Andres Gómez García
2005-Feb-12 20:46 UTC
[Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
Hi all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've not luck (oRTP errors in console) even to p2p connection between 2 linphone client computers or sipomatic. I've tried GNOMEMeeting also. It works fine with a P2P client connections (ALSA works fine) but, even when I success connecting to an asterisk server, I haven't hear anything. I mean, I don't hear the demo successfull messages. I've looking the GNOMEMeeting logs and it says that it closes the sound channel as soon as it connects to the asterisk server. This is my h323.conf file: [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs context=default and my extensions.conf file: [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=Zap/g2 TRUNKMSD=1 . . . [demo] exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,BackGround(demo-congrats) exten => s,6,BackGround(demo-instruct) . . . [default] include => demo . . . I've also can see how asterisk says it actually plays these sound files in the CLI. Any idea? Thanks in advance. -- Andr?s G?mez Garc?a Ingeniero en Inform?tica Telf: +34 981 91 39 91 Fax: +34 981 91 39 49 mailto:agomez@igalia.com http://personales.igalia.com/agomez IGALIA, S.L. http://www.igalia.com
On Sun, 2005-02-13 at 04:46 +0100, Andres G?mez Garc?a wrote:> I've tried GNOMEMeeting also. It works fine with a P2P client > connections (ALSA works fine) but, even when I success connecting to an > asterisk server, I haven't hear anything. I mean, I don't hear the demo > successfull messages. I've looking the GNOMEMeeting logs and it says > that it closes the sound channel as soon as it connects to the asterisk > server. This is my h323.conf file:Had the same issue with Debian Sarge. I didn't actually investigate it, but I strongly suspect the openh323/pwlib packages don't work with the asterisk-h323 package. The H323 README specifically says btw to don't use the packages of the distribution but rather the versions recommended there. I finally decided to compile * 1.0.5 from scratch, as well as use chan_oh323 instead of chan_h323, and all works well now. As to the linphone problems, don't know, it should work. If not, it'd be rather a linphone issue. As to an IAX phone, the only choice on linux currently seems to be iaxcomm/iaxclient. For me, it's not really usable because of latency issues, but to test the * installation it'll suffice anyway. Regards, Bruno.