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

Julien REVERET jreveret at osiatis.com
Wed Jul 18 10:58:19 CEST 2007


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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-- 
Julien REVERET
OSIATIS - LIFO https://www.lifo-osiatis.com






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