Thanks a lot Ivalio, i'll do some tests and report back.
Sent from my Huawei Mobile
Hello,
Comment inline.
Hope I'm helping,
Ivo
On 11/21/2017 12:32 PM, Amar Tinawi wrote:
In order to route the call to Asterisk you should define it as AS/MMtel in your HSS. Define proper TP and iFC for the subscriber profiles.Thank you Daniel :
Actually i followed your tutorial, but in case of IMS the modifications should be on (my assumption) S-CSCF, in this case all routing block should be routed towards AS ?
For sure you should configure Asterisk to handle originating and terminating call legs from/to S-CSCF. From Asterisk perspective the easiest way is S-CSCF to be a SIP trunk. (Note that this may not be 100% true and depends on the design)i followed also Mr Carsten Bock videos on kamailio World conferences for the years 2015 - 2016 - 2017 and he mentioned clearly that any application that talk sip can operate as an AS, and he mentioned Asterisk and FreeSwitch as an exampleThanks Mojtaba, i followed this video but the video at the end shows that the users still in offline mode after registration, is that right ?My questions are :- should i configure Asterisk in a way to be AS for kamailio IMS ?
As far I know there is no DIAMETER/Sh support in Asterisk. Related and to Q below too.- should the users be identified in Asterisk Database in order to make a correlation ? or asterisk can pull users profiles from HSS ?
You have 2 options in general here, to have or not subscribers in Asterisk. Easiest way is to not. Assuming that all your subscribers are same from service point of view you can handle everything in routing w/o having any subscriber profile in Asterisk. Second option is a matter of design and there is no simple answer.- Do we have to perform any configuration over HSS to identify sh interface with Asterisk ?
Keep in mind that SIP and IMS services differs. When using * or FreeSWITCH as SIP MMtel w/ SIP clients in IMS env, you can relatively easy implement known SIP services. Implementing MMTel IMS services w/ IMS clients is a different story.
if you can send me the Astricon video it would be great
Thanks,Ammar
On Tue, Nov 21, 2017 at 10:29 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,
personally I do not have any time these days to search or look at
documents/videos/etc... to see what TAS are supposed to be -- a lot of
stuff to do for upcoming kamailio 5.1 in the spare time.
On the other hand, at the last edition of Astricon, Carsten Bock (one of
the ims extensions developer in kamailio) had a presentation about using
Asterisk for IMS -- I had to be at the Kamailio booth in Astricon expo
area, so I couldn't attend it -- maybe the videos from Astricon are
already published and you can watch it, if not, probably they will be
released soon.
Cheers,
Daniel
Daniel-Constantin Mierla
On 21.11.17 09:06, Mojtaba wrote:
> Hello,
> Maybe this link be useful to understanding the concept of Asterisk
> working as Telephone Application Servers(TAS) in IMS core network.
> https://www.youtube.com/watch?v=3nVRpzkUbIg
> With Regards. Mojtaba
>
> On Mon, Nov 20, 2017 at 5:14 PM, Amar Tinawi <amar.tinawi@gmail.com> wrote:
>> Dears;
>> Good day !
>>
>> is it possible to use Asterisk pbx server as an Application server for the
>> kamailio IMS users ? as if i want to correlate an IMS user with an extension
>> number.
>>
>> My though is kamailio ims will handle the registration process and
>> authentication and Asterisk will deal with IMS users as they are PBX users
>> to get all asterisk's benefits (call conference, monitoring, waiting, voice
>> mail ... etc)
>>
>> i'm thinking to integrate Asterisk with S-cscf, and when an IMS user
>> register to the IMS to domain, a trigger will initiate the S-cscf to send
>> notify to asterisk .
>>
>>
>>
>> i didnt find any solution or tutorial for this procedure, and idont know if
>> it is possible to be done.
>>
>> Regards
>> Ammar
>>
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