Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA)


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
jmr.richardson at gmai...
Guest





PostPosted: Thu May 08, 2008 5:17 pm    Post subject: [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP Reply with quote

Quote:
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.


I had intermittent echo when I first deployed TNT's as well. It took
a while to track down. Adjust the volume on the TNT lower until the
echo goes away. Here is what I had to set mine to:

In each T1 config:

set line-interface voip-gain-control input-pad = 3db-loss
set line-interface voip-gain-control output-pad = 3db-loss

Hope this helps.

JR
--
JR Richardson
Engineering for the Masses
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services