pjsip/rest_api: Introduce a test for ChannelTransfer event #77
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Attempt to use the dialplan function over a "well-known" variable. |
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REMINDER: If this PR applies to other branches, please add a comment with the appropriate "cherry-pick-to" headers as per the [Create a Pull Request](https://docs.asterisk.org/Development/Policies-and-Procedures/Code-Contribution) process.
If you don't want it cherry-picked, please add a "cherry-pick-to: none" comment so we don't keep asking. If, after adding "cherry-pick-to" comments, you change your mind, please edit the comment to DELETE the header lines and add "cherry-pick-to: none". The currently active branches are now 20, 21, 22 and master. |
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cherry-pick-to: 20 |
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Test that a SIP REFER is translated into a ChannelTransfer event when TRANSFERHANDLING=ari-only is set on the channel and that transfer_progress generates the necessary SIP NOTIFY messages. Be strict on the first ChannelTransfer event being received. Then only check that the transfer state is changing from not being present, to progress and finally answered. When answered delete all the channels and the bridge and the test wull succeed.
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I am looking at https://github.com/asterisk/asterisk/actions/runs/13272885836/job/37056440524 which failed when asterisk/asterisk#1102 was merged. While I don't see a way to get the "asterisk" binary that was used to run the test, the full.txt has this as the output: What is the probability that the "20-ari" test didn't use an Asterisk 20 binary with the patch to be cherry picked? |
Test that a SIP REFER is translated into a ChannelTransfer event when TRANSFERHANDLING=ari-only is set on the channel and that transfer_progress generates the necessary SIP NOTIFY messages.
Be strict on the first ChannelTransfer event being received. Then only check that the transfer state is changing from not being present, to progress and finally answered. When answered delete all the channels and the bridge and the test wull succeed.