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(Updated June 25, 2014, 2:42 a.m.) Status ------ This change has been marked as submitted. Review request for Asterisk Developers, Joshua Colp and Matt Jordan. Changes ------- Committed in revision 417208 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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