When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar.
I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk it should open channel 16, when it according to Asterisk should be 17, since 16 is the D-channel. This mismatch then follows all the way up to the last channel. I've read some stuff about Q.SIG. And according to that information Q.SIG has the posibility to renumber b-channels, but Asterisk doesn't seem to care about that. I have connected our PBX to other PBX'es before, so I do know that the PRI/Q.SIG actually works with other implementations. For now I have changed chan_zap.c so that it loads the channels differently, when it configures the prioffset parameter, I just lower it by one, if it's greater than 15. This actually solved all my problems, and now both incoming and outgoing calls works just fine. I know this is not a good solutions in the long run, but it will have to do for the time being :) Mvh Peter Olsson Visionutveckling AB Tel: 0303-72 92 00 -----Ursprungligt meddelande----- Från: Peter Olsson Skickat: den 25 april 2006 17:41 Till: asterisk-users@lists.digium.com Ämne: Updated: No audio when dialing in via PRI with Q.SIG After lots of testing I discovered that I could get the sound to work. The only thing I had been testing was MeetMe and Voicemail. But when I dialed a SIP-phone, or routed back to other phones via the PRI interface, everything works just great! The problem only seem to occur when dialing directly into Asterisk, when Asterisk sends the audio output. I have also discovered that the PRI never seem to get the signal that the call has been connected when dialing into MeetMe, it thinks it's still in the ringing state - I've discovered this by watching TAPI events showing up on my other PBX. Is this some kinf of known bug in Asterisk? I guess it's because of this I won't get any sound on these calls.... When dialing to a SIP phone I get all information. If anyone have any idea, I'd appreciate it. If it helps I could also send some debug logs from ISDN. Best regards, Peter Olsson Visionutveckling AB _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
