[Asterisk-Dev] SIP Extrange Problem

[Conroy, Lawrence (SMTP)]

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,
AND
*   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,
   Lawrence

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
>  

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