Error making call: Missing route set

[alberto.bitto@xxxxxxxxx (Alberto Bitto)]

Hello,

I'm trying to implement a very simple iOS app that uses pjsua. I
succesfully built and imported the pjsip library in my sample project, have
a Kamailio remote server up and running and now I'm trying to make a very
simple call to the server.

Using the code found on this tutorial
http://www.xianwenchen.com/blog/2014/06/30/how-to-make-an-ios-voip-app-with-pjsip-part-3/,
I do the following:

1) create pjsua status:

        status = pjsua_create();

2) init configuration and logging:

        pjsua_config cfg;
        pjsua_config_default (&cfg);

        cfg.cb.on_incoming_call = &on_incoming_call;
        cfg.cb.on_call_media_state = &on_call_media_state;
        cfg.cb.on_call_state = &on_call_state;

        pjsua_logging_config log_cfg;
        pjsua_logging_config_default(&log_cfg);
        log_cfg.console_level = 4;

        status = pjsua_init(&cfg, &log_cfg, NULL);

3) add UDP and TCP transport:

        pjsua_transport_config cfg;
        pjsua_transport_config_default(&cfg);
        cfg.port = 5080;

        status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);

(and the same for TCP)

4) start pjsua:

            status = pjsua_start();

5) register the account on the server:

            pjsua_acc_id acc_id;
            pjsua_acc_config cfg;

            pjsua_acc_config_default(&cfg);

            cfg.id = pj_str("sip:" SIP_USER "@" SIP_URL);
            cfg.reg_uri = pj_str("sip:" SIP_URL);
            cfg.cred_count = 1;
            cfg.cred_info[0].realm = pj_str(SIP_REALM);
            cfg.cred_info[0].scheme = pj_str("digest");
            cfg.cred_info[0].username = pj_str(SIP_USER);
            cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
            cfg.cred_info[0].data = pj_str(SIP_PASSWD);

            status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);

where SIP_URL is the IP of my Kamailio server and SIP_REALM = "*"

6) if everything is fine, make the call:

            pj_str_t uri = pj_str(SIP_URL ":" SIP_PORT);
            status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL
);

where SIP_PORT is the port 5061 on the Kamailio server.

When I run the app and actually make the call, on the log I can see the
pjsua status change from NULL up to RUNNING and the account successfully
added to the server; when it tries to make the call, after the sound media
setup, I get this error:

*Error making call: Missing route set (for tel: URI) (PJSIP_ENOROUTESET)
[status=171005]*

and pjsua starts to shut everything down.

What am I missing?

Thanks in advance.

Alberto Bitto
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