[asterisk-users] Re: Asterisk hangs up after 10-15 minutes whenSIPPhone is on mute

[Rich Adamson]

Several Linksys models have had a problem in the past allowing multiple 
devices on the inside lan to nat properly with something on the outside wan.

Ordinarily a sip phone on the inside of the lan attempts to register 
with an external asterisk box, and the Linksys keeps track of source IP, 
source port, destination IP, and destination port. (That is part of 
every nat box.) But, on some Linksys models, they do not seem to track 
the source info, thus two sip phones appear exactly the same from an 
outside perspective. The issue can be seen in several forms including 
multiple sip phones, vpn clients, etc.  Not sure exactly which models 
fall into the category, but I know from experience there have been 
multiple models over the years.

You might also check to be sure your running the latest firmware on the 
Linksys.


Mike wrote:
> That would be problematic.  I am using a cheap Linksys router where my
> Polycom 501 is located and I see no such setting.  It probably is hardcoded.
> 
> Can I force the Polycom 501 to send empty RTP packet?
> 
>  (actually, I tried using comfort noise but I got an asterisk error message
> rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC
> 3389). Please turn off on client if possible. Client IP: xx.xxx.xxx.xx
> 
> Mike 
>