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    <tt>Hi Duane,<br>
      <br>
      AFAIK, b2b can do transfer only for established calls - this is
      why you get that err message.<br>
      <br>
      Regards,<br>
    </tt>
    <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
    <br>
    On 06/08/2012 08:10 PM, Duane Larson wrote:
    <blockquote
cite="mid:CAFcM1Er7-aLtUOHeryOMUqiGFQwuPb+kqHELeXwf_A1ft3YGLw@mail.gmail.com"
      type="cite">Here is a SIP Trace of my Call Pickup<br>
      <br>
      <a moz-do-not-send="true" href="http://pastebin.com/beARSgmj">http://pastebin.com/beARSgmj</a><br>
      <br>
      If you do a search for <b>Replaces</b> you will see the INVITE
      that is sent in order to do a Call Pickup of the call coming in
      from the SIP Provider.<br>
      <br>
      <br>
      <br>
      <div class="gmail_quote">On Wed, Jun 6, 2012 at 9:26 PM,
        osiris123d <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</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;">
          In the past I used a VoIP SIP trunk provider that I guess used
          Asterisk<br>
          servers as their gateways and I was able to perform the Call
          Pickup feature<br>
          with OpenSIPS and the SIP Trunk Provider. Now I have a new
          provider and I<br>
          need to use the B2B modules for Transfers. &nbsp;Currently I am not
          able to<br>
          perform Call Pickup with calls coming from the PSTN to
          internal customers.<br>
          I tried to use the b2b_bridge_request but when the PSTN Caller
          calls<br>
          internal_user_A and internal_user_B tries to do a call pickup
          I see the<br>
          following error in syslog<br>
          <br>
          &nbsp;ERROR:b2b_logic:b2bl_bridge_msg: Wrong state for entity ek=
          [B2B.106.390],<br>
          tk=[536.0]<br>
          <br>
          When I do an "opensipsctl fifo b2b_list" all the states are
          zero.<br>
          <br>
          So does the state of the call have to be a "on call" state
          instead of a<br>
          "calling" state? &nbsp;How can you bridge a call that is in process
          of ringing?<br>
          <br>
          --<br>
          View this message in context: <a moz-do-not-send="true"
href="http://opensips-open-sip-server.1449251.n2.nabble.com/B2B-with-Call-Pickup-tp7580224.html"
            target="_blank">http://opensips-open-sip-server.1449251.n2.nabble.com/B2B-with-Call-Pickup-tp7580224.html</a><br>
          Sent from the OpenSIPS - Users mailing list archive at
          Nabble.com.<br>
          <br>
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            href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
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            href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users"
            target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
        </blockquote>
      </div>
      <br>
      <br clear="all">
      <br>
      -- <br>
      --<br>
      *--*--*--*--*--*<br>
      Duane<br>
      *--*--*--*--*--*<br>
      --<br>
      <pre wrap="">
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</pre>
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