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Author	SHA1	Message	Date
Joshua Colp	601602f44b	ChangeLog: Updated for certified/13.8-cert1-rc2	9 years ago
Joshua Colp	13461bb9a6	Release summaries: Add summaries for certified/13.8-cert1-rc2	9 years ago
Joshua Colp	cadb5c4e64	Release summaries: Remove previous versions	9 years ago
Joshua Colp	d4d5548ef8	.version: Update for certified/13.8-cert1-rc2	9 years ago
Joshua Colp	a5bc40ae51	.lastclean: Update for certified/13.8-cert1-rc2	9 years ago
Joshua Colp	2b6df52c66	realtime: Add database scripts for certified/13.8-cert1-rc2	9 years ago
Alexei Gradinari	c4426f1035	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	9 years ago
Joshua Colp	c69e0f1813	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	9 years ago
Joshua Colp	a95dbe6fee	Merge "app_queue: queue members can receive multiple calls" into certified/13.8	9 years ago
Kevin Harwell	eebe8b3dd3	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	9 years ago
George Joseph	5cbd4b9799	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	9 years ago
zuul	7a18ebaa0c	Merge "res_stasis: Handle re-enter stasis bridge with swap channel." into certified/13.8	9 years ago
zuul	35f9b6f38b	Merge "bridge: Hold off more than one imparting channel at a time." into certified/13.8	9 years ago
Richard Mudgett	bc51227ef8	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	9 years ago
Richard Mudgett	8dd79720e6	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	9 years ago
George Joseph	2a2e754d15	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	9 years ago
Joshua Colp	188ce34aff	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	9 years ago
Joshua Colp	da80f40014	app_talkdetect: Enable for testsuite tests. ... Change-Id: I9acf2e2210f7a15cdd2c63c4c8dcb92de6b47d43	9 years ago
Mark Michelson	9f3ecf0a8d	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	9 years ago
Joshua Colp	84edbae3ab	Merge "res_pjsip_transport_management: Allow unload to occur." into certified/13.8	9 years ago
Mark Michelson	39b4742db1	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	9 years ago
Joshua Colp	4caa57f6b3	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	9 years ago
Mark Michelson	0b35582bbb	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	9 years ago
Mark Michelson	9f8b803a29	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	9 years ago
Joshua Colp	3a845d27fd	Merge "Rename res_pjsip_keepalive res_pjsip_transport_management" into certified/13.8	9 years ago
Mark Michelson	a35d3eb73b	Rename res_pjsip_keepalive res_pjsip_transport_management ... ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433	9 years ago
Mark Michelson	3de37dee68	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	9 years ago
George Joseph	e378c18815	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 d963a33749)	9 years ago
Joshua Colp	dd93204a84	ChangeLog: Updated for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	6d29a919d4	Release summaries: Add summaries for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	4fa3428247	Release summaries: Remove previous versions	9 years ago
Joshua Colp	b418e14998	.version: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	69b6cf2368	.lastclean: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	847dc5c7d7	realtime: Add database scripts for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	c86b5e369e	Merge "res_pjsip: Handle deferred SDP hold/unhold properly." into certified/13.8	9 years ago
Joshua Colp	c23bf7c8df	ChangeLog: Updated for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	4f94668022	Release summaries: Add summaries for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	454daec0e1	Release summaries: Remove previous versions	9 years ago
Joshua Colp	4ba2b5e92c	.version: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	e6f27ca09c	.lastclean: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	08dbdd5996	realtime: Add database scripts for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	ec7a89771d	ChangeLog: Updated for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	ffcb651205	Release summaries: Add summaries for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	97499f717a	Release summaries: Remove previous versions	9 years ago
Joshua Colp	99d52771b5	.version: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	eb9e193c65	.lastclean: Update for certified/13.8-cert1-rc1	9 years ago
Joshua Colp	8ec588b8b1	realtime: Add database scripts for certified/13.8-cert1-rc1	9 years ago
Mark Michelson	4b87a773dc	res_pjsip: Handle deferred SDP hold/unhold properly. ... Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. At this point, PJMedia's SDP negotiator saves Asterisk's local state as being recvonly. Now, when the device attempts to unhold, it again uses a deferred SDP reinvite, so we end up doing the following: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating recvonly on streams * Device sends ACK with SDP indicating sendonly on streams The problem here is that Asterisk offered recvonly, and by RFC 3264's rules, if an offer is recvonly, the answer has to be sendonly. The result is that the device is not taken off hold. What is supposed to happen is that Asterisk should indicate sendrecv in the 200 OK that it sends. This way, the device has the freedom to indicate sendrecv if it wants the stream taken off hold, or it can continue to respond with sendonly if the purpose of the reinvite was something else (like a session timer refresher). The fix here is to alter the SDP negotiator's state when we receive a reinvite with no SDP. If the negotiator's state is currently in the recvonly or inactive state, then we alter our local state to be sendrecv. This way, we allow the device to indicate the stream state as desired. ASTERISK-25854 #close Reported by Robert McGilvray Change-Id: I7615737276165eef3a593038413d936247dcc6ed	9 years ago
Joshua Colp	c29e2e3fb7	.version: Update for certified/13.8 ... Change-Id: I37e5a8e36c2f4f9137f8f230c99220005424e514	9 years ago
Matthew Jordan	3c796e694e	Disable extended support modules ... Change-Id: Ia2e359021b3eccecce20028098c5b6d1099c3f9e	9 years ago