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

[Selim]

Have you tried to register directely to you VoIP provider using a soft
phone (such as
X-Lite) to check that your account (user/passwd) is activated ?

You can also try different values for the fromdomain parameter ...

On Tue, 14 Sep 2004 12:45:07 +0200, Evert Meulie <evert at witelcom.com> wrote:
> 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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> >>
> 
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