[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.

Matt Jordan (Code Review) asteriskteam at digium.com
Mon Apr 6 11:22:03 CDT 2015


Matt Jordan has posted comments on this change.

Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
......................................................................


Patch Set 2:

(1 comment)

While my comment is piddly, I think a little bit more debug logging thrown into the Asterisk 1.8/11 case would be pretty helpful. It's often very difficult to figure out why a transfer test for those Asterisk versions fails - particularly since the event model presented by AMI can be rather confusing.

https://gerrit.asterisk.org/#/c/29/2/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
File tests/channels/SIP/sip_attended_transfer/attended_transfer.py:

Line 153:         if numchans == 1:
        :             self.controller.call_carol()
        :         elif numchans == 2:
        :             self.controller.transfer_call()
Since Asterisk 11 Bridge events are 'weird' (aka: confusing and prone to breakage), a few DEBUG statements in here might be nice.


-- 
To view, visit https://gerrit.asterisk.org/29
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: comment
Gerrit-Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d
Gerrit-PatchSet: 2
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-HasComments: Yes



More information about the asterisk-dev mailing list