[SR-Users] Route to PSTN no works

[Daniel-Constantin Mierla]

Hello,

which log message you see in the syslog?

***********ROUTE PSTN***********

or:

PSTN ACTIVADO

or none of them?

Cheers,
Daniel

On 03.07.20 21:21, sip user wrote:
> Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams
> calls.
> For Teams -> Asterisk calls I'd worked using extension and register
> Asterisk with that extension.
>
> But I'd like to use direct routing with IP.
>
> In kamailio.cfg I activate define WITH_PSTN.
> I configured the IP and PORT for my PSTN.
>
> I'm using the default route[PSTN]:
>
> route[PSTN] {
> #!ifdef WITH_PSTN
> Â  Â  Â  Â  # check if PSTN GW IP is defined
> Â  Â  Â  Â  xlog("L_INFO","PSTN ACTIVADO");
> Â  Â  Â  Â  if (strempty($sel(cfg_get.pstn.gw_ip))) {
> Â  Â  Â  Â  Â  Â  Â  Â  xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not
> defined\n");
> Â  Â  Â  Â  Â  Â  Â  Â  return;
> Â  Â  Â  Â  }
>
> Â  Â  Â  Â  # route to PSTN dialed numbers starting with '+' or '00'
> Â  Â  Â  Â  # Â  Â  (international format)
> Â  Â  Â  Â  # - update the condition to match your dialing rules for PSTN
> routing
> Â  Â  Â  Â  if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){
> Â  Â  Â  Â  Â  Â  Â  Â  xlog("L_INFO", "Error en el formato numerico!!");
> Â  Â  Â  Â  Â  Â  Â  Â  return;
> Â  Â  Â  Â  }
>
> Â  Â  Â  Â  # only local users allowed to call
> Â  Â  Â  Â  if(from_uri!=myself) {
> Â  Â  Â  Â  Â  Â  Â  Â  sl_send_reply("403", "Not Allowed");
> Â  Â  Â  Â  Â  Â  Â  Â  exit;
> Â  Â  Â  Â  }
>
> Â  Â  Â  Â  # normalize target number for pstn gateway
> Â  Â  Â  Â  # - convert leading 00 to +
> Â  Â  Â  Â  #if (starts_with("$rU", "00")) {
> Â  Â  Â  Â  # Â  Â  Â  strip(2);
> Â  Â  Â  Â  # Â  Â  Â  prefix("+");
> Â  Â  Â  Â  #}
>
> Â  Â  Â  Â  if (strempty($sel(cfg_get.pstn.gw_port))) {
> Â  Â  Â  Â  Â  Â  Â  Â  #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
> Â  Â  Â  Â  Â  Â  Â  Â  xlog("L_INFO","SELECCION CON PUERTO");
> Â  Â  Â  Â  Â  Â  Â  Â  $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  + $sel(cfg_get.pstn.gw_port);
> Â  Â  Â  Â  } else {
> Â  Â  Â  Â  Â  Â  Â  Â  xlog("L_INFO","SELECCION CON PUERTO");
> Â  Â  Â  Â  Â  Â  Â  Â  $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  + $sel(cfg_get.pstn.gw_port);
> Â  Â  Â  Â  }
>
> Â  Â  Â  Â  route(RELAY);
> Â  Â  Â  Â  exit;
> #!endif
>
> Â  Â  Â  Â  return;
> }
>
> And in my request_route:
>
> Â remove_hf("Route");
> Â  Â  Â  Â  if (is_method("INVITE|SUBSCRIBE")) {
> Â  Â  Â  Â  Â  Â  Â  Â  if($src_ip != "IP ASTERISK"){
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  xlog("L_INFO", "***********ROUTE
> PSTN***********");
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  route(PSTN);
> Â  Â  Â  Â  Â  Â  Â  Â  } else {
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  xlog("L_INFO","LLamada desde $si con puerto $sp");
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  record_route_preset("FQND:5061;transport=tls",
> "IP KAMAILIO:5060");
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  add_rr_param(";r2=on");
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  route(DISPATCH);
> Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  Â  route(RELAY);
> Â  Â  Â  Â  Â  Â  Â  Â  }
> Â  Â  Â  Â  }
>
> But never see that the call go to PSTN route..
>
> I'd made any wrong??
>
> Thanks
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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