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    <p>Here is my current zone file:</p>
    <p>$ORIGIN mnc001.mcc001.3gppnetwork.org.<br>
      $TTL 1W<br>
      @                       1D IN SOA       localhost. root.localhost.
      (<br>
                                              1               ; serial<br>
                                              3H              ; refresh<br>
                                              15M             ; retry<br>
                                              1W              ; expiry<br>
                                              1D )            ; minimum<br>
      <br>
                              1D IN NS        ns<br>
      ns                      1D IN A         10.82.10.56<br>
      <br>
      pcscf                   1D IN A         10.82.10.56<br>
      _sip._udp.pcscf         1D SRV 0 0 5060 pcscf<br>
      _sip._tcp.pcscf         1D SRV 0 0 5060 pcscf<br>
      <br>
      icscf                   1D IN A         10.82.10.56<br>
      _sip._udp               1D SRV 0 0 4060 icscf<br>
      _sip._tcp               1D SRV 0 0 4060 icscf<br>
      _sip._udp.ims           1D SRV 0 0 4060 icscf<br>
      _sip._tcp.ims           1D SRV 0 0 4060 icscf<br>
      <br>
      scscf                   1D IN A         10.82.10.56<br>
      _sip._udp.scscf         1D SRV 0 0 6060 scscf<br>
      _sip._tcp.scscf         1D SRV 0 0 6060 scscf<br>
      <br>
      as                      1D IN A         10.82.10.56<br>
      _sip._udp.as            1D SRV 0 0 5062 as<br>
      _sip._tcp.as            1D SRV 0 0 5062 as<br>
      <br>
      hss                     1D IN A         10.82.10.56<br>
      <br>
      <br>
      How do I modify it in order to make this work ?<br>
      <br>
      Tnx.<br>
      <br>
    </p>
    <div class="moz-cite-prefix">On 28.01.19 г. 11:50 ч., Mojtaba wrote:<br>
    </div>
    <blockquote type="cite"
cite="mid:CABVi_Ez=0+u3868nKqDkhhRgJhmFL9yhWfP9H0d=N5-ibkiALw@mail.gmail.com">
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      <div dir="ltr">
        <div dir="ltr">
          <div>Hi Tsvetan, <br>
          </div>
          <div>Why do you send call back to S-CSCF? You should send call
            back to I-CSCF.  Actually in resolve of domain "<a
              class="gmail-m_3097791201919026857moz-txt-link-abbreviated"
href="mailto:SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20"
              target="_blank" moz-do-not-send="true">mnc001.mcc001.3gppnetwork.org"</a>,
            The ICSCF's IP should be returned.</div>
          <div> Make sure entry SRV recordd in DNS server are true.</div>
          <div>This kind of call back to IMS is true, But make sure you
            won't have any issue in DNS resolve.</div>
          <div>  exten => 972551000002,1,Dial(<a
              class="gmail-m_3097791201919026857moz-txt-link-abbreviated"
href="mailto:SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20"
              target="_blank" moz-do-not-send="true">SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20</a>);</div>
        </div>
        <div><br>
        </div>
        <div>With Regards.Mojtaba<br>
        </div>
        <div class="gmail_quote">
          <div dir="ltr" class="gmail_attr">On Mon, Jan 28, 2019 at
            12:34 PM 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:0px 0px 0px
            0.8ex;border-left:1px solid
            rgb(204,204,204);padding-left:1ex">
            <div bgcolor="#FFFFFF">
              <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="gmail-m_3097791201919026857moz-txt-link-freetext"
href="https://community.asterisk.org/t/how-can-i-configure-asterisk-to-act-like-a-sip-proxy-server/18464"
                  target="_blank" moz-do-not-send="true">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="gmail-m_3097791201919026857moz-txt-link-abbreviated"
href="mailto:SIP/972551000002@mnc001.mcc001.3gppnetwork.org,20"
                  target="_blank" moz-do-not-send="true">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="gmail-m_3097791201919026857moz-txt-link-abbreviated"
href="mailto:SIP/972551000002@scscf.mnc001.mcc001.3gppnetwork.org,20"
                  target="_blank" moz-do-not-send="true">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="gmail-m_3097791201919026857moz-cite-prefix">On
                22.12.18 г. 0:06 ч., Mojtaba wrote:<br>
              </div>
              <blockquote type="cite">
                <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"
                          target="_blank" moz-do-not-send="true">tsvetan.filev@inno-networks.com</a>>
                        wrote:<br>
                      </div>
                      <blockquote class="gmail_quote" style="margin:0px
                        0px 0px 0.8ex;border-left:1px solid
                        rgb(204,204,204);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>
                        _______________________________________________<br>
                        Kamailio (SER) - Users Mailing List<br>
                        <a href="mailto:sr-users@lists.kamailio.org"
                          rel="noreferrer" target="_blank"
                          moz-do-not-send="true">sr-users@lists.kamailio.org</a><br>
                        <a
                          href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users"
                          rel="noreferrer noreferrer" target="_blank"
                          moz-do-not-send="true">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><br>
                      </blockquote>
                    </div>
                  </div>
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                <br>
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                  class="gmail-m_3097791201919026857mimeAttachmentHeader"></fieldset>
                <pre class="gmail-m_3097791201919026857moz-quote-pre">_______________________________________________
Kamailio (SER) - Users Mailing List
<a class="gmail-m_3097791201919026857moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org" target="_blank" moz-do-not-send="true">sr-users@lists.kamailio.org</a>
<a class="gmail-m_3097791201919026857moz-txt-link-freetext" href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank" moz-do-not-send="true">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a>
</pre>
              </blockquote>
            </div>
          </blockquote>
        </div>
        <br clear="all">
        <br>
        -- <br>
        <div dir="ltr" class="gmail_signature">--Mojtaba Esfandiari.S</div>
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