VoIP Mailing List Archives

[g.grandis at invidea.it]

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:

[g.grandis at invidea.it]

Thanks Henry,
anyway the phone is always registered when i get the busy tone

* Name : 502
Secret : <Set>
MD5Secret : <Not set>
Context : local
Language : it
FromUser :
FromDomain :
Callgroup : 1 (2)
Pickupgroup : 1 (2)
Mailbox :
LastMsgsSent : -1
Dynamic : Yes
Expire : 703 seconds
Expiry : 900
Insecure : No
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
DTMFmode : info
LastMsg : 0
ToHost :
Addr->IP : 192.168.13.171 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Username : 502
Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263)
Codec Order : (alaw|ulaw|gsm|g729|g723)
Status : OK (22 ms)
Useragent : Grandstream GXP2000 1.1.5.15
Full Contact : sip:502 at 192.168.13.171:5060;transport=udp;user=phone

Any idea?

Thanks again to all
-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Henry Devito
Inviato: mercoled? 13 febbraio 2008 22.01
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

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

[Guest]

Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have not seen the issue that you are describing, but I had some very strange issues like call hang-ups when I was using INFO. After switching the issues were gone and I have had no further troubles.

Hope this helps you.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giordano Grandis
Sent: Thursday, February 14, 2008 3:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9

Thanks Henry,
anyway the phone is always registered when i get the busy tone

* Name : 502
Secret : <Set>
MD5Secret : <Not set>
Context : local
Language : it
FromUser :
FromDomain :
Callgroup : 1 (2)
Pickupgroup : 1 (2)
Mailbox :
LastMsgsSent : -1
Dynamic : Yes
Expire : 703 seconds
Expiry : 900
Insecure : No
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
DTMFmode : info
LastMsg : 0
ToHost :
Addr->IP : 192.168.13.171 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Username : 502
Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263)
Codec Order : (alaw|ulaw|gsm|g729|g723)
Status : OK (22 ms)
Useragent : Grandstream GXP2000 1.1.5.15
Full Contact : sip:502 at 192.168.13.171:5060;transport=udp;user=phone

Any idea?

Thanks again to all
-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Henry Devito
Inviato: mercoled? 13 febbraio 2008 22.01
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9

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

[hdevito at mchsi.com]

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: 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: