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(a)free.fr> To: users(a)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]*(a)sip.test.com to a SIP/PSTN gateway provider without using asterisk? Thanks _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Julien REVERET OSIATIS - LIFO https://www.lifo-osiatis.com

Hi, El Wednesday 18 July 2007 10:02:26 Marc LEURENT escribió: I suposse you need to use de UAC module. http://www.openser.org/docs/modules/stable/uac.html Ciao -- http://debaser.homelinux.com/