Tuesday, September 18, 2007

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...

[root@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@xorcom.com >:
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?

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