1 Jun
2015
1 Jun
'15
11:46 a.m.
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(a)ing.una.py>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
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