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@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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Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users