We had attempted adding the 'r' to the dial command parameter and that
didn't seem to have an effect. We played around with the progressinband
a bit and tried to see if we could find a solution and only ended up
with same results no matter if it were set to "yes", "no", or "never".
We set everything back to where it was in the beginning and it seems to
be working now somehow. It has been running just fine and ringing since
midday yesterday. Thanks for the help Philipp and Faisal!
On 7/26/2010 11:37 PM, Faisal Hanif wrote:
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
--
*Chris Ramirez*
TELE-ONE COMMUNICATIONS, INC.
[email protected]
903-531-0777
--
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