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

[dtryba]

> On July 21, 2014, 6:31 p.m., Mark Michelson wrote:
> > /trunk/channels/sip/dialplan_functions.c, line 80
> > <https://reviewboard.asterisk.org/r/3781/diff/1/?file=63311#file63311line80>
> >
> >     There's an inline function in include/asterisk/netsock2.h called ast_sockaddr_stringify_port() that you can use in place of the ast_sockaddr_stringify_fmt() call.

Didn't spot that one. Updated patch, same results as expected.


- dtryba


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On July 22, 2014, 11:44 a.m., dtryba wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3781/
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> 
> (Updated July 22, 2014, 11:44 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Bugs: ASTERISK-24040
>     https://issues.asterisk.org/jira/browse/ASTERISK-24040
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   /trunk/channels/sip/dialplan_functions.c 418610 
> 
> Diff: https://reviewboard.asterisk.org/r/3781/diff/
> 
> 
> Testing
> -------
> 
> 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.
> 
> 
> Thanks,
> 
> dtryba
> 
>

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