[asterisk-users] GXP2000 and asterisk 1.0.9

C F shmaltz at gmail.com
Wed Feb 13 14:09:00 CST 2008


Just check DND if it's on on the phone or not.
What is the CLI output when you try making a phone call?
Why don't you try it with a later version of astrisk and a Phone?

On Feb 13, 2008 10:58 AM, Giordano Grandis <g.grandis at invidea.it> wrote:
>
>
> Hi all gusy,
> i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few
> go in "busy" state, if you call it get the busy tone but the phone can male
> any type of call.
> This is my sip.conf
>
> [502]
> language = it
> username = 502
> secret = <password>
> host = dynamic
> type = friend
> context = local
> canreinvite = yes
> dtmfmode = info
> callgroup = 1
> pickupgroup = 1
> callerid = 502 <502>
>
> Under Grandstream's support suggest, I set "Use randmom port" to yes and
> "Nat traversal (STUN)" to "No, but send keep alive" but without success.
> This is the firmware version: Program-- 1.1.5.15    Bootloader-- 1.1.5.6
>
> Anyone can help me ?
>
> Thanks in advance
>
> Giordano
>
>
> No virus found in this outgoing message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008
> 15.20
>
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