> On June 6, 2014, 8:35 a.m., Matt Jordan wrote: > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/attended_transfer.py, > > lines 15-21 > > <https://reviewboard.asterisk.org/r/3560/diff/2/?file=58800#file58800line15> > > > > I'm not sure I understand why this is needed. The SIPp scenarios > > themselves should be able to govern when the channels are hung up.
In these tests, there are four SIPp scenarios running. Only two of these scenarios are communicating via 3PCC (referee.xml and referer[_uas].xml). The other two scenarios are just basic UAC or UAS scenarios that wait for a hangup. - opticron ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3560/#review12076 ----------------------------------------------------------- On May 22, 2014, 3:20 p.m., opticron wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3560/ > ----------------------------------------------------------- > > (Updated May 22, 2014, 3:20 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-23643 > https://issues.asterisk.org/jira/browse/ASTERISK-23643 > > > Repository: testsuite > > > Description > ------- > > This adds tests for nominal callee- and caller-initiated attended transfer > scenarios using SIPp scenarios and SIPp's 3PCC to coordinate call state. This > also adds another sample SIPp scenario for handling the REFER initiator > behavior as a UAS that works with referee.xml (the UAC attended transfer > coordinator). > > > Diffs > ----- > > asterisk/trunk/tests/channels/pjsip/transfers/tests.yaml 5043 > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/tests.yaml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/test-config.yaml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/sipp/uas.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/sipp/referer.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/sipp/referee.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/configs/ast1/pjsip.conf > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/configs/ast1/extensions.conf > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/caller_initiated/attended_transfer.py > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/test-config.yaml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/sipp/uas.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/sipp/uac-no-hangup.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/sipp/referer_uas.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/sipp/referee.xml > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/configs/ast1/pjsip.conf > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/configs/ast1/extensions.conf > PRE-CREATION > > asterisk/trunk/tests/channels/pjsip/transfers/attended_transfer/callee_initiated/attended_transfer.py > PRE-CREATION > asterisk/trunk/contrib/sipp/transfer/referer_uas.xml PRE-CREATION > asterisk/trunk/contrib/sipp/table_of_contents 5043 > > Diff: https://reviewboard.asterisk.org/r/3560/diff/ > > > Testing > ------- > > Ensured that the tests performed as expected. > > > Thanks, > > opticron > >
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