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

[dtryba]

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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.


Changes
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ast_sockaddr_stringify_port instead of ast_sockaddr_stringify_fmt


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 (updated)
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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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