VoIP Mailing List Archives :: View topic - [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)

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[asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)

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Joe at myl2n.com
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Posted: Thu May 08, 2008 11:22 am Post subject: [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080508/fffb4f5f/attachment.htm

ploeppky at porchlight.ca
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Posted: Thu May 08, 2008 11:41 am Post subject: [asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP

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:

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.

Thanks in advance..
-Joe

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