> On March 23, 2015, 3:01 p.m., Matt Jordan wrote: > > Thanks for the patch! I've clicked the Ship It button, although the same > > statement about requiring tests for things going into Asterisk 13 that I > > made on the DTMF review applies here as well. > > > > In this particular case, a test for this patch should be done using SIPp, > > as it is pretty easy to construct an inbound INVITE request and put an > > OPTION request in-dialog with that INVITE request. > > > > Most of the tests in channels/pjsip use SIPp to drive the tests, and so > > there is a lot of material to base a test on. We also have sample SIPp > > scenarios to use as a template in the contrib/sipp folder. > > > > If you have any questions about where to start with that, please don't > > hesitate to ask on the asterisk-dev mailing list/#asterisk-dev. > > Joshua Colp wrote: > A testsuite test has now been published for review at > https://gerrit.asterisk.org/#/c/37/
Since the test is up, I'm going to go ahead and commit this. yaron: Thanks a lot for the patch! - Matt ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4499/#review14774 ----------------------------------------------------------- On March 18, 2015, 4:01 a.m., yaron nahum wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4499/ > ----------------------------------------------------------- > > (Updated March 18, 2015, 4:01 a.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24862 > https://issues.asterisk.org/jira/browse/ASTERISK-24862 > > > Repository: Asterisk > > > Description > ------- > > Respond to OPTIONS message sent on an existing dialog with 200OK. > This feature is vital in order to interoperate with some switches that send > OPTIONS message periodically per active call to make sure it is still alive. > Not responding would cause the switch to disconnect the call. > This functionality used to work on the old SIP channel, but was not > implemented on PJSIP. > > > Diffs > ----- > > /trunk/res/res_pjsip_dlg_options.c PRE-CREATION > > Diff: https://reviewboard.asterisk.org/r/4499/diff/ > > > Testing > ------- > > > Thanks, > > yaron nahum > >
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