+2016-05-03 12:55 +0000 Asterisk Development Team <asteriskteam@digium.com>
+
+ * asterisk certified/13.8-cert1-rc2 Released.
+
+2016-05-03 07:54 +0000 [cadb5c4e64] Joshua Colp <jcolp@digium.com>
+
+ * Release summaries: Remove previous versions
+
+2016-05-03 07:54 +0000 [d4d5548ef8] Joshua Colp <jcolp@digium.com>
+
+ * .version: Update for certified/13.8-cert1-rc2
+
+2016-05-03 07:54 +0000 [a5bc40ae51] Joshua Colp <jcolp@digium.com>
+
+ * .lastclean: Update for certified/13.8-cert1-rc2
+
+2016-05-03 07:54 +0000 [2b6df52c66] Joshua Colp <jcolp@digium.com>
+
+ * realtime: Add database scripts for certified/13.8-cert1-rc2
+
+2016-04-15 11:59 +0000 [c4426f1035] Alexei Gradinari <alex2grad@gmail.com>
+
+ * res_pjsip: disable multi domain to improve realtime performace
+
+ This patch added new global pjsip option 'disable_multi_domain'.
+ Disabling Multi Domain can improve Realtime performance by reducing
+ number of database requests.
+
+ ASTERISK-25930 #close
+
+ Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
+
+2016-04-26 05:48 +0000 [c69e0f1813] Joshua Colp <jcolp@digium.com>
+
+ * app_queue: Fix crash when unloading module.
+
+ When unloading the app_queue module the members in each queue are
+ destroyed and as part of this they are removed from the pending
+ members container. Unfortunately a crash would occur as the container
+ was destroyed before the members were removed.
+
+ This change tweaks ordering so the container destruction occurs
+ after the members are destroyed.
+
+ ASTERISK-16115
+
+ Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b
+
+2016-04-21 14:23 +0000 [eebe8b3dd3] Kevin Harwell <kharwell@digium.com>
+
+ * app_queue: queue members can receive multiple calls
+
+ It was possible for a queue member that is a member of at least 2 or more
+ queues to receive mulitiple calls at the same time. This happened because
+ of a race between when a member was being rung and when the device state
+ notified the other queue(s) member object of the state change.
+
+ This patch makes it so when a queue member is being rung it gets added to
+ a global pool of queue members. If that same member is tried again, e.g.
+ from another queue, and it is found to already exist in the pending member
+ container then it will not ring that member.
+
+ ASTERISK-16115 #close
+
+ Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48
+
+2016-04-22 17:53 +0000 [5cbd4b9799] gtjoseph <gjoseph@digium.com>
+
+ * res_agi: Prevent run_agi from eating frames it shouldn't
+
+ The run_agi function is eating control frames when it shouldn't be. This is
+ causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond
+ transfer.
+
+ Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie
+ answers.
+
+ Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE
+ and is left thinking he's connected to Bob.
+
+ In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls
+ an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on
+ Charlie's channel.
+
+ The fix was to accumulate deferrable frames in the "forever" loop instead of
+ dropping them, and re-queue them just before running the actual agi command
+ or exiting.
+
+ ASTERISK-25951 #close
+
+ Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645
+
+2016-04-15 14:36 +0000 [bc51227ef8] Richard Mudgett <rmudgett@digium.com>
+
+ * res_stasis: Handle re-enter stasis bridge with swap channel.
+
+ We lose the fact that there is a swap channel if there is one. We
+ currently wind up rejoining the stasis bridge as a normal join after the
+ swap channel has already been kicked from the bridge.
+
+ This patch preserves the swap channel so the AMI/ARI events can note that
+ the channel joining the bridge is swapping with another channel. Another
+ benefit to swaqpping in one operation is if there are any channels that
+ get lonely (MOH, bridge playback, and bridge record channels). The lonely
+ channels won't leave before the joining channel has a chance to come back
+ in under stasis if the swap channel is the only reason the lonely channels
+ are staying in the bridge.
+
+ ASTERISK-25947 #close
+ Reported by: Richard Mudgett
+
+ ASTERISK-24649
+ Reported by: John Bigelow
+
+ ASTERISK-24782
+ Reported by: John Bigelow
+
+ Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee
+
+2016-04-19 16:58 +0000 [8dd79720e6] Richard Mudgett <rmudgett@digium.com>
+
+ * bridge: Hold off more than one imparting channel at a time.
+
+ An earlier patch blocked the ast_bridge_impart() call until the channel
+ either entered the target bridge or it failed. Unfortuantely, if the
+ target bridge is stasis and the imprted channel is not a stasis channel,
+ stasis bounces the channel out of the bridge to come back into the bridge
+ as a proper stasis channel. When the channel is bounced out, that
+ released the block on ast_bridge_impart() to continue. If the impart was
+ a result of a transfer, then it became a race to see if the swap channel
+ would get hung up before the imparted channel could come back into the
+ stasis bridge. If the imparted channel won then everything is fine. If
+ the swap channel gets hung up first then the transfer will fail because
+ the swap channel is leaving the bridge.
+
+ * Allow a chain of ast_bridge_impart()'s to happen before any are
+ unblocked to prevent the race condition described above. When the channel
+ finally joins the bridge or completely fails to join the bridge then the
+ ast_bridge_impart() instances are unblocked.
+
+ ASTERISK-25947
+ Reported by: Richard Mudgett
+
+ ASTERISK-24649
+ Reported by: John Bigelow
+
+ ASTERISK-24782
+ Reported by: John Bigelow
+
+ Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1
+
+2016-04-19 17:52 +0000 [2a2e754d15] gtjoseph <george.joseph@fairview5.com>
+
+ * res_pjsip_callerid: Clear out display name if id->name is not valid
+
+ When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning
+ the From header, then it overwrites the display name and uri from the channel's
+ connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was
+ leaving the display name from the From header in the new RPID or PAI header.
+ On an attended transfer where the originator had a caller id number set but not
+ a display name, the re-INVITE to the final transferee had the number of the
+ originator but the display name of the transferer.
+
+ Added a check to clear out the display name in the new header if
+ connected.id.name was invalid.
+
+ ASTERISK-25942 #close
+
+ Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b
+
+2016-04-19 13:02 +0000 [188ce34aff] Joshua Colp <jcolp@digium.com>
+
+ * app_talkdetect: Make the module core supported.
+
+ This module is used as part of testsuite tests to confirm
+ stuff works. I'm accordingly marking it as core as it is
+ required by those tests.
+
+ Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88
+
+2016-04-19 13:00 +0000 [da80f40014] Joshua Colp <jcolp@digium.com>
+
+ * app_talkdetect: Enable for testsuite tests.
+
+ Change-Id: I9acf2e2210f7a15cdd2c63c4c8dcb92de6b47d43
+
+2016-04-18 12:12 +0000 [9f3ecf0a8d] Mark Michelson <mmichelson@digium.com>
+
+ * PJSIP: Remove PJSIP parsing functions from uri length validation.
+
+ The PJSIP parsing functions provide a nice concise way to check the
+ length of a hostname in a SIP URI. The problem is that in order to use
+ those parsing functions, it's required to use them from a thread that
+ has registered with PJLib.
+
+ On startup, when parsing AOR configuration, the permanent URI handler
+ may not be run from a PJLib-registered thread. Specifically, this could
+ happen when Asterisk was started in daemon mode rather than
+ console-mode. If PJProject were compiled with assertions enabled, then
+ this would cause Asterisk to crash on startup.
+
+ The solution presented here is to do our own parsing of the contact URI
+ in order to ensure that the hostname in the URI is not too long. The
+ parsing does not attempt to perform a full SIP URI parse/validation,
+ since the hostname in the URI is what is important.
+
+ ASTERISK-25928 #close
+ Reported by Joshua Colp
+
+ Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
+
+2016-04-18 17:00 +0000 [39b4742db1] Mark Michelson <mmichelson@digium.com>
+
+ * res_pjsip_registrar: Fix bad memory-ness with user_agent.
+
+ Recent changes to the PJSIP registrar resulted in tests failing due to
+ missing AOR_CONTACT_ADDED test events. The reason for this was that the
+ user_agent string had junk values in it, resulting in being unable to
+ generate the event.
+
+ I'm going to be honest here, I have no idea why this was happening. Here
+ are the steps needed for the user_agent variable to get messed up:
+ * REGISTER is received
+ * First contact in the REGISTER results in a contact being removed
+ * Second contact in the REGISTER results in a contact being added
+ * The contact, AOR, expiration, and user agent all have to be passed as
+ format parameters to the creation of a string. Any subset of those
+ parameters would not be enough to cause the problem.
+
+ Looking into what was happening, the thing that struck me as odd was
+ that the user_agent variable was meant to be set to the value of the
+ User-Agent SIP header in the incoming REGISTER. However, when removing a
+ contact, the user_agent variable would be set (via ast_strdupa inside a
+ loop) to the stored contact's user_agent. This means that the
+ user_agent's value would be incorrect when attempting to process further
+ contacts in the incoming REGISTER.
+
+ The fix here is to use a different variable for the stored user agent
+ when removing a contact. Correcting the behavior to be correct also
+ means the memory usage is less weird, and the issue no longer occurs.
+
+ ASTERISK-25929 #close
+ Reported by Joshua Colp
+
+ Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
+
+2016-04-18 13:41 +0000 [4caa57f6b3] Joshua Colp <jcolp@digium.com>
+
+ * res_pjsip_transport_management: Allow unload to occur.
+
+ At shutdown it is possible for modules to be unloaded that wouldn't
+ normally be unloaded. This allows the environment to be cleaned up.
+
+ The res_pjsip_transport_management module did not have the unload
+ logic in it to clean itself up causing the res_pjsip module to not
+ get unloaded. As a result the res_pjsip monitor thread kept going
+ processing traffic and timers when it shouldn't.
+
+ Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
+
+2016-04-14 13:49 +0000 [0b35582bbb] Mark Michelson <mmichelson@digium.com>
+
+ * transport management: Register thread with PJProject.
+
+ The scheduler thread that kills idle TCP connections was not registering
+ with PJProject properly and causing assertions if PJProject was built in
+ debug mode.
+
+ This change registers the thread with PJProject the first time that the
+ scheduler callback executes.
+
+ AST-2016-005
+
+ Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283
+
+2016-03-08 12:12 +0000 [9f8b803a29] Mark Michelson <mmichelson@digium.com>
+
+ * res_pjsip_transport_management: Kill idle TCP connections.
+
+ "Idle" here means that someone connects to us and does not send a SIP
+ request. PJProject will not automatically time out such connections, so
+ it's up to Asterisk to do it instead.
+
+ When we receive an incoming TCP connection, we will start a timer
+ (equivalent to transaction timer D) waiting to receive an incoming
+ request. If we do not receive a request in that timeframe, then we will
+ shut down the TCP connection.
+
+ ASTERISK-25796 #close
+ Reported by George Joseph
+
+ AST-2016-005
+
+ Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6
+
+2016-03-08 10:52 +0000 [a35d3eb73b] Mark Michelson <mmichelson@digium.com>
+
+ * Rename res_pjsip_keepalive res_pjsip_transport_management
+
+ ASTERISK-25796
+ Reported by George Joseph
+
+ AST-2016-005
+
+ Change-Id: Id322a05f927392293570599730050bc677d99433
+
+2016-04-14 07:15 +0000 [3de37dee68] Mark Michelson <mmichelson@digium.com>
+
+ * AST-2016-004: Fix crash on REGISTER with long URI.
+
+ Due to some ignored return values, Asterisk could crash if processing an
+ incoming REGISTER whose contact URI was above a certain length.
+
+ ASTERISK-25707 #close
+ Reported by George Joseph
+
+ Patches:
+ 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
+
+ AST-2016-004
+
+ Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55
+
+2016-03-23 08:59 +0000 [e378c18815] gtjoseph <george.joseph@fairview5.com>
+
+ * pjproject-bundled: Cleanups for reported issues
+
+ PortAudio should no longer be required
+ PJSIP_MAX_PKT_LEN is now 6000
+ Older autoconf issue fixed. (CentOS 6)
+
+ Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd
+ (cherry picked from commit d963a3374991c64594cf196e90a5c74964c8ba7c)
+
2016-04-06 16:01 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk certified/13.8-cert1-rc1 Released.
projects / thirdparty / asterisk.git / commitdiff
