[Asterisk-Users] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g722ulaw On asterisk (if this is a right example of pass-thru utilization), I download the codec from http://kvin.lv/pub/Linux/Asterisk/freebsd/ (my processor is a Sempron 2.2, then I download codec_g729-gcc-athlon-sse.so and codec_g729-gcc-debug.so files) and put it in my codec directory /usr/local/lib/asterisk/modules/. I remove the dummy codec first, then on sip.conf: disallow=all allow=g729 allow=alaw allow=ulaw The ISP sip services have support of g729. When I try to make a call from cisco phone to ISP, I see something on CME that seems codec g729 doesn't work: barahir#sh voice call summary PORT CODECVAD VTSP STATEVPM STATE == === == 2/0.1 - - - 2/0.2 - - - 2/1.1 - - - 2/1.2 - - - 50/0/1 .1 g711alaw n S_CONNECT EFXS_CONNECT 50/0/1 .2 - - - EFXS_ONHOOK 50/0/2 .1 - - - EFXS_INIT 50/0/2 .2 - - - EFXS_INIT 50/0/3 .1 - - - EFXS_ONHOOK 50/0/4 .1 - - - EFXS_ONHOOK 50/0/4 .2 - - - EFXS_ONHOOK Where is my mistake? Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDchFRMakHrsrHP9wRAoElAKDxrAxtMOOyRLO6kWaG/hvLVwAj8QCfW/TO LkuPpXb7DVpjUkoi6uV1PNU= =qwXR -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. -Greg On Wed, 2005-11-09 at 16:10 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g722ulaw On asterisk (if this is a right example of pass-thru utilization), I download the codec from http://kvin.lv/pub/Linux/Asterisk/freebsd/ (my processor is a Sempron 2.2, then I download codec_g729-gcc-athlon-sse.so and codec_g729-gcc-debug.so files) and put it in my codec directory /usr/local/lib/asterisk/modules/. I remove the dummy codec first, then on sip.conf: disallow=all allow=g729 allow=alaw allow=ulaw The ISP sip services have support of g729. When I try to make a call from cisco phone to ISP, I see something on CME that seems codec g729 doesn't work: barahir#sh voice call summary PORT CODECVAD VTSP STATEVPM STATE == === == 2/0.1 - - - 2/0.2 - - - 2/1.1 - - - 2/1.2 - - - 50/0/1 .1 g711alaw n S_CONNECT EFXS_CONNECT 50/0/1 .2 - - - EFXS_ONHOOK 50/0/2 .1 - - - EFXS_INIT 50/0/2 .2 - - - EFXS_INIT 50/0/3 .1 - - - EFXS_ONHOOK 50/0/4 .1 - - - EFXS_ONHOOK 50/0/4 .2 - - - EFXS_ONHOOK Where is my mistake? Any advice will be appreciated Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDchFRMakHrsrHP9wRAoElAKDxrAxtMOOyRLO6kWaG/hvLVwAj8QCfW/TO LkuPpXb7DVpjUkoi6uV1PNU= =qwXR -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me? http://www.nesys.it/snap/debug_voice_ccapi.txt thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDch8XMakHrsrHP9wRAkO2AJ9W15cGdtnWF+oWl0Yd/ai7HTHs+wCg1oUD X8BxszRaAVFpPkQzd1w5jEg= =Jsnv -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDciEJMakHrsrHP9wRArwvAJ9/lz+D1xVL8WnU3dyNLfpkh62nJwCgm8DD /9HE2UKACZ/OOJkZpC8c6Ss= =+5Iw -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
Post up your dial-peer 500 config as well. It is doing codec 0x2 (g.711Alaw) from the get go. Also post relevant config for the phone from asterisk and dialplan entry used. -Greg On Wed, 2005-11-09 at 17:08 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me? http://www.nesys.it/snap/debug_voice_ccapi.txt thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDch8XMakHrsrHP9wRAkO2AJ9W15cGdtnWF+oWl0Yd/ai7HTHs+wCg1oUD X8BxszRaAVFpPkQzd1w5jEg= =Jsnv -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsoreby Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
Just put codec g729(whatever version you need) in your dialpeer. I do not see what the voice-class codec 1 is without that section. -Greg On Wed, 2005-11-09 at 17:17 +0100, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME. Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDciEJMakHrsrHP9wRArwvAJ9/lz+D1xVL8WnU3dyNLfpkh62nJwCgm8DD /9HE2UKACZ/OOJkZpC8c6Ss= =+5Iw -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] problem with g729 and CME-Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 5:18 PM, Greg Oliver wrote: Post up your dial-peer 500 config as well. It is doing codec 0x2 (g.711Alaw) from the get go. Also post relevant config for the phone from asterisk and dialplan entry used. the call flows are: [ISDN in only] -- ntte [CME] [VOIP in] -- 5600 [asterisk] -- 601 [CME] (codec g711a)* [VOIP out] -- [asterisk] -- CME (codec g729 if possible) * multiple sip UA are registered with forwarding to 5600 -- 601 on CME maybe that's not a pass-thru solution, that is maybe I could'n use g729 without license, isn't it? Cisco (only voip out): ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711alaw codec preference 3 g711ulaw ! dial-peer voice 500 voip description ITA through Messagenet destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad ! ephone-dn 3 number 603 secondary xx no-reg label Home call-forward noan timeout 30 ! Asterisk: sip.conf - [general] context=cme-pbx language=it realm=sip.nesys.it port=5060 bindaddr=192.168.17.10 srvlookup=yes useragent=Nesys Asterisk PBX disallow=all allow=g729 allow=alaw allow=ulaw tos=0xb8 nat=yes register = xxx:[EMAIL PROTECTED]:5061/5600 ... [5600] type=friend host=192.168.17.10 dtmfmode=rfc2833 canreinvite=yes context=myphones qualify=yes [cme-pbx] type=peer canreinvite=no host=192.168.17.1 qualify=yes [60x] type=friend language=it host=dynamic dtmfmode=rfc2833 canreinvite=yes [EMAIL PROTECTED] context=myphones qualify=yes [messagenet-MI-out] context=cme-pbx type=friend language=it username=xxx fromuser=xxx fromdomain=sip.messagenet.it secret=yyy host=sip.messagenet.it port=5061 nat=yes canreinvite=no insecure=very qualify=yes extensions.conf - --- [myphones] include = cme-pbx include = messagenet-ITA-out [messagenet-ITA-out] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,r) exten = _X.,2,Playback(invalid) exten = _X.,3,Hangup [cme-pbx] exten = _6XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _6XX,2,Playback(invalid) exten = _6XX,3,Hangup exten = 5600,1,Dial(SIP/601,45) include = messagenet-ITA-out I know, that's a complicated implementation, the confs will be better ;) Thanks for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFDcitSMakHrsrHP9wRAuf3AJ9q1TAQcngi5h+rlBzviWs5/GsjugCfd+fG J87utc0S2yvKZT27w/cn4Dc= =RJnY -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
