[Users] Using # for Sip 2 Sip calls

[Kenny Chua]

So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I  try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have:
  
  route[3] {
  
      # -----------------------------------------------------------------
      # INVITE Message Handler
      # -----------------------------------------------------------------
  
      if (!proxy_authorize("","subscriber")) {
          proxy_challenge("","0");
          return;
      } else if (!check_from()) {
          sl_send_reply("403", "Use From=ID");
          return;
      };
  
       consume_credentials();
  
      if (nat_uac_test("19")) {
          setflag(6);
      }
  
      lookup("aliases");
      if (uri!=myself) {
          route(4);
          route(1);
          return;
      };
      
      if (uri=~"^sip:\*[0-9]*@"){
          xlog("Sip 2 Sip\n");
          strip(1); #strip the * because we dont need it
          route(4);
          route(6);
          route(1);
          return;
          
      };
      
      if  (!lookup("location")){
      
          if (uri=~"^sip:[0-9]*@") {        # International PSTN
              xlog("PSTN Gateway\n");
              route(4);
              route(5);
              return;
          };
      
          sl_send_reply("404", "User Not Found");
          return;
      };
  
      route(4);
      route(1);
  }
  
  route[4] {
  
      # -----------------------------------------------------------------
      # NAT Traversal Section
      #  -----------------------------------------------------------------
  
      if (isflagset(6)) {
          force_rport();
          fix_nated_contact();
          force_rtp_proxy();
      }
  }
  
  route[5] {
  
      # -----------------------------------------------------------------
      # PSTN Handler
      # -----------------------------------------------------------------
      xlog("Routed to route 5\n");
      rewritehostport("pstn.gateway:5060");  
  
      avp_write("i:45", "inv_timeout");
  
      route(1);
  }
  
  route[6] {
      
      if (!proxy_authorize("","subscriber")) {
          proxy_challenge("","0");
           return;
      } else if (!check_from()) {
          sl_send_reply("403", "Use From=ID");
          return;
      };
  }
  
  
  onreply_route[1] {
  
      if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
          if (!search("^Content-Length:[ ]*0")) {
              force_rtp_proxy();
          };
      };
  
      if (nat_uac_test("1")) {
          fix_nated_contact();
      };
  }
  
  
 Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote: Hi,
 
 that's right.  For example SIPURA ATAs with two lines but online one 
 terminal use # for line selection....
 you better use a digit that does not overlap with the PSTN dialling plan.
 
 regards,
 bogdan
 
 Glenn Dalgliesh wrote:
 
 >Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice.
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 >
 >-----Original Message-----
 >From: Kenny Chua 
>Date: Wednesday, Jun 28, 2006 10:56 pm
>Subject: [Users] Using # for Sip 2 Sip calls
>
>Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?
> 
> I came up with something like this:  
>     lookup("aliases");
>     if (uri=~"^sip:#[0-9]*@"){
>         xlog("Sip 2 SIP\n");
>         route(4);
>         route(1);
>         return;
>     };
> 
> Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.
> 
>   
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>Content-Type: text/html; charset=iso-8859-1
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>
>Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?
 
 I came up with something like this: 
     lookup("aliases");
      if (uri=~"^sip:#[0-9]*@"){
         xlog("Sip 2 SIP\n");
         route(4);
         route(1);
         return;
     };
 
 Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.
   
>  
 
 			
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