[Asterisk-Users] RE: SOLVED: No audio when dialing in via PRI with Q.SIG
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
SV: [Asterisk-Users] Updated: No audio when dialing in via PRIwithQ.SIG
I've already tried that, but the result is the same... :( I've also seen the same error reported a long time ago, on this link: http://lists.digium.com/pipermail/asterisk-users/2004-August/053365.html. But I can't find a solution anywhere... Best regards, Peter Olsson Visionutveckling AB Från: [EMAIL PROTECTED] genom Alexander Lopez Skickat: ti 2006-04-25 18:07 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: RE: [Asterisk-Users] Updated: No audio when dialing in via PRIwithQ.SIG Add an Answer() as your first step in your dialplan and see if that help. snip ___ --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 !DSPAM:444e4a53124942303717053! ___ --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
[Asterisk-Users] 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
[Asterisk-Users] No sound in one calling direction, men using PRI with E1 and Q.SIG
I've been trying lots of configurations now. And the problem that I can't solve is this: I have a Digium T205P card. I have connected one of the connections to our internal PBX (NEC 2000 IPS). The Asterisk is configured as pri_cpe, and the NEC is configured to be the network side of the connection. Both ends are using b-channels 1-15 and 17-31, the d-channel is on 16. When I start everything, the link is ok on both ends, and it says that the D-channel is up (on both ends). Now, when I try to dial from our internal PBX to the Asterisk, the call connects ok, but there is no sound. But when I dial from the Asterisk to our NEC PBX everything works just fine, and the sound is working perfectly. One more strange thing is that when I dial from the NEC to Asterisk, every time a new B-channel is connecting the call (which I guess is normal). But it NEVER uses channel 31, it skips from 30 to 1. But then it seems to try to connect the call on channel 16, which is the D-channel(!), and that of course fails. Both ends seem to be setup correctly, and since the D-channel is initialized correctly, both ends must be using the correct channel for this, but still Asterisk tries to connect the incomming call B-channel on channel 1-30, instead of 1-15,17-31. Could this be caused by something in the Q.SIG protocol? I have used the NEC PBX with other PBX's, using Q.SIG. And everything has been working just fine. I've tried to look at the PRI debug output, but not much help there... What information is needed from me to get any help with this? 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
