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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3781/
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Review request for Asterisk Developers.
Bugs: ASTERISK-24040
https://issues.asterisk.org/jira/browse/ASTERISK-24040
Repository: Asterisk
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.
Diffs
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/trunk/channels/sip/dialplan_functions.c 418610
Diff: https://reviewboard.asterisk.org/r/3781/diff/
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.
Thanks,
dtryba
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