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 <[email protected]> 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 > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > 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 > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List [email protected] https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
