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ChangeLog: Updated for 13.8.2

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Joshua Colp
2016-04-20 05:46:18 -05:00
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2016-04-20 10:46 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.8.2 Released.
2016-04-20 05:45 +0000 [26d67ce885] Joshua Colp <jcolp@digium.com>
* Release summaries: Remove previous versions
2016-04-20 05:45 +0000 [d9909232ed] Joshua Colp <jcolp@digium.com>
* .version: Update for 13.8.2
2016-04-20 05:45 +0000 [fc57bb9b15] Joshua Colp <jcolp@digium.com>
* .lastclean: Update for 13.8.2
2016-04-20 05:45 +0000 [ac04474f38] Joshua Colp <jcolp@digium.com>
* realtime: Add database scripts for 13.8.2
2016-04-18 17:00 +0000 [91a3e1184f] 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
(cherry picked from commit f436b9ab111f1ff57c6dd3970051f123b42c1103)
2016-04-18 13:41 +0000 [70e25ced60] 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
(cherry picked from commit 49bfdc9ac029e0ef17cd8e85d8d7b7731387a34e)
2016-04-18 12:12 +0000 [856931edc2] 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
(cherry picked from commit efae187217750e322cd6167705a33f888b631427)
2016-04-14 20:26 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.8.1 Released.