[Asterisk-Users] how to route these outgoing calls?

[Evert Meulie]

Tried that. Now I get:

Sip read:
SIP/2.0 403 Forbidden (From header is not a Trust host or gateway)
From: "Evert"<sip:<yourUsername>@<yourProvider>>;tag=as0687982f
To: <sip:069101701@<yourProvider>>;tag=87f2a0d5-13c4-4146e66c-1a8baa18-5e5
Call-ID: 2112969600583cc607abf82955a3de49 at 217.13.2.82
CSeq: 102 INVITE
Via:SIP/2.0/UDP 217.13.2.82:5060;branch=z9hG4bK3dc10bb5
Content-Length:0

>You can try this:
>
>In your sip.conf add the following entry
>
>[yourProvider]
>type = peer
>secret = <yourPassword>
>username = <yourUsername>
>host = <yourProvider>
>fromuser = <yourUsername>  ; some prviders need this parameter
>fromdomain = <yourProvider>  ; some prviders need this parameter
>
>In your extension.conf add the following entry:
>
>exten => _NXXXXXXX,1,Dial(SIP/${EXTEN}@yourProvider,30,r)
>
>This config is only for outgoing calls.
>
>On Tue, 14 Sep 2004 10:01:37 +0200, Evert Meulie <evert at witelcom.com> wrote:
>  
>
>>Hi everyone!
>>
>>I now have obtained a couple of SIP-accounts at a local VOIP-provider.
>>How do I specify that ALL outgoing calls to _NXXXXXXX go out via one of
>>these accounts?
>>
>>Regards,
>>   Evert
>>
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>>    
>>