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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: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3781/ > ----------------------------------------------------------- > > (Updated Aug. 4, 2014, 8:25 p.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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