Hello, can i use different audio codecs when i calling between sip devices ( snom phones ) and different when i making call from isdn to sip or from sip to isdn ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind is like a parachute... it only works when it's open.
Yes, you can, you have to modify a little bit chan_sip to set up a codec that you need for certain extensions. It's going to be hardcoded. ps. maybe this will be a feature soon in the asterisk regards Martin On 19 Feb 2003, Marian Danisek wrote:> Hello, > > can i use different audio codecs when i calling between sip devices ( > snom phones ) and different when i making call from isdn to sip or from > sip to isdn ? > > best regards > > Marian > > -- > SUNTEQ s. r. o. > Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic > Tel: +421-46-5430 754 # Fax: +421-46-5439 144 > http://www.sunteq.sk/ > ------------------------------------------------------------ > A mind is like a parachute... it only works when it's open. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I know that Digium has a deal in place with Voiceage to supply the G.729 codec per DS0 basis. Is there any chance that G.723 will be supported in a similar way? I've been "playing" with the G.723 code that comes with asterisk (for development purposes only) and have not been too impressesed with the voice quality (there is a definite "beating" with the decoded voice stream on the TDM side of the conversation). I'm hoping a Voiceage codec would be better optimized and tuned for x86 running on Linux. As a side note, I'm using X-lite as a client on my PC --> Asterisk X100P --> PSTN Thanks. Lenny Quick Link Communications Ltd. http://www.qlccom.com
I have just compiled H323 support in Asterisk .Now I get these errors ERROR[1024]: File chan_oh323.c, Line 1304 (load_module): CAP_NSUP_ER. WARNING[1024]: File loader.c, Line 273 (ast_load_resource): chan_oh323.so: load_module failed, returning -1 DEBUG[1024]: File chan_oh323.c, Line 1330 (unload_module): Cleaning up OpenH323 channel driver... DEBUG[1024]: File chan_oh323.c, Line 1334 (unload_module): Incoming calls = 0. DEBUG[1024]: File chan_oh323.c, Line 1335 (unload_module): Outgoing calls = 0. DEBUG[1024]: File chan_oh323.c, Line 1360 (unload_module): Done... WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module chan_oh323.so failed! Ouch ... error while writing audio data: : Broken pipe What could be the problem? Kenneth
Just went throught that. Add the LD_LIBRARY= for both pwlib and openh323 into your ENV, reboot, and KAPOW! laters, darran ----- Original Message ----- From: "Makerere University" <kkabagambe@tech.mak.ac.ug> To: <asterisk-users@lists.digium.com> Sent: Wednesday, April 30, 2003 12:42 PM Subject: [Asterisk-Users] H323> I have just compiled H323 support in Asterisk .Now I get these errors > > ERROR[1024]: File chan_oh323.c, Line 1304 (load_module): CAP_NSUP_ER. > WARNING[1024]: File loader.c, Line 273 (ast_load_resource): chan_oh323.so: > load_module failed, returning -1 > DEBUG[1024]: File chan_oh323.c, Line 1330 (unload_module): Cleaning up > OpenH323 channel driver... > DEBUG[1024]: File chan_oh323.c, Line 1334 (unload_module): Incoming calls > 0. > DEBUG[1024]: File chan_oh323.c, Line 1335 (unload_module): Outgoing calls > 0. > DEBUG[1024]: File chan_oh323.c, Line 1360 (unload_module): Done... > WARNING[1024]: File loader.c, Line 368 (load_modules): Loading module > chan_oh323.so failed! > Ouch ... error while writing audio data: : Broken pipe > > What could be the problem? > Kenneth > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I havn't used the h323 channel of Asterisk for a while, but today I needed to test a few things only I found out that Asterisk/H323 crashes my Siemens optipoint 400 phone. It seems to be the audio codecs that's causing it. Is something broken in chan_h323? - -- Regards, Tais M. Hansen ComX -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/Jl7g2TEAILET3McRAmTzAJwNCGxGKJI5ZK92Vnri0CUJM+hFpQCfZo5o vf6vMWBmYjseyvtczNuM688=8xpI -----END PGP SIGNATURE-----