[SR-Users] R: Call subscriber online

Stefano Larosa stefano.larosa at 10993.it
Tue Mar 15 12:03:26 CET 2011


Hello,

 

thank you for your answer.

 

I’ve added in my route the lookup("location") and the subscriber phone
rings.

 

route[TOPROXYUSER] {

 

                               xlog("L_NOTICE", "TOPROXYUSER $mi
route[$rm][0] $fu -> $ru START PROCESSING MESSAGE\n");

                

                               if (is_method("BYE|CANCEL")) { 

                                                route(FAIL_ONE);

                               } else if (is_method("INVITE")){ 

                                                if (!lookup("location")) {

 
sl_send_reply("404", "Not Found");

                                                                       exit;

                                               }else{    

                                                               route(RELAY);


                                               };

                                }; 

                               exit;

}

 

My first call was ringing but with no rtp audio so I’ve installed rtpproxy
with 

Apt-get install rtpproxy

 

And then started RTPPROXY

rtpproxy -l _your_public_ip_ -s udp:localhost:7722 –F

 

after that everything seems to work fine.

 

Thank you,

Stivu.

 

 

Da: Daniel-Constantin Mierla [mailto:miconda at gmail.com] 
Inviato: lunedì 14 marzo 2011 10.43
A: Stefano Larosa
Cc: sr-users at lists.sip-router.org
Oggetto: Re: [SR-Users] Call subscriber online

 

Hello,

shouldn't the call go to location service before relaying to subscriber B?
Is B at a fix address an port and that is local host port 5060? Are you
doing all in your computer for testing purposes, because otherwise an
application bound to localhost (like could be the softphone B) cannot really
communicate with the inter/intra-network?

Cheers,
Daniel


On 3/11/11 4:50 PM, Stefano Larosa wrote: 

Hi,

 

I’m new on Kamailio 3.0

 

This is the scenario I would like to build:

 

 1 Subscriber A -> 2 kamailio -> 3 asterisk -> 4 Kamailio -> 5 Subscriber B

 

Everything is working fine until the last step

 

This is the code that manage the call from asterisk to kamailio

 

if(is_method("INVITE") && (src_ip==80.169.xx.xx) )

    {

                              route(TOPROXYUSER);

    }

 

And this is the code that should end the call the the subscriber

 

route[TOPROXYUSER] {

                               xlog("L_NOTICE", "$mi route[$rm][0] $fu ->
$ru START PROCESSING MESSAGE\n");

                               rewritehostport("127.0.0.1:5060");

                               if (is_method("BYE|CANCEL")) { 

                                                route(FAIL_ONE);

                               } else if (is_method("INVITE")){ 

                                               route(RELAY);

                               }; 

                               exit;                      

}

 

 

Thank you,

 

Stivu.

 

 
 
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users at lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users





-- 
Daniel-Constantin Mierla
http://www.asipto.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110315/b8d3700f/attachment.htm>


More information about the sr-users mailing list