[Asterisk-Users] Asterisk Newbie ...

[Andy Powell]

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
>>
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>> Asterisk-Users at lists.digium.com
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>>
>
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