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<p>Hi Mojtaba.</p>
<p>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).<br>
I found this old question here
<a class="moz-txt-link-freetext" href="https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464">https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464</a>
that describes exactly the same issue.<br>
Here is my dial plan:</p>
<p>exten => 972551000002,1,Progress()<br>
exten => 972551000002,n,Playback(vm-starmain, noanswer)<br>
exten => 972551000002,n,Wait(3)<br>
exten => 972551000002,n,Hangup()<br>
; This will send the call to the pcscf again<br>
; exten =>
972551000002,1,Dial(<a class="moz-txt-link-abbreviated" href="mailto:SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20">SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20</a>);<br>
; This will send the call to scscf but it will be rejected as
domain not supported<br>
; exten =>
972551000002,1,Dial(<a class="moz-txt-link-abbreviated" href="mailto:SIP/972551000002@scscf.mnc001.mcc001.3gppnetwork.org,20">SIP/972551000002@scscf.mnc001.mcc001.3gppnetwork.org,20</a>);<br>
</p>
<p>Can I use kamailio as an AS and implement the same ?<br>
</p>
<p>Regards.<br>
</p>
<div class="moz-cite-prefix">On 22.12.18 г. 0:06 ч., Mojtaba wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CABVi_EwBe2=dBesR4WDxyouB8BZ6ug=uupxzV7dULag+g9DjuQ@mail.gmail.com">
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<div dir="auto">
<div>Hello Tsvetan.</div>
<div dir="auto">Actually you could use SIP Early media in AS and
also with cscf.</div>
<div dir="auto">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.</div>
<div dir="auto">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.</div>
<div dir="auto"><br>
</div>
<div dir="auto">Wih Regards.Mojtaba Esfandiari.S</div>
<div dir="auto"><br>
<div class="gmail_quote" dir="auto">
<div dir="ltr">On Fri, 21 Dec 2018, 16:34 Tsvetan Filev,
<<a href="mailto:tsvetan.filev@inno-networks.com"
moz-do-not-send="true">tsvetan.filev@inno-networks.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all.<br>
<br>
I want to use SIP early media to play music to the caller
in kamailio <br>
IMS installation like this: <br>
<a
href="http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html"
rel="noreferrer noreferrer" target="_blank"
moz-do-not-send="true">http://www.sharetechnote.com/html/IMS_SIP_EarlyMedia.html</a><br>
<br>
I looked a little bit but didn't find ready solution. The
information is <br>
vague on this topic.<br>
Should this be done through a module or application server
?<br>
May I need to handle ringing in onreply_route and send OK
with SDP to <br>
the caller in SCSCF ?<br>
<br>
Regards.<br>
<br>
<br>
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target="_blank" rel="noreferrer" moz-do-not-send="true">sr-users@lists.kamailio.org</a><br>
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href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users"
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</blockquote>
</div>
</div>
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<br>
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<pre class="moz-quote-pre" wrap="">_______________________________________________
Kamailio (SER) - Users Mailing List
<a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a>
<a class="moz-txt-link-freetext" href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
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