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ChangeLog: Updated for certified/13.8-cert1-rc2

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changes/64/2764/1 certified/13.8-cert1-rc2
Joshua Colp 9 years ago
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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.

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