----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3753/#review12614 -----------------------------------------------------------
Ship it! /team/group/media_formats-reviewed-trunk/main/channel.c <https://reviewboard.asterisk.org/r/3753/#comment22899> This may have been wrong anyway for the surviving channel. After a masquerade, the surviving channel could be using the preferred codec instead of what was negotiated. The guts of SIP/100 (clonechan) masquerades into SIP/200 (original) for call pickup. SIP/100 (codec list: gsm, alaw) and was using alaw After the pickup masquerade it would be using gsm. - rmudgett On July 11, 2014, 9:53 p.m., Matt Jordan wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3753/ > ----------------------------------------------------------- > > (Updated July 11, 2014, 9:53 p.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > ------- > > When a masquerade occurs, a newly created channel replaces an existing > channel and steals its private data structure. This includes swapping the > formats and capabilities. > > Since the newly created channel only contains ast_format_none, this causes an > assert to fire when the masqueraded channel is destroyed. > > Removing the assert isn't a good idea. However, there's also no real need to > do the accessing that fires the asserts either: the masqueraded channel will > be destroyed, the references will be cleaned up appropriately, and life will > go on. free_translation is also called in a channel destructor, and again, > things will be cleaned up appropriately without going through the accessors > that have the asserts. > > > Diffs > ----- > > /team/group/media_formats-reviewed-trunk/main/channel.c 418435 > > Diff: https://reviewboard.asterisk.org/r/3753/diff/ > > > Testing > ------- > > The crashing test (tests/apps/dial/dial_dtmf_hangup_cancel) now passes > > > Thanks, > > Matt Jordan > >
-- _____________________________________________________________________ -- 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
