No one can help me?
A hint or a documentation to read, please!
Thank you
Max
Il giorno 27/nov/2014, alle ore 13:19, Massimo Varriale (IPZeta) ha scritto:
> Hi,
> is there someone could give me a hint on how to configure this module?
> I can see that the call is going out to my external gateway interrogating the
> routing table, however I'm getting a CANCEL from Kamailio.
>
> If using this cr_route I can see that users can't place calls themself
> (internal calls) because all calls are going out to the external IP.
> In my opinion the cr_route should be placed replacing PSTN route, right?
> However I'm still getting the problem.
>
> Thanks
> Max
>
>
>
>
>
> Inizio messaggio inoltrato:
>
>> Da: "Massimo Varriale (IPZeta)" <m.varri...@ipzeta.it>
>> Oggetto: [Kamailio 4.2.0] Carrieroute Routing Questions
>> Data: 25 novembre 2014 16:27:05 GMT+01:00
>> A: sr-users@lists.sip-router.org
>>
>> Hi Guys,
>> I'm new on Kamailio and this list, so be patient with me :)
>>
>> I've built an almost "complete" working SIP server on Ubuntu 14.04 LTS.
>> I told almost complete because my problem is with carrierroute module
>> because I'm not understanding the routing file and where I should put the
>> code to use function cr_route
>>
>> I would like to allow SIP calls/video between users for free and send calls
>> to different external Gateways / Switches, etc based on the dialled
>> destination.
>>
>> In particular, I can send outbound calls a prepaid platform that will allow
>> to bill calls based on CLI validation, but in all my tests, it seems that
>> calls are going out always to the IP address of the remote gateway found
>> into the carrierroute table ignoring the SIP user I'm calling.....what is
>> the correct routing?
>>
>> I'm finding some docs around there, but some refer to very old versions of
>> Kamailio and some others are not clear to me.
>>
>> What's wrong with my routing logic?
>>
>> Thank you so much!
>> Max
>>
>>
>>
>>
>>
>>
>>
>> route {
>>
>> # per request initial checks
>> route(REQINIT);
>>
>> # NAT detection
>> route(NAT);
>>
>> # handle requests within SIP dialogs
>> route(WITHINDLG);
>>
>> ### only initial requests (no To tag)
>>
>> # CANCEL processing
>> if (is_method("CANCEL"))
>> {
>> if (t_check_trans())
>> t_relay();
>> exit;
>> }
>>
>> t_check_trans();
>>
>> # authentication
>> route(AUTH);
>>
>> # record routing for dialog forming requests (in case they are routed)
>> # - remove preloaded route headers
>> remove_hf("Route");
>> if (is_method("INVITE|SUBSCRIBE"))
>> record_route();
>>
>> # account only INVITEs
>> if (is_method("INVITE"))
>> {
>> setflag(FLT_ACC); # do accounting
>> }
>>
>> # dispatch requests to foreign domains
>> route(SIPOUT);
>>
>> ### requests for my local domains
>>
>> # handle presence related requests
>> route(PRESENCE);
>>
>> # handle registrations
>> route(REGISTRAR);
>>
>> if ($rU==$null)
>> {
>> # request with no Username in RURI
>> sl_send_reply("484","Address Incomplete");
>> exit;
>> }
>>
>> # dispatch destinations to PSTN
>> #route(PSTN);
>>
>>
>> # CARRIERROUTE MODULE routing logic
>> # check table for carrier default and domain default
>> if(!cr_route("default", "default", "$rU", "$rU", "call_id")){
>> sl_send_reply("403", "Not allowed");
>> } else {
>> # In case of failure, re-route the request
>> t_on_failure("1");
>> # Relay the request to the gateway
>> t_relay();
>> }
>>
>>
>>
>> # user location service
>> route(LOCATION);
>>
>> route(RELAY);
>> }
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
