> On June 25, 2014, 6:45 a.m., Joshua Colp wrote: > > /team/group/media_formats-reviewed-trunk/res/res_rtp_asterisk.c, lines > > 1683-1685 > > <https://reviewboard.asterisk.org/r/3671/diff/1/?file=60623#file60623line1683> > > > > SRSLY!
I actually has srsly at first, then decided that I'd be somewhat more "professional" (whatever that means). This check still annoys me. - Matt ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3671/#review12307 ----------------------------------------------------------- On June 25, 2014, 7:14 a.m., Corey Farrell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3671/ > ----------------------------------------------------------- > > (Updated June 25, 2014, 7:14 a.m.) > > > Review request for Asterisk Developers, Joshua Colp and Matt Jordan. > > > Repository: Asterisk > > > Description > ------- > > This update gives media_formats the ability to receive a call using chan_sip. > Possibly other channel drivers might work, I haven't tried them. > > * ast_format_cap_is_compatible_format needs to be checked against > AST_FORMAT_CMP_NOT_EQUAL, not zero/non-zero. All calls to > ast_format_cap_is_compatible_format were fixed. > * res_rtp_asterisk was updated by Matt Jordan, along with related changes to > codec.c, codec.h, format.c, format.c and codec_builtin.c. > * Switch ast_format_copy from function to macro to ao2_bump. This allows > REF_DEBUG to give better results. > > > Diffs > ----- > > /team/group/media_formats-reviewed-trunk/res/res_speech.c 417190 > /team/group/media_formats-reviewed-trunk/res/res_rtp_asterisk.c 417190 > /team/group/media_formats-reviewed-trunk/main/translate.c 417190 > /team/group/media_formats-reviewed-trunk/main/frame.c 417190 > /team/group/media_formats-reviewed-trunk/main/format.c 417190 > /team/group/media_formats-reviewed-trunk/main/codec_builtin.c 417190 > /team/group/media_formats-reviewed-trunk/main/codec.c 417190 > /team/group/media_formats-reviewed-trunk/main/channel.c 417190 > /team/group/media_formats-reviewed-trunk/main/bridge.c 417190 > /team/group/media_formats-reviewed-trunk/include/asterisk/format.h 417190 > /team/group/media_formats-reviewed-trunk/include/asterisk/codec.h 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_unistim.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_skinny.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_sip.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_pjsip.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_oss.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_nbs.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_motif.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_mgcp.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_jingle.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_h323.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_gtalk.c 417190 > /team/group/media_formats-reviewed-trunk/channels/chan_alsa.c 417190 > /team/group/media_formats-reviewed-trunk/addons/chan_ooh323.c 417190 > /team/group/media_formats-reviewed-trunk/addons/chan_mobile.c 417190 > > Diff: https://reviewboard.asterisk.org/r/3671/diff/ > > > Testing > ------- > > Called from Asterisk 11 to a test server with this code, I was able to hear > the 'invalid' message, everything seemed during the call. I received TONS of > ao2 frack's when stopping Asterisk. The sip.conf peer on both Asterisk > servers was setup for disallow=all / allow=ulaw. > > > Thanks, > > Corey Farrell > >
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