12 msgapp_queue New Function ToggleQueueMemberUse()
1 msgDial Command with the L switch again
1 msgVUC Friday May 9 @ 12 Noon EDT: Asterisk 3rd pa...
2 msgRe: Newbie IVR: How toread()beforeplayback()is ...
2 msg-zapg729toulaw did not update samples 160
1 msgt38modem
1 msgZaptel ring voltage detection
3 msgText for built-in recordings
1 msgI hear noise in the line
1 msgRe: Lucent Max TNT PRI Agg --> * --> SIP ...
2 msgMOH and Licensed G729 codec
4 msgRe: Out-Going Callerid
12 msgZap Channels Collide (Incoming & Outgoing)
6 msgWhich Cepstral Voice to license
1 msgchan_sip Maximum retries exceeded on transmission

Lucent Max TNT PRI Agg --> * --> SIP DEV ...
\ Joe Carroll (8 May 2008)
. \ Peter (8 May 2008)

7 msghelp with rotating number plan
8 msgManager API - Setvar not working
13 msgSLN File Format
4 msgNewbie Queue: tricky problem with MOH
Subject:Re: Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)
Group:Asterisk-users
From:Peter
Date:8 May 2008


 
I've found an interesting link. It might help you out.

http://www.cisco.com/en/US/docs/ios/solutions_docs/voip_solutions/EA_ISD.html

Peter

Joe Carroll wrote:
> Hello...
> We're attempting to track down an intermittent echo issue. Our setup is
> <phone>sip<asterisk>sip<tnt>pri to carriers. We have less than 2 ms latency on the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is the subscriber using sip. The PSTN users does not hear the echo.
>
> We should be note that there is zero echo when calling sip to sip with or without reinvites enabled.
>
> We have several different phones; linksys, polycom, & grandstream (both atas and phones). It's difficult to reproduce the problem regularly so isolation is an issue.
>
>
> Thanks in advance..
> -Joe
>
>
>
>
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