[Asterisk-Dev] SIP Extrange Problem

[Sergio Serrano Revuelto]

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
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20040226/00851ea9/attachment.htm