-----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 CODEC VAD VTSP STATE VPM 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-----
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 CODEC VAD VTSP STATE VPM 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
-----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-----
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