[Asterisk-Users] Grandstream and Snome remote sip stops taking calls

[The VoIP Connection]

Jason,

Your NAT is closing on you, so you need to do something to keep it open.
With the snom you can register more often (every minute or so usually works)
or you can use QUALIFY. Grandstreams have a NAT keep-alive on the phone
which is enabled using NAT TRAVERSAL = YES.  This mechanism sends an empty
packet at a regular interval to your server keeping the NAT port open. The
default keep alive interval is usually fine. Note that if you have more than
one phone at a location with you should set USE RANDOM PORT = YES.

Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:611 at voiceserver.thevoipconnection.com


> -----Original Message-----
> From: Jason [mailto:jason at a2artifacts.com] 
> Sent: Wednesday, January 04, 2006 3:34 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc: 'Manny A. Wise'
> Subject: [Asterisk-Users] Grandstream and Snome remote sip 
> stops taking calls
> 
> 
> I have remote users that are setup to sip into the Asterisk server.
> Problem is that if you call there extension after they have 
> been registered For a while there phones don't ring.
> If I do a sip show peers they can be seen as registered in.
> Also the user can dial out.
> If they reset the phone they can receive calls.
> This seems to be more of an issue with the Grand stream phones.
> 
> The Grandstream has these two settings I am un sure of.
> NAT Traversal (STUN):  currently set to no SUBSCRIBE for MWI: 
> currently set to no
> 
> Any ideas?
> 
> -Jason
> 
> 
>