Re: [asterisk-users] Randomly half-voice at sip/zap
Péter Tóth wrote: > 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? > Maybe you're monitoring the wrong zap channel? If you put '1' to refer to group 1, that's not the way ztmonitor works -- it wants a specific zap channel number. Just an idea, I'm not sure why else it wouldn't work. Maybe set up a call and switch ztmonitor from channel to channel until you find the one that's in use. Further, I don't recall your setup -- make sure you're actually using a ZAP channel in your bridged call ;) Sorry if all this seems obvious, but I can't imagine any other reasons it wouldn't work. 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
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] Randomly half-voice at sip/zap
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
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
Re: [asterisk-users] Randomly half-voice at sip/zap
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? -- 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
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
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
[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
