I am trying to setup a really simple (I hope so) IMS platform, with some basic services (like voicemail, IVR, videoconference), but I'm kind of stuck. I have installed Kamailio IMS on debian and a OpenIMS VM also. But my little problem begins when I try to configure a application server (in this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk), asterisk alone, and read some about Restcomm. For Elastix and asterisk, I get stuck trying to setup the trigger point. For Restcomm, I only find documentation for integration with Clearwater IMS. So, I need your help. Somebody has something like that working? What TAS (or components for a TAS) do you recommend? Tutorials, examples, manuals that you know can help me? Thanks for reading, and many many thanks for any advice. David
Hi David,
Sorry, for not answering earlier, but we've had some public holidays as well as KamailioWorld last week, so the time was quite limited... :-)
I've attached a really simple example for an AS implemented in Kamailio. The functionality is really minimalistic:
1) if the dialed number (RURI) starts with a "0", strip the first character in RURI and prepend "49" for Germany 2) Forward the INVITE back to the network (I know some carriers would actually deploy an full-blown Oracle SBC for this functionality, since they are not allowed to use open-source... :-D )
You can basically use any Endpoint you want as an AS, as long as it speaks SIP. In case the AS terminates the session (e.g. in order to play an announcement or for a Video-Conference), it's rather simple. The AS just has to answer the session, no big deal.
If you want to forward your request back to the network, it becomes a little bit more complicated, e.g.:
U 2015/06/01 10:22:13.419694 178.62.224.219:5060 -> 178.62.224.185:5060 INVITE sip:040524759398@ims.voiceblue.com SIP/2.0. Record-Route: sip:mo@178.62.224.219;lr=on;ftag=0FFE09DE-556C163500064B51-1F6EF700;did=d51.5492. Route: sip:mmtel-1.ams.voiceblue.com:5060;lr, sip:iscmark@scscf-1.ams.voiceblue.com;lr;s=1;h=0;d=0;a=74656c3a343934303436383937373434. [...]
The AS has to evaluate the Route Header (the first Route is itself), the next Route is the originating S-CSCF. In order to do proper Routing, your AS has to keep the Route-Header intact, when returning the SIP-Message back to the network, so the S-CSCF can determine, that this request was coming from an AS. You will have to modify the SIP-Stack from Freeswitch or Asterisk to achieve this, but that should be not too hard.
Thanks, Carsten
2015-05-20 3:43 GMT+02:00 dfretes@ing.una.py:
I am trying to setup a really simple (I hope so) IMS platform, with some basic services (like voicemail, IVR, videoconference), but I'm kind of stuck. I have installed Kamailio IMS on debian and a OpenIMS VM also. But my little problem begins when I try to configure a application server (in this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk), asterisk alone, and read some about Restcomm. For Elastix and asterisk, I get stuck trying to setup the trigger point. For Restcomm, I only find documentation for integration with Clearwater IMS. So, I need your help. Somebody has something like that working? What TAS (or components for a TAS) do you recommend? Tutorials, examples, manuals that you know can help me? Thanks for reading, and many many thanks for any advice. David
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