[asterisk-users] Strange Problem

[Danny Nicholas]

Monitor the successful and failing calls from a CLI session with "core set
verbose 5".  This should show you what is different between the two calls.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru
Oniciuc
Sent: Monday, February 08, 2010 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Strange Problem

 

Hello list!

 

I've run into a strange problem today and I was hoping that someone here has
seen this before and maybe can give me a hand:

 

I'm using asterisk 1.6.0.22 in this config:

 

(A)PATTON ISDN ->(B) ASTERISK -> (C)PATTON PRI -> PSTN -> (D)OTHER PBX

 

Strange Problem:

 

USER A calls makes a call to a PBX over the PSTN and ends into an IVR. When
the user makes a selection and gets his call passed to an extension of that
PBX (USER D), USER D has no sound while USER A hears the voice just fine.

 

 If USER A makes a direct call to USER D, calling directly his extension,
the call has audio both ways and its all working fine.

The same thing if USER A calls directly mobile phones or numbers that aren't
managed by IVRs.

 

I've verified this with a few PBXs(different manufacturers), and the problem
is there every time an IVR gets the control of the call.

 

A sip debug in asterisk confirmed that the SIP Session is not renegotiated
when the call exits USER's D IVR and ends up to his extension.

 

Any idea what might be causing this?

 

Thank you in advance!

 

Alex

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