Multi-site internal calls end early when send to voicemail

Unanswered Question
Sep 24th, 2009

Hi,

We have a multi-site configuration with Nuvox SIP trunks.  If you make a call from an internal extension at one site to an extension at the other site and it ends up in voicemail, you hear the beginning of the voicemail greeting and the user's name - then it ends the call abrubtly, never giving you a chance to leave a message.

Same-site voicemail calls work.  Must be a codec issue, but oddly every 1 in 10 calls works.

Debugs attached.  The first file is the head end, the second is the receiving end.  The receiving end file is split in half - the top is one that happened to actually work, the bottom is a call that dropped before completion.

Thanks,

Joe

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Steven Smith Fri, 09/25/2009 - 07:32

Can you post a debug ccsip messages for a working and a broken call as well?

Joe Gadell Mon, 09/28/2009 - 15:08

Steve,

Here is a working & broken call "debug ccsip messages" from the receiving end.

Thanks.

Joe Gadell Mon, 09/28/2009 - 21:20

Opened a TAC case.  They fixed by adding modifying the dial peer configuration to the following for site-to-site calls.  We had originally gotten this configuration from the Cisco SBCS doc "Multi Site Deployment for Data nad Voice for SBCS" ...

Here is the text from the engineer:

At the main and second sides, the incoming and outgoing dial-peers were using the H323 protocol and the dial-peer go to AA or voicemail is using the SIP protocol. Although we have configured the allow H323 to SIP, but we are facing some problems.

What I have changed is that I change the dial-peer protocol and dtmf-relay information under the incoming and outgoing dial-peer between 2 sides to the same as the voicemail dial-peer. And also I removed the commands: voice source-group CCA_SIP_SOURCE_GROUP

Which may cause some unknown issues.

Change:

Main site:

  dial-peer voice 3055 voip

           description OUTGOING TO EUREKA EXT 20XX

           destination-pattern 20..

           session target ipv4:10.0.1.2

           dtmf-relay h245-alphanumeric

           codec g711ulaw

           no vad

add session-protocol sipv2

change dtmf-relay sip-notify

Second site:

  dial-peer voice 3050 voip

           description INCOMING FROM STL TO 20XX

           incoming called-number 20..

dtmf-relay h245-alphanumeric

           codec g711ulaw

           no vad

add session-protocol sipv2

change dtmf-relay sip-notify

TAC said that they weren't exactly sure why H323->SIP wasn't working properly, but that this would solve the issue.  CUE's dial-peer is configured similarly.

Thanks.

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