[asterisk-dev] [Code Review] 2423: Pimp my SIP: Messaging

[Kevin Harwell]

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(Updated April 11, 2013, 10:40 p.m.)


Review request for Asterisk Developers.


Changes
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Addressed review issues:

changed call to free to ast_free and removed NULL check.
added an inc/dec the refcount to the associated session object when sendtext is called.


Bugs: ASTERISK-21076
    https://issues.asterisk.org/jira/browse/ASTERISK-21076


Repository: Asterisk


Description
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Adds messaging support to the new SIP work being done in Asterisk.  Both out of call and in-dialog MESSAGE requests are handled.


Diffs (updated)
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  /team/group/pimp_my_sip/channels/chan_gulp.c 385389 
  /team/group/pimp_my_sip/res/res_sip.c 385389 
  /team/group/pimp_my_sip/res/res_sip_messaging.c PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/2423/diff/


Testing
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Wrote/modified several tests for the testsuite that include tests for both out of call and in-dialog MESSAGE requests, custom header propagation, MessageSend with specified from parameter, and the AMI MessageSend action.


Thanks,

Kevin Harwell

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