[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

[Mark Michelson]

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Ship it!


Ship It!

- Mark Michelson


On May 9, 2013, 4:38 a.m., Michael Young wrote:
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> (Updated May 9, 2013, 4:38 a.m.)
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> Review request for Asterisk Developers.
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> Bugs: ASTERISK-21374
>     https://issues.asterisk.org/jira/browse/ASTERISK-21374
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> Repository: Asterisk
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> Description
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> Okay... so... I created a crash problem which will not make many people very happy.
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> The code that was submitted failed to take into consideration that not all outgoing calls will be to a peer.  Shame on me.
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> 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.
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> 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.
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> Diffs
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>   /branches/11/channels/chan_sip.c 388111 
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> Diff: https://reviewboard.asterisk.org/r/2524/diff/
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> Testing
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> Tested on a production machine for some days now and on a couple of dev machines.
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> Thanks,
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> Michael Young
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