[Asterisk-Users] Asterisk Newbie ...

[Julien]

Done ...

[1945]
type=friend
username=1945
secret=1945
host=dynamic
context=from-sip
mailbox=1945
defaultip=192.168.0.6     (and if i use DHCP ???)

But it doesn't work well. It's strange, because calls between ip phones or
soft phones work well ...but the voice mail is unavailable..
I'm searching .... and thanks for your help.

Julien.
----- Original Message ----- 
From: "Andy Powell" <andy at beagles-den.demon.co.uk>
To: <asterisk-users at lists.digium.com>
Sent: Sunday, August 10, 2003 4:49 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...


Julien,

try adding defaultip=<ip of phones> in your sip.conf for each phone
definition

Andy

*********** REPLY SEPARATOR  ***********

On 10/08/2003 at 16:28 Julien wrote:

>Yes, the voice mail is at 2999 , but it doesnt work when i call it from
>the
>ata .I talked about the 600 (echo test) but i removed it from the
>extension.conf, sorry.
>
>In sjphone , i've got this error
>
>15:06:42 INFO Session rejected. Reason: 404 Not Found
>15:06:42 INFO Call 153 ended: Session rejected: 404 Not Found
>15:06:42 INFO SIP: Session terminated.
>
>And the configuratin of sjphone works, i can call the ata from this pc.
>On the console the 2999 is available, i can acces my voicemail.
>
>I think something is missing ... but it's difficult for me to understand
>how
>everything works in asterik. I read all night long documentation on the net
>, and the handbok-draft without succes.
>For the moment i don't want to use a card to connect to my phone line ...
>Just 1 ata with 2 phones , and a soft phone with their own voicemail.
>
>Julien.
>
>----- Original Message ----- 
>From: "Florian Overkamp" <florian at obsimref.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Sunday, August 10, 2003 3:47 PM
>Subject: Re: [Asterisk-Users] Asterisk Newbie ...
>
>
>> At 15:13 10-8-2003 +0200, you wrote:
>> >If i want to call the sjphone from the ata or call the ata from de
>sjphone
>> >everything is ok.
>> >My problem is ,that i can't call the voicemail or any other phone number
>> >..as 600 for exemple from the ata or the jphone.
>> >I don't know why but i looked after a long time..
>> >
>> >[from-sip]
>> >exten => 1943,1,Dial(SIP/1943,5)
>> >exten => 1943,2,Voicemail(u1943)
>> >exten => 1943,102,Voicemail(b1943)
>> >exten => 1943,103,Hangup
>> >
>> >exten => 1945,1,Dial(SIP/1945,6)
>> >exten => 1945,2,Voicemail(u1945)
>> >exten => 1945,102,Voicemail(b1945)
>> >exten => 1945,103,Hangup
>> >
>> >exten => 2999,1,VoicemailMain(${CALLERIDNUM})
>>
>> Call me silly, but this does mean your Voicemailbox is at extension 2999,
>> not 600. Or did I misunderstand you ?
>>
>> BTW, any other phonenumber not being callable would make sense, since you
>> simply don't have anything else in the [from-sip] context. You could
>> include the IAXtel gateway or add a device to connect to your phone
>line...
>>
>> Florian
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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