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ChangeLog: Updated for 13.8.2
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ChangeLog
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ChangeLog
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2016-04-20 10:46 +0000 Asterisk Development Team <asteriskteam@digium.com>
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* asterisk 13.8.2 Released.
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2016-04-20 05:45 +0000 [26d67ce885] Joshua Colp <jcolp@digium.com>
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* Release summaries: Remove previous versions
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2016-04-20 05:45 +0000 [d9909232ed] Joshua Colp <jcolp@digium.com>
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* .version: Update for 13.8.2
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2016-04-20 05:45 +0000 [fc57bb9b15] Joshua Colp <jcolp@digium.com>
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* .lastclean: Update for 13.8.2
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2016-04-20 05:45 +0000 [ac04474f38] Joshua Colp <jcolp@digium.com>
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* realtime: Add database scripts for 13.8.2
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2016-04-18 17:00 +0000 [91a3e1184f] Mark Michelson <mmichelson@digium.com>
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* res_pjsip_registrar: Fix bad memory-ness with user_agent.
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Recent changes to the PJSIP registrar resulted in tests failing due to
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missing AOR_CONTACT_ADDED test events. The reason for this was that the
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user_agent string had junk values in it, resulting in being unable to
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generate the event.
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I'm going to be honest here, I have no idea why this was happening. Here
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are the steps needed for the user_agent variable to get messed up:
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* REGISTER is received
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* First contact in the REGISTER results in a contact being removed
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* Second contact in the REGISTER results in a contact being added
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* The contact, AOR, expiration, and user agent all have to be passed as
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format parameters to the creation of a string. Any subset of those
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parameters would not be enough to cause the problem.
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Looking into what was happening, the thing that struck me as odd was
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that the user_agent variable was meant to be set to the value of the
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User-Agent SIP header in the incoming REGISTER. However, when removing a
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contact, the user_agent variable would be set (via ast_strdupa inside a
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loop) to the stored contact's user_agent. This means that the
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user_agent's value would be incorrect when attempting to process further
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contacts in the incoming REGISTER.
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The fix here is to use a different variable for the stored user agent
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when removing a contact. Correcting the behavior to be correct also
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means the memory usage is less weird, and the issue no longer occurs.
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ASTERISK-25929 #close
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Reported by Joshua Colp
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Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
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(cherry picked from commit f436b9ab111f1ff57c6dd3970051f123b42c1103)
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2016-04-18 13:41 +0000 [70e25ced60] Joshua Colp <jcolp@digium.com>
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* res_pjsip_transport_management: Allow unload to occur.
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At shutdown it is possible for modules to be unloaded that wouldn't
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normally be unloaded. This allows the environment to be cleaned up.
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The res_pjsip_transport_management module did not have the unload
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logic in it to clean itself up causing the res_pjsip module to not
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get unloaded. As a result the res_pjsip monitor thread kept going
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processing traffic and timers when it shouldn't.
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Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a
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(cherry picked from commit 49bfdc9ac029e0ef17cd8e85d8d7b7731387a34e)
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2016-04-18 12:12 +0000 [856931edc2] Mark Michelson <mmichelson@digium.com>
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* PJSIP: Remove PJSIP parsing functions from uri length validation.
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The PJSIP parsing functions provide a nice concise way to check the
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length of a hostname in a SIP URI. The problem is that in order to use
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those parsing functions, it's required to use them from a thread that
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has registered with PJLib.
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On startup, when parsing AOR configuration, the permanent URI handler
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may not be run from a PJLib-registered thread. Specifically, this could
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happen when Asterisk was started in daemon mode rather than
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console-mode. If PJProject were compiled with assertions enabled, then
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this would cause Asterisk to crash on startup.
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The solution presented here is to do our own parsing of the contact URI
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in order to ensure that the hostname in the URI is not too long. The
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parsing does not attempt to perform a full SIP URI parse/validation,
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since the hostname in the URI is what is important.
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ASTERISK-25928 #close
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Reported by Joshua Colp
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Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60
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(cherry picked from commit efae187217750e322cd6167705a33f888b631427)
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2016-04-14 20:26 +0000 Asterisk Development Team <asteriskteam@digium.com>
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2016-04-14 20:26 +0000 Asterisk Development Team <asteriskteam@digium.com>
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* asterisk 13.8.1 Released.
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* asterisk 13.8.1 Released.
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