[asterisk-dev] [Code Review] 3245: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

Kevin Harwell reviewboard at asterisk.org
Tue Feb 25 11:47:32 CST 2014

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(Updated Feb. 25, 2014, 11:47 a.m.)


This change has been marked as submitted.

Review request for Asterisk Developers.

Repository: Asterisk


Added the ability for transferring directly to voicemail on digium phones.  Added a new module that checks for the presence of a custom header and/or diversion header within a sip REFER.  If either is found and they specify a sending to voicemail action then variables are added to the channel allowing the user access to them in the dialplan.  Dialplan can then be written that branches based upon these values allowing, for instace, for a single number to be used for dialing and/or accessing voicemail directly.

Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip channels through (checked to make sure it has the correct channel type before proceeding).


  branches/12/res/res_pjsip_send_to_voicemail.c PRE-CREATION 
  branches/12/res/res_pjsip_header_funcs.c 408875 

Diff: https://reviewboard.asterisk.org/r/3245/diff/


Ran various scenarios manually with digium phones to make sure user were able to transfer callers directly to voicemail.  Also wrote a testsuite test that checks the presence of those headers/values in the dialplan: https://reviewboard.asterisk.org/r/3246/


Kevin Harwell

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