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shmaltz at gmail.com
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 Posted: Wed Feb 13, 2008 3:09 pm Post subject: [asterisk-users] GXP2000 and asterisk 1.0.9 |
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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
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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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hdevito at mchsi.com
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| Posted: Wed Feb 13, 2008 4:01 pm Post subject: [asterisk-users] GXP2000 and asterisk 1.0.9 |
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Is your phone actually registered to the switch. go to the CLI and do a
'sip show peers' see if extension 502 is registered. Making an outbound
call does not prove that the phone is registered.
----- Original Message -----
From: "C F" <shmaltz at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Wednesday, February 13, 2008 2:09 PM
Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9
| Quote: | 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
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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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