[asterisk-users] Bad Echo between SIP calls
Matthew Fredrickson
creslin at digium.com
Mon Jun 11 16:25:57 CDT 2007
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.
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