----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/#review13371 -----------------------------------------------------------
Thanks for supplying a working test. I wouldn't commit it as-is, but I'd understand it if you don't want to clean it up yourself. /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <https://reviewboard.asterisk.org/r/4006/#comment23860> You're not Russel. /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <https://reviewboard.asterisk.org/r/4006/#comment23861> Unused. /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <https://reviewboard.asterisk.org/r/4006/#comment23862> PEP says: remove ' ' before ':'. -error_file is not populated because you don't have -trace_err. /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <https://reviewboard.asterisk.org/r/4006/#comment23863> - Remove red blob. - Why the sleep(10)? I see no benefits of this. /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test <https://reviewboard.asterisk.org/r/4006/#comment23867> This depends on the -pause_ign_msg. If we alter the test to retransmit manually, we wouldn't need to grep any logs. We could probably do without the run-tests entirely. See how sip_outbound_address/test-config.yaml is created. /asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml <https://reviewboard.asterisk.org/r/4006/#comment23866> Decrease this for more speed. But see my other comment about doing the retransmit manually. /asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml <https://reviewboard.asterisk.org/r/4006/#comment23864> Irrelevant comment. - wdoekes On Sept. 19, 2014, 3:51 p.m., Torrey Searle wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4006/ > ----------------------------------------------------------- > > (Updated Sept. 19, 2014, 3:51 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24335 > https://issues.asterisk.org/jira/browse/ASTERISK-24335 > > > Repository: testsuite > > > Description > ------- > > This is a test for the test suite to reproduce the issue described in > ASTERISK-24335 > > > Diffs > ----- > > /asterisk/trunk/tests/channels/SIP/tests.yaml 5608 > /asterisk/trunk/tests/channels/SIP/invite_retransmit/test-config.yaml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/sip.conf > PRE-CREATION > > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/extensions.conf > PRE-CREATION > > Diff: https://reviewboard.asterisk.org/r/4006/diff/ > > > Testing > ------- > > test passes when 4003 patch applied, fails when patch not applied > > > Thanks, > > Torrey Searle > >
-- _____________________________________________________________________ -- 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
