It would be like these lines with afew changes:          1D IN A           1D IN NAPTR 10 50 "s" "SIP+D2U"    ""    _sip._udp          1D IN NAPTR 20 50 "s" "SIP+D2T"    ""    _sip._tcp

On Mon, Jan 28, 2019 at 1:37 PM Tsvetan Filev <> wrote:

Here is my current zone file:

@                       1D IN SOA       localhost. root.localhost. (
                                        1               ; serial
                                        3H              ; refresh
                                        15M             ; retry
                                        1W              ; expiry
                                        1D )            ; minimum

                        1D IN NS        ns
ns                      1D IN A

pcscf                   1D IN A
_sip._udp.pcscf         1D SRV 0 0 5060 pcscf
_sip._tcp.pcscf         1D SRV 0 0 5060 pcscf

icscf                   1D IN A
_sip._udp               1D SRV 0 0 4060 icscf
_sip._tcp               1D SRV 0 0 4060 icscf
_sip._udp.ims           1D SRV 0 0 4060 icscf
_sip._tcp.ims           1D SRV 0 0 4060 icscf

scscf                   1D IN A
_sip._udp.scscf         1D SRV 0 0 6060 scscf
_sip._tcp.scscf         1D SRV 0 0 6060 scscf

as                      1D IN A            1D SRV 0 0 5062 as            1D SRV 0 0 5062 as

hss                     1D IN A

How do I modify it in order to make this work ?


On 28.01.19 г. 11:50 ч., Mojtaba wrote:
Hi Tsvetan,
Why do you send call back to S-CSCF? You should send call back to I-CSCF.  Actually in resolve of domain "", The ICSCF's IP should be returned.
 Make sure entry SRV recordd in DNS server are true.
This kind of call back to IMS is true, But make sure you won't have any issue in DNS resolve.
  exten => 972551000002,1,Dial(SIP/,20);

With Regards.Mojtaba
On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <> wrote:

Hi Mojtaba.

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 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(SIP/,20);
  ; This will send the call to scscf but it will be rejected as domain not supported
  ;  exten => 972551000002,1,Dial(SIP/,20);

Can I use kamailio as an AS and implement the same ?


On 22.12.18 г. 0:06 ч., Mojtaba wrote:
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, <> wrote:
Hi all.

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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--Mojtaba Esfandiari.S

--Mojtaba Esfandiari.S