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    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 :)<br>
    <br>
    <blockquote
      cite="mid:BANLkTikh_DOBBJQpaMLO0KBbnF-dRHEDDQ@mail.gmail.com"
      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">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>
          &nbsp; &nbsp; 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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    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
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