[asterisk-dev] [Code Review] 3781: Retrieve the source port of an incoming (chan_sip) SIP invite

[wdoekes]

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http://lists.digium.com/pipermail/asterisk-commits/2014-August/069629.html

It'd be nice if the documentation got updated too...

I know it's in an illogical place, in funcs/func_channel.c, but it should be updated nevertheless.

/w


- wdoekes


On Aug. 4, 2014, 8:25 p.m., dtryba wrote:
> 
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> (Updated Aug. 4, 2014, 8:25 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24040
>     https://issues.asterisk.org/jira/browse/ASTERISK-24040
> 
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> Repository: Asterisk
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> Description
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> 
> Retrieve the source port of an incoming (chan_sip) SIP invite in the dialplan with ${CHANNEL(recvport)}
> To complement ${CHANNEL(recvip)} and enable me to make dialplan decisions based on source port (in a peerless setup, handle everything as guests using AGI to lookup source ip/port for routing/handling).
> 
> pjsip appears to have this capability through the CHANNEL function (pjsip,local_addr/remote_addr).
> 
> Simple 2 line patch using ast_sockaddr_stringify_fmt(const struct ast_sockaddr *sa, int format)
> to return the port as a string.
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> Diffs
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>   /trunk/channels/sip/dialplan_functions.c 418610 
> 
> Diff: https://reviewboard.asterisk.org/r/3781/diff/
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> Testing
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> Tested on 11.10.2 (Debian Jessie) and trunk (418610) using IPv4. Having a few SIP endpoints connect from different address/ports combinations 
> Logged ${CHANNEL(recvip)}:${CHANNEL(recvport)} corresponds with source ip:port in packetdumps on the asterisk machine.
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> 
> Thanks,
> 
> dtryba
> 
>

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