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

[Tom Vile]

What phone? What firmware version? and have you set qualify=yes for
the phones in question?
On 1/4/06, Jason <jason at a2artifacts.com> wrote:
>
> 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
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax:     518-631-2856