[asterisk-users] Problem with Zap-Sip calls.

[Faisal Hanif]

  You may need to add "r" as option perameter to dial command.

Regards,

Faisal Hanif

On 7/26/2010 9:39 PM, Chris Ramirez wrote:
> The problem we are having with Asterisk is when we initiate a call via 
> a Zap line and it goes out on a Sip line. When it goes out via Sip we 
> hear no sound until the party we are calling answers the line. If the 
> call were to go out Sip-Sip or Zap-Zap it works perfectly fine. It is 
> only with the Zap-Sip calls. If anyone knows anything that could 
> possibly help it would be greatly appreciated. I have checked many 
> different things already and tried comparing Zap-Zap and Zap-Sip call 
> logs. Thanks!
> -- 
> *Chris Ramirez*
> TELE-ONE COMMUNICATIONS, INC.
> cramirez at tele-onecom.com
> 903-531-0777
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