[asterisk-users] Randomly half-voice at sip/zap

Tue Sep 18 05:22:48 CDT 2007

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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               Tzafrir Cohen       
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