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(Updated July 22, 2014, 11:44 a.m.) Review request for Asterisk Developers. Changes ------- ast_sockaddr_stringify_port instead of ast_sockaddr_stringify_fmt 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 (updated) ----- /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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