[asterisk-dev] [Code Review] 2466: Pimp my SIP: Call forwarding & diversion headers

[svnbot]

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https://reviewboard.asterisk.org/r/2466/
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(Updated May 3, 2013, 11:32 a.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers, Joshua Colp and Mark Michelson.


Changes
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Committed in revision 387582


Bugs: ASTERISK-21426
    https://issues.asterisk.org/jira/browse/ASTERISK-21426


Repository: Asterisk


Description
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Adds call forwarding support (Josh's patch) to the new SIP work being done in Asterisk.  This also includes the ability to add a diversion header to an outgoing response/request when appropriate.  The diversion header feature can be turned off by setting the send_diversion=false (defaults to true) on an endpoint within the configuration file.


Diffs
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  /team/group/pimp_my_sip/res/res_sip_diversion.c PRE-CREATION 
  /team/group/pimp_my_sip/res/res_sip/sip_configuration.c 386621 
  /team/group/pimp_my_sip/res/res_sip.c 386621 
  /team/group/pimp_my_sip/include/asterisk/res_sip.h 386621 
  /team/group/pimp_my_sip/channels/chan_gulp.c 386621 
  /team/group/pimp_my_sip/res/res_sip_session.c 386621 

Diff: https://reviewboard.asterisk.org/r/2466/diff/


Testing
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Wrote a few testsuite tests to make sure calls are being forwarded and the diversion header is added/propagated appropriately.


Thanks,

Kevin Harwell

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