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]+@.*") # 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@free.fr To: users@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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Hello,

El Wednesday 18 July 2007 10:02:26 Marc LEURENT escribió:

...

Does anyone succeed in redirecting SIP calls like [0-9] *@sip.test.com to a SIP/PSTN gateway provider without using asterisk?

I suposse you need to use de UAC module. http://www.openser.org/docs/modules/stable/uac.html

You only need UAC module if digest authentication is mandatory (and UAC module has some limitations with authentication).

The easiest way is that your provider authenticates your traffic by source ip address (and maybe a prefix).

Regards.

Saludos JesusR.

------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr@voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------