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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100726/4c7e0690/attachment.htm
> 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.Search for "progress" and/or "progressinband".
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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100727/745d59a9/attachment.htm