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:[EMAIL PROTECTED] http://personales.igalia.com/agomez IGALIA, S.L. http://www.igalia.com _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
