g.grandis at invidea.it
Guest
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 Posted: Tue Mar 18, 2008 11:42 am Post subject: [asterisk-users] [SOLVED] GXP2000 and asterisk 1.0.9 |
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Switching the dtmf mode to RFC2833 solved my problem, thanks a lot Sam
Good work everyone
-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Lutgring, Sam
Inviato: gioved? 14 febbraio 2008 13.55
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9
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
| 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
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