[SR-Users] Route to PSTN no works

[BALL SUN]

Hello All

I am also start to integrate Kamailio IMS with PSTN, may I know my
understanding below is correct or not?

1. Inbound calls need to point to the I-CSCF
2. Outbound gateways are defined in Dispatcher on the Serving-CSCF

Also, what configuration should I change to support the above configuration?

any help is appreciated.

- RBK

On Tue, Jul 7, 2020 at 2:21 PM Daniel-Constantin Mierla
<miconda at gmail.com> wrote:
>
> 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
>         # check if PSTN GW IP is defined
>         xlog("L_INFO","PSTN ACTIVADO");
>         if (strempty($sel(cfg_get.pstn.gw_ip))) {
>                 xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n");
>                 return;
>         }
>
>         # route to PSTN dialed numbers starting with '+' or '00'
>         #     (international format)
>         # - update the condition to match your dialing rules for PSTN routing
>         if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){
>                 xlog("L_INFO", "Error en el formato numerico!!");
>                 return;
>         }
>
>         # only local users allowed to call
>         if(from_uri!=myself) {
>                 sl_send_reply("403", "Not Allowed");
>                 exit;
>         }
>
>         # normalize target number for pstn gateway
>         # - convert leading 00 to +
>         #if (starts_with("$rU", "00")) {
>         #       strip(2);
>         #       prefix("+");
>         #}
>
>         if (strempty($sel(cfg_get.pstn.gw_port))) {
>                 #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
>                 xlog("L_INFO","SELECCION CON PUERTO");
>                 $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
>                                         + $sel(cfg_get.pstn.gw_port);
>         } else {
>                 xlog("L_INFO","SELECCION CON PUERTO");
>                 $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
>                                         + $sel(cfg_get.pstn.gw_port);
>         }
>
>         route(RELAY);
>         exit;
> #!endif
>
>         return;
> }
>
> And in my request_route:
>
>  remove_hf("Route");
>         if (is_method("INVITE|SUBSCRIBE")) {
>                 if($src_ip != "IP ASTERISK"){
>                         xlog("L_INFO", "***********ROUTE PSTN***********");
>                         route(PSTN);
>                 } else {
>                         xlog("L_INFO","LLamada desde $si con puerto $sp");
>                         record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060");
>                         add_rr_param(";r2=on");
>                         route(DISPATCH);
>                         route(RELAY);
>                 }
>         }
>
> But never see that the call go to PSTN route..
>
> I'd made any wrong??
>
> Thanks
>
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>
> --
> 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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