[asterisk-users] R: GXP2000 and asterisk 1.0.9

Henry Devito hdevito at mchsi.com
Thu Feb 14 09:12:12 CST 2008


I had GXP-2000's running on 1.0 versions of asterisk even earlier.  So I 
know it does work.  I upgraded one of my customers GXP's to the latest 
firmware in it still works.  Can you post the output of the CLI with verbose 
set to 99 and the the output from the asterisk log file that has the call in 
it.  You can usually do a 'tail /var/log/asterisk/full -n 400' right after 
the call fails.

I will be glad to help, just need a little more info to narrow down the 
issue.

Thanks
Henry


----- Original Message ----- 
From: "Giordano Grandis" <g.grandis at invidea.it>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, February 14, 2008 2:15 AM
Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9


1. The phone has not the DND active, i checked it several times
2. Outbound calls always success, the problem is when the phone receive a 
call, it repsnds with busy signalling.
3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade 
asterisk.

Thanks for all

-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com 
[mailto:asterisk-users-bounces at lists.digium.com] Per conto di C F
Inviato: mercoledì 13 febbraio 2008 21.09
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

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
>
>
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