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.
[email protected]
903-531-0777
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
