[Asterisk-Dev] SIP Extrange Problem

Conroy, Lawrence (SMTP) lwc at roke.co.uk
Thu Feb 26 17:01:57 MST 2004

Hi Sergio, Folks,
  "after a few seconds" is interesting.
I wonder if this is a re-INVITE issue?

First, switch on "sip debug" in the asterisk console, and
see if there are sip messages approximately when the audio dies.

One quick way to *TEST* this is to:
*   set the phones in sip.conf to canreinvite=no,
*   edit rtp.c to always return -2, (a couple of lines before the 
existing conditional -2 return).

That way, * should always stay in the call and should NOT try to drop 
out by sending a re-INVITE to the
phones after the initial call setup.
Then retry the calls to see if the problem has "gone away".

If so, then the re-INVITE issue has bitten you. It can cause strange 
effects just like you're seeing (especially with some kinds of NAT/ALG 
between the phones). The problem is in the NAT/ALG, but it's
triggered by the re-INVITE behaviour of *.

all the best,

On 26 Feb 2004, at 10:07 am, Sergio Serrano Revuelto wrote:

> Hi all,
>     For a few days we have a very extrange problem. We have an 
> intranet with Budgetone and others SIP Phones.
>  In the extranet We Have Budgetone Phones. The whole system was 
> working well between the extranet and the intranet until a few days 
> ago.
>  When we try to speak with a Budgetone of the intranet, we can speak 
> during a few seconds but after a time the audio is cut in the sense of 
> intranet-extranet.
>  The problem is not only it, but if a budgetone of the intranet speaks 
> with another phone of the intranet the same thing happens.
>  After a time of conversation the audio is cut in the sense of the 
> budgetone to another phone. I see the next meesage in debug:
> Feb 26 10:50:04 DEBUG[50193]: Didn't get a frame from channel: 
> SIP/707-996a
> I have checked the files of configuration. It does not appear at all 
> any more in the files of logs and I do not know that to do.
>  Can it be a problem of the internal network? of the switches? Is 
> there any bug in the budgetones?
> Any idea?
> Thanks,
> srsergio

Registered Office: Roke Manor Research Ltd, Siemens House, Oldbury, Bracknell,
Berkshire. RG12 8FZ

The information contained in this e-mail and any attachments is confidential to
Roke Manor Research Ltd and must not be passed to any third party without
permission. This communication is for information only and shall not create or
change any contractual relationship.

More information about the asterisk-dev mailing list