Mark Michelson has uploaded a new patch set (#2). Change subject: Rewrite sip_attended_transfer test to stop failing. ......................................................................
Rewrite sip_attended_transfer test to stop failing. The sip_attended_transfer test has been bouncing for a while. There are two major fixes introduced here to prevent the bouncing. First, by converting to using the testsuite's PJSUA module, we no longer are using the native Python threading library. Instead, we are using a method that works better with the Twisted framework. Second, the test is more strict about when the transfer may be performed. The previous test would attempt the transfer when Asterisk reported that the call was bridged. The problem is that Asterisk may report the call as bridged before PJSUA has properly processed the 200 OK that Asterisk has sent to it. By waiting, we can be sure that all parties are prepared when the transfer is attempted. The test has also been rewritten to only work with Asterisk 12+. A new separate test will be written to work on Asterisk 11. This helps the code to be a little less cluttered. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e --- A tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/configs/ast1/extensions.conf M tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf D tests/channels/SIP/sip_attended_transfer/run-test M tests/channels/SIP/sip_attended_transfer/test-config.yaml 5 files changed, 242 insertions(+), 216 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/19/19/2 -- To view, visit https://gerrit.asterisk.org/19 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson <mmichel...@digium.com> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev