----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4362/#review14269 -----------------------------------------------------------
After reading through the analysis on the underlying ASTERISK issue, I don't have any findings with the patch. I'm always a little concerned when we have to add a new state to keep track of on the sip_pvt, but right now I can't think of another property that would be appropriate. It'd probably be good for someone who has spent more time in the chan_sip transfer code to look at this as well, just to make sure I'm not missing anything. - Matt Jordan On Jan. 20, 2015, 12:36 p.m., Jeremiah Gowdy wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4362/ > ----------------------------------------------------------- > > (Updated Jan. 20, 2015, 12:36 p.m.) > > > Review request for Asterisk Developers and Matt Jordan. > > > Bugs: https://issues.asterisk.org/jira/browse/ASTERISK-22436 > > https://issues.asterisk.org/jira/browse/https://issues.asterisk.org/jira/browse/ASTERISK-22436 > > > Repository: Asterisk > > > Description > ------- > > chan_sip: This patch fixes a bug in chan_sip's handling of Invite: Replaces > which currently never hangs up on the replaced call. It adds an additional > flag to track the fact that we're doing a replaces and then uses that flag to > determine if we should send a BYE. > > > Diffs > ----- > > /branches/11/channels/sip/include/sip.h 430836 > /branches/11/channels/chan_sip.c 430836 > > Diff: https://reviewboard.asterisk.org/r/4362/diff/ > > > Testing > ------- > > This is running in production for a beta product we have now. Our > development and QA staff have done manual testing and found no issues. > > > Thanks, > > Jeremiah Gowdy > >
-- _____________________________________________________________________ -- 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
