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

Mike list at virtutel.ca
Thu Sep 7 10:56:36 MST 2006

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


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dr. Michael J.
Sent: September 7, 2006 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Asterisk hangs up after 10-15 minutes
whenSIPPhone is on mute

Mike wrote:
> Thanks Tony.  Its possible that the phone stops sending RTP stream 
> (but it certainly is receiving some!). How do I get Asterisk to stop 
> caring whether it receives RTP or not?
> Yes there is a NAT between the phone the the Internet.  The Asterisk 
> server doesn't have NAT though.

My Sonicwall NAT/firewall has a 15 minute default inactivity timeout for TCP
NAT connections, which is suggestive (it can be increased, though). 
If the signaling vanishes in one direction, maybe your NAT device is
"forgetting" about the connection.

- Mike
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