On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote:

Hi,
          We have a PRI connection & when its was on test networks we had echo problems withoutside line. 

So I bought a TE212P card resolve the echo problem.  Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4.


But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear.

I am not sure whats the problem.  Also there's slight echo when calling Digium support.

Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for..

Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP.


---
Matthew Fredrickson
Software Engineer
Digium, Inc.

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to