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#16849755 jabber:[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 _________________
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