[asterisk-dev] [Code Review] 2524: Fix The Crash Caused By The Fix For One-way Audio With auto_* NAT Settings When SIP Calls Are Initiated By PBX

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(Updated May 13, 2013, 4:05 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
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Committed in revision 388601


Bugs: ASTERISK-21374
    https://issues.asterisk.org/jira/browse/ASTERISK-21374


Repository: Asterisk


Description
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Okay... so... I created a crash problem which will not make many people very happy.

The code that was submitted failed to take into consideration that not all outgoing calls will be to a peer.  Shame on me.

This patch checks if there is a related peer involved.  If there is, then go about checking its NAT settings to see if it is using the new auto_* NAT settings and then handle the call accordingly.

Also, I discovered an issue with real time peers.  If the global setting has auto_force_rport set and we issued a "sip reload" while a peer is still registered, which, resets the flags for the NAT settings on the peer to off (and we haven't had a chance to check if the peer might be behind NAT in order to turn the force_rport flag on for the peer), we were always setting the contact address of the peer to that of the full contact.  This patch corrects that condition as well.


Diffs
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  /branches/11/channels/chan_sip.c 388111 

Diff: https://reviewboard.asterisk.org/r/2524/diff/


Testing
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Tested on a production machine for some days now and on a couple of dev machines.


Thanks,

Michael Young

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