Trying to configure Asterisk 11 Cert with G722.1C. I have installed the latest binary for Siren14: srv-echo*CLI> siren14 show version Digium Siren14 Module Version 11.0_1.0.5 (optimized for opteron_sse3_64) According to this list post in 2012 Asterisk supports G.722.1 Annex C (also known as Siren14), and this can be configured in a config file as either 'g7221c' or 'siren14': http://lists.digium.com/pipermail/asterisk-users/2012-February/270218.html However when setting allow=g7221c Asterisk will throw the following error: [Mar 27 00:32:39] WARNING[1950]: frame.c:821 ast_parse_allow_disallow: Cannot allow unknown format 'g7221c' [Mar 27 00:32:39] WARNING[1950]: chan_sip.c:31786 reload_config: Codec configuration errors found in line 9 : allow = g7221c The setting for allow=siren14 does work. The issue is the Polycom VVX600 phones do not support Siren14, rather the G722.1C implementation. The module was loaded properly as well: [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1004 load_module: ITU G.722.1 Annex C (Siren14, licensed from Polycom) transcoding module version 11.0_1.0.5 (This seems like a good start) [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1005 load_module: Copyright (C) 1999-2009 Digium, Inc. [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1006 load_module: This module is supplied under a commercial license granted by Digium, Inc. [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1007 load_module: Please see the full license text supplied in the accompanying [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1008 load_module: "LICENSE" file, or ask for a copy from Digium. [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1009 load_module: This product includes software from the Speex library. Please see [Mar 26 12:59:11] NOTICE[51170]: codec_siren14.c:1010 load_module: the accompanying "SPEEX_LICENSE" file for license information. [Mar 26 12:59:11] == Registered translator 'siren14tolin16' from format siren14 to slin16, table cost, 960000, computational cost 14044 [Mar 26 12:59:11] == Registered translator 'lin16tosiren14' from format slin16 to siren14, table cost, 825000, computational cost 19581 [Mar 26 12:59:11] == Registered translator 'siren14tolin8' from format siren14 to slin, table cost, 960000, computational cost 18572 [Mar 26 12:59:11] == Registered translator 'lin8tosiren14' from format slin to siren14, table cost, 825000, computational cost 17958 [Mar 26 12:59:11] Loaded codec_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom) Encoder/Decoder (optimized for opteron_sse3_64)) When trying to setup a call with just allow=siren14 the VVX600 would offer this: [Mar 26 13:12:03] --- (16 headers 14 lines) --- [Mar 26 13:12:03] Sending to XX.XX.XX.XX:38447 (NAT) [Mar 26 13:12:03] Using INVITE request as basis request - 53bbb6bb-1866e166-f at 10.10.10.75 [Mar 26 13:12:03] Found peer '0004F' for '0004F' from XX.XX.XX.XX:38447 [Mar 26 13:12:03] == Using SIP RTP TOS bits 184 [Mar 26 13:12:03] == Using SIP RTP CoS mark 5 [Mar 26 13:12:03] Found RTP audio format 9 [Mar 26 13:12:03] Found RTP audio format 0 [Mar 26 13:12:03] Found RTP audio format 8 [Mar 26 13:12:03] Found RTP audio format 18 [Mar 26 13:12:03] Found RTP audio format 127 [Mar 26 13:12:03] Found audio description format G722 for ID 9 [Mar 26 13:12:03] Found audio description format PCMU for ID 0 [Mar 26 13:12:03] Found audio description format PCMA for ID 8 [Mar 26 13:12:03] Found audio description format G729 for ID 18 [Mar 26 13:12:03] Found audio description format telephone-event for ID 127 [Mar 26 13:12:03] NOTICE[46737][C-00000bce]: chan_sip.c:10423 process_sdp: No compatible codecs, not accepting this offer! The Polycom RealPresence Trio 8800 however does support Siren14, and calls do work: [Mar 26 13:16:49] User-Agent: Polycom/5.4.1.17597 PolycomRealPresenceTrio-Trio_8800-UA/5.4.1.17597 [Mar 26 13:16:49] Allow-Events: conference,talk,hold [Mar 26 13:16:49] Accept-Language: en [Mar 26 13:16:49] Content-Type: application/sdp [Mar 26 13:16:49] Content-Length: 247 [Mar 26 13:16:49] [Mar 26 13:16:49] v=0 [Mar 26 13:16:49] o=- 1459020408 1459020408 IN IP4 10.241.125.120 [Mar 26 13:16:49] s=Polycom IP Phone [Mar 26 13:16:49] c=IN IP4 10.241.125.120 [Mar 26 13:16:49] t=0 0 [Mar 26 13:16:49] a=sendrecv [Mar 26 13:16:49] m=audio 2234 RTP/AVP 115 101 [Mar 26 13:16:49] a=rtpmap:115 G7221/32000 [Mar 26 13:16:49] a=fmtp:115 bitrate=48000 [Mar 26 13:16:49] a=rtpmap:101 telephone-event/8000 [Mar 26 13:16:49] a=sendrecv [Mar 26 13:16:49] <-------------> [Mar 26 13:16:49] --- (14 headers 11 lines) --- [Mar 26 13:16:49] Found RTP audio format 115 [Mar 26 13:16:49] Found RTP audio format 101 [Mar 26 13:16:49] Found audio description format G7221 for ID 115 [Mar 26 13:16:49] Found audio description format telephone-event for ID 101 [Mar 26 13:16:49] Capabilities: us - (ulaw|g729|g722|siren14), peer - audio=(siren14)/video=(nothing)/text=(nothing), combined - (siren14) [Mar 26 13:16:49] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Transcoding from G722 to Siren14 did also throw this error when using jitterbuffer going to a VVX600: [Mar 26 13:16:51] WARNING[51171][C-00000bc4]: abstract_jb.c:284 ast_jb_put: SIP/0004F-000013bd received frame with invalid timing info: has_timing_info=0, len=0, ts=0, src=lin16tog722 Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... 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