will associate the
- User-Agent header with all contacts that were bound in that
- REGISTER request. ........ Merged revisions 408270 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-16 03:25 +0000 [r408199-408227] Matthew Jordan <mjordan@digium.com>
-
- * /, main/pbx.c: pbx: Handle a completely empty dialplan during a
- context merge It is highly unlikely, but - at least in Asterisk
- 12 - theoretically possible to load Asterisk with no dialplan
- whatsoever. If that occurs, and some other module (that is not a
- pbx module) attempts to merge its contexts into the dialplan, the
- existing merge routine will crash. This is because it is not
- insane, and rightly believes that you provided some sort of
- dialplan, somewhere. This patch will gracefully merge the
- contexts in such a case. Note that this is highly unlikely to
- occur in 1.8/11, as features will most likely provide some
- dialplan via parking. However, in Asterisk 12, parking is now
- provided by res_parking, and hence may create its dialplan later.
- (closes issue ASTERISK-23297) Reported by: CJ Oster Review:
- https://reviewboard.asterisk.org/r/3222 ........ Merged revisions
- 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 408201 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408220 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, Makefile: buildsystem: Unbreak the build (infloop) on Asterisk
- 11+ Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/
- ) broke the build. This patch fixes it by ignoring the .lastclean
- dependencies if the MENUSELECT_EMBED variable is not defined.
- patches: tmp.diff uploaded by wdoekes (License 5674) Review:
- https://reviewboard.asterisk.org/r/3228/ ........ Merged
- revisions 408193 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408194 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-14 21:44 +0000 [r408139-408141] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/stasis_endpoints.c, /: ARI: correct upper/lower case URI
- discrepancies URI's are supposed to be case sensitive and all
- lower case. In practice some portions of URI's in ARI are case
- insensitive and others are not, such as TECH, which in one
- instance would match a lower case name and in another would not.
- In this patch, the ast_endpoint_lastest_snapshot() function is
- modified to change the TECH portion to full upper case before
- lookup. This resolves the discrepancy noted by the reporter.
- However I chose to avoid forcing the /ari prefix of the URI's to
- be lower case for now. Except for the two cases here, all URI's
- should be lower case, unless they are part of a resource name or
- id. Review: https://reviewboard.asterisk.org/r/3211/ Reported by:
- Zane Conkle (closes issue ASTERISK-23125) ........ Merged
- revisions 408140 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/format.c, /: format.c: correct possible null pointer
- dereference In ast_format_sdp_parse and ast_format_sdp_generate
- the check checks for a valid interface and function were
- potentially confusing, and hid an error in the test of the
- presence of the function that is called later. This patch clears
- up and corrects the test. Review:
- https://reviewboard.asterisk.org/r/3208/ (closes issue
- ASTERISK-23098) Reported by: marcelloceschia Patches:
- main_format.patch uploaded by marcelloceschia (license 6036)
- ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
- ........ Merged revisions 408137 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408138 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-14 13:31 +0000 [r408086] Walter Doekes <walter+asterisk@wjd.nu>
-
- * Makefile, /: buildsystem: Don't force main to depend on
- everything else. Directory 'main' only needs to depend on
- embedded modules. If no module embedding is selected, the
- dependency is dropped. Review:
- https://reviewboard.asterisk.org/r/3212/ ........ Merged
- revisions 408083 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 408084 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 408085 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-14 12:41 +0000 [r408070] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER
- prior to calling bridge blind transfer This patch moves setting
- SIP_DEFER_BY_ON_TRANSFER prior to calling
- ast_bridge_transfer_blind. This prevents a BYE from being sent
- prior to the NOTIFY request that informs the transferor if the
- transfer succeeded or failed. This patch also clears said flag
- from the off nominal NOTIFY paths in the local_attended_transfer
- code, as once we've sent the NOTIFY request it is safe to send by
- the BYE request. This was caught by the
- blind-transfer-accountcode test in the Asterisk Test Suite.
- (closes issue ASTERISK-23290) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3214/ ........ Merged
- revisions 408069 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-14 08:52 +0000 [r408059] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * Makefile, build_tools/install_subst (added): install_subst:
- helper script for installing with path substitution A helper
- script to copy a source file substituting any
- __ASTERISK_<foo>_DIR__ with the content of $AST<foo>DIR. Review:
- https://reviewboard.asterisk.org/r/3202/
-
-2014-02-13 18:52 +0000 [r407990-408006] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pubsub.c, /, res/res_pjsip_mwi.c: Remove all PJSIP
- MWI-specific use from our MWI code. PJSIP has built-in MWI code
- that could be useful to some degree, but our utilization of the
- API actually made our code a bit more cluttered since we had to
- have special cases peppered throughout. With this change, we move
- to using the pjsip_evsub API instead, which streamlines the code
- by removing special cases. Review:
- https://reviewboard.asterisk.org/r/3205 ........ Merged revisions
- 408005 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/location.c: Fix crash in AMI PJSIPShowEndpoint
- action. If an AOR has no permanent contacts, then the
- permanent_contacts container is never allocated. This makes the
- code safe in the face of NULLs. I also changed the variable that
- counts contacts from "num" to "total_contacts" since there are
- now two variables that are indicate numbers of things. ........
- Merged revisions 407988 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-13 15:51 +0000 [r407989] Kinsey Moore <kmoore@digium.com>
-
- * main/logger.c, CHANGES: Logger: Add dynamic logger channels This
- adds the ability to dynamically add and remove logger channels
- from Asterisk via the CLI. (closes issue AST-1150) Review:
- https://reviewboard.asterisk.org/r/3185/
-
-2014-02-12 08:25 +0000 [r407970] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, main/config.c: realtime: Fix ast_update2_realtime() on
- raspberry pi. The old code depended on undefined va_arg
- behaviour: calling a function twice with the same va_list
- parameter and expecting it to continue where it left off. The
- changed code behaves like the manpage says it should. Also added
- a bunch of early returns to trap errors (e.g. OOM) instead of
- crashing. The problem was found by Julian Lyndon-Smith. The
- deviant behaviour on the raspberry PI also uncovered another bug
- (fixed in r407875) in the res_config_pgsql.so driver. Reported
- by: jmls Tested by: jmls Review:
- https://reviewboard.asterisk.org/r/3201/ ........ Merged
- revisions 407968 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-11 20:17 +0000 [r407958] Joshua Colp <jcolp@digium.com>
-
- * main/sched.c: scheduler: Remove hashtab usage. This is a first
- stab at tweaking the performance profile of the scheduler.
- Removing the hashtab usage removes an extra memory allocation
- when scheduling something and makes it so rescheduling does not
- incur any memory allocation at all. Review:
- https://reviewboard.asterisk.org/r/3199/
-
-2014-02-11 03:18 +0000 [r407940] Matthew Jordan <mjordan@digium.com>
-
- * res/ari/resource_channels.c, /: ari/resource_channels: Add
- channel variables earlier in the creation process This patch
- tweaks the behaviour of POST /channels with channel variables
- such that the variables are passed into the pbx.c routines that
- perform the origination. This allows the variables to be assigned
- to the newly created channels immediately upon their
- construction, as opposed to be assigned after the originate has
- completed. The upshot of this is that the variables are available
- on the channels if they execute in the dialplan, as opposed to
- only being available once the channels are answered. Review:
- https://reviewboard.asterisk.org/r/3183/ ........ Merged
- revisions 407937 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-10 18:28 +0000 [r407926] Corey Farrell <git@cfware.com>
-
- * channels/sip/include/reqresp_parser.h,
- channels/sip/include/route.h (added), channels/chan_sip.c,
- channels/sip/route.c (added), channels/sip/include/sip.h:
- chan_sip: Isolate code that manages struct sip_route. * Move
- route code to sip/route.c + sip/include/route.h * Rename
- functions to sip_route_* * Replace ad-hoc list code with macro's
- from linkedlists.h * Create sip_route_process_header() to
- processes Path and Record-Route headers (previously done with
- different code in build_route and build_path) * Add use of const
- where possible * Move struct uriparams, struct contact and
- contactliststruct from sip.h to reqresp_parser.h. sip/route.c
- uses reqresp_parser.h but not sip.h, this was a problem. These
- moved declares are not used outside of reqresp_parser. * While
- modifying reqprep() the lack of {} caused me trouble. I added
- them. * Code outside route.c treats sip_route as an opaque
- structure, using macro's or procedures for all access. (closes
- issue ASTERISK-22582) Reported by: Corey Farrell Review:
- https://reviewboard.asterisk.org/r/3173/
-
-2014-02-10 16:49 +0000 [r407876] Walter Doekes <walter+asterisk@wjd.nu>
-
- * res/res_config_pgsql.c, /: res_config_pgsql: Fix
- ast_update2_realtime calls. Fix so multiple updates from a single
- call works (add missing ','). Remove bogus ast_free's that
- weren't supposed to be there. Moved a few spaces for readability.
- Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged
- revisions 407873 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407874 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407875 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-10 16:01 +0000 [r407859] Kinsey Moore <kmoore@digium.com>
-
- * apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c,
- apps/confbridge/conf_state_empty.c,
- apps/confbridge/conf_config_parser.c,
- configs/confbridge.conf.sample, /,
- apps/confbridge/include/confbridge.h, UPGRADE.txt: ConfBridge:
- Correct prompt playback target Currently, when the first marked
- user enters the conference that contains waitmarked users, a
- prompt is played indicating that the user is being placed into
- the conference. Unfortunately, this prompt is played to the
- marked user and not the waitmarked users which is not very
- helpful. This patch changes that behavior to play a prompt
- stating "The conference will now begin" to the entire conference
- after adding and unmuting the waitmarked users since the design
- of confbridge is not conducive to playing a prompt to a subset of
- users in a conference in an asynchronous manner. (closes issue
- PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/
- Reported by: Steve Pitts ........ Merged revisions 407857 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407858 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 20:52 +0000 [r407767] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL
- checks to a routine already full of them. ........ Merged
- revisions 407764 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407765 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407766 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 20:17 +0000 [r407752] Matthew Jordan <mjordan@digium.com>
-
- * /, main/security_events.c: security_events: Fix assertion failure
- in dev-mode on optional IE parsing When formatting an optional
- IE, the value is, of course, optional. As such, it is entirely
- appropriate for ast_json_object_get to return NULL. If that
- occurs, we now simply skip the IE that was requested, as it was
- not provided by the entity that raised the event. Thanks to
- George Joseph (gtjoseph) for catching this and reporting it in
- #asterisk-dev ........ Merged revisions 407750 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 20:01 +0000 [r407749] Joshua Colp <jcolp@digium.com>
-
- * main/timing.c, res/res_timing_pthread.c, res/res_timing_dahdi.c,
- res/res_timing_timerfd.c, include/asterisk/timing.h,
- res/res_timing_kqueue.c: timing: Improve performance for most
- timing implementations. This change allows timing implementation
- data to be stored directly on the timer itself thus removing the
- requirement for many implementations to do a container lookup for
- the same information. This means that API calls into timing
- implementations can directly access the information they need
- instead of having to find it. Review:
- https://reviewboard.asterisk.org/r/3175/
-
-2014-02-07 19:40 +0000 [r407748] Matthew Jordan <mjordan@digium.com>
-
- * /, funcs/func_cdr.c: funcs/func_cdr: Handle empty time values
- when extracting parsed values When extracting timestamps that are
- parsed, time stamp values that are not set (time values of
- 0.000000) should not actually result in a parsed string. The
- value should be skipped, and the result of the CDR function
- should be an empty string. Prior to this patch, the result was
- fed to the time formatting, which would result in an output of a
- date/time in 1969. ........ Merged revisions 407747 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 18:29 +0000 [r407731] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_iax2.c, include/asterisk/frame.h,
- configs/iax.conf.sample, /: chan_iax2: Block unnecessary control
- frames to/from the wire. Establishing an IAX2 call between
- Asterisk v1.4 and v1.8 (or later) results in an unexpected call
- disconnect. The problem happens because newer values in the enum
- ast_control_frame_type are not consistent between the branch
- versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
- using IAX2 2) v1.8 answers and sends a connected line update
- control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
- receives the control frame as an end-of-q (on v1.4
- AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
- receive queue becomes empty. Several things are done by this
- patch to fix the problem and attempt to prevent it from happening
- again in the future: * Added a warning at the definition of enum
- ast_control_frame_type about how to add new control frame values.
- * Made block sending and receiving control frames that have no
- reason to go over the wire. * Extended the connectedline iax.conf
- parameter to also include the redirecting information updates. *
- Updated the connectedline iax.conf parameter documentation to
- include a notice that the parameter must be "no" when the peer is
- an Asterisk v1.4 instance. (closes issue AST-1302) Review:
- https://reviewboard.asterisk.org/r/3174/ ........ Merged
- revisions 407678 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407727 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407729 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 16:47 +0000 [r407677] Matthew Jordan <mjordan@digium.com>
-
- * /, main/security_events.c: security_events: Fix error caused by
- DTD validation error The appdocsxml.dtd specifies that a
- "required" attribute in a parameter may have a value of yes, no,
- true, or false. On some systems, specifying "False" instead of
- "false" would cause a validation error. This patch fixes the
- casing to explicitly match the DTD. ........ Merged revisions
- 407676 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-07 13:15 +0000 [r407625] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * /, configs/indications.conf.sample: indications.conf: add stutter
- tone; end properly * If the "stutter" (voicemail indication) tone
- is indeed a stutter tone, and it ends with a constant tone, make
- sure that it is the dial tone. This was done for India (in),
- Mexico (mx) and the Philippines (ph). * If no "stutter" tone
- exists for a country, provide one. This was done for Spain (es),
- Malaysia (my) and Venezuela (ve). Review:
- https://reviewboard.asterisk.org/r/3158/ ........ Merged
- revisions 407622 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407623 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407624 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-06 21:24 +0000 [r407602] Matthew Jordan <mjordan@digium.com>
-
- * /, main/security_events.c, UPGRADE.txt, CHANGES: security_events:
- Add AMI documentation; output optional fields This patch adds
- documentation for the Security Events that are emited over AMI.
- It also notes these events in the UPGRADE/CHANGES file. ........
- Merged revisions 407589 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-06 19:58 +0000 [r407588] Rusty Newton <rnewton@digium.com>
-
- * /, configs/pjsip.conf.sample: configs/pjsip.conf.sample:
- Configuration section naming in pjsip.conf.sample needs a little
- clarification There is a bit of nuance to how you name things in
- pjsip.conf. This is a documentation patch to at least clear it up
- a little for users. Review:
- https://reviewboard.asterisk.org/r/3180/ ........ Merged
- revisions 407587 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-06 18:11 +0000 [r407574] Kevin Harwell <kharwell@digium.com>
-
- * /,
- contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py:
- pjsip realtime: already created enum failure for postgresql If an
- enum had been previously created the alembic script would attempt
- to re-create it and an error would be generated while running
- migrations for a postgresql server. The work around for this is
- to use the ENUM object type for postgres as opposed to the
- generic enum type used by sqlalchemy. Using this type in the
- script seems to work properly for both postgres and mysql.
- ........ Merged revisions 407572 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-06 17:55 +0000 [r407573] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_pjsip_logger.c,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
- include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- res/res_pjsip/config_auth.c, /, res/res_pjsip/location.c,
- res/res_pjsip_outbound_registration.c,
- res/res_pjsip_endpoint_identifier_ip.c,
- include/asterisk/res_pjsip_cli.h, res/res_pjsip/pjsip_cli.c,
- res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/config_domain_aliases.c: res_pjsip: Updates and
- adds more PJSIP CLI commands. * Adds identify, transport, and
- registration support to the PJSIP CLI. * Creates three additional
- callbacks, one for an iterator, one for a comparator, and one for
- a container. This eliminates the link dependency from higher
- level modules to lower level ones. * Eliminates duplicate sorting
- in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. *
- Pushes CLI command registration down to the implementing source
- file. * Adds several ast_sip_destroy_sorcery functions to
- complement existing ast_sip_sorcery_initialize functions. The
- destroy functions unregister PJSIP CLI commands and PJSIP CLI
- formatters. Reported by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3104/ ........ Merged
- revisions 407568 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 23:04 +0000 [r407514] Rusty Newton <rnewton@digium.com>
-
- * /, formats/format_wav.c: formats/format_wav: enhancing log
- message "Not a wav file" to be clear on what is supported
- Modifying the log message to be more specific as to what is
- supported. Specifically it seems format_wav supports only PCM
- encoded versions with a lower-case '.wav' extension. (closes
- issues ASTERISK-22310) Reported by: Jim Credland Review:
- https://reviewboard.asterisk.org/r/3188/ ........ Merged
- revisions 407511 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407512 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407513 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 20:56 +0000 [r407462] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, /: CHANGES: Improved description of Name/Creator changes
- to bridge ARI, adds AMI The changes log was written with language
- that was a little too internal Asterisk specific, so it's been
- changed to be more in the frame of reference of an ARI user.
- Also, previously the AMI event changes were omitted from the
- change log as well as the ability to include a bridge name in the
- ARI post bridges command. ........ Merged revisions 407461 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 20:43 +0000 [r407459] Kinsey Moore <kmoore@digium.com>
-
- * main/logger.c, /: Logger: Fix handling of absolute paths This
- fixes path handling for log files so that an extra / is not
- appended to the file path when the path is absolute (begins with
- /). This would previously result in different but functionally
- equivalent paths in the output of 'logger show channels'.
- ........ Merged revisions 407455 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407456 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 19:42 +0000 [r407443] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip/config_global.c, /: res_pjsip: When no global type
- the debug option defaults to "yes" If the global section was not
- specified in pjsip.conf then the configuration object does not
- exist in sorcery so when retrieving "debug" option it would
- return NULL. Then the NULL result was passed to ast_false utils
- function which would return false because it wasn't set to some
- representation of false, thus enabling sip debug logging. Made it
- so if the global config object does not exist then it will return
- a default of "no" for sip debugging. (issue ASTERISK-23038)
- Reported by: Rusty Newton ........ Merged revisions 407442 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 17:42 +0000 [r407422-407425] Jonathan Rose <jrose@digium.com>
-
- * CHANGES: CHANGES: Update changes log to include r403414 entry
- Adds note of additional 0 for operator option on app_record
-
- * CHANGES, /: CHANGES: Update changes log to include new bridge
- fields added in r404042 ........ Merged revisions 407419 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-05 15:29 +0000 [r407407] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/playbacks.json, UPGRADE.txt,
- rest-api/api-docs/sounds.json, rest-api/resources.json, CHANGES,
- include/asterisk/manager.h, rest-api/api-docs/bridges.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/mailboxes.json,
- rest-api/api-docs/asterisk.json,
- rest-api/api-docs/applications.json,
- rest-api/api-docs/channels.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
- /: ARI/AMI: Update versions; update UPGRADE/CHANGES notes for
- 12.1.0 changes Due to backwards compatible changes made to
- AMI/ARI, the version needs to be bumped to 1.1.0/2.1.0,
- respectively. ........ Merged revisions 407402 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-04 20:15 +0000 [r407275-407340] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/devicestate.h, /, main/devicestate.c:
- devicestate: Make ast_devstate_changed_literal() return value and
- doxygen consistent. Nothing actually cares about the value
- anyway. (closes issue ASTERISK-23178) Reported by: Jonathan Rose
- ........ Merged revisions 407337 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407338 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407339 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix assertion
- for pjsip.conf authorization list options. (closes issue
- ASTERISK-23168) Reported by: George Joseph Review:
- https://reviewboard.asterisk.org/r/3143/ ........ Merged
- revisions 407324 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * configs/sip.conf.sample, main/tcptls.c, /: tcptls.c: Made TLS
- handle a certificate chain file. Thanks to Guillaume Martres for
- doing the necessary research to validate the change. (closes
- issue ASTERISK-17727) Reported by: LN Patches:
- use_certificate_chain.patch (license #5864) patch uploaded by st
- documente_certificate_chain.patch (license #6576) patch uploaded
- by Guillaume Martres ........ Merged revisions 407272 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407273 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407274 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-04 16:55 +0000 [r407260] Matthew Jordan <mjordan@digium.com>
-
- * /, funcs/func_cdr.c: funcs/func_cdr: Fix non-epoch timestamps
- broken by improper char array deref Thanks to snuffy for pointing
- this issue out and fixing it. (closes issue ASTERISK-23250)
- Reported by: snuffy patches: func_cdr-fix.diff uploaded by snuffy
- (License 5024) ........ Merged revisions 407259 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-04 02:22 +0000 [r407217] Joshua Colp <jcolp@digium.com>
-
- * res/res_clialiases.c, /: res_clialiases: Fix crash when reloading
- and re-aliasing an alias that is in use. The code assumed that
- unregistering the alias would always succeed while in practice
- this is not actually true. A common case is the "reload" command
- itself. If the cli_aliases.conf configuration file was changed
- and reload executed the command would fail to unregister and
- ultimately point to freed memory. The reload process now checks
- whether unregistering succeeded or not and if not the old CLI
- alias is retained. (closes issue ASTERISK-19773) Reported by:
- Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth
- Blades ........ Merged revisions 407205 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407210 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407213 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-04 02:07 +0000 [r407198] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny - Fix deadlock when pickup of
- no call. Locking issues in skinny when picking up a call that
- doesn't exist. Cleaned up sub locking by fully removing and using
- the chan lock instead. Also changed ast_call_pickup to check
- whether chan was masq'd. (closes issue ASTERISK-23249) Reported
- by: wedhorn Tested by: snuffy, myself Patches:
- skinny-locking01.diff uploaded by wedhorn (license 5019) ........
- Merged revisions 407197 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-03 01:31 +0000 [r407169] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: cdrs: Check for applications to lock onto during
- dial begin handling This patch brings CDR processing further in
- line with r407085. During some dial operations, the application
- would not be locked to the Dial application and would instead
- continue to show the previously known application. In particular,
- this would occur when a Parked call would time out. This was due
- to a previous snapshot already locking the application to Park -
- processing this in a Dial Begin allows the Dial application to
- reassert its rightful place. (CDRs. Ugh.) But hooray for the
- Parked Call tests for catching this in the Asterisk Test Suite.
- ........ Merged revisions 407166 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-01 16:26 +0000 [r407154] Joshua Colp <jcolp@digium.com>
-
- * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
- res/stasis/app.c, res/ari/ari_model_validators.c,
- res/res_stasis.c, main/stasis_bridges.c: res_stasis: Enable
- transfers and provide events when they occur. This change enables
- transfers within ARI created bridges and adds events for when
- they occur. Unlike other events these will be received if *any*
- subscribed object is involved in the transfer. (closes issue
- ASTERISK-22984) Reported by: David M. Lee Review:
- https://reviewboard.asterisk.org/r/3120/ ........ Merged
- revisions 407153 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-02-01 00:25 +0000 [r407105] Corey Farrell <git@cfware.com>
-
- * apps/app_stack.c, /: app_stack: protect against missing
- parameters to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2
- parameters and LOCAL_PEEK requires 1 parameter. This protects
- against situations where those parameters are blank or missing by
- logging an error and returning. (closes issue ASTERISK-23220)
- Reported by: James Sharp ........ Merged revisions 407100 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 407103 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407104 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 23:40 +0000 [r407083-407085] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_dial.c, main/cdr.c, main/pbx.c, /, main/bridge_after.c,
- UPGRADE.txt, main/manager_channels.c: CDRs: fix a variety of dial
- status problems, h/hangup handler creating CDRs This patch fixes
- a number of small-ish problems that were noticed when witnessing
- the records that the FreePBX dialplan produces: (1) Mid-call
- events (as well as privacy options) have the ability to change
- the overall state of the Dial operation after the called party
- answers. This means that publishing the DialEnd event when the
- called party is premature; we have to wait for the execution of
- these subroutines to complete before we can signal the overall
- status of the DialEnd. This patch moves that publication and adds
- handlers for the mid-call events. (2) The AST_FLAG_OUTGOING
- channel flag is cleared if an after bridge goto datastore is
- detected. This flag was preventing CDRs from being recorded for
- all outbound channels that had a 'continue' option enabled on
- them by the Dial application. (3) The CDR engine now locks the
- 'Dial' application as being the CDR application if it detects
- that the current CDR has entered that app. This is similar to the
- logic that is done for Parking. In general, if we entered into
- Dial, then we want that CDR to record the application as such -
- this prevents pre-dial handlers, mid-call handlers, and other
- shenaniganry from changing the application value. (4) The CDR
- engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more
- places to determine if the channel is in hangup logic or dead. In
- either case, we don't want to record changes in the channel. (5)
- The default option for "endbeforehexten" has been changed to
- "yes". In general, you don't want to see CDRs in the 'h' exten or
- in hangup logic. Since the semantics of that option changed in
- 12, it made sense to update the default value as well. (6)
- Finally, because we now have the ability to synchronize on the
- messages published to the CDR topic, on shutdown the CDR engine
- will now synchronize to the messages currently in flight. This
- helps to ensure that all in-flight CDRs are written before
- shutting down. (closes issue ASTERISK-23164) Reported by: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/3154 ........
- Merged revisions 407084 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_dial.c, /: app_dial: Allow macro/gosub pre-bridge
- execution to occur on priorities The parsing for the destination
- of the macro/gosub uses the '^' character to separate out
- context, extension, and priority. However, the logic for the
- macro/gosub execution was written such that it would only do the
- actual macro/gosub jump if a '^' character existed. This doesn't
- apply when the macro/gosub jump occurs in a priority/priority
- label. This patch changes the logic so that the parsing still
- occurs, but the jump will occur even for priorities/priority
- labels. (issue ASTERISK-23164) Review:
- https://reviewboard.asterisk.org/r/3154 ........ Merged revisions
- 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 407074 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 407082 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 23:15 +0000 [r407035-407037] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_logger.c, CHANGES, res/res_pjsip.c,
- include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c,
- contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py
- (added), /, configs/pjsip.conf.sample, UPGRADE.txt: res_pjsip:
- Config option to enable PJSIP logger at load time. Added a
- "debug" configuration option for res_pjsip that when set to "yes"
- enables SIP messages to be logged. It is specified under the
- "system" type. Also added an alembic script to add the option to
- realtime. (closes issue ASTERISK-23038) Reported by: Rusty Newton
- Review: https://reviewboard.asterisk.org/r/3148/ ........ Merged
- revisions 407036 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_exten_state.c, /: res_pjsip_exten_state: Exporting
- global symbols caused load order issues Removed the exportation
- of global symbols from the module as it is no longer needed and
- it could potentially cause load problems as on some systems it
- would try to load before res_pjsip_pubsub ........ Merged
- revisions 407034 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 23:04 +0000 [r407033] Richard Mudgett <rmudgett@digium.com>
-
- * CHANGES, apps/app_chanspy.c: ChanSpy: Add ability to specify
- channel uniqueids as well as channel names. * Made ChanSpy accept
- a channel uniqueid or a fully specified channel name as the
- chanprefix parameter if the 'u' option is specified. (closes
- issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/
-
-2014-01-31 22:39 +0000 [r407030-407032] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/res_pjsip_presence_xml.h (added), /: Add file
- that apparently got missed in the merge. ........ Merged
- revisions 407031 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pidf_body_generator.c (added),
- include/asterisk/res_pjsip_exten_state.h (removed),
- res/res_pjsip_pubsub.exports.in, /,
- include/asterisk/res_pjsip_body_generator_types.h (added),
- res/res_pjsip_mwi.c, res/res_pjsip_xpidf_body_generator.c
- (added), res/res_pjsip_mwi_body_generator.c (added),
- res/res_pjsip_pubsub.c, res/res_pjsip_pidf.c (removed),
- res/res_pjsip_pidf_eyebeam_body_supplement.c (added),
- res/res_pjsip_exten_state.c, res/res_pjsip/presence_xml.c
- (added), include/asterisk/res_pjsip_pubsub.h: Decouple
- subscription handling from NOTIFY/PUBLISH body generation. When
- the PJSIP pubsub framework was created, subscription handlers
- were required to state what event they handled along with what
- body types they knew how to generate. While this serves well when
- implementing a base RFC, it has problems when trying to extend
- the body to support non-standard or proprietary body elements.
- The code also was NOTIFY-specific, meaning that when the time
- comes that we start writing code to send out PUBLISH requests
- with MWI or presence bodies, we would likely find ourselves
- duplicating code that had previously been written. This changeset
- introduces the concept of body generators and body supplements. A
- body generator is responsible for allocating a native structure
- for a given body type, providing the primary body content,
- converting the native structure to a string, and deallocating
- resources. A body supplement takes the primary body content (the
- native structure, not a string) generated by the body generator
- and adds nonstandard elements to the body. With these elements
- living in their own module, it becomes easy to extend our support
- for body types and to re-use resources when sending a PUBLISH
- request. Body generators and body supplements register themselves
- with the pubsub core, similar to how subscription and publish
- handlers had done. Now, subscription handlers do not need to know
- what type of body content they generate, but they still need to
- inform the pubsub core about what the default body type for a
- given event package is. The pubsub core keeps track of what body
- generators and body supplements have been registered. When a
- SUBSCRIBE arrives, the pubsub core will check that there is a
- subscription handler for the event in the SUBSCRIBE, then it will
- check that there is a body generator that can provide the content
- specified in the Accept header(s). Because of the nature of body
- generators and supplements, it means res_pjsip_exten_state and
- res_pjsip_mwi have been completely gutted. They no longer worry
- about body types, instead calling
- ast_sip_pubsub_generate_body_content() when they need to generate
- a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150
- ........ Merged revisions 407016 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 22:23 +0000 [r407015-407029] Kevin Harwell <kharwell@digium.com>
-
- * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py,
- contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py,
- /, UPGRADE.txt: alembic: script modifications due to errors A
- couple of the scripts had errors that would not allow a full
- migration to take place. The extensions table needed to make its
- 'id' column a primary key in order to work with mysql. The other
- script ...add_endpoints... was missing tables that it was trying
- to add columns to. Added the primary key on id for extensions and
- added the tables in for the missing pjsip configuration options.
- While it is not ideal to modify already released scripts this was
- a case where it had to be done due to errors in the script and
- lacking a better alternative. Review:
- https://reviewboard.asterisk.org/r/3167/ ........ Merged
- revisions 407019 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Subscribe fails when
- missing aor name When subscribing to MWI (res_pjsip_mwi) and the
- sip uri did not contain a name (ex: sip:<ip address>) then the
- subscription would fail since it would be unable to locate an
- associated aor. This patch makes it so that when a subscribe
- comes with no aor name then it will subscribe to all aors on the
- located endpoint. (closes issue ASTERISK-23072) Reported by: Bob
- M Review: https://reviewboard.asterisk.org/r/3164/ ........
- Merged revisions 407014 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 15:08 +0000 [r407001] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip_nat.c, /: PJSIP: Fix address for ACK in NAT
- situations In NAT scenarios where a call is placed to a
- Grandstream phone, res_pjsip will sometimes send the ACK to a 200
- OK to the private address of the device behind the NAT instead of
- the address of the NAT device. This corrects that behavior by
- rewriting the address in the Contact header in the incoming 200
- OK and the dialog's target address if necessary (since it has
- already been rewritten to the incorrect private address). (closes
- issue ASTERISK-23106) Review:
- https://reviewboard.asterisk.org/r/3168/ Reported by: Matt Jordan
- ........ Merged revisions 407000 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-31 05:31 +0000 [r406988] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny: fix up possible double unlock
- of chan. Return before chan is possibly unlocked a second time
- when hanging up a channel in SUBSTATE_OFFHOOK. ........ Merged
- revisions 406987 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-30 20:36 +0000 [r406936] Corey Farrell <git@cfware.com>
-
- * main/udptl.c, res/res_rtp_asterisk.c, /: res_rtp_asterisk &
- udptl: fix port selection to work with SELinux restrictions
- ast_bind to a port reserved for another program by SELinux causes
- errno == EACCES. This caused random failures when binding rtp or
- udptl sockets. Treat EACCES as a non-fatal error, try next port.
- (closes issue ASTERISK-23134) Reported by: Corey Farrell ........
- Merged revisions 406933 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406934 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406935 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-30 17:35 +0000 [r406920] Sean Bright <sean@malleable.com>
-
- * main/manager.c, /: Make a NOTICE about an invalid channel name
- more useful. ........ Merged revisions 406918 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406919 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-29 00:44 +0000 [r406863] Russell Bryant <russell@russellbryant.com>
-
- * /, configs/queues.conf.sample: queues.conf.sample Fix documented
- default for persistentmembers Closes issue ASTERISK-22662
- ........ Merged revisions 406860 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406861 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406862 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-28 23:40 +0000 [r406789-406848] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_pubsub.c, /: res_pjsip_pubsub: potential crash on
- timeout What seems to be happening is if a subscription has been
- terminated and the subscription timeout/expires is less than the
- time it takes for all pending transactions (currently on the
- subscription) to end then the subscription timer will not have
- been canceled yet and sub will be null. Since the subscription
- has already been canceled nothing needs to be done so a null
- check in the asterisk code is sufficient in working around this
- problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins
- ........ Merged revisions 406847 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * cdr/cdr_radius.c, cel/cel_radius.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: cdr_radius,
- cel_radius: build agains libfreeradius-client Asterisk's RADIUS
- module currently build against libradiusclient-ng, but this
- project has been superseeded by libfreeradius-client. The API is
- 99% compatible except that the header name has changed, the
- library name has changed, and the configuration file location has
- changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé
- Patches: freeradius-client.patch uploaded by sharky (license
- 6561) ........ Merged revisions 406801 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406802 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406803 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip/include/res_pjsip_private.h, /,
- include/asterisk/compat.h: res_pjsip,compat: INFINITY and NAN
- undefined On some systems the values for INFINITY and NAN are not
- defined thus causing a build error on those systems. Added
- definitions for those if they had not previously been defined.
- (closes issue ASTERISK-23056) Reported by: capouch Patches:
- inf-nan-patch.txt uploaded by capouch (license 6564) ........
- Merged revisions 406788 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-28 19:19 +0000 [r406778] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_stasis_device_state.c: ARI: Make double subscribe
- respond with success Currently, attempting to subscribe an
- application to a device state that it has already subscribed to
- will generate a 500 error response. This will now be treated as a
- subscription refresh even though ARI subscriptions don't
- currently support lifetimes and will respond with the normal
- response for a successful subscription (200 OK). (closes issue
- ASTERISK-23143) Reported by: Matt Jordan ........ Merged
- revisions 406775 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-28 16:43 +0000 [r406724] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/rtp_engine.c, /: rtp_engine: improved handling of
- get_rtp_info failure In ast_rtp_instance_make_compatible(), after
- a failure of channel tech call get_rtp_info() to return
- peer_instance, the null pointer would be passed to ao2_ref,
- producing an error that looked like a refernce counting problem
- but is not. This patch corrects that and adds helpful LOG_ERROR
- messages to indicate which failure path occurred. (issue
- AST-1276) Review: https://reviewboard.asterisk.org/r/3156/
- ........ Merged revisions 406721 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406722 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406723 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-28 00:20 +0000 [r406710] Richard Mudgett <rmudgett@digium.com>
-
- * /, tests/test_cel.c, tests/test_cdr.c: test_cdr.c, test_cel.c:
- Correctly destroy created bridges. * Fixed the
- test_cel_attended_transfer_bridges_link unit test to also account
- for the local channel link being destroyed now that the bridges
- are actually destroyed. * Made CDR unit test use its own version
- of do_sleep() from the CEL unit tests. ........ Merged revisions
- 406707 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-27 22:54 +0000 [r406647-406696] Kevin Harwell <kharwell@digium.com>
-
- * CHANGES: manager: ExtensionStatus event status human readable
- Added a note in the changes file about the new 'StatusText' field
- that was added to the 'ExtensionStatus' event. (issue
- ASTERISK-23154) Reported by: Jonathan Rose
-
- * main/manager.c: manager: ExtensionStatus event status human
- readable When an 'ExtensionStatus' event was raised it included
- the status as a numerical value, but did not include a text
- description of the status. Added a 'StatusText' field to the
- event which is a string representation of the extension status.
- Also added this to the 'Extension State' command response.
- (closes issue ASTERISK-23154) Reported by: Jonathan Rose
-
-2014-01-27 20:38 +0000 [r406646] Russell Bryant <russell@russellbryant.com>
-
- * main/config.c, /: Allow nested #includes in extconfig.conf
- extconfig.conf was hard-coded to not allow nested includes for
- some reason. The code has been this way since a patch was merged
- for ASTERISK-3333 (revision 4889), which was a significant update
- to this code ("Merge config updates"). I can't figure out any
- good reason why this should be limited. This patch just removes
- the limit and uses the default nesting depth limit. Closes issue
- ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/
- ........ Merged revisions 406643 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406644 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406645 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-27 08:17 +0000 [r406618] Walter Doekes <walter+asterisk@wjd.nu>
-
- * main/manager.c, UPGRADE.txt, configs/manager.conf.sample:
- manager: The eventfilter= option now takes an extended regex. In
- pre-trunk versions (...12) it accepts a basic regex, which is
- confusing because all other regexes in asterisk are of the
- extended kind. Review: https://reviewboard.asterisk.org/r/3147/
-
-2014-01-27 01:25 +0000 [r406595] Russell Bryant <russell@russellbryant.com>
-
- * main/file.c, include/asterisk/channel.h, main/channel.c, /:
- Protect ast_filestream object when on a channel The
- ast_filestream object gets tacked on to a channel via
- chan->timingdata. It's a reference counted object, but the
- reference count isn't used when putting it on a channel. It's
- theoretically possible for another thread to interfere with the
- channel while it's unlocked and cause the filestream to get
- destroyed. Use the astobj2 reference count to make sure that as
- long as this code path is holding on the ast_filestream and
- passing it into the file.c playback code, that it knows it's
- valid. Bug reported by Leif Madsen. Review:
- https://reviewboard.asterisk.org/r/3135/ ........ Merged
- revisions 406566 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406567 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406574 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-26 23:04 +0000 [r406517] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/tcptls.c: tcptls.c: Add missing cleanup on off nominal
- path. ........ Merged revisions 406514 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406515 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406516 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-26 14:19 +0000 [r406503] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * contrib/scripts/live_ast: live_ast: run wrapped programs with
- exec live_ast can be used as a wrapper script to run asterisk,
- gdb or valgrind. In those cases it runs them and returns the
- result. It is more useful to use 'exec' to avoid having another
- odd process in the chain. Review:
- https://reviewboard.asterisk.org/r/3110/
-
-2014-01-26 02:11 +0000 [r406490] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Be less strict
- with core requested outgoing capabilities. The core may
- (depending on circumstances) request a single codec on outgoing
- calls. Many channel drivers ignore or treat this as a suggestion
- while still including configured codecs. The res_pjsip_session
- logic treated this as an explicit request, leaving out other
- configured codecs. This change makes res_pjsip_session behave
- like other channel driver and simply adds the requested codec to
- the list. (closes issue ASTERISK-23082) Reported by: xrobau
- Review: https://reviewboard.asterisk.org/r/3140/ ........ Merged
- revisions 406489 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-24 23:33 +0000 [r406466] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/cel.c: CEL: Protect data structures during reload and
- shutdown. The CEL data structures need to be protected during a
- configuration reload and shutdown. Asterisk crashed during a
- shutdown because CEL events were still in flight and the CEL data
- structures were already destroyed. * Protected the cel_backends,
- cel_dialstatus_store, and cel_linkedids ao2 containers with a
- global ao2 object wrapper. * Added NULL checks before use of the
- cel_backends, cel_dialstatus_store, and cel_linkedids ao2
- containers in case the CEL module is already shutdown. * Fixed
- overloading of the cel_linkedids held objects reference count.
- During shutdown any held objects would be leaked. * Fixed memory
- leak of cel_linkedids held objects if the LINKEDID_END is not
- being tracked. The objects in the cel_linkedids container were
- not removed if the LINKEDID_END event is not used. * Added access
- protection to the cel_backends container during the CLI "cel show
- status" command. * Made cel_backends, cel_dialstatus_store, and
- cel_linkedids use the standard ao2 callback templates for the
- hash and cmp functions. * Eliminated unnecessary uses of
- RAII_VAR(). * Made ast_cel_engine_init() cleanup alocated
- resources on failure. (closes issue AST-1253) Reported by:
- Guenther Kelleter Review:
- https://reviewboard.asterisk.org/r/3128/ ........ Merged
- revisions 406417 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406418 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406465 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-24 22:34 +0000 [r406416] Jonathan Rose <jrose@digium.com>
-
- * main/utils.c, CHANGES: Thread Debugging: Add LWP to core show
- locks output This patch adds the LWP to core show locks output if
- it is available. Review: https://reviewboard.asterisk.org/r/3142/
-
-2014-01-24 22:18 +0000 [r406407] Richard Mudgett <rmudgett@digium.com>
-
- * main/manager.c, /: manager: Register atexit shutdown routine only
- once. * Made register atexit shutdown routine only once in
- __init_manager(). * Fixed some initial load failure conditions in
- __init_manager(). * Made reset options to defaults on reload when
- the reload will actually happen. * Removed unnecessary container
- traversals of the white/black filters during manager_free_user().
- * ast_free() does not need a NULL check before calling. ........
- Merged revisions 406359 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406400 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406401 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-24 21:46 +0000 [r406399] Jonathan Rose <jrose@digium.com>
-
- * res/res_config_pgsql.c, /: res_config_pgsql: Fix a memory leak
- and use RAII_VAR for cleanup when practical Review:
- https://reviewboard.asterisk.org/r/3141/ ........ Merged
- revisions 406360 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406361 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406389 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-24 18:13 +0000 [r406343] Richard Mudgett <rmudgett@digium.com>
-
- * main/manager.c, /: manager: Protect data structures during
- shutdown. Occasionally, the manager module would get an
- "INTERNAL_OBJ: bad magic number" error on a "core restart
- gracefully" command if an AMI connection is established. * Added
- ao2_global_obj protection to the sessions global container. *
- Fixed the order of unreferencing a session object in
- session_destroy(). * Removed unnecessary container traversals of
- the white/black filters during session_destructor(). (closes
- issue AST-1242) Reported by: Guenther Kelleter Review:
- https://reviewboard.asterisk.org/r/3144/ ........ Merged
- revisions 406341 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406342 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-23 23:43 +0000 [r406328] Mark Michelson <mmichelson@digium.com>
-
- * /: Today is not my day for writing code that compiles. ........
- Merged revisions 406327 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-23 22:56 +0000 [r406312] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, addons/res_config_mysql.c: res_config_mysql: Fix Setting The
- Column Name Incorrectly When support for a realtime sorcery
- module was added in revision 386731, the wrong property was
- accidentally used for setting the column name to be updated in
- the database table. This patch fixes the typo. (closes issue
- ASTERISK-23177) Reported by: Denis Tested by: Denis Patches:
- asterisk-23177-use-field-name.diff by Michael L. Young (license
- 5026) ........ Merged revisions 406311 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-23 21:18 +0000 [r406298] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pidf.c, /: Multiple revisions 406294-406295
- ........ r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu,
- 23 Jan 2014) | 11 lines Fix presence body errors found during
- testing: * PIDF bodies were reporting an "open" state in many
- cases where it should have been reporting "closed" * XPIDF bodies
- had XML nodes placed incorrectly within the hierarchy. * SIP URIs
- in XPIDF bodies did not go through XML sanitization * XML
- sanitization had some errors: * Right angle bracket was being
- replaced with "&rt;" instead of ">" * Double quote,
- apostrophe, and ampersand were not being escaped. ........
- r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan
- 2014) | 11 lines Fix presence body errors found during testing: *
- PIDF bodies were reporting an "open" state in many cases where it
- should have been reporting "closed" * XPIDF bodies had XML nodes
- placed incorrectly within the hierarchy. * SIP URIs in XPIDF
- bodies did not go through XML sanitization * XML sanitization had
- some errors: * Right angle bracket was being replaced with "&rt;"
- instead of ">" * Double quote, apostrophe, and ampersand were
- not being escaped. ........ Merged revisions 406294-406295 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-22 22:24 +0000 [r406269] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/pbx.c, /, utils/extconf.c: pbx.c: Pre-initialize timezone to
- avoid crash on destroy In ast_build_timing, initialize the
- timezone value to NULL in order to avoid deferencing an
- uninitialized value later when calling ast_destroy_timing. The
- timezone value could be uninitialized if ast_build_timing were to
- fail due to a zero length time string. (closes issue
- ASTERISK-22861) Reported by: Sebastian Murray-Roberts Review:
- https://reviewboard.asterisk.org/r/3134/ Patches:
- ast_build_timing-initialize-timezone.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 406241 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406245 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406264 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-22 19:36 +0000 [r406153-406224] Kinsey Moore <kmoore@digium.com>
-
- * /, apps/app_confbridge.c: ConfBridge: Fix channel parameter
- documentation Confbridge AMI and CLI commands for mute, unmute,
- and setting the single video source can accept channel prefixes
- in lieu of a full channel name, but documentation states only
- that it is required and is a channel name. This corrects the
- documentation. (closes issue PQ-1397) Reported by: Steve Pitts
- ........ Merged revisions 406217 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406223 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: chan_sip: Decline image streams on
- unsupported transports This change allows chan_sip to decline
- individual image streams over unsupported transports in the SDP
- of the 200 response. Previously, an image stream offer with
- RTP/AVP as the transport would cause chan_sip to respond with a
- 488. (closes issue ASTERISK-22988) Reported by: adomjan Original
- patch by: adomjan ........ Merged revisions 406170 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406171 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406172 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_playback.c, /: res_stasis_playback: Correct error
- argument order Several of the playback error messages for invalid
- media input in res_stasis_playback.c had the media name and
- channel name reversed. They now correctly identify the channel
- name and media name. Reported by: skrusty ........ Merged
- revisions 406152 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-21 21:48 +0000 [r406134] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip.c: res_pjsip: Documentation improvement for
- Endpoint and AOR mailbox options. Making the help text for both
- more explicit regarding the format of mailbox identifiers. i.e.
- clarifying the format for app_voicemail mailboxes vs mailboxes
- from external MWI sources through modules such as
- res_external_mwi. ........ Merged revisions 406133 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-21 21:08 +0000 [r406082] Walter Doekes <walter+asterisk@wjd.nu>
-
- * main/manager.c, /, configs/manager.conf.sample: manager: Clarify
- eventfilter documentation. Textual changes only. Review:
- https://reviewboard.asterisk.org/r/3133/ ........ Merged
- revisions 406079 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406080 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406081 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-21 20:28 +0000 [r406006-406078] Kinsey Moore <kmoore@digium.com>
-
- * channels/chan_mgcp.c, /: chan_mgcp: Enforce locking for oseq This
- restricts direct usage of global oseq so that all accesses are
- locked and threads are not racing to get oseq values that they
- did not claim. This also fixes a build error in res_pktccops
- under dev mode. (closes issue ASTERISK-23100) Reported by:
- adomjan Patch by: adomjan ........ Merged revisions 406037 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 406038 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 406049 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_outbound_registration.c, res/res_pjsip.c: PJSIP:
- Handle headers in a list appropriately The PJSIP header parsing
- function (pjsip_parse_hdr) can generate more than one header
- instance from a single header field. These header instances exist
- as a list attached to the returned header and must be handled
- appropriately when they are added to a message or else only the
- first header instance will be used. This changes the linked list
- functions used in outbound proxy code to merge the lists
- properly. ........ Merged revisions 406020 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_sounds.h, res/ari/resource_bridges.h,
- res/ari/resource_device_states.h, res/ari/resource_mailboxes.h,
- res/ari/resource_asterisk.h, rest-api/api-docs/channels.json,
- res/ari/resource_applications.h, res/ari/resource_channels.c,
- res/res_ari_playbacks.c, res/res_ari_sounds.c,
- rest-api-templates/asterisk_processor.py,
- res/ari/resource_channels.h, res/res_ari_bridges.c, /,
- res/res_ari_device_states.c,
- rest-api-templates/ari_resource.h.mustache,
- res/res_ari_mailboxes.c, res/res_ari_asterisk.c,
- res/res_ari_applications.c,
- rest-api-templates/res_ari_resource.c.mustache,
- rest-api-templates/body_parsing.mustache (added),
- res/res_ari_channels.c, res/ari/resource_playbacks.h,
- rest-api-templates/param_parsing.mustache: ARI: Support channel
- variables in originate This adds back in support for specifying
- channel variables during an originate without compromising the
- ability to specify query parameters in the JSON body. This was
- accomplished by generating the body-parsing code in a separate
- function instead of being integrated with the URI query parameter
- parsing code such that it could be called by paths with body
- parameters. This is transparent to the user of the API and
- prevents manual duplication of code or data structures. (closes
- issue ASTERISK-23051) Review:
- https://reviewboard.asterisk.org/r/3122/ Reported by: Matt Jordan
- ........ Merged revisions 406003 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-20 23:25 +0000 [r405985] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny: fix up handling of fragmented
- packets. Bad offset in reading second or more fragment of skinny
- packets. Fixed to offset by char (single byte) rather than size
- of req. ........ Merged revisions 405982 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-20 22:23 +0000 [r405947] Richard Mudgett <rmudgett@digium.com>
-
- * channels/sig_pri.c, /: chan_dahdi/PRI: Suppress CONNECTED_LINE
- updates when nothing in the udpate is valid. * Also simplified
- some subddress handling code. (closes issue ASTERISK-23008)
- Reported by: Michael Cargile ........ Merged revisions 405926
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 405927 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405928 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-20 21:56 +0000 [r405925] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny: fix up session logging.
- Logging from the skinny session loop was providing some incorrect
- reasons for exiting the loop. Cleaned up messages and handling so
- correct reason displayed. ........ Merged revisions 405924 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-20 18:18 +0000 [r405910] Jonathan Rose <jrose@digium.com>
-
- * channels/chan_pjsip.c, /: chan_pjsip: Provide a means for
- tracking device state when holding/unholding Previously PJSIP did
- not track hold/unhold and it would always simply be 'inuse'. This
- patch fixes that. review:
- https://reviewboard.asterisk.org/r/3129/ ........ Merged
- revisions 405908 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-19 00:01 +0000 [r405894] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny: fix reversed device reset from
- CLI. Existing code would do a full device restart when "skinny
- reset device" was entered at the CLI and do a reset when "skinny
- reset device restart" entered. ........ Merged revisions 405893
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-17 22:09 +0000 [r405878] Sean Bright <sean@malleable.com>
-
- * /, channels/chan_sip.c: Make sure the maxptime attribute is added
- to the correct offers. ........ Merged revisions 405877 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-17 21:33 +0000 [r405862-405876] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/format_pref.c, main/sorcery.c, main/frame.c, /,
- include/asterisk/format_pref.h, res/res_pjsip_sdp_rtp.c: pjsip:
- fix support for allow=all This change adds improvements to
- support for allow=all in pjsip.conf so that it functions as
- intended. Previously, the allow/disallow socery configuration
- would set & clear codecs from the media.codecs and media.prefs
- list, but if all was specified the prefs list was not updated.
- Then a call would fail when create_outgoing_sdp_stream() created
- an SDP with no audio codecs. A new function
- ast_codec_pref_append_all() is provided to add all codecs to the
- prefs list - only those not already on the list. This enables the
- configuration to specify a codec preference, but still add all
- codecs, and even then remove some codecs, as shown in this
- example: allow = ulaw, alaw, all, !g729, !g723 Also, the display
- order of allow in cli output is updated to match the
- configuration by using prefs instead of caps when generating a
- human readable string. Finally, a change to
- create_outgoing_sdp_stream() skips a codec when it does not have
- a payload code instead of the call failing. (closes issue
- ASTERISK-23018) Reported by: xrobau Review:
- https://reviewboard.asterisk.org/r/3131/ ........ Merged
- revisions 405875 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/http.c: http: supported chunked Transfer-Encoding This
- change implements support for HTTP Transfer-Encoding chunked in
- both JSON and Form (post vars) body content. A new function
- ast_http_get_contents() handles both regular and chunked mode
- body, returning after the entire body is received. (closes issue
- ASTERISK-23068) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3125/ ........ Merged
- revisions 405861 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-17 18:55 +0000 [r405778-405844] Rusty Newton <rnewton@digium.com>
-
- * res/res_pjsip.c, /: Fixing some XML syntax issues with my
- previous commit at r405777 for ASTERISK-23071 ........ Merged
- revisions 405843 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c, doc/asterisk.8, main/features.c,
- configs/sip.conf.sample, apps/app_queue.c, apps/app_transfer.c,
- channels/chan_iax2.c: Documentation: doc fixes across various
- parts of the code for ASTERISK issues 23061,23028,23046,23027
- Fixes typos of "transfered" instead of "transferred" in various
- code. Fixes incorrect gosub param help text for app_queue. Fixes
- Asterisk man pages containing unquoted minus signs. Adds note
- about the "textsupport" option in sip.conf.sample. (issue
- ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046)
- (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes
- issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue
- ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis
- Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine
- (license 6561) hyphen.patch uploaded by Jeremy Laine (license
- 6561) sip.conf.sample.patch uploaded by Eugene (license 6360)
- ........ Merged revisions 405791 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 405792 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405829 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip.c, /: res_pjsip: enhance documentation for
- mailboxes options, for both endpoints and aors Made documentation
- more explicit as to the use of the both options. (issue
- ASTERISK-23071) (closes issue ASTERISK-23071) Reported by: Matt
- Jordan ........ Merged revisions 405777 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-17 14:17 +0000 [r405766] Walter Doekes <walter+asterisk@wjd.nu>
-
- * res/res_musiconhold.c, CHANGES: Enable wide band audio in
- musiconhold streams. Review:
- https://reviewboard.asterisk.org/r/3112/
-
-2014-01-16 20:06 +0000 [r405747-405749] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip/pjsip_options.c, /: res_pjsip: AOR option
- qualify_frequency not respected on startup If an endpoint had
- previously dynamically registered a contact and the contact
- information was successfully stored in astdb then upon restart
- the qualify notifications would not be sent out if the
- qualify_frequency was set. This was due to the fact that only
- permanent contacts were being checked and scheduled for qualifies
- on startup. Modified the code to check and schedule all
- registered contacts at startup. (closes issue ASTERISK-23062)
- Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/3124/ ........ Merged
- revisions 405748 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: manager: Originate doesn't abort on failed
- format_cap allocation action_originate responds to the remote
- system with an error when cap==NULL, but doesn't return (abort
- the originate). Patched to return. (closes issue ASTERISK-23034)
- Reported by: Corey Farrell Patches: ASTERISK-23034.patch uploaded
- by coreyfarrell (license 5909) ........ Merged revisions 405745
- from http://svn.asterisk.org/svn/asterisk/branches/11 ........
- Merged revisions 405746 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-16 19:33 +0000 [r405744] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip.c: PJSIP: Fix outbound OPTIONS support When path
- support was added and contacts were made available during request
- creation and transmission, the code path used by outbound qualify
- support was not modified correctly and was causing request
- creation to fail. This ensures that outbound request creation
- with only a contact and no dialog, endpoint, or uri can succeed
- which restores qualify support. Reported by: gtjoseph Reported
- by: kharwell ........ Merged revisions 405743 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-16 19:13 +0000 [r405644-405695] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_fax.c, configs/res_fax.conf.sample: res_fax:
- check_modem_rate() returned incorrect rate for V.27 According to
- the new standard for V.27 and V.32 they are able to transmit at a
- bit rate of 4,800 or 9,600. The check_mode_rate function needed
- to be updated to reflect this. Also, because of this change the
- default 'minrate' value was updated to be 4800. (closes issue
- ASTERISK-22790) Reported by: Paolo Compagnini Patches:
- res_fax.txt uploaded by looserouting (license 6548) ........
- Merged revisions 405656 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 405693 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405694 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_pjsip.c: chan_pjsip: initial device state on
- endpoints is INVALID When endpoints get loaded their device state
- gets set to 'INVALID' because the channel driver has not been
- loaded yet. Fixed by updating the device state for every endpoint
- upon load of the channel driver. (closes issue ASTERISK-23065)
- Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/3123/ ........ Merged
- revisions 405643 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-15 16:51 +0000 [r405586-405589] Jonathan Rose <jrose@digium.com>
-
- * CHANGES: Make 12 - 12.1 CHANGES log the same as in 12
-
- * CHANGES, /: Include CHANGES info for r405553 ........ Merged
- revisions 405585 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-15 16:36 +0000 [r405584] Joshua Colp <jcolp@digium.com>
-
- * /, cel/cel_manager.c: cel_manager: Don't crash if configuration
- file is invalid. The cel_manager module did not properly handle
- the case where the configuration file was invalid. The module
- will now output a warning message and disable itself if this
- occurs. Reported by: Bryan Walters ........ Merged revisions
- 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 405582 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405583 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-15 13:16 +0000 [r405566] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip/location.c, res/res_pjsip_outbound_registration.c,
- res/res_pjsip_path.c (added), res/res_pjsip_mwi.c,
- res/res_pjsip/pjsip_distributor.c, res/res_pjsip_diversion.c,
- channels/chan_pjsip.c, res/res_pjsip_registrar.c,
- res/res_pjsip_refer.c, include/asterisk/res_pjsip.h,
- include/asterisk/res_pjsip_session.h, res/res_pjsip_notify.c, /,
- res/res_pjsip_messaging.c, res/res_pjsip_caller_id.c,
- res/res_pjsip_t38.c, res/res_pjsip.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip_nat.c,
- res/res_pjsip_session.c,
- contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py
- (added), res/res_pjsip_header_funcs.c: PJSIP: Add Path header
- support This adds Path support to chan_pjsip in res_pjsip_path.c
- with minimal additions in res_pjsip_registrar.c to store the path
- and additions in res_pjsip_outbound_registration.c to enable
- advertisement of path support to registrars and intervening
- proxies. Path information is stored on contacts and is enabled
- via Address of Record (AoRs) and Registration configuration
- sections. While adding path support, it became necessary to be
- able to add SIP supplements that handled messages outside of
- sessions, so a framework for handling these types of hooks was
- added in parallel to the already-existing session supplements and
- several senders of out-of-dialog requests were refactored as a
- result. (closes issue ASTERISK-21084) Review:
- https://reviewboard.asterisk.org/r/3050/ ........ Merged
- revisions 405565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-14 23:44 +0000 [r405554] Jonathan Rose <jrose@digium.com>
-
- * res/res_stasis_mailbox.exports.in (added),
- res/ari/ari_model_validators.h, rest-api/api-docs/mailboxes.json
- (added), include/asterisk/stasis_app_mailbox.h (added),
- res/ari/resource_mailboxes.c (added), /, res/ari.make,
- res/res_ari_mailboxes.c (added), res/ari/resource_mailboxes.h
- (added), res/res_stasis_mailbox.c (added),
- rest-api/resources.json, res/ari/ari_model_validators.c: ARI: Add
- mailboxes resource for controlling and polling external MWI Adds
- the following AMI commands: PUT mailboxes/mailboxName modifies
- mailbox state and implicitly creates new mailboxes GET
- mailboxes/mailboxName retrieves a JSON representation of a single
- mailbox if it exists GET mailboxes retrieves a JSON array of all
- mailboxes DELETE mailbox/mailboxName deletes a mailbox Note that
- res_mwi_external must be loaded for these functions to actually
- do anything. Review: https://reviewboard.asterisk.org/r/3117/
- ........ Merged revisions 405553 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-14 21:46 +0000 [r405542] Richard Mudgett <rmudgett@digium.com>
-
- * main/strings.c, /: string container: Remove unnecessary RAII_VAR
- usage and string object lock. ........ Merged revisions 405541
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-14 18:15 +0000 [r405437] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: fix Local From tag on outbound
- register regression In ASTERISK-12117, an improvement to insure
- consistant local from tags on outbound registrations resulted in
- an undesirable behavior - caused by leftover unexpired sip_pvt
- dialogs (with the previous cseq number), resulting in many
- uncessary REGISTER requests. Instead of significant rework of
- transmit_register(), this change deletes the dialogs after a 200
- OK response indiciating a successful registration, keeping the
- old dialogs from interfering with normal operation. (closes issue
- ASTERISK-22946) Reported by: Stephan Eisvogel Review:
- https://reviewboard.asterisk.org/r/3109/ ........ Merged
- revisions 405433 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 405434 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405435 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-14 18:14 +0000 [r405436] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_verbose.c, main/asterisk.c, configs/logger.conf.sample,
- main/cli.c, include/asterisk/logger.h, main/pbx.c,
- main/manager.c, /, funcs/func_timeout.c, apps/app_dumpchan.c,
- main/logger.c, UPGRADE.txt: verbosity: Fix performance of console
- verbose messages. The per console verbose level feature as
- previously implemented caused a large performance penalty. The
- fix required some minor incompatibilities if the new rasterisk is
- used to connect to an earlier version. If the new rasterisk
- connects to an older Asterisk version then the root console
- verbose level is always affected by the "core set verbose"
- command of the remote console even though it may appear to only
- affect the current console. If an older version of rasterisk
- connects to the new version then the "core set verbose" command
- will have no effect. * Fixed the verbose performance by not
- generating a verbose message if nothing is going to use it and
- then filtered any generated verbose messages before actually
- sending them to the remote consoles. * Split the "core set debug"
- and "core set verbose" CLI commands to remove the per module
- verbose support that cannot work with the per console verbose
- level. * Added a silent option to the "core set verbose" command.
- * Fixed "core set debug off" tab completion. * Made "core show
- settings" list the current console verbosity in addition to the
- root console verbosity. * Changed the default verbose level of
- the 'verbose' setting in the logger.conf [logfiles] section. The
- default is now to once again follow the current root console
- level. As a result, using the AMI Command action with "core set
- verbose" could again set the root console verbose level and
- affect the verbose level logged. (closes issue AST-1252) Reported
- by: Guenther Kelleter Review:
- https://reviewboard.asterisk.org/r/3114/ ........ Merged
- revisions 405431 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405432 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-14 16:43 +0000 [r405420] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip/pjsip_distributor.c: Fix erroneous behavior when
- sending auth rejection to artificial endpoint. We were not
- including an authentication challenge when sending a 401 response
- to unmatched endpoints. This was due to the conversion to use a
- vector for authentication section names on an endpoint. The
- vector for artificial endpoints was empty, resulting in the
- challenge being sent back containing no challenges. This is
- worked around by placing a bogus value in the artificial
- endpoint's auth vector. This value is never looked up by
- anything, since they instead will directly call
- ast_sip_get_artificial_auth().
-
-2014-01-14 03:27 +0000 [r405369] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Skinny: do not add call to missed
- calls list if answered elsewhere. Patch updates skinny devices
- with a SKINNY_CONNECTED callstate if an inbound ringing or
- callwaiting call is answered elsewhere. ........ Merged revisions
- 405367 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-13 13:34 +0000 [r405339] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip/pjsip_cli.c: res_pjsip: Fix CLI tab completion
- issues This fixes several issues with the new res_pjsip CLI tab
- completion such as output of headers during tab completion and
- being able to tab-complete more items than the code actually
- handled (further items would simply be ignored). (closes issue
- ASTERISK-23081) Review: https://reviewboard.asterisk.org/r/3115/
- Reported by: xrobau ........ Merged revisions 405338 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-12 22:24 +0000 [r405326] Joshua Colp <jcolp@digium.com>
-
- * res/ari/resource_playbacks.c, res/ari/resource_channels.c,
- include/asterisk/ari.h, res/ari/resource_bridges.c,
- res/ari/resource_recordings.c, res/ari/resource_device_states.c,
- res/res_ari.c, res/ari/resource_endpoints.c, /,
- res/ari/resource_applications.c: res_ari: Fix various memory
- leaks. This change fixes a few memory leaks that were found based
- on a mailing list post. 1. Some JSON response messages were never
- freed. This was caused by the documentation stating that message
- references were stolen when in reality they were not. The code
- now follows the documentation and usage has been updated. 2. HTTP
- response headers were never freed. 3. The variable list for
- wildcards paths was never freed. (closes issue ASTERISK-23128)
- Reported by: Kenneth Watson (on list) Review:
- https://reviewboard.asterisk.org/r/3119/ ........ Merged
- revisions 405325 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-12 22:13 +0000 [r405313-405314] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h,
- apps/app_cdr.c, main/cdr.c: CDRs: Synchronize dialplan
- applications that manipulate CDRs with the engine In
- https://reviewboard.asterisk.org/r/3057/, applications and
- functions that manipulate CDRs were made to interact over Stasis.
- This was done to synchronize manipulations of CDRs from the
- dialplan with the updates the engine itself receives over the
- message bus. This change rested on a faulty premise: that
- messages published to the CDR topic or to a topic that forwards
- to the CDR topic are synchronized with the messages handled by
- the CDR topic subscription in the CDR engine. This is not the
- case. There is no ordering guaranteed for two messages published
- to the same topic; ordering is only guaranteed if a message is
- published to the same subscriber. Stasis was modified in r405311
- to allow a publisher to synchronize on the subscriber. This patch
- uses that API to synchronize the CDR publishers with the CDR
- engine message router, which maintains the overall topic
- subscription. (closes issue ASTERISK-22884) Reported by: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........
- Merged revisions 405312 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis.c, main/stasis_message_router.c, /,
- include/asterisk/stasis.h,
- include/asterisk/stasis_message_router.h, tests/test_stasis.c:
- stasis: Add methods to allow for synchronous publishing to
- subscriber This patch adds an API call to Stasis that allows a
- publisher to publish a stasis message that will not return until
- a specific subscriber handles the message. Since a subscriber can
- have their own forwarding topic which orders messages from many
- topics, this allows a publisher who knows of that subscriber to
- synchronize to that subscriber regardless of the forwarding
- relationships between topics. This is of particular use for
- dialplan applications that need to synchronize on a particular
- subscriber's handling of a message. (issue ASTERISK-22884)
- Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3099/ ........ Merged
- revisions 405311 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-10 20:00 +0000 [r405299] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip/security_events.c: Print "<unknown>" for
- artificial endpoint in PJSIP security events. Previously, this
- printed a UUID, which was not very clear when dealing with an
- artificial endpoint. Review:
- https://reviewboard.asterisk.org/r/3113 ........ Merged revisions
- 405298 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-10 18:17 +0000 [r405284] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/logger.c: Logging callid: Fix some sizeof() references
- per coding guidelines. ........ Merged revisions 405281 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405282 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 23:52 +0000 [r405270] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_session.c: PJSIP: Add unhold on reinvite without
- SDP behavior Review: https://reviewboard.asterisk.org/r/3106/
-
-2014-01-09 23:50 +0000 [r405269] Damien Wedhorn <voip@facts.com.au>
-
- * channels/chan_dahdi.c, /: Fix chan_dahdi copile issue in
- dev-mode. Error "unused variable i in dahdi_create_channel_range"
- when compiling in dev-mode. Small restructure to
- dahdi_create_channel_range to move the for(x) loop and int i,x to
- a block within the IFDEF. ........ Merged revisions 405268 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 23:39 +0000 [r405267] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip.c, /, res/res_pjsip_messaging.c:
- res_pjsip_messaging: potential for field values in from/to
- headers to be missing Added in ability to specify display name
- format ("name" <sip:name@ipaddr:port>) for a given URI and made
- sure it was fully propagated to the outgoing message. Also made
- it so outoing messages in res_pjsip always send as "sip:".
- (closes issue ASTERISK-22924) Reported by: Anthony Messina
- Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged
- revisions 405266 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 20:34 +0000 [r405254] Kinsey Moore <kmoore@digium.com>
-
- * main/astobj2.c, res/res_pjsip_session.c, /,
- include/asterisk/astobj2.h: astobj2: Correct ao2_iterator opacity
- violations This corrects the ao2_iterator opacity violations in
- res_pjsip_session.c by adding a global function to get the number
- of elements inside the container hidden behind the iterator.
- (closes issue ASTERISK-23053) Review:
- https://reviewboard.asterisk.org/r/3111/ Reported by: Richard
- Mudgett ........ Merged revisions 405253 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 16:52 +0000 [r405236] Kevin Harwell <kharwell@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fails to resume
- WebRTC call from hold In ast_rtp_ice_start if the ice session
- create check list failed, start check was never initiated and
- ice_started was never set to true. Upon re-entering the function
- (for instance, [un]hold) it would try to create the check list
- again with duplicate remote candidates. Fixed so that if the
- create check list fails the necessary data structures are
- properly re-initialized for any subsequent retries. Note, it was
- decided to not stop ice support (by calling ast_rtp_ice_stop) on
- a check list failure because it possible things might still work.
- However, a debug message was added to help with any future
- troubleshooting. (closes issue ASTERISK-22911) Reported by: Vytis
- Valentinavičius Patches: works_on_my_machine.patch uploaded by
- xytis (license 6558) ........ Merged revisions 405234 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405235 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 15:50 +0000 [r405217] Matthew Jordan <mjordan@digium.com>
-
- * /, apps/app_confbridge.c,
- apps/confbridge/conf_state_multi_marked.c: app_confbridge: Fix
- crash caused when waitmarked/marked users leave together When
- waitmarked users join a ConfBridge, the conference state is
- transitioned from EMPTY -> INACTIVE. In this state, the users are
- maintined in a waiting users list. When a marked user joins, the
- ConfBridge conference transitions from INACTIVE -> MULTI_MARKED,
- and all users are put onto the active list of users. This process
- works correctly. When the marked user leaves, if they are the
- last marked user, the MULTI_MARKED state does the following: (1)
- It plays back a message to the bridge stating that the leader has
- left the conference. This requires an unlocking of the bridge.
- (2) It moves waitmarked users back to the waiting list (3) It
- transitions to the appropriate state: in this case, INACTIVE
- However, because it plays the prompt back to the bridge before
- moving the users and before finishing the state transition, this
- creates a race condition: with the bridge unlocked, waitmarked
- users who leave the conference (or are kicked from it) can cause
- a state transition of the bridge to another state before the
- conference is transitioned to the INACTIVE state. This causes the
- state machine to get a bit wonky, often leading to a crash when
- the MULTI_MARKED state attempts to conclude its processing. This
- patch fixes this problem: (1) It prevents kicked users from being
- kicked again. That's just a nicety. (2) More importantly, it
- fixes the race condition by only playing the prompt once the
- state has transitioned correctly to INACTIVE. If waitmarked users
- sneak out during the prompt being played, no harm no foul.
- Review: https://reviewboard.asterisk.org/r/3108/ Note that the
- patch committed here is essentially the same as uploaded by Simon
- Moxon on ASTERISK-22740, with the addition of the double kick
- prevention. (closes issue AST-1258) Reported by: Steve Pitts
- (closes issue ASTERISK-22740) Reported by: Simon Moxon patches:
- ASTERISK-22740.diff uploaded by Simon Moxon (license 6546)
- ........ Merged revisions 405215 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405216 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-09 14:15 +0000 [r405163] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, apps/app_dumpchan.c: "Minimun" typo. ........ Merged revisions
- 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 405161 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405162 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-08 17:23 +0000 [r405144] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip/security_events.c: Use proper case for checking
- if digest authentication is used. ........ Merged revisions
- 405131 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-08 16:34 +0000 [r405129-405130] Kinsey Moore <kmoore@digium.com>
-
- * /, configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support
- for Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
- available on newer operating systems. (closes issue
- ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
- Reported by: George Joseph Patch by: George Joseph ........
- Merged revisions 405090 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 405091 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405124 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: Add the missing part of r400140 When the
- patch to add retry-on-forbidden-response was committed, part of
- the patch for chan_sip was not committed which caused the feature
- to be entirely nonfunctional. This corrects the code in question.
- (closes issue ASTERISK-17138) Review:
- https://reviewboard.asterisk.org/r/2874 ........ Merged revisions
- 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 405081 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 405083 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-07 19:56 +0000 [r405020-405035] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_acl.c: res_pjsip_acl: Fix another case of
- assuming a contact will always contain a URI. ........ Merged
- revisions 405034 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_nat.c: res_pjsip_nat: Don't assume a Contact
- header will always contain a URI. If the 'rewrite_contact' option
- was enabled and a Contact header was received which contained a
- '*' a crash would occur. This change makes the res_pjsip_nat
- module ignore the Contact header if it contains only a '*'.
- (closes issue ASTERISK-23101) Reported by: Matt Jordan ........
- Merged revisions 405019 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-06 21:55 +0000 [r404953-405007] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_voicemail.c, /: app_voicemail: Explicitly set
- defaultenabled=yes ........ Merged revisions 405006 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_mwi_external_ami.c (added): External MWI AMI support.
- The external MWI AMI interface provides a thin wrapper around the
- core external MWI resource. The resource adds the following AMI
- actions: MWIGet, MWIDelete, and MWIUpdate. (closes issue AFS-46)
- Review: https://reviewboard.asterisk.org/r/3061/ ........ Merged
- revisions 404954 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_mwi_external.c (added), configs/sorcery.conf.sample,
- include/asterisk/res_mwi_external.h (added),
- res/res_mwi_external.exports.in (added), apps/app_voicemail.c:
- External MWI core support. * The core external MWI resource
- provides for MWI message counts persistence using sorcery. With
- sorcery, the user is able to configure which sorcery wizzard
- backend to use if the default astdb is not desired. * The core
- external MWI resoruce provides some debugging CLI commands
- enabled by defining MWI_DEBUG_CLI. The debugging CLI commands
- are: "mwi delete all", "mwi delete like <regex>", "mwi delete
- mailbox <mailbox>", "mwi list all", "mwi list like <regex>", "mwi
- show mailbox <mailbox>", and "mwi update mailbox <mailbox> [<new>
- [<old>]]". (closes issue AFS-43) Review:
- https://reviewboard.asterisk.org/r/3061/ ........ Merged
- revisions 404952 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-05 16:01 +0000 [r404924-404936] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Don't assume that a registration
- client will always exist. ........ Merged revisions 404935 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Create registration client in pj
- thread. Depending on which threading was loading the outbound
- registration it was possible for the registration client to be
- allocated outside of a pj thread. This change moves the creation
- inside the synchronous task where it is guaranteed it will occur
- in a pj thread. Reported by: Rob Thomas ........ Merged revisions
- 404923 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-04 10:52 +0000 [r404912] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * main/asterisk.c, /: asterisk.c: suppress live_dangerously warning
- on rasterisk Even since the fixes of AST-2013-007, Asterisk
- prints the following warning on startup if the user decided to
- live dangerously: Privilege escalation protection disabled! See
- https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
- message is intended for the logs and interactive startup. No need
- for it to appear on a remote console. This commit removes it from
- there. (closes issue ASTERISK-23084) Review:
- https://reviewboard.asterisk.org/r/3101/ ........ Merged
- revisions 404861 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404888 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404911 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 22:00 +0000 [r404860] Kevin Harwell <kharwell@digium.com>
-
- * cel/cel_pgsql.c, /: cel_pgsql: module not correctly reloading
- Upon reload the module unconditionally "unloaded" the module
- (freeing memory and setting pointers to NULL) and then when
- attempting a "load" if the config file had not changed then
- nothing would be reinitialized. By moving the "unload" to occur
- conditionally (reload only) after an attempted configuration
- load, but before module "loading" alleviates the issue. The
- module now loads/unloads/reloads correctly. (closes issue
- ASTERISK-22871) Reported by: Matteo ........ Merged revisions
- 404857 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 404858 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404859 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 21:45 +0000 [r404844-404856] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_pjsip_logger.c: res_pjsip_logger: Add the
- ASTERISK_FILE_VERSION macro Registering yourself with the
- Asterisk core is the nice thing to do, even when you're a logging
- module. ........ Merged revisions 404855 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_authenticator_digest.c, tests/test_utils.c:
- res_pjsip_authenticator_digest: Fix md5 hash buffer An md5 hash
- is 32 bytes long. The char buffer must be at least 33 bytes to
- avoid clobbering of the stack. This patch also fixes a potential
- clobbering in test_utils.c. Thanks to Andrew Nagy for reporting
- and testing this out in #asterisk-dev Reported by: Andrew Nagy
- Tested by: Andrew Nagy ........ Merged revisions 404843 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 20:02 +0000 [r404787-404832] Kevin Harwell <kharwell@digium.com>
-
- * main/manager.c: manager: UserEvent including action on output AMI
- action UserEvent event response would include the action header
- in its keyvalue pairs list. Adjusted the start of the header loop
- to skip over the action part. (closes issue ASTERISK-22899)
- Reported by: outtolunc Patches:
- svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license
- 5198)
-
- * channels/chan_dahdi.c, /: chan_dahdi: dahdi show channels slices
- PRI channel dnid on output dahdi show channels output slices the
- callerid (which is dnid copied over on PRI channels). If the
- channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
- then the output slices 1408409XXXX down to 1408409XXX. This patch
- just opens it up to 15 chars so you can see the whole thing.
- (closes issue ASTERISK-22918) Reported by: outtolunc Patches:
- svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
- (license 5198) ........ Merged revisions 404784 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404785 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404786 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 18:33 +0000 [r404783] Richard Mudgett <rmudgett@digium.com>
-
- * tests/test_stasis.c, /: test_stasis.c: Fix ref leak in normal
- execution path. ........ Merged revisions 404764 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 18:31 +0000 [r404782] Kevin Harwell <kharwell@digium.com>
-
- * /, apps/app_meetme.c: app_meetme: compiler warning Fixed a
- compiler warning (errors in 'dev-mode') given by gcc version
- 4.8.1. The one in app_meetme involved the
- 'sizeof-pointer-memaccess' (see:
- http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so it
- would no longer issue a warning and can compile again in
- 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
- ........ Merged revisions 404742 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404773 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404781 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-03 17:27 +0000 [r404726-404738] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip/pjsip_configuration.c, /, res/res_pjsip/location.c:
- res_pjsip: Ensure more URI validation happens in pj threads.
- ........ Merged revisions 404737 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_outbound_registration.c:
- res_pjsip_outbound_registration: Ensure URI validation happens in
- a pjlib thread. This change moves outbound registration URI
- validation into the task executed within a pjlib thread. Reported
- by: Andrew Nagy ........ Merged revisions 404725 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-02 19:38 +0000 [r404677] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, funcs/func_strings.c: func_strings: use memmove to prevent
- overlapping memory on strcpy When calling REPLACE() with an empty
- replace-char argument, strcpy is used to overwrite the the
- matching <find-char>. However as the src and dest arguments to
- strcpy must not overlap, it causes other parts of the string to
- be overwritten with adjacent characters and the result is
- mangled. Patch replaces call to strcpy with memmove and adds a
- test suite case for REPLACE. (closes issue ASTERISK-22910)
- Reported by: Gareth Palmer Review:
- https://reviewboard.asterisk.org/r/3083/ Patches:
- func_strings.patch uploaded by Gareth Palmer (license 5169)
- ........ Merged revisions 404674 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404675 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404676 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2014-01-02 19:08 +0000 [r404664] Kevin Harwell <kharwell@digium.com>
-
- * channels/chan_pjsip.c, include/asterisk/res_pjsip.h, /,
- configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c,
- CHANGES, res/res_pjsip.c: res_pjsip: add 'set_var' support on
- endpoints Added a new 'set_var' option for ast_sip_endpoint(s).
- For each variable specified that variable gets set upon creation
- of a pjsip channel involving the endpoint. (closes issue
- ASTERISK-22868) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/3095/ ........ Merged
- revisions 404663 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-31 22:51 +0000 [r404620-404653] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_pjsip.c, res/res_pjsip_session.c, /: chan_pjsip:
- Handle hanging up before calling. Channel creation in Asterisk is
- broken up into two steps: requesting and calling. In some cases a
- channel may be requested but never called. This happens in the
- ChanIsAvail dialplan application for determining if something is
- reachable or not. The PJSIP channel driver did not take this
- situation into account and attempted to end a session that was
- never called out on. The code now checks the session state to
- determine if the session has been called out on and if not
- terminates it instead of ending it. (closes issue ASTERISK-23074)
- Reported by: Kilburn ........ Merged revisions 404652 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_endpoint_identifier_ip.c:
- res_pjsip_endpoint_identifier_ip: Accept hostnames in the 'match'
- field. Hostnames specified in the 'match' field will be resolved
- and all addresses returned. Each address will be added to the
- endpoint identifier for the matching process. Reported by: Rob
- Thomas ........ Merged revisions 404613 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-31 21:39 +0000 [r404606] Kevin Harwell <kharwell@digium.com>
-
- * cel/cel_pgsql.c, /: cel_pgsql: deadlock on unload and
- core_event_dispatcher A deadlock can happen between a thread
- unloading or reloading the cel_pgsql module and the
- core_event_dispatcher taskprocessor thread. Description of what
- is happening: Thread 1 (for example, a netconsole thread): a
- "module reload cel_pgsql" is launched the thread enter the
- "my_unload_module" function (cel_pgsql.c) the thread acquire the
- write lock on psql_columns the thread enter the
- "ast_event_unsubscribe" function (event.c) the thread try to
- acquire the write lock on ast_event_subs[sub->type] Thread 2
- (core_event_dispatcher taskprocessor thread): the taskprocessor
- pop a CEL event the thread enter the "handle_event" function
- (event.c) the thread acquire the read lock on
- ast_event_subs[sub->type] the thread callback the "pgsql_log"
- function (cel_pgsql.c), since it's a subscriber of CEL events the
- thread try to acquire a read lock on psql_columns (closes issue
- ASTERISK-22854) Reported by: Etienne Lessard Patches:
- cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
- 6394) ........ Merged revisions 404603 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404604 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404605 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-31 20:27 +0000 [r404593] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_outbound_registration.c, /:
- res_pjsip_outbound_registration: Add validation for 'server_uri'
- and 'client_uri'. When applying configuration for outbound
- registrations the 'server_uri' and 'client_uri' fields were not
- validated. The code will now confirm that they exist and that
- they contain parseable SIP URIs. Reported by: Andrew Nagy
- ........ Merged revisions 404592 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-30 23:25 +0000 [r404582] Kevin Harwell <kharwell@digium.com>
-
- * main/channel.c, /: channels.c: core show channeltypes slicing
- 'core show channeltypes' type column is being sliced, resulting
- in incomplete type names. (closes issue ASTERISK-22919) Reported
- by: outtolunc Patches: svn_channel.c.format_15.diff.txt uploaded
- by outtolunc (license 5198) ........ Merged revisions 404579 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404581 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-24 17:12 +0000 [r404567-404569] David M. Lee <dlee@digium.com>
-
- * UPGRADE-12.txt, /: Added note to UPGRADE.txt about the default
- value of live_dangerously changing ........ Merged revisions
- 404568 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/http.c: http: Properly reject requests with
- Transfer-Encoding set Asterisk does not support any of the
- transfer encodings specified in HTTP/1.1, other than the default
- "identity" encoding. According to RFC 2616: A server which
- receives an entity-body with a transfer-coding it does not
- understand SHOULD return 501 (Unimplemented), and close the
- connection. A server MUST NOT send transfer-codings to an
- HTTP/1.0 client. This patch adds the 501 Unimplemented response,
- instead of the hard work of actually implementing other
- recordings. This behavior is especially problematic for Node.js
- clients, which use chunked encoding by default. (closes issue
- ASTERISK-22486) Review: https://reviewboard.asterisk.org/r/3092/
- ........ Merged revisions 404565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-24 02:20 +0000 [r404554] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_pubsub.c: res_pjsip_pubsub: Ensure dialog
- manipulation happens on proper thread. When destroying a
- subscription we remove the serializer from its dialog and
- decrease its reference count. Depending on which thread dropped
- the subscription reference count to 0 it was possible for this to
- occur in a thread where it is not possible. (closes issue
- ASTERISK-22952) Reported by: Matt Jordan ........ Merged
- revisions 404553 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-23 16:38 +0000 [r404542] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
- UPGRADE-12.txt: chan_dahdi: enable ignore_failed_channels by
- default If ignore_failed_channels is set to "true" for a channel,
- the channel will continue to be configured even if configuring it
- has failed. This allows Asterisk to start before all the DAHDI
- initialization is done and thus not force the starting order
- dahdi -> asterisk. Review:
- https://reviewboard.asterisk.org/r/3063/
-
-2013-12-21 03:35 +0000 [r404532] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_pjsip/pjsip_cli.c: res_pjsip/pjsip_cli: fix
- compilation error caused by passing ast_free When wanting to pass
- *free as a function pointer, ast_free_ptr has to be used instead
- of ast_free. This allows it to be compiled with MALLOC_DEBUG
- enabled. ........ Merged revisions 404531 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 22:04 +0000 [r404511-404512] David M. Lee <dlee@digium.com>
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c, res/ari/resource_channels.h, /,
- rest-api/api-docs/applications.json: ari: Remove support for
- specifying channel vars during origination. When we added support
- for specifying channel variables for an origination, we didn't
- consider how that would interact with another feature, namely
- specifying request parameters in a JSON request body. The method
- of specifying channel variables (as a flat JSON object passed in
- the JSON body) interferes with parsing parameters out of the
- request body. Unfortunately, fixing this would be a backward
- incompatible change. In the interest of keeping the API sane and
- keeping our release schedule, we're dropping the feature for
- specifying channel variables in the origination request. We will
- bring the feature back soon, as a backward compatible addition to
- the API. (closes issue ASTERISK-23051) Review:
- https://reviewboard.asterisk.org/r/3088 ........ Merged revisions
- 404509 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /: Remove automerge properties ........ Merged revisions 404488
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 21:32 +0000 [r404507] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/config.h, main/config.c, main/channel.c,
- res/res_pjsip/location.c, include/asterisk/res_pjsip_cli.h
- (added), res/res_pjsip/pjsip_cli.c (added),
- include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip_registrar.c, main/sorcery.c,
- include/asterisk/res_pjsip.h, CREDITS,
- res/res_pjsip/config_auth.c, /,
- res/res_pjsip_endpoint_identifier_ip.c: res_pjsip: Add PJSIP CLI
- commands Implements the following cli commands: pjsip list aors
- pjsip list auths pjsip list channels pjsip list contacts pjsip
- list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show
- channels pjsip show endpoint(s) Also... Minor modifications made
- to the AMI command implementations to facilitate reuse. New
- function ast_variable_list_sort added to config.c and config.h to
- implement variable list sorting. (issue ASTERISK-22610) patches:
- pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
- ........ Merged revisions 404480 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 21:18 +0000 [r404461] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, main/say.c: say.c: correct time for polish In
- ast_say_date_with_format_pl(), change ast_say_number() to use
- tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
- by: Robert Mordec Review:
- https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
- uploaded by veilen (license 6555) ........ Merged revisions
- 404456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 404457 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 20:28 +0000 [r404452] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_refer.c: Fix issue where PJSIP blind transferer
- dialog may not complete as planned. When transferring to a
- dialplan extension that will not place any outbound calls, the
- only control frames that the PJSIP REFER framehook will receive
- are inconsequential (such as unhold or srcchange). As such, we
- shouldn't allow for the reception of those types of frames
- prevent us from signaling to the transferring party that the
- transfer has completed successfully once voice frames are read.
- Thanks to Jonathan Rose for pointing this out. ........ Merged
- revisions 404439 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 20:05 +0000 [r404438] Matthew Jordan <mjordan@digium.com>
-
- * /, res/ari/resource_applications.h,
- res/res_stasis_device_state.c: res_stasis_device_state: Set
- resource type for subscriptions to deviceState The documentation
- for ARI already specifies that the device state resource when
- used for subscribing for events is "deviceState", not
- "device_state". The code, however, used "device_state"; although
- this was inconsistent as well in doxygen comments in
- resource_applications. Because the actual resource being
- subscribed to is /deviceStates/{device}/, it makes sense for the
- resource type specifier to be deviceState. Note that the key
- value in the events is still "device_state". ........ Merged
- revisions 404437 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 20:00 +0000 [r404436] Richard Mudgett <rmudgett@digium.com>
-
- * res/ari/resource_channels.c, tests/test_scoped_lock.c,
- tests/test_stasis.c, res/parking/parking_manager.c,
- res/ari/resource_bridges.c, res/ari/resource_endpoints.c, /,
- res/res_pjsip/location.c, tests/test_cel.c: ao2_iterator:
- Mini-audit of the ao2_iterator loops in the new code files. *
- Fixed several places where ao2_iterator_destroy() was not called.
- * Fixed several iterator loop object variable reference problems.
- * Fixed res_parking AMI actions returning non-zero. Only the AMI
- logoff action can return non-zero. Review:
- https://reviewboard.asterisk.org/r/3087/ ........ Merged
- revisions 404434 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 19:25 +0000 [r404433] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/manager.h, /: manager: bump version to 2.0.0 AMI
- has received substantial updates over the past year. Not only has
- the syntax been vastly improved and made consistent (which
- entails many event changes), but the underlying things that those
- events convey have changed substantially as well. After some
- conversation in #asterisk-dev, it was agreed that this is a good
- time to jump to 2. At the same time, since ARI will most likely
- use semantic versioning, we might as well use that for AMI as
- well. That also affords us greater meaning for the AMI version.
- ........ Merged revisions 404421 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 19:06 +0000 [r404420] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/sounds_index.c: Whitespace fixes. ........ Merged
- revisions 404419 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-20 17:22 +0000 [r404406] Rusty Newton <rnewton@digium.com>
-
- * /, configs/pjsip.conf.sample: Documentation: Updates for info
- about NAT-related settings and fixes for pjsip.conf.sample Added
- another NAT example to pjsip.conf.sample. We had a few mentions
- of NAT configuration throughout the sample, but I added another
- for a little bit more clarity. Additionally many pjsip options
- were affected by the change to snake case, so I fixed any
- instances of those options in pjsip.conf. I regenerated the
- config option list (at the bottom of the file) from a new xml
- config doc dump, so all the snake case changes should be
- reflected there, as well as any other changes to those options.
- (issue ASTERISK-23004) (closes issue ASTERISK-23004) Reported by:
- Matt Jordan Review: https://reviewboard.asterisk.org/r/3086/
- ........ Merged revisions 404405 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 20:48 +0000 [r404387] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/security_events.c: security_events: log events with
- descriptive names This patch updates the log messages to include
- descriptive names for event types. This is an improvement over
- having only cryptic type numbers. (closes issue ASTERISK-22909)
- Reported by: outtolunc Review:
- https://reviewboard.asterisk.org/r/3081/ Patches:
- svn_security_events.c.names.diff.txt uploaded by outtolunc
- (license 5198)
-
-2013-12-19 18:16 +0000 [r404376] Richard Mudgett <rmudgett@digium.com>
-
- * /, CHANGES: Put notice in CHANGES as well as UPGRADE.txt.
- ........ Merged revisions 404375 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 18:00 +0000 [r404370-404372] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip/pjsip_outbound_auth.c, /: res_pjsip: Ignore 401/407
- responses for transactions and dialogs we don't know about. Under
- normal conditions it is unlikely we will ever receive a response
- for a transaction or dialog we don't know about but if any are
- received ignore them. ........ Merged revisions 404371 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_session.c: res_pjsip_session: Fix SDP
- negotiation when resending an INVITE with authentication. The
- process for resending an INVITE with authentication involves
- restarting the UAC session. We were incorrectly passing in that a
- new offer is being sent, causing the SDP negotiation to get into
- a (technically speaking) funky state. ........ Merged revisions
- 404369 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 17:45 +0000 [r404368] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/channel.h, res/res_pjsip.c, main/channel.c, /,
- include/asterisk/autochan.h: Fix a deadlock that occurred due to
- a conflict of masquerades. For the explanation, here is a
- copy-paste of the review board explanation: Initially, it was
- discovered that performing an attended transfer of a multiparty
- bridge with a PJSIP channel would cause a deadlock. A PBX thread
- started a masquerade and reached the point where it was calling
- the fixup() callback on the "original" channel. For chan_pjsip,
- this involves pushing a synchronous task to the session's
- serializer. The problem was that a task ahead of the fixup task
- was also attempting to perform a channel masquerade. However,
- since masquerades are designed in a way to only allow for one to
- occur at a time, the task ahead of the fixup could not continue
- until the masquerade already in progress had completed. And of
- course, the masquerade in progress could not complete until the
- task ahead of the fixup task had completed. Deadlock. The initial
- fix was to change the fixup task to be asynchronous. While this
- prevented the deadlock from occurring, it had the frightful side
- effect of potentially allowing for tasks in the session's
- serializer to operate on a zombie channel. Taking a step back
- from this particular deadlock, it became clear that the problem
- was not really this one particular issue but that masquerades
- themselves needed to be addressed. A PJSIP attended transfer
- operation calls ast_channel_move(), which attempts to both set up
- and execute a masquerade. The problem was that after it had set
- up the masquerade, the PBX thread had swooped in and tried to
- actually perform the masquerade. Looking at changes that had been
- made to Asterisk 12, it became clear that there never is any time
- now that anyone ever wants to set up a masquerade and allow for
- the channel thread to actually perform the masquerade. Everyone
- always is calling ast_channel_move(), performs the masquerade
- itself before returning. In this patch, I have removed all blocks
- of code from channel.c that will attempt to perform a masquerade
- if ast_channel_masq() returns true. Now, there is no distinction
- between setting up a masquerade and performing the masquerade. It
- is one operation. The only remaining checks for
- ast_channel_masq() and ast_channel_masqr() are in ast_hangup()
- since we do not want to interrupt a masquerade by hanging up the
- channel. Instead, now ast_hangup() will wait for a masquerade to
- complete before moving forward with its operation. The
- ast_channel_move() function has been modified to basically
- in-line the logic that used to be in ast_channel_masquerade().
- ast_channel_masquerade() has been killed off for real.
- ast_channel_move() now has a lock associated with it that is used
- to prevent any simultaneous moves from occurring at once. This
- means there is no need to make sure that ast_channel_masq() or
- ast_channel_masqr() are already set on a channel when
- ast_channel_move() is called. It also means the channel container
- lock is not pulling double duty by both keeping the container
- locked and preventing multiple masquerades from occurring
- simultaneously. The ast_do_masquerade() function has been renamed
- to do_channel_masquerade() and is now internal to channel.c. The
- function now takes explicit arguments of which channels are
- involved in the masquerade instead of a single channel. While it
- probably is possible to do some further refactoring of this
- method, I feel that I would be treading dangerously. Instead, all
- I did was change some comments that no longer are true after this
- changeset. The other more minor change introduced in this patch
- is to res_pjsip.c to make ast_sip_push_task_synchronous() run the
- task in-place if we are already a SIP servant thread. This is
- related to this patch because even when we isolate the channel
- masquerade to only running in the SIP servant thread, we would
- still deadlock when the fixup() callback is reached since we
- would essentially be waiting forever for ourselves to finish
- before actually running the fixup. This makes it so the fixup is
- run without having to push a task into a serializer at all.
- (closes issue ASTERISK-22936) Reported by Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/3069 ........ Merged revisions
- 404356 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 17:13 +0000 [r404355] Richard Mudgett <rmudgett@digium.com>
-
- * main/udptl.c, addons/chan_ooh323.c, /, channels/chan_sip.c,
- include/asterisk/udptl.h: udptl: Dead code elimination.
- ast_udptl_bridge was not used. Removing dead code starting with
- ast_udptl_bridge() eliminated the code in this change. Note: This
- code has actually been dead since Asterisk v1.4 when it was first
- put in. Review: https://reviewboard.asterisk.org/r/3079/ ........
- Merged revisions 404354 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 17:03 +0000 [r404353] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, res/res_fax.c: res_fax.c: crash on framehook with no dsp in
- fax detect In fax_detect_framehook() a null pointer reference can
- occur where a voice frame is processed but no dsp is attached to
- the fax detection structure. The code block that rejects frames
- that detection cannot be processed on is checking for dsp but
- falls through when it should instead return, as this change
- implements. (closes issue ASTERISK-22942) Reported by: adomjan
- Review: https://reviewboard.asterisk.org/r/3076/ ........ Merged
- revisions 404351 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404352 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 16:52 +0000 [r404350] Richard Mudgett <rmudgett@digium.com>
-
- * configs/skinny.conf.sample, res/res_xmpp.c, res/res_jabber.c,
- CHANGES, channels/chan_iax2.c, channels/sig_pri.c,
- channels/h323/chan_h323.h, configs/iax.conf.sample,
- channels/sig_pri.h, channels/chan_dahdi.c,
- include/asterisk/app.h, channels/chan_skinny.c,
- channels/chan_dahdi.h, channels/chan_h323.c, main/app.c,
- UPGRADE-12.txt, configs/sip.conf.sample,
- channels/sip/include/sip.h, channels/chan_mgcp.c,
- apps/app_voicemail.c, channels/chan_unistim.c,
- configs/chan_dahdi.conf.sample, /, channels/chan_sip.c,
- configs/voicemail.conf.sample, funcs/func_vmcount.c: Voicemail:
- Remove mailbox identifier format (box@context) assumptions in the
- system. This change is in preparation for external MWI support.
- Removed code from the system for normal mailbox handling that
- appends @default to the mailbox identifier if it does not have a
- context. The only exception is the legacy hasvoicemail users.conf
- option. The legacy option will only work for app_voicemail
- mailboxes. The system cannot make any assumptions about the
- format of the mailbox identifer used by app_voicemail. chan_sip
- and chan_dahdi/sig_pri had the most changes because they both
- tried to interpret the mailbox identifier. chan_sip just stored
- and compared the two components. chan_dahdi actually used the box
- information. The ISDN MWI support configuration options had to be
- reworked because chan_dahdi was parsing the box@context format to
- get the box number. As a result the mwi_vm_boxes chan_dahdi.conf
- option was added and is documented in the chan_dahdi.conf.sample
- file. Review: https://reviewboard.asterisk.org/r/3072/ ........
- Merged revisions 404348 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 16:33 +0000 [r404346] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/db.c, /: astdb: crash in sqlite3 during shutdown When
- Asterisk is shut down, the astdb_atexit() function releases
- (finalize) the previously initiated (prepared) SQL statements in
- sqlite3. Another thread making a subsequent request can cause a
- crash in sqlite3. This patch eliminates that issue by resetting
- the statement pointer after it is released/cleared. The sqlite3
- code detects the null pointer, and aborts the operation cleanly.
- (closes issue AST-1265) Reported by: Alexander Hömig (closes
- issue ASTERISK-22350) Reported by: Birger "WIMPy" Harzenetter
- Review: https://reviewboard.asterisk.org/r/3078/ ........ Merged
- revisions 404344 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404345 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 12:18 +0000 [r404333] Joshua Colp <jcolp@digium.com>
-
- * main/channel.c, /: channel: Add a missing ast_channel_unlock when
- allocating a Surrogate channel. ........ Merged revisions 404332
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 08:35 +0000 [r404321] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooGkClient.c,
- addons/chan_ooh323.c, /, addons/ooh323c/src/ooGkClient.h: Handle
- temporary failures on gk registration Introduce new 'stopped'
- state for gk client and restart gk client on failures Remove
- ooh323 stack command lock as it is not need now. (closes issue
- ASTERISK-21960) Reported by: Dmitry Melekhov Patches:
- ASTERISK-21960.patch ASTERISK-21960-stacklockup-2.patch Tested
- by: Dmitry Melekhov ........ Merged revisions 404318 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404320 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 02:59 +0000 [r404307] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Fixup some skinny bugs causing Fracks
- and ao2 cleanup issues. Moved channel locking into setsubstate so
- that a process can complete working on a sub before another
- starts changing it. The existing code was causing some Fracks
- with schedule deletion. Removed multiple rtp cleanup. Now only
- cleansup up once, fixing ao2 object cleanup issues. ........
- Merged revisions 404306 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 00:50 +0000 [r404295] Matthew Jordan <mjordan@digium.com>
-
- * include/asterisk/cdr.h, CHANGES, apps/app_cdr.c, main/cdr.c,
- apps/app_forkcdr.c, main/pbx.c, /, funcs/func_cdr.c,
- apps/app_disa.c, UPGRADE-12.txt: app_cdr,app_forkcdr,func_cdr:
- Synchronize with engine when manipulating state When doing the
- rework of the CDR engine that pushed all of the logic into cdr.c
- and made it respond to changes in channel state over Stasis, we
- knew that accessing the CDR engine from the dialplan would be
- "slightly" non-deterministic. Dialplan threads would be accessing
- CDRs while Stasis threads would be updating the state of said
- CDRs - whereas in the past, everything happened on the dialplan
- threads. Tests have shown that "slightly" is in reality "very".
- This patch synchronizes things by making the dialplan
- applications/functions that manipulate CDRs do so over Stasis.
- ForkCDR, NoCDR, ResetCDR, CDR, and CDR_PROP now all use Stasis to
- send their requests over to the CDR engine, and synchronize on
- the channel Stasis topic via a subscription so that they return
- their values/control to the dialplan at the appropriate time.
- While going through this, the following changes were also made: *
- DISA, which can reset the CDR when a user successfully
- authenticates, now just uses the ResetCDR app to do this. This
- prevents having to duplicate the same Stasis synchronization
- logic in that application. * Answer no longer disables CDRs. It
- actually didn't work anyway - calling DISABLE on the channel's
- CDR doesn't stop the CDR from getting the Answer time - it just
- kills all CDRs on that channel, which isn't what the caller would
- intend. (closes issue ASTERISK-22884) (closes issue
- ASTERISK-22886) Review: https://reviewboard.asterisk.org/r/3057/
- ........ Merged revisions 404294 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-19 00:32 +0000 [r404293] Damien Wedhorn <voip@facts.com.au>
-
- * /, channels/chan_skinny.c: Fixup skinny registration following
- network issues. On session registration, if device is already
- reporting that it is connected to a device, an innocuous packet
- (update time) is sent to the already connected device. If the tcp
- connection is down, the device will be unregistered and the new
- connection allowed. Without this patch, network issues can see a
- situation where a device can not reregister until after
- 3*timeout. ........ Merged revisions 404292 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 23:00 +0000 [r404280] Jason Parker <jparker@digium.com>
-
- * main/manager.c, /: Add AMI event for presence state. Review:
- https://reviewboard.asterisk.org/r/3039/ ........ Merged
- revisions 404275 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404279 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 21:12 +0000 [r404264] Richard Mudgett <rmudgett@digium.com>
-
- * addons/ooh323c/src/ooTimer.c, /: ooh323c: Fix gcc 4.6.3 compiler
- warnings. ........ Merged revisions 404212 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404219 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404263 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 20:48 +0000 [r404260-404262] Kevin Harwell <kharwell@digium.com>
-
- * channels/chan_oss.c, /: chan_oss.c: channel being locked twice
- and unlocked once Removed channel lock as it is now being down in
- ast_channel_alloc ........ Merged revisions 404261 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c,
- addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c,
- channels/chan_pjsip.c, res/parking/parking_manager.c,
- channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c,
- funcs/func_timeout.c, /, apps/app_meetme.c, main/bridge.c,
- tests/test_stasis_channels.c, include/asterisk/channel.h,
- channels/chan_gtalk.c, channels/sig_pri.c, apps/app_queue.c,
- main/cel.c, main/stasis_bridges.c, channels/chan_jingle.c,
- channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c,
- channels/sig_analog.c, include/asterisk/stasis_channels.h,
- res/res_agi.c, channels/chan_motif.c, tests/test_cel.c,
- apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c,
- apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc,
- addons/chan_ooh323.c, main/pickup.c, include/asterisk/aoc.h,
- include/asterisk/stasis_bridges.h, apps/app_userevent.c,
- apps/app_disa.c, channels/chan_console.c,
- include/asterisk/channelstate.h, main/core_local.c,
- channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c,
- res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
- main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c:
- channel locking: Add locking for channel snapshot creation
- Original commit message by mmichelson (asterisk 12 r403311):
- "This adds channel locks around calls to create channel snapshots
- as well as other functions which operate on a channel and then
- end up creating a channel snapshot. Functions that expect the
- channel to be locked prior to being called have had their
- documentation updated to indicate such." The above was initially
- committed and then reverted at r403398. The problem was found to
- be in core_local.c in the publish_local_bridge_message function.
- The ast_unreal_lock_all function locks and adds a reference to
- the returned channels and while they were being unlocked they
- were not being unreffed when no longer needed. Fixed by unreffing
- the channels. Also in bridge.c a lock was obtained on
- "other->chan", but then an attempt was made to unlock "other" and
- not the previously locked channel. Fixed by unlocking
- "other->chan" (closes issue ASTERISK-22709) Reported by: John
- Bigelow ........ Merged revisions 404237 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 19:36 +0000 [r404211] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c, configs/ooh323.conf.sample: Introduce new
- config option 'aniasdni'. If yes then asterisk set dialed number
- as own id back to the caller on incoming h.323 calls. Option can
- be set globally or per user section. (closes issue
- ASTERISK-22020) Reported by: Ross Beer
-
-2013-12-18 19:28 +0000 [r404210] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_mgcp.c, main/pbx.c, channels/chan_sip.c,
- apps/confbridge/conf_chan_record.c, tests/test_app.c,
- tests/test_stasis_channels.c, main/core_unreal.c,
- include/asterisk/channel.h, channels/chan_console.c,
- channels/chan_oss.c, channels/chan_jingle.c,
- channels/chan_misdn.c, channels/chan_h323.c, tests/test_cel.c,
- channels/chan_nbs.c, channels/chan_pjsip.c, res/res_calendar.c,
- apps/app_voicemail.c, channels/chan_unistim.c,
- tests/test_substitution.c, channels/chan_vpb.cc,
- addons/chan_ooh323.c, channels/chan_multicast_rtp.c, /,
- apps/app_meetme.c, res/res_stasis_snoop.c, channels/chan_gtalk.c,
- channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
- channels/chan_phone.c, channels/chan_skinny.c,
- res/parking/parking_tests.c, channels/chan_motif.c,
- tests/test_voicemail_api.c, channels/chan_alsa.c, main/message.c,
- addons/chan_mobile.c, tests/test_cdr.c: channels: Return
- allocated channels locked. This change makes ast_channel_alloc
- return allocated channels locked. By doing so no other thread can
- acquire, lock, and manipulate the channel before it is completely
- set up. (closes issue AST-1256) Review:
- https://reviewboard.asterisk.org/r/3067/ ........ Merged
- revisions 404204 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 19:10 +0000 [r404198] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c: Implement module reload command for
- chan_ooh323 (close issue ASTERISK-22817) Patches:
- ooh323_module_reload.patch
-
-2013-12-18 12:46 +0000 [r404185] Matthew Jordan <mjordan@digium.com>
-
- * rest-api/api-docs/applications.json,
- rest-api/api-docs/playbacks.json,
- rest-api/api-docs/channels.json, rest-api/api-docs/sounds.json,
- rest-api/resources.json, rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json,
- rest-api/api-docs/deviceStates.json,
- rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
- /, rest-api/api-docs/asterisk.json: ari: Bump the version of ARI
- to 1.0.0 (closes issue ASTERISK-23007) ........ Merged revisions
- 404184 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 12:01 +0000 [r404138] Joshua Colp <jcolp@digium.com>
-
- * res/res_calendar.c, /: res_calendar: Protect channel when adding
- datastore. This change adds a missing channel lock when adding a
- datastore to a channel. ........ Merged revisions 404135 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404136 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404137 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 00:36 +0000 [r404100] Rusty Newton <rnewton@digium.com>
-
- * /, funcs/func_strings.c: func_strings: Documentation fix for
- QUOTE() Example output was inaccurate. (issue ASTERISK-22970)
- (closes issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
- func_strings.patch uploaded by Gareth Palmer (license 5169)
- ........ Merged revisions 404081 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 404087 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 404099 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-18 00:17 +0000 [r404051] Matthew Jordan <mjordan@digium.com>
-
- * /, LICENSE: LICENSE: Update language to include ARI ........
- Merged revisions 404050 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 23:57 +0000 [r404049] Jonathan Rose <jrose@digium.com>
-
- * /, tests/test_cel.c, tests/test_cdr.c: tests: fix
- ast_bridge_base_new calls not using the additional arguments
- r404042 gave ast_bridge_base_new two new arguments for setting a
- bridge creator and name. Unfortunately since a couple test
- modules aren't compiled by default, I missed the fact that this
- change impacted those tests and caused compilation failures
- against them. ........ Merged revisions 404048 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 23:38 +0000 [r404047] Rusty Newton <rnewton@digium.com>
-
- * include/asterisk/test.h, main/channel.c, main/rtp_engine.c, /,
- channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c:
- Several components: fixing Typos in comments and code,
- "avaliable" instead of "available" (issue ASTERISK-23021) (closes
- issue ASTERISK-23021) Reported by: Jeremy Lainé Tested by: Rusty
- Newton Patches: available.patch uploaded by Jeremy Lainé (license
- 6561) ........ Merged revisions 404046 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 23:25 +0000 [r404043] Jonathan Rose <jrose@digium.com>
-
- * apps/app_bridgewait.c, res/ari/ari_model_validators.c,
- doc/appdocsxml.xslt, main/stasis_bridges.c,
- rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
- apps/app_agent_pool.c, res/parking/parking_bridge.c,
- res/ari/ari_model_validators.h, main/manager_bridges.c,
- res/ari/resource_bridges.h, include/asterisk/bridge_internal.h,
- apps/app_confbridge.c, res/res_stasis.c,
- include/asterisk/bridge.h, res/res_ari_bridges.c, /,
- main/bridge.c, main/bridge_basic.c,
- include/asterisk/stasis_bridges.h, include/asterisk/stasis_app.h:
- bridging: Give bridges a name and a known creator Bridges have
- two new optional properties, a creator and a name. Certain
- consumers of bridges will automatically provide bridges that they
- create with these properties. Examples include app_bridgewait,
- res_parking, app_confbridge, and app_agent_pool. In addition, a
- name may now be provided as an argument to the POST function for
- creating new bridges via ARI. (closes issue AFS-47) Review:
- https://reviewboard.asterisk.org/r/3070/ ........ Merged
- revisions 404042 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 18:35 +0000 [r404028-404030] Joshua Colp <jcolp@digium.com>
-
- * res/res_sorcery_config.c, /: res_sorcery_config: Output an error
- message when an object can't be created. If object creation fails
- an error message will now be output with the id, type, and
- configuration file. ........ Merged revisions 404029 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/framehook.c: framehooks: Re-iterate if framehook provides
- different frame. Framehooks can be used in a reactive manner to
- execute specific logic when a frame is received with a certain
- type and payload. Since it is possible for framehooks to provide
- frames it was possible for this reactive framehook to be unaware
- of frames it is looking for. This change makes it so that when
- framehooks return a modified frame the code will now re-iterate
- (from the beginning) and call any previous framehooks that have
- not provided a modified frame themselves. Review:
- https://reviewboard.asterisk.org/r/3046/ ........ Merged
- revisions 404027 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 14:41 +0000 [r404008-404009] David M. Lee <dlee@digium.com>
-
- * /, configs/asterisk.conf.sample, main/asterisk.c: Changed the
- default for live_dangerously to no ........ Merged revisions
- 404006 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/pjsip, /: Setting svn:ignore ........ Merged revisions
- 403748 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-17 12:59 +0000 [r403994] Matthew Jordan <mjordan@digium.com>
-
- * /, res/ari/resource_channels.c: ari/resource_channels: When
- creating a channel, specify a default format (SLIN) When creating
- channels via ARI, the current code fails to provide any default
- format capabilities. For non-virtual channels this isn't really a
- problem - the channels typically receive their capabilities as a
- result of the underlying channel driver configuration. For
- virtual channels (such as Local channels), the lack of any format
- capabilities causes the Asterisk core to make some 'odd' choices
- with respect to the translation paths. The issue reporter had
- some paths that had 3 hops on each channel leg, causing multiple
- transcodings and some really crappy audio/performance. By
- specifying a baseline of SLIN, we prevent that from occurring.
- Note that this is what AMI does when it performs an Originate, as
- does res_clioriginate. Review:
- https://reviewboard.asterisk.org/r/3068/ (issue ASTERISK-22962)
- Reported by: Matt DiMeo ........ Merged revisions 403993 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-16 19:11 +0000 [r403960] David M. Lee <dlee@digium.com>
-
- * include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c,
- main/pbx.c, main/tcptls.c, funcs/func_db.c, /,
- README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample,
- funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c,
- UPGRADE-12.txt: security: Inhibit execution of privilege
- escalating functions This patch allows individual dialplan
- functions to be marked as 'dangerous', to inhibit their execution
- from external sources. A 'dangerous' function is one which
- results in a privilege escalation. For example, if one were to
- read the channel variable SHELL(rm -rf /) Bad Things(TM) could
- happen; even if the external source has only read permissions.
- Execution from external sources may be enabled by setting
- 'live_dangerously' to 'yes' in the [options] section of
- asterisk.conf. Although doing so is not recommended. Also, the
- ABI was changed to something more reasonable, since Asterisk 12
- does not yet have a public release. (closes issue ASTERISK-22905)
- Review: http://reviewboard.digium.internal/r/432/ ........ Merged
- revisions 403913 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 403917 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403959 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-16 18:31 +0000 [r403958] Jonathan Rose <jrose@digium.com>
-
- * /, main/bridge.c: transfers: Fix bug setting both BLINDTRANSFER
- and ATTENDEDTRANSFER The ast_bridge_set_transfer_variables
- function is supposed to wipe whichever variable isn't being set.
- Instead it was setting both to the new value. Oops. (issue
- AFS-24) ........ Merged revisions 403957 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-16 16:12 +0000 [r403857-403865] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/pbx.c, /: pbx.c: put copy of ast_exten.data on stack to
- prevent memory corruption During dialplan execution in
- pbx_extension_helper(), the contexts global read lock prevents
- link list corruption, but was released with a pointer to the
- ast_exten and data later used in variable substitution. Instead,
- this patch removes pbx_substitute_variables() and locates a copy
- of the ast_exten data on the stack before releasing the lock,
- where ast_exten could get free'd by another thread performing a
- module reload. (issue AST-1179) Reported by: Thomas Arimont
- (issue AST-1246) Reported by: Alexander Hömig Review:
- https://reviewboard.asterisk.org/r/3055/ ........ Merged
- revisions 403862 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 403863 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403864 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_sms.c: app_sms: BufferOverflow when receiving odd
- length 16 bit message This patch prevents an infinite loop
- overwriting memory when a message is received into the
- unpacksms16() function, where the length of the message is an odd
- number of bytes. (closes issue ASTERISK-22590) Reported by: Jan
- Juergens Tested by: Jan Juergens ........ Merged revisions 403856
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-15 01:39 +0000 [r403824] Matthew Jordan <mjordan@digium.com>
-
- * channels/pjsip/dialplan_functions.c, /: pjsip/dialplan_functions:
- Use the right buffer length when printing URIs While
- entertaining, sizeof(buflen) is not the same as buflen. Doh.
- ........ Merged revisions 403823 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-14 17:28 +0000 [r403810-403812] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c,
- res/res_pjsip/pjsip_options.c, res/res_pjsip.c: res_pjsip: Apply
- outbound proxy to all SIP requests. Objects which are involved in
- SIP request creation and sending now allow an outbound proxy to
- be specified. For cases where an endpoint is used the outbound
- proxy specified there will be applied. (closes issue
- ASTERISK-22673) Reported by: Antti Yrjola Review:
- https://reviewboard.asterisk.org/r/3022/ ........ Merged
- revisions 403811 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_channels.c, apps/app_queue.c,
- res/ari/ari_model_validators.c, apps/app_dial.c,
- res/ari/ari_model_validators.h, main/dial.c,
- include/asterisk/stasis_channels.h,
- rest-api/api-docs/events.json, /, res/stasis/app.c: res_stasis:
- Expose event for call forwarding and follow forwarded channel.
- This change adds an event for when an originated call is
- redirected to another target. This event contains the original
- channel and the newly created channel. If a stasis subscription
- exists on the original originated channel for a stasis
- application then a new subscription will also be created on the
- stasis application to the redirected channel. This allows the
- application to follow the call path completely. (closes issue
- ASTERISK-22719) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/3054/ ........ Merged
- revisions 403808 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 21:35 +0000 [r403797] Jonathan Rose <jrose@digium.com>
-
- * /, res/res_pjsip_messaging.c, main/message.c: documentation: Add
- PJSIP technology to messaging documentation ........ Merged
- revisions 403796 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 20:17 +0000 [r403784] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/test.c: test.c: Fix too sticky unit test failed status.
- Rerunning a failed unit test after loading any required modules
- should allow the test to report a pass status if it now passes.
- ........ Merged revisions 403782 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 20:13 +0000 [r403783] Jonathan Rose <jrose@digium.com>
-
- * /, main/bridge.c, main/bridge_basic.c, include/asterisk/bridge.h,
- res/parking/parking_bridge_features.c,
- res/parking/parking_manager.c: Transfers: Make Asterisk set
- ATTENDEDTRANSFER/BLINDTRANSFER more reliably There were still a
- few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be
- set on channels involved with blind and attended transfers. This
- would happen with features that were initialized by channel
- driver specific mechanisms in multiparty calls. This patch
- resolves those cases while attempted to keep the behavior for
- setting those variables as consistent as possible. (closes issue
- AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........
- Merged revisions 403781 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 18:33 +0000 [r403750-403768] Kevin Harwell <kharwell@digium.com>
-
- * main/channel.c, /, channels/chan_sip.c,
- include/asterisk/channel.h, bridges/bridge_native_rtp.c,
- channels/chan_pjsip.c: bridge_native_rtp: Deadlock during 4-way
- conference creation The change contains a slightly adjusted patch
- that was on the issue (submitted by kmoore). A fix was made by
- adding in a bridge lock while calling bridge_start/stop from the
- framehook callback. Since the framehook callback is not called
- from the bridging core the bridge is not locked, but needs to be
- before calling bridge_start. (closes issue ASTERISK-22749)
- Reported by: Kinsey Moore Review:
- https://reviewboard.asterisk.org/r/3066/ Patches:
- lock_inversion.diff uploaded by kmoore (license 6273) ........
- Merged revisions 403767 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c, res/ari/resource_channels.h, /,
- main/http.c: ARI: Allow specifying channel variables during a
- POST /channels Added the ability to specify channel variables
- when creating/originating a channel in ARI. The variables are
- sent in the body of the request and should be formatted as a
- single level JSON object. No nested objects allowed. For example:
- {"variable1": "foo", "variable2": "bar"}. (closes issue
- ASTERISK-22872) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3052/ ........ Merged
- revisions 403752 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_answer.c, rest-api/api-docs/bridges.json,
- res/ari/resource_bridges.c, res/res_ari_bridges.c,
- res/stasis/command.c, res/res_stasis_playback.c, /,
- res/stasis/control.c, res/stasis/command.h,
- include/asterisk/stasis_app.h,
- include/asterisk/stasis_app_impl.h, res/res_stasis_recording.c:
- ARI: Adding a channel to a bridge while a live recording is
- active blocks Added the ability to have rules that are checked
- when adding and/or removing channels to/from a bridge. In this
- case, if a channel is currently recording and someone attempts to
- add it to a bridge an "is recording" rule is checked, fails, and
- a 409 conflict is returned. Also command functions now return an
- integer value that can be descriptive of what kind of problems,
- if any, occurred before or during execution. (closes issue
- ASTERISK-22624) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/2947/ ........ Merged
- revisions 403749 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 05:00 +0000 [r403737] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/Makefile: channels/Makefile: clean pjsip directory
- ........ Merged revisions 403736 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-13 00:40 +0000 [r403726] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/app.h, tests/test_voicemail_api.c, main/app.c:
- test_voicemail_api: Add check for a registered voicemail provider
- before tests. It is much nicer diagnosing a test failure if
- app_voicemail is actually loaded.
-
-2013-12-12 19:46 +0000 [r403714] Scott Griepentrog <sgriepentrog@digium.com>
-
- * contrib/ast-db-manage/config/versions/581a4264e537_adding_extensions.py
- (added), /: realtime: Create extensions in alembic ast-db-manage
- contribution When the alembic scripts were written for creating
- Asterisk realtime databases the extensions table for dialplan
- wasn't included. This update creates the extensions table.
- (closes issue ASTERISK-22815) Reported by: Zone Conkle Review:
- https://reviewboard.asterisk.org/r/3064/ ........ Merged
- revisions 403713 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-12 19:18 +0000 [r403707] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_pjsip.c: chan_pjsip: Revert r403587 This patch
- was intended to eliminate a deadlock that occurs when masquerades
- occur in pjsip channels, but has some potential side effects.
- Mark Michelson is currently working on addressing this problem
- from another angle. (issue ASTERISK-22936) Reported by: Jonathan
- Rose ........ Merged revisions 403705 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-11 20:24 +0000 [r403687] Kevin Harwell <kharwell@digium.com>
-
- * include/asterisk/res_pjsip.h, res/res_pjsip/config_global.c, /,
- configs/pjsip.conf.sample, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip_messaging.c,
- res/res_pjsip/include/res_pjsip_private.h, res/res_pjsip.c:
- res_pjsip_messaging: send message to a default outbound endpoint
- In some cases messages need to be sent to a direct URI (sip:<ip
- address>). This patch adds in that support by using a default
- outbound endpoint. When sending messages, if no endpoint can be
- found then the default one is used. To facilitate this a new
- default_outbound_endpoint option was added to the globals section
- for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/
- ........ Merged revisions 403680 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-11 19:22 +0000 [r403652] Russell Bryant <russell@russellbryant.com>
-
- * /, channels/chan_sip.c: Reset peer outboundproxy on sip.conf
- reload If you set a peer's outboundproxy and then removed it from
- the config, this would not get picked up in a config reload. This
- patch fixes that by resetting it in set_peer_defaults(). Closes
- ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
- ........ Merged revisions 403634 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 403635 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403639 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-11 19:19 +0000 [r403643] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_voicemail.c, include/asterisk/app.h,
- include/asterisk/doxyref.h, main/app.c: app_voicemail: Voicemail
- callback registration/unregistration function improvements. * The
- voicemail registration/unregistration functions now take a struct
- of callbacks instead of a lengthy parameter list of callbacks. *
- The voicemail registration/unregistration functions now prevent a
- competing module from interfering with an already registered
- callback supplying module.
-
-2013-12-11 13:06 +0000 [r403617-403619] Matthew Jordan <mjordan@digium.com>
-
- * channels/pjsip/dialplan_functions.c,
- include/asterisk/res_pjsip_session.h, channels/pjsip (added), /,
- funcs/func_channel.c, channels/pjsip/include,
- channels/pjsip/include/dialplan_functions.h, res/res_pjsip_t38.c,
- channels/pjsip/include/chan_pjsip.h, channels/Makefile,
- channels/chan_pjsip.c, main/xmldoc.c: func_channel, chan_pjsip:
- Add CHANNEL read function support for chan_pjsip This patch adds
- CHANNEL read support for chan_pjsip. This allows the dialplan to
- use the CHANNEL function on a chan_pjsip channel to obtain
- run-time information about the channel from the PJSIP channel
- driver and the PJSIP stack. This includes: * RTP information,
- including source/destination media addresses, whether or not the
- media is secure, held, and other properties. * RTCP information.
- This includes sets of parseable information, as well as
- individual statistic attriutes. * PJSIP information. This
- includes URIs, local/remote signalling addresses, whether or not
- the signalling is secure, and other properties. * The endpoint
- name. This can be used in conjunction with the PJSIP_ENDPOINT
- function to obtain more detailed endpoint information. Review:
- https://reviewboard.asterisk.org/r/3038/ ........ Merged
- revisions 403618 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * Makefile, funcs/func_pjsip_endpoint.c (added), doc/snapshots.xslt
- (removed), /, doc/appdocsxml.xslt (added), doc/appdocsxml.dtd,
- main/sorcery.c: func_pjsip_endpoint: Add PJSIP_ENDPOINT function
- for querying endpoint details This patch adds a new function,
- PJSIP_ENDPOINT, which lets the dialplan query, for any endpoint,
- any property configured on an endpoint. This function is a
- companion to the CHANNEL function, which can be used to extract
- the endpoint name for a channel. Review:
- https://reviewboard.asterisk.org/r/3035 ........ Merged revisions
- 403616 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-10 15:15 +0000 [r403605] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_authenticator_digest.c: Fix correct authentication
- behavior for artificial endpoint. When switching to using a
- vector for authentication, I initialized the vector for the
- artificial endpoint to be of size 1. However, this does not
- result in AST_VECTOR_SIZE() returning 1 since there isn't
- actually anything in the vector. Rather than trifle with the
- vector by putting unnecessary elements in, I simply changed the
- callback in res_pjsip_authenticator_digest.c to explicitly report
- that the artificial endpoint requires authentication. Thanks to
- Joshua Colp for pointing this out.
-
-2013-12-09 22:59 +0000 [r403576-403588] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_pjsip.c: chan_pjsip: Fix a sticking channel lock
- caused by channel masquerades (closes issue ASTERISK-22936)
- Reported by: Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/3042/ ........ Merged
- revisions 403587 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * CHANGES, main/dial.c, apps/app_page.c, include/asterisk/dial.h:
- app_page: Add predial handlers for app_page. (closes issue
- AFS-14) Review: https://reviewboard.asterisk.org/r/3045/
-
-2013-12-09 19:24 +0000 [r403544-403560] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_sorcery_astdb.c: Reverting regex part of -r403545 at
- request of file. res_sorcery_astdb.c: Fix get multiple records by
- regex. * Fix sorcery_astdb_retrieve_regex() pattern matching. Let
- the regexec() function match the stored key values instead of
- having astdb prefilter them. Previoiusly you could only use a
- simple regex pattern when the pattern began with '^'. ........
- Merged revisions 403559 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_sorcery_astdb.c: res_sorcery_astdb.c: Fix get multiple
- records by regex. * Fix sorcery_astdb_retrieve_regex() pattern
- matching. Let the regexec() function match the stored key values
- instead of having astdb prefilter them. Previoiusly you could
- only use a simple regex pattern when the pattern began with '^'.
- * Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
- ........ Merged revisions 403545 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/sorcery.c, /: sorcery: Eliminate shadowing a varaible that
- caused confusion. * Eliminated shadowing of the
- __ast_sorcery_apply_config() name parameter causing confusion. *
- Fix potential crash from sorcery.conf user input in
- __ast_sorcery_apply_config() if the user supplied a malformed
- config line that is missing the sorcery object type name. *
- Remove redundant test in __ast_sorcery_apply_config(). !config
- and config == CONFIGS_STATUS_FILEMISSING are identical. ........
- Merged revisions 403541 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-09 18:32 +0000 [r403543] Joshua Colp <jcolp@digium.com>
-
- * /, main/endpoints.c: endpoints: Keep a reference to channel ids
- when creating snapshot. The snapshot process for endpoints uses
- the channel ids present on the endpoint itself. Without keeping a
- reference it was possible for the strings to be freed underneath
- any consumer of an endpoint snapshot. A reference is now held by
- the snapshot to the channel ids and released when the snapshot is
- destroyed. (issue ASTERISK-22801) Reported by: Matt Jordan
- ........ Merged revisions 403542 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-09 18:14 +0000 [r403528] Richard Mudgett <rmudgett@digium.com>
-
- * main/sorcery.c, /: sorcery: Whitespace You would think that a new
- file would start off without any whitespace oddities. ........
- Merged revisions 403527 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-09 17:29 +0000 [r403512-403526] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_confbridge.c, CHANGES,
- apps/confbridge/conf_state_multi_marked.c: Add a
- CONFBRIDGE_RESULT channel variable to discern why a channel left
- a ConfBridge. Review: https://reviewboard.asterisk.org/r/3009
-
- * CHANGES, apps/app_mixmonitor.c: Create function for retrieving
- Mixmonitor instance data. For the time, this is only useful for
- retrieving the filename. The purpose of this function is to
- better facilitate multiple mixmonitors per channel. Setting a
- MIXMONITOR_FILENAME channel variable is not conducive to such
- behavior, so allowing finer grained access to individual
- mixmonitor properties improves the situation. The
- MIXMONITOR_FILENAME channel variable is still set, though, so
- there is no worry about backwards compatibility. Review:
- https://reviewboard.asterisk.org/r/3023
-
-2013-12-09 16:41 +0000 [r403511] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_nat.c, /: res_pjsip_nat: Add NAT module to session
- dialogs. Due to the way pjproject internally works it was
- possible for the NAT module to not be invoked on messages with-in
- a session dialog. This means that the various parts of the
- message would not get rewritten with the source IP address and
- port. This change uses a session supplement to add the NAT module
- to the dialog on the first incoming or outgoing INVITE. (closes
- issue ASTERISK-22941) Reported by: Leif Madsen ........ Merged
- revisions 403510 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-09 16:10 +0000 [r403499] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip/config_auth.c,
- res/res_pjsip_outbound_authenticator_digest.c,
- res/res_pjsip_authenticator_digest.c,
- res/res_pjsip_outbound_registration.c,
- res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip/pjsip_distributor.c, res/res_pjsip.c,
- include/asterisk/res_pjsip.h: Switch PJSIP auth to use a vector.
- Since Asterisk has a vector API now, places where arrays are
- manually resized don't really make sense any more. Since the auth
- work in PJSIP was freshly-written, it was easy to reform it to
- use a vector. Review: https://reviewboard.asterisk.org/r/3044
-
-2013-12-09 03:21 +0000 [r403436-403466] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38
- session to avoid crashes during state change Prior to this patch,
- res_fax_spandsp was conservative with how it initialized the
- spandsp T.38 context. It would only initialize it if the driver
- thought the current state was a T.38 fax. While this works fine
- in nominal situations, in certain off nominal situations,
- res_fax_spandsp can believe that a T.38 fax will not occur when
- in fact one has started. In particular, this was discovered when
- res_fax would fall back to audio after timing out on a T.38
- upgrade. The SIP channel driver would continue to retry the
- re-INVITE and - if the remote end responded after res_fax timed
- out with a 200 OK - a T.38 frame would be delivered to the
- res_fax stack when it no longer expected it. As it turns out,
- there does not appear to be any downside to always initializing
- the T.38 context, other than the actual memory allocation. Since
- that avoids this off nominal situation (and others which are
- equally likely hard to predict), this is the safest way to avoid
- this problem. Much thanks to Torrey as well for providing a
- scenario that reproduces this issue. (closes issue
- ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
- Searle patches: always-init-t38.patch uploaded by awinters
- (License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
- ........ Merged revisions 403449 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 403450 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_config_sqlite.c: res_config_sqlite: Check for CDR
- unregistration failures If the CDR unregistration fails due to an
- inflight CDR, the res_config_sqlite module needs to bail on
- unloading itself. Otherwise, the config could be unloaded
- (including the CDR table name) while the CDR engine posts a CDR
- to the still registered backend, resulting in a crash. ........
- Merged revisions 403435 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-05 23:40 +0000 [r403414] Jonathan Rose <jrose@digium.com>
-
- * apps/app_record.c: app_record: Add an option that allows DTMF '0'
- to act as an additional terminator Using this terminator when
- active results in ${RECORD_STATUS} being set to 'OPERATOR'
- instead of 'DTMF' (closes issue AFS-7) Review:
- https://reviewboard.asterisk.org/r/3041/
-
-2013-12-05 22:10 +0000 [r403402-403404] David M. Lee <dlee@digium.com>
-
- * addons/chan_mobile.c, main/bridge_channel.c, tests/test_cdr.c,
- channels/chan_pjsip.c, res/parking/parking_manager.c,
- channels/chan_mgcp.c, channels/chan_unistim.c, main/pbx.c, /,
- apps/app_meetme.c, funcs/func_timeout.c, main/bridge.c,
- tests/test_stasis_channels.c, main/core_unreal.c,
- include/asterisk/channel.h, channels/chan_gtalk.c, main/cel.c,
- apps/app_queue.c, channels/sig_pri.c, main/stasis_bridges.c,
- channels/chan_jingle.c, channels/chan_phone.c,
- channels/chan_dahdi.c, main/dial.c, channels/sig_analog.c,
- include/asterisk/stasis_channels.h, res/res_agi.c,
- channels/chan_motif.c, channels/chan_h323.c, tests/test_cel.c,
- apps/app_confbridge.c, res/res_stasis.c, res/res_pjsip_refer.c,
- apps/app_voicemail.c, apps/app_dial.c, channels/chan_vpb.cc,
- addons/chan_ooh323.c, channels/chan_sip.c, main/pickup.c,
- include/asterisk/aoc.h, include/asterisk/stasis_bridges.h,
- apps/app_userevent.c, apps/app_disa.c, main/core_local.c,
- include/asterisk/channelstate.h, channels/chan_console.c,
- channels/chan_iax2.c, main/endpoints.c, channels/chan_oss.c,
- res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
- main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c,
- pbx/pbx_realtime.c, channels/chan_alsa.c, main/stasis_channels.c,
- channels/chan_nbs.c: Reverting r403311. It's causing ARI tests to
- hang. ........ Merged revisions 403398 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/stasis/control.c: ari: Fix deadlock problem with functions
- that use autoservice. The code for getting channel variables from
- ARI assumed that you needed to lock the channel in order to
- properly execute functions and read channel variables.
- Apparently, this is not the case, since any dialplan function
- that puts the channel into autoservice deadlocks when attempting
- to remove the channel from autoservice. ........ Merged revisions
- 403342 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /: Multiple revisions 403304,403310 ........ r403304 | dlee |
- 2013-12-02 12:34:50 -0600 (Mon, 02 Dec 2013) | 1 line Fixed the
- filename for the ari.conf docs ........ r403310 | file |
- 2013-12-03 10:32:12 -0600 (Tue, 03 Dec 2013) | 5 lines Revert
- revision 403304: Fixed the filename for the ari.conf docs The
- changed value refers to the name of the module. The name of the
- configuration file is specified in the configFile section.
- ........ Merged revisions 403304,403310 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-04 21:42 +0000 [r403378] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_registrar.c: res_pjsip_registrar: undefined
- function pointer symbol Used a static wrapper around the
- offending function to alleviate the issue. Reported by: rmudgett
- ........ Merged revisions 403377 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-04 20:54 +0000 [r403365] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_t38.c, /: res_pjsip_t38: Don't pass T.38 control
- frames through to other hooks. This crept up during gateway
- testing where the gateway would receive the request to negotiate
- and assume it came from the remote side, causing the gateway
- state machine to go a little, to a use a technical term, "wonky".
- ........ Merged revisions 403364 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-04 18:41 +0000 [r403350] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip.c: Initialize the hash value argument to
- pj_hash_get() to 0. Passing a non-zero value causes PJLIB to use
- the given input as the hash value. Passing zero causes the
- parameter to become an output parameter that receives the hash
- value that was computed based on the given key. This change
- essentially makes ast_sip_dict_get() properly retrieve the
- desired value. ........ Merged revisions 403349 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-03 18:01 +0000 [r403330] Joshua Colp <jcolp@digium.com>
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- res/res_pjsip_session.c: res_pjsip_session: Add support for
- PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. Newer versions of PJSIP
- have changed to using a flag for the
- PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds
- a configure check to detect the presence of the flag and use it
- if found. ........ Merged revisions 403329 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-03 17:35 +0000 [r403327] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/sorcery.h, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip_registrar_expire.c, res/res_pjsip/pjsip_options.c,
- tests/test_sorcery.c, include/asterisk/bucket.h, main/sorcery.c,
- /, main/bucket.c: sorcery, bucket: Change observer remove calls
- to take const callbacks struct. * Make
- ast_sorcery_observer_remove() accept a const callbacks struct. *
- Make ast_sorcery_observer_remove() tolerant of the sorcery
- parameter being NULL. Now it can be called within a module unload
- routine if the sorcery initialization fails. * Fix
- ast_sorcery_observer_add() to fail if the container link fails.
- ........ Merged revisions 403324 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-03 17:07 +0000 [r403314] Mark Michelson <mmichelson@digium.com>
-
- * channels/chan_nbs.c, main/bridge_channel.c, res/res_stasis.c,
- channels/chan_pjsip.c, res/parking/parking_manager.c,
- apps/app_voicemail.c, channels/chan_unistim.c,
- channels/chan_vpb.cc, addons/chan_ooh323.c, /,
- include/asterisk/aoc.h, apps/app_meetme.c, main/bridge.c,
- apps/app_userevent.c, channels/chan_gtalk.c,
- channels/chan_iax2.c, main/endpoints.c, main/stasis_bridges.c,
- main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
- main/dial.c, channels/sig_analog.c, channels/chan_skinny.c,
- res/res_agi.c, channels/chan_motif.c, pbx/pbx_realtime.c,
- channels/chan_alsa.c, main/stasis_channels.c,
- apps/app_confbridge.c, addons/chan_mobile.c, tests/test_cdr.c,
- res/res_pjsip_refer.c, channels/chan_mgcp.c, apps/app_dial.c,
- main/pbx.c, channels/chan_sip.c, main/pickup.c,
- funcs/func_timeout.c, tests/test_stasis_channels.c,
- main/core_unreal.c, include/asterisk/stasis_bridges.h,
- apps/app_disa.c, include/asterisk/channel.h, main/core_local.c,
- include/asterisk/channelstate.h, channels/chan_console.c,
- main/cel.c, apps/app_queue.c, channels/sig_pri.c,
- channels/chan_oss.c, res/parking/parking_bridge_features.c,
- apps/app_agent_pool.c, channels/chan_jingle.c,
- channels/chan_misdn.c, include/asterisk/stasis_channels.h,
- channels/chan_h323.c, tests/test_cel.c: Add channel locking for
- channel snapshot creation. This adds channel locks around calls
- to create channel snapshots as well as other functions which
- operate on a channel and then end up creating a channel snapshot.
- Functions that expect the channel to be locked prior to being
- called have had their documentation updated to indicate such.
- ........ Merged revisions 403311 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-03 16:39 +0000 [r403313] Joshua Colp <jcolp@digium.com>
-
- * main/media_index.c, /: media_index: Make media indexing tolerable
- of bad symlinks. Media indexing will now skip over files and
- directories that stat will not return information about. This can
- occur under normal conditions when a symbolic link points to a
- location that no longer exists. ........ Merged revisions 403312
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-02 18:12 +0000 [r403292] Alexandr Anikin <may@telecom-service.ru>
-
- * addons/chan_ooh323.c, /: Check and reject non-digits e164 values
- on peers and general sections in ooh323.conf Regenerate e164
- endpoint list on reload ooh323 (issue ASTERISK-22901) Reported
- by: Cyril CONSTANTIN Patches: ASTERISK-22901.patch ........
- Merged revisions 403288 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403290 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-12-01 21:13 +0000 [r403257-403272] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_session.c: res_pjsip_session: Apply fromuser and
- fromdomain to all requests as documented. ........ Merged
- revisions 403271 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_t38.c, /: res_pjsip_t38: Add the framehook to the
- channel only on first INVITE. The check for determining whether
- the T.38 framehook should be added to the channel or not has now
- been changed to guarantee adding only occurs on the first
- incoming or outgoing INVITE. ........ Merged revisions 403258
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip/security_events.c, res/res_pjsip/pjsip_options.c,
- res/res_pjsip.c, res/res_pjsip_transport_websocket.c,
- include/asterisk/res_pjsip.h, /, res/res_pjsip/location.c:
- res_pjsip_transport_websocket: Fix security events and simplify
- implementation. Transport type determination for security events
- has been simplified to use the type present on the message itself
- instead of searching through configured transports to find the
- transport used. The actual WebSocket transport has also been
- simplified. It now leverages the existing PJSIP transport manager
- for finding the active WebSocket transport for outgoing messages.
- This removes the need for res_pjsip_transport_websocket to store
- a mapping itself. (closes issue ASTERISK-22897) Reported by: Max
- E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/
- ........ Merged revisions 403256 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-30 14:12 +0000 [r403241] Joshua Colp <jcolp@digium.com>
-
- * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /,
- res/ari/ari_model_validators.c: res_ari: Add Recording events to
- the validator. ........ Merged revisions 403240 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-28 02:12 +0000 [r403208-403224] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Don't produce an
- invalid media stream with no formats. Depending on configuration
- it was possible for a media stream to be created without any
- media formats. The produced SDP would fail internal validation
- and cause a crash. The code will now no longer add media streams
- with no formats to the SDP, allowing it to pass validation and
- work. (closes issue ASTERISK-22858) Reported by: Anthony Messina
- ........ Merged revisions 403223 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_header_funcs.c, /: res_pjsip_header_funcs: Don't
- add headers to re-INVITEs. When sending a re-INVITE to an
- endpoint it was possible for received headers to be added as well
- (since they are stored for retrieval using the PJSIP_HEADER
- dialplan function). This caused a broken (and potentially large)
- SIP INVITE to be produced and sent. This changes the module so it
- will no longer add headers to re-INVITEs. (closes issue
- ASTERISK-22882) Reported by: David M. Lee ........ Merged
- revisions 403221 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_playback.c, /: res_stasis_playback: Add 'number',
- 'digits', and 'characters' URI scheme implementations. This
- change adds new URI scheme implementations for playing numbers,
- digits, and characters. This is done as part of the normal
- playback mechanism and can be used with queueing to create a
- combined sentence. Review:
- https://reviewboard.asterisk.org/r/3028/ ........ Merged
- revisions 403209 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c,
- res/res_pjsip_session.c, include/asterisk/res_pjsip.h:
- res_pjsip_session: Add configurable behavior for redirects. The
- action taken when a redirect occurs is now configurable on a
- per-endpoint basis. The redirect can either be treated as a
- redirect to a local extension, to a URI that is dialed through
- the Asterisk core, or to a URI that is dialed within PJSIP
- itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged
- revisions 403207 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-27 17:32 +0000 [r403192] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/astdb.h: astdb: Tweak some doxygen comments.
-
-2013-11-27 16:12 +0000 [r403180] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Fix crash when
- reloading certain configurations. Certain options available that
- specify a SIP URI perform validation on the provided URI using
- the PJSIP URI parser. This operation requires that the thread
- executing it be registered with the PJLIB library. During reloads
- this was done on a thread which was NOT registered with it. This
- fixes the problem by creating a task which reloads the
- configuration on a PJSIP thread. (closes issue ASTERISK-22923)
- Reported by: Anthony Messina ........ Merged revisions 403179
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-27 15:48 +0000 [r403177] David M. Lee <dlee@digium.com>
-
- * res/res_ari_channels.c, include/asterisk/ari.h,
- rest-api-templates/param_parsing.mustache,
- include/asterisk/http.h, res/res_ari_recordings.c,
- res/res_ari_endpoints.c, main/http.c,
- rest-api-templates/swagger_model.py, res/res_ari_playbacks.c,
- res/res_ari_sounds.c, rest-api-templates/asterisk_processor.py,
- res/res_ari_bridges.c, tests/test_ari.c, res/res_ari.c, /,
- res/res_ari_device_states.c, res/res_ari_asterisk.c,
- rest-api-templates/res_ari_resource.c.mustache,
- res/res_ari_applications.c: ari:Add application/json parameter
- support The patch allows ARI to parse request parameters from an
- incoming JSON request body, instead of requiring the request to
- come in as query parameters (which is just weird for POST and
- DELETE) or form parameters (which is okay, but a bit asymmetric
- given that all of our responses are JSON). For any operation that
- does _not_ have a parameter defined of type body (i.e.
- "paramType": "body" in the API declaration), if a request
- provides a request body with a Content type of
- "application/json", the provided JSON document is parsed and
- searched for parameters. The expected fields in the provided JSON
- document should match the query parameters defined for the
- operation. If the parameter has 'allowMultiple' set, then the
- field in the JSON document may optionally be an array of values.
- (closes issue ASTERISK-22685) Review:
- https://reviewboard.asterisk.org/r/2994/
-
-2013-11-27 15:31 +0000 [r403161-403174] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/pjsip_configuration.c: res_pjsip: Update
- handling of some options to work with new option names. Some
- options (such as call_group and pickup_group) share the same
- configuration handler and decide what logic to use based on the
- name of the option. These handlers were not updated to check for
- the new option names and were treating the options as invalid.
- This change simply updates the handlers with the proper names of
- the options. (closes issue ASTERISK-22922) Reported by: Anthony
- Messina ........ Merged revisions 403173 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Fix
- a configure issue with PJSIP transaction group lock detection.
- The configure check did not use the provided paths for pjproject
- if provided when looking for transaction group lock support.
- ........ Merged revisions 403160 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-23 17:48 +0000 [r403133-403135] Kevin Harwell <kharwell@digium.com>
-
- * res/ari.make, rest-api/api-docs/applications.json,
- res/ari/resource_device_states.h (added),
- include/asterisk/stasis_app_device_state.h (added),
- res/ari/resource_applications.h, res/res_stasis.c,
- include/asterisk/devicestate.h, rest-api/api-docs/events.json,
- res/res_stasis_device_state.exports.in (added), res/stasis/app.c,
- res/res_ari_device_states.c (added), /,
- include/asterisk/stasis_app.h, main/devicestate.c,
- res/stasis/app.h, rest-api/resources.json,
- res/res_stasis_device_state.c (added),
- res/ari/ari_model_validators.c, res/ari/ari_model_validators.h,
- res/ari/resource_device_states.c (added),
- rest-api/api-docs/deviceStates.json (added),
- rest-api-templates/ari.make.mustache: ARI: Implement device state
- API Created a data model and implemented functionality for an ARI
- device state resource. The following operations have been added
- that allow a user to manipulate an ARI controlled device:
- Create/Change the state of an ARI controlled device PUT
- /deviceStates/{deviceName}&{deviceState} Retrieve all ARI
- controlled devices GET /deviceStates Retrieve the current state
- of a device GET /deviceStates/{deviceName} Destroy a device-state
- controlled by ARI DELETE /deviceStates/{deviceName} The ARI
- controlled device must begin with 'Stasis:'. An example
- controlled device name would be Stasis:Example. A
- 'DeviceStateChanged' event has also been added so that an
- application can subscribe and receive device change events. Any
- device state, ARI controlled or not, can be subscribed to. While
- adding the event, the underlying subscription control mechanism
- was refactored so that all current and future resource
- subscriptions would be the same. Each event resource must now
- register itself in order to be able to properly handle
- [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged
- revisions 403134 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_registrar.c, main/sorcery.c,
- include/asterisk/res_pjsip.h, include/asterisk/acl.h,
- res/res_pjsip/config_auth.c, include/asterisk/utils.h,
- res/res_pjsip.exports.in, /,
- res/res_pjsip_endpoint_identifier_ip.c, main/acl.c, main/utils.c,
- res/res_pjsip.c, res/res_pjsip_exten_state.c,
- include/asterisk/res_pjsip_pubsub.h, res/res_pjsip/location.c,
- res/res_pjsip_outbound_registration.c, res/res_pjsip_mwi.c,
- res/res_pjsip/pjsip_configuration.c, include/asterisk/sorcery.h,
- include/asterisk/strings.h,
- res/res_pjsip/include/res_pjsip_private.h,
- res/res_pjsip_pubsub.c, res/res_pjsip/config_transport.c:
- res_pjsip: AMI commands and events. Created the following AMI
- commands and corresponding events for res_pjsip:
- PJSIPShowEndpoints - Provides a listing of all pjsip endpoints
- and a few select attributes on each. Events: EndpointList - for
- each endpoint a few attributes. EndpointlistComplete - after all
- endpoints have been listed. PJSIPShowEndpoint - Provides a detail
- list of attributes for a specified endpoint. Events:
- EndpointDetail - attributes on an endpoint. AorDetail - raised
- for each AOR on an endpoint. AuthDetail - raised for each
- associated inbound and outbound auth TransportDetail - transport
- attributes. IdentifyDetail - attributes for the identify object
- associated with the endpoint. EndpointDetailComplete - last event
- raised after all detail events. PJSIPShowRegistrationsInbound -
- Provides a detail listing of all inbound registrations. Events:
- InboundRegistrationDetail - inbound registration attributes for
- each registration. InboundRegistrationDetailComplete - raised
- after all detail records have been listed.
- PJSIPShowRegistrationsOutbound - Provides a detail listing of all
- outbound registrations. Events: OutboundRegistrationDetail -
- outbound registration attributes for each registration.
- OutboundRegistrationDetailComplete - raised after all detail
- records have been listed. PJSIPShowSubscriptionsInbound - A
- detail listing of all inbound subscriptions and their attributes.
- Events: SubscriptionDetail - on each subscription detailed
- attributes SubscriptionDetailComplete - raised after all detail
- records have been listed. PJSIPShowSubscriptionsOutbound - A
- detail listing of all outboundbound subscriptions and their
- attributes. Events: SubscriptionDetail - on each subscription
- detailed attributes SubscriptionDetailComplete - raised after all
- detail records have been listed. (issue ASTERISK-22609) Reported
- by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/
- ........ Merged revisions 403131 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-23 12:52 +0000 [r403118-403120] Joshua Colp <jcolp@digium.com>
-
- * res/res_stasis_playback.c, rest-api/api-docs/events.json, /,
- res/res_stasis_recording.c, res/ari/ari_model_validators.c,
- rest-api/api-docs/recordings.json,
- res/ari/ari_model_validators.h: ari: Add events for playback and
- recording. While there were events defined for playback and
- recording these were not actually sent. This change implements
- the to_json handlers which produces them. (closes issue
- ASTERISK-22710) Reported by: Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/3026/ ........ Merged
- revisions 403119 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_snoop.exports.in (added), /,
- include/asterisk/stasis_app_snoop.h (added),
- rest-api/api-docs/channels.json, res/res_stasis_snoop.c (added),
- main/audiohook.c, res/ari/resource_channels.c,
- res/res_ari_channels.c, res/ari/resource_channels.h: ari: Add
- Snoop operation for spying/whispering on channels. The Snoop
- operation can be invoked on a channel to spy or whisper on it. It
- returns a channel that any channel operations can then be invoked
- on (such as record to do monitoring). (closes issue
- ASTERISK-22780) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/3003/ ........ Merged
- revisions 403117 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-23 00:22 +0000 [r403106] Rusty Newton <rnewton@digium.com>
-
- * apps/app_voicemail.c: app_voicemail: when forwarding a message,
- play vm-msgforwarded instead of vm-msgsaved In the last release
- of sounds, 1.4.25 we added a vm-msgforwarded prompt for various
- core languages. Now we use that prompt. (issue ASTERISK-21413)
- (closes issue ASTERISK-21413) Reported by: netwrkr Tested by:
- newtonr
-
-2013-11-22 23:57 +0000 [r403095] Kinsey Moore <kmoore@digium.com>
-
- * tests/test_stasis.c, /, tests/test_stasis_channels.c: Make sure
- unit tests compile This fixes the unit tests that were broken by
- r403069 and several functions requiring a new parameter for
- sanitization of JSON messages generated from object snapshots.
- ........ Merged revisions 403094 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-22 22:37 +0000 [r403083] Kevin Harwell <kharwell@digium.com>
-
- * /,
- contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py,
- res/res_pjsip/pjsip_configuration.c: res_pjsip: convert
- configuration settings names to snake case some more Updated the
- alembic script for pjsip. Also, the dtls config parsing stuff was
- expecting strings with no underscores, so removed the underscores
- from the option name before passing it to the parser. ........
- Merged revisions 403082 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-22 20:10 +0000 [r403070] Kinsey Moore <kmoore@digium.com>
-
- * res/res_stasis.c, main/stasis_endpoints.c,
- res/ari/resource_endpoints.c, main/rtp_engine.c, /,
- res/stasis/app.c, include/asterisk/stasis_bridges.h,
- include/asterisk/stasis.h, include/asterisk/stasis_app.h,
- main/stasis_bridges.c, res/ari/resource_bridges.c, main/json.c,
- main/stasis_message.c, include/asterisk/stasis_channels.h,
- main/stasis_channels.c, res/ari/resource_channels.c,
- include/asterisk/stasis_endpoints.h: ARI: Don't leak
- implementation details This change prevents channels used as
- implementation details from leaking out to ARI. It does this by
- preventing creation of JSON blobs of channel snapshots created
- from those channels and sanitizing JSON blobs of bridge snapshots
- as they are created. This introduces a framework for excluding
- information from output targeted at Stasis applications on a
- consumer-by-consumer basis using channel sanitization callbacks
- which could be extended to bridges or endpoints if necessary.
- This prevents unhelpful error messages from being generated by
- ast_json_pack. This also corrects a bug where BridgeCreated
- events would not be created. (closes issue ASTERISK-22744)
- Review: https://reviewboard.asterisk.org/r/2987/ Reported by:
- David M. Lee ........ Merged revisions 403069 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-22 17:27 +0000 [r403051] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_acl.c, res/res_pjsip.c,
- res/res_pjsip/config_transport.c, res/res_pjsip/config_global.c,
- /, configs/pjsip.conf.sample, res/res_pjsip/config_system.c,
- contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
- res/res_pjsip/pjsip_configuration.c: res_pjsip: convert
- configuration settings names to snake case Renamed, where
- appropriate, the configuration options for chan/res_pjsip to use
- snake case (compound words separated by an underscore). For
- example, faxdetect will become fax_detect, recordofffeature will
- become record_off_feature, etc... Review:
- https://reviewboard.asterisk.org/r/3002/ ........ Merged
- revisions 403022 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-22 17:12 +0000 [r403017] Joshua Colp <jcolp@digium.com>
-
- * /, main/translate.c: translate: Move freeing of frame to after it
- is used. When translating from one format to another it is
- possible to inform the translation function that the source frame
- should be freed. This was previously done immediately but shortly
- afterwards the frame that was freed was accessed and used again.
- This change moves code around a bit so that the frame is now
- freed after it has been completely used. (closes issue
- ASTERISK-22788) Reported by: Corey Farrell Patches:
- translate-access-after-free-11up.patch uploaded by coreyfarrell
- (license 5909) translate-access-after-free-1.8.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 403014 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 403015 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 403016 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-22 16:43 +0000 [r403013] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_directed_pickup.c, CHANGES: PickupChan: Add ability to
- specify channel uniqueids as well as channel names. * Made
- PickupChan() search by channel uniqueids if the search could not
- find a channel by name. * Ensured PickupChan() never considers
- the picking channel for pickup. * Made PickupChan() option p use
- a common search by name routine. The original search was
- erroneously case sensitive. (issue AFS-42) Review:
- https://reviewboard.asterisk.org/r/3017/
-
-2013-11-21 22:38 +0000 [r402995] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, apps/app_directory.c: app_directory: Set variable
- indicating reason directory exited By the time the directory
- application exits, a channel variable DIRECTORY_RESULT will be
- set for the channel that invoked it which can be used to
- determine the reason for exit. The changes log and the
- app_directory documentation contain specific details about each
- of the possible values for DIRECTORY_RESULT. Review:
- https://reviewboard.asterisk.org/r/3016/
-
-2013-11-21 22:36 +0000 [r402982-402994] David M. Lee <dlee@digium.com>
-
- * rest-api-templates/ari_resource.c.mustache, /,
- rest-api-templates/res_ari_resource.c.mustache: ari: Fix #include
- to match generated headers for snakeCase resource files ........
- Merged revisions 402993 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * rest-api-templates/make_ari_stubs.py, /: ari: Fix generators for
- resources with camelCase names. For the new deviceState resource,
- we need to properly generate device_state.[ch] files. ........
- Merged revisions 402981 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-21 19:22 +0000 [r402969] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_session.c, /: res_pjsip_session: Fix memory leak of
- direct media format capabilities The direct media format
- capabilities are always allocated in ast_sip_session_alloc and
- were not freed in the session destructor. Whoops. (This being the
- third whoops caught by Scott and Nitesh's valgrind work for the
- Asterisk Test Suite. Nifty!) ........ Merged revisions 402968
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-21 19:09 +0000 [r402945-402957] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/app.h, /: voicemail: Fixup some doxygen
- comments. ........ Merged revisions 402956 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/bucket.c: bucket: Fix scheme ref leak in
- __ast_bucket_scheme_register(). ........ Merged revisions 402944
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-21 17:53 +0000 [r402942-402943] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, /: res_pjsip_sdp_rtp: Fix use of
- uninitialized value in PJSIP In PJMEDIA,
- pjmedia_sdp_rtpmap_to_attr will attempt to use the string
- rtpmap.param regardless of its length value. Simply setting the
- length to 0 does not prevent the garbage on the stack in
- rtpmap.param.ptr from being formatted in a sprintf call. This
- patch initializes the string to NULL so that at the very least,
- something is provided to the function that is predictable.
- ........ Merged revisions 402941 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_mwi.c: res_pjsip_mwi: Fix memory leak of MWI
- subscriptions container This patch fixes a reference counting
- memory leak on the ao2_container created as part of
- create_mwi_subscriptions. When we create the container in this
- routine, the intent is to hand lifetime ownership over to the
- global container unsolicited_mwi. When
- ao2_global_obj_replace_unref is called, the reference count on
- mwi_subscriptions (the container) will be bumped by 1; however,
- the function does not decrement the reference count on
- mwi_subscriptions when this occurs. This will prevent the
- container from being fully disposed of when Asterisk exits (or on
- any subsequent call to this operation, such as during a reload).
- ........ Merged revisions 402940 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-21 15:57 +0000 [r402928-402929] David M. Lee <dlee@digium.com>
-
- * res/res_stasis.c, /: stasis: Fixed scoping problem with bridge
- tracking. ........ Merged revisions 402817 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_channels.c, res/res_ari_channels.c,
- res/ari/resource_channels.h, /, res/stasis/control.c,
- include/asterisk/stasis_app.h, rest-api/api-docs/channels.json:
- ari: Add silence generator controls This patch adds the ability
- to start a silence generator on a channel via ARI. This generator
- will play silence on the channel (avoiding audio timeouts on the
- peer) until it is stopped, or some other media operation is
- started (like playing media, starting music on hold, etc.).
- (closes issue ASTERISK-22514) Review:
- https://reviewboard.asterisk.org/r/3019/ ........ Merged
- revisions 402926 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-19 23:17 +0000 [r402892] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_caller_id.c: res_pjsip_caller_id: Don't
- overwrite user portion of the From header when fromuser is set.
- The fromuser option is used to explicitly set the user within the
- From header. The res_pjsip_caller_id module did not take this
- setting into account when determining if the From header could be
- modified or not. (closes issue ASTERISK-22866) Reported by:
- Anthony Messina ........ Merged revisions 402891 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-16 13:51 +0000 [r402865] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip/pjsip_distributor.c, /, configure,
- include/asterisk/autoconfig.h.in, configure.ac: res_pjsip: Add
- support for building against pjproject with SIP transaction group
- lock support. SIP transaction group lock support has been
- backported into our pjproject. Since the code now internally uses
- a group lock the code is now changed to unlock it if present.
- Note that the act of finding the transaction is what actually
- returns it locked. For further information about group locks
- check out the wiki page at:
- http://trac.pjsip.org/repos/wiki/Group_Lock (issue
- ASTERISK-22818) Reported by: Matt Jordan ........ Merged
- revisions 402864 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-15 22:38 +0000 [r402854] Jonathan Rose <jrose@digium.com>
-
- * apps/app_confbridge.c, CHANGES,
- apps/confbridge/conf_config_parser.c,
- configs/confbridge.conf.sample,
- apps/confbridge/include/confbridge.h: Confbridge: Add option to
- review the recording similar to announce_join_leave Review:
- https://reviewboard.asterisk.org/r/3008/
-
-2013-11-15 14:37 +0000 [r402839] Kinsey Moore <kmoore@digium.com>
-
- * /, main/cel.c: CEL: Fix crash when using CELGenUserEvent This
- fixes a crash when CELGenUserEvent is called from the dialplan
- while CEL is disabled. Currently, CEL does not create its topics
- and forwards if it is not enabled and external entities may
- depend on these topics blindly since they should always be
- available. This patch breaks up route creation and topic/forward
- creation such that the CEL topics and forwards will always exist
- while the router and its associated routes will be torn down and
- recreated as necessary. (closes issue ASTERISK-22799) Review:
- https://reviewboard.asterisk.org/r/3010/ Reported by: Matt Jordan
- ........ Merged revisions 402838 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-14 21:36 +0000 [r402820-402829] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_directed_pickup.c: Pickup: Pickup() and PickupChan()
- parameter parsing improvements. * Made Pickup() and PickupChan()
- tollerate empty pickup values. i.e., You can now have
- Pickup(&&exten@context). * Made PickupChan() use the standard
- option flag parsing code.
-
- * apps/app_directed_pickup.c: Pickup: Ensure using PICKUPMARK never
- considers the picking channel.
-
-2013-11-14 20:32 +0000 [r402819] Jonathan Rose <jrose@digium.com>
-
- * CHANGES, main/pbx.c, apps/app_sayunixtime.c: Say: If
- SAY_DTMF_INTERRUPT is set to an ast_true value, jump on DTMF
- Similar to how background works, if a say application is called
- with this variable set to 'true', 'yes', 'on', etc. then using
- DTMF while the say action is in progress will result in the
- channel jumping to that extension in the dialplan. Review:
- https://reviewboard.asterisk.org/r/3011/
-
-2013-11-13 23:11 +0000 [r402805] Joshua Colp <jcolp@digium.com>
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c, res/ari/resource_channels.h, /,
- res/stasis/control.c, include/asterisk/stasis_app.h:
- res_ari_channels: Add the ability to stop locally generated
- ringing on a channel. Using the 'ring' operation it is possible
- to start locally generated ringback if the channel is answered.
- This change adds the ability to stop it by using DELETE. ........
- Merged revisions 402804 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 23:17 +0000 [r402788-402795] Kevin Harwell <kharwell@digium.com>
-
- * res/ari/resource_endpoints.c, /: ari endpoints: GET
- /ari/endpoints/{invalid-tech} should return a 404 Was returning a
- 404 on a valid technology with an empty list of endpoints. Now
- checking against the channel tech to make sure the tech itself is
- valid and not just an empty list of endpoints. (issue
- ASTERISK-22803) Reported by: David M. Lee ........ Merged
- revisions 402793 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * rest-api/api-docs/endpoints.json, res/ari/resource_endpoints.c,
- /, res/res_ari_endpoints.c: ari endpoints: GET
- /ari/endpoints/{invalid-tech} should return a 404 Implementation
- listing endpoints by technology returned an empty array if no
- matching endpoints were found. Fixed so a "404 Not Found" will be
- returned instead. (closes issue ASTERISK-22803) Reported by:
- David M. Lee ........ Merged revisions 402787 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 19:38 +0000 [r402768-402778] Mark Michelson <mmichelson@digium.com>
-
- * /, main/channel.c: Switch to a scoped lock to avoid missing
- unlocks in failure returns. ........ Merged revisions 402769 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/channel.c, /: Move a NULL check to a place that makes more
- sense. Two variables were being checked for NULLity immediately
- after being declared NULL. I moved the NULL check until after the
- variables are allocated. This allows for the "channelvars" option
- in manager.conf to work as intended again. ........ Merged
- revisions 402767 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 16:49 +0000 [r402758] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_messaging.c, res/res_pjsip_header_funcs.c, /:
- pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer
- dereferences Both res_pjsip_messaging and res_pjsip_header_funcs
- were causing asterisk to crash because they were trying to
- dereference a NULL pointer. In the case of res_pjsip_messaging it
- was attempting to "print" a contact header that did not exist. In
- fact contact headers should not be part of a SIP MESSAGE, so the
- offending code was simply removed. In the case of
- res_pjsip_header_funcs a null private channel tech was being
- passed to the function and then later dereferenced. Added null
- checks (and error logging) to the read/write function handlers to
- guard against crashing. (closes issue ASTERISK-22821) Reported
- by: Anthony Messina ........ Merged revisions 402757 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 16:34 +0000 [r402756] Kinsey Moore <kmoore@digium.com>
-
- * /, apps/app_celgenuserevent.c: CELGenUserEvent: Fix error message
- from ast_json_pack This prevents NULL from being passed into an
- ast_json_pack call when no extra information is passed to the
- application which prevents an error message about NULL arguments
- from being generated. ........ Merged revisions 402755 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 15:27 +0000 [r402741] David M. Lee <dlee@digium.com>
-
- * res/ari/ari_model_validators.h, rest-api/api-docs/events.json, /:
- Fixed a typ. ........ Merged revisions 402738 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-12 15:03 +0000 [r402711] Kinsey Moore <kmoore@digium.com>
-
- * channels/chan_dahdi.c, /: chan_dahdi: Fix crash during caller ID
- read Asterisk will sometimes core dump during caller id read on
- analog channels due to a negative return value from the read() in
- my_get_callerid that slips through as a negative length argument
- to callerid_feed() if the errno returned by DAHDI is ELAST. This
- change ensures that the negative return is treated properly even
- when it is ELAST. (closes issue ASTERISK-22746) Reported by:
- Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
- uploaded by Michael Walton (License 6502) ........ Merged
- revisions 402708 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 402709 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402710 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-11 20:28 +0000 [r402698] Jonathan Rose <jrose@digium.com>
-
- * apps/app_confbridge.c: Confbridge: add test events for dynamic
- menus test Adds a couple of test events for conference menu
- actions so that it's easy to discern when those menu actions have
- been triggered. (issue ASTERISK-22760) Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/2999/
-
-2013-11-11 19:31 +0000 [r402688] Mark Michelson <mmichelson@digium.com>
-
- * apps/app_confbridge.c, /: Get rid of some inaccurate comments.
- I'm doing some unrelated work in app_confbridge and finding these
- "invalid pin" comments to be annoying. Get out! ........ Merged
- revisions 402686 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402687 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-11 15:37 +0000 [r402648] Kinsey Moore <kmoore@digium.com>
-
- * /, apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
- current app_queue code from 1.8 up to trunk the upper and lower
- penalties can be set to 0 but the value is interpreted to be
- disabled instead of actually setting limits. This is especially
- evident if min and max limits are set to 0 and members with
- penalties of 0 and 1 are in the queue since the member with
- penalty 1 will still receive calls. This patch adjusts the
- special disabled value to be INT_MAX instead of 0. (closes issue
- ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
- Reported by: Schmooze Com ........ Merged revisions 402645 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 402646 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402647 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 23:07 +0000 [r402607] Scott Griepentrog <sgriepentrog@digium.com>
-
- * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
- keep same local (from) tag for outgoing register requests For
- outbound register requests the tag on the From line was updated
- every 20 seconds prior to a successful registration and also once
- for each registration renewal. That behavior can possibly cause
- the registration to be denied because of the different tag, and
- is not aligned with the intention of RFC 3261 8.1.3.5 "...
- request constitutes a new transaction and SHOULD have the same
- value of the Call-ID, To, and From of the previous request...".
- This updates chan_sip to have a field to keep the local tag in
- the registration structure and use that tag for registration
- requests where the callid is also unchanged. (closes issue
- ASTERISK-12117) Reported by: Pawel Pierscionek Review:
- https://reviewboard.asterisk.org/r/2988/ ........ Merged
- revisions 402604 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 402605 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402606 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 20:37 +0000 [r402595] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_stasis.c: res_stasis.c: Fix locking issues with the
- app_bridge_moh container. * Fix unlinking from the
- app_bridges_moh container in remove_bridge_moh() without a lock
- under normal circumstances. * Made check
- ast_bridge_set_after_callback() return value in
- bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK()
- locking over too much scope in stasis_app_bridge_moh_channel()
- and stasis_app_bridge_moh_stop(). * Fixed unusual usage of
- ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge
- from off nominal path in stasis_app_bridge_create(). * Fixed
- strange construct in stasis_app_unsubscribe(). From a bad merge?
- * Made load_module() cleanup on failure. Review:
- https://reviewboard.asterisk.org/r/2962/ ........ Merged
- revisions 402593 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 19:33 +0000 [r402585] Jonathan Rose <jrose@digium.com>
-
- * /, main/security_events.c, configs/manager.conf.sample, CHANGES,
- include/asterisk/manager.h, main/manager.c: security_events: Push
- out security events over AMI events Security Events will now be
- written to any listener of the new 'security' class Review:
- https://reviewboard.asterisk.org/r/2998/ ........ Merged
- revisions 402584 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 19:22 +0000 [r402583] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip.c, /: Clarify an ambiguous error message. ........
- Merged revisions 402582 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 18:53 +0000 [r402571-402572] David M. Lee <dlee@digium.com>
-
- * /, res/res_pjsip/config_system.c: res_pjsip: Print a helpful
- error message if sorcery registration fails ........ Merged
- revisions 402570 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_playbacks.h, /: Changes from make ari-stubs
- after r402560 ........ Merged revisions 402561 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 17:59 +0000 [r402562] Kevin Harwell <kharwell@digium.com>
-
- * rest-api/resources.json, res/ari/resource_playback.h (removed),
- res/res_ari_playbacks.c (added), res/ari/resource_playbacks.h
- (added), /, res/ari.make, rest-api/api-docs/playback.json
- (removed), res/ari/resource_playback.c (removed),
- res/res_ari_playback.c (removed),
- rest-api/api-docs/playbacks.json (added),
- res/ari/resource_playbacks.c (added): ARI playback: Rename ARI
- Playback to Playbacks Before playback was the only non plural
- resource. It has been renamed to playbacks for consistency.
- (closes issue ASTERISK-22737) Reported by: Paul Belanger ........
- Merged revisions 402560 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 17:29 +0000 [r402557] David M. Lee <dlee@digium.com>
-
- * res/res_ari.c, main/manager.c, /, main/http.c: ari: Add
- application/x-www-form-urlencoded parameter support ARI POST
- calls only accept parameters via the URL's query string. While
- this works, it's atypical for HTTP API's in general, and
- specifically frowned upon with RESTful API's. This patch adds
- parsing for application/x-www-form-urlencoded request bodies if
- they are sent in with the request. Any variables parsed this way
- are prepended to the variable list supplied by the query string.
- (closes issue ASTERISK-22743) Review:
- https://reviewboard.asterisk.org/r/2986/ ........ Merged
- revisions 402555 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-08 14:58 +0000 [r402546] Kevin Harwell <kharwell@digium.com>
-
- * apps/app_dahdiras.c, utils/extconf.c, main/asterisk.c:
- app_dahdiras: Use waitpid instead of wait4. Several places in the
- code were using wait4 while other places were using waitpid. This
- change makes all places use waitpid in order to make things more
- consistent and since the 'rusage' object passed in/out of wait4
- was never used. (closes issue ASTERISK-22557) Reported by:
- YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman
- (license 6537)
-
-2013-11-07 23:42 +0000 [r402538] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_authenticator_digest.c, /: PJSIP: Improve error
- handling in digest authenticator Previously, regardless of
- whether failure to authenticate was due to lacking any
- authentication or actually failing authentication, the Digest
- Authenticator would simply return that a challenge was still
- needed. It will continue to do that when no authentication
- information is in the received SIP digest, but when
- authentication information is present and does not pass
- authentication, that will be treated as an authentication error.
- This is to ensure that PJSIP will issue security events indicated
- failed auths. ........ Merged revisions 402537 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-07 21:10 +0000 [r402529] David M. Lee <dlee@digium.com>
-
- * res/ari/resource_applications.c, res/ari/resource_playback.c,
- rest-api/api-docs/channels.json, res/ari/resource_applications.h,
- res/ari/resource_channels.c, res/ari/resource_playback.h,
- rest-api/api-docs/recordings.json, res/ari/resource_recordings.c,
- rest-api-templates/ari_resource.c.mustache,
- rest-api-templates/asterisk_processor.py,
- res/ari/resource_channels.h, rest-api/api-docs/endpoints.json,
- res/ari/resource_endpoints.c, res/ari/resource_recordings.h,
- res/ari/resource_events.c, res/res_ari_playback.c,
- res/res_ari_applications.c, res/ari/resource_endpoints.h,
- res/ari/resource_events.h, rest-api/api-docs/sounds.json,
- res/ari/resource_sounds.c, res/res_ari_channels.c,
- rest-api/api-docs/bridges.json, res/ari/resource_bridges.c,
- res/ari/resource_sounds.h, res/res_ari_recordings.c,
- res/ari/resource_bridges.h, rest-api/api-docs/asterisk.json,
- res/ari/resource_asterisk.c, res/res_ari_endpoints.c,
- rest-api/api-docs/applications.json,
- rest-api/api-docs/playback.json, res/res_ari_events.c,
- res/ari/resource_asterisk.h, rest-api-templates/swagger_model.py,
- res/res_ari_sounds.c, res/res_ari_bridges.c, /,
- rest-api-templates/ari_resource.h.mustache,
- rest-api-templates/rest_handler.mustache, res/res_ari_asterisk.c,
- rest-api-templates/res_ari_resource.c.mustache: ari: User better
- nicknames for ARI operations While working on building client
- libraries from the Swagger API, I noticed a problem with the
- nicknames. channel.deleteChannel() channel.answerChannel()
- channel.muteChannel() Etc. We put the object name in the nickname
- (since we were generating C code), but it makes OO generators
- redundant. This patch makes the nicknames more OO friendly. This
- resulted in a lot of name changing within the res_ari_*.so
- modules, but not much else. There were a couple of other fixed I
- made in the process. * When reversible operations (POST /hold,
- POST /unhold) were made more RESTful (POST /hold, DELETE
- /unhold), the path for the second operation was left in the API
- declaration. This worked, but really the two operations should
- have been on the same API. * The POST /unmute operation had still
- not been REST-ified. Review:
- https://reviewboard.asterisk.org/r/2940/ ........ Merged
- revisions 402528 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-06 21:58 +0000 [r402518] Kevin Harwell <kharwell@digium.com>
-
- * /, apps/app_queue.c: app_queue: crash if first agent is "busy" If
- the first agent/member (via CLI "queue show") in a queue is
- "busy" (dnd, circuit busy, etc...) and no agents answered then
- app_queue would crash. This occurred because while the calling of
- agent(s) remained valid the channel on "busy" agent would be set
- to NULL and then later dereferenced upon a second "rna" function
- call. The original intention of the code is to have only valid
- "call attempt" objects (channels != NULL) checked while
- attempting to call agent(s). It does this by building a
- "call_next" list of valid "call attempt" objects. In the case of
- the "busy" agent subsequent builds of the valid "call attempt"
- list would sometimes include (the case mentioned above) an
- invalid "call attempt" object. The fix was to make sure the "call
- attempt" list was appropriately built on every iteration. A NULL
- sanity check was also added at the original offending spot of the
- crash just in case another one slipped by somehow. (closes issue
- ASTERISK-22644) Reported by: Marco Signorini Review:
- https://reviewboard.asterisk.org/r/2983/ ........ Merged
- revisions 402517 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-05 21:17 +0000 [r402502-402508] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Use AST_AF* defined constant
- when calling ast_get_ip While the structure passed to ast_get_ip
- should be set memset to 0, thus initializing the ss_family member
- to 0, explicitly setting it to AST_AF_UNSPEC is more portable.
- ........ Merged revisions 402507 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Fix incorrect usage of
- ast_get_ip involving uninitialized struct This started off as a
- fix for the failing IAX2 acl_call test in the Asterisk Test
- Suite. When inspecting why that test was failing, it became clear
- that all attempts to bind to any local loopback address was
- failing: [Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding
- IAX2 to address 127.0.0.1:4569 [Nov 2 15:56:28] DEBUG[15787]
- netsock2.c: Splitting '127.0.0.1' into... [Nov 2 15:56:28]
- DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''. [Nov 2
- 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1",
- "(null)", ...): ai_family not supported [Nov 2 15:56:28]
- WARNING[15787] acl.c: Unable to lookup '127.0.0.1' While there's
- conceivably other ways for getaddrino to return EAI_FAMILY, the
- most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not
- provided as the desired family. The culprit was the call to
- ast_get_ip, defined in acl.h. This function uses the family from
- the passed in addr object (which it will also populate when it
- returns!) when it eventually calls getaddrinfo. This patch fixes
- the use of ast_get_ip that were not specifying the family in
- chan_iax2. This prevents uninitialized use of the structure, so
- that the addresses resolve correctly. Review:
- https://reviewboard.asterisk.org/r/2991 ........ Merged revisions
- 402505 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/acl.h, /, include/asterisk/netsock2.h: netsock2:
- Define AST_AF_* enum constants to their AF_* equivalents This
- patch explicitly defines AST_AF_* enum constants to their
- sys/socket.h defined equivalents. It is certainly unclear why
- these constants actually have to exist, given that netsock2.h
- includes sys/socket.h; however, since the code base is already
- liberally sprinkled with the usage of AST_AF_* (as well as with
- direct calls to AF_*), this will at least keep the semantics
- consistent between their usage across systems. ........ Merged
- revisions 402503 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_channels.c, /: stasis_channels: Don't give preference
- to ANI info in channel snapshots When publishing channel
- snapshots, we currently compute the caller ID name and number by
- giving preference first to ani.{name|number}, then to
- id.{name|number}. However, when a channel driver (such as
- chan_sip) updates the caller ID, it typically only updates the
- caller ID stored in id.{name|number}. This means that we are
- currently giving preference to stale information. When looking at
- the rest of the code base, the only other place where we appear
- to use this same logic is in app_amd. Everywhere else, we treat
- the party information in ani as being separate to the party
- information in id. This patch publishes only the caller ID name
- and number in the snapshot field for caller_name and caller_num.
- Note that the information in ANI is still available in
- caller_ani. Review: https://reviewboard.asterisk.org/r/2992/
- ........ Merged revisions 402501 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-04 21:02 +0000 [r402453] Kevin Harwell <kharwell@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: notify dialog info ignores
- presentation indicator in callerid The presentation indicator in
- a callerid (e.g. set by dialplan function
- Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
- Info Notifies are generated during extension monitoring. Added a
- check to make sure the name and/or number presentations on the
- callee (remote identity) are set to allow. If they are restricted
- then "anonymous" is used instead. (closes issue AST-1175)
- Reported by: Thomas Arimont Review:
- https://reviewboard.asterisk.org/r/2976/ ........ Merged
- revisions 402450 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402452 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-02 04:30 +0000 [r402406-402439] Richard Mudgett <rmudgett@digium.com>
-
- * main/stasis.c, main/stasis_message_router.c, /,
- include/asterisk/vector.h: vector: Uppercase API to follow C
- convention. C does not support templates like C++. ........
- Merged revisions 402438 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/lock.h, main/stasis.c,
- main/stasis_message_router.c, /, include/asterisk/vector.h:
- vector: Update API to be more flexible. Made the vector macro API
- be more like linked lists. 1) Added a name parameter to
- ast_vector() to name the vector struct. 2) Made the API take a
- pointer to the vector struct instead of the struct itself. 3)
- Added an element cleanup macro/function parameter when removing
- an element from the vector for ast_vector_remove_cmp_unordered()
- and ast_vector_remove_elem_unordered(). 4) Added
- ast_vector_get_addr() in case the vector element is not a simple
- pointer. * Converted an inline vector usage in
- stasis_message_router to use the vector API. It needed the API
- improvements so it could be converted. * Fixed topic reference
- leak in router_dtor() when the stasis_message_router is
- destroyed. * Fixed deadlock potential in stasis_forward_all() and
- stasis_forward_cancel(). Locking two topics at the same time
- requires deadlock avoidance. * Made internal_stasis_subscribe()
- tolerant of a NULL topic. * Made stasis_message_router_add(),
- stasis_message_router_add_cache_update(),
- stasis_message_router_remove(), and
- stasis_message_router_remove_cache_update() tolerant of a NULL
- message_type. * Promoted a LOG_DEBUG message to LOG_ERROR as
- intended in dispatch_message(). Review:
- https://reviewboard.asterisk.org/r/2903/ ........ Merged
- revisions 402429 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/confbridge/conf_state_single.c,
- apps/confbridge/conf_state_inactive.c,
- apps/confbridge/conf_state_single_marked.c, /,
- apps/confbridge/include/confbridge.h,
- apps/confbridge/conf_state_multi.c, apps/app_confbridge.c,
- apps/confbridge/conf_state_multi_marked.c,
- apps/confbridge/conf_state.c: confbridge: Separate user muting
- from system muting overrides. The system overrides the user
- muting requests when MOH is playing or a waitmarked user is
- waiting for a marked user to join. System muting overrides
- interfere with what the user may wish the muting to be when the
- system override ends. * User muting requests are now independent
- of the system muting overrides. The effective muting is now the
- logical or of the user request and system override. * Added a
- Muted flag to the CLI "confbridge list <conference>" command. *
- Added a Muted header to the AMI ConfbridgeList action
- ConfbridgeList event. (closes issue AST-1102) Reported by: John
- Bigelow Review: https://reviewboard.asterisk.org/r/2960/ ........
- Merged revisions 402425 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402427 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/config.c, apps/confbridge/conf_config_parser.c,
- configs/confbridge.conf.sample, /: config: Allow ConfBridge DTMF
- menus to have '#' as the first digit. ConfBridge allows custom
- DTMF menus to be created in the confbridge.conf file by assigning
- a DTMF key sequence to a sequence of actions as follows:
- DTMF-sequence = action,action... Unfortunately, the normal config
- file processing code interprets an initial '#' character as
- starting a directive such as #include. * Add the ability to
- escape the first non-blank character in a config line so the '#'
- character can be used without triggering the directive processing
- code. (closes issue AFS-2) (closes issue ASTERISK-22478) Reported
- by: Nicolas Tanski Patches: jira_asterisk_22478_v11.patch
- (license #5621) patch uploaded by rmudgett (modified) Review:
- https://reviewboard.asterisk.org/r/2969/ ........ Merged
- revisions 402407 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402416 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/app.h, /, main/app.c: voicemail: Simplify
- callback pointer declarations and add doxygen. * Typedefed and
- added doxegen for the voicemail callback functions. * Simplified
- the prototypes for ast_install_vm_functions() and
- ast_install_vm_test_functions() to use the new function typedefs.
- * Simplified the voicemail callback function pointer variable
- declarations to use the new function typedefs. ........ Merged
- revisions 402398 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-01 22:48 +0000 [r402397] Jonathan Rose <jrose@digium.com>
-
- * apps/confbridge/conf_config_parser.c,
- apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
- CHANGES: app_confbridge: Make the CONFBRIDGE function be able to
- create dynamic menus Also adds the ability to clear all profile
- items and makes behavior more consistent with documentation as
- when choosing whether to use CONFBRIDGE datastore profiles or the
- application arguments to the confbridge application. (closes
- issue ASTERISK-22760) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2971/
-
-2013-11-01 21:51 +0000 [r402388] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/manager_bridges.c, /, main/bridge.c,
- include/asterisk/bridge.h: Manager: Add equivalent AMI actions
- for the bridge CLI commands. Adds the following AMI events,
- closely following their CLI counterparts: BridgeDestroy
- BridgeKick BridgeTechnologyList BridgeTechnologySuspend
- BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge,
- where BridgeKick kicks just one channel off the bridge. When
- kicking a channel, specifying the bridge also (optional) insures
- it is not removed from the wrong bridge. The BridgeTechnology
- events allow viewing and changing suspension status, which
- affects only subsequent not active bridging. (closes
- ASTERISK-22356) Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/2973/ ........ Merged
- revisions 402387 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-01 16:31 +0000 [r402368] David M. Lee <dlee@digium.com>
-
- * /, rest-api-templates/api.wiki.mustache: ari wiki docs: add notes
- about allowMultiple parameters. This patch adds a note to any
- parameter that has 'allowMultiple' set in the Swagger
- documentation. (closes issue ASTERISK-22704) ........ Merged
- revisions 402367 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-01 14:38 +0000 [r402359] Joshua Colp <jcolp@digium.com>
-
- * include/asterisk/stasis_app.h, rest-api/api-docs/channels.json,
- res/ari/resource_channels.c, res/res_ari_channels.c,
- res/ari/resource_channels.h, res/res_stasis_playback.c, /,
- res/stasis/control.c: res_ari_channels: Add ring operation, dtmf
- operation, hangup reasons, and tweak early media. The ring
- operation sends ringing to the specified channel it is invoked
- on. The dtmf operation can be used to send DTMF digits to the
- specified channel of a specific length with a wait time in
- between. Finally hangup reasons allow you to specify why a
- channel is being hung up (busy, congestion). Early media behavior
- has also been tweaked slightly. When playing media to a channel
- it will no longer automatically answer. If it has not been
- answered a progress indication is sent instead. (closes issue
- ASTERISK-22701) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2916/ ........ Merged
- revisions 402358 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-01 12:40 +0000 [r402349] Kinsey Moore <kmoore@digium.com>
-
- * res/res_rtp_asterisk.c, /, channels/chan_sip.c,
- include/asterisk/rtp_engine.h: chan_sip: Fix RTCP port for SRFLX
- ICE candidates This corrects one-way audio between Asterisk and
- Chrome/jssip as a result of Asterisk inserting the incorrect RTCP
- port into RTCP SRFLX ICE candidates. This also exposes an ICE
- component enumeration to extract further details from candidates.
- (closes issue ASTERISK-21383) Reported by: Shaun Clark Review:
- https://reviewboard.asterisk.org/r/2967/ ........ Merged
- revisions 402345 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402348 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-11-01 12:33 +0000 [r402337-402347] Joshua Colp <jcolp@digium.com>
-
- * /, include/asterisk/stasis_app.h, res/ari/resource_channels.c:
- res_ari_channels: Fix a deadlock when originating multiple
- channels close to eachother. If a Stasis application is specified
- an implicit subscription is done on the originated channel. This
- was previously done with the channel lock held which is dangerous
- as the underlying code locks the container and iterates items.
- This change releases the lock on the originated channel before
- subscribing occurs. (closes issue ASTERISK-22768) Reported by:
- Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/
- ........ Merged revisions 402346 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/stasis/control.c: res_stasis: Ensure the channel is always
- departed from the bridge when it leaves. This change adds a
- command to the command queue to explicitly depart the channel
- from the bridge when it is told it has left. If the channel has
- already been departed or has entered a different bridge this
- command will become a no-op. (closes issue ASTERISK-22703)
- Reported by: John Bigelow (closes issue ASTERISK-22634) Reported
- by: Kevin Harwell Review:
- https://reviewboard.asterisk.org/r/2965/ ........ Merged
- revisions 402336 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-31 22:09 +0000 [r402328] Mark Michelson <mmichelson@digium.com>
-
- * /, contrib/scripts/sip_to_pjsip/sip_to_pjsip.py,
- contrib/scripts/sip_to_res_sip (removed),
- contrib/scripts/sip_to_pjsip (added),
- contrib/scripts/sip_to_pjsip/astconfigparser.py,
- contrib/scripts/sip_to_pjsip/astdicts.py: Update the conversion
- script from sip.conf to pjsip.conf (closes issue ASTERISK-22374)
- Reported by Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2846 ........ Merged revisions
- 402327 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-31 16:06 +0000 [r402286-402290] Matthew Jordan <mjordan@digium.com>
-
- * main/loader.c, /: core/loader: Don't call dlclose in a while loop
- For awhile now, we've noticed continuous integration builds
- hanging on CentOS 6 64-bit build agents. After resolving a number
- of problems with symbols, strange locks, and other shenanigans,
- the problem has persisted. In all cases, gdb shows the Asterisk
- process stuck in loader.c on one of the infinite while loops that
- calls dlclose repeatedly until success. The documentation of
- dlclose states that it returns 0 on success; any other value on
- error. It does not state that repeatedly calling it will
- eventually clear those errors. Most likely, the repeated calls to
- dlclose was to force a close by exhausting the references on the
- library; however, that will never succeed if: (a) There is some
- fundamental error at work in the loaded library that precludes
- unloading it (b) Some other loaded module is referencing a symbol
- in the currently loaded module This results in Asterisk sitting
- forever. Since we have matching pairs of dlopen/dlclose, this
- patch opts to only call dlclose once, and log out as an ERROR if
- dlclose fails to return success. If nothing else, this might help
- to determine why on the CentOS 6 64-bit build agent things are
- not closing successfully. Review:
- https://reviewboard.asterisk.org/r/2970 ........ Merged revisions
- 402287 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 402288 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402289 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/media_index.c, /: medix_index: Display errors when library
- calls fail Based on feedback from ipengineer in #asterisk, when
- the media indexer cannot access a sound file on the system (or
- otherwise fails) Asterisk displays a "Cannot frob file" error but
- fails to tell you why. This is especially problematic as the
- media_indexer failing will rpevent Asterisk from starting, as it
- is in the core. We now display the errno error messages so folks
- can figure out what they've done wrong. ........ Merged revisions
- 402285 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-31 14:45 +0000 [r402277] David M. Lee <dlee@digium.com>
-
- * /, res/stasis/app.c: stasis: add functions embarrassingly missing
- from r400522 I neglected to implement two of the endpoint
- subscription functions when I did the work. Normally, you'll only
- hit that when you unsubscribe from a specific endpoint. ........
- Merged revisions 402276 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-30 17:54 +0000 [r402266] Kevin Harwell <kharwell@digium.com>
-
- * channels/chan_pjsip.c, /, res/res_pjsip_messaging.c:
- pjsip_messaging: Added debug for in dialog messaging (issue
- ASTERISK-22777) Reported by: Matt Jordan ........ Merged
- revisions 402265 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-29 23:43 +0000 [r402227] Rusty Newton <rnewton@digium.com>
-
- * /, sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14
- extra sounds, plus new en_GB language set The new sound packages
- relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
- ASTERISK-20782 Modified sounds/Makefile for the new sound
- versions and to account for the new en_GB language set. (issue
- ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
- ASTERISK-22411) (closes issue ASTERISK-22544) ........ Merged
- revisions 402224 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 402225 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402226 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-29 12:57 +0000 [r402155] Matthew Jordan <mjordan@digium.com>
-
- * main/xmldoc.c, main/channel.c, main/pbx.c, /, main/translate.c:
- Remove some spammy debug messages; improve clarity of others
- Debug messages aren't free. Even when the debug level is
- sufficiently low such that the messages are never evaluated,
- there is a cost to having to parse Asterisk logs that contain
- debug messages that (a) fail to convey sufficient information or
- (b) occur so frequently as to be next to meaningless. Based on
- having to stare at lots of DEBUG messages, this patch makes the
- following changes: * channel.c: When copying variables from a
- parent channel to a child channel, specify the channels involved.
- Do not log anything for a variable that is not inherited; the
- fact that it doesn't have an _ or __ already signifies that it
- won't be inherited. * pbx.c: Specify what function evaluation has
- occurred that created the result. * translate.c: Bump up the
- translator path messages to 10. I've never once had to use these
- debug messages, and for each format that is registered (on
- startup) and unregistered (on shutdown) the entire f^2 matrix is
- logged out. For short tests in the Asterisk Test Suite, this
- should make finding the actual test much easier. * xmldoc.c: The
- debug message that 'blah' is not found in the tree is expected.
- Often, description elements - which are not required - are not
- provided. This debug message adds no additional value, as it is
- not indicative of an error or helpful in debugging which element
- did not contain a 'blah' element as a child. If an element is
- supposed to contain a child element, then that XML tree should
- have failed validation in the first place. Review:
- https://reviewboard.asterisk.org/r/2966/ ........ Merged
- revisions 402150 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 402151 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402154 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-29 12:51 +0000 [r402149-402153] Kinsey Moore <kmoore@digium.com>
-
- * rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c, res/ari/resource_channels.h, /: ARI:
- Remove channels/{channelId}/dial This removes the
- /ari/channels/{channelId}/dial URI since it is redundant, overly
- complex, is likely to become more externally complex over time,
- and is too high-level compared with other ARI operations. See the
- following for further information:
- http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html
- (closes issue ASTERISK-22784) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2968/ ........ Merged
- revisions 402152 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * bridges/bridge_native_rtp.c, /: bridge_native_rtp: Ensure bridge
- is torn down When a bridge transitions away from one tech to
- another, the tech going away is provided a dummy bridge with no
- channels in it to tear down. Currently this means that the
- teardown code exits prematurely and does not tear anything down.
- This change tears down RTP bridging for the channel provided in
- the leave bridge tech callback. This also reverts the majority of
- r400403 since it is now redundant. (closes issue ASTERISK-22628)
- (closes issue ASTERISK-22676) Reported by: John Bigelow Reported
- by: Kevin Harwell Tested by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/2905/ Patches:
- native_rtp_fix.diff uploaded by Kinsey Moore (License 6273)
- ........ Merged revisions 402148 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-29 11:15 +0000 [r402140] Joshua Colp <jcolp@digium.com>
-
- * /, rest-api/api-docs/playback.json, res/res_ari_playback.c:
- res_ari_playback: Add missing 404 error response for GET and
- DELETE. (closes issue ASTERISK-22722) Reported by: Richard
- Mudgett ........ Merged revisions 402139 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-28 22:10 +0000 [r402128-402130] David M. Lee <dlee@digium.com>
-
- * /, doc: Ignore full docs ........ Merged revisions 402127 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /: Put back several merge revisions that were lost in r402054
-
- * /: Put back several merge revisions that were lost in r401962
-
-2013-10-28 15:08 +0000 [r402113-402117] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, UPGRADE-11.txt, UPGRADE-12.txt: Fix UPGRADE.txt Due To Merging
- From Branch 11 When merging in the patch for ASTERISK-22728, the
- UPGRADE.txt file was changed incorrectly. That change should have
- gone into ASTERISK-11.txt. This commit is to fix that. Also,
- another comment in the UPGRADE-11.txt was missing and this commit
- adds that as well. ........ Merged revisions 402115 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c, UPGRADE-12.txt: chan_sip: Clarify
- 'Forcerport' Setting Displayed When Running "sip show peers"
- While looking at ASTERISK-22236, Walter Doekes pointed out that
- when running "sip show peers", the setting being displayed can be
- confusing. The display of "N" used to mean NAT (i.e. yes). The
- NAT setting has gone through many different changes resulting in
- the display of different characters to try and convey what the
- current setting is for 'Forcerport' (A for Auto and Forcerport is
- currently on, a for Auto but Forcerport is off, Y for yes, and N
- for no). During the initial code review to try and clarify these
- settings (especially since "N" no longer meant what it used to
- mean in prior versions of Asterisk), Mark Michelson suggested
- using the full space available to display the settings which
- helped to make the settings very clear. That was a great
- suggestion. Therefore, this patch does the following: * The
- column for 'Forcerport' now will show: Auto (Yes), Auto (No),
- Yes, or No. * A column for the 'Comedia' setting has been added.
- It too will display the setting in a non-cryptic way: Auto (Yes),
- Auto (No), Yes, or No. * UPGRADE.txt has been updated to document
- this change. (closes issue ASTERISK-22728) Reported by: Walter
- Doekes Tested by: Michael L. Young Patches:
- asterisk-forcerport-display-clarification_v3.diff uploaded by
- Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2941 ........ Merged revisions
- 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
- ........ Merged revisions 402112 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-27 23:22 +0000 [r402073-402091] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: Filter out internal channels from dial message
- handling Surrogate channels would pop up from time to time in
- dial message handling. This would cause a WARNING message to
- appear, indicating that the Surrogate channel had no CDR. This
- patch filters out those channels that have the internal
- implementation flag set, such that the WARNING message isn't
- displayed. ........ Merged revisions 402090 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * cdr/cdr_sqlite3_custom.c, /, cdr/cdr_syslog.c, cdr/cdr_sqlite.c,
- cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
- include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c,
- cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c,
- cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c: Prevent CDR backends
- from unregistering while billing data is in flight This patch
- makes it so that CDR backends cannot be unregistered while active
- CDR records exist. This helps to prevent billing data from being
- lost during restarts and shutdowns. Review:
- https://reviewboard.asterisk.org/r/2880/ ........ Merged
- revisions 402081 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, contrib/ast-db-manage/config/env.py,
- contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py,
- contrib/ast-db-manage/voicemail/env.py: Update Alembic database
- scripts for external scripting and PostgreSQL, Oracle This patch
- does the following: 1) The env scripts have been updated to be
- tolerant of a NULL configuration file. This occurs when
- configuration is provided by an external script, such that the
- actual config.ini file is not used. 2) Enum types have all been
- given names. This is needed for PostgreSQL script generation. 3)
- The identifier meetme_confno_starttime_endtime is greater than 30
- characters, and hence invalid for Oracle databases. This has been
- truncated down to meetme_confno_start_end. ........ Merged
- revisions 400383 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-26 12:56 +0000 [r402065] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_pjsip.c, include/asterisk/res_pjsip_session.h, /:
- chan_pjsip: Fix a crash when direct media is enabled and an ACK
- is received after the channel is hung up. (closes issue
- ASTERISK-22731) Reported by: Kinsey Moore ........ Merged
- revisions 402064 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-26 00:36 +0000 [r402056] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_stasis.c, /: res_stasis.c: Made use the ao2_container
- callback templates. * Made res_stasis.c use the OBJ_SEARCH_XXX
- defines. ........ Merged revisions 402055 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-26 00:27 +0000 [r402054] Scott Griepentrog <sgriepentrog@digium.com>
-
- * main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine:
- fix rtp payloads copy and improve argument names In function
- ast_rtp_instance_early _bridge_make_compatible the use of
- instance 0/1 as arguments doesn't clearly communicate a direction
- that the copying of payloads from the source channel to the
- destination channel will occur, making it more probable to have
- the arguments to ast_rtp_codecs_payloads_copy() put in the
- reverse order. This patch renames the arguments with _dst and
- _src suffixes and corrects the copy direction. (closes issue
- ASTERISK-21464) Reported by: Kevin Stewart Review:
- https://reviewboard.asterisk.org/r/2894/ ........ Merged
- revisions 402000 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows
- rtpmap:119 being copied per this change, but is not in sip invite
- ........ Merged revisions 402042 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 402043 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 23:58 +0000 [r402004-402045] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/taskprocessor.c: taskprocessor: Made use pthread_equal()
- to compare thread ids. * Removed another silly use of RAII_VAR().
- RAII_VAR() and SCOPED_LOCK() are not silver bullets that allow
- you to turn off your brain. ........ Merged revisions 402044 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/stasis/app.c: You'd think that new files would be free of
- whitespace issues. But you would be wrong. ........ Merged
- revisions 402003 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 22:01 +0000 [r401999-402002] Jonathan Rose <jrose@digium.com>
-
- * res/ari/resource_bridges.c, res/res_ari_bridges.c, /,
- rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c, rest-api/api-docs/bridges.json: ARI:
- channel/bridge recording errors when invalid format specified
- Asterisk will now issue 422 if recording is requested against
- channels or bridges with an unknown format (closes issue
- ASTERISK-22626) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/2939/ ........ Merged
- revisions 402001 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_recording.c, rest-api/api-docs/channels.json,
- res/ari/resource_channels.c, res/ari/ari_model_validators.c,
- res/res_ari_channels.c, rest-api/api-docs/bridges.json,
- rest-api/api-docs/recordings.json, res/ari/resource_bridges.c,
- res/ari/ari_model_validators.h, res/res_ari_bridges.c,
- rest-api/api-docs/events.json, /: ARI recordings: Issue HTTP
- failures for recording requests with file conflicts If a file
- already exists in the recordings directory with the same name as
- what we would record, issue a 422 instead of relying on the
- internal failure and issuing success. (closes issue
- ASTERISK-22623) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/2922/ ........ Merged
- revisions 401973 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 20:51 +0000 [r401962] Scott Griepentrog <sgriepentrog@digium.com>
-
- * include/asterisk/pbx.h, main/pbx.c, /: pbx.c: fix confused match
- caller id that deleted exten still in hash This fixes a bug where
- a zero length callerid match adjacent to a no match callerid
- extension entry would be deleted together, which then resulted in
- hashtable references to free'd memory. A third state of the
- matchcid value has been added to indicate match to any extension
- which allows enforcing comparison of matchcid on/off without
- errors. (closes issue AST-1235) Reported by: Guenther Kelleter
- Review: https://reviewboard.asterisk.org/r/2930/ ........ Merged
- revisions 401959 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401960 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401961 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 17:41 +0000 [r401898-401939] Jonathan Rose <jrose@digium.com>
-
- * /, res/res_pjsip/pjsip_distributor.c,
- res/res_pjsip_endpoint_identifier_user.c: PJSIP: Add log messages
- when requests are received for non-existent endpoints (closes
- issue ASTERISK-22552) Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/2934/ ........ Merged
- revisions 401938 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * utils/clicompat.c, utils/refcounter.c, /: Put clicompat-r2.patch
- back in We've figured out how to resolve the problems this was
- causing in 12/trunk, so this can go back in now. (issue
- ASTERISK-22467) Reported by: Corey Farrell Patches:
- clicompat-r2.patch uploaded by coreyfarrell (license 5909)
- ........ Merged revisions 401914 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401935 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401936 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, utils/clicompat.c: revert clicompat-r2.patch from r401704
- Patch caused the following build errors against testsuite
- https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
- (issue ASTERISK-22467) Reported by: Corey Farrell ........ Merged
- revisions 401895 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401896 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401897 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 16:09 +0000 [r401886] Kevin Harwell <kharwell@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Allow a sip peer to accept both
- AVP and AVPF calls Adapts the behaviour of avpf to only impact
- the format of outgoing calls. For inbound calls, both AVP and
- AVPF calls will be accepted regardless of the value of avpf in
- the configuration. (closes issue ASTERISK-22005) Reported by:
- Torrey Searle Patches: optional_avpf_trunk.patch uploaded by
- tsearle (license 5334) ........ Merged revisions 401884 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401885 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-25 13:49 +0000 [r401873] David M. Lee <dlee@digium.com>
-
- * tests/test_json.c, /: test_json: Fix deprecation warnings After a
- series of upgrades over recent weeks, I've discovered that
- test_json.c won't compile in dev mode any more for me. One of
- gcc-4.8.2, OS X Mavericks or Xcode 5 has decided to deprecate
- tempnam. Which, in general, is a good thing. But for test code
- that just needs a temporary file, it's just annoying. This patch
- replaces usage of tempname with mkstemp, avoiding the deprecation
- warning. It also removes the temporary files when the test is
- complete, which apparently we weren't doing before (oops).
- Review: https://reviewboard.asterisk.org/r/2957/ ........ Merged
- revisions 401872 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-24 21:06 +0000 [r401836] Kevin Harwell <kharwell@digium.com>
-
- * /, main/logger.c: Logging: Logging types ignored after specifying
- a verbose level If one specified a verbose level within a logging
- facility in logger.conf then any component after it was ignored.
- Fixed so all values are correctly read. (closes issue
- ASTERISK-22456) Reported by: Kevin Harwell ........ Merged
- revisions 401833 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401835 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-24 20:48 +0000 [r401834] David M. Lee <dlee@digium.com>
-
- * rest-api-templates/models.wiki.mustache,
- rest-api/api-docs/events.json, /,
- rest-api-templates/swagger_model.py,
- rest-api-templates/ari_model_validators.c.mustache: The Swagger
- 1.2 specification for type extension ended up being slightly
- different than my proposal. Instead of putting an 'extends' field
- on the subtype, the base type has a 'subTypes' field, which is a
- list of the subTypes. Given that its a messaging model and not an
- object model, kinda makes sense. This patch changes the
- events.json api-doc, and the python translators to take the new
- format into account. Other changes that are in Swagger 1.2 were
- not adopted, since the spec is still in flux, and could change
- before it's finalized. A summary of changes to the Swagger-1.2
- spec can be found at
- https://github.com/wordnik/swagger-core/wiki/1.2-transition.
- (closes issue ASTERISK-22440) Review:
- https://reviewboard.asterisk.org/r/2909/ ........ Merged
- revisions 401701 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-24 20:34 +0000 [r401622-401832] Jonathan Rose <jrose@digium.com>
-
- * /, main/utils.c: utils: Fix memory leaks and missed
- unregistration of CLI commands on shutdown Final set of patches
- in a series of memory leak/cleanup patches by Corey Farrell
- (closes issue ASTERISK-22467) Reported by: Corey Farrell Patches:
- main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
- main-utils-11.patch uploaded by coreyfarrell (license 5909)
- main-utils-12up.patch uploaded by coreyfarrell (license 5909)
- ........ Merged revisions 401829 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401830 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401831 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, tests/test_linkedlists.c: test_linkedlists: Fix memory leak
- (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
- test_linkedlists-1.8.patch uploaded by coreyfarrell (license
- 5909) test_linkedlists-11up.patch uploaded by coreyfarrell
- (license 5909) ........ Merged revisions 401790 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401791 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401792 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
- reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
- jitterbuf-jb_reset-leak-1.8.patch
- jitterbuf-jb_reset-leak-11up.patch ........ Merged revisions
- 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 401787 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401788 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/astobj2.c, /: astobj2: Unregister debug CLI commands at exit
- (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
- astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
- (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 401781 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401783 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401784 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_voicemail.c, /: app_voicemail: Memory Leaks against
- tests (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
- app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
- app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
- ........ Merged revisions 401743 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401744 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401745 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/app.c, main/asterisk.c, utils/clicompat.c,
- channels/chan_dahdi.c, codecs/ilbc/doCPLC.c, main/data.c, /:
- memory leaks: Memory leak cleanup patch by Corey Farrell (second
- set) Also covers ast_app_parse_timelen-fail-zero-length.patch,
- but the patch was replaced with one of my own. (issue
- ASTERISK-22467) Reported by: Corey Farrell Patches:
- chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license
- 5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909)
- codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
- data-cleanup-test-registration.patch uploaded by coreyfarrell
- (license 5909) main-asterisk-kill-listener.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 401704 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401705 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401706 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, tests/test_dlinklists.c, funcs/func_math.c,
- channels/sip/reqresp_parser.c, main/test.c,
- main/editline/readline.c: memory leaks: Memory leak cleanup patch
- by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
- Corey Farrell Patches:
- chan_sip-parse_contact_header_test-free-contacts.patch uploaded
- by coreyfarrell (license 5909) cli-filename-completion-leak.patch
- uploaded by coreyfarrell (license 5909) func_math.patch uploaded
- by corefarrell (license 5909) main-test-cleanup.patch uploaded by
- coreyfarrell (license 5909) test_dlinklists.patch uploaded by
- coreyfarrell (license 5909) ........ Merged revisions 401660 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401661 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401662 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
- Address jittery DTMF events in RTP streams (closes issue
- ASTERISK-21170) Reported by: NITESH BANSAL Patches:
- dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
- Review: https://reviewboard.asterisk.org/r/2938/ ........ Merged
- revisions 401619 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401620 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401621 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-23 16:52 +0000 [r401582] Richard Mudgett <rmudgett@digium.com>
-
- * /, cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a
- filter when the CDR value is empty. Extra CDR records are written
- if a filtered CDR value is empty because the filter is not
- checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull
- Chavarria ........ Merged revisions 401577 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401579 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401581 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-23 16:48 +0000 [r401580] John Bigelow <jbigelow@digium.com>
-
- * /, main/bridge_channel.c: Add a test suite event to indicate when
- the atxfer 3-way feature is detected This adds a test suite event
- that indicates to tests when the attended transfer three-way call
- feature is detected. Review:
- https://reviewboard.asterisk.org/r/2912/ ........ Merged
- revisions 401578 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-23 15:23 +0000 [r401540] Kinsey Moore <kmoore@digium.com>
-
- * channels/chan_mgcp.c, /: chan_mgcp: Properly handle malformed
- media lines This corrects a situation in which a media line was
- not parsed properly and resulted in a crash. (closes issue
- ASTERISK-21190) Reported by: adomjan Patches:
- chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
- ........ Merged revisions 401537 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401538 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401539 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-23 11:16 +0000 [r401500] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix an issue where an
- incompatible audio format may be added to SDP. If preferred
- codecs included any non-audio format the code would mistakenly
- add the audio format, even if it was not a joint capability with
- the remote side. (closes issue ASTERISK-21131) Reported by:
- nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by
- nbougues (license 6470) ........ Merged revisions 401497 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401498 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401499 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-23 02:36 +0000 [r401489] Michael L. Young <elgueromexicano@gmail.com>
-
- * channels/chan_iax2.c, configs/iax.conf.sample, /: chan_iax2: Fix
- Binding To Multiple Addresses Again When reworking chan_iax2 for
- IPv6, the ability to bind to multiple addresses was removed by
- mistake. This patch restores this functionality and adds notes
- about IPv6 addresses in the sample config. (closes issue
- ASTERISK-22741) Reported by: Joshua Colp Tested by: Michael L.
- Young Patches: asterisk-22741-fix-binding-multiple-addr.diff
- uploaded by Michael L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2945/ ........ Merged
- revisions 401488 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-22 23:10 +0000 [r401450] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP
- is not available during SSRC change In r400089, a patch was put
- in to correct erroneous RTCP statistic resets. Unfortunately,
- ast_rtp_read can be called on an RTP instance that does not have
- RTCP information. This patch prevents that crash by only
- resetting the statistics if we do actually have an RTCP instance.
- (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
- Bigelow ........ Merged revisions 401445 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401446 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401447 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-22 19:04 +0000 [r401421-401435] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_queue.c, /: app_queue: Fix CLI "queue remove member"
- queue_log entry. The queue_log entry resulting from CLI "queue
- remove member" when log_membername_as_agent is enabled is wrong.
- It always uses the interface name instead of the member name in
- the queue_log entry. * Get the queue member before removing it
- from the queue so the member name is available for the queue_log
- entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve
- Patches: fix_membername.diff (license #6505) patch uploaded by
- Oscar Esteve (modified to fix potential ref leak) ........ Merged
- revisions 401433 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401434 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/bridge_channel.c,
- include/asterisk/bridge_channel_internal.h, /, main/bridge.c:
- Bridging: Fix orphaned bridge if neither of the joining channels
- can join. The original issue noted that the bridge is orphaned
- when res_parking.so is not loaded and a call uses the dial kK
- flags. A similar issue happens when only one of the park flags is
- used. In this case you have the bridge with one or the other
- channel left in it. The channel and bridge will stay around until
- the channel hangs up. * Fixed the initial bridge channel push
- failure to act as if the channel were kicked out of the bridge.
- The bridge then decides if it needs to be dissolved. (closes
- issue ASTERISK-22629) Reported by: Kevin Harwell Review:
- https://reviewboard.asterisk.org/r/2928/ ........ Merged
- revisions 401424 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/parking/parking_bridge_features.c,
- res/parking/parking_bridge.c, /: res_parking: Give parking
- timeout comebacktoorigin channel DTMF features. Parking timeouts
- did not set any DTMF features for the channel calling the parker
- back. * Added code to set the parkedcalltransfers,
- parkedcallreparking, parkedcallhangup, and parkedcallrecording
- options appropriately for the channels when a parking timeout
- occurs. The recall channel DTMF options are set using the
- BRIDGE_FEATURES channel variable to allow the other timeout
- options to have the DTMF features available. (closes issue
- ASTERISK-22630) Reported by: Kevin Harwell Review:
- https://reviewboard.asterisk.org/r/2942/ ........ Merged
- revisions 401422 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_parking.c: res_parking: Update XML documention for
- DTMF features after parking timeout. * Updated the XML
- documentation to indicate that the parkedcalltransfers,
- parkedcallreparking, parkedcallhangup, and parkedcallrecording
- configuration options also apply to parking timeouts. (issue
- ASTERISK-22630) Reported by: Kevin Harwell Review:
- https://reviewboard.asterisk.org/r/2942/ ........ Merged
- revisions 401420 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-22 15:17 +0000 [r401411] Joshua Colp <jcolp@digium.com>
-
- * apps/app_dial.c: Add an 'R' option to Dial which sends ringing
- until early media has been received. (closes issue
- ASTERISK-10487) Reported by: Gaspar Zoltan Patches: 10487.patch
- uploaded by n8ideas (license 6075)
-
-2013-10-21 21:06 +0000 [r401365] Mark Michelson <mmichelson@digium.com>
-
- * /, main/bridge_channel.c: Remove a noisy debug message from
- bridging code. This particular debug message, during a stress
- test, was logged so often that it appeared that there may be a
- memory leak in the logger code. In actuality, there was no memory
- leak, but the logger thread was having a hard time keeping up
- with the demands of the rest of the system. Since this debug
- message has no value at all, the best way to fix the problem was
- to just remove the message. (closes issue AST-1225) reported by
- John Bigelow Patches: spammy_log.diff uploaded by Mark Michelson
- (License #5049) ........ Merged revisions 401364 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-21 19:50 +0000 [r401328] Kevin Harwell <kharwell@digium.com>
-
- * /, main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
- tgetstr(), when libncurses5-dev isn't installed Include the
- appropriate declarations when not using termcap, but term+curses
- and [n]curses do not exist. (closes issue ASTERISK-22351)
- Reported by: A. Iglesias Patches:
- issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
- by wdoekes (license 5674) ........ Merged revisions 401325 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401326 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401327 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-21 18:59 +0000 [r401316-401317] David M. Lee <dlee@digium.com>
-
- * rest-api/api-docs/channels.json, /: Fixing r401281; the model
- name is Channel, with a capital C ........ Merged revisions
- 401315 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_ari.c, /: Fixed malformed Access-Control-Allow-Methods
- header. Was causing Safari to barf on POST and DELETE. ........
- Merged revisions 401106 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-19 21:57 +0000 [r401292] Kinsey Moore <kmoore@digium.com>
-
- * /, channels/chan_iax2.c: Fix IAX2 incoming call address lookups
- This fixes address lookup for incoming calls without a peer
- definition. The address family was unset instead of being set to
- AST_AF_UNSPEC which was causing lookup failures on "127.0.0.1".
- This is one of the causes of the current failure of the app_page
- integration test. Review:
- https://reviewboard.asterisk.org/r/2933/ ........ Merged
- revisions 401291 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-19 14:45 +0000 [r401282] Joshua Colp <jcolp@digium.com>
-
- * res/ari/resource_channels.h, main/pbx.c, /,
- rest-api/api-docs/channels.json, res/ari/resource_channels.c,
- res/res_ari_channels.c: Return a channel snapshot when
- originating using ARI, and subscribe the Stasis application to
- it. This change allows a user of ARI to know what channel it has
- originated and also follow any progress. If a Stasis application
- is provided it will be automatically subscribed to the originated
- channel immediately. (closes issue ASTERISK-22485) Reported by:
- David Lee Review: https://reviewboard.asterisk.org/r/2910/
- ........ Merged revisions 401281 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 22:52 +0000 [r401272] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/parking/parking_controller.c: res_parking: Remove setting
- useless flag. ........ Merged revisions 401271 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 21:51 +0000 [r401263] David M. Lee <dlee@digium.com>
-
- * contrib/scripts/get_swagger_ui.sh (added), Makefile, /,
- static-http: This is just a quick script for dumping swagger-ui
- into static-http, so that it can be served by the Asterisk web
- server. I had to change the Makefile in order to recursively
- install content from the static-http directory, hence the code
- review instead of just putting it in. Review:
- https://reviewboard.asterisk.org/r/2924/ ........ Merged
- revisions 401261 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 18:44 +0000 [r401249] Mark Michelson <mmichelson@digium.com>
-
- * main/sorcery.c, main/cli.c, main/manager.c, /, main/bridge.c,
- main/bucket.c: Resolve some memory leaks due to incorrect for
- loop / ao2 ref usage. A common idiom in Asterisk is to due
- something like: for (ao2_obj = list_beginning; ao2_obj =
- next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice
- because it automatically takes care of the object references for
- you. However, there is a pitfall here. If a break statement is in
- the for loop, then the current reference is not cleaned up. In
- some cases, this is on purpose, but in others there is a leak.
- This commit fixes the leak cases. ........ Merged revisions
- 401248 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 16:59 +0000 [r401233-401240] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_fax.c, include/asterisk/channel.h, apps/app_dial.c,
- main/channel.c: Add channel lock protection around translation
- path setup. Most callers of ast_channel_make_compatible() happen
- before the channels enter a two party bridge. With the new
- bridging framework, two party bridging technologies may also call
- ast_channel_make_compatible() when there is more than one thread
- involved with the two channels. * Added channel lock protection
- in set_format() and ast_channel_make_compatible_helper() when
- dealing with the channel's native formats while setting up a
- translation path. * Fixed best_src_fmt and best_dst_fmt usage
- consistency in ast_channel_make_compatible_helper(). The call to
- ast_translator_best_choice() got them backwards. * Updated some
- callers of ast_channel_make_compatible() and the function
- documentation. There is actually a difference between the two
- channels passed in. * Fixed the deadlock potential in res_fax.c
- dealing with ast_channel_make_compatible(). The deadlock
- potential was already there anyway because res_fax called
- ast_channel_make_compatible() with chan locked. (closes issue
- ASTERISK-22542) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2915/ ........ Merged
- revisions 401239 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, include/asterisk/bridge.h: Tweak ast_bridge_depart() doxygen.
- ........ Merged revisions 401232 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 16:06 +0000 [r401216-401224] Mark Michelson <mmichelson@digium.com>
-
- * include/asterisk/bridge.h, /: Remove the bit about requiring
- ast_bridge_depart() to be called before ast_bridge_destroy().
- ........ Merged revisions 401223 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/bridge.h, /: Clarify in ast_bridge_destroy()
- about how departable channels must be handled. ........ Merged
- revisions 401212 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 15:14 +0000 [r401184] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, channels/chan_sip.c: Remove Port Restriction When Checking For
- NAT When trying to determine if a peer is behind NAT, we should
- not be using the ports when comparing addresses. This patch
- removes the port from being checked and just useds the addresses
- now. (closes issue ASTERISK-22729) Reported by: Michael L. Young
- Tested by: Michael L. Young Patches:
- asterisk-remove-using-port-for-nat-check.diff uploaded by Michael
- L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2927/ ........ Merged
- revisions 401182 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401183 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-18 14:50 +0000 [r401181] Walter Doekes <walter+asterisk@wjd.nu>
-
- * main/channel.c, /: Properly copy/remove the device state cache
- flag over a masquerade. In r378303 the
- AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the
- devstate system to not cache states for non-real devices.
- However, when optimizing away channels (ast_do_masquerade), that
- flag wasn't copied. In my case, using Local devices as queue
- members created a situation where the endpoint was considered in
- use, but the state change of the device being available again was
- ignored (not cached). The endpoint channel was optimized into the
- (previously) Local channel, but kept the do-not-cache flag. The
- end result being that the queue member apparently stayed in use
- forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes
- Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged
- revisions 401178 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401179 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401180 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-17 20:39 +0000 [r401169] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, channels/chan_sip.c: Fix Setting A chan_sip Dialog's
- SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix
- ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was
- set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the
- dialog. This condition should not have been there since it
- assumed that if Asterisk is in an environment where NAT is
- involved, that the auto_* nat settings or force_rport setting
- would be on in the global settings. If the nat setting in the
- global setting is set to 'nat=no' and then turned on for peers
- (which is not quite the recommended way, although it is allowed)
- this flag is never copied to the dialog resulting in problems
- like, REGISTER replies going to the wrong port. This patch
- removes this conditional check and will now always use the peer's
- flag which by this point in the code the checks on whether the
- peer is behind NAT or not (if using auto_force_rport) have
- already been run. (closes issue ASTERISK-22236) Reported by:
- Filip Frank Tested by: Michael L. Young Patches:
- asterisk-2236-always-set-rport.diff uploaded by Michael L. Young
- (license 5026) Review: https://reviewboard.asterisk.org/r/2919/
- ........ Merged revisions 401167 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401168 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-17 18:25 +0000 [r401159] Jonathan Rose <jrose@digium.com>
-
- * res/res_parking.c, /: res_parking: Fix bug where reloading
- immediately wipes new parkpos extensions (closes issue
- ASTERISK-22631) Reported by: Kevin Harwell ........ Merged
- revisions 401158 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-17 15:41 +0000 [r401122] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_xmpp.c, res/res_jabber.c: Reduce log level of a
- non-pubsub error message Drop an error log message to debug level
- 1 since distributed device state functions correctly when
- receiving this message and it spams the logs. (closes issue
- ASTERISK-22410) Reported by: abelbeck Patches:
- asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
- uploaded by abelbeck (License 5903)
- asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded
- by abelbeck (License 5903) ........ Merged revisions 401119 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401120 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401121 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-16 21:22 +0000 [r401108] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/ari/resource_playback.c: ARI: Fix crash when POST
- /playback/{id}/control does not have an operation parameter.
- (closes issue ASTERISK-22680) Reported by: John Bigelow ........
- Merged revisions 401107 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-16 17:01 +0000 [r401097] David M. Lee <dlee@digium.com>
-
- * rest-api/resources.json, /: Oops. Leftover /stasis reference
- ........ Merged revisions 401096 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-16 14:02 +0000 [r401088] Kinsey Moore <kmoore@digium.com>
-
- * rest-api/api-docs/bridges.json, res/ari/resource_channels.h, /,
- res/ari/resource_bridges.h, rest-api/api-docs/channels.json:
- Clarify documentation for channel and bridge list This makes it
- clear that the ARI API calls for listing channels and bridges
- will list all channels or bridges in the system and not just
- those that are in or are controlled by a Stasis application.
- (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........
- Merged revisions 401087 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-16 12:19 +0000 [r401079] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, apps/app_queue.c: Don't check all realtime queues when doing
- "queue show some_queue". When using realtime queues, queues have
- to be fetched from the database every now and then to see if any
- info has been changed or to see if the queue has been removed.
- When fetching info for an individual queue, the pruning of other
- queues is unnecessarily costly. Review:
- https://reviewboard.asterisk.org/r/2907/ ........ Merged
- revisions 401049 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 401076 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401077 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-16 00:12 +0000 [r401041] Paul Belanger <paul.belanger@polybeacon.com>
-
- * /, rest-api/api-docs/bridges.json, res/res_ari_bridges.c: Use
- POST / DELETE to toggle ARI bridge moh Review:
- https://reviewboard.asterisk.org/r/2911/ ........ Merged
- revisions 401040 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-15 23:44 +0000 [r401020-401039] Richard Mudgett <rmudgett@digium.com>
-
- * main/translate.c: translate.c: Some minor code tweaks. *
- Consistently compare format2index() return value so matrix_get()
- cannot get passed negative values. * Optimize
- ast_translator_best_choice() to defer initializing things until
- needed. Also cached the matrix_get() return value rather than
- repeatedly calling it.
-
- * /, channels/dahdi/bridge_native_dahdi.c: bridge_native_dahdi:
- Return channel join failure if could not make the channels
- compatible. ........ Merged revisions 401030 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_iax2.c: chan_iax2: Fix channel left locked in
- off nominal code path. ........ Merged revisions 401016 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 401017 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-15 20:03 +0000 [r401019] Kinsey Moore <kmoore@digium.com>
-
- * rest-api/api-docs/bridges.json, res/res_ari_bridges.c, /: Ensure
- bridge record error responses validate This adds the list of
- expected errors to the /bridges/{bridgeId}/record ARI
- documentation so that outbound 4xx errors validate properly.
- Previously, this would result in a response validation failure.
- (closes issue ASTERISK-22627) Reported by: Joshua Colp ........
- Merged revisions 401018 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-15 15:30 +0000 [r401007] Paul Belanger <paul.belanger@polybeacon.com>
-
- * rest-api/api-docs/channels.json, res/res_ari_channels.c, /: Use
- POST / DELETE to toggle hold / moh for ARI channels This change
- updates how we handle toggle events, rather then create two
- different function names, we'll just use POST / DELETE from HTTP
- to handle it. Review: https://reviewboard.asterisk.org/r/2906/
- ........ Merged revisions 400999 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-15 15:26 +0000 [r400998] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: Prevent chan_sip from sending duplicate
- BYEs. When a 200 OK for an initial INVITE is received, we were
- doing the right thing by ACKing and sending an immediate BYE.
- However, we also were doing the wrong thing and queuing an answer
- frame, thus causing the call to be answered. This would cause the
- call to be hung up by the channel thread, thus resulting in a
- second BYE being sent out. In this fix, I also have set the
- hangupcause to be correct since the initial BYE being sent by
- Asterisk had an unknown hangup cause. I have changed to using
- "Bearer capabilty not available" since the call was hung up due
- to an SDP offer/answer error. (closes issue ASTERISK-22621)
- reported by Kinsey Moore ........ Merged revisions 400970 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400971 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400984 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-15 13:44 +0000 [r400959] David M. Lee <dlee@digium.com>
-
- * /, rest-api-templates/asterisk_processor.py: My doc correction in
- r400842 had a silly bug. Because I added a wiki_description to
- models and not their properties, the rendered wiki page had the
- model description instead of the property descriptions, which
- looks very silly indeed. (closes issue ASTERISK-22705) ........
- Merged revisions 400958 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-14 22:52 +0000 [r400913-400950] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample,
- channels/chan_dahdi.h: chan_dahdi: Add config support for hwgain
- settings. * Add hwtxgain and hwrxgain config options to
- chan_dahdi.conf with documentation in chan_dahdi.conf.sample.
- (closes issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
- jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch
- uploaded by rmudgett
-
- * channels/chan_dahdi.c, /, channels/chan_dahdi.h: chan_dahdi:
- Reflect the set software gain in the CLI "dahdi show channel"
- output. * Remember the swgain setting from CLI "dahdi set swgain"
- command so the CLI "dahdi show channel" output will reflect the
- current setting. * Updated CLI "dahdi set hwgain" and "dahdi set
- swgain" documentation. (issue ASTERISK-22429) Reported by: Jaco
- Kroon Patches: jira_asterisk_22429_v1.8_v2.patch (license #5621)
- patch uploaded by rmudgett ........ Merged revisions 400907 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400909 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400911 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-14 22:03 +0000 [r400912] Mark Michelson <mmichelson@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Do not increment the SDP
- version between 183 and 200 responses. Bumping the SDP version
- number can cause interoperability problems since receivers of the
- responses will expect that a 200 SDP will be identical to a
- previous 183 SDP. (closes issue ASTERISK-21204) reported by
- NITESH BANSAL Patches:
- dont-increment-session-version-in-2xx-after-183.patch uploaded by
- NITESH BANSAL (License #6418) ........ Merged revisions 400906
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 400908 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400910 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-14 15:54 +0000 [r400891] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c: pjsip outbound
- registration: Log message says received a 408 when we didn't If
- the server didn't exist that we are trying to register to the log
- message would say that a 408 was received from that server when
- in reality one wasn't. Added log messages stating no response was
- received if the response does not exist. (closes issue
- ASTERISK-22554) Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/2893/ ........ Merged
- revisions 400890 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-14 15:01 +0000 [r400882] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_mwi.c, /: Remove duplicate module info block The
- module info block was repeated twice. Once is sufficient.
- ........ Merged revisions 400881 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-13 15:42 +0000 [r400873] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_session.c, /: Fix a race condition in
- res_pjsip_session with rapidly terminating the session. The
- INVITE session state callback wrongly assumes that a session will
- always exist, but when rapidly terminating the session this
- assumption goes out the window. As all handler code for the
- INVITE session state callback requires the session it will now
- just exit immediately if no session exists. (closes issue
- ASTERISK-22668) Reported by: John Bigelow ........ Merged
- revisions 400872 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-12 16:53 +0000 [r400864] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_pjsip_outbound_authenticator_digest.c: Fix realm
- comparison for outbound auth When generating the list of
- authentication credentials to pass to PJSIP, Asterisk was using
- the raw pointer of a pj_str_t which is not always
- NULL-terminated. This sometimes resulted in incorrect text for
- the realm and a failure to match the realm for authentication
- purposes which was causing the outbound nominal auth pjsip basic
- call test to bounce. This now uses the pj_str_t that contains the
- realm instead of generating a new one. Thanks to John Bigelow for
- helping to narrow this down. ........ Merged revisions 400863
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-11 17:05 +0000 [r400855] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/channel.h, /: channel.h: whitespace changes.
- ........ Merged revisions 400854 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-11 16:36 +0000 [r400851-400852] David M. Lee <dlee@digium.com>
-
- * /, res/ari/resource_bridges.h, rest-api/api-docs/playback.json,
- rest-api-templates/api.wiki.mustache, res/res_ari_playback.c,
- rest-api/api-docs/channels.json, res/ari/resource_playback.h,
- rest-api/api-docs/bridges.json,
- rest-api-templates/asterisk_processor.py,
- res/ari/resource_channels.h,
- rest-api-templates/models.wiki.mustache: Multiple revisions
- 400508,400842-400843,400848 ........ r400508 | dlee | 2013-10-03
- 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line Corrected response
- class for stopPlayback ........ r400842 | dlee | 2013-10-10
- 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line Correct some ARI wiki
- rendering errors ........ r400843 | dlee | 2013-10-10 14:26:19
- -0500 (Thu, 10 Oct 2013) | 1 line Updated /play resource docs.
- The playback of http: resources isn't implemented... yet ........
- r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5
- lines Fix a stupid copy/paste error in ARI docs. Patches:
- ari-doc-patch.txt uploaded by jbigelow (license 5091) ........
- Merged revisions 400508,400842-400843,400848 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /: Fixed merge tracking for r400360, which was somehow lost
-
-2013-10-11 16:28 +0000 [r400850] Richard Mudgett <rmudgett@digium.com>
-
- * bridges/bridge_softmix.c, /: Softmix: Fix crash when switching
- from softmix to another bridge technology. The crash is caused by
- a race condition when switching between native RTP and softmix
- bridging technologies. In this situation, the bridging technology
- is switched from native RTP to softmix, and then back to native
- RTP fast enough that the softmix private data gets destroyed
- before the softmix mixing thread gets started. Thanks to Kinsey
- Moore for the crash analysis. * Fix race condition when starting
- the softmix mixing thread and switching to another bridge
- technology. (closes issue ASTERISK-22678) Reported by: John
- Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621)
- patch uploaded by rmudgett Tested by: John Bigelow ........
- Merged revisions 400849 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-10 18:21 +0000 [r400825-400834] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip/location.c: Perform validation of permanent
- contacts on AORs in res_pjsip. ........ Merged revisions 400833
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_configuration.c, res/res_pjsip.c: Fix an
- assertion in res_pjsip when specifying an invalid outbound proxy.
- This change fixes two issues when setting an outbound proxy: 1.
- The outbound proxy URI was not parsed and validated during
- configuration. 2. If an outgoing dialog was created and the
- outbound proxy could not be set an assertion would occur because
- the usage count on the dialog was not decremented. The
- documentation has also been updated to specify that a full URI
- must be specified for the outbound proxy. (closes issue
- ASTERISK-22672) Reported by: Antti Yrjola ........ Merged
- revisions 400824 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-09 11:02 +0000 [r400772-400813] Matthew Jordan <mjordan@digium.com>
-
- * res/res_pjsip_header_funcs.c, /: Use 'z' as the format specifier
- for size_t Using 'lu' will produce a compiler warning for some
- versions of gcc and on some architectures. 'z' should be portable
- as a format specifier for size_t. ........ Merged revisions
- 400812 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_header_funcs.c (added), /: Add PJSIP_HEADER
- function for manipulation of SIP headers in the PJSIP stack This
- patch adds support to the PJSIP stack in Asterisk for SIP header
- manipulation. Note that this is analagous to
- SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming
- supplemental session callback is registered that takes the
- pjsip_hdrs from the incoming session and stores them in a linked
- list in the session datastore. Calls to PJSIP_HEADER traverse
- over the list and return the nth matching header where 'n' is the
- 'number' argument to the function. When adding a header, the
- first call creates a datastore and linked list and adds the
- datastore to the session. The header is then created as a
- pjsip_hdr and added to the list. An outgoing supplemental session
- callback then traverses the list and adds the headers to the
- outgoing pjsip_msg. When removing a header, the list created with
- PJSIP_HEADER(add,...) is traversed and all matching entries are
- removed. (closes issue ASTERISK-22498) Reported by: George Joseph
- patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph
- (License 6322) ........ Merged revisions 400771 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 22:33 +0000 [r400770] Kinsey Moore <kmoore@digium.com>
-
- * /, configure, configure.ac: Add warning when compiling with iODBC
- support When running configure, libiodbc2 development headers
- will fulfill the requirement for ODBC development headers, but
- will not function properly. This adds a warning when libiodbc2
- development headers are detected instead of unixodbc development
- headers. (closes issue ASTERISK-22459) Reported by: Patrick
- Maille Tested by: Walter Doekes Patches:
- issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
- (License 5674) ........ Merged revisions 400767 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400768 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400769 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 21:20 +0000 [r400759] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_agent_pool.c, /: app_agent_pool: Fix AMI/CLI AgentLogoff
- soft preventing agents from logging back in. * Clear the
- deferred_logoff flag when an agent logs in. (closes issue
- ASTERISK-22669) Reported by: John Bigelow ........ Merged
- revisions 400754 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 20:52 +0000 [r400750] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip.c, res/res_pjsip/config_transport.c: Switch from
- using pjsip_strerror to pj_strerror. pjsip_strerror is only aware
- of PJSIP-specific error codes. pj_strerror() is aware of all
- PJProject error codes and OS-specific error codes. This
- specifically fixes an oft-seen error in transport configuration
- code where EADDRINUSE would result in "Unknown PJSIP error
- 120098" instead of a useful message. ........ Merged revisions
- 400749 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 20:18 +0000 [r400728-400744] Richard Mudgett <rmudgett@digium.com>
-
- * configs/confbridge.conf.sample, /,
- apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
- CHANGES, apps/confbridge/conf_config_parser.c: app_confbridge:
- Can now set the language used for announcements to the
- conference. ConfBridge now has the ability to set the language of
- announcements to the conference. The language can be set on a
- bridge profile in confbridge.conf or by the dialplan function
- CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983)
- Reported by: Jonathan White Patches: M19983_rev2.diff (license
- #5138) patch uploaded by junky (modified) Tested by: rmudgett
- ........ Merged revisions 400741 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400742 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/confbridge/conf_config_parser.c, /: app_confbridge: Fix
- duplicate default_user profile. * Fixed looking in the wrong
- profiles container to see if the default_user profile is already
- created in verify_default_profiles(). The bridge profile
- container is never going to hold user profiles. :) ........
- Merged revisions 400723 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400724 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 18:19 +0000 [r400684-400704] Kinsey Moore <kmoore@digium.com>
-
- * funcs/func_config.c, /: Fix func_config list entry allocation The
- AST_CONFIG dialplan function defined in func_config.c allocates
- its config file list entries using ast_malloc. List entry
- allocations destined for use with Asterisk's linked list API must
- be ast_calloc()d or otherwise initialized so that list pointers
- are set to NULL. These uses of ast_malloc have been replaced by
- ast_calloc to prevent dereferencing of uninitialized pointer
- values when traversing the list. (closes issue ASTERISK-22483)
- Reported by: Brian Scott ........ Merged revisions 400694 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400697 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400701 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: Fix STUN crash when using IPv6 any
- address Ensure that when chan_sip binds to the IPv6 any address
- ([::]), IPv4 candidates are also added. (closes issue
- ASTERISK-21917) Reported by: Torrey Searle Patches:
- 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License
- 5334) ........ Merged revisions 400681 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400682 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 15:44 +0000 [r400683] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip/pjsip_options.c, /: Push CLI qualify into the
- threadpool. If you run Asterisk in the background and then
- connect to it through a separate console, the thread that runs
- CLI commands is not registered with PJLIB. Thus PJLIB does not
- like it when you attempt to send OPTIONS requests from that
- thread. So now we push the task into the threadpool, which we
- know to be registered with PJLIB. Thanks to Antti Yrjola for
- reporting this. ........ Merged revisions 400680 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-08 15:12 +0000 [r400662-400672] Richard Mudgett <rmudgett@digium.com>
-
- * /, res/res_agi.c, apps/app_queue.c: Make app_queue and res_agi
- independent of AMI being enabled. The
- https://reviewboard.asterisk.org/r/2888/ review changes manager
- to not subscribe to stasis when it is disabled for performance
- reasons. When manager is disabled app_queue and res_agi decline
- to load and fail to clean up what they have already allocated. *
- Made app_queue and res_agi clean up allocated resources when they
- decline to load. * Made app_queue and res_agi use their own
- subscriptions to the stasis topics instead of borrowing manager's
- message router structure inappropriately. (closes issue
- ASTERISK-22604) Reported by: rmudgett Review:
- https://reviewboard.asterisk.org/r/2902/ ........ Merged
- revisions 400671 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, include/asterisk/stasis.h, apps/app_queue.c,
- include/asterisk/manager.h: Miscellaneous stand alone comment
- cleanups. ........ Merged revisions 400661 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-06 17:13 +0000 [r400625] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, apps/app_queue.c: app_queue: Fix Queuelog EXITWITHKEY only
- logging two of four fields Commit r62462 added two extra fields
- for logging "the original position the caller entered the queue
- at, and the amount of time the caller was waiting in the queue."
- But when r75969 was merged from 1.4 into trunk (r75977), these
- two fields disappeared. Those two extra fields were not logged in
- 1.4 and when the patch was merged, those fields went away.
- Therefore, this is a regression and was caught by the reporter
- because he was reading the awesome "Asterisk: The Definitive
- Guide" book. (closes issue ASTERISK-22197) Reported by: Dalius M.
- Tested by: Dalius M. Patches:
- asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
- Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2901/ ........ Merged
- revisions 400622 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400623 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400624 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-05 00:59 +0000 [r400593] Richard Mudgett <rmudgett@digium.com>
-
- * /, channels/iax2/include/parser.h: chan_iax2: Fix compile error.
- ........ Merged revisions 400588 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-04 21:41 +0000 [r400568] Michael L. Young <elgueromexicano@gmail.com>
-
- * main/acl.c, include/asterisk/netsock2.h, CHANGES,
- channels/chan_iax2.c, channels/iax2/parser.c, main/netsock.c,
- main/netsock2.c, /, channels/iax2/include/parser.h: Add IPv6
- Support To chan_iax2 This patch adds IPv6 support to chan_iax2.
- Yay! (closes issue ASTERISK-22025) Patches:
- iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026)
- Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged
- revisions 400567 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-04 19:32 +0000 [r400553] David M. Lee <dlee@digium.com>
-
- * rest-api/api-docs/applications.json (added), /: Added missing
- file from r400522 ........ Merged revisions 400552 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-04 19:11 +0000 [r400533-400543] Jonathan Rose <jrose@digium.com>
-
- * res/res_pjsip_logger.c, /: chan_pjsip: Make logger togglable
- without loading/unloading This patch makes the res_pjsip_logger
- do a few things... First, it will be built and installed by
- default now, so end users won't need to enable it in menuselect.
- Second, while it is loaded, it no longer will immediately issue
- log messages. Upon loading, it is in the disabled state and must
- be turned on with the new CLI command. The CLI command 'pjsip set
- logger <on/off/host> has been added and can be used to do the
- following: pjsip set logger on: Enables logger for all PJSIP
- traffic pjsip set logger off: Disables logger for all PJSIP
- traffic pjsip set logger host <host>: Enables logger for the
- specific host Review: https://reviewboard.asterisk.org/r/2900/
- ........ Merged revisions 400542 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /,
- contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py
- (added), configs/extconfig.conf.sample,
- configs/sorcery.conf.sample,
- contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py:
- chan_pjsip: Add alembic scripts for generating db tables for
- PJSIP Also updates sample configurations for sorcery and
- extconfig to demonstrate how to use databases created by that
- alembic script. (closes issue ASTERISK-22133) Reported by: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........
- Merged revisions 400532 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-04 16:01 +0000 [r400523] Matthew Jordan <mjordan@digium.com>
-
- * res/res_stasis.c, main/asterisk.c,
- rest-api/api-docs/endpoints.json, rest-api/api-docs/events.json,
- res/stasis/app.c, /,
- rest-api-templates/ari_model_validators.h.mustache,
- include/asterisk/endpoints.h, res/res_ari_applications.c (added),
- res/ari/resource_endpoints.h, include/asterisk/stasis_app.h,
- res/stasis/app.h, rest-api/resources.json,
- include/asterisk/_private.h, res/ari/ari_model_validators.c,
- main/endpoints.c, res/ari/ari_model_validators.h, main/json.c,
- res/res_ari_model.c, res/ari.make,
- res/ari/resource_applications.c (added),
- res/ari/resource_applications.h (added): ARI: Add subscription
- support This patch adds an /applications API to ARI, allowing
- explicit management of Stasis applications. * GET /applications -
- list current applications * GET /applications/{applicationName} -
- get details of a specific application * POST
- /applications/{applicationName}/subscription - explicitly
- subscribe to a channel, bridge or endpoint * DELETE
- /applications/{applicationName}/subscription - explicitly
- unsubscribe from a channel, bridge or endpoint Subscriptions work
- by a reference counting mechanism: if you subscript to an event
- source X number of times, you must unsubscribe X number of times
- to stop receiveing events for that event source. Review:
- https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451)
- Reported by: Matt Jordan ........ Merged revisions 400522 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-04 15:49 +0000 [r400511-400521] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip.c: Enclose the To URI and update its user
- portion if a request user has been specified. ........ Merged
- revisions 400520 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_session.c, /: Replace the connection address at the
- SDP level if altering the SDP with the external media address.
- ........ Merged revisions 400510 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 23:20 +0000 [r400482] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Don't ignore expires value in
- contact header if it lacks semicolon (closes issue
- ASTERISK-22574) Reported by: Filip Jenicek Patches:
- chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
- ........ Merged revisions 400469 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400470 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400471 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 21:46 +0000 [r400461] Matthew Jordan <mjordan@digium.com>
-
- * /, main/channel_internal_api.c: Remove publication of a channel
- snapshot when the technology is set This patch removes said
- publication for a few reasons: (1) It is unnecessary. Association
- of the channel technology with a specific channel is an
- implementation detail that should be assumed to "just happen",
- and consumers of Stasis don't need to be informed about it. (2)
- Publication of said message can now cause crashes, as the actual
- creation of a channel in normal locations now stages its
- messages. As a result, things that create dummy channels (such as
- the SIP RTP QOS unit test) and associate them with a channel
- technology were now crashing, as the channel itself was not known
- by Stasis. ........ Merged revisions 400460 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 20:22 +0000 [r400452] Mark Michelson <mmichelson@digium.com>
-
- * bridges/bridge_native_rtp.c, /,
- include/asterisk/bridge_technology.h: Fix assumption in
- bridge_native_rtp.c regarding number of participants in a bridge.
- When a party leaves a bridge, there may be more participants in
- the bridge than expected. As such, it is important not to make
- assumptions regarding the list of channels in a bridge. This
- change makes it so that when a party leaves a native RTP bridge,
- we unbridge it and the party it was bridged with. Previously, the
- first and last channels in the list were unbridged since it was
- assumed that these were the two channels that had been bridged.
- As previously stated, a new party had been inserted into the
- bridge, so this logic did not work properly. (closes issue
- ASTERISK-22615) reported by Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2899 ........ Merged revisions
- 400403 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 19:32 +0000 [r400443] Joshua Colp <jcolp@digium.com>
-
- * /, main/cdr.c: When serializing CDR variables (like for "core
- show channels") don't output an error if CDRs aren't enabled.
- ........ Merged revisions 400442 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 19:30 +0000 [r400441] Kinsey Moore <kmoore@digium.com>
-
- * /, main/security_events.c: Fix security events for AMI invalid
- password In r337595, additional security events were added for
- chan_sip authentication failures. The new IEs added to the
- existing invalid password event were defined as required IEs, but
- existing users of the event did not set the new IEs and could not
- since they didn't apply to existing uses. They are now marked as
- optional IEs. (closes issue ASTERISK-22578) Reported by: Matt
- Jordan ........ Merged revisions 400421 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400440 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 19:06 +0000 [r400402] Joshua Colp <jcolp@digium.com>
-
- * res/ari/resource_channels.c, /: Fix a crash caused by muting and
- unmuting a channel in ARI without specifying a direction. (closes
- issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by
- Matt Jordan, whose office I have taken over in the name of
- Canada. ........ Merged revisions 400401 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 18:51 +0000 [r400399] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/cel.c: cel: Some whitespace cleanups ........ Merged
- revisions 400398 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 18:32 +0000 [r400385-400397] Kinsey Moore <kmoore@digium.com>
-
- * res/res_rtp_multicast.c, /: res_rtp_multicast: Ensure SSRC is set
- properly This fixes a bug where the SSRC field on multicast RTP
- can be stuck at 0 which can cause problems for endpoints trying
- to make sense of incoming streams. (closes issue ASTERISK-22567)
- Reported by: Simone Camporeale Patches:
- 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
- (License 6536) ........ Merged revisions 400393 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400394 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400395 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, configure, include/asterisk/autoconfig.h.in, configure.ac,
- main/xml.c: Detect and use xsltCleanupGlobals when available This
- introduces usage of an additional libxslt cleanup function,
- xsltCleanupGlobals, when the configure script detects that it is
- available. Early versions of the library did not include this
- function. (closes issue ASTERISK-22570) Reported by: Corey
- Farrell Patches: xsltCleanupGlobals.patch uploaded by Corey
- Farrell (License 5909) ........ Merged revisions 400384 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 16:28 +0000 [r400374] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_vpb.cc, /: chan_vpb: Make compile again. ........
- Merged revisions 400373 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 14:59 +0000 [r400363-400364] Mark Michelson <mmichelson@digium.com>
-
- * tests/test_cel.c, /: Get rid of uses of stasis_topic_wait()
- ........ Merged revisions 400362 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * pbx/pbx_spool.c, main/manager.c, main/format_cap.c,
- channels/chan_skinny.c, res/res_agi.c, channels/chan_motif.c,
- channels/chan_alsa.c, apps/app_confbridge.c,
- addons/chan_mobile.c, channels/chan_mgcp.c,
- res/res_clioriginate.c, channels/chan_bridge_media.c,
- channels/chan_sip.c, tests/test_format_api.c,
- res/res_pjsip_sdp_rtp.c, bridges/bridge_simple.c,
- apps/app_originate.c, res/parking/parking_applications.c,
- main/core_local.c, channels/chan_console.c, channels/chan_oss.c,
- include/asterisk/format_cap.h, res/res_pjsip_session.c,
- res/ari/resource_bridges.c, channels/chan_jingle.c,
- channels/chan_misdn.c, channels/dahdi/bridge_native_dahdi.c,
- res/res_pjsip/pjsip_configuration.c, main/file.c,
- channels/chan_h323.c, channels/chan_nbs.c,
- bridges/bridge_native_rtp.c, tests/test_config.c,
- res/res_stasis.c, channels/chan_pjsip.c, channels/chan_unistim.c,
- channels/chan_multicast_rtp.c, addons/chan_ooh323.c,
- main/rtp_engine.c, /, main/ccss.c, apps/app_meetme.c,
- bridges/bridge_holding.c, main/bridge_basic.c,
- bridges/bridge_softmix.c, channels/chan_gtalk.c,
- channels/chan_iax2.c, main/media_index.c, main/channel.c,
- channels/chan_phone.c, channels/chan_dahdi.c, main/dial.c: Cache
- string values of formats on ast_format_cap() to save processing.
- Channel snapshots have string representations of the channel's
- native formats. Prior to this change, the format strings were
- re-created on ever channel snapshot creation. Since channel
- native formats rarely change, this was very wasteful. Now, string
- representations of formats may optionally be stored on the
- ast_format_cap for cases where string representations may be
- requested frequently. When formats are altered, the string cache
- is marked as invalid. When strings are requested, the cache
- validity is checked. If the cache is valid, then the cached
- strings are copied. If the cache is invalid, then the string
- cache is rebuilt and copied, and the cache is marked as being
- valid again. Review: https://reviewboard.asterisk.org/r/2879
- ........ Merged revisions 400356 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-03 14:52 +0000 [r400361] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, res/res_pjsip_t38.c, /: Fix crashes in
- res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and
- external_media_address is set. The callback function for changing
- the media address in streams wrongly assumes that a connection
- line will always be present. This is false as no line is present
- if a stream has been rejected. (closes issue ASTERISK-22645)
- Reported by: Rusty Newton ........ Merged revisions 400360 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 22:22 +0000 [r400335] Mark Michelson <mmichelson@digium.com>
-
- * main/stasis_wait.c (removed), res/ari/resource_endpoints.c, /,
- include/asterisk/stasis.h, tests/test_cel.c,
- include/asterisk/stasis_endpoints.h, channels/chan_pjsip.c,
- main/stasis.c, main/stasis_endpoints.c: Multiple revisions
- 400318-400319 ........ r400318 | mmichelson | 2013-10-02 17:08:49
- -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from
- stasis. Since caches are updated on publisher threads, there is
- no need to wait for the cache updates to occur after a stasis
- message is published. In the case of chan_pjsip device state
- changes, this set of changes caused an improvement to
- performance. Review: https://reviewboard.asterisk.org/r/2890
- ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed,
- 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........
- Merged revisions 400318-400319 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 21:33 +0000 [r400317] Michael L. Young <elgueromexicano@gmail.com>
-
- * channels/chan_iax2.c, /: Cast Integer Argument To Unsigned Char
- The member reg in the peercnt structure is an unsigned char and
- peercnt_modify() is expecting an unsigned char argument which
- gets assigned to peercnt->reg. This patch fixes that by casting
- the integer argument being passed to peercnt_modify to unsigned
- char. ........ Merged revisions 400314 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400315 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400316 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 21:26 +0000 [r400313] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, main/manager.c, /, main/cel.c: Only create Stasis
- subscriptions when enabled Subscribing to Stasis isn't free. As
- such, this patch makes AMI, CDR, and CEL - the "big 3" - only
- subscribe when enabled. Toggling their availability via a .conf
- file will unsubscribe/subscribe as appropriate. Review:
- https://reviewboard.asterisk.org/r/2888/ ........ Merged
- revisions 400312 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 20:31 +0000 [r400304] Richard Mudgett <rmudgett@digium.com>
-
- * main/pbx.c, /: Originate: Make setting caller id on outgoing call
- use either name or number. Previous code was requiring both name
- and number to be available. Also restored a comment block on why
- caller id is also set on an outgoing call leg in addition to
- connected line from earlier versions of Asterisk. ........ Merged
- revisions 400303 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 19:20 +0000 [r400295] Kinsey Moore <kmoore@digium.com>
-
- * /, rest-api/api-docs/asterisk.json: Correct allowable values for
- ARI general information filter ........ Merged revisions 400291
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 19:17 +0000 [r400287] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: Fix the CDR CLI command 'cdr show active
- {channel}' When the switch from channel names to channel unique
- IDs happened, the poor CLI command got left in the dust. This
- fixes the command so that users can once again see how Asterisk
- is messing up your billing information. ........ Merged revisions
- 400286 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 18:44 +0000 [r400285] Joshua Colp <jcolp@digium.com>
-
- * /, res/res_pjsip_t38.c: Fix a crash in res_pjsip_t38 caused by
- the wrong assumption that a session will always have a channel.
- When starting up or shutting down this assumption is false.
- ........ Merged revisions 400284 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 18:28 +0000 [r400282] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
-
- * Makefile, doc/astdb2sqlite3.8 (added), /, doc/astdb2bdb.8
- (added): man pages for astdb2bdb and astdb2sqlite3 Review:
- https://reviewboard.asterisk.org/r/2898/ ........ Merged
- revisions 400279 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400281 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 17:12 +0000 [r400269-400271] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_stack.c, res/stasis_recording/stored.c, main/json.c,
- main/stasis_cache.c, res/res_ari.c, /, main/utils.c:
- MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is
- enabled. * There were several places in ARI where an external
- library was mallocing memory that must always be released with
- free(). When MALLOC_DEBUG is enabled, free() is redirected to the
- MALLOC_DEBUG version. Since the external library call still uses
- the normal malloc(), MALLOC_DEBUG complains that the freed memory
- block is not registered and will not free it. These cases must
- use ast_std_free(). * Changed calls to asprintf() and vasprintf()
- to the equivalent ast_asprintf() and ast_vasprintf() versions
- respectively. ........ Merged revisions 400270 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/sig_ss7.c, /: sig_ss7: Fix compiler warnings. ........
- Merged revisions 400268 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-02 16:23 +0000 [r400246-400266] Joshua Colp <jcolp@digium.com>
-
- * channels/chan_alsa.c, main/stasis_channels.c, channels/sig_ss7.c,
- channels/chan_pjsip.c, channels/chan_mgcp.c,
- channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, /,
- channels/chan_sip.c, main/bridge.c, include/asterisk/channel.h,
- channels/chan_gtalk.c, channels/chan_console.c,
- channels/sig_pri.c, channels/chan_iax2.c, channels/chan_jingle.c,
- main/channel.c, channels/chan_dahdi.c, main/dial.c,
- include/asterisk/stasis_channels.h, channels/chan_skinny.c,
- channels/chan_motif.c: Reduce channel snapshot creation and
- publishing by up to 50%. This change introduces the ability to
- stage channel snapshot creation and publishing by suppressing the
- implicit creation and publishing that some functions have. Once
- all operations are executed the staging is marked as done and a
- single snapshot is created and published. Review:
- https://reviewboard.asterisk.org/r/2889/ ........ Merged
- revisions 400265 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_session.c, /: Fix a random one way audio issue in
- PJSIP. Due to the asynchronous design of the PJMEDIA SDP
- negotiator it was possible for the SDP to be negotiated *after* a
- channel was created and after it was being wait on by an
- application. It is only after negotiation occurs that the file
- descriptors for RTP are placed on the channel. Since the channel
- was already being waited on these file descriptors were not
- monitored, causing incoming media to never be read. This change
- wakes up any application waiting on the channel so that added
- file descriptors end up being monitored. (closes issue AST-1227)
- Reported by: John Bigelow ........ Merged revisions 400256 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/stasis/control.c, include/asterisk/stasis_app.h,
- res/ari/resource_channels.c: Allow specifying a channel to dial
- an extension and context in an ARI dial operation. (issue
- ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged
- revisions 400254 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip_session.c: Retrieve and store the hostname only
- once so multiple threads do not potentially initialize it at the
- same time. ........ Merged revisions 400245 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-01 21:19 +0000 [r400228-400237] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_dahdi.c, channels/sig_analog.c, /: chan_dahdi: Fix
- analog parking using flash-hook. Transferring an analog call
- using a flash-hook to parking would fail to park the call and
- result in an invalid ao2 object unref. * Park the correct bridged
- channel. ........ Merged revisions 400236 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/features_config.c, /: Features: Rearm the parking config
- options have moved warning for each reload. ........ Merged
- revisions 400227 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-10-01 15:54 +0000 [r400218] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: Filter out internal channels for bridge leave
- messages and parked call messages Granted, if you manage to park
- a Conference announcer channel, something has gone horrifically
- wrong. ........ Merged revisions 400217 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-30 21:40 +0000 [r400206] Jonathan Rose <jrose@digium.com>
-
- * configs/features.conf.sample, /, configs/res_parking.conf.sample:
- configuration samples: Pull all parking related stuff out of
- features.conf This patch also adds documentation for parking from
- features.conf to res_parking.conf ........ Merged revisions
- 400205 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-30 19:58 +0000 [r400195-400197] Matthew Jordan <mjordan@digium.com>
-
- * /, funcs/func_cdr.c: Parse arguments passed to the CDR_PROP
- function correctly I can only blame this on a bad merge, because
- this in no way worked properly the way it was written. Mea culpa.
- The function should now parse its arguments correctly and
- function properly. (Note that the API used by the CDR_PROP
- function has working unit tests... this was merely bad coding of
- the actual registered function) (closes issue ASTERISK-22613)
- Reported by: Private Name ........ Merged revisions 400196 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: Remove spurious event raised when CDRs are
- reloaded The Reload event is now raised by the module loading
- core. As such, the Reload event in the CDR engine was a duplicate
- and not needed. ........ Merged revisions 400194 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-30 18:55 +0000 [r400186] David M. Lee <dlee@digium.com>
-
- * tests/test_devicestate.c, include/asterisk/sem.h (added),
- tests/test_taskprocessor.c, res/res_pjsip_mwi.c,
- res/res_pjsip/include/res_pjsip_private.h, tests/test_stasis.c,
- res/parking/parking_manager.c, res/res_security_log.c,
- channels/chan_mgcp.c, main/stasis_cache_pattern.c, main/pbx.c,
- include/asterisk/vector.h (added), /, main/ccss.c,
- apps/app_meetme.c, include/asterisk/taskprocessor.h,
- configs/stasis.conf.sample (removed), configure.ac,
- res/parking/parking_applications.c, channels/sig_pri.c,
- apps/app_queue.c, main/cel.c, main/stasis.c,
- channels/chan_dahdi.c, funcs/func_presencestate.c,
- main/stasis_message_router.c, configure,
- apps/confbridge/confbridge_manager.c, res/res_agi.c,
- main/manager_system.c, res/res_stasis_test.c, main/sem.c (added),
- main/manager_channels.c, res/res_pjsip_refer.c,
- main/manager_mwi.c, apps/app_voicemail.c, main/stasis_cache.c,
- main/stasis_wait.c, main/stasis_config.c (removed),
- include/asterisk/stasis_internal.h, res/stasis/app.c,
- channels/chan_sip.c, include/asterisk/autoconfig.h.in,
- main/manager_endpoints.c, main/channel_internal_api.c,
- include/asterisk/stasis.h, main/devicestate.c,
- main/taskprocessor.c, res/res_xmpp.c, main/sounds_index.c,
- include/asterisk/stasis_message_router.h, channels/chan_iax2.c,
- res/res_jabber.c, main/endpoints.c, main/astobj2.c,
- res/res_chan_stats.c, res/parking/parking_bridge_features.c,
- tests/test_stasis_endpoints.c, main/cdr.c, main/channel.c,
- main/manager_bridges.c, main/manager.c, channels/chan_skinny.c:
- Multiple revisions 399887,400138,400178,400180-400181 ........
- r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1
- line Minor performance bump by not allocate manager variable
- struct if we don't need it ........ r400138 | dlee | 2013-09-30
- 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance
- improvements This patch addresses several performance problems
- that were found in the initial performance testing of Asterisk
- 12. The Stasis dispatch object was allocated as an AO2 object,
- even though it has a very confined lifecycle. This was replaced
- with a straight ast_malloc(). The Stasis message router was
- spending an inordinate amount of time searching hash tables. In
- this case, most of our routers had 6 or fewer routes in them to
- begin with. This was replaced with an array that's searched
- linearly for the route. We more heavily rely on AO2 objects in
- Asterisk 12, and the memset() in ao2_ref() actually became
- noticeable on the profile. This was #ifdef'ed to only run when
- AO2_DEBUG was enabled. After being misled by an erroneous comment
- in taskprocessor.c during profiling, the wrong comment was
- removed. Review: https://reviewboard.asterisk.org/r/2873/
- ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep
- 2013) | 24 lines Taskprocessor optimization; switch Stasis to use
- taskprocessors This patch optimizes taskprocessor to use a
- semaphore for signaling, which the OS can do a better job at
- managing contention and waiting that we can with a mutex and
- condition. The taskprocessor execution was also slightly
- optimized to reduce the number of locks taken. The only
- observable difference in the taskprocessor implementation is that
- when the final reference to the taskprocessor goes away, it will
- execute all tasks to completion instead of discarding the
- unexecuted tasks. For systems where unnamed semaphores are not
- supported, a really simple semaphore implementation is provided.
- (Which gives identical performance as the original taskprocessor
- implementation). The way we ended up implementing Stasis caused
- the threadpool to be a burden instead of a boost to performance.
- This was switched to just use taskprocessors directly for
- subscriptions. Review: https://reviewboard.asterisk.org/r/2881/
- ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep
- 2013) | 28 lines Optimize how Stasis forwards are dispatched This
- patch optimizes how forwards are dispatched in Stasis.
- Originally, forwards were dispatched as subscriptions that are
- invoked on the publishing thread. This did not account for the
- vast number of forwards we would end up having in the system, and
- the amount of work it would take to walk though the forward
- subscriptions. This patch modifies Stasis so that rather than
- walking the tree of forwards on every dispatch, when forwards and
- subscriptions are changed, the subscriber list for every topic in
- the tree is changed. This has a couple of benefits. First, this
- reduces the workload of dispatching messages. It also reduces
- contention when dispatching to different topics that happen to
- forward to the same aggregation topic (as happens with all of the
- channel, bridge and endpoint topics). Since forwards are no
- longer subscriptions, the bulk of this patch is simply changing
- stasis_subscription objects to stasis_forward objects (which,
- admittedly, I should have done in the first place.) Since this
- required me to yet again put in a growing array, I finally
- abstracted that out into a set of ast_vector macros in
- asterisk/vector.h. Review:
- https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee
- | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove
- dispatch object allocation from Stasis publishing While looking
- for areas for performance improvement, I realized that an unused
- feature in Stasis was negatively impacting performance. When a
- message is sent to a subscriber, a dispatch object is allocated
- for the dispatch, containing the topic the message was published
- to, the subscriber the message is being sent to, and the message
- itself. The topic is actually unused by any subscriber in
- Asterisk today. And the subscriber is associated with the
- taskprocessor the message is being dispatched to. First, this
- patch removes the unused topic parameter from Stasis subscription
- callbacks. Second, this patch introduces the concept of
- taskprocessor local data, data that may be set on a taskprocessor
- and provided along with the data pointer when a task is pushed
- using the ast_taskprocessor_push_local() call. This allows the
- task to have both data specific to that taskprocessor, in
- addition to data specific to that invocation. With those two
- changes, the dispatch object can be removed completely, and the
- message is simply refcounted and sent directly to the
- taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/
- ........ Merged revisions 399887,400138,400178,400180-400181 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-30 15:57 +0000 [r400142] Kinsey Moore <kmoore@digium.com>
-
- * /, channels/chan_sip.c, configs/pjsip.conf.sample,
- res/res_pjsip_outbound_registration.c, configs/sip.conf.sample,
- CHANGES: chan_sip: Allow Asterisk to retry after 403 on register
- This adds a global option in chan_sip to allow it to continue
- attempting registration if a 403 is received, clearing the cached
- nonce and treating it as a non-fatal response. Normally, this
- would cause registration attempts to that endpoint to stop. This
- also adds a similar per-outbound-registration option to
- chan_pjsip which allows the retry interval to be altered for 403
- responses to REGISTER requests. (closes issue ASTERISK-17138)
- Review: https://reviewboard.asterisk.org/r/2874/ Reported by:
- Rudi ........ Merged revisions 400137 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400140 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400141 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-28 22:57 +0000 [r400059-400122] Matthew Jordan <mjordan@digium.com>
-
- * /, res/res_pjsip_notify.c, configs/pjsip_notify.conf.sample
- (added): res_pjsip_notify: Add documentation We forgot to add
- documentation for res_pjsip_notify, which would prevent it from
- being loaded. Whoops. This patch also updates res_pjsip_notify to
- use pjsip_notify.conf, which now has its own sample file in the
- configs directory as well. Review:
- https://reviewboard.asterisk.org/r/2835/ ........ Merged
- revisions 400121 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Correct erroneous
- lost packet information in RTCP reports RTCP's calculation of the
- number of lost packets in an RTP stream is based on that stream's
- sequence number count, the number of received packets, and how
- many packets we expect to receive. When the SSRC for an RTP
- stream changes, there can - and almost always will be - a large
- jump in the next packet's timestamp and sequence number. If we
- don't reset the number of received packets, sequence number
- count, and other metrics used by RTCP, the next RR/SR report will
- use the previous SSRC's values to calculate the lost packet count
- for the new SSRC - resulting in a very large number of lost
- packets. This patch modifies res_rtp_asterisk such that, if it
- detects a SSRC change, it will reset the various values used by
- the RTCP calculations. From the perspective of RTCP, this appears
- as a new media stream - which is what it is. Review:
- https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
- Reported by: Thomas Arimont ........ Merged revisions 400089 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400093 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400108 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, configure, configure.ac: Add check for openSUSE when detecting
- bfd library In ASTERISK-17842, some additional library checks
- were added to the configure script so that the bfd library could
- be found on CentOS and Fedora systems. As it turns out, openSUSE
- requires an additional library. This patch adds another check to
- the configure script for openSUSE that will add that library.
- Review: https://reviewboard.asterisk.org/r/2885/ (closes issue
- AST-1169) Reported by: Guenther Kelleter ........ Merged
- revisions 400073 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400075 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400077 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: CDR: Improve handling of parking; resolve
- assertion when originating into park This patch covers two
- problems: 1) Currently, when a call is transferred into a parking
- lot from a bridge (using either the blind transfer or one touch
- parking mechanisms), the application fails to be set to "Park" in
- the resulting CDR record for the parked channel. This is due to
- the ParkedCall message arriving before the BridgeEnter for the
- channel entering the parking bridge. The ParkedCall message isn't
- handled as the CDR for the channel has already been finalized
- (due to the channel having left its two party bridge), and the
- BridgeEnter - which creates the new CDR - doesn't have the
- parking information. This patch modifies the behavior so that
- reception of a ParkedCall message will - if not handled by a CDR
- chain - cause a new CDR to be created and put into the Parking
- state. 2) It fixes a FRACK that occurred when a channel is
- originated into a parking space. The DialedPending state - which
- occurs for both Dialed and Originated channels - assumed that it
- couldn't handle the parking transitions due to it having a Party
- B; however, Originated channels don't have a Party B. As such,
- the existing CDR needs to transition into the parking state -
- this patch does that. Review:
- https://reviewboard.asterisk.org/r/2877/ (closes issue
- ASTERISK-22482) Reported by: Richard Mudgett ........ Merged
- revisions 400062 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_queue.c: app_queue: Make manager events tolerant of
- Local channel shenanigans app_queue currently attempts to handle
- Local channel optimizations in an effort to provide accurate
- information in Stasis messages (and their corresponding AMI
- events) as well as the Queue log. Sometimes, however, things
- don't go as planned. Consider the following scenario: SIP/foo <->
- L;1 <-> L;2 <-> SIP/agent SIP/agent answers, triggering a Local
- channel optimization. app_queue will normally do the following: *
- Listen for the Local optimization events and update our agent
- accordingly to SIP/agent in the queue log and messages * When we
- get a hangup, publish the AgentComplete event based on our
- information (SIP/foo and SIP/agent) However, as with all things
- that depend on sanity from something as capricious as Local
- channels, things can go wrong: (1) SIP/agent immediately hangs up
- upon answering. This triggers a race condition between
- termination messages coming from SIP/agent and the ongoing Local
- channel optimization messages. (Note that this can also occur
- with SIP/foo) (2) In a race condition, Asterisk can (rarely)
- deliver the hangup messages prior to the Local channel
- optimization. In that case, the messages *may* arrive to
- app_queue in the following order: * Hangup SIP/Agent * Hangup
- SIP/foo * Optimize L;1/L;2 * Hangup L;2 * Hangup L;1 When
- app_queue receives the hangup of the agent or the caller, it will
- attempt to publish the AgentComplete event. However, it now has a
- problem - it thinks its agent is the ;1 side of the Local
- channel, as it never received the optimization event. At the same
- time, that channel is already gone. This results in getting NULL
- from the Stasis cache. What's more, we can't really wait for the
- optimization message, as we are currently handling the hangup of
- the channel that the optimization event would tell us to use.
- This patch modifies the behavior in app_queue such that, since we
- still have a lot of pertinent queue information (interface, queue
- name, etc.), we now raise the event with what information we
- know. The channels involved now may or may not be present. Users
- will still at least get the "AgentComplete" event, which
- "completes" the known Agent information. Review:
- https://reviewboard.asterisk.org/r/2878/ (closes issue
- ASTERISK-22507) Reported by: Richard Mudgett ........ Merged
- revisions 400060 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/manager.c, /: manager: Fix crash when appending a manager
- channel variable In r399887, a minor performance improvement was
- introduced by not allocating the manager variable struct if it
- wasn't used. Unfortunately, when directly accessing an
- ast_channel struct, manager assumed that the struct was always
- allocated. Since this was no longer the case, things got a bit
- crashy. This fixes that problem by simply bypassing appending
- variables if the manager channel variable struct isn't there.
- ........ Merged revisions 400058 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 21:58 +0000 [r400016-400021] Richard Mudgett <rmudgett@digium.com>
-
- * apps/app_cdr.c, res/res_parking.c, /: app_cdr and res_parking:
- Fix some resource leaks. * app_cdr left the ResetCDR application
- registered. * res_parking leaked a ref to config global. (closes
- issue ASTERISK-22566) Reported by: Corey Farrell Patches:
- ASTERISK-22566-r2.patch (license #5909) patch uploaded by Corey
- Farrell ........ Merged revisions 400020 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/sip/reqresp_parser.c, /, channels/chan_sip.c: chan_sip:
- Increase some scratch buffer sizes dealing with caller id. *
- Eliminated an unnecessary initialization in check_user_full().
- (closes issue ASTERISK-22477) Reported by: Michael Shepelev
- ........ Merged revisions 400013 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 400014 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 400015 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 19:18 +0000 [r400000] Sean Bright <sean@malleable.com>
-
- * configs/sip.conf.sample: Remove some trailing whitespace and
- steal revision 400000.
-
-2013-09-27 18:28 +0000 [r399991] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip.c, res/res_pjsip_session.c,
- include/asterisk/res_pjsip.h, res/res_pjsip.exports.in:
- res_pjsip: crash when using localnet and
- external_signaling_address options There was a collision of
- mod_data use on the transaction between using a nat hook and an
- session response callback. During state change it was assumed
- what was in the mod_data was nothing or the response callback.
- However, it was possible for it to also contain a nat hook thus
- resulting in a bad cast and a crash. Added the ability to store
- multiple data elements in mod_data via a hash table. In this
- instance, mod_data now stores a hash table of the two values that
- can be retrieved using an associated string key. (closes issue
- ASTERISK-22394) Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/2843/ ........ Merged
- revisions 399990 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 17:46 +0000 [r399978] Jonathan Rose <jrose@digium.com>
-
- * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
- Reject calls on 200 OKs if no SDP has been received When Asterisk
- receives a 200 OK in response to an invite, that peer should have
- sent an SDP at some point by then. If the channel has never
- received an SDP, media won't have been set and the remote address
- won't be known. Endpoints in general should not be doing this.
- This patch makes it so that Asterisk will simply hang up a call
- if it sends a 200 OK at this point. So far this odd behavior for
- endpoints has only been observed in tests which involved manually
- created SIP transactions in SIPp. (closes issue ASTERISK-22424)
- Reported by: Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/2827/ ........ Merged
- revisions 399939 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399962 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399976 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 17:11 +0000 [r399938] Richard Mudgett <rmudgett@digium.com>
-
- * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c,
- /: astobj2: Remove OBJ_CONTINUE support. OBJ_CONTINUE was a
- strange feature that came into the world under suspicious
- circumstances to support an abuse of the ao2_container by
- chan_iax2. Since chan_iax2 no longer uses OBJ_CONTINUE, it is
- safe to remove it. The simplified code should help performance
- slightly and make understanding the code easier. Review:
- https://reviewboard.asterisk.org/r/2887/ ........ Merged
- revisions 399937 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 14:35 +0000 [r399925] Mark Michelson <mmichelson@digium.com>
-
- * /, bridges/bridge_native_rtp.c: Fix refleaks of ast_rtp_instance
- structures. These refleaks were causing bridged calls not to
- close their RTP ports. Thus a call would leave open 4 ports (RTP
- for party A, RTCP for party A, RTP for party B, and RTCP for
- party B). This led to an eventual depletion of available RTP
- ports. ........ Merged revisions 399924 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-27 14:08 +0000 [r399913] Kinsey Moore <kmoore@digium.com>
-
- * tests/test_cel.c, main/cel.c, /, include/asterisk/cel.h: Restore
- usefulness of the CEL Peer field This change makes the CEL peer
- field useful again for BRIDGE_ENTER and BRIDGE_EXIT events and
- fills the field with a comma-separated list of all channels in
- the bridge other than the channel that is entering or exiting the
- bridge. Review: https://reviewboard.asterisk.org/r/2840/ (closes
- issue ASTERISK-22393) ........ Merged revisions 399912 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-26 18:51 +0000 [r399898] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_registrar.c, include/asterisk/res_pjsip.h,
- res/res_pjsip.exports.in, /, res/res_pjsip/security_events.c:
- pjsip: race condition in registrar While handling a registration
- request a race condition could occur if/when two+ clients
- registered at the same time. This happened when one request
- obtained a copy of the current contacts for an AOR and another
- request did the same before the first request updated. Thus the
- second would update and overwrite the first (or vice-versa
- depending on which actually updated first). In the case of it
- being the same contact two "add" events would be raised. pjsip
- registration handling is now serialized to alleviate this issue.
- (closes issue AST-1213) Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/2860/ ........ Merged
- revisions 399897 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-26 14:13 +0000 [r399875] Rusty Newton <rnewton@digium.com>
-
- * /, apps/app_dial.c: Adding a few words to the Dial option 'r'
- help text to clarify its tone argument description ........
- Merged revisions 399874 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-25 20:38 +0000 [r399844] Richard Mudgett <rmudgett@digium.com>
-
- * channels/sig_ss7.c, channels/chan_dahdi.c, /: chan_dahdi: CLI
- "core stop gracefully" has needless delay for PRI and SS7. The
- PRI and SS7 link control threads are not stopped correctly when
- the chan_dahdi.so module is unloaded. The link control threads
- pri_dchannel() and ss7_linkset() are not awakened from a poll()
- to cancel the thread. * Added a SIGURG signal after requesting
- the thread cancel to break the link control thread poll()
- immediately. For SS7 it was slightly worse, the link poll()
- timeout would always be whatever was the last libss7 scheduled
- event time used. If no libss7 scheduled event was pending, the
- thread could run more often than necessary. * Set nextms to 60
- seconds for the ss7_linkset() poll() if there is no other libss7
- scheduled event. ........ Merged revisions 399818 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399834 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399842 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-25 19:43 +0000 [r399799] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip.c: Broke the build - Fixing XML DTD violation
- added in r399782, missing <para> tags inside a <note> ........
- Merged revisions 399798 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-25 19:29 +0000 [r399797] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, channels/chan_sip.c: chan_sip: Fix Realtime Peer Update
- Problem When Un-registering And Expires Header In 200ok 1st Issue
- When a realtime peer sends an un-REGISTER request, Asterisk
- un-registers the peer but the database table record still has
- regseconds and fullcontact for the peer. This results in calls
- attempting to be routed to the peer which is no longer
- registered. The expected behavior is to get busy/congested when
- attempting to call an un-registered peer through the dialplan.
- What was discovered is that we are clearing out the peer's
- registration in the database in parse_register_contact() when
- calling expire_register() but then upon returning from
- parse_register_contact(), update_peer() is run which stores back
- in the database table regseconds and fullcontact. 2nd Issue The
- reporter pointed out that the 200 ok being returned by Asterisk
- after un-registering a peer contains a Contact header with
- ;expires= and the Expires header is not set to 0. This is
- actually a regression. Tests were created for this second issue
- (ASTERISK-22548). The tests have been reviewed and a Ship It! was
- received on those tests. This patch does the following: * Do not
- ignore the Expires header value even when it is set to 0. The
- patch sets the pvt->expiry earlier on in the function so that it
- is set properly and used. * If pvt->expiry is 0, do not call
- update_peer since that means the peer has already been
- un-registered and there is no need to update the database record
- again since nothing has changed. (closes issue ASTERISK-22428)
- Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L.
- Young Patches:
- asterisk-22428-rt-peer-update-and-expires-header.diff by Michael
- L. Young (license 5026) Review:
- https://reviewboard.asterisk.org/r/2869/ ........ Merged
- revisions 399794 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399795 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399796 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-25 18:38 +0000 [r399782] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip.c: Fixing documentation for the configOption
- "external_media_address" of both Endpoints and Transports
- Re-using some of Mark Michelson's text from an E-mail discussion
- for: * Modifying synopsis for both options * Adding description
- to both options * Changing name of "external_media_address" for
- Endpoint configuration to "media_address" in anticipation of the
- option name being changed. (As it is not really specific to
- external destinations) (issue ASTERISK-22405) (closes issue
- ASTERISK-22405) Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/2850/ ........ Merged
- revisions 399781 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-24 22:55 +0000 [r399737-399750] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/astobj2.c: astobj2: Made use OBJ_SEARCH_xxx identifiers
- as field enum values internally. * Made ao2_unlink to protect
- itself from stray OBJ_SEARCH_xxx values passed in. ........
- Merged revisions 399749 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Prevent some needless
- breaking of the native IAX2 bridge. * Clean up some twisted code
- in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and
- AST_CONTROL_SRCCHANGE to a list of frames to prevent the native
- bridge loop from breaking. * Passing the
- AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a
- native IAX2 bridge. (issue ABE-2912) Review:
- https://reviewboard.asterisk.org/r/2870/ ........ Merged
- revisions 399697 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399708 from
- http://svn.asterisk.org/svn/asterisk/branches/11 For v12 and
- above this is really just documentation until IAX2 native
- bridging is restored. ........ Merged revisions 399736 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-24 19:22 +0000 [r399667-399696] Matthew Jordan <mjordan@digium.com>
-
- * apps/app_queue.c, /: app_queue: Don't be quite so aggressive in
- initializing the array We only need the first character. ........
- Merged revisions 399695 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_queue.c, /: app_queue: Initialize array holding
- MixMonitor exec options If the channel variable MONITOR_EXEC is
- set, app_queue will pass the specified execution parameters to
- the MixMonitor application when a queue is recorded. If that
- channel variable is not set, the buffer that holds the escaped
- value was not being initialized to NULL, and so would be passed
- to the MixMonitor application with garbage. Hilarity ensued as
- app_mixmonitor attempted to execute gobeldy-gook. ........ Merged
- revisions 399681 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/stasis_bridges.c, tests/test_cdr.c, main/cdr.c, /: Fix a
- performance problem CDRs There is a large performance price
- currently in the CDR engine. We currently perform two
- ao2_callback calls on a container that has an entry for every
- channel in the system. This is done to create matching pairs
- between channels in a bridge. As such, the portion of the CDR
- logic that this patch deals with is how we make pairings when a
- channel enters a mixing bridge. In general, when a channel enters
- such a bridge, we need to do two things: (1) Figure out if anyone
- in the bridge can be this channel's Party B. (2) Make pairings
- with every other channel in the bridge that is not already our
- Party B. This is a two step process. In the first step, we look
- through everyone in the bridge and see if they can be our Party B
- (single_state_process_bridge_enter). If they can - yay! We mark
- our CDR as having gotten a Party B. If not, we keep searching. If
- we don't find one, we wait until someone joins who can be our
- Party B. Step 2 is where we changed the logic
- (handle_bridge_pairings and bridge_candidate_process).
- Previously, we would first find candidates - those channels in
- the bridge with us - from the active_cdrs_by_channel container.
- Because a channel could be a candidate if it was Party B to an
- item in the container, the code implemented multiple
- ao2_container callbacks to get all the candidates. We also had to
- store them in another container with some other meta information.
- This was rather complex and costly, particularly if you have 300
- Local channels (600 channels!) going at once. Luckily, none of it
- is needed: when a channel enters a bridge (which is when we're
- figuring all this stuff out), the bridge snapshot tells us the
- unique IDs of everyone already in the bridge. All we need to do
- is: For all channels in the bridge: If the channel is us or our
- Party B that we got in step 1, skip it Compare us and the
- candidate to figure out who is Party A (based on some specific
- rules) If we are Party A: Make a new CDR for us, append it to our
- chain, and set the candidate as Party B If they are Party A: If
- they don't have a Party B: Make a new CDR for them, append us to
- their chain, and us as Party B Otherwise: Copy us over as Party B
- on their existing CDR. This patch does that. Because we now use
- channel unique IDs to find the candidates during bridging,
- active_cdrs_by_channel now looks up things using uniqueid instead
- of channel name. This makes the more complex code simpler; it
- does, however, have the drawback that dialplan applications and
- functions will be slightly slower as they have to iterate through
- the container looking for the CDR by name. That's a small price
- to pay however as the bridging code will be called a lot more
- often. This patch also does two other minor changes: (1) It
- reduces the container size of the channels in a bridge snapshot
- to 1. In order to be predictable for multi-party bridges, the
- order of the channels in the container must be stable; that is,
- it must always devolve to a linked list. (2) CDRs and the
- multi-party test was updated to show the relationship between two
- dialed channels. You still want to know if they talked -
- previously, dialed channels were always ignored, which is wrong
- when they have managed to get a Party B. (closes issue
- ASTERISK-22488) Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/2861/ ........ Merged
- revisions 399666 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-23 12:03 +0000 [r399625] Joshua Colp <jcolp@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_session.c, /: Fix crash in
- res_pjsip on load if error occurs, and prevent unloading of
- res_pjsip and res_pjsip_session. During load time in res_pjsip if
- an error occurred the operation would attempt to rollback all
- operations done during load. This is not permitted by PJSIP as it
- will assert if the operation has not been done. This fix changes
- the code so it will only rollback what has been initialized
- already. Further changes also prevent res_pjsip and
- res_pjsip_session from being unloaded. This is due to limitations
- within PJSIP itself. The library environment can only be changed
- to a certain extent and does not provide the ability, currently,
- to deinitialize certain required functionality. (closes issue
- ASTERISK-22474) Reported by: Corey Farrell ........ Merged
- revisions 399624 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-21 04:49 +0000 [r399578-399608] Richard Mudgett <rmudgett@digium.com>
-
- * res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix ref leaks in
- ast_rtcp_read(). Moved rtcp_report RAII_VAR declaration into the
- loop so it is unref'ed after every loop. Moved message_blob to
- loop and switched it to a regular variable. The regular variable
- was used since message_blob is used in a very contained way.
- (closes issue ASTERISK-22565) Reported by: Corey Farrell Patches:
- rtcp_report-leak.patch (license #5909) patch uploaded by Corey
- Farrell Tested by: Corey Farrell ........ Merged revisions 399607
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/media_index.c: media_index: Fix
- process_description_file() memory leak of file_id_persist.
- ........ Merged revisions 399596 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/features_config.c: features_config: Fix config ref leak
- of parkinglots. This leak happend for just about every channel
- created. ........ Merged revisions 399585 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, apps/app_queue.c: app_queue: Fix json blob ref leak. The json
- ref from queue_member_blob_create() was never released. ........
- Merged revisions 399583 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/json.c, /: json: Make it obvious that ast_json_unref() is
- NULL safe. It looked like the safety check was done after the
- NULL pointer was used. ........ Merged revisions 399576 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-20 22:44 +0000 [r399566] Kinsey Moore <kmoore@digium.com>
-
- * main/config_options.c, /: Ensure global types in the config
- framework are initialized If a config object was allocated but
- one of its global objects was never encountered, then the global
- object's defaults were never applied. Ensure that global objects
- are initialized properly upon allocation instead of on
- configuration. Review: https://reviewboard.asterisk.org/r/2866/
- ........ Merged revisions 399564 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399565 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-20 22:06 +0000 [r399554] Jonathan Rose <jrose@digium.com>
-
- * main/dial.c, /: originate/call forwarding: Fix a crash when
- forwarding a call from originate (closes issue ASTERISK-22487)
- Reported by: David M. Lee Review:
- https://reviewboard.asterisk.org/r/2868/ ........ Merged
- revisions 399553 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-20 16:18 +0000 [r399533] Joshua Colp <jcolp@digium.com>
-
- * /, channels/chan_pjsip.c: Add a missing session supplement
- unregistration in chan_pjsip for ACKs. (closes issue
- ASTERISK-22453) Reported by: Corey Farrell Patches:
- chan_pjsip_session_unregister_supplement.patch uploaded by Corey
- Farrell (license 5909) ........ Merged revisions 399531 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-20 14:26 +0000 [r399515] Kevin Harwell <kharwell@digium.com>
-
- * /, main/logger.c: Fix memory leak in logger. Fixed a memory leak
- discovered in the logger where a temporary string buffer was not
- being freed. (closes issue ASTERISK-22540) Reported by: John
- Hardin ........ Merged revisions 399513 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399514 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-19 23:20 +0000 [r399503] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/optional_api.c: optional_api: Make always use the
- standard malloc functions even with MALLOC_DEBUG. ........ Merged
- revisions 399501 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-19 17:01 +0000 [r399459] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_sip.c: chan_sip: Make direct media reinvites for
- T38 put Asterisk in the media path Prior to this patch, Asterisk
- would incorrectly use the previous endpoint addresses in SDP in
- spite of providing its own port. T38 is never meant to be done
- through directmedia and Asterisk should always be in the media
- path for these streams. (closes issue ASTERISK-17273) Reported
- by: Kevin Stewart (closes issue ASTERISK-18706) Reported by:
- Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/
- ........ Merged revisions 399456 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399457 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-18 20:04 +0000 [r399405] Kinsey Moore <kmoore@digium.com>
-
- * /, main/abstract_jb.c: Fix jitter buffer log file creation This
- adjusts '/'-to-'#' replacement to replace all instances of '/'
- instead of just the first to ensure that the jitter buffer log
- file gets the correct name as per Richard Kenner's suggestion.
- (closes issue ASTERISK-21036) Reported by: Richard Kenner
- ........ Merged revisions 399402 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399403 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399404 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-18 17:23 +0000 [r399368-399378] Matthew Jordan <mjordan@digium.com>
-
- * /, build_tools/prep_tarball: Update prep_tarball with new
- documentation files on the Asterisk wiki This will now pull both
- a command reference for the version being prepared, as well as an
- Admin Guide that applies to all versions of Asterisk. (issue
- ASTERISK-22439) Reported by: Olle Johansson ........ Merged
- revisions 399351 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399373 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399376 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when
- a timing module isn't loaded If bridge_softmix fails to be
- created because no timing source is present in Asterisk, this
- will currently fail gracefully but with (most likely) a generic
- error message by whatever module tried to create the softmix
- bridge. This patch adds a more explicit warning so you can
- actually diagnose and fix the problem. Review:
- https://reviewboard.asterisk.org/r/2857/ ........ Merged
- revisions 399353 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399365 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-18 17:15 +0000 [r399352] Richard Mudgett <rmudgett@digium.com>
-
- * main/config_options.c: Make config framework able to reload
- module configs with multiple config files. The config framework
- is supposed to be able to load configs that come from multiple
- config files. The principle example is chan_sip's sip.conf and
- users.conf. Unfortunately, it only does this correctly on initial
- load. This patch causes the module's config to be reloaded
- entirely if any of the config files change. (closes issue
- ASTERISK-22009) Reported by: Richard Mudgett Review:
- https://reviewboard.asterisk.org/r/2859/
-
-2013-09-18 14:56 +0000 [r399340] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_messaging.c, /: res_pjsip_messaging: Register
- message technology as pjsip pjsip's message technology was being
- registered as 'sip', which was causing it to not load due it
- conflicting with chan_sip's registered 'sip' technology for
- messaging. It now registers as 'pjsip'. However, due to this
- change the "to" field for outgoing pjsip messages need to be
- prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to
- res_pjsip_messaging will automatically have their "to" fields
- altered in order to accommodate the change. Outgoing messages
- also handle changing it back to 'sip' before being sent so the
- pjsip library will properly handle it. (closes issue
- ASTERISK-22445) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2833/ ........ Merged
- revisions 399339 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-18 00:13 +0000 [r399295] Michael L. Young <elgueromexicano@gmail.com>
-
- * /, main/features_config.c: Fix Segfault In features-config.c When
- Application Has No Arguments Some applications do not require
- arguments. Therefore, when parsing application maps in
- features.conf, it is possible that app_data will be set to NULL.
- * This patch sets app_data to "" if it is NULL. Review:
- https://reviewboard.asterisk.org/r/2804 ........ Merged revisions
- 399294 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-17 23:10 +0000 [r399284] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_sdp_rtp.c, res/res_pjsip/pjsip_configuration.c,
- res/res_pjsip_t38.c, include/asterisk/res_pjsip.h, /: Change the
- "external_media_address" PJSIP endpoint option to
- "media_address". The endpoint option does not apply to
- communication with external entities. Rather, the option is
- applied to all communications with the endpoint. The
- external_media_address transport configuration option may
- override the endpoint option if it turns out that we are going to
- be communicating with an external entity. Two things of note: 1)
- I have not updated the XML documentation. This is being taken
- care of by Rusty as part of his work on issue ASTERISK-22405 2)
- This commit is likely to cause testsuite failures since there are
- tests that use the external_media_address endpoint option, and
- they will need to be changed over. Well, I'm planning to get that
- updated ASAP after this commit. (closes issue ASTERISK-22528)
- reported by Rusty Newton ........ Merged revisions 399283 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-17 18:44 +0000 [r399269] Kevin Harwell <kharwell@digium.com>
-
- * main/logger.c, main/asterisk.c, /: Remote console: more output
- discrepancies The remote console continued to have issues with
- its output. In this case CLI command output would either not show
- up (if verbose level = 0) or would contain verbose prefixes (if
- verbose level > 0) once log messages were sent to the remote
- console. The fix now now adds verbose prefix data to all new
- lines contained in a verbose log string. (closes issue
- ASTERISK-22450) Reported by: David Brillert (closes issue
- AST-1193) Reported by: Guenther Kelleter Review:
- https://reviewboard.asterisk.org/r/2825/ ........ Merged
- revisions 399267 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399268 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-17 17:55 +0000 [r399258] Richard Mudgett <rmudgett@digium.com>
-
- * /, include/asterisk/features_config.h: Fix doxygen to use correct
- units of features.conf options. ........ Merged revisions 399257
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-17 17:10 +0000 [r399238-399248] Mark Michelson <mmichelson@digium.com>
-
- * main/bridge_basic.c, main/features_config.c, /: Fix other
- timeouts (atxferloopdelay and atxfernoanswertimeout) to use
- seconds instead of milliseconds. Thanks to Richard Mudgett for
- pointing this out. ........ Merged revisions 399247 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/features_config.c, /, include/asterisk/features_config.h,
- main/bridge_basic.c: Switch transferdigittimeout to be configured
- as seconds instead of milliseconds. This was an unintentional
- consequence of the update of features.conf to use the config
- framework in Asterisk 12. Thanks to Marco Signorini on the
- Asterisk developers list for pointing out the problem. ........
- Merged revisions 399237 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-17 14:58 +0000 [r399226] Kevin Harwell <kharwell@digium.com>
-
- * apps/confbridge/conf_state_multi_marked.c, /: Confbridge: empty
- conference not being torn down Confbridge would not properly tear
- down an empty conference bridge when all users were kicked via
- end_marked=yes and at least one user was also set to wait_marked.
- This occurred because while end_marked users were being kicked
- and at least one was also set to wait_marked then the leave
- wait_marked handler would be called on that user, but there would
- be no waiting user (still considered active). The waiting users
- would decrement and now be negative. The conference would remain,
- but be put into an inactive state. The solution was to move from
- the active list to the wait list, those users with wait_marked
- set right before kicking. This allows both the active and wait
- users to decrement correctly and the confbridge to tear down
- properly. A crashed also occurred when trying to list the
- specific conference from the CLI. This happened because the
- conference specified was invalid. Since the conference properly
- tears down now there is no way to reference it thus alleviating
- the crash as well. (closes issue ASTERISK-21859) Reported by:
- Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/
- ........ Merged revisions 399222 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399225 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-16 18:36 +0000 [r399161-399208] Richard Mudgett <rmudgett@digium.com>
-
- * tests/test_ari_model.c, /: Fix module load errors for
- test_ari_model.so. You cannot use a function pointer variable
- with an external function from another dynamically loaded module
- because data variables are always resolved even with RTLD_LAZY. *
- Added wrapper functions for ast_ari_validate_int() and
- ast_ari_validate_string() to use instead for the function pointer
- variable. (closes issue ASTERISK-22457) Reported by: David M. Lee
- ........ Merged revisions 399207 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_speech_utils.c, /, res/res_speech.exports.in:
- app_speech_utils: Fix unresolved symbol ast_speech_get_setting().
- Fixes regression introduced by -r374096. * Made
- res_speech.export.in export ast_* symbols instead of specific
- functions. * Made app_speech_utils.c declare that it is dependent
- upon res_speech. (issue ASTERISK-17136) Reported by: Richard
- Kenner ........ Merged revisions 399197 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry
- time in astdb. When a new IAX2 client registers, the astdb
- database is updated with the value of minregexpire defined in
- iax.conf instead of using the expiry time that is provided by the
- client. The provided expiry time of the client is updated after
- inserting the astdb entry. As a consequence, restarting or
- reloading asterisk creates clients whose registration may expire
- before they reregister. The clients are therefore unavailable
- after minregexpire seconds until they reregister. * Move updating
- of the expiry time to before inserting into the astdb. (closes
- issue ASTERISK-22504) Reported by: Stefan Wachtler Patches:
- chan_iax2.c.patch (license #6533) patch uploaded by Stefan
- Wachtler ........ Merged revisions 399158 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399159 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399160 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-16 02:37 +0000 [r399147] Matthew Jordan <mjordan@digium.com>
-
- * main/cdr.c, /: Filter internal channels out of bridge enter/leave
- message handling Some channels exist merely as an implementation
- detail in Asterisk, such as ConfBridge's announcer/recorder
- channels. These channels should never be exposed to the outside
- world, or to interfaces that report on Asterisk. We already
- filter out such channels in snapshot processing; however, we
- failed to filter out bridge related messages that involved these
- channels. This patch filters out bridge related messages that are
- for such channels. This prevents a spurious WARNING message from
- being displayed when those channels move in and out of bridges.
- ........ Merged revisions 399146 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 22:19 +0000 [r399138] Richard Mudgett <rmudgett@digium.com>
-
- * res/parking/parking_bridge_features.c, apps/app_agent_pool.c,
- include/asterisk/features.h, main/channel.c,
- res/parking/parking_tests.c, include/asterisk/bridge_channel.h,
- main/features.c, tests/test_cel.c, main/bridge_channel.c,
- tests/test_cdr.c, apps/confbridge/conf_chan_announce.c,
- include/asterisk/bridge.h, res/res_pjsip_refer.c, /,
- channels/chan_sip.c, res/stasis/control.c, main/bridge.c,
- main/bridge_basic.c, main/core_unreal.c,
- res/parking/parking_applications.c, main/core_local.c: Restore
- Dial, Queue, and FollowMe 'I' option support. The Dial, Queue,
- and FollowMe applications need to inhibit the bridging initial
- connected line exchange in order to support the 'I' option. *
- Replaced the pass_reference flag on ast_bridge_join() with a
- flags parameter to pass other flags defined by enum
- ast_bridge_join_flags. * Replaced the independent flag on
- ast_bridge_impart() with a flags parameter to pass other flags
- defined by enum ast_bridge_impart_flags. * Since the Dial, Queue,
- and FollowMe applications are now the only callers of
- ast_bridge_call() and ast_bridge_call_with_flags(), changed the
- calling contract to require the initial COLP exchange to already
- have been done by the caller. * Made all callers of
- ast_bridge_impart() check the return value. It is important. As a
- precaution, I also made the compiler complain now if it is not
- checked. * Did some cleanup in parking_tests.c as a result of
- checking the ast_bridge_impart() return value. An independent,
- but associated change is: * Reduce stack usage in
- ast_indicate_data() and add a dropping redundant connected line
- verbose message. (closes issue ASTERISK-22072) Reported by:
- Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/
- ........ Merged revisions 399136 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 20:55 +0000 [r399101] David M. Lee <dlee@digium.com>
-
- * /, main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
- defined. If MALLOC_DEBUG is enabled, then the debug destructor
- for the container is used, which would erroneously write to
- /tmp/refs. This patch only uses the debug destructor if ref_debug
- is used. (closes issue ASTERISK-22536) ........ Merged revisions
- 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 399099 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399100 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 14:50 +0000 [r399082-399084] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip.c, res/res_pjsip_pubsub.c, res/res_pjsip_session.c,
- include/asterisk/res_pjsip.h, res/res_pjsip.exports.in, /: Create
- more accurate Contact headers for dialogs when we are the UAS.
- (closes issue AST-1207) reported by John Bigelow Review:
- https://reviewboard.asterisk.org/r/2842 ........ Merged revisions
- 399083 from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip/config_auth.c, /,
- res/res_pjsip_outbound_authenticator_digest.c,
- res/res_pjsip_authenticator_digest.c: Change how realms are
- handled for outbound authentication. With this change, if no
- realm is specified in an outbound auth section, then we will
- simply match the realm that was present in the 401/407 challenge.
- (closes issue ASTERISK-22471) Reported by George Joseph (closes
- issue ASTERISK-22386) Reported by Rusty Newton Patches:
- outbound_auth_realm_v4.patch uploaded by George Joseph (License
- #6322) ........ Merged revisions 399059 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 14:43 +0000 [r399080-399081] David M. Lee <dlee@digium.com>
-
- * /: Recorded merge of revisions 399035,399049 from
- http://svn.asterisk.org/svn/asterisk/branches/12 These were lost
- in r399071
-
- * /: Put merge tracking for r399039 back.
-
-2013-09-13 14:27 +0000 [r399071] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip_endpoint_identifier_ip.c: Broke the build!
- Forgot para tags within my description.
- https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304
- ........ Merged revisions 399064 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 14:22 +0000 [r399042-399051] David M. Lee <dlee@digium.com>
-
- * res/res_pjsip_log_forwarder.c (added), res/res_pjsip_logger.c,
- res/res_rtp_asterisk.c, /: res_pjsip: Forward PJSIP logging to
- Asterisk logging This patch uses PJSIP's pj_log_set_log_func() to
- forward PJSIP's log messages to Asterisk's logger. This is done
- in a new module: res_pjsip_log_forwarder.so. This patch sets
- defaultenabled on the existing res_pjsip_logger.so to no, since
- logging every SIP packet seems a bit odd to do by default, and is
- (hopefully) less necessary with regular PJSIP logging. It also
- removes res_rtp_asterisk's disabling of PJSIP logging. (closes
- issue ASTERISK-22360) Reported by: Joshua Colp Review:
- https://reviewboard.asterisk.org/r/2830/ ........ Merged
- revisions 399049 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_http_websocket.c: ARI: Fix WebSocket response when
- subprotocol isn't specified When I moved the ARI WebSocket from
- /ws to /ari/events, I added code to allow a WebSocket to connect
- without specifying the subprotocol if there's only one
- subprotocol handler registered for the WebSocket. Naively, I
- coded it to always respond with the subprotocol in use.
- Unfortunately, according to RFC 6455, if the server's response
- includes a subprotocol header field that "indicates the use of a
- subprotocol that was not present in the client's handshake [...],
- the client MUST _Fail the WebSocket Connection_.", emphasis
- theirs. This patch correctly omits the Sec-WebSocket-Protocol if
- one is not specified by the client. (closes issue ASTERISK-22441)
- Review: https://reviewboard.asterisk.org/r/2828/ ........ Merged
- revisions 399039 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 14:17 +0000 [r399036] Kinsey Moore <kmoore@digium.com>
-
- * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This
- change ensures that MeetMeAdmin commands requiring a user
- actually get a user and fixes another issue where an extra
- dereference could occur for a last-entered user being ejected if
- a user identifier was also provided. (closes issue
- ASTERISK-21907) Reported by: Alex Epshteyn Review:
- https://reviewboard.asterisk.org/r/2844/ ........ Merged
- revisions 399033 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 399034 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 399035 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-13 13:28 +0000 [r399032] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip_endpoint_identifier_ip.c: 'identify'
- configObject doesn't have a synopsis Add a straightforward
- synopsis and description to the identify config object in XML
- documentation. (issue ASTERISK-22311) (closes issue
- ASTERISK-22311) Reported By: Rusty Newton ........ Merged
- revisions 399031 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 23:42 +0000 [r399020-399022] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/bridge.c: CLI bridge: Fix "bridge destroy <id>" and
- "bridge kick <id> <chan>" tab completion. These two commands must
- deal with the live bridges container for tab completion and not
- the stasis cache. ........ Merged revisions 399021 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/bridge.c, /: astobj2: Register the bridges container for
- debug inspection. ........ Merged revisions 399019 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 23:23 +0000 [r399018] Rusty Newton <rnewton@digium.com>
-
- * /, res/res_pjsip_acl.c: Documentation fix and improvements to XML
- configuration help res_pjsip_acl * One bug fix. Made the synopsis
- for "type" to accurate. * changing the usage of "IP-domains" to
- "IP addresses" * clarifying the usage for the options, by adding
- a relevant description for each * modified other areas of the XML
- help for clarity, such as the module description and a few
- synopsis changes here and there. See the patch. (issue
- ASTERISK-22458) (closes issue ASTERISK-22458) Reported By: Rusty
- Newton Review: https://reviewboard.asterisk.org/r/2823/ ........
- Merged revisions 399017 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 20:27 +0000 [r399006] Jonathan Rose <jrose@digium.com>
-
- * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
- Revert r398835 due to failing tests involving originate (issue
- ASTERISK-22424) Reported by: Jonathan Rose ........ Merged
- revisions 398977 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398986 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398991 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 16:44 +0000 [r398939] Richard Mudgett <rmudgett@digium.com>
-
- * main/core_unreal.c, /: core_local: Fix memory corruption race
- condition. The masquerade super test is failing on v12 with high
- fence violations and crashing. The fence violations are showing
- that party id allocated memory strings are somehow getting
- corrupted in the bridge_reconfigured_connected_line_update()
- function. The invalid string values happen to be the freed memory
- fill pattern. After much puzzling, I deduced that the
- bridge_reconfigured_connected_line_update() is copying a string
- out of the source channel's caller party id struct just as
- another thread is updating it with a new value. The copying
- thread is using the old string pointer being freed by the
- updating thread. A search of the code found the
- unreal_colp_redirect_indicate() routine updating the caller party
- id's without holding the channel lock. A latent bug in v1.8 and
- v11 hatched in v12 because of the bridging and connected line
- changes. :) (issue ASTERISK-22221) Reported by: Matt Jordan
- Review: https://reviewboard.asterisk.org/r/2839/ ........ Merged
- revisions 398938 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 15:23 +0000 [r398928] David M. Lee <dlee@digium.com>
-
- * res/res_pjsip.c, /: Fix symbol collision with pjsua. We shouldn't
- be exporting any symbols that start with pjsip_. ........ Merged
- revisions 398927 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-12 00:04 +0000 [r398883-398887] Rusty Newton <rnewton@digium.com>
-
- * /, apps/app_queue.c: 'queue add member' help text correction You
- are adding dial strings to the queue, not channels. An aribitrary
- string could be used, but you are typically referencing a
- channel. Correcting the command help text. (issue ASTERISK-22263)
- (closes issue ASTERISK-22263) Reported By: Rusty Newton ........
- Merged revisions 398884 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398885 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398886 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * configs/chan_dahdi.conf.sample, /: Documentation fix -
- waitfordialtone is not boolean, it's time in milliseconds
- Changing text in chan_dahdi.conf sample to be accurate. (issue
- ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
- Malcolm Davenport ........ Merged revisions 398880 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398881 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398882 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-11 20:03 +0000 [r398838] Jonathan Rose <jrose@digium.com>
-
- * /, channels/chan_sip.c, channels/sip/include/sip.h: chan_sip:
- Reject calls without prior SDP on 200 OK If we receive a 200 OK
- without SDP, we will now check to see if the remote address has
- been established for that channel's RTP session and if the to tag
- for that channel has changed from the most recent to tag in a
- response less than 200. If either a change has been made since
- the last to-tag was received or the remote address is unset, then
- we will drop the call. (closes issue ASTERISK-22424) Reported by:
- Jonathan Rose Review:
- https://reviewboard.asterisk.org/r/2827/diff/#index_header
- ........ Merged revisions 398835 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398836 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398837 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-11 18:03 +0000 [r398822] Russell Bryant <russell@russellbryant.com>
-
- * configs/confbridge.conf.sample, /: Fix typo in
- confbridge.conf.sample The denoise filter requires func_speex,
- not codec_speex. Fix this in the description of the denoise=yes
- option in confbridge.conf. ........ Merged revisions 398820 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398821 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-11 14:23 +0000 [r398808] Kevin Harwell <kharwell@digium.com>
-
- * res/res_pjsip_caller_id.c, channels/chan_pjsip.c, /: pjsip:
- reinvite for connected line updates occurs when it should not
- Connected line updates are now only sent out if an actual update
- needs to occur. This happens under the following conditions: 1.
- The endpoint we are sending to is trusted. 2. Either a
- P-Asserted-Identity or Remote Party-ID header needs to be
- added/sent. 3. The connected id's number and name are valid. Also
- added an SDP when an update is sent out. (closes issue AST-1212)
- Reported by: John Bigelow Review:
- https://reviewboard.asterisk.org/r/2831/ ........ Merged
- revisions 398806 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-10 18:05 +0000 [r398760] Richard Mudgett <rmudgett@digium.com>
-
- * main/event.c, res/res_musiconhold.c, main/indications.c,
- main/asterisk.c, main/xmldoc.c, main/cli.c, /,
- funcs/func_dialgroup.c, main/heap.c,
- res/res_pjsip/pjsip_configuration.c: Fix incorrect usages of
- ast_realloc(). There are several locations in the code base where
- this is done: buf = ast_realloc(buf, new_size); This is going to
- leak the original buf contents if the realloc fails. Review:
- https://reviewboard.asterisk.org/r/2832/ ........ Merged
- revisions 398757 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398758 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398759 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-10 17:50 +0000 [r398751-398755] David M. Lee <dlee@digium.com>
-
- * utils/check_expr.c, /: Fixed utils directory breakage from
- r398748, this time with extra hate. ........ Merged revisions
- 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 398753 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398754 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * utils/check_expr.c, /, utils/ael_main.c, utils/conf2ael.c: Fixed
- utils directory breakage from r398648 ........ Merged revisions
- 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 398749 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398750 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-09 23:29 +0000 [r398732] Richard Mudgett <rmudgett@digium.com>
-
- * main/astmm.c, /: MALLOC_DEBUG: Change fence magic number to be
- completely different from the freed magic number. Race conditions
- between freeing a nul terminated string and ast_strdup()'ing it
- are more likely to be detected if the fence and freed magic
- numbers are completely different. ........ Merged revisions
- 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 398721 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398726 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-09 22:00 +0000 [r398695] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_endpoint_identifier_ip.c, /: Add extra debugging to
- res_pjsip_endpoint_identifier_ip ........ Merged revisions 398694
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-09 20:13 +0000 [r398641-398652] David M. Lee <dlee@digium.com>
-
- * /, main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
- DEBUG_THREADS when lock is acquired in __constructor__ This patch
- fixes some long-standing bugs in debug threads that were
- exacerbated with recent Optional API work in Asterisk 12. With
- debug threads enabled, on some systems, there's a lock ordering
- problem between our mutex and glibc's mutex protecting its module
- list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
- thread, the module list will be locked before acquiring our
- mutex. In another thread, our mutex will be locked before locking
- the module list (which happens in the depths of calling
- backtrace()). This patch fixes this issue by moving backtrace()
- calls outside of critical sections that have the mutex acquired.
- The bigger change was to reentrancy tracking for
- ast_cond_{timed,}wait, which wrongly assumed that waiting on the
- mutex was equivalent to a single unlock (it actually suspends all
- recursive locks on the mutex). (closes issue ASTERISK-22455)
- Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged
- revisions 398648 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398649 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398651 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/ari/resource_channels.h, /, rest-api/api-docs/channels.json:
- Multiple revisions 398638-398639 ........ r398638 | dlee |
- 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note
- about expected behavior of originate ........ r398639 | dlee |
- 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note
- about expected behavior of originate (the rest of the commit)
- ........ Merged revisions 398638-398639 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-08 23:30 +0000 [r398629] Matthew Jordan <mjordan@digium.com>
-
- * tests/test_cdr.c, /: Update CDR Unit tests to reflect container
- changes in r398579 When a channel joins a multi-party bridge, the
- ordering of the CDRs that is created is determined by the
- ordering of the channels who happen to be in that bridge. When
- r398579 changed the number of buckets in the container to
- something sensible, it changed the ordering that the CDRs was
- created in, causing one of the multiparty tests to fail. This
- fixes the test with the now expected ordering. ........ Merged
- revisions 398628 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-07 01:03 +0000 [r398603-398620] Kinsey Moore <kmoore@digium.com>
-
- * /, res/res_xmpp.c: Prevent XMPP timeout on blank responses
- Sometimes the Google Voice servers have a bad habit of sending
- out 1 byte replies to the xmpp resource. When a blank 1 byte
- reply is received from the socket the buffer attempts to wait
- (endlessly) for the rest of the reply from google which
- effectively blocks the socket and google voice calls will no
- longer come into the server. This patch allows the xmpp module to
- correctly detect empty packets and send out ping replies to
- google. It also sets a socket timeout on the default socket which
- prevents the xmpp socket from closing and preventing future
- google voice calls from coming into the server. Furthermore
- instead of sending an empty reply back to google we send a proper
- xmpp ping reply back. This also adds several more socket
- messages. (closes issue ASTERISK-22347) Reported by: Andrew Nagy
- Review: https://reviewboard.asterisk.org/r/2771 Patches:
- xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524) ........
- Merged revisions 398618 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398619 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_xmpp.c, res/res_jabber.c: Multiple revisions
- 398558,398577 ........ r398558 | kmoore | 2013-09-06 14:28:16
- -0500 (Fri, 06 Sep 2013) | 17 lines Fix Jabber/XMPP distributed
- MWI The mailbox and context are swapped on the receiving end for
- all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and
- all more recent versions. This swaps those values to be correct
- when publishing to the internal event system from Jabber/XMPP
- distributed MWI state. (closes issue ASTERISK-22435) Reported by:
- abelbeck Tested by: Michael Keuter Patches:
- asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
- abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
- uploaded by abelbeck ........ Merged revisions 398523 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) |
- 10 lines Commit the remainder of r398523 This is a missing part
- of the commit in revision 398523 that corrects the name of a
- variable. (issue ASTERISK-22435) ........ Merged revisions 398576
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 398558,398577 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398580 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-06 21:17 +0000 [r398557-398583] Richard Mudgett <rmudgett@digium.com>
-
- * main/cdr.c, /: cdr: Change the number of container buckets to be
- similar to the channels container. * Fix the temporary cdr
- candidate containers to use a prime number of buckets. ........
- Merged revisions 398579 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/core_local.c, /: core_local: Fix LocalOptimizationBegin AMI
- event missing Source channel snapshot. * Fix the
- LocalOptimizationBegin AMI event by eliminating an artificial
- buffer size limitation that is too small anyway. ........ Merged
- revisions 398572 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: cdr: Fix some ref leaks. * Added missing
- unregister of the cdr container in cdr_engine_shutdown(). * Fixed
- ref leak in off nominal path of cdr_object_alloc(). * Removed
- some unnecessary NULL checks in cdr_object_dtor(). ........
- Merged revisions 398562 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/astobj2.h, main/cel.c, main/features_config.c,
- apps/app_agent_pool.c, main/cdr.c, main/udptl.c, /,
- main/parking.c, main/stasis_config.c: astobj2: Add warn unused
- attribute to some functions. * Fixed resulting warnings with
- improper use of ao2_global_obj_replace(). * Made a couple uses of
- ao2_global_obj_replace_unref(x, NULL) into the equivalent and
- more appropriate ao2_global_obj_release() call. ........ Merged
- revisions 398533 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-06 18:53 +0000 [r398512-398522] Kinsey Moore <kmoore@digium.com>
-
- * main/http.c, /, res/stasis/app.c: Fix build warnings When
- AST_DEVMODE is not defined, ast_asserts are not compiled into the
- binary. In some cases, this means variables are not referenced or
- are set but unused which causes warnings to show up. (closes
- issue ASTERISK-22446) Reported by: Jason Parker (qwell) ........
- Merged revisions 398521 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_h323.c: Fix chan_h323 compilation This fixes the
- things in chan_h323 that were missed or ignored in the great
- channel opaquification and gets chan_h323 back into a compiling
- state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov
- Patches: chan_h323.patch uploaded by Dmitry Melekhov ........
- Merged revisions 398510 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398511 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-05 21:48 +0000 [r398384-398499] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/astobj2.c: astobj2: Only define ao2_bt() once. * Make
- ao2_bt() not use single char variable names. * Fix ao2_bt()
- formatting. ........ Merged revisions 398498 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in
- __attempt_transmit(). * Reduce indentation in
- __attempt_transmit(). * Don't update the static last error time
- variable every time in __schedule_action() and socket_read().
- ........ Merged revisions 398456 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398457 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398458 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker
- thread idle_list. * Fix stray reference to idle_list in
- cleanup_thread_list(). This may be the reason for the note in
- iax2_process_thread() about threads not being removed from the
- task lists. * Move cleanup_thread_list(&idle_list) to after the
- other lists are cleaned up. ........ Merged revisions 398416 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398417 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398418 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock
- avoidance. * Fix bridgecallno deadlock avoidance. When doing
- deadlock avoidance, you need to retest the status of values for
- each loop to see if you still need the lock for bridgecallno. *
- As a safety check, after acquiring the bridgecallno lock you
- should check if iaxs[bridgecallno] is NULL just like the current
- callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
- to after processing any deferred frames to ensure that the
- iostate is IDLE when it is placed back into the idle list.
- defer_full_frame() tries to ensure iax2_process_thread() wakes up
- to process the frame. ........ Merged revisions 398379 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398380 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398381 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-05 14:10 +0000 [r398369] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c: Clarify server_uri and
- client_uri registration settings. Used some of Rusty's suggested
- language plus also included more SIPesque descriptions of where
- the URIs are actually used in an outgoing REGISTER. (closes issue
- ASTERISK-22390) reported by Rusty Newton ........ Merged
- revisions 398368 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-04 23:07 +0000 [r398304] Richard Mudgett <rmudgett@digium.com>
-
- * channels/iax2/parser.c, /: chan_iax2: Add missing control frame
- names to debug frame decode output. ........ Merged revisions
- 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
- ........ Merged revisions 398302 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398303 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-04 22:49 +0000 [r398300] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_outbound_authenticator_digest.c: Give more
- detail regarding failures to create request with auth
- credentials. (issue ASTERISK-22386) ........ Merged revisions
- 398299 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-04 21:37 +0000 [r398284-398287] Jonathan Rose <jrose@digium.com>
-
- * /, tests/test_voicemail_api.c: unit tests: test_voicemail_api
- leaks stringfields from snapshots (closes issue ASTERISK-22414)
- Reported by: Corey Farrell Patches:
- test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
- (license 5909) ........ Merged revisions 398285 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398286 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * apps/app_voicemail.c, /: app_voicemail: Fix leaking config
- objects when msg_id doesn't match (issues ASTERISK-22414)
- Reported by: Corey Farrell Patch:
- test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
- (license 5909) ........ Merged revisions 398281 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398283 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-04 16:03 +0000 [r398238] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output
- printed with arbitrary verbose levels. Fix the misdn debug output
- to remote consoles. chan_misdn uses ast_console_puts() which
- doesn't know about verbose levels. Better to use ast_verbose()
- instead. Without this patch the misdn debug messages are appended
- to the verbose level which ever was set by the message sent to
- the console before, i.e. any undefined level. (closes issue
- AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch
- (license #6372) patch uploaded by Guenther Kelleter ........
- Merged revisions 398235 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398236 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398237 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-04 14:32 +0000 [r398227] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c: Debug messages for
- pjsip outbound registration Added debug messages indicating that
- an outbound registration attempt was made and it was successful
- in pjsip. (closes issue ASTERISK-22388) Reported by: Rusty Newton
- ........ Merged revisions 398226 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-03 20:28 +0000 [r398217] Alexandr Anikin <may@telecom-service.ru>
-
- * /, addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling
- on empty tcs received ........ Merged revisions 398214 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398215 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-03 18:09 +0000 [r398207] Kinsey Moore <kmoore@digium.com>
-
- * res/res_pjsip_dtmf_info.c, /: Prevent a crash in
- res_pjsip_dtmf_info.c This change makes sure that a content type
- header exists before checking the contents of the header against
- known SIP INFO DTMF content types. ........ Merged revisions
- 398206 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-03 17:19 +0000 [r398205] David M. Lee <dlee@digium.com>
-
- * Makefile, /: Fixed 'make clean' for wiki docs ........ Merged
- revisions 398198 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-09-03 14:29 +0000 [r398197] Walter Doekes <walter+asterisk@wjd.nu>
-
- * /, cel/cel_custom.c: Be a little more verbose when loading
- cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
- ........ Merged revisions 398167 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398168 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398196 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 20:58 +0000 [r398150] David M. Lee <dlee@digium.com>
-
- * main/asterisk.c, include/asterisk/optional_api.h, /,
- main/optional_api.c: Fix graceful shutdown crash. The cleanup
- code for optional_api needs to happen after all of the optional
- API users and providers have unused/unprovided. Unfortunately,
- regsitering the atexit() handler at the beginning of main() isn't
- soon enough, since module destructors run after that. ........
- Merged revisions 398149 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 20:37 +0000 [r398148] Rusty Newton <rnewton@digium.com>
-
- * /, configs/pjsip.conf.sample: New pjsip.conf.sample (issue
- ASTERISK-22145) (closes issue ASTERISK-22145) Reported By: Matt
- Jordan Review: https://reviewboard.asterisk.org/r/2811/ ........
- Merged revisions 398147 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 19:55 +0000 [r398124-398140] Kevin Harwell <kharwell@digium.com>
-
- * /, res/res_pjsip_outbound_registration.c,
- include/asterisk/sorcery.h, res/res_pjsip.c,
- res/res_pjsip/config_transport.c, main/sorcery.c: Add a
- reloadable option for sorcery type objects Some configuration
- objects currently won't place nice if reloaded. Specifically, in
- this case the pjsip transport objects. Now when registering an
- object in sorcery one may specify that the object is allowed to
- be reloaded or not. If the object is set to not reload then upon
- reloading of the configuration the objects of that type will not
- be reloaded. The initially loaded objects of that type however
- will remain. While the transport objects will not longer be
- reloaded it is still possible for a user to configure an endpoint
- to an invalid transport. A couple of log messages were added to
- help diagnose this problem if it occurs. (closes issue
- ASTERISK-22382) Reported by: Rusty Newton (closes issue
- ASTERISK-22384) Reported by: Rusty Newton Review:
- https://reviewboard.asterisk.org/r/2807/ ........ Merged
- revisions 398139 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/config.c, res/res_security_log.c, /, channels/chan_sip.c,
- main/translate.c, main/named_acl.c, main/indications.c: Fix
- various memory leaks main/config.c - cleanup cache fie includes
- res/res_security_log.c - unregister logger level
- channesl/chan_sip.c - cleanup io context and notify_types
- main/translator.c - cleanup at shutdown main/named_acl.c -
- cleanup cli commands main/indications.c -
- ast_get_indication_tone() unref default_tone_zone if used (closes
- issues ASTERISK-22378) Reported by: Corey Farrell Patches:
- config_shutdown.patch uploaded by coreyfarrell (license 5909)
- res_security_log.patch uploaded by coreyfarrell (license 5909)
- chan_sip-11.patch uploaded by coreyfarrell (license 5909)
- indications_refleak.patch uploaded by coreyfarrell (license 5909)
- named_acl-cli_unreg-trunk.patch uploaded by coreyfarrell (license
- 5909) translate_shutdown.patch uploaded by coreyfarrell (license
- 5909) ........ Merged revisions 398102 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398103 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398116 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 18:38 +0000 [r398101] Matthew Jordan <mjordan@digium.com>
-
- * /, UPGRADE-12.txt (added), UPGRADE.txt: Update UPGRADE.txt file
- for Asterisk 12 This simply pulls in the changes that were
- breaking from the CHANGES file and updates a few other areas
- accordingly. It also removes the 10 => 11 notes, which are
- traditionally removed from each major version and stored in the
- appropriate UPGRADE-X.txt file. ........ Merged revisions 398100
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 18:30 +0000 [r398064-398099] Jonathan Rose <jrose@digium.com>
-
- * main/features_config.c, /, main/config_options.c:
- features_config: Ignore parkinglots in features.conf instead of
- failing to load Parkinglots are defined in res_features.conf now,
- but this patch fixes features_config so that features don't fail
- to load when parkinglots are present in features.conf Review:
- https://reviewboard.asterisk.org/r/2801/ ........ Merged
- revisions 398068 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/features_config.c, main/udptl.c, /: features_config: Don't
- require features.conf to be present for Asterisk to load (closes
- issue ASTERISK-22426) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2806/ ........ Merged
- revisions 398020 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 17:59 +0000 [r398063] Kevin Harwell <kharwell@digium.com>
-
- * main/manager.c, /, res/res_agi.c: Memory leak fix
- ast_xmldoc_printable returns an allocated block that must be
- freed by the caller. Fixed manager.c and res_agi.c to stop
- leaking these results. (closes issue ASTERISK-22395) Reported by:
- Corey Farrell Patches: manager-leaks-12.patch uploaded by
- coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
- by coreyfarrell (license 5909) ........ Merged revisions 398060
- from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
- Merged revisions 398061 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398062 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 17:11 +0000 [r398024-398026] Richard Mudgett <rmudgett@digium.com>
-
- * tests/test_substitution.c, /: test_substitution: Fix failing
- test. Revert the -r392190 change. The original test was correct.
- The CDR code was actually returning an unititialized buffer.
- ........ Merged revisions 398025 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * tests/test_substitution.c, /: test_substituition: Fix failed test
- reporting to actually report failure. You cannot put the "Testing
- <blah> pass/fail" on a single line before actually performing the
- test. Now any additional failure information is logged before the
- test pass/fail announcement. * Added an additional CDR(answer,u)
- test. ........ Merged revisions 398018 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 398019 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398023 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 16:27 +0000 [r398003-398017] Kevin Harwell <kharwell@digium.com>
-
- * /, apps/app_mixmonitor.c: Fix memory leaks (closes issue
- ASTERISK-22368) Reported by: Corey Farrell Patches:
- issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes
- (license 5674) ........ Merged revisions 398004 from
- http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
- revisions 398011 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398016 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/asterisk.c, /: Check return value on fwrite ........ Merged
- revisions 398000 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 398002 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 13:40 +0000 [r397987-397990] David M. Lee <dlee@digium.com>
-
- * rest-api-templates/swagger_model.py, res/ari/ari_websockets.c,
- channels/sip/include/sip.h, main/asterisk.c, res/res_ari.c,
- tests/test_optional_api.c (added), /, channels/chan_sip.c,
- include/asterisk/autoconfig.h.in, configure.ac,
- rest-api-templates/res_ari_resource.c.mustache,
- res/ari/internal.h, res/res_http_websocket.c, CHANGES,
- include/asterisk/compiler.h, include/asterisk/ari.h,
- main/loader.c, include/asterisk/optional_api.h,
- build_tools/cflags.xml, configure, res/res_ari_events.c,
- include/asterisk/http_websocket.h, main/optional_api.c (added):
- optional_api: Fix linking problems between modules that export
- global symbols With the new work in Asterisk 12, there are some
- uses of the optional_api that are prone to failure. The details
- are rather involved, and captured on [the wiki][1]. This patch
- addresses the issue by removing almost all of the magic from the
- optional API implementation. Instead of relying on weak symbol
- resolution, a new optional_api.c module was added to Asterisk
- core. For modules providing an optional API, the pointer to the
- implementation function is registered with the core. For modules
- that use an optional API, a pointer to a stub function, along
- with a optional_ref function pointer are registered with the
- core. The optional_ref function pointers is set to the
- implementation function when it's provided, or the stub function
- when it's now. Since the implementation no longer relies on
- magic, it is now supported on all platforms. In the spirit of
- choice, an OPTIONAL_API flag was added, so we can disable the
- optional_api if needed (maybe it's buggy on some bizarre platform
- I haven't tested on) The AST_OPTIONAL_API*() macros themselves
- remained unchanged, so existing code could remain unchanged. But
- to help with debugging the optional_api, the patch limits the
- #include of optional API's to just the modules using the API.
- This also reduces resource waste maintaining optional_ref
- pointers that aren't used. Other changes made as a part of this
- patch: * The stubs for http_websocket that wrap system calls set
- errno to ENOSYS. * res_http_websocket now properly increments
- module use count. * In loader.c, the while() wrappers around
- dlclose() were removed. The while(!dlclose()) is actually an
- anti-pattern, which can lead to infinite loops if the module
- you're attempting to unload exports a symbol that was directly
- linked to. * The special handling of nonoptreq on systems without
- weak symbol support was removed, since we no longer rely on weak
- symbols for optional_api. [1]:
- https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue
- ASTERISK-22296) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2797/ ........ Merged
- revisions 397989 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_stasis_playback.c, /,
- include/asterisk/stasis_app_recording.h,
- res/ari/resource_recordings.h, res/res_stasis_recording.c,
- res/Makefile, res/ari/ari_model_validators.c,
- rest-api/api-docs/recordings.json, res/stasis_recording (added),
- res/ari/resource_recordings.c, res/ari/ari_model_validators.h,
- res/res_ari_recordings.c: ARI: Implement /recordings/stored API's
- his patch implements the ARI API's for stored recordings. While
- the original task only specified deleting a recording, it was
- simple enough to implement the GET for all recordings, and for an
- individual recording. The recording playback operation was
- modified to use the same code for accessing the recording as the
- REST API, so that they will behave consistently. There were
- several problems with the api-docs that were also fixed, bringing
- the ARI spec in line with the implementation. There were some
- 'wishful thinking' fields on the stored recording model (duration
- and timestamp) that were removed, because I ended up not
- implementing a metadata file to go along with the recording to
- store such information. The GET /recordings/live operation was
- removed, since it's not really that useful to get a list of all
- recordings that are currently going on in the system. (At least,
- if we did that, we'd probably want to also list all of the
- current playbacks. Which seems weird.) (closes issue
- ASTERISK-21582) Review: https://reviewboard.asterisk.org/r/2693/
- ........ Merged revisions 397985 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /: Multiple revisions 397975-397976 ........ r397975 | rmudgett |
- 2013-08-29 20:00:00 -0500 (Thu, 29 Aug 2013) | 1 line pbx.c: Make
- ast_str_substitute_variables_full() not mask variables. ........
- r397976 | rmudgett | 2013-08-29 20:00:41 -0500 (Thu, 29 Aug 2013)
- | 1 line Revert last commit. ........ Merged revisions
- 397975-397976 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 01:20 +0000 [r397978] Richard Mudgett <rmudgett@digium.com>
-
- * main/pbx.c, /: pbx.c: Make pbx_substitute_variables_helper_full()
- not mask variables. ........ Merged revisions 397977 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-30 00:11 +0000 [r397962-397969] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip_pidf.c, /: Sanitize XML output for PIDF bodies.
- PJSIP's PIDF API does not replace angle brackets with their
- appropriate counterparts for XML. So we have to do it ourself. In
- this particular case, the problem had to do with attempting to
- place an unsanitized SIP URI into an XML node. Now we don't get a
- 488 from recipients of our PIDF NOTIFYs. ........ Merged
- revisions 397968 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_pidf.c, /: Fix method for creating activities
- string in PIDF bodies. The previous method did not allocate
- enough space to create the entire string, but adjusted the
- string's slen value to be larger than the actual allocation. This
- resulted in garbled text in NOTIFY requests from Asterisk. This
- method allocates the proper amount of space first and then writes
- the content into the buffer. ........ Merged revisions 397960
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 22:49 +0000 [r397959] Kevin Harwell <kharwell@digium.com>
-
- * apps/app_dumpchan.c, main/logger.c, apps/app_verbose.c,
- main/asterisk.c, channels/chan_misdn.c, /: Verbose logging
- discrepancies Refactored cases where a combination of
- ast_verbose/options_verbose were present. Also in general tried
- to eliminate, in as many places as possible, where the
- options_verbose global variable was being used. Refactored the
- way local and remote consoles handle verbose message logging in
- an attempt to solve the various discrepancies that sometimes
- would show between the two. (closes issue AST-1193) Reported by:
- Guenther Kelleter Review:
- https://reviewboard.asterisk.org/r/2798/ ........ Merged
- revisions 397948 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 397958 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 22:26 +0000 [r397956-397957] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_pubsub.c: Fix when the subscription_terminated
- callback is called for subscription handlers. The previous
- placement would result in the resubscribe() callback called
- instead of the subscription_terminated() callback being called
- when a subscription was ended via a SUBSCRIBE request. This would
- result in confusing PJSIP and having it throw an assertion.
- ........ Merged revisions 397955 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * res/res_pjsip_session.c, /: Fix a race condition where a canceled
- call was answered. RFC 5407 section 3.1.2 details a scenario
- where a UAC sends a CANCEL at the same time that a UAS sends a
- 200 OK for the INVITE that the UAC is canceling. When this
- occurs, it is the role of the UAC to immediately send a BYE to
- terminate the call. This scenario was reproducible by have a
- Digium phone with two lines place a call to a second phone that
- forwarded the call to the second line on the original phone. The
- Digium phone, upon realizing that it was connecting to itself,
- would attempt to cancel the call. The timing of this happened to
- trigger the aforementioned race condition about 80% of the time.
- Asterisk was not doing its job of sending a BYE when receiving a
- 200 OK on a cancelled INVITE. The result was that the ast_channel
- structure was destroyed but the underlying SIP session, as well
- as the PJSIP inv_session and dialog, were still alive. Attempting
- to perform an action such as a transfer, once in this state,
- would result in Asterisk crashing. The circumstances are now
- detected properly and the session is ended as recommended in RFC
- 5407. (closes issue AST-1209) reported by John Bigelow ........
- Merged revisions 397945 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 21:37 +0000 [r397947] Kevin Harwell <kharwell@digium.com>
-
- * main/file.c, main/app.c, main/config_options.c, main/cel.c,
- main/asterisk.c, main/cdr.c, main/manager.c, /,
- main/stasis_config.c: Memory leaks fix (closes ASTERISK-22376)
- Reported by: John Hardin Patches: memleak.patch uploaded by
- jhardin (license 6512) memleak2.patch uploaded by jhardin
- (license 6512) ........ Merged revisions 397946 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 20:22 +0000 [r397939] Matthew Jordan <mjordan@digium.com>
-
- * configs/safe_asterisk.conf.sample (removed), /, CHANGES,
- contrib/scripts/safe_asterisk, Makefile: Revert r394939 due to
- (numerous) objections The patch from ASTERISK-21965 was committed
- perhaps a bit too hastily. Walter and Tzafrir have pointed out
- numerous issues with the approach and have propsed an alternative
- in r/2757. Since it's not a time critical issue and is not worth
- holding up the release of 12 for it, I've gone ahead and reverted
- r394939 from 12/trunk and re-opened ASTERISK-21965. ........
- Merged revisions 397938 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 16:21 +0000 [r397932] David M. Lee <dlee@digium.com>
-
- * rest-api-templates/make_ari_stubs.py, /,
- rest-api-templates/api.wiki.mustache,
- rest-api-templates/asterisk_processor.py: Account for {} in
- Swagger notes ........ Merged revisions 397927 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 16:05 +0000 [r397925] Matthew Jordan <mjordan@digium.com>
-
- * Makefile, /: Recursively search for '.c' files when making
- documentation with 'make full' Without this, documentation
- defined in sub-folders is ignored. Since having properly
- generated documentation is especially important in Asterisk 12 -
- not having it can cause a module to not load - 'make full' needs
- to look in all .c files. ........ Merged revisions 397924 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 15:43 +0000 [r397923] Mark Michelson <mmichelson@digium.com>
-
- * /, apps/app_queue.c, main/cel.c, main/stasis_bridges.c: Multiple
- revisions 397921-397922 ........ r397921 | mmichelson |
- 2013-08-29 10:42:10 -0500 (Thu, 29 Aug 2013) | 6 lines Resolve
- assumptions that bridge snapshots would be non-NULL for transfer
- stasis events. Attempting to transfer an unbridged call would
- result in crashes in either CEL code or in the conversion to AMI
- messages. ........ r397922 | mmichelson | 2013-08-29 10:42:29
- -0500 (Thu, 29 Aug 2013) | 3 lines Remove extra debug message.
- ........ Merged revisions 397921-397922 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-29 12:30 +0000 [r397912] Matthew Jordan <mjordan@digium.com>
-
- * contrib/ast-db-manage/config,
- contrib/ast-db-manage/config/script.py.mako,
- contrib/ast-db-manage/voicemail.ini.sample,
- contrib/ast-db-manage/voicemail/env.py,
- contrib/ast-db-manage/voicemail,
- contrib/ast-db-manage/voicemail/script.py.mako,
- contrib/ast-db-manage/README.md,
- contrib/ast-db-manage/config/versions,
- contrib/ast-db-manage/voicemail/versions/a2e9769475e_create_tables.py,
- contrib/ast-db-manage (added),
- contrib/ast-db-manage/voicemail/versions, /,
- contrib/ast-db-manage/config.ini.sample,
- contrib/ast-db-manage/config/env.py,
- contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py:
- Actually *add* the database schema management utilities In
- r397874, the scripts were removed... but not replaced. Thanks to
- Michael Young for noticing this! ........ Merged revisions 397911
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-28 23:15 +0000 [r397886-397903] Richard Mudgett <rmudgett@digium.com>
-
- * main/cdr.c, /, funcs/func_cdr.c, main/stdtime/localtime.c: Fix
- some uninitialized buffers for CDR handling valgrind found. *
- Made ast_strftime_locale() ensure that the output buffer is
- initialized. The std library strftime() returns 0 and does not
- touch the buffer if it has an error. However, the function can
- also return 0 without an error. (closes issue ASTERISK-22412)
- Reported by: rmudgett ........ Merged revisions 397902 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: Fixed problems with ast_cdr_serialize_variables().
- * Fixed return value of ast_cdr_serialize_variables() on error.
- It needs to return 0 indicating no CDR variables found. * Made
- ast_cdr_serialize_variables() check the return value of
- cdr_object_format_property() and assert if nonzero. A member of
- the cdr_readonly_vars[] was not handled. * Removed unused
- elements from cdr_readonly_vars[]: total_duration, total_billsec,
- first_start, and first_answer. ........ Merged revisions 397900
- from http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: Made the on/off in CLI "cdr set debug [on|off]"
- case insensitive. ........ Merged revisions 397898 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/cdr.c, /: Make CDR variable name chandling consistently case
- insensitive. ........ Merged revisions 397896 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, main/cdr.c: Make CDR code deal with channel names case
- insensitively. ........ Merged revisions 397894 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, funcs/func_cdr.c, main/cdr.c: Some CDR code optimization.
- ........ Merged revisions 397892 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, funcs/func_cdr.c: Whitespace and curly braces. ........ Merged
- revisions 397885 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-28 21:09 +0000 [r397877] Mark Michelson <mmichelson@digium.com>
-
- * /, res/res_pjsip_refer.c: Improve detection of answer on SIP
- blind transfer. A problem encountered during testing was that
- res_pjsip_refer would not ever send a NOTIFY with a 200 OK
- sipfrag. This is because the framehook that was supposed to send
- the NOTIFY would never be told that an answer had occurred. This
- happened for two reasons: 1) The transferee channel on which the
- framehook was on was already up. 2) Answers are rarely if ever
- written to channels. Rather, the ast_answer() or ast_raw_answer()
- function is used to answer channels. Thanks to a suggestion by
- Matt Jordan, the best way to detect that the call had been
- answered was to find out when the transferee channel joined a
- bridge. With stasis this is an easy task. So now, in addition to
- the framehook logic, there is a stasis subscription used to
- determine when the transferee has entered a bridge. Once it has
- entered, an appropriate NOTIFY is sent. ........ Merged revisions
- 397876 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-28 20:55 +0000 [r397872-397875] Matthew Jordan <mjordan@digium.com>
-
- * contrib/realtime/mysql/queue_log.sql,
- contrib/realtime/mysql/voicemail.sql,
- contrib/realtime/mysql/sippeers.sql, /,
- contrib/realtime/mysql/iaxfriends.sql,
- contrib/realtime/mysql/meetme.sql,
- contrib/realtime/mysql/voicemail_messages.sql,
- contrib/realtime/postgresql/realtime.sql,
- contrib/realtime/mysql/voicemail_data.sql, CHANGES,
- contrib/realtime/mysql/musiconhold.sql: Add database schema
- management using Alembic This patch replaces contrib/realtime/
- with a new setup for managing the database schema required for
- database integration with Asterisk. In addition to initializing a
- database with the proper schema, alembic can do a database
- migration to assist with upgrading Asterisk in the future.
- Hopefully this helps make setting up and operating Asterisk with
- a database easier. With this the schema only needs to be
- maintained in one place instead of once per database. The schemas
- I have added here have a bit of improvement over the examples
- that were there before (some added consistency and added some
- missing indexes). Managing the schema in one place here also
- applies to all databases supported by SQLAlchemy. See
- contrib/ast-db-manage/README.md for more details. Review:
- https://reviewboard.asterisk.org/r/2731 patch by Russell Bryant
- (license 6300) ........ Merged revisions 397874 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * CHANGES, /: Update CHANGES file for Asterisk 12 This updates the
- Asterisk 12 CHANGES file with the things that were missed during
- the development cycle. Review:
- https://reviewboard.asterisk.org/r/2795/ ........ Merged
- revisions 397870 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-28 16:13 +0000 [r397857-397860] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/pbx.c: pbx.c: Make ast_str_substitute_variables_full()
- not mask variables. ........ Merged revisions 397859 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * main/chanvars.c: ast_free() is null tollerant.
-
- * include/asterisk/threadstorage.h, /: Match use of ast_free() with
- ast_calloc() and add some curly braces. ........ Merged revisions
- 397856 from http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-28 15:43 +0000 [r397855] Mark Michelson <mmichelson@digium.com>
-
- * res/res_pjsip/pjsip_distributor.c, /: Fix dialog matching in the
- SIP distributor. Dialog matching is performed in the distributor
- for the sole purpose of retrieving an associated serializer so
- the request may be serialized. This patch fixes two problems.
- First, incoming CANCEL requests that had no to-tag (which really
- should be *all* CANCEL requests) would not match with a dialog.
- An earlier bug fix to deal with early CANCEL requests would
- result in the CANCEL being replied to with a 481. The fix for
- this is to find the matching INVITE transaction and get the
- dialog from that transaction. Second, no SIP responses were
- matching dialogs. This is because we were inverting the tags that
- we were passing into PJSIP's dialog finding function. This logic
- has been corrected by setting local and remote tag variables
- based on whether the incoming message is a request or response.
- ........ Merged revisions 397854 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-27 19:19 +0000 [r397820] David M. Lee <dlee@digium.com>
-
- * rest-api-templates/param_parsing.mustache, res/res_ari_bridges.c,
- /, res/stasis/app.c, res/res_ari_events.c,
- res/res_ari_asterisk.c,
- rest-api-templates/res_ari_resource.c.mustache, res/stasis/app.h,
- res/res_stasis.c, main/stasis_bridges.c: ARI: WebSocket event
- cleanup Stasis events (which get distributed over the ARI
- WebSocket) are created by subscribing to the channel_all_cached
- and bridge_all_cached topics, filtering out events for
- channels/bridges currently subscribed to. There are two issues
- with that. First was a race condition, where messages in-flight
- to the master subscribe-to-all-things topic would get sent out,
- even though the events happened before the channel was put into
- Stasis. Secondly, as the number of channels and bridges grow in
- the system, the work spent filtering messages becomes excessive.
- Since r395954, individual channels and bridges have caching
- topics, and can be subscribed to individually. This patch takes
- advantage, so that channels and bridges are subscribed to on
- demand, instead of filtering the global topics. The one case
- where filtering is still required is handling BridgeMerge
- messages, which are published directly to the bridge_all topic.
- Other than the change to how subscriptions work, this patch
- mostly just moves code around. Most of the work generating JSON
- objects from messages was moved to .to_json handlers on the
- message types. The callback functions handling app subscriptions
- were moved from res_stasis (b/c they were global to the model) to
- stasis/app.c (b/c they are local to the app now). (closes issue
- ASTERISK-21969) Reported by: Matt Jordan Review:
- https://reviewboard.asterisk.org/r/2754/ ........ Merged
- revisions 397816 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-27 18:52 +0000 [r397811] Richard Mudgett <rmudgett@digium.com>
-
- * /, main/astmm.c: Made MALLOC_DEBUG less CPU intensive by default.
- Storing a backtrace for each allocation in anticipation of a
- memory management problem is very CPU intensive. * Added the CLI
- "memory backtrace {on|off}" command to request that the backtrace
- be gathered only on request. The backtrace is off by default.
- (issue ASTERISK-22221) Reported by: Matt Jordan ........ Merged
- revisions 397809 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-27 18:10 +0000 [r397753-397760] Matthew Jordan <mjordan@digium.com>
-
- * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
- SDP If the SIP channel driver processes an invalid SDP that
- defines media descriptions before connection information, it may
- attempt to reference the socket address information even though
- that information has not yet been set. This will cause a crash.
- This patch adds checks when handling the various media
- descriptions that ensures the media descriptions are handled only
- if we have connection information suitable for that media. Thanks
- to Walter Doekes, OSSO B.V., for reporting, testing, and
- providing the solution to this problem. (closes issue
- ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
- issueA22007_sdp_without_c_death.patch uploaded by wdoekes
- (License 5674) ........ Merged revisions 397756 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 397757 from
- http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
- revisions 397758 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 397759 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK
- on dialog that has no channel A remote exploitable crash
- vulnerability exists in the SIP channel driver if an ACK with SDP
- is received after the channel has been terminated. The handling
- code incorrectly assumed that the channel would always be
- present. This patch adds a check such that the SDP will only be
- parsed and applied if Asterisk has a channel present that is
- associated with the dialog. Note that the patch being applied was
- modified only slightly from the patch provided by Walter Doekes
- of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
- Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
- issueA21064_fix.patch uploaded by wdoekes (License 5674) ........
- Merged revisions 397710 from
- http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
- revisions 397711 from
- http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged
- revisions 397712 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 397713 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-27 16:51 +0000 [r397746] Richard Mudgett <rmudgett@digium.com>
-
- * channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
- channels/chan_dahdi.c, channels/sig_analog.c, /,
- channels/chan_sip.c, channels/chan_motif.c: Fix uninitialized
- value in struct ast_control_pvt_cause_code usage. ........ Merged
- revisions 397744 from
- http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
- revisions 397745 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-26 23:48 +0000 [r397691] Matthew Jordan <mjordan@digium.com>
-
- * /, main/bridge_channel.c: Better handle clearing the OUTGOING
- flag when a channel leaves a bridge When a channel with the
- OUTGOING flag leaves a bridge, and it will survive being pulled
- from the bridge (either because it will execute dialplan, go into
- another bridge, or live in a friendly autoloop), we have to clear
- the OUTGOING flag. This is the signal to the CDR engine that this
- channel is no longer a second class citizen, i.e., it is not
- "dialed". The soft hangup flags are only half the picture. If a
- channel is being moved from one bridge to another, the soft
- hangup flags aren't set; however, the state of the bridge_channel
- will not be hung up. Since the channel does not have one of the
- two hang up states, that implies that the channel is still
- technically alive. This patch modifies the check so that it
- checks both the soft hangup flags as well as the bridge_channel
- state. If either suggests that the channel is going to persist,
- we clear the OUTGOING flag. ........ Merged revisions 397690 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-26 21:32 +0000 [r397674] David M. Lee <dlee@digium.com>
-
- * /, main/bucket.c: Fixed bucket.c for systems where tv_usec is not
- an unsigned long. ........ Merged revisions 397673 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-26 16:25 +0000 [r397644-397651] Richard Mudgett <rmudgett@digium.com>
-
- * /, include/asterisk/bridge_channel.h, main/bridge_channel.c:
- bridging: Fix a livelock with local channel optimization. Use a
- better means of waking up the bridge channel thread. ........
- Merged revisions 397650 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * channels/Makefile, /: chan_dahdi: Add some missing build cleanup.
- ........ Merged revisions 397643 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-25 18:12 +0000 [r397622-397631] Matthew Jordan <mjordan@digium.com>
-
- * tests/test_bucket.c, /: Fix bucket unit tests After the review
- for buckets was completed (r2715), the handling of names in the
- bucket core was deferred to the wizards. As such, the bucket unit
- tests cannot expect that passing a URI with a scheme specified
- but no actual resource name will automatically fail. The tests
- have been updated to not make this check. ........ Merged
- revisions 397630 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * include/asterisk/config_options.h, /, main/config_options.c,
- tests/test_config.c: Fix the config_options_test The config
- options test requires the entire configuration item to be
- transparent from the documentation system. So we let it do that
- too. As an aside, please do not use this power for evil.
- Documentation is your friend, and you really should document your
- configurations. Hiding your module's configuration information
- from the system attempting to enforce some sanity in the universe
- is something only a Bond villain would contemplate. ........
- Merged revisions 397628 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
- * /, res/res_pjsip/pjsip_configuration.c: Add rtpengine
- configuration parameter The rtpengine configuration parameter was
- documented in the XML documentation, but it was not actually
- registered with the sorcery object. This adds the parameter with
- a default of "asterisk", such that res_rtp_asterisk is chosen as
- the default RTP implementation. (closes issue ASTERISK-22380)
- Reported by: Rusty Newton Tested by: Rusty Newton ........ Merged
- revisions 397621 from
- http://svn.asterisk.org/svn/asterisk/branches/12
-
-2013-08-23 22:40 +0000 [r397615] Matthew Jordan <mjordan@digium.com>
-
- * /: Set new merge properties on 12
-
-2013-08-23 22:20 +0000 [r397613] Joshua Colp <jcolp@digium.com>
-
- * main/bucket.c: Fix building of trunk. Note: This is why I commit
- on the weekend.
-
projects / thirdparty / asterisk.git / commitdiff
