[SR-Users] IMS early media

[Mojtaba]

Hi Tsvetan,
Why do you send call back to S-CSCF? You should send call back to I-CSCF.
Actually in resolve of domain "mnc001.mcc001.3gppnetwork.org"
<SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20>, 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/972551000002 at mnc001.mcc001.3gppnetwork.org,20);

With Regards.Mojtaba
On Mon, Jan 28, 2019 at 12:34 PM Tsvetan Filev <
tsvetan.filev at inno-networks.com> 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
> 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(
> SIP/972551000002 at mnc001.mcc001.3gppnetwork.org,20);
>   ; This will send the call to scscf but it will be rejected as domain not
> supported
>   ;  exten => 972551000002,1,Dial(
> SIP/972551000002 at scscf.mnc001.mcc001.3gppnetwork.org,20);
>
> Can I use kamailio as an AS and implement the same ?
>
> Regards.
> 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, <tsvetan.filev at inno-networks.com>
> wrote:
>
>> Hi all.
>>
>> I want to use SIP early media to play music to the caller in kamailio
>> IMS installation like this:
>> http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html
>>
>> 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 ?
>>
>> Regards.
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
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>

-- 
--Mojtaba Esfandiari.S
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