Re: [asterisk-users] Randomly half-voice at sip/zap
Ok, so i made the terminal screen wider, but during the call nothing changes: ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* Rx: 10736 (10736) Tx: 0 (0) What could be the reason? THx 2007/10/10, Mojo with Horan Company, LLC [EMAIL PROTECTED]: Péter Tóth wrote: When i try ztmonitor as follows, it gives strange output... [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* R ###* R If ztmonitor keeps scrolling down the screen, you need to make your terminal wider. The '#' marks should jump back and forth left and right like a level monitor, and there will only be one row of them (but with two levels, one for RX and one for TX). The screen won't scroll at all. Try this again :) Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Hi! Today one more strange thing happened. An incoming ISDN call's voice become like a robot (or when you hear very fast chopping?!), in both direction ... I did bri debug, but i can't see anything. After amportal stop and service zaptel restart the problem terminated. Has anybody similar experience? Thanks! _ Tóth Péter Tel.: +36703834578 _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Horrible problem - calls losing sound
Hi John! I have the same problem, the system contains 1 port Billion ISDN BRI card, and 1 sip trunk. This is a trixbox with Asterisk 1.2.22-BRIstuffed-0.3.0-PRE-1y-i The ISDN call is forwarded to a ring-group. The 6 sip phones are welltech lp399 series. If incoming the call get wrong, we can not hear the other side, but they hear us. In my case the rtp debug shows there are no incoming rtp packets from asterisk to SIP phone. If somebody experienced this problem, please help US! Thanks! 2007/9/20, John Hughes [EMAIL PROTECTED]: We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI). We're using Asterisk 1.4.11 (1.4.11-BRIstuffed-0.4.0-test4) on a Debian system, with chan-capi 1.0.1. Any idea what could be going wrong, where to look c? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Randomly half-voice at sip/zap
Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The incoming route directed to a ring group. Sometimes the incoming calls - from pstn - are not, the caller do not hear any voice from us. When i call out on the sip line, it happens indirectly, so i can't hear nothing from the other side, especially when i call my sip telco provider. (10 try, 2 wrong) If they're calling me, everything is ok! Please help me! Thanks in advance! _ Peter Toth _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Randomly half-voice at sip/zap
What do you mean on direct call? The error is more frequently on my sip trunk. Should I make a sip debug? My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem? Anyway i will watch the bri debug, and try to make a wrong and a correct call. Thanks 2007/9/18, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Sep 18, 2007 at 10:20:14AM +0200, Péter Tóth wrote: Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The incoming route directed to a ring group. Sometimes the incoming calls - from pstn - are not, the caller do not hear any voice from us. When i call out on the sip line, it happens indirectly, so i can't hear nothing from the other side, especially when i call my sip telco provider. (10 try, 2 wrong) If they're calling me, everything is ok! Is the call a direct call? Can you hear / see the audio in ztmonitor? The next step would probably be to enable 'bri debug span 1' and get traces from a good call and from a bad call. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Randomly half-voice at sip/zap
Hi! Yes, the echo test worked perfectly. When i try ztmonitor as follows, it gives strange output... [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* And so on... Is this normal? Thanks! 2007/9/18, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Sep 18, 2007 at 12:07:20PM +0200, Péter Tóth wrote: What do you mean on direct call? The error is more frequently on my sip trunk. Should I make a sip debug? My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem? Anyway i will watch the bri debug, and try to make a wrong and a correct call. Can you successfully call an echo-test extension? (Echo() ) from SIP? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
