From: Asterisk Autobuilder Date: Mon, 8 Dec 2014 17:18:17 +0000 (+0000) Subject: Importing files for 13.1.0-rc1 release. X-Git-Tag: 13.1.0-rc1~1 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=bdc565a5a59816effabf421c62b62dc3e0554307;p=thirdparty%2Fasterisk.git Importing files for 13.1.0-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/13.1.0-rc1@429107 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/.lastclean b/.lastclean new file mode 100644 index 0000000000..425151f3a4 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +40 diff --git a/.version b/.version new file mode 100644 index 0000000000..1090053014 --- /dev/null +++ b/.version @@ -0,0 +1 @@ +13.1.0-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..c8bcb82157 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,20848 @@ +2014-12-08 Asterisk Development Team + + * Asterisk 13.1.0-rc1 Released. + +2014-12-08 16:53 +0000 [r429091] Matthew Jordan + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 : + + * configs/samples/manager.conf.sample: ASTERISK-23512, correct + inaccurate comment in manager.conf.sample + +2014-10-28 16:40 +0000 [r426368-426432] Matthew Jordan + + * /, 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 : + + * 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 + + * 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 + + * 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 + + * 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 + + * main/manager.c: Documentation: Improve documentation for + ExtensionStatus AMI events Review: + https://reviewboard.asterisk.org/r/4085/ + +2014-10-24 Asterisk Development Team + + * Asterisk 13.0.0 Released. + +2014-10-22 21:27 +0000 [r426097] Shaun Ruffell + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * channels/chan_motif.c: chan_motif: Cleanup + jingle_tech.capabilities only once. + +2014-10-15 19:05 +0000 [r425611] Jonathan Rose + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, + 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, + 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 + + * 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 ' 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * channels/chan_skinny.c: Skinny: Fixup compile warning for non + dev-mode. + +2014-08-18 20:19 +0000 [r421337] George Joseph + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + . 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 + + * /, 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 + + * 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 + + * 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 + + * main/manager.c: AMI: Improve documentation for Status action + +2014-08-13 07:52 +0000 [r420899] Walter Doekes + + * /, 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 + + * 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 + + * apps/app_voicemail.c: app_voicemail: Fix the + "test_voicemail_vm_info" unit test. + +2014-08-11 20:53 +0000 [r420837] Richard Mudgett + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * UPGRADE.txt: Update UPGRADE file for 13 branch + + * /: Remove old properties + + * / (added): ___ _ _ _ __ _____ / _ \ | | (_) | | / ||____ | / /_\ + \___| |_ ___ _ __ _ ___| | __ `| | / / | _ / __| __/ _ | '__| / + __| |/ / | | \ \ | | | \__ | || __| | | \__ | < _| |.___/ / \_| + |_|___/\__\___|_| |_|___|_|\_\ \___\____/ + +2014-08-07 21:58 +0000 [r420437] Richard Mudgett + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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\" ", + "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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 tag in the Asterisk XML + Documentation This patch adds support for an 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 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * res/res_fax_spandsp.c: Fix build in dev-mode + +2014-07-18 17:55 +0000 [r418961-418963] Scott Griepentrog + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 : + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * channels/chan_skinny.c: Skinny: cleanup some log messages around + sessions. + +2014-06-24 02:50 +0000 [r417167] Corey Farrell + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/http.c: http.c: Remove dead code. + +2014-05-09 17:03 +0000 [r413557] Jonathan Rose + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * main/tcptls.c: tcptls.c : Log errors as ERROR, not warning or + something else. + +2014-04-25 19:26 +0000 [r413012] Matthew Jordan + + * 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 + + * /, + 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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=). 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * main/audiohook.c: Formatting: Remove invisible characters + + * main/audiohook.c: Formatting only. + +2014-04-11 02:59 +0000 [r412154] Matthew Jordan + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, + 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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__DIR__ with the content of $ASTDIR. Review: + https://reviewboard.asterisk.org/r/3202/ + +2014-02-13 18:52 +0000 [r407990-408006] Mark Michelson + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, + 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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:) 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /: 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, res/res_pjsip/security_events.c: Print "" 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 + + * /, 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 + + * 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 + + * 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 + + * 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" ) 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 ", "mwi delete + mailbox ", "mwi list all", "mwi list like ", "mwi + show mailbox ", and "mwi update mailbox [ + []]". (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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 . 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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:). 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * include/asterisk/astdb.h: astdb: Tweak some doxygen comments. + +2013-11-27 16:12 +0000 [r403180] Joshua Colp + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, + 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 " 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 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 : 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * configs/sip.conf.sample: Remove some trailing whitespace and + steal revision 400000. + +2013-09-27 18:28 +0000 [r399991] Kevin Harwell + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, res/res_pjsip.c: Broke the build - Fixing XML DTD violation + added in r399782, missing tags inside a ........ + Merged revisions 399798 from + http://svn.asterisk.org/svn/asterisk/branches/12 + +2013-09-25 19:29 +0000 [r399797] Michael L. Young + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /: 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, main/bridge.c: CLI bridge: Fix "bridge destroy " and + "bridge kick " 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + + * /, 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 + + * 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + + * /: Set new merge properties on 12 + +2013-08-23 22:20 +0000 [r397613] Joshua Colp + + * main/bucket.c: Fix building of trunk. Note: This is why I commit + on the weekend. + diff --git a/contrib/realtime/mysql/mysql_cdr.sql b/contrib/realtime/mysql/mysql_cdr.sql new file mode 100644 index 0000000000..95d2282f3f --- /dev/null +++ b/contrib/realtime/mysql/mysql_cdr.sql @@ -0,0 +1,32 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR(20), + src VARCHAR(80), + dst VARCHAR(80), + dcontext VARCHAR(80), + clid VARCHAR(80), + channel VARCHAR(80), + dstchannel VARCHAR(80), + lastapp VARCHAR(80), + lastdata VARCHAR(80), + start DATETIME, + answer DATETIME, + end DATETIME, + duration INTEGER, + billsec INTEGER, + disposition VARCHAR(45), + amaflags VARCHAR(45), + userfield VARCHAR(256), + uniqueid VARCHAR(150), + linkedid VARCHAR(150), + peeraccount VARCHAR(20), + sequence INTEGER +); + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d'); + diff --git a/contrib/realtime/mysql/mysql_config.sql b/contrib/realtime/mysql/mysql_config.sql new file mode 100644 index 0000000000..de713fe5d1 --- /dev/null +++ b/contrib/realtime/mysql/mysql_config.sql @@ -0,0 +1,705 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> 4da0c5f79a9c + +CREATE TABLE sippeers ( + id INTEGER NOT NULL AUTO_INCREMENT, + name VARCHAR(40) NOT NULL, + ipaddr VARCHAR(45), + port INTEGER, + regseconds INTEGER, + defaultuser VARCHAR(40), + fullcontact VARCHAR(80), + regserver VARCHAR(20), + useragent VARCHAR(20), + lastms INTEGER, + host VARCHAR(40), + type ENUM('friend','user','peer'), + context VARCHAR(40), + permit VARCHAR(95), + deny VARCHAR(95), + secret VARCHAR(40), + md5secret VARCHAR(40), + remotesecret VARCHAR(40), + transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp'), + dtmfmode ENUM('rfc2833','info','shortinfo','inband','auto'), + directmedia ENUM('yes','no','nonat','update'), + nat VARCHAR(29), + callgroup VARCHAR(40), + pickupgroup VARCHAR(40), + language VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + insecure VARCHAR(40), + trustrpid ENUM('yes','no'), + progressinband ENUM('yes','no','never'), + promiscredir ENUM('yes','no'), + useclientcode ENUM('yes','no'), + accountcode VARCHAR(40), + setvar VARCHAR(200), + callerid VARCHAR(40), + amaflags VARCHAR(40), + callcounter ENUM('yes','no'), + busylevel INTEGER, + allowoverlap ENUM('yes','no'), + allowsubscribe ENUM('yes','no'), + videosupport ENUM('yes','no'), + maxcallbitrate INTEGER, + rfc2833compensate ENUM('yes','no'), + mailbox VARCHAR(40), + `session-timers` ENUM('accept','refuse','originate'), + `session-expires` INTEGER, + `session-minse` INTEGER, + `session-refresher` ENUM('uac','uas'), + t38pt_usertpsource VARCHAR(40), + regexten VARCHAR(40), + fromdomain VARCHAR(40), + fromuser VARCHAR(40), + qualify VARCHAR(40), + defaultip VARCHAR(45), + rtptimeout INTEGER, + rtpholdtimeout INTEGER, + sendrpid ENUM('yes','no'), + outboundproxy VARCHAR(40), + callbackextension VARCHAR(40), + timert1 INTEGER, + timerb INTEGER, + qualifyfreq INTEGER, + constantssrc ENUM('yes','no'), + contactpermit VARCHAR(95), + contactdeny VARCHAR(95), + usereqphone ENUM('yes','no'), + textsupport ENUM('yes','no'), + faxdetect ENUM('yes','no'), + buggymwi ENUM('yes','no'), + auth VARCHAR(40), + fullname VARCHAR(40), + trunkname VARCHAR(40), + cid_number VARCHAR(40), + callingpres ENUM('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib'), + mohinterpret VARCHAR(40), + mohsuggest VARCHAR(40), + parkinglot VARCHAR(40), + hasvoicemail ENUM('yes','no'), + subscribemwi ENUM('yes','no'), + vmexten VARCHAR(40), + autoframing ENUM('yes','no'), + rtpkeepalive INTEGER, + `call-limit` INTEGER, + g726nonstandard ENUM('yes','no'), + ignoresdpversion ENUM('yes','no'), + allowtransfer ENUM('yes','no'), + dynamic ENUM('yes','no'), + path VARCHAR(256), + supportpath ENUM('yes','no'), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX sippeers_name ON sippeers (name); + +CREATE INDEX sippeers_name_host ON sippeers (name, host); + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port); + +CREATE INDEX sippeers_host_port ON sippeers (host, port); + +CREATE TABLE iaxfriends ( + id INTEGER NOT NULL AUTO_INCREMENT, + name VARCHAR(40) NOT NULL, + type ENUM('friend','user','peer'), + username VARCHAR(40), + mailbox VARCHAR(40), + secret VARCHAR(40), + dbsecret VARCHAR(40), + context VARCHAR(40), + regcontext VARCHAR(40), + host VARCHAR(40), + ipaddr VARCHAR(40), + port INTEGER, + defaultip VARCHAR(20), + sourceaddress VARCHAR(20), + mask VARCHAR(20), + regexten VARCHAR(40), + regseconds INTEGER, + accountcode VARCHAR(20), + mohinterpret VARCHAR(20), + mohsuggest VARCHAR(20), + inkeys VARCHAR(40), + outkeys VARCHAR(40), + language VARCHAR(10), + callerid VARCHAR(100), + cid_number VARCHAR(40), + sendani ENUM('yes','no'), + fullname VARCHAR(40), + trunk ENUM('yes','no'), + auth VARCHAR(20), + maxauthreq INTEGER, + requirecalltoken ENUM('yes','no','auto'), + encryption ENUM('yes','no','aes128'), + transfer ENUM('yes','no','mediaonly'), + jitterbuffer ENUM('yes','no'), + forcejitterbuffer ENUM('yes','no'), + disallow VARCHAR(200), + allow VARCHAR(200), + codecpriority VARCHAR(40), + qualify VARCHAR(10), + qualifysmoothing ENUM('yes','no'), + qualifyfreqok VARCHAR(10), + qualifyfreqnotok VARCHAR(10), + timezone VARCHAR(20), + adsi ENUM('yes','no'), + amaflags VARCHAR(20), + setvar VARCHAR(200), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX iaxfriends_name ON iaxfriends (name); + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host); + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port); + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port); + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port); + +CREATE TABLE voicemail ( + uniqueid INTEGER NOT NULL AUTO_INCREMENT, + context VARCHAR(80) NOT NULL, + mailbox VARCHAR(80) NOT NULL, + password VARCHAR(80) NOT NULL, + fullname VARCHAR(80), + alias VARCHAR(80), + email VARCHAR(80), + pager VARCHAR(80), + attach ENUM('yes','no'), + attachfmt VARCHAR(10), + serveremail VARCHAR(80), + language VARCHAR(20), + tz VARCHAR(30), + deletevoicemail ENUM('yes','no'), + saycid ENUM('yes','no'), + sendvoicemail ENUM('yes','no'), + review ENUM('yes','no'), + tempgreetwarn ENUM('yes','no'), + operator ENUM('yes','no'), + envelope ENUM('yes','no'), + sayduration INTEGER, + forcename ENUM('yes','no'), + forcegreetings ENUM('yes','no'), + callback VARCHAR(80), + dialout VARCHAR(80), + exitcontext VARCHAR(80), + maxmsg INTEGER, + volgain NUMERIC(5, 2), + imapuser VARCHAR(80), + imappassword VARCHAR(80), + imapserver VARCHAR(80), + imapport VARCHAR(8), + imapflags VARCHAR(80), + stamp DATETIME, + PRIMARY KEY (uniqueid) +); + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox); + +CREATE INDEX voicemail_context ON voicemail (context); + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context); + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser); + +CREATE TABLE meetme ( + bookid INTEGER NOT NULL AUTO_INCREMENT, + confno VARCHAR(80) NOT NULL, + starttime DATETIME, + endtime DATETIME, + pin VARCHAR(20), + adminpin VARCHAR(20), + opts VARCHAR(20), + adminopts VARCHAR(20), + recordingfilename VARCHAR(80), + recordingformat VARCHAR(10), + maxusers INTEGER, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +); + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime); + +CREATE TABLE musiconhold ( + name VARCHAR(80) NOT NULL, + mode ENUM('custom','files','mp3nb','quietmp3nb','quietmp3'), + directory VARCHAR(255), + application VARCHAR(255), + digit VARCHAR(1), + sort VARCHAR(10), + format VARCHAR(10), + stamp DATETIME, + PRIMARY KEY (name) +); + +INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c'); + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TABLE ps_endpoints ( + id VARCHAR(40) NOT NULL, + transport VARCHAR(40), + aors VARCHAR(200), + auth VARCHAR(40), + context VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + direct_media ENUM('yes','no'), + connected_line_method ENUM('invite','reinvite','update'), + direct_media_method ENUM('invite','reinvite','update'), + direct_media_glare_mitigation ENUM('none','outgoing','incoming'), + disable_direct_media_on_nat ENUM('yes','no'), + dtmf_mode ENUM('rfc4733','inband','info'), + external_media_address VARCHAR(40), + force_rport ENUM('yes','no'), + ice_support ENUM('yes','no'), + identify_by ENUM('username'), + mailboxes VARCHAR(40), + moh_suggest VARCHAR(40), + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + rewrite_contact ENUM('yes','no'), + rtp_ipv6 ENUM('yes','no'), + rtp_symmetric ENUM('yes','no'), + send_diversion ENUM('yes','no'), + send_pai ENUM('yes','no'), + send_rpid ENUM('yes','no'), + timers_min_se INTEGER, + timers ENUM('forced','no','required','yes'), + timers_sess_expires INTEGER, + callerid VARCHAR(40), + callerid_privacy ENUM('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable'), + callerid_tag VARCHAR(40), + `100rel` ENUM('no','required','yes'), + aggregate_mwi ENUM('yes','no'), + trust_id_inbound ENUM('yes','no'), + trust_id_outbound ENUM('yes','no'), + use_ptime ENUM('yes','no'), + use_avpf ENUM('yes','no'), + media_encryption ENUM('no','sdes','dtls'), + inband_progress ENUM('yes','no'), + call_group VARCHAR(40), + pickup_group VARCHAR(40), + named_call_group VARCHAR(40), + named_pickup_group VARCHAR(40), + device_state_busy_at INTEGER, + fax_detect ENUM('yes','no'), + t38_udptl ENUM('yes','no'), + t38_udptl_ec ENUM('none','fec','redundancy'), + t38_udptl_maxdatagram INTEGER, + t38_udptl_nat ENUM('yes','no'), + t38_udptl_ipv6 ENUM('yes','no'), + tone_zone VARCHAR(40), + language VARCHAR(40), + one_touch_recording ENUM('yes','no'), + record_on_feature VARCHAR(40), + record_off_feature VARCHAR(40), + rtp_engine VARCHAR(40), + allow_transfer ENUM('yes','no'), + allow_subscribe ENUM('yes','no'), + sdp_owner VARCHAR(40), + sdp_session VARCHAR(40), + tos_audio INTEGER, + tos_video INTEGER, + cos_audio INTEGER, + cos_video INTEGER, + sub_min_expiry INTEGER, + from_domain VARCHAR(40), + from_user VARCHAR(40), + mwi_fromuser VARCHAR(40), + dtls_verify VARCHAR(40), + dtls_rekey VARCHAR(40), + dtls_cert_file VARCHAR(200), + dtls_private_key VARCHAR(200), + dtls_cipher VARCHAR(200), + dtls_ca_file VARCHAR(200), + dtls_ca_path VARCHAR(200), + dtls_setup ENUM('active','passive','actpass'), + srtp_tag_32 ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id); + +CREATE TABLE ps_auths ( + id VARCHAR(40) NOT NULL, + auth_type ENUM('md5','userpass'), + nonce_lifetime INTEGER, + md5_cred VARCHAR(40), + password VARCHAR(80), + realm VARCHAR(40), + username VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_auths_id ON ps_auths (id); + +CREATE TABLE ps_aors ( + id VARCHAR(40) NOT NULL, + contact VARCHAR(40), + default_expiration INTEGER, + mailboxes VARCHAR(80), + max_contacts INTEGER, + minimum_expiration INTEGER, + remove_existing ENUM('yes','no'), + qualify_frequency INTEGER, + authenticate_qualify ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_aors_id ON ps_aors (id); + +CREATE TABLE ps_contacts ( + id VARCHAR(40) NOT NULL, + uri VARCHAR(40), + expiration_time VARCHAR(40), + qualify_frequency INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_contacts_id ON ps_contacts (id); + +CREATE TABLE ps_domain_aliases ( + id VARCHAR(40) NOT NULL, + domain VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id); + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + `match` VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id); + +UPDATE alembic_version SET version_num='43956d550a44'; + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id BIGINT NOT NULL AUTO_INCREMENT, + context VARCHAR(40) NOT NULL, + exten VARCHAR(40) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR(40) NOT NULL, + appdata VARCHAR(256) NOT NULL, + PRIMARY KEY (id, context, exten, priority), + UNIQUE (id) +); + +UPDATE alembic_version SET version_num='581a4264e537'; + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TABLE ps_systems ( + id VARCHAR(40) NOT NULL, + timer_t1 INTEGER, + timer_b INTEGER, + compact_headers ENUM('yes','no'), + threadpool_initial_size INTEGER, + threadpool_auto_increment INTEGER, + threadpool_idle_timeout INTEGER, + threadpool_max_size INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_systems_id ON ps_systems (id); + +CREATE TABLE ps_globals ( + id VARCHAR(40) NOT NULL, + max_forwards INTEGER, + user_agent VARCHAR(40), + default_outbound_endpoint VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_globals_id ON ps_globals (id); + +CREATE TABLE ps_transports ( + id VARCHAR(40) NOT NULL, + async_operations INTEGER, + bind VARCHAR(40), + ca_list_file VARCHAR(200), + cert_file VARCHAR(200), + cipher VARCHAR(200), + domain VARCHAR(40), + external_media_address VARCHAR(40), + external_signaling_address VARCHAR(40), + external_signaling_port INTEGER, + method ENUM('default','unspecified','tlsv1','sslv2','sslv3','sslv23'), + local_net VARCHAR(40), + password VARCHAR(40), + priv_key_file VARCHAR(200), + protocol ENUM('udp','tcp','tls','ws','wss'), + require_client_cert ENUM('yes','no'), + verify_client ENUM('yes','no'), + verifiy_server ENUM('yes','no'), + tos ENUM('yes','no'), + cos ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_transports_id ON ps_transports (id); + +CREATE TABLE ps_registrations ( + id VARCHAR(40) NOT NULL, + auth_rejection_permanent ENUM('yes','no'), + client_uri VARCHAR(40), + contact_user VARCHAR(40), + expiration INTEGER, + max_retries INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + retry_interval INTEGER, + forbidden_retry_interval INTEGER, + server_uri VARCHAR(40), + transport VARCHAR(40), + support_path ENUM('yes','no'), + UNIQUE (id) +); + +CREATE INDEX ps_registrations_id ON ps_registrations (id); + +ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN redirect_method ENUM('user','uri_core','uri_pjsip'); + +ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT; + +ALTER TABLE ps_endpoints CHANGE mwi_fromuser mwi_from_user VARCHAR(40) NULL; + +ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN path TEXT; + +ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER; + +ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN support_path ENUM('yes','no'); + +UPDATE alembic_version SET version_num='2fc7930b41b3'; + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40); + +UPDATE alembic_version SET version_num='21e526ad3040'; + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TABLE queues ( + name VARCHAR(128) NOT NULL, + musiconhold VARCHAR(128), + announce VARCHAR(128), + context VARCHAR(128), + timeout INTEGER, + ringinuse ENUM('yes','no'), + setinterfacevar ENUM('yes','no'), + setqueuevar ENUM('yes','no'), + setqueueentryvar ENUM('yes','no'), + monitor_format VARCHAR(8), + membermacro VARCHAR(512), + membergosub VARCHAR(512), + queue_youarenext VARCHAR(128), + queue_thereare VARCHAR(128), + queue_callswaiting VARCHAR(128), + queue_quantity1 VARCHAR(128), + queue_quantity2 VARCHAR(128), + queue_holdtime VARCHAR(128), + queue_minutes VARCHAR(128), + queue_minute VARCHAR(128), + queue_seconds VARCHAR(128), + queue_thankyou VARCHAR(128), + queue_callerannounce VARCHAR(128), + queue_reporthold VARCHAR(128), + announce_frequency INTEGER, + announce_to_first_user ENUM('yes','no'), + min_announce_frequency INTEGER, + announce_round_seconds INTEGER, + announce_holdtime VARCHAR(128), + announce_position VARCHAR(128), + announce_position_limit INTEGER, + periodic_announce VARCHAR(50), + periodic_announce_frequency INTEGER, + relative_periodic_announce ENUM('yes','no'), + random_periodic_announce ENUM('yes','no'), + retry INTEGER, + wrapuptime INTEGER, + penaltymemberslimit INTEGER, + autofill ENUM('yes','no'), + monitor_type VARCHAR(128), + autopause ENUM('yes','no','all'), + autopausedelay INTEGER, + autopausebusy ENUM('yes','no'), + autopauseunavail ENUM('yes','no'), + maxlen INTEGER, + servicelevel INTEGER, + strategy ENUM('ringall','leastrecent','fewestcalls','random','rrmemory','linear','wrandom','rrordered'), + joinempty VARCHAR(128), + leavewhenempty VARCHAR(128), + reportholdtime ENUM('yes','no'), + memberdelay INTEGER, + weight INTEGER, + timeoutrestart ENUM('yes','no'), + defaultrule VARCHAR(128), + timeoutpriority VARCHAR(128), + PRIMARY KEY (name) +); + +CREATE TABLE queue_members ( + queue_name VARCHAR(80) NOT NULL, + interface VARCHAR(80) NOT NULL, + uniqueid VARCHAR(80) NOT NULL, + membername VARCHAR(80), + state_interface VARCHAR(80), + penalty INTEGER, + paused INTEGER, + PRIMARY KEY (queue_name, interface) +); + +UPDATE alembic_version SET version_num='28887f25a46f'; + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints CHANGE tos_audio tos_audio VARCHAR(10) NULL; + +ALTER TABLE ps_endpoints CHANGE tos_video tos_video VARCHAR(10) NULL; + +ALTER TABLE ps_transports CHANGE tos tos VARCHAR(10) NULL; + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio; + +ALTER TABLE ps_endpoints DROP COLUMN cos_video; + +ALTER TABLE ps_transports DROP COLUMN cos; + +ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER; + +ALTER TABLE ps_transports ADD COLUMN cos INTEGER; + +UPDATE alembic_version SET version_num='4c573e7135bd'; + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40); + +UPDATE alembic_version SET version_num='3855ee4e5f85'; + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals CHANGE user_agent user_agent VARCHAR(255) NULL; + +ALTER TABLE ps_contacts CHANGE id id VARCHAR(255) NULL; + +ALTER TABLE ps_contacts CHANGE uri uri VARCHAR(255) NULL; + +ALTER TABLE ps_contacts CHANGE user_agent user_agent VARCHAR(255) NULL; + +ALTER TABLE ps_registrations CHANGE client_uri client_uri VARCHAR(255) NULL; + +ALTER TABLE ps_registrations CHANGE server_uri server_uri VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='e96a0b8071c'; + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR(40) NOT NULL, + packet VARCHAR(2048), + src_name VARCHAR(128), + src_port INTEGER, + transport_key VARCHAR(64), + local_name VARCHAR(128), + local_port INTEGER, + cseq INTEGER, + tag VARCHAR(128), + endpoint VARCHAR(40), + expires INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id); + +UPDATE alembic_version SET version_num='c6d929b23a8'; + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD COLUMN force_avp ENUM('yes','no'); + +ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport ENUM('yes','no'); + +UPDATE alembic_version SET version_num='51f8cb66540e'; + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20); + +UPDATE alembic_version SET version_num='1d50859ed02e'; + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers CHANGE useragent useragent VARCHAR(255) NULL; + +UPDATE alembic_version SET version_num='1758e8bbf6b'; + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid; + +ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL; + +ALTER TABLE queue_members ADD UNIQUE (uniqueid); + +ALTER TABLE queue_members CHANGE uniqueid uniqueid INTEGER NOT NULL AUTO_INCREMENT; + +UPDATE alembic_version SET version_num='5139253c0423'; + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR(80) NOT NULL, + time VARCHAR(32) NOT NULL, + min_penalty VARCHAR(32) NOT NULL, + max_penalty VARCHAR(32) NOT NULL +); + +UPDATE alembic_version SET version_num='d39508cb8d8'; + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports CHANGE verifiy_server verify_server ENUM('yes','no') NULL; + +UPDATE alembic_version SET version_num='5950038a6ead'; + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +ALTER TABLE sippeers CHANGE directmedia directmedia ENUM('yes','no','nonat','update','outgoing') NULL; + +UPDATE alembic_version SET version_num='10aedae86a32'; + +-- Running upgrade 10aedae86a32 -> eb88a14f2a + +ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic ENUM('yes','no'); + +UPDATE alembic_version SET version_num='eb88a14f2a'; + diff --git a/contrib/realtime/mysql/mysql_voicemail.sql b/contrib/realtime/mysql/mysql_voicemail.sql new file mode 100644 index 0000000000..ff5b6205f8 --- /dev/null +++ b/contrib/realtime/mysql/mysql_voicemail.sql @@ -0,0 +1,34 @@ +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR(255) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR(80), + macrocontext VARCHAR(80), + callerid VARCHAR(80), + origtime INTEGER, + duration INTEGER, + recording BLOB, + flag VARCHAR(30), + category VARCHAR(30), + mailboxuser VARCHAR(30), + mailboxcontext VARCHAR(30), + msg_id VARCHAR(40) +); + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum); + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir); + +INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e'); + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages CHANGE recording recording BLOB(4294967295) NULL; + +UPDATE alembic_version SET version_num='39428242f7f5'; + diff --git a/contrib/realtime/oracle/oracle_cdr.sql b/contrib/realtime/oracle/oracle_cdr.sql new file mode 100644 index 0000000000..66302fc94f --- /dev/null +++ b/contrib/realtime/oracle/oracle_cdr.sql @@ -0,0 +1,46 @@ +SET TRANSACTION READ WRITE + +/ + +CREATE TABLE alembic_version ( + version_num VARCHAR2(32 CHAR) NOT NULL +) + +/ + +-- Running upgrade None -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR2(20 CHAR), + src VARCHAR2(80 CHAR), + dst VARCHAR2(80 CHAR), + dcontext VARCHAR2(80 CHAR), + clid VARCHAR2(80 CHAR), + channel VARCHAR2(80 CHAR), + dstchannel VARCHAR2(80 CHAR), + lastapp VARCHAR2(80 CHAR), + lastdata VARCHAR2(80 CHAR), + "start" DATE, + answer DATE, + end DATE, + duration INTEGER, + billsec INTEGER, + disposition VARCHAR2(45 CHAR), + amaflags VARCHAR2(45 CHAR), + userfield VARCHAR2(256 CHAR), + uniqueid VARCHAR2(150 CHAR), + linkedid VARCHAR2(150 CHAR), + peeraccount VARCHAR2(20 CHAR), + sequence INTEGER +) + +/ + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d') + +/ + +COMMIT + +/ + diff --git a/contrib/realtime/oracle/oracle_config.sql b/contrib/realtime/oracle/oracle_config.sql new file mode 100644 index 0000000000..ff6f7d0e8e --- /dev/null +++ b/contrib/realtime/oracle/oracle_config.sql @@ -0,0 +1,994 @@ +SET TRANSACTION READ WRITE + +/ + +CREATE TABLE alembic_version ( + version_num VARCHAR2(32 CHAR) NOT NULL +) + +/ + +-- Running upgrade None -> 4da0c5f79a9c + +CREATE TABLE sippeers ( + id INTEGER NOT NULL, + name VARCHAR2(40 CHAR) NOT NULL, + ipaddr VARCHAR2(45 CHAR), + port INTEGER, + regseconds INTEGER, + defaultuser VARCHAR2(40 CHAR), + fullcontact VARCHAR2(80 CHAR), + regserver VARCHAR2(20 CHAR), + useragent VARCHAR2(20 CHAR), + lastms INTEGER, + host VARCHAR2(40 CHAR), + type VARCHAR(6 CHAR), + context VARCHAR2(40 CHAR), + permit VARCHAR2(95 CHAR), + deny VARCHAR2(95 CHAR), + secret VARCHAR2(40 CHAR), + md5secret VARCHAR2(40 CHAR), + remotesecret VARCHAR2(40 CHAR), + transport VARCHAR(7 CHAR), + dtmfmode VARCHAR(9 CHAR), + directmedia VARCHAR(6 CHAR), + nat VARCHAR2(29 CHAR), + callgroup VARCHAR2(40 CHAR), + pickupgroup VARCHAR2(40 CHAR), + language VARCHAR2(40 CHAR), + disallow VARCHAR2(200 CHAR), + allow VARCHAR2(200 CHAR), + insecure VARCHAR2(40 CHAR), + trustrpid VARCHAR(3 CHAR), + progressinband VARCHAR(5 CHAR), + promiscredir VARCHAR(3 CHAR), + useclientcode VARCHAR(3 CHAR), + accountcode VARCHAR2(40 CHAR), + setvar VARCHAR2(200 CHAR), + callerid VARCHAR2(40 CHAR), + amaflags VARCHAR2(40 CHAR), + callcounter VARCHAR(3 CHAR), + busylevel INTEGER, + allowoverlap VARCHAR(3 CHAR), + allowsubscribe VARCHAR(3 CHAR), + videosupport VARCHAR(3 CHAR), + maxcallbitrate INTEGER, + rfc2833compensate VARCHAR(3 CHAR), + mailbox VARCHAR2(40 CHAR), + "session-timers" VARCHAR(9 CHAR), + "session-expires" INTEGER, + "session-minse" INTEGER, + "session-refresher" VARCHAR(3 CHAR), + t38pt_usertpsource VARCHAR2(40 CHAR), + regexten VARCHAR2(40 CHAR), + fromdomain VARCHAR2(40 CHAR), + fromuser VARCHAR2(40 CHAR), + qualify VARCHAR2(40 CHAR), + defaultip VARCHAR2(45 CHAR), + rtptimeout INTEGER, + rtpholdtimeout INTEGER, + sendrpid VARCHAR(3 CHAR), + outboundproxy VARCHAR2(40 CHAR), + callbackextension VARCHAR2(40 CHAR), + timert1 INTEGER, + timerb INTEGER, + qualifyfreq INTEGER, + constantssrc VARCHAR(3 CHAR), + contactpermit VARCHAR2(95 CHAR), + contactdeny VARCHAR2(95 CHAR), + usereqphone VARCHAR(3 CHAR), + textsupport VARCHAR(3 CHAR), + faxdetect VARCHAR(3 CHAR), + buggymwi VARCHAR(3 CHAR), + auth VARCHAR2(40 CHAR), + fullname VARCHAR2(40 CHAR), + trunkname VARCHAR2(40 CHAR), + cid_number VARCHAR2(40 CHAR), + callingpres VARCHAR(21 CHAR), + mohinterpret VARCHAR2(40 CHAR), + mohsuggest VARCHAR2(40 CHAR), + parkinglot VARCHAR2(40 CHAR), + hasvoicemail VARCHAR(3 CHAR), + subscribemwi VARCHAR(3 CHAR), + vmexten VARCHAR2(40 CHAR), + autoframing VARCHAR(3 CHAR), + rtpkeepalive INTEGER, + "call-limit" INTEGER, + g726nonstandard VARCHAR(3 CHAR), + ignoresdpversion VARCHAR(3 CHAR), + allowtransfer VARCHAR(3 CHAR), + dynamic VARCHAR(3 CHAR), + path VARCHAR2(256 CHAR), + supportpath VARCHAR(3 CHAR), + PRIMARY KEY (id), + UNIQUE (name), + CONSTRAINT type_values CHECK (type IN ('friend', 'user', 'peer')), + CONSTRAINT sip_transport_values CHECK (transport IN ('udp', 'tcp', 'tls', 'ws', 'wss', 'udp,tcp', 'tcp,udp')), + CONSTRAINT sip_dtmfmode_values CHECK (dtmfmode IN ('rfc2833', 'info', 'shortinfo', 'inband', 'auto')), + CONSTRAINT sip_directmedia_values CHECK (directmedia IN ('yes', 'no', 'nonat', 'update')), + CONSTRAINT yes_no_values CHECK (trustrpid IN ('yes', 'no')), + CONSTRAINT sip_progressinband_values CHECK (progressinband IN ('yes', 'no', 'never')), + CONSTRAINT yes_no_values CHECK (promiscredir IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (useclientcode IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (callcounter IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowoverlap IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowsubscribe IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (videosupport IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (rfc2833compensate IN ('yes', 'no')), + CONSTRAINT sip_session_timers_values CHECK ("session-timers" IN ('accept', 'refuse', 'originate')), + CONSTRAINT sip_session_refresher_values CHECK ("session-refresher" IN ('uac', 'uas')), + CONSTRAINT yes_no_values CHECK (sendrpid IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (constantssrc IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (usereqphone IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (textsupport IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (faxdetect IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (buggymwi IN ('yes', 'no')), + CONSTRAINT sip_callingpres_values CHECK (callingpres IN ('allowed_not_screened', 'allowed_passed_screen', 'allowed_failed_screen', 'allowed', 'prohib_not_screened', 'prohib_passed_screen', 'prohib_failed_screen', 'prohib')), + CONSTRAINT yes_no_values CHECK (hasvoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (subscribemwi IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (autoframing IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (g726nonstandard IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (ignoresdpversion IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowtransfer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (dynamic IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (supportpath IN ('yes', 'no')) +) + +/ + +CREATE INDEX sippeers_name ON sippeers (name) + +/ + +CREATE INDEX sippeers_name_host ON sippeers (name, host) + +/ + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port) + +/ + +CREATE INDEX sippeers_host_port ON sippeers (host, port) + +/ + +CREATE TABLE iaxfriends ( + id INTEGER NOT NULL, + name VARCHAR2(40 CHAR) NOT NULL, + type VARCHAR(6 CHAR), + username VARCHAR2(40 CHAR), + mailbox VARCHAR2(40 CHAR), + secret VARCHAR2(40 CHAR), + dbsecret VARCHAR2(40 CHAR), + context VARCHAR2(40 CHAR), + regcontext VARCHAR2(40 CHAR), + host VARCHAR2(40 CHAR), + ipaddr VARCHAR2(40 CHAR), + port INTEGER, + defaultip VARCHAR2(20 CHAR), + sourceaddress VARCHAR2(20 CHAR), + mask VARCHAR2(20 CHAR), + regexten VARCHAR2(40 CHAR), + regseconds INTEGER, + accountcode VARCHAR2(20 CHAR), + mohinterpret VARCHAR2(20 CHAR), + mohsuggest VARCHAR2(20 CHAR), + inkeys VARCHAR2(40 CHAR), + outkeys VARCHAR2(40 CHAR), + language VARCHAR2(10 CHAR), + callerid VARCHAR2(100 CHAR), + cid_number VARCHAR2(40 CHAR), + sendani VARCHAR(3 CHAR), + fullname VARCHAR2(40 CHAR), + trunk VARCHAR(3 CHAR), + auth VARCHAR2(20 CHAR), + maxauthreq INTEGER, + requirecalltoken VARCHAR(4 CHAR), + encryption VARCHAR(6 CHAR), + transfer VARCHAR(9 CHAR), + jitterbuffer VARCHAR(3 CHAR), + forcejitterbuffer VARCHAR(3 CHAR), + disallow VARCHAR2(200 CHAR), + allow VARCHAR2(200 CHAR), + codecpriority VARCHAR2(40 CHAR), + qualify VARCHAR2(10 CHAR), + qualifysmoothing VARCHAR(3 CHAR), + qualifyfreqok VARCHAR2(10 CHAR), + qualifyfreqnotok VARCHAR2(10 CHAR), + timezone VARCHAR2(20 CHAR), + adsi VARCHAR(3 CHAR), + amaflags VARCHAR2(20 CHAR), + setvar VARCHAR2(200 CHAR), + PRIMARY KEY (id), + UNIQUE (name), + CONSTRAINT type_values CHECK (type IN ('friend', 'user', 'peer')), + CONSTRAINT yes_no_values CHECK (sendani IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (trunk IN ('yes', 'no')), + CONSTRAINT iax_requirecalltoken_values CHECK (requirecalltoken IN ('yes', 'no', 'auto')), + CONSTRAINT iax_encryption_values CHECK (encryption IN ('yes', 'no', 'aes128')), + CONSTRAINT iax_transfer_values CHECK (transfer IN ('yes', 'no', 'mediaonly')), + CONSTRAINT yes_no_values CHECK (jitterbuffer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcejitterbuffer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (qualifysmoothing IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (adsi IN ('yes', 'no')) +) + +/ + +CREATE INDEX iaxfriends_name ON iaxfriends (name) + +/ + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host) + +/ + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port) + +/ + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port) + +/ + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port) + +/ + +CREATE TABLE voicemail ( + uniqueid INTEGER NOT NULL, + context VARCHAR2(80 CHAR) NOT NULL, + mailbox VARCHAR2(80 CHAR) NOT NULL, + password VARCHAR2(80 CHAR) NOT NULL, + fullname VARCHAR2(80 CHAR), + alias VARCHAR2(80 CHAR), + email VARCHAR2(80 CHAR), + pager VARCHAR2(80 CHAR), + attach VARCHAR(3 CHAR), + attachfmt VARCHAR2(10 CHAR), + serveremail VARCHAR2(80 CHAR), + language VARCHAR2(20 CHAR), + tz VARCHAR2(30 CHAR), + deletevoicemail VARCHAR(3 CHAR), + saycid VARCHAR(3 CHAR), + sendvoicemail VARCHAR(3 CHAR), + review VARCHAR(3 CHAR), + tempgreetwarn VARCHAR(3 CHAR), + operator VARCHAR(3 CHAR), + envelope VARCHAR(3 CHAR), + sayduration INTEGER, + forcename VARCHAR(3 CHAR), + forcegreetings VARCHAR(3 CHAR), + callback VARCHAR2(80 CHAR), + dialout VARCHAR2(80 CHAR), + exitcontext VARCHAR2(80 CHAR), + maxmsg INTEGER, + volgain NUMERIC(5, 2), + imapuser VARCHAR2(80 CHAR), + imappassword VARCHAR2(80 CHAR), + imapserver VARCHAR2(80 CHAR), + imapport VARCHAR2(8 CHAR), + imapflags VARCHAR2(80 CHAR), + stamp DATE, + PRIMARY KEY (uniqueid), + CONSTRAINT yes_no_values CHECK (attach IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (deletevoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (saycid IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (sendvoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (review IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (tempgreetwarn IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (operator IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (envelope IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcename IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcegreetings IN ('yes', 'no')) +) + +/ + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox) + +/ + +CREATE INDEX voicemail_context ON voicemail (context) + +/ + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context) + +/ + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser) + +/ + +CREATE TABLE meetme ( + bookid INTEGER NOT NULL, + confno VARCHAR2(80 CHAR) NOT NULL, + starttime DATE, + endtime DATE, + pin VARCHAR2(20 CHAR), + adminpin VARCHAR2(20 CHAR), + opts VARCHAR2(20 CHAR), + adminopts VARCHAR2(20 CHAR), + recordingfilename VARCHAR2(80 CHAR), + recordingformat VARCHAR2(10 CHAR), + maxusers INTEGER, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +) + +/ + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime) + +/ + +CREATE TABLE musiconhold ( + name VARCHAR2(80 CHAR) NOT NULL, + "mode" VARCHAR(10 CHAR), + directory VARCHAR2(255 CHAR), + application VARCHAR2(255 CHAR), + digit VARCHAR2(1 CHAR), + sort VARCHAR2(10 CHAR), + format VARCHAR2(10 CHAR), + stamp DATE, + PRIMARY KEY (name), + CONSTRAINT moh_mode_values CHECK ("mode" IN ('custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3')) +) + +/ + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TABLE ps_endpoints ( + id VARCHAR2(40 CHAR) NOT NULL, + transport VARCHAR2(40 CHAR), + aors VARCHAR2(200 CHAR), + auth VARCHAR2(40 CHAR), + context VARCHAR2(40 CHAR), + disallow VARCHAR2(200 CHAR), + allow VARCHAR2(200 CHAR), + direct_media VARCHAR(3 CHAR), + connected_line_method VARCHAR(8 CHAR), + direct_media_method VARCHAR(8 CHAR), + direct_media_glare_mitigation VARCHAR(8 CHAR), + disable_direct_media_on_nat VARCHAR(3 CHAR), + dtmf_mode VARCHAR(7 CHAR), + external_media_address VARCHAR2(40 CHAR), + force_rport VARCHAR(3 CHAR), + ice_support VARCHAR(3 CHAR), + identify_by VARCHAR(8 CHAR), + mailboxes VARCHAR2(40 CHAR), + moh_suggest VARCHAR2(40 CHAR), + outbound_auth VARCHAR2(40 CHAR), + outbound_proxy VARCHAR2(40 CHAR), + rewrite_contact VARCHAR(3 CHAR), + rtp_ipv6 VARCHAR(3 CHAR), + rtp_symmetric VARCHAR(3 CHAR), + send_diversion VARCHAR(3 CHAR), + send_pai VARCHAR(3 CHAR), + send_rpid VARCHAR(3 CHAR), + timers_min_se INTEGER, + timers VARCHAR(8 CHAR), + timers_sess_expires INTEGER, + callerid VARCHAR2(40 CHAR), + callerid_privacy VARCHAR(23 CHAR), + callerid_tag VARCHAR2(40 CHAR), + 100rel VARCHAR(8 CHAR), + aggregate_mwi VARCHAR(3 CHAR), + trust_id_inbound VARCHAR(3 CHAR), + trust_id_outbound VARCHAR(3 CHAR), + use_ptime VARCHAR(3 CHAR), + use_avpf VARCHAR(3 CHAR), + media_encryption VARCHAR(4 CHAR), + inband_progress VARCHAR(3 CHAR), + call_group VARCHAR2(40 CHAR), + pickup_group VARCHAR2(40 CHAR), + named_call_group VARCHAR2(40 CHAR), + named_pickup_group VARCHAR2(40 CHAR), + device_state_busy_at INTEGER, + fax_detect VARCHAR(3 CHAR), + t38_udptl VARCHAR(3 CHAR), + t38_udptl_ec VARCHAR(10 CHAR), + t38_udptl_maxdatagram INTEGER, + t38_udptl_nat VARCHAR(3 CHAR), + t38_udptl_ipv6 VARCHAR(3 CHAR), + tone_zone VARCHAR2(40 CHAR), + language VARCHAR2(40 CHAR), + one_touch_recording VARCHAR(3 CHAR), + record_on_feature VARCHAR2(40 CHAR), + record_off_feature VARCHAR2(40 CHAR), + rtp_engine VARCHAR2(40 CHAR), + allow_transfer VARCHAR(3 CHAR), + allow_subscribe VARCHAR(3 CHAR), + sdp_owner VARCHAR2(40 CHAR), + sdp_session VARCHAR2(40 CHAR), + tos_audio INTEGER, + tos_video INTEGER, + cos_audio INTEGER, + cos_video INTEGER, + sub_min_expiry INTEGER, + from_domain VARCHAR2(40 CHAR), + from_user VARCHAR2(40 CHAR), + mwi_fromuser VARCHAR2(40 CHAR), + dtls_verify VARCHAR2(40 CHAR), + dtls_rekey VARCHAR2(40 CHAR), + dtls_cert_file VARCHAR2(200 CHAR), + dtls_private_key VARCHAR2(200 CHAR), + dtls_cipher VARCHAR2(200 CHAR), + dtls_ca_file VARCHAR2(200 CHAR), + dtls_ca_path VARCHAR2(200 CHAR), + dtls_setup VARCHAR(7 CHAR), + srtp_tag_32 VARCHAR(3 CHAR), + UNIQUE (id), + CONSTRAINT yesno_values CHECK (direct_media IN ('yes', 'no')), + CONSTRAINT pjsip_connected_line_method_values CHECK (connected_line_method IN ('invite', 'reinvite', 'update')), + CONSTRAINT pjsip_connected_line_method_values CHECK (direct_media_method IN ('invite', 'reinvite', 'update')), + CONSTRAINT pjsip_direct_media_glare_mitigation_values CHECK (direct_media_glare_mitigation IN ('none', 'outgoing', 'incoming')), + CONSTRAINT yesno_values CHECK (disable_direct_media_on_nat IN ('yes', 'no')), + CONSTRAINT pjsip_dtmf_mode_values CHECK (dtmf_mode IN ('rfc4733', 'inband', 'info')), + CONSTRAINT yesno_values CHECK (force_rport IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (ice_support IN ('yes', 'no')), + CONSTRAINT pjsip_identify_by_values CHECK (identify_by IN ('username')), + CONSTRAINT yesno_values CHECK (rewrite_contact IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (rtp_ipv6 IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (rtp_symmetric IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_diversion IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_pai IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_rpid IN ('yes', 'no')), + CONSTRAINT pjsip_timer_values CHECK (timers IN ('forced', 'no', 'required', 'yes')), + CONSTRAINT pjsip_cid_privacy_values CHECK (callerid_privacy IN ('allowed_not_screened', 'allowed_passed_screened', 'allowed_failed_screened', 'allowed', 'prohib_not_screened', 'prohib_passed_screened', 'prohib_failed_screened', 'prohib', 'unavailable')), + CONSTRAINT pjsip_100rel_values CHECK (100rel IN ('no', 'required', 'yes')), + CONSTRAINT yesno_values CHECK (aggregate_mwi IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (trust_id_inbound IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (trust_id_outbound IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (use_ptime IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (use_avpf IN ('yes', 'no')), + CONSTRAINT pjsip_media_encryption_values CHECK (media_encryption IN ('no', 'sdes', 'dtls')), + CONSTRAINT yesno_values CHECK (inband_progress IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (fax_detect IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (t38_udptl IN ('yes', 'no')), + CONSTRAINT pjsip_t38udptl_ec_values CHECK (t38_udptl_ec IN ('none', 'fec', 'redundancy')), + CONSTRAINT yesno_values CHECK (t38_udptl_nat IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (t38_udptl_ipv6 IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (one_touch_recording IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (allow_transfer IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (allow_subscribe IN ('yes', 'no')), + CONSTRAINT pjsip_dtls_setup_values CHECK (dtls_setup IN ('active', 'passive', 'actpass')), + CONSTRAINT yesno_values CHECK (srtp_tag_32 IN ('yes', 'no')) +) + +/ + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id) + +/ + +CREATE TABLE ps_auths ( + id VARCHAR2(40 CHAR) NOT NULL, + auth_type VARCHAR(8 CHAR), + nonce_lifetime INTEGER, + md5_cred VARCHAR2(40 CHAR), + password VARCHAR2(80 CHAR), + realm VARCHAR2(40 CHAR), + username VARCHAR2(40 CHAR), + UNIQUE (id), + CONSTRAINT pjsip_auth_type_values CHECK (auth_type IN ('md5', 'userpass')) +) + +/ + +CREATE INDEX ps_auths_id ON ps_auths (id) + +/ + +CREATE TABLE ps_aors ( + id VARCHAR2(40 CHAR) NOT NULL, + contact VARCHAR2(40 CHAR), + default_expiration INTEGER, + mailboxes VARCHAR2(80 CHAR), + max_contacts INTEGER, + minimum_expiration INTEGER, + remove_existing VARCHAR(3 CHAR), + qualify_frequency INTEGER, + authenticate_qualify VARCHAR(3 CHAR), + UNIQUE (id), + CONSTRAINT yesno_values CHECK (remove_existing IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (authenticate_qualify IN ('yes', 'no')) +) + +/ + +CREATE INDEX ps_aors_id ON ps_aors (id) + +/ + +CREATE TABLE ps_contacts ( + id VARCHAR2(40 CHAR) NOT NULL, + uri VARCHAR2(40 CHAR), + expiration_time VARCHAR2(40 CHAR), + qualify_frequency INTEGER, + UNIQUE (id) +) + +/ + +CREATE INDEX ps_contacts_id ON ps_contacts (id) + +/ + +CREATE TABLE ps_domain_aliases ( + id VARCHAR2(40 CHAR) NOT NULL, + domain VARCHAR2(80 CHAR), + UNIQUE (id) +) + +/ + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id) + +/ + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR2(40 CHAR) NOT NULL, + endpoint VARCHAR2(40 CHAR), + match VARCHAR2(80 CHAR), + UNIQUE (id) +) + +/ + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id) + +/ + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id NUMBER(19) NOT NULL, + context VARCHAR2(40 CHAR) NOT NULL, + exten VARCHAR2(40 CHAR) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR2(40 CHAR) NOT NULL, + appdata VARCHAR2(256 CHAR) NOT NULL, + PRIMARY KEY (id, context, exten, priority), + UNIQUE (id) +) + +/ + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TABLE ps_systems ( + id VARCHAR2(40 CHAR) NOT NULL, + timer_t1 INTEGER, + timer_b INTEGER, + compact_headers VARCHAR(3 CHAR), + threadpool_initial_size INTEGER, + threadpool_auto_increment INTEGER, + threadpool_idle_timeout INTEGER, + threadpool_max_size INTEGER, + UNIQUE (id), + CONSTRAINT yesno_values CHECK (compact_headers IN ('yes', 'no')) +) + +/ + +CREATE INDEX ps_systems_id ON ps_systems (id) + +/ + +CREATE TABLE ps_globals ( + id VARCHAR2(40 CHAR) NOT NULL, + max_forwards INTEGER, + user_agent VARCHAR2(40 CHAR), + default_outbound_endpoint VARCHAR2(40 CHAR), + UNIQUE (id) +) + +/ + +CREATE INDEX ps_globals_id ON ps_globals (id) + +/ + +CREATE TABLE ps_transports ( + id VARCHAR2(40 CHAR) NOT NULL, + async_operations INTEGER, + bind VARCHAR2(40 CHAR), + ca_list_file VARCHAR2(200 CHAR), + cert_file VARCHAR2(200 CHAR), + cipher VARCHAR2(200 CHAR), + domain VARCHAR2(40 CHAR), + external_media_address VARCHAR2(40 CHAR), + external_signaling_address VARCHAR2(40 CHAR), + external_signaling_port INTEGER, + method VARCHAR(11 CHAR), + local_net VARCHAR2(40 CHAR), + password VARCHAR2(40 CHAR), + priv_key_file VARCHAR2(200 CHAR), + protocol VARCHAR(3 CHAR), + require_client_cert VARCHAR(3 CHAR), + verify_client VARCHAR(3 CHAR), + verifiy_server VARCHAR(3 CHAR), + tos VARCHAR(3 CHAR), + cos VARCHAR(3 CHAR), + UNIQUE (id), + CONSTRAINT pjsip_transport_method_values CHECK (method IN ('default', 'unspecified', 'tlsv1', 'sslv2', 'sslv3', 'sslv23')), + CONSTRAINT pjsip_transport_protocol_values CHECK (protocol IN ('udp', 'tcp', 'tls', 'ws', 'wss')), + CONSTRAINT yesno_values CHECK (require_client_cert IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (verify_client IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (verifiy_server IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (tos IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (cos IN ('yes', 'no')) +) + +/ + +CREATE INDEX ps_transports_id ON ps_transports (id) + +/ + +CREATE TABLE ps_registrations ( + id VARCHAR2(40 CHAR) NOT NULL, + auth_rejection_permanent VARCHAR(3 CHAR), + client_uri VARCHAR2(40 CHAR), + contact_user VARCHAR2(40 CHAR), + expiration INTEGER, + max_retries INTEGER, + outbound_auth VARCHAR2(40 CHAR), + outbound_proxy VARCHAR2(40 CHAR), + retry_interval INTEGER, + forbidden_retry_interval INTEGER, + server_uri VARCHAR2(40 CHAR), + transport VARCHAR2(40 CHAR), + support_path VARCHAR(3 CHAR), + UNIQUE (id), + CONSTRAINT yesno_values CHECK (auth_rejection_permanent IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (support_path IN ('yes', 'no')) +) + +/ + +CREATE INDEX ps_registrations_id ON ps_registrations (id) + +/ + +ALTER TABLE ps_endpoints ADD media_address VARCHAR2(40 CHAR) + +/ + +ALTER TABLE ps_endpoints ADD redirect_method VARCHAR(9 CHAR) + +/ + +ALTER TABLE ps_endpoints ADD CONSTRAINT pjsip_redirect_method_values CHECK (redirect_method IN ('user', 'uri_core', 'uri_pjsip')) + +/ + +ALTER TABLE ps_endpoints ADD set_var CLOB + +/ + +ALTER TABLE ps_endpoints RENAME COLUMN mwi_fromuser TO mwi_from_user + +/ + +ALTER TABLE ps_contacts ADD outbound_proxy VARCHAR2(40 CHAR) + +/ + +ALTER TABLE ps_contacts ADD path CLOB + +/ + +ALTER TABLE ps_aors ADD maximum_expiration INTEGER + +/ + +ALTER TABLE ps_aors ADD outbound_proxy VARCHAR2(40 CHAR) + +/ + +ALTER TABLE ps_aors ADD support_path VARCHAR(3 CHAR) + +/ + +ALTER TABLE ps_aors ADD CONSTRAINT yesno_values CHECK (support_path IN ('yes', 'no')) + +/ + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD debug VARCHAR2(40 CHAR) + +/ + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TABLE queues ( + name VARCHAR2(128 CHAR) NOT NULL, + musiconhold VARCHAR2(128 CHAR), + announce VARCHAR2(128 CHAR), + context VARCHAR2(128 CHAR), + timeout INTEGER, + ringinuse VARCHAR(3 CHAR), + setinterfacevar VARCHAR(3 CHAR), + setqueuevar VARCHAR(3 CHAR), + setqueueentryvar VARCHAR(3 CHAR), + monitor_format VARCHAR2(8 CHAR), + membermacro VARCHAR2(512 CHAR), + membergosub VARCHAR2(512 CHAR), + queue_youarenext VARCHAR2(128 CHAR), + queue_thereare VARCHAR2(128 CHAR), + queue_callswaiting VARCHAR2(128 CHAR), + queue_quantity1 VARCHAR2(128 CHAR), + queue_quantity2 VARCHAR2(128 CHAR), + queue_holdtime VARCHAR2(128 CHAR), + queue_minutes VARCHAR2(128 CHAR), + queue_minute VARCHAR2(128 CHAR), + queue_seconds VARCHAR2(128 CHAR), + queue_thankyou VARCHAR2(128 CHAR), + queue_callerannounce VARCHAR2(128 CHAR), + queue_reporthold VARCHAR2(128 CHAR), + announce_frequency INTEGER, + announce_to_first_user VARCHAR(3 CHAR), + min_announce_frequency INTEGER, + announce_round_seconds INTEGER, + announce_holdtime VARCHAR2(128 CHAR), + announce_position VARCHAR2(128 CHAR), + announce_position_limit INTEGER, + periodic_announce VARCHAR2(50 CHAR), + periodic_announce_frequency INTEGER, + relative_periodic_announce VARCHAR(3 CHAR), + random_periodic_announce VARCHAR(3 CHAR), + retry INTEGER, + wrapuptime INTEGER, + penaltymemberslimit INTEGER, + autofill VARCHAR(3 CHAR), + monitor_type VARCHAR2(128 CHAR), + autopause VARCHAR(3 CHAR), + autopausedelay INTEGER, + autopausebusy VARCHAR(3 CHAR), + autopauseunavail VARCHAR(3 CHAR), + maxlen INTEGER, + servicelevel INTEGER, + strategy VARCHAR(11 CHAR), + joinempty VARCHAR2(128 CHAR), + leavewhenempty VARCHAR2(128 CHAR), + reportholdtime VARCHAR(3 CHAR), + memberdelay INTEGER, + weight INTEGER, + timeoutrestart VARCHAR(3 CHAR), + defaultrule VARCHAR2(128 CHAR), + timeoutpriority VARCHAR2(128 CHAR), + PRIMARY KEY (name), + CONSTRAINT yesno_values CHECK (ringinuse IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setinterfacevar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setqueuevar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setqueueentryvar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (announce_to_first_user IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (relative_periodic_announce IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (random_periodic_announce IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (autofill IN ('yes', 'no')), + CONSTRAINT queue_autopause_values CHECK (autopause IN ('yes', 'no', 'all')), + CONSTRAINT yesno_values CHECK (autopausebusy IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (autopauseunavail IN ('yes', 'no')), + CONSTRAINT queue_strategy_values CHECK (strategy IN ('ringall', 'leastrecent', 'fewestcalls', 'random', 'rrmemory', 'linear', 'wrandom', 'rrordered')), + CONSTRAINT yesno_values CHECK (reportholdtime IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (timeoutrestart IN ('yes', 'no')) +) + +/ + +CREATE TABLE queue_members ( + queue_name VARCHAR2(80 CHAR) NOT NULL, + interface VARCHAR2(80 CHAR) NOT NULL, + uniqueid VARCHAR2(80 CHAR) NOT NULL, + membername VARCHAR2(80 CHAR), + state_interface VARCHAR2(80 CHAR), + penalty INTEGER, + paused INTEGER, + PRIMARY KEY (queue_name, interface) +) + +/ + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints MODIFY tos_audio VARCHAR2(10 CHAR) + +/ + +ALTER TABLE ps_endpoints MODIFY tos_video VARCHAR2(10 CHAR) + +/ + +ALTER TABLE ps_transports MODIFY tos VARCHAR2(10 CHAR) + +/ + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio + +/ + +ALTER TABLE ps_endpoints DROP COLUMN cos_video + +/ + +ALTER TABLE ps_transports DROP COLUMN cos + +/ + +ALTER TABLE ps_endpoints ADD cos_audio INTEGER + +/ + +ALTER TABLE ps_endpoints ADD cos_video INTEGER + +/ + +ALTER TABLE ps_transports ADD cos INTEGER + +/ + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD message_context VARCHAR2(40 CHAR) + +/ + +ALTER TABLE ps_contacts ADD user_agent VARCHAR2(40 CHAR) + +/ + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals MODIFY user_agent VARCHAR2(255 CHAR) + +/ + +ALTER TABLE ps_contacts MODIFY id VARCHAR2(255 CHAR) + +/ + +ALTER TABLE ps_contacts MODIFY uri VARCHAR2(255 CHAR) + +/ + +ALTER TABLE ps_contacts MODIFY user_agent VARCHAR2(255 CHAR) + +/ + +ALTER TABLE ps_registrations MODIFY client_uri VARCHAR2(255 CHAR) + +/ + +ALTER TABLE ps_registrations MODIFY server_uri VARCHAR2(255 CHAR) + +/ + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR2(40 CHAR) NOT NULL, + packet VARCHAR2(2048 CHAR), + src_name VARCHAR2(128 CHAR), + src_port INTEGER, + transport_key VARCHAR2(64 CHAR), + local_name VARCHAR2(128 CHAR), + local_port INTEGER, + cseq INTEGER, + tag VARCHAR2(128 CHAR), + endpoint VARCHAR2(40 CHAR), + expires INTEGER, + UNIQUE (id) +) + +/ + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id) + +/ + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD force_avp VARCHAR(3 CHAR) + +/ + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (force_avp IN ('yes', 'no')) + +/ + +ALTER TABLE ps_endpoints ADD media_use_received_transport VARCHAR(3 CHAR) + +/ + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_use_received_transport IN ('yes', 'no')) + +/ + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD accountcode VARCHAR2(20 CHAR) + +/ + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers MODIFY useragent VARCHAR2(255 CHAR) + +/ + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid + +/ + +ALTER TABLE queue_members ADD uniqueid INTEGER NOT NULL + +/ + +ALTER TABLE queue_members ADD UNIQUE (uniqueid) + +/ + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR2(80 CHAR) NOT NULL, + time VARCHAR2(32 CHAR) NOT NULL, + min_penalty VARCHAR2(32 CHAR) NOT NULL, + max_penalty VARCHAR2(32 CHAR) NOT NULL +) + +/ + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports MODIFY verifiy_server VARCHAR(3 CHAR) + +/ + +ALTER TABLE ps_transports RENAME COLUMN verifiy_server TO verify_server + +/ + +ALTER TABLE ps_transports ADD CONSTRAINT yesno_values CHECK (verifiy_server IN ('yes', 'no')) + +/ + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +ALTER TABLE sippeers DROP CONSTRAINT sip_directmedia_values + +/ + +ALTER TABLE sippeers MODIFY directmedia VARCHAR(8 CHAR) + +/ + +ALTER TABLE sippeers ADD CONSTRAINT sip_directmedia_values_v2 CHECK (directmedia IN ('yes', 'no', 'nonat', 'update', 'outgoing')) + +/ + +-- Running upgrade 10aedae86a32 -> eb88a14f2a + +ALTER TABLE ps_endpoints ADD media_encryption_optimistic VARCHAR(3 CHAR) + +/ + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_optimistic IN ('yes', 'no')) + +/ + +INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a') + +/ + +COMMIT + +/ + diff --git a/contrib/realtime/oracle/oracle_voicemail.sql b/contrib/realtime/oracle/oracle_voicemail.sql new file mode 100644 index 0000000000..2c663e76e5 --- /dev/null +++ b/contrib/realtime/oracle/oracle_voicemail.sql @@ -0,0 +1,52 @@ +SET TRANSACTION READ WRITE + +/ + +CREATE TABLE alembic_version ( + version_num VARCHAR2(32 CHAR) NOT NULL +) + +/ + +-- Running upgrade None -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR2(255 CHAR) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR2(80 CHAR), + macrocontext VARCHAR2(80 CHAR), + callerid VARCHAR2(80 CHAR), + origtime INTEGER, + duration INTEGER, + recording BLOB, + flag VARCHAR2(30 CHAR), + category VARCHAR2(30 CHAR), + mailboxuser VARCHAR2(30 CHAR), + mailboxcontext VARCHAR2(30 CHAR), + msg_id VARCHAR2(40 CHAR) +) + +/ + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum) + +/ + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir) + +/ + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages MODIFY recording BLOB + +/ + +INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5') + +/ + +COMMIT + +/ + diff --git a/contrib/realtime/postgresql/postgresql_cdr.sql b/contrib/realtime/postgresql/postgresql_cdr.sql new file mode 100644 index 0000000000..8aa1d97374 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_cdr.sql @@ -0,0 +1,36 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR(20), + src VARCHAR(80), + dst VARCHAR(80), + dcontext VARCHAR(80), + clid VARCHAR(80), + channel VARCHAR(80), + dstchannel VARCHAR(80), + lastapp VARCHAR(80), + lastdata VARCHAR(80), + start TIMESTAMP WITHOUT TIME ZONE, + answer TIMESTAMP WITHOUT TIME ZONE, + "end" TIMESTAMP WITHOUT TIME ZONE, + duration INTEGER, + billsec INTEGER, + disposition VARCHAR(45), + amaflags VARCHAR(45), + userfield VARCHAR(256), + uniqueid VARCHAR(150), + linkedid VARCHAR(150), + peeraccount VARCHAR(20), + sequence INTEGER +); + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d'); + +COMMIT; + diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql new file mode 100644 index 0000000000..cb71ba6766 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_config.sql @@ -0,0 +1,739 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> 4da0c5f79a9c + +CREATE TYPE type_values AS ENUM ('friend','user','peer'); + +CREATE TYPE sip_transport_values AS ENUM ('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp'); + +CREATE TYPE sip_dtmfmode_values AS ENUM ('rfc2833','info','shortinfo','inband','auto'); + +CREATE TYPE sip_directmedia_values AS ENUM ('yes','no','nonat','update'); + +CREATE TYPE yes_no_values AS ENUM ('yes','no'); + +CREATE TYPE sip_progressinband_values AS ENUM ('yes','no','never'); + +CREATE TYPE sip_session_timers_values AS ENUM ('accept','refuse','originate'); + +CREATE TYPE sip_session_refresher_values AS ENUM ('uac','uas'); + +CREATE TYPE sip_callingpres_values AS ENUM ('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib'); + +CREATE TABLE sippeers ( + id SERIAL NOT NULL, + name VARCHAR(40) NOT NULL, + ipaddr VARCHAR(45), + port INTEGER, + regseconds INTEGER, + defaultuser VARCHAR(40), + fullcontact VARCHAR(80), + regserver VARCHAR(20), + useragent VARCHAR(20), + lastms INTEGER, + host VARCHAR(40), + type type_values, + context VARCHAR(40), + permit VARCHAR(95), + deny VARCHAR(95), + secret VARCHAR(40), + md5secret VARCHAR(40), + remotesecret VARCHAR(40), + transport sip_transport_values, + dtmfmode sip_dtmfmode_values, + directmedia sip_directmedia_values, + nat VARCHAR(29), + callgroup VARCHAR(40), + pickupgroup VARCHAR(40), + language VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + insecure VARCHAR(40), + trustrpid yes_no_values, + progressinband sip_progressinband_values, + promiscredir yes_no_values, + useclientcode yes_no_values, + accountcode VARCHAR(40), + setvar VARCHAR(200), + callerid VARCHAR(40), + amaflags VARCHAR(40), + callcounter yes_no_values, + busylevel INTEGER, + allowoverlap yes_no_values, + allowsubscribe yes_no_values, + videosupport yes_no_values, + maxcallbitrate INTEGER, + rfc2833compensate yes_no_values, + mailbox VARCHAR(40), + "session-timers" sip_session_timers_values, + "session-expires" INTEGER, + "session-minse" INTEGER, + "session-refresher" sip_session_refresher_values, + t38pt_usertpsource VARCHAR(40), + regexten VARCHAR(40), + fromdomain VARCHAR(40), + fromuser VARCHAR(40), + qualify VARCHAR(40), + defaultip VARCHAR(45), + rtptimeout INTEGER, + rtpholdtimeout INTEGER, + sendrpid yes_no_values, + outboundproxy VARCHAR(40), + callbackextension VARCHAR(40), + timert1 INTEGER, + timerb INTEGER, + qualifyfreq INTEGER, + constantssrc yes_no_values, + contactpermit VARCHAR(95), + contactdeny VARCHAR(95), + usereqphone yes_no_values, + textsupport yes_no_values, + faxdetect yes_no_values, + buggymwi yes_no_values, + auth VARCHAR(40), + fullname VARCHAR(40), + trunkname VARCHAR(40), + cid_number VARCHAR(40), + callingpres sip_callingpres_values, + mohinterpret VARCHAR(40), + mohsuggest VARCHAR(40), + parkinglot VARCHAR(40), + hasvoicemail yes_no_values, + subscribemwi yes_no_values, + vmexten VARCHAR(40), + autoframing yes_no_values, + rtpkeepalive INTEGER, + "call-limit" INTEGER, + g726nonstandard yes_no_values, + ignoresdpversion yes_no_values, + allowtransfer yes_no_values, + dynamic yes_no_values, + path VARCHAR(256), + supportpath yes_no_values, + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX sippeers_name ON sippeers (name); + +CREATE INDEX sippeers_name_host ON sippeers (name, host); + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port); + +CREATE INDEX sippeers_host_port ON sippeers (host, port); + +CREATE TYPE iax_requirecalltoken_values AS ENUM ('yes','no','auto'); + +CREATE TYPE iax_encryption_values AS ENUM ('yes','no','aes128'); + +CREATE TYPE iax_transfer_values AS ENUM ('yes','no','mediaonly'); + +CREATE TABLE iaxfriends ( + id SERIAL NOT NULL, + name VARCHAR(40) NOT NULL, + type type_values, + username VARCHAR(40), + mailbox VARCHAR(40), + secret VARCHAR(40), + dbsecret VARCHAR(40), + context VARCHAR(40), + regcontext VARCHAR(40), + host VARCHAR(40), + ipaddr VARCHAR(40), + port INTEGER, + defaultip VARCHAR(20), + sourceaddress VARCHAR(20), + mask VARCHAR(20), + regexten VARCHAR(40), + regseconds INTEGER, + accountcode VARCHAR(20), + mohinterpret VARCHAR(20), + mohsuggest VARCHAR(20), + inkeys VARCHAR(40), + outkeys VARCHAR(40), + language VARCHAR(10), + callerid VARCHAR(100), + cid_number VARCHAR(40), + sendani yes_no_values, + fullname VARCHAR(40), + trunk yes_no_values, + auth VARCHAR(20), + maxauthreq INTEGER, + requirecalltoken iax_requirecalltoken_values, + encryption iax_encryption_values, + transfer iax_transfer_values, + jitterbuffer yes_no_values, + forcejitterbuffer yes_no_values, + disallow VARCHAR(200), + allow VARCHAR(200), + codecpriority VARCHAR(40), + qualify VARCHAR(10), + qualifysmoothing yes_no_values, + qualifyfreqok VARCHAR(10), + qualifyfreqnotok VARCHAR(10), + timezone VARCHAR(20), + adsi yes_no_values, + amaflags VARCHAR(20), + setvar VARCHAR(200), + PRIMARY KEY (id), + UNIQUE (name) +); + +CREATE INDEX iaxfriends_name ON iaxfriends (name); + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host); + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port); + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port); + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port); + +CREATE TABLE voicemail ( + uniqueid SERIAL NOT NULL, + context VARCHAR(80) NOT NULL, + mailbox VARCHAR(80) NOT NULL, + password VARCHAR(80) NOT NULL, + fullname VARCHAR(80), + alias VARCHAR(80), + email VARCHAR(80), + pager VARCHAR(80), + attach yes_no_values, + attachfmt VARCHAR(10), + serveremail VARCHAR(80), + language VARCHAR(20), + tz VARCHAR(30), + deletevoicemail yes_no_values, + saycid yes_no_values, + sendvoicemail yes_no_values, + review yes_no_values, + tempgreetwarn yes_no_values, + operator yes_no_values, + envelope yes_no_values, + sayduration INTEGER, + forcename yes_no_values, + forcegreetings yes_no_values, + callback VARCHAR(80), + dialout VARCHAR(80), + exitcontext VARCHAR(80), + maxmsg INTEGER, + volgain NUMERIC(5, 2), + imapuser VARCHAR(80), + imappassword VARCHAR(80), + imapserver VARCHAR(80), + imapport VARCHAR(8), + imapflags VARCHAR(80), + stamp TIMESTAMP WITHOUT TIME ZONE, + PRIMARY KEY (uniqueid) +); + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox); + +CREATE INDEX voicemail_context ON voicemail (context); + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context); + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser); + +CREATE TABLE meetme ( + bookid SERIAL NOT NULL, + confno VARCHAR(80) NOT NULL, + starttime TIMESTAMP WITHOUT TIME ZONE, + endtime TIMESTAMP WITHOUT TIME ZONE, + pin VARCHAR(20), + adminpin VARCHAR(20), + opts VARCHAR(20), + adminopts VARCHAR(20), + recordingfilename VARCHAR(80), + recordingformat VARCHAR(10), + maxusers INTEGER, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +); + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime); + +CREATE TYPE moh_mode_values AS ENUM ('custom','files','mp3nb','quietmp3nb','quietmp3'); + +CREATE TABLE musiconhold ( + name VARCHAR(80) NOT NULL, + mode moh_mode_values, + directory VARCHAR(255), + application VARCHAR(255), + digit VARCHAR(1), + sort VARCHAR(10), + format VARCHAR(10), + stamp TIMESTAMP WITHOUT TIME ZONE, + PRIMARY KEY (name) +); + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TYPE yesno_values AS ENUM ('yes','no'); + +CREATE TYPE pjsip_connected_line_method_values AS ENUM ('invite','reinvite','update'); + +CREATE TYPE pjsip_direct_media_glare_mitigation_values AS ENUM ('none','outgoing','incoming'); + +CREATE TYPE pjsip_dtmf_mode_values AS ENUM ('rfc4733','inband','info'); + +CREATE TYPE pjsip_identify_by_values AS ENUM ('username'); + +CREATE TYPE pjsip_timer_values AS ENUM ('forced','no','required','yes'); + +CREATE TYPE pjsip_cid_privacy_values AS ENUM ('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable'); + +CREATE TYPE pjsip_100rel_values AS ENUM ('no','required','yes'); + +CREATE TYPE pjsip_media_encryption_values AS ENUM ('no','sdes','dtls'); + +CREATE TYPE pjsip_t38udptl_ec_values AS ENUM ('none','fec','redundancy'); + +CREATE TYPE pjsip_dtls_setup_values AS ENUM ('active','passive','actpass'); + +CREATE TABLE ps_endpoints ( + id VARCHAR(40) NOT NULL, + transport VARCHAR(40), + aors VARCHAR(200), + auth VARCHAR(40), + context VARCHAR(40), + disallow VARCHAR(200), + allow VARCHAR(200), + direct_media yesno_values, + connected_line_method pjsip_connected_line_method_values, + direct_media_method pjsip_connected_line_method_values, + direct_media_glare_mitigation pjsip_direct_media_glare_mitigation_values, + disable_direct_media_on_nat yesno_values, + dtmf_mode pjsip_dtmf_mode_values, + external_media_address VARCHAR(40), + force_rport yesno_values, + ice_support yesno_values, + identify_by pjsip_identify_by_values, + mailboxes VARCHAR(40), + moh_suggest VARCHAR(40), + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + rewrite_contact yesno_values, + rtp_ipv6 yesno_values, + rtp_symmetric yesno_values, + send_diversion yesno_values, + send_pai yesno_values, + send_rpid yesno_values, + timers_min_se INTEGER, + timers pjsip_timer_values, + timers_sess_expires INTEGER, + callerid VARCHAR(40), + callerid_privacy pjsip_cid_privacy_values, + callerid_tag VARCHAR(40), + "100rel" pjsip_100rel_values, + aggregate_mwi yesno_values, + trust_id_inbound yesno_values, + trust_id_outbound yesno_values, + use_ptime yesno_values, + use_avpf yesno_values, + media_encryption pjsip_media_encryption_values, + inband_progress yesno_values, + call_group VARCHAR(40), + pickup_group VARCHAR(40), + named_call_group VARCHAR(40), + named_pickup_group VARCHAR(40), + device_state_busy_at INTEGER, + fax_detect yesno_values, + t38_udptl yesno_values, + t38_udptl_ec pjsip_t38udptl_ec_values, + t38_udptl_maxdatagram INTEGER, + t38_udptl_nat yesno_values, + t38_udptl_ipv6 yesno_values, + tone_zone VARCHAR(40), + language VARCHAR(40), + one_touch_recording yesno_values, + record_on_feature VARCHAR(40), + record_off_feature VARCHAR(40), + rtp_engine VARCHAR(40), + allow_transfer yesno_values, + allow_subscribe yesno_values, + sdp_owner VARCHAR(40), + sdp_session VARCHAR(40), + tos_audio INTEGER, + tos_video INTEGER, + cos_audio INTEGER, + cos_video INTEGER, + sub_min_expiry INTEGER, + from_domain VARCHAR(40), + from_user VARCHAR(40), + mwi_fromuser VARCHAR(40), + dtls_verify VARCHAR(40), + dtls_rekey VARCHAR(40), + dtls_cert_file VARCHAR(200), + dtls_private_key VARCHAR(200), + dtls_cipher VARCHAR(200), + dtls_ca_file VARCHAR(200), + dtls_ca_path VARCHAR(200), + dtls_setup pjsip_dtls_setup_values, + srtp_tag_32 yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id); + +CREATE TYPE pjsip_auth_type_values AS ENUM ('md5','userpass'); + +CREATE TABLE ps_auths ( + id VARCHAR(40) NOT NULL, + auth_type pjsip_auth_type_values, + nonce_lifetime INTEGER, + md5_cred VARCHAR(40), + password VARCHAR(80), + realm VARCHAR(40), + username VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_auths_id ON ps_auths (id); + +CREATE TABLE ps_aors ( + id VARCHAR(40) NOT NULL, + contact VARCHAR(40), + default_expiration INTEGER, + mailboxes VARCHAR(80), + max_contacts INTEGER, + minimum_expiration INTEGER, + remove_existing yesno_values, + qualify_frequency INTEGER, + authenticate_qualify yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_aors_id ON ps_aors (id); + +CREATE TABLE ps_contacts ( + id VARCHAR(40) NOT NULL, + uri VARCHAR(40), + expiration_time VARCHAR(40), + qualify_frequency INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_contacts_id ON ps_contacts (id); + +CREATE TABLE ps_domain_aliases ( + id VARCHAR(40) NOT NULL, + domain VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id); + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40), + match VARCHAR(80), + UNIQUE (id) +); + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id); + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id BIGSERIAL NOT NULL, + context VARCHAR(40) NOT NULL, + exten VARCHAR(40) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR(40) NOT NULL, + appdata VARCHAR(256) NOT NULL, + PRIMARY KEY (id, context, exten, priority), + UNIQUE (id) +); + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TYPE pjsip_redirect_method_values AS ENUM ('user','uri_core','uri_pjsip'); + +CREATE TABLE ps_systems ( + id VARCHAR(40) NOT NULL, + timer_t1 INTEGER, + timer_b INTEGER, + compact_headers yesno_values, + threadpool_initial_size INTEGER, + threadpool_auto_increment INTEGER, + threadpool_idle_timeout INTEGER, + threadpool_max_size INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_systems_id ON ps_systems (id); + +CREATE TABLE ps_globals ( + id VARCHAR(40) NOT NULL, + max_forwards INTEGER, + user_agent VARCHAR(40), + default_outbound_endpoint VARCHAR(40), + UNIQUE (id) +); + +CREATE INDEX ps_globals_id ON ps_globals (id); + +CREATE TYPE pjsip_transport_method_values AS ENUM ('default','unspecified','tlsv1','sslv2','sslv3','sslv23'); + +CREATE TYPE pjsip_transport_protocol_values AS ENUM ('udp','tcp','tls','ws','wss'); + +CREATE TABLE ps_transports ( + id VARCHAR(40) NOT NULL, + async_operations INTEGER, + bind VARCHAR(40), + ca_list_file VARCHAR(200), + cert_file VARCHAR(200), + cipher VARCHAR(200), + domain VARCHAR(40), + external_media_address VARCHAR(40), + external_signaling_address VARCHAR(40), + external_signaling_port INTEGER, + method pjsip_transport_method_values, + local_net VARCHAR(40), + password VARCHAR(40), + priv_key_file VARCHAR(200), + protocol pjsip_transport_protocol_values, + require_client_cert yesno_values, + verify_client yesno_values, + verifiy_server yesno_values, + tos yesno_values, + cos yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_transports_id ON ps_transports (id); + +CREATE TABLE ps_registrations ( + id VARCHAR(40) NOT NULL, + auth_rejection_permanent yesno_values, + client_uri VARCHAR(40), + contact_user VARCHAR(40), + expiration INTEGER, + max_retries INTEGER, + outbound_auth VARCHAR(40), + outbound_proxy VARCHAR(40), + retry_interval INTEGER, + forbidden_retry_interval INTEGER, + server_uri VARCHAR(40), + transport VARCHAR(40), + support_path yesno_values, + UNIQUE (id) +); + +CREATE INDEX ps_registrations_id ON ps_registrations (id); + +ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40); + +ALTER TABLE ps_endpoints ADD COLUMN redirect_method pjsip_redirect_method_values; + +ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT; + +ALTER TABLE ps_endpoints RENAME mwi_fromuser TO mwi_from_user; + +ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN path TEXT; + +ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER; + +ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40); + +ALTER TABLE ps_aors ADD COLUMN support_path yesno_values; + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40); + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TYPE queue_autopause_values AS ENUM ('yes','no','all'); + +CREATE TYPE queue_strategy_values AS ENUM ('ringall','leastrecent','fewestcalls','random','rrmemory','linear','wrandom','rrordered'); + +CREATE TABLE queues ( + name VARCHAR(128) NOT NULL, + musiconhold VARCHAR(128), + announce VARCHAR(128), + context VARCHAR(128), + timeout INTEGER, + ringinuse yesno_values, + setinterfacevar yesno_values, + setqueuevar yesno_values, + setqueueentryvar yesno_values, + monitor_format VARCHAR(8), + membermacro VARCHAR(512), + membergosub VARCHAR(512), + queue_youarenext VARCHAR(128), + queue_thereare VARCHAR(128), + queue_callswaiting VARCHAR(128), + queue_quantity1 VARCHAR(128), + queue_quantity2 VARCHAR(128), + queue_holdtime VARCHAR(128), + queue_minutes VARCHAR(128), + queue_minute VARCHAR(128), + queue_seconds VARCHAR(128), + queue_thankyou VARCHAR(128), + queue_callerannounce VARCHAR(128), + queue_reporthold VARCHAR(128), + announce_frequency INTEGER, + announce_to_first_user yesno_values, + min_announce_frequency INTEGER, + announce_round_seconds INTEGER, + announce_holdtime VARCHAR(128), + announce_position VARCHAR(128), + announce_position_limit INTEGER, + periodic_announce VARCHAR(50), + periodic_announce_frequency INTEGER, + relative_periodic_announce yesno_values, + random_periodic_announce yesno_values, + retry INTEGER, + wrapuptime INTEGER, + penaltymemberslimit INTEGER, + autofill yesno_values, + monitor_type VARCHAR(128), + autopause queue_autopause_values, + autopausedelay INTEGER, + autopausebusy yesno_values, + autopauseunavail yesno_values, + maxlen INTEGER, + servicelevel INTEGER, + strategy queue_strategy_values, + joinempty VARCHAR(128), + leavewhenempty VARCHAR(128), + reportholdtime yesno_values, + memberdelay INTEGER, + weight INTEGER, + timeoutrestart yesno_values, + defaultrule VARCHAR(128), + timeoutpriority VARCHAR(128), + PRIMARY KEY (name) +); + +CREATE TABLE queue_members ( + queue_name VARCHAR(80) NOT NULL, + interface VARCHAR(80) NOT NULL, + uniqueid VARCHAR(80) NOT NULL, + membername VARCHAR(80), + state_interface VARCHAR(80), + penalty INTEGER, + paused INTEGER, + PRIMARY KEY (queue_name, interface) +); + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints ALTER COLUMN tos_audio TYPE VARCHAR(10); + +ALTER TABLE ps_endpoints ALTER COLUMN tos_video TYPE VARCHAR(10); + +ALTER TABLE ps_transports ALTER COLUMN tos TYPE VARCHAR(10); + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio; + +ALTER TABLE ps_endpoints DROP COLUMN cos_video; + +ALTER TABLE ps_transports DROP COLUMN cos; + +ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER; + +ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER; + +ALTER TABLE ps_transports ADD COLUMN cos INTEGER; + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40); + +ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40); + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals ALTER COLUMN user_agent TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN id TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN uri TYPE VARCHAR(255); + +ALTER TABLE ps_contacts ALTER COLUMN user_agent TYPE VARCHAR(255); + +ALTER TABLE ps_registrations ALTER COLUMN client_uri TYPE VARCHAR(255); + +ALTER TABLE ps_registrations ALTER COLUMN server_uri TYPE VARCHAR(255); + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR(40) NOT NULL, + packet VARCHAR(2048), + src_name VARCHAR(128), + src_port INTEGER, + transport_key VARCHAR(64), + local_name VARCHAR(128), + local_port INTEGER, + cseq INTEGER, + tag VARCHAR(128), + endpoint VARCHAR(40), + expires INTEGER, + UNIQUE (id) +); + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id); + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD COLUMN force_avp yesno_values; + +ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport yesno_values; + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20); + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers ALTER COLUMN useragent TYPE VARCHAR(255); + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid; + +ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL; + +ALTER TABLE queue_members ADD UNIQUE (uniqueid); + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR(80) NOT NULL, + time VARCHAR(32) NOT NULL, + min_penalty VARCHAR(32) NOT NULL, + max_penalty VARCHAR(32) NOT NULL +); + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports ALTER COLUMN verifiy_server TYPE yesno_values; + +ALTER TABLE ps_transports RENAME verifiy_server TO verify_server; + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +CREATE TYPE sip_directmedia_values_v2 AS ENUM ('yes','no','nonat','update','outgoing'); + +ALTER TABLE sippeers ALTER COLUMN directmedia TYPE sip_directmedia_values_v2 USING directmedia::text::sip_directmedia_values_v2; + +DROP TYPE sip_directmedia_values; + +-- Running upgrade 10aedae86a32 -> eb88a14f2a + +ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values; + +INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a'); + +COMMIT; + diff --git a/contrib/realtime/postgresql/postgresql_voicemail.sql b/contrib/realtime/postgresql/postgresql_voicemail.sql new file mode 100644 index 0000000000..20caf394a4 --- /dev/null +++ b/contrib/realtime/postgresql/postgresql_voicemail.sql @@ -0,0 +1,36 @@ +BEGIN; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +-- Running upgrade None -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR(255) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR(80), + macrocontext VARCHAR(80), + callerid VARCHAR(80), + origtime INTEGER, + duration INTEGER, + recording BYTEA, + flag VARCHAR(30), + category VARCHAR(30), + mailboxuser VARCHAR(30), + mailboxcontext VARCHAR(30), + msg_id VARCHAR(40) +); + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum); + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir); + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA; + +INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5'); + +COMMIT; + diff --git a/contrib/realtime/sqlserver/mssql_cdr.sql b/contrib/realtime/sqlserver/mssql_cdr.sql new file mode 100644 index 0000000000..dcd433f0ce --- /dev/null +++ b/contrib/realtime/sqlserver/mssql_cdr.sql @@ -0,0 +1,42 @@ +BEGIN TRANSACTION; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +GO + +-- Running upgrade None -> 210693f3123d + +CREATE TABLE cdr ( + accountcode VARCHAR(20) NULL, + src VARCHAR(80) NULL, + dst VARCHAR(80) NULL, + dcontext VARCHAR(80) NULL, + clid VARCHAR(80) NULL, + channel VARCHAR(80) NULL, + dstchannel VARCHAR(80) NULL, + lastapp VARCHAR(80) NULL, + lastdata VARCHAR(80) NULL, + start DATETIME NULL, + answer DATETIME NULL, + [end] DATETIME NULL, + duration INTEGER NULL, + billsec INTEGER NULL, + disposition VARCHAR(45) NULL, + amaflags VARCHAR(45) NULL, + userfield VARCHAR(256) NULL, + uniqueid VARCHAR(150) NULL, + linkedid VARCHAR(150) NULL, + peeraccount VARCHAR(20) NULL, + sequence INTEGER NULL +); + +GO + +INSERT INTO alembic_version (version_num) VALUES ('210693f3123d'); + +GO + +COMMIT; + diff --git a/contrib/realtime/sqlserver/mssql_config.sql b/contrib/realtime/sqlserver/mssql_config.sql new file mode 100644 index 0000000000..4834bece58 --- /dev/null +++ b/contrib/realtime/sqlserver/mssql_config.sql @@ -0,0 +1,990 @@ +BEGIN TRANSACTION; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +GO + +-- Running upgrade None -> 4da0c5f79a9c + +CREATE TABLE sippeers ( + id INTEGER NOT NULL IDENTITY(1,1), + name VARCHAR(40) NOT NULL, + ipaddr VARCHAR(45) NULL, + port INTEGER NULL, + regseconds INTEGER NULL, + defaultuser VARCHAR(40) NULL, + fullcontact VARCHAR(80) NULL, + regserver VARCHAR(20) NULL, + useragent VARCHAR(20) NULL, + lastms INTEGER NULL, + host VARCHAR(40) NULL, + type VARCHAR(6) NULL, + context VARCHAR(40) NULL, + permit VARCHAR(95) NULL, + [deny] VARCHAR(95) NULL, + secret VARCHAR(40) NULL, + md5secret VARCHAR(40) NULL, + remotesecret VARCHAR(40) NULL, + transport VARCHAR(7) NULL, + dtmfmode VARCHAR(9) NULL, + directmedia VARCHAR(6) NULL, + nat VARCHAR(29) NULL, + callgroup VARCHAR(40) NULL, + pickupgroup VARCHAR(40) NULL, + language VARCHAR(40) NULL, + disallow VARCHAR(200) NULL, + allow VARCHAR(200) NULL, + insecure VARCHAR(40) NULL, + trustrpid VARCHAR(3) NULL, + progressinband VARCHAR(5) NULL, + promiscredir VARCHAR(3) NULL, + useclientcode VARCHAR(3) NULL, + accountcode VARCHAR(40) NULL, + setvar VARCHAR(200) NULL, + callerid VARCHAR(40) NULL, + amaflags VARCHAR(40) NULL, + callcounter VARCHAR(3) NULL, + busylevel INTEGER NULL, + allowoverlap VARCHAR(3) NULL, + allowsubscribe VARCHAR(3) NULL, + videosupport VARCHAR(3) NULL, + maxcallbitrate INTEGER NULL, + rfc2833compensate VARCHAR(3) NULL, + mailbox VARCHAR(40) NULL, + [session-timers] VARCHAR(9) NULL, + [session-expires] INTEGER NULL, + [session-minse] INTEGER NULL, + [session-refresher] VARCHAR(3) NULL, + t38pt_usertpsource VARCHAR(40) NULL, + regexten VARCHAR(40) NULL, + fromdomain VARCHAR(40) NULL, + fromuser VARCHAR(40) NULL, + qualify VARCHAR(40) NULL, + defaultip VARCHAR(45) NULL, + rtptimeout INTEGER NULL, + rtpholdtimeout INTEGER NULL, + sendrpid VARCHAR(3) NULL, + outboundproxy VARCHAR(40) NULL, + callbackextension VARCHAR(40) NULL, + timert1 INTEGER NULL, + timerb INTEGER NULL, + qualifyfreq INTEGER NULL, + constantssrc VARCHAR(3) NULL, + contactpermit VARCHAR(95) NULL, + contactdeny VARCHAR(95) NULL, + usereqphone VARCHAR(3) NULL, + textsupport VARCHAR(3) NULL, + faxdetect VARCHAR(3) NULL, + buggymwi VARCHAR(3) NULL, + auth VARCHAR(40) NULL, + fullname VARCHAR(40) NULL, + trunkname VARCHAR(40) NULL, + cid_number VARCHAR(40) NULL, + callingpres VARCHAR(21) NULL, + mohinterpret VARCHAR(40) NULL, + mohsuggest VARCHAR(40) NULL, + parkinglot VARCHAR(40) NULL, + hasvoicemail VARCHAR(3) NULL, + subscribemwi VARCHAR(3) NULL, + vmexten VARCHAR(40) NULL, + autoframing VARCHAR(3) NULL, + rtpkeepalive INTEGER NULL, + [call-limit] INTEGER NULL, + g726nonstandard VARCHAR(3) NULL, + ignoresdpversion VARCHAR(3) NULL, + allowtransfer VARCHAR(3) NULL, + dynamic VARCHAR(3) NULL, + path VARCHAR(256) NULL, + supportpath VARCHAR(3) NULL, + PRIMARY KEY (id), + UNIQUE (name), + CONSTRAINT type_values CHECK (type IN ('friend', 'user', 'peer')), + CONSTRAINT sip_transport_values CHECK (transport IN ('udp', 'tcp', 'tls', 'ws', 'wss', 'udp,tcp', 'tcp,udp')), + CONSTRAINT sip_dtmfmode_values CHECK (dtmfmode IN ('rfc2833', 'info', 'shortinfo', 'inband', 'auto')), + CONSTRAINT sip_directmedia_values CHECK (directmedia IN ('yes', 'no', 'nonat', 'update')), + CONSTRAINT yes_no_values CHECK (trustrpid IN ('yes', 'no')), + CONSTRAINT sip_progressinband_values CHECK (progressinband IN ('yes', 'no', 'never')), + CONSTRAINT yes_no_values CHECK (promiscredir IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (useclientcode IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (callcounter IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowoverlap IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowsubscribe IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (videosupport IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (rfc2833compensate IN ('yes', 'no')), + CONSTRAINT sip_session_timers_values CHECK ([session-timers] IN ('accept', 'refuse', 'originate')), + CONSTRAINT sip_session_refresher_values CHECK ([session-refresher] IN ('uac', 'uas')), + CONSTRAINT yes_no_values CHECK (sendrpid IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (constantssrc IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (usereqphone IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (textsupport IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (faxdetect IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (buggymwi IN ('yes', 'no')), + CONSTRAINT sip_callingpres_values CHECK (callingpres IN ('allowed_not_screened', 'allowed_passed_screen', 'allowed_failed_screen', 'allowed', 'prohib_not_screened', 'prohib_passed_screen', 'prohib_failed_screen', 'prohib')), + CONSTRAINT yes_no_values CHECK (hasvoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (subscribemwi IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (autoframing IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (g726nonstandard IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (ignoresdpversion IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (allowtransfer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (dynamic IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (supportpath IN ('yes', 'no')) +); + +GO + +CREATE INDEX sippeers_name ON sippeers (name); + +GO + +CREATE INDEX sippeers_name_host ON sippeers (name, host); + +GO + +CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port); + +GO + +CREATE INDEX sippeers_host_port ON sippeers (host, port); + +GO + +CREATE TABLE iaxfriends ( + id INTEGER NOT NULL IDENTITY(1,1), + name VARCHAR(40) NOT NULL, + type VARCHAR(6) NULL, + username VARCHAR(40) NULL, + mailbox VARCHAR(40) NULL, + secret VARCHAR(40) NULL, + dbsecret VARCHAR(40) NULL, + context VARCHAR(40) NULL, + regcontext VARCHAR(40) NULL, + host VARCHAR(40) NULL, + ipaddr VARCHAR(40) NULL, + port INTEGER NULL, + defaultip VARCHAR(20) NULL, + sourceaddress VARCHAR(20) NULL, + mask VARCHAR(20) NULL, + regexten VARCHAR(40) NULL, + regseconds INTEGER NULL, + accountcode VARCHAR(20) NULL, + mohinterpret VARCHAR(20) NULL, + mohsuggest VARCHAR(20) NULL, + inkeys VARCHAR(40) NULL, + outkeys VARCHAR(40) NULL, + language VARCHAR(10) NULL, + callerid VARCHAR(100) NULL, + cid_number VARCHAR(40) NULL, + sendani VARCHAR(3) NULL, + fullname VARCHAR(40) NULL, + trunk VARCHAR(3) NULL, + auth VARCHAR(20) NULL, + maxauthreq INTEGER NULL, + requirecalltoken VARCHAR(4) NULL, + encryption VARCHAR(6) NULL, + transfer VARCHAR(9) NULL, + jitterbuffer VARCHAR(3) NULL, + forcejitterbuffer VARCHAR(3) NULL, + disallow VARCHAR(200) NULL, + allow VARCHAR(200) NULL, + codecpriority VARCHAR(40) NULL, + qualify VARCHAR(10) NULL, + qualifysmoothing VARCHAR(3) NULL, + qualifyfreqok VARCHAR(10) NULL, + qualifyfreqnotok VARCHAR(10) NULL, + timezone VARCHAR(20) NULL, + adsi VARCHAR(3) NULL, + amaflags VARCHAR(20) NULL, + setvar VARCHAR(200) NULL, + PRIMARY KEY (id), + UNIQUE (name), + CONSTRAINT type_values CHECK (type IN ('friend', 'user', 'peer')), + CONSTRAINT yes_no_values CHECK (sendani IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (trunk IN ('yes', 'no')), + CONSTRAINT iax_requirecalltoken_values CHECK (requirecalltoken IN ('yes', 'no', 'auto')), + CONSTRAINT iax_encryption_values CHECK (encryption IN ('yes', 'no', 'aes128')), + CONSTRAINT iax_transfer_values CHECK (transfer IN ('yes', 'no', 'mediaonly')), + CONSTRAINT yes_no_values CHECK (jitterbuffer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcejitterbuffer IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (qualifysmoothing IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (adsi IN ('yes', 'no')) +); + +GO + +CREATE INDEX iaxfriends_name ON iaxfriends (name); + +GO + +CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host); + +GO + +CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port); + +GO + +CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port); + +GO + +CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port); + +GO + +CREATE TABLE voicemail ( + uniqueid INTEGER NOT NULL IDENTITY(1,1), + context VARCHAR(80) NOT NULL, + mailbox VARCHAR(80) NOT NULL, + password VARCHAR(80) NOT NULL, + fullname VARCHAR(80) NULL, + alias VARCHAR(80) NULL, + email VARCHAR(80) NULL, + pager VARCHAR(80) NULL, + attach VARCHAR(3) NULL, + attachfmt VARCHAR(10) NULL, + serveremail VARCHAR(80) NULL, + language VARCHAR(20) NULL, + tz VARCHAR(30) NULL, + deletevoicemail VARCHAR(3) NULL, + saycid VARCHAR(3) NULL, + sendvoicemail VARCHAR(3) NULL, + review VARCHAR(3) NULL, + tempgreetwarn VARCHAR(3) NULL, + operator VARCHAR(3) NULL, + envelope VARCHAR(3) NULL, + sayduration INTEGER NULL, + forcename VARCHAR(3) NULL, + forcegreetings VARCHAR(3) NULL, + callback VARCHAR(80) NULL, + dialout VARCHAR(80) NULL, + exitcontext VARCHAR(80) NULL, + maxmsg INTEGER NULL, + volgain NUMERIC(5, 2) NULL, + imapuser VARCHAR(80) NULL, + imappassword VARCHAR(80) NULL, + imapserver VARCHAR(80) NULL, + imapport VARCHAR(8) NULL, + imapflags VARCHAR(80) NULL, + stamp DATETIME NULL, + PRIMARY KEY (uniqueid), + CONSTRAINT yes_no_values CHECK (attach IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (deletevoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (saycid IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (sendvoicemail IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (review IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (tempgreetwarn IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (operator IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (envelope IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcename IN ('yes', 'no')), + CONSTRAINT yes_no_values CHECK (forcegreetings IN ('yes', 'no')) +); + +GO + +CREATE INDEX voicemail_mailbox ON voicemail (mailbox); + +GO + +CREATE INDEX voicemail_context ON voicemail (context); + +GO + +CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context); + +GO + +CREATE INDEX voicemail_imapuser ON voicemail (imapuser); + +GO + +CREATE TABLE meetme ( + bookid INTEGER NOT NULL IDENTITY(1,1), + confno VARCHAR(80) NOT NULL, + starttime DATETIME NULL, + endtime DATETIME NULL, + pin VARCHAR(20) NULL, + adminpin VARCHAR(20) NULL, + opts VARCHAR(20) NULL, + adminopts VARCHAR(20) NULL, + recordingfilename VARCHAR(80) NULL, + recordingformat VARCHAR(10) NULL, + maxusers INTEGER NULL, + members INTEGER NOT NULL, + PRIMARY KEY (bookid) +); + +GO + +CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime); + +GO + +CREATE TABLE musiconhold ( + name VARCHAR(80) NOT NULL, + mode VARCHAR(10) NULL, + directory VARCHAR(255) NULL, + application VARCHAR(255) NULL, + digit VARCHAR(1) NULL, + sort VARCHAR(10) NULL, + format VARCHAR(10) NULL, + stamp DATETIME NULL, + PRIMARY KEY (name), + CONSTRAINT moh_mode_values CHECK (mode IN ('custom', 'files', 'mp3nb', 'quietmp3nb', 'quietmp3')) +); + +GO + +-- Running upgrade 4da0c5f79a9c -> 43956d550a44 + +CREATE TABLE ps_endpoints ( + id VARCHAR(40) NOT NULL, + transport VARCHAR(40) NULL, + aors VARCHAR(200) NULL, + auth VARCHAR(40) NULL, + context VARCHAR(40) NULL, + disallow VARCHAR(200) NULL, + allow VARCHAR(200) NULL, + direct_media VARCHAR(3) NULL, + connected_line_method VARCHAR(8) NULL, + direct_media_method VARCHAR(8) NULL, + direct_media_glare_mitigation VARCHAR(8) NULL, + disable_direct_media_on_nat VARCHAR(3) NULL, + dtmf_mode VARCHAR(7) NULL, + external_media_address VARCHAR(40) NULL, + force_rport VARCHAR(3) NULL, + ice_support VARCHAR(3) NULL, + identify_by VARCHAR(8) NULL, + mailboxes VARCHAR(40) NULL, + moh_suggest VARCHAR(40) NULL, + outbound_auth VARCHAR(40) NULL, + outbound_proxy VARCHAR(40) NULL, + rewrite_contact VARCHAR(3) NULL, + rtp_ipv6 VARCHAR(3) NULL, + rtp_symmetric VARCHAR(3) NULL, + send_diversion VARCHAR(3) NULL, + send_pai VARCHAR(3) NULL, + send_rpid VARCHAR(3) NULL, + timers_min_se INTEGER NULL, + timers VARCHAR(8) NULL, + timers_sess_expires INTEGER NULL, + callerid VARCHAR(40) NULL, + callerid_privacy VARCHAR(23) NULL, + callerid_tag VARCHAR(40) NULL, + [100rel] VARCHAR(8) NULL, + aggregate_mwi VARCHAR(3) NULL, + trust_id_inbound VARCHAR(3) NULL, + trust_id_outbound VARCHAR(3) NULL, + use_ptime VARCHAR(3) NULL, + use_avpf VARCHAR(3) NULL, + media_encryption VARCHAR(4) NULL, + inband_progress VARCHAR(3) NULL, + call_group VARCHAR(40) NULL, + pickup_group VARCHAR(40) NULL, + named_call_group VARCHAR(40) NULL, + named_pickup_group VARCHAR(40) NULL, + device_state_busy_at INTEGER NULL, + fax_detect VARCHAR(3) NULL, + t38_udptl VARCHAR(3) NULL, + t38_udptl_ec VARCHAR(10) NULL, + t38_udptl_maxdatagram INTEGER NULL, + t38_udptl_nat VARCHAR(3) NULL, + t38_udptl_ipv6 VARCHAR(3) NULL, + tone_zone VARCHAR(40) NULL, + language VARCHAR(40) NULL, + one_touch_recording VARCHAR(3) NULL, + record_on_feature VARCHAR(40) NULL, + record_off_feature VARCHAR(40) NULL, + rtp_engine VARCHAR(40) NULL, + allow_transfer VARCHAR(3) NULL, + allow_subscribe VARCHAR(3) NULL, + sdp_owner VARCHAR(40) NULL, + sdp_session VARCHAR(40) NULL, + tos_audio INTEGER NULL, + tos_video INTEGER NULL, + cos_audio INTEGER NULL, + cos_video INTEGER NULL, + sub_min_expiry INTEGER NULL, + from_domain VARCHAR(40) NULL, + from_user VARCHAR(40) NULL, + mwi_fromuser VARCHAR(40) NULL, + dtls_verify VARCHAR(40) NULL, + dtls_rekey VARCHAR(40) NULL, + dtls_cert_file VARCHAR(200) NULL, + dtls_private_key VARCHAR(200) NULL, + dtls_cipher VARCHAR(200) NULL, + dtls_ca_file VARCHAR(200) NULL, + dtls_ca_path VARCHAR(200) NULL, + dtls_setup VARCHAR(7) NULL, + srtp_tag_32 VARCHAR(3) NULL, + UNIQUE (id), + CONSTRAINT yesno_values CHECK (direct_media IN ('yes', 'no')), + CONSTRAINT pjsip_connected_line_method_values CHECK (connected_line_method IN ('invite', 'reinvite', 'update')), + CONSTRAINT pjsip_connected_line_method_values CHECK (direct_media_method IN ('invite', 'reinvite', 'update')), + CONSTRAINT pjsip_direct_media_glare_mitigation_values CHECK (direct_media_glare_mitigation IN ('none', 'outgoing', 'incoming')), + CONSTRAINT yesno_values CHECK (disable_direct_media_on_nat IN ('yes', 'no')), + CONSTRAINT pjsip_dtmf_mode_values CHECK (dtmf_mode IN ('rfc4733', 'inband', 'info')), + CONSTRAINT yesno_values CHECK (force_rport IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (ice_support IN ('yes', 'no')), + CONSTRAINT pjsip_identify_by_values CHECK (identify_by IN ('username')), + CONSTRAINT yesno_values CHECK (rewrite_contact IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (rtp_ipv6 IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (rtp_symmetric IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_diversion IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_pai IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (send_rpid IN ('yes', 'no')), + CONSTRAINT pjsip_timer_values CHECK (timers IN ('forced', 'no', 'required', 'yes')), + CONSTRAINT pjsip_cid_privacy_values CHECK (callerid_privacy IN ('allowed_not_screened', 'allowed_passed_screened', 'allowed_failed_screened', 'allowed', 'prohib_not_screened', 'prohib_passed_screened', 'prohib_failed_screened', 'prohib', 'unavailable')), + CONSTRAINT pjsip_100rel_values CHECK ([100rel] IN ('no', 'required', 'yes')), + CONSTRAINT yesno_values CHECK (aggregate_mwi IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (trust_id_inbound IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (trust_id_outbound IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (use_ptime IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (use_avpf IN ('yes', 'no')), + CONSTRAINT pjsip_media_encryption_values CHECK (media_encryption IN ('no', 'sdes', 'dtls')), + CONSTRAINT yesno_values CHECK (inband_progress IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (fax_detect IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (t38_udptl IN ('yes', 'no')), + CONSTRAINT pjsip_t38udptl_ec_values CHECK (t38_udptl_ec IN ('none', 'fec', 'redundancy')), + CONSTRAINT yesno_values CHECK (t38_udptl_nat IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (t38_udptl_ipv6 IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (one_touch_recording IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (allow_transfer IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (allow_subscribe IN ('yes', 'no')), + CONSTRAINT pjsip_dtls_setup_values CHECK (dtls_setup IN ('active', 'passive', 'actpass')), + CONSTRAINT yesno_values CHECK (srtp_tag_32 IN ('yes', 'no')) +); + +GO + +CREATE INDEX ps_endpoints_id ON ps_endpoints (id); + +GO + +CREATE TABLE ps_auths ( + id VARCHAR(40) NOT NULL, + auth_type VARCHAR(8) NULL, + nonce_lifetime INTEGER NULL, + md5_cred VARCHAR(40) NULL, + password VARCHAR(80) NULL, + realm VARCHAR(40) NULL, + username VARCHAR(40) NULL, + UNIQUE (id), + CONSTRAINT pjsip_auth_type_values CHECK (auth_type IN ('md5', 'userpass')) +); + +GO + +CREATE INDEX ps_auths_id ON ps_auths (id); + +GO + +CREATE TABLE ps_aors ( + id VARCHAR(40) NOT NULL, + contact VARCHAR(40) NULL, + default_expiration INTEGER NULL, + mailboxes VARCHAR(80) NULL, + max_contacts INTEGER NULL, + minimum_expiration INTEGER NULL, + remove_existing VARCHAR(3) NULL, + qualify_frequency INTEGER NULL, + authenticate_qualify VARCHAR(3) NULL, + UNIQUE (id), + CONSTRAINT yesno_values CHECK (remove_existing IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (authenticate_qualify IN ('yes', 'no')) +); + +GO + +CREATE INDEX ps_aors_id ON ps_aors (id); + +GO + +CREATE TABLE ps_contacts ( + id VARCHAR(40) NOT NULL, + uri VARCHAR(40) NULL, + expiration_time VARCHAR(40) NULL, + qualify_frequency INTEGER NULL, + UNIQUE (id) +); + +GO + +CREATE INDEX ps_contacts_id ON ps_contacts (id); + +GO + +CREATE TABLE ps_domain_aliases ( + id VARCHAR(40) NOT NULL, + domain VARCHAR(80) NULL, + UNIQUE (id) +); + +GO + +CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id); + +GO + +CREATE TABLE ps_endpoint_id_ips ( + id VARCHAR(40) NOT NULL, + endpoint VARCHAR(40) NULL, + match VARCHAR(80) NULL, + UNIQUE (id) +); + +GO + +CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id); + +GO + +-- Running upgrade 43956d550a44 -> 581a4264e537 + +CREATE TABLE extensions ( + id BIGINT NOT NULL IDENTITY(1,1), + context VARCHAR(40) NOT NULL, + exten VARCHAR(40) NOT NULL, + priority INTEGER NOT NULL, + app VARCHAR(40) NOT NULL, + appdata VARCHAR(256) NOT NULL, + PRIMARY KEY (id, context, exten, priority), + UNIQUE (id) +); + +GO + +-- Running upgrade 581a4264e537 -> 2fc7930b41b3 + +CREATE TABLE ps_systems ( + id VARCHAR(40) NOT NULL, + timer_t1 INTEGER NULL, + timer_b INTEGER NULL, + compact_headers VARCHAR(3) NULL, + threadpool_initial_size INTEGER NULL, + threadpool_auto_increment INTEGER NULL, + threadpool_idle_timeout INTEGER NULL, + threadpool_max_size INTEGER NULL, + UNIQUE (id), + CONSTRAINT yesno_values CHECK (compact_headers IN ('yes', 'no')) +); + +GO + +CREATE INDEX ps_systems_id ON ps_systems (id); + +GO + +CREATE TABLE ps_globals ( + id VARCHAR(40) NOT NULL, + max_forwards INTEGER NULL, + user_agent VARCHAR(40) NULL, + default_outbound_endpoint VARCHAR(40) NULL, + UNIQUE (id) +); + +GO + +CREATE INDEX ps_globals_id ON ps_globals (id); + +GO + +CREATE TABLE ps_transports ( + id VARCHAR(40) NOT NULL, + async_operations INTEGER NULL, + bind VARCHAR(40) NULL, + ca_list_file VARCHAR(200) NULL, + cert_file VARCHAR(200) NULL, + cipher VARCHAR(200) NULL, + domain VARCHAR(40) NULL, + external_media_address VARCHAR(40) NULL, + external_signaling_address VARCHAR(40) NULL, + external_signaling_port INTEGER NULL, + method VARCHAR(11) NULL, + local_net VARCHAR(40) NULL, + password VARCHAR(40) NULL, + priv_key_file VARCHAR(200) NULL, + protocol VARCHAR(3) NULL, + require_client_cert VARCHAR(3) NULL, + verify_client VARCHAR(3) NULL, + verifiy_server VARCHAR(3) NULL, + tos VARCHAR(3) NULL, + cos VARCHAR(3) NULL, + UNIQUE (id), + CONSTRAINT pjsip_transport_method_values CHECK (method IN ('default', 'unspecified', 'tlsv1', 'sslv2', 'sslv3', 'sslv23')), + CONSTRAINT pjsip_transport_protocol_values CHECK (protocol IN ('udp', 'tcp', 'tls', 'ws', 'wss')), + CONSTRAINT yesno_values CHECK (require_client_cert IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (verify_client IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (verifiy_server IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (tos IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (cos IN ('yes', 'no')) +); + +GO + +CREATE INDEX ps_transports_id ON ps_transports (id); + +GO + +CREATE TABLE ps_registrations ( + id VARCHAR(40) NOT NULL, + auth_rejection_permanent VARCHAR(3) NULL, + client_uri VARCHAR(40) NULL, + contact_user VARCHAR(40) NULL, + expiration INTEGER NULL, + max_retries INTEGER NULL, + outbound_auth VARCHAR(40) NULL, + outbound_proxy VARCHAR(40) NULL, + retry_interval INTEGER NULL, + forbidden_retry_interval INTEGER NULL, + server_uri VARCHAR(40) NULL, + transport VARCHAR(40) NULL, + support_path VARCHAR(3) NULL, + UNIQUE (id), + CONSTRAINT yesno_values CHECK (auth_rejection_permanent IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (support_path IN ('yes', 'no')) +); + +GO + +CREATE INDEX ps_registrations_id ON ps_registrations (id); + +GO + +ALTER TABLE ps_endpoints ADD media_address VARCHAR(40) NULL; + +GO + +ALTER TABLE ps_endpoints ADD redirect_method VARCHAR(9) NULL; + +GO + +ALTER TABLE ps_endpoints ADD CONSTRAINT pjsip_redirect_method_values CHECK (redirect_method IN ('user', 'uri_core', 'uri_pjsip')); + +GO + +ALTER TABLE ps_endpoints ADD set_var TEXT NULL; + +GO + +EXEC sp_rename 'ps_endpoints.mwi_fromuser', mwi_from_user, 'COLUMN'; + +GO + +ALTER TABLE ps_contacts ADD outbound_proxy VARCHAR(40) NULL; + +GO + +ALTER TABLE ps_contacts ADD path TEXT NULL; + +GO + +ALTER TABLE ps_aors ADD maximum_expiration INTEGER NULL; + +GO + +ALTER TABLE ps_aors ADD outbound_proxy VARCHAR(40) NULL; + +GO + +ALTER TABLE ps_aors ADD support_path VARCHAR(3) NULL; + +GO + +ALTER TABLE ps_aors ADD CONSTRAINT yesno_values CHECK (support_path IN ('yes', 'no')); + +GO + +-- Running upgrade 2fc7930b41b3 -> 21e526ad3040 + +ALTER TABLE ps_globals ADD debug VARCHAR(40) NULL; + +GO + +-- Running upgrade 21e526ad3040 -> 28887f25a46f + +CREATE TABLE queues ( + name VARCHAR(128) NOT NULL, + musiconhold VARCHAR(128) NULL, + announce VARCHAR(128) NULL, + context VARCHAR(128) NULL, + timeout INTEGER NULL, + ringinuse VARCHAR(3) NULL, + setinterfacevar VARCHAR(3) NULL, + setqueuevar VARCHAR(3) NULL, + setqueueentryvar VARCHAR(3) NULL, + monitor_format VARCHAR(8) NULL, + membermacro VARCHAR(512) NULL, + membergosub VARCHAR(512) NULL, + queue_youarenext VARCHAR(128) NULL, + queue_thereare VARCHAR(128) NULL, + queue_callswaiting VARCHAR(128) NULL, + queue_quantity1 VARCHAR(128) NULL, + queue_quantity2 VARCHAR(128) NULL, + queue_holdtime VARCHAR(128) NULL, + queue_minutes VARCHAR(128) NULL, + queue_minute VARCHAR(128) NULL, + queue_seconds VARCHAR(128) NULL, + queue_thankyou VARCHAR(128) NULL, + queue_callerannounce VARCHAR(128) NULL, + queue_reporthold VARCHAR(128) NULL, + announce_frequency INTEGER NULL, + announce_to_first_user VARCHAR(3) NULL, + min_announce_frequency INTEGER NULL, + announce_round_seconds INTEGER NULL, + announce_holdtime VARCHAR(128) NULL, + announce_position VARCHAR(128) NULL, + announce_position_limit INTEGER NULL, + periodic_announce VARCHAR(50) NULL, + periodic_announce_frequency INTEGER NULL, + relative_periodic_announce VARCHAR(3) NULL, + random_periodic_announce VARCHAR(3) NULL, + retry INTEGER NULL, + wrapuptime INTEGER NULL, + penaltymemberslimit INTEGER NULL, + autofill VARCHAR(3) NULL, + monitor_type VARCHAR(128) NULL, + autopause VARCHAR(3) NULL, + autopausedelay INTEGER NULL, + autopausebusy VARCHAR(3) NULL, + autopauseunavail VARCHAR(3) NULL, + maxlen INTEGER NULL, + servicelevel INTEGER NULL, + strategy VARCHAR(11) NULL, + joinempty VARCHAR(128) NULL, + leavewhenempty VARCHAR(128) NULL, + reportholdtime VARCHAR(3) NULL, + memberdelay INTEGER NULL, + weight INTEGER NULL, + timeoutrestart VARCHAR(3) NULL, + defaultrule VARCHAR(128) NULL, + timeoutpriority VARCHAR(128) NULL, + PRIMARY KEY (name), + CONSTRAINT yesno_values CHECK (ringinuse IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setinterfacevar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setqueuevar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (setqueueentryvar IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (announce_to_first_user IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (relative_periodic_announce IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (random_periodic_announce IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (autofill IN ('yes', 'no')), + CONSTRAINT queue_autopause_values CHECK (autopause IN ('yes', 'no', 'all')), + CONSTRAINT yesno_values CHECK (autopausebusy IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (autopauseunavail IN ('yes', 'no')), + CONSTRAINT queue_strategy_values CHECK (strategy IN ('ringall', 'leastrecent', 'fewestcalls', 'random', 'rrmemory', 'linear', 'wrandom', 'rrordered')), + CONSTRAINT yesno_values CHECK (reportholdtime IN ('yes', 'no')), + CONSTRAINT yesno_values CHECK (timeoutrestart IN ('yes', 'no')) +); + +GO + +CREATE TABLE queue_members ( + queue_name VARCHAR(80) NOT NULL, + interface VARCHAR(80) NOT NULL, + uniqueid VARCHAR(80) NOT NULL, + membername VARCHAR(80) NULL, + state_interface VARCHAR(80) NULL, + penalty INTEGER NULL, + paused INTEGER NULL, + PRIMARY KEY (queue_name, interface) +); + +GO + +-- Running upgrade 28887f25a46f -> 4c573e7135bd + +ALTER TABLE ps_endpoints ALTER COLUMN tos_audio VARCHAR(10); + +GO + +ALTER TABLE ps_endpoints ALTER COLUMN tos_video VARCHAR(10); + +GO + +ALTER TABLE ps_transports ALTER COLUMN tos VARCHAR(10); + +GO + +ALTER TABLE ps_endpoints DROP COLUMN cos_audio; + +GO + +ALTER TABLE ps_endpoints DROP COLUMN cos_video; + +GO + +ALTER TABLE ps_transports DROP COLUMN cos; + +GO + +ALTER TABLE ps_endpoints ADD cos_audio INTEGER NULL; + +GO + +ALTER TABLE ps_endpoints ADD cos_video INTEGER NULL; + +GO + +ALTER TABLE ps_transports ADD cos INTEGER NULL; + +GO + +-- Running upgrade 4c573e7135bd -> 3855ee4e5f85 + +ALTER TABLE ps_endpoints ADD message_context VARCHAR(40) NULL; + +GO + +ALTER TABLE ps_contacts ADD user_agent VARCHAR(40) NULL; + +GO + +-- Running upgrade 3855ee4e5f85 -> e96a0b8071c + +ALTER TABLE ps_globals ALTER COLUMN user_agent VARCHAR(255); + +GO + +ALTER TABLE ps_contacts ALTER COLUMN id VARCHAR(255); + +GO + +ALTER TABLE ps_contacts ALTER COLUMN uri VARCHAR(255); + +GO + +ALTER TABLE ps_contacts ALTER COLUMN user_agent VARCHAR(255); + +GO + +ALTER TABLE ps_registrations ALTER COLUMN client_uri VARCHAR(255); + +GO + +ALTER TABLE ps_registrations ALTER COLUMN server_uri VARCHAR(255); + +GO + +-- Running upgrade e96a0b8071c -> c6d929b23a8 + +CREATE TABLE ps_subscription_persistence ( + id VARCHAR(40) NOT NULL, + packet VARCHAR(2048) NULL, + src_name VARCHAR(128) NULL, + src_port INTEGER NULL, + transport_key VARCHAR(64) NULL, + local_name VARCHAR(128) NULL, + local_port INTEGER NULL, + cseq INTEGER NULL, + tag VARCHAR(128) NULL, + endpoint VARCHAR(40) NULL, + expires INTEGER NULL, + UNIQUE (id) +); + +GO + +CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id); + +GO + +-- Running upgrade c6d929b23a8 -> 51f8cb66540e + +ALTER TABLE ps_endpoints ADD force_avp VARCHAR(3) NULL; + +GO + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (force_avp IN ('yes', 'no')); + +GO + +ALTER TABLE ps_endpoints ADD media_use_received_transport VARCHAR(3) NULL; + +GO + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_use_received_transport IN ('yes', 'no')); + +GO + +-- Running upgrade 51f8cb66540e -> 1d50859ed02e + +ALTER TABLE ps_endpoints ADD accountcode VARCHAR(20) NULL; + +GO + +-- Running upgrade 1d50859ed02e -> 1758e8bbf6b + +ALTER TABLE sippeers ALTER COLUMN useragent VARCHAR(255); + +GO + +-- Running upgrade 1758e8bbf6b -> 5139253c0423 + +ALTER TABLE queue_members DROP COLUMN uniqueid; + +GO + +ALTER TABLE queue_members ADD uniqueid INTEGER NOT NULL; + +GO + +ALTER TABLE queue_members ADD UNIQUE (uniqueid); + +GO + +-- Running upgrade 5139253c0423 -> d39508cb8d8 + +CREATE TABLE queue_rules ( + rule_name VARCHAR(80) NOT NULL, + time VARCHAR(32) NOT NULL, + min_penalty VARCHAR(32) NOT NULL, + max_penalty VARCHAR(32) NOT NULL +); + +GO + +-- Running upgrade d39508cb8d8 -> 5950038a6ead + +ALTER TABLE ps_transports ALTER COLUMN verifiy_server VARCHAR(3); + +GO + +EXEC sp_rename 'ps_transports.verifiy_server', verify_server, 'COLUMN'; + +GO + +ALTER TABLE ps_transports ADD CONSTRAINT yesno_values CHECK (verifiy_server IN ('yes', 'no')); + +GO + +-- Running upgrade 5950038a6ead -> 10aedae86a32 + +ALTER TABLE sippeers DROP CONSTRAINT sip_directmedia_values; + +GO + +ALTER TABLE sippeers ALTER COLUMN directmedia VARCHAR(8); + +GO + +ALTER TABLE sippeers ADD CONSTRAINT sip_directmedia_values_v2 CHECK (directmedia IN ('yes', 'no', 'nonat', 'update', 'outgoing')); + +GO + +-- Running upgrade 10aedae86a32 -> eb88a14f2a + +ALTER TABLE ps_endpoints ADD media_encryption_optimistic VARCHAR(3) NULL; + +GO + +ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_optimistic IN ('yes', 'no')); + +GO + +INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a'); + +GO + +COMMIT; + diff --git a/contrib/realtime/sqlserver/mssql_voicemail.sql b/contrib/realtime/sqlserver/mssql_voicemail.sql new file mode 100644 index 0000000000..815d24efb5 --- /dev/null +++ b/contrib/realtime/sqlserver/mssql_voicemail.sql @@ -0,0 +1,48 @@ +BEGIN TRANSACTION; + +CREATE TABLE alembic_version ( + version_num VARCHAR(32) NOT NULL +); + +GO + +-- Running upgrade None -> a2e9769475e + +CREATE TABLE voicemail_messages ( + dir VARCHAR(255) NOT NULL, + msgnum INTEGER NOT NULL, + context VARCHAR(80) NULL, + macrocontext VARCHAR(80) NULL, + callerid VARCHAR(80) NULL, + origtime INTEGER NULL, + duration INTEGER NULL, + recording IMAGE NULL, + flag VARCHAR(30) NULL, + category VARCHAR(30) NULL, + mailboxuser VARCHAR(30) NULL, + mailboxcontext VARCHAR(30) NULL, + msg_id VARCHAR(40) NULL +); + +GO + +ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum); + +GO + +CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir); + +GO + +-- Running upgrade a2e9769475e -> 39428242f7f5 + +ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE; + +GO + +INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5'); + +GO + +COMMIT; +