Re: [asterisk-users] Problem with Zap-Sip calls.
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. crami...@tele-onecom.com 903-531-0777 -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 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
[asterisk-users] Problem with Zap-Sip calls.
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. crami...@tele-onecom.com 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
Re: [asterisk-users] Problem with Zap-Sip calls.
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. -- _ -- 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
Re: [asterisk-users] Problem with Zap-Sip calls.
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. crami...@tele-onecom.com 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
