Error making call: Missing route set

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



Maybe because you're using port 5061 it requires additional SSL setup? 
Try 5060 perhaps? - Bill

On 1/12/2015 11:18 AM, Alberto Bitto wrote:
> 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_configcfg;
>         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_configlog_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_configcfg;
> 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_idacc_id;
> pjsua_acc_configcfg;
>
> 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_turi = 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
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
> pjsip at lists.pjsip.org
> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150112/f1be27ac/attachment.html>


[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux