[asterisk-dev] [Code Review] external sip attended transfer test

[David Vossel]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/662/
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Review request for Asterisk Developers.


Summary
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This is an external test to exercise the native sip method of doing attended transfers using the Replaces header.

A, B, and C are SIP endpoints.

A calls B
A calls C
A transfers B to C

This test is verified using the 'Status' manager action to make sure the channel's are bridged correctly after the transfer.


To run this test, pjsip's pjsua app is required.  It can be found here, http://www.pjsip.org/download.htm.  After installing rename and move ..../pjsip-apps/bin/pjsua-x86-unknown-linux-gnu to /usr/sbin/pjsua. This puts pjsua into PATH, and this script expects it to be named 'pjsua'.


Diffs
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  /asterisk/trunk/tests/sip_attended_transfer/configs/extensions.conf PRE-CREATION 
  /asterisk/trunk/tests/sip_attended_transfer/configs/manager.conf PRE-CREATION 
  /asterisk/trunk/tests/sip_attended_transfer/configs/sip.conf PRE-CREATION 
  /asterisk/trunk/tests/sip_attended_transfer/run-test PRE-CREATION 
  /asterisk/trunk/tests/sip_attended_transfer/test-config.yaml PRE-CREATION 

Diff: https://reviewboard.asterisk.org/r/662/diff


Testing
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Tested with 1.4 and Trunk.  Trunk r263541 currently causes this test to segfault, but earlier revisions pass.


Thanks,

David