[OpenSER-Users] Redirect to Trunk IP SIP/PSTN gateway + billing

[Marc LEURENT]

Thanks! But where can I put the login and password of my sip account of 
the gateway?
Best Regards,

Marc LEURENT

Julien REVERET a écrit :
> You can redirect calls to a PSTN gateway using this kind of routing :
> if (method=="INVITE")
>  {
>   if (uri=~"sip:011[0-9]+ at .*")  # Here we check the number dialed
>   {
>    #authorize if a call is going to PSTN 
>    if(!proxy_authorize("domain.net", "subscriber"))
>    {
>     proxy_challenge("domain.net", "0");
>     return;
>    };
>  
>    xlog("L_INFO", "CALL: Call to international number\n");
>    rewritehostport("voip_gw.domain.net:5060");  # rewriting SIP headers
>    route(1);
>   }
>
>
> By checking the uri and rewriting destination host you can route your PSTN calls to a PSTN gateway. The gateway can be an Asterisk PBX, a SIP/PSTN appliance or any kind of SIP provider, I guees you already know that. The example above is taken from openser and asterisk realtime integration.
>
> ----- Original Message -----
> From: "Marc LEURENT" <lftsy at free.fr>
> To: users at openser.org
> Sent: mercredi 18 juillet 2007 10 h 02 (GMT+0100) Europe/Berlin
> Subject: [OpenSER-Users] Redirect to Trunk IP SIP/PSTN gateway + billing
>
> Does anyone succeed in redirecting SIP calls like [0-9]*@sip.test.com to 
> a SIP/PSTN gateway provider without using asterisk?
> Thanks
>
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