--- /dev/null
+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
