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    Hello!<br>
    <br>
        Thank You, Steve!<br>
    <blockquote
      cite="mid:BANLkTim4hoBnmSXjRDyqqU4z7YzQ7s1CnQ@mail.gmail.com"
      type="cite">Yes, either that or 200 (you can skip ringing to go
      straight to being answered).<br>
      <br>
      Generally with 183/180 you'll get ringback but not hear voice in
      early media, you wouldn't normally hear voice until the 200 OK
      answers the call.<br>
      <br>
      Media can't start up without SDP in the body of the 180/183/200.
      SDP contains the IP/port used for media and the selected codec.
      Media can't work without those details. Sometimes you'll see
      180/183 without SDP there'll still be no early media containing
      ringback, in that case the client'll usually generate its own
      ringback but you'll always see SDP in the 200 OK.<br>
      <br>
      -Steve<br>
      <br>
      <br>
      <br>
      <br>
      <div class="gmail_quote">On 30 June 2011 13:00, Boris Kovalenko <span
          dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:boris@tagnet.ru">boris@tagnet.ru</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <div text="#000000" bgcolor="#ffffff"> Hello!<br>
            <br>
                But even reading this RFC, I found that 100 Trying
            should be followed by one of 18x response to get media
            channels connected, isn't? <br>
            <div>
              <div class="h5"> <br>
                <blockquote type="cite">Btw, this is the latest SIP
                  spec, better than the draft spec for 183 you posted
                  before.<br>
                  <br>
                  <a moz-do-not-send="true"
                    href="http://www.ietf.org/rfc/rfc3261.txt"
                    target="_blank">http://www.ietf.org/rfc/rfc3261.txt</a><br>
                  <br>
                  -Steve<br>
                  <br>
                  <br>
                  <br>
                  <div class="gmail_quote"> On 30 June 2011 09:55, Boris
                    Kovalenko <span dir="ltr">&lt;<a
                        moz-do-not-send="true"
                        href="mailto:boris@tagnet.ru" target="_blank">boris@tagnet.ru</a>&gt;</span>
                    wrote:<br>
                    <blockquote class="gmail_quote" style="margin: 0pt
                      0pt 0pt 0.8ex; border-left: 1px solid rgb(204,
                      204, 204); padding-left: 1ex;">
                      <div text="#000000" bgcolor="#ffffff"> Hello!<br>
                        <br>
                        Yes, this is all. INVITE, 100 and RTP. No other
                        SIP messages. So my knowledge of SIP if right.
                        And customer lies to me or forgot to configure
                        something :)
                        <div>
                          <div><br>
                            <br>
                            <blockquote type="cite">Is that all the
                              signalling?<br>
                              <br>
                              100 Trying only acknowledges the INVITE.<br>
                              <br>
                              That *may* be followed by 183 or 180
                              which'll indicate ringing. The Cisco would
                              only be able to send early media if that
                              was a 183/SDP or 180/SDP.<br>
                              <br>
                              The voice will only start when it sends
                              200/SDP which is when the call is
                              answered.<br>
                              <br>
                              -Steve<br>
                              <br>
                              <br>
                              <div class="gmail_quote">On 30 June 2011
                                06:00, Boris Kovalenko <span dir="ltr">&lt;<a
                                    moz-do-not-send="true"
                                    href="mailto:boris@tagnet.ru"
                                    target="_blank">boris@tagnet.ru</a>&gt;</span>
                                wrote:<br>
                                <blockquote class="gmail_quote"
                                  style="margin: 0pt 0pt 0pt 0.8ex;
                                  border-left: 1px solid rgb(204, 204,
                                  204); padding-left: 1ex;">Hello!<br>
                                  <br>
                                      I have a customers with Cisco 2821
                                  which want to use IVR on it. He<br>
                                  has configured Cisco and told me there
                                  is no voice when number is<br>
                                  called. The configuration is:<br>
                                  <br>
                                  Cisco 2821 &lt;-- FreeSwitch (full
                                  proxy mode) --&gt; PSTN<br>
                                  <br>
                                  I looked at tcpdump and the call flow
                                  is:<br>
                                  FS (INVITE) --&gt; Cisco 2821<br>
                                  Cisco 2821 (100 Trying) --&gt; FS<br>
                                  Cisco 2821 (RTP) --&gt; FS<br>
                                  <br>
                                  I think the call flow is wrong in that
                                  case and SIP100 can not go<br>
                                  throught Freeswitch<br>
                                  (<a moz-do-not-send="true"
                                    href="http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt"
                                    target="_blank">http://jdrosen.net/papers/draft-donovan-mmusic-183-00.txt</a>)
                                  and there<br>
                                  should be SIP180 to get it work. But
                                  customers said this is OK and he<br>
                                  has no troubles with same
                                  configuration with other provider.
                                  What may I<br>
                                  do in this sutiation? Any specific
                                  Freeswitch configuration?<br>
                                  <br>
                                  P.S. FreeSWITCH Version 1.0.head
                                  (git-1c95ad9 2011-01-20 22-43-50
                                  -0300)<br>
                                  <br>
                                  --<br>
                                  Regards,<br>
                                  Boris<br>
                                  <br>
                                  <br>
                                  <br>
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                        <pre cols="72">-- 
С уважением,
  Борис Коваленко
  ЗАО "Тагнет"
  тел. +7 (3435) 230001
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            <pre cols="72">-- 
Regards,
Boris

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    <pre class="moz-signature" cols="72">-- 
Regards,
Boris

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