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Summary:ASTERISK-24574: Bouncy transfer tests
Reporter:Joshua C. Colp (jcolp)Labels:
Date Opened:2014-12-01 10:13:35.000-0600Date Closed:2014-12-31 15:29:15.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Tests/testsuite
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Description:Some transfer tests currently rely on SIP signaling to trigger the transfer operation. This is problematic as the channels may not be in the right state within Asterisk to allow the transfer (they may still be joining a bridge or may still be moving around).

A quick hack to add a callLater of a second before doing the transfer fixed the issue in my environment consistently but this is of course not a reliable fix.
Comments:By: Matt Jordan (mjordan) 2014-12-06 22:11:38.333-0600

I've gone ahead and fixed the following:
* {{tests/channels/pjsip/transfers/blind_transfer/callee_refer_only}}
* {{tests/channels/pjsip/transfers/blind_transfer/caller_refer_only}}

These were failing for the reason described on this issue. We now only initiate the transfer once both of the original parties are in a bridge together.

The direct media tests, however, still bounce from time to time. This may be for similar reasons, or it may be specific to the re-INVITEs incurred during the direct media negotiation.

By: Matt Jordan (mjordan) 2014-12-31 15:29:16.029-0600

This ended up getting fixed due to a combination of other test changes and the synchronization of answering a channel in {{chan_pjsip}}.