[asterisk-users] [SOLVED] GXP2000 and asterisk 1.0.9

Tue Mar 18 11:42:02 CDT 2008

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: giovedì 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: mercoledì 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

> 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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>
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No virus found in this incoming message.
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