I implemented the AS way and was able to play sound to the caller
but In order to continue the call and send the invite to SCSCF I
need to use proxy in the Dial application which is a problem
(Asterisk is B2BUA not a proxy).
I found this old question here https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464 that describes exactly the same issue.
Here is my dial plan:
exten => 972551000002,1,Progress()
exten => 972551000002,n,Playback(vm-starmain, noanswer)
exten => 972551000002,n,Wait(3)
exten => 972551000002,n,Hangup()
; This will send the call to the pcscf again
; exten => 972551000002,1,Dial(SIPemail@example.com,20);
; This will send the call to scscf but it will be rejected as domain not supported
; exten => 972551000002,1,Dial(SIPfirstname.lastname@example.org,20);
Can I use kamailio as an AS and implement the same ?
Hello Tsvetan.Actually you could use SIP Early media in AS and also with cscf.If you choice the first way, i think it is very simple and strightforward because you just use early media functions on your AS. For example in Astrisk you could use Progress application and 'm' option in Dial application in your dialplan.In second way you should check in Reply-Route block,if you got 180 ringing, you have to use rtpproxy-stream funtion for doing sip early.
Wih Regards.Mojtaba Esfandiari.S
On Fri, 21 Dec 2018, 16:34 Tsvetan Filev, <email@example.com> wrote:
I want to use SIP early media to play music to the caller in kamailio
IMS installation like this:
I looked a little bit but didn't find ready solution. The information is
vague on this topic.
Should this be done through a module or application server ?
May I need to handle ringing in onreply_route and send OK with SDP to
the caller in SCSCF ?
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