[Asterisk-Users] Asterisk Newbie ...

Julien fabia at free.fr
Sun Aug 10 07:28:56 MST 2003


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
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>




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