Tuesday, September 18, 2007

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 <tzafrir.cohen@xorcom.com>:
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:tzafrir@jabber.org
+972-50-7952406           mailto:tzafrir.cohen@xorcom.com
http://www.xorcom.com  iax:guest@local.xorcom.com/tzafrir

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