[asterisk-users] GXP2000 and asterisk 1.0.9

Henry Devito hdevito at mchsi.com
Wed Feb 13 15:01:08 CST 2008


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


> 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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