[asterisk-dev] [Code Review] 3246: res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

[Kevin Harwell]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3246/
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(Updated Feb. 27, 2014, 2:32 p.m.)


Review request for Asterisk Developers.


Changes
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Addressed review issues.


Repository: testsuite


Description
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Testuite test for https://reviewboard.asterisk.org/r/3245/


Diffs (updated)
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  asterisk/trunk/tests/channels/pjsip/tests.yaml 4726 
  asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/test-config.yaml PRE-CREATION 
  asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/refer.xml PRE-CREATION 
  asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/sipp/invite.xml PRE-CREATION 
  asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/pjsip.conf PRE-CREATION 
  asterisk/trunk/tests/channels/pjsip/refer_send_to_vm/configs/ast1/extensions.conf PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/3246/diff/


Testing
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Ran test and it passed.


Thanks,

Kevin Harwell

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