shmaltz at gmail.com
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 Posted: Thu Feb 14, 2008 7:36 pm Post subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 |
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On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito <hdevito at mchsi.com> wrote:
| Quote: | 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
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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.
| Quote: | 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: mercoled? 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:
| Quote: |
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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