[asterisk-users] R: GXP2000 and asterisk 1.0.9

[C F]

On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito <hdevito at mchsi.com> wrote:
> 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

I'm not sure you are right, since I have had Polycoms that didn't
work, it's quite possible you should have GPXs that do work.

>  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
>  >
>  >
>  > 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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>  No virus found in this incoming message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008
>  15.20
>
>
>  No virus found in this outgoing message.
>  Checked by AVG Free Edition.
>  Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008
>  20.00
>
>
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