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 _________________
On Tue, Sep 18, 2007 at 10:20:14AM +0200, P?ter T?th 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#16849755 jabber:tzafrir at jabber.org +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
Hi! Yes, the echo test worked perfectly. When i try ztmonitor as follows, it gives strange output... [root at asterisk1 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 <tzafrir.cohen at xorcom.com>:> > On Tue, Sep 18, 2007 at 12:07:20PM +0200, P?ter T?th 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? >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070918/dba9d475/attachment.htm