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Ship it! Just minor nits. Ship it! /branches/12/channels/chan_pjsip.c <https://reviewboard.asterisk.org/r/4008/#comment23883> The ternary operator is redundant here. Just do generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHO_INVITE); /branches/12/res/res_pjsip_session.c <https://reviewboard.asterisk.org/r/4008/#comment23884> There's a red blob at the end of this line. - Mark Michelson On Sept. 19, 2014, 5:04 p.m., Joshua Colp wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4008/ > ----------------------------------------------------------- > > (Updated Sept. 19, 2014, 5:04 p.m.) > > > Review request for Asterisk Developers and Mark Michelson. > > > Repository: Asterisk > > > Description > ------- > > Currently it is possible for ast_sip_session_refresh to be called at times > where the state of the dialog and INVITE session does not allow it to send a > request. Trying to send a request actually results in an assertion within > PJSIP. This change adds additional checks for deferral of these, stops > generating new SDP on COLP UPDATEs, makes it so deferral does not always > result in SDP generation, and ensures that after a provisional response that > any pending UPDATE occurs. > > * Note: Currently there is still a bug within pjproject which causes an > UPDATE without SDP sent after a provisional response to cancel the pending > SDP negotiation when it should not. This has been reported to Teluu and a fix > is being worked on. > > > Diffs > ----- > > /branches/12/res/res_pjsip_session.c 423546 > /branches/12/channels/chan_pjsip.c 423546 > > Diff: https://reviewboard.asterisk.org/r/4008/diff/ > > > Testing > ------- > > Modified the dialing API to change callerID at certain points (after call but > before handling responses, after handling responses). Confirmed that new code > correctly defers sending COLP updates. > > > Thanks, > > Joshua Colp > >
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