Matthew Schumacher
2005-Aug-23 11:21 UTC
[Asterisk-Users] Can't get G729 working after buying a license.
List, I purchased 2 g729 licenses but I can't get it to answer a g729 call from a cisco router with a vwic card. In the debug output below you will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) when it should support g729 according to the config also listed below. The real odd thing is I can place g729 calls to the router, just not from the router to *. Anyone have any ideas on how to fix this? Another problem I am having is I want to use the info dtmf mode, but the sip packet that asterisk sends does not announce info in the Allow string. Thanks, schu in debug: 20 headers, 13 lines Using latest request as basis request Sending to 192.168.77.254 : 5060 (non-NAT) Found no matching peer or user for '192.168.77.254:49206' Found RTP audio format 18 Found RTP audio format 101 Found RTP audio format 19 Peer audio RTP is at port 192.168.77.254:16494 Found description format G729 Found description format telephone-event Found description format CN Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (g723), peer - 0x3 (g723|gsm), combined - 0x1 (g723) Aug 23 09:54:43 NOTICE[1379]: chan_sip.c:2792 process_sdp: No compatible codecs! Transmitting (no NAT): SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.77.254:5060 From: <sip:874@192.168.77.254>;tag=4194CB3C-F91 To: <sip:9999@192.168.11.17>;tag=as4ebd30b1 Call-ID: CB361638-133511DA-988CF03C-BF8FDD9A@192.168.77.254 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:9999@192.168.11.17> Content-Length: 0 in sip.conf: [router] type=friend context=default host=192.168.77.254 dtmfmode=info disallow=all allow=g729 nat=no canreinvite=yes qualify=yes in debug: [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx == Found license 'G729-XXXXXXXX' providing 2 channels == Found total of 2 G.729 licenses == Registered translator 'g729tolin' from format g729 to slin, cost 2 == Registered translator 'lintog729' from format slin to g729, cost 11
Michael D Schelin
2005-Aug-23 12:08 UTC
[Asterisk-Users] Can't get G729 working after buying a license.
Call Digum. They support the license codec install. Matthew Schumacher wrote:>List, > >I purchased 2 g729 licenses but I can't get it to answer a g729 call >from a cisco router with a vwic card. In the debug output below you >will see that asterisk thinks it only supports: (gsm|ulaw|alaw|h263) >when it should support g729 according to the config also listed below. > >The real odd thing is I can place g729 calls to the router, just not >from the router to *. Anyone have any ideas on how to fix this? > >Another problem I am having is I want to use the info dtmf mode, but the >sip packet that asterisk sends does not announce info in the Allow string. > >Thanks, >schu > >in debug: >20 headers, 13 lines >Using latest request as basis request >Sending to 192.168.77.254 : 5060 (non-NAT) >Found no matching peer or user for '192.168.77.254:49206' >Found RTP audio format 18 >Found RTP audio format 101 >Found RTP audio format 19 >Peer audio RTP is at port 192.168.77.254:16494 >Found description format G729 >Found description format telephone-event >Found description format CN >Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 >(g729)/video=0x0 (nothing), combined - 0x0 (nothing) >Non-codec capabilities: us - 0x1 (g723), peer - 0x3 (g723|gsm), combined >- 0x1 (g723) >Aug 23 09:54:43 NOTICE[1379]: chan_sip.c:2792 process_sdp: No compatible >codecs! >Transmitting (no NAT): >SIP/2.0 488 Not acceptable here >Via: SIP/2.0/UDP 192.168.77.254:5060 >From: <sip:874@192.168.77.254>;tag=4194CB3C-F91 >To: <sip:9999@192.168.11.17>;tag=as4ebd30b1 >Call-ID: CB361638-133511DA-988CF03C-BF8FDD9A@192.168.77.254 >CSeq: 101 INVITE >User-Agent: Asterisk PBX >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >Contact: <sip:9999@192.168.11.17> >Content-Length: 0 > >in sip.conf: >[router] >type=friend >context=default >host=192.168.77.254 >dtmfmode=info >disallow=all >allow=g729 >nat=no >canreinvite=yes >qualify=yes > >in debug: >[codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec >Translator) > == G.729 Host-ID: >xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx:xx > == Found license 'G729-XXXXXXXX' providing 2 channels > == Found total of 2 G.729 licenses > == Registered translator 'g729tolin' from format g729 to slin, cost 2 > == Registered translator 'lintog729' from format slin to g729, cost 11 > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >