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    <div class="moz-cite-prefix">Hi Vlad,<br>
      yes, it show connected.<br>
      I'm trying to call an external SIP URI.<br>
      <br>
      It works only inside @opensips.org?<br>
      <br>
      Thanks<br>
      Stefano<br>
      <br>
      Il 05/07/2013 14.14, Vlad Paiu ha scritto:<br>
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    <blockquote cite="mid:51D6B8C1.1090605@opensips.org" type="cite">
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      <div class="moz-cite-prefix">Hello,<br>
        <br>
        So when you go to 'web calls' and hit the 'Login' button, the
        app successfully registers your SIP account against opensips.org
        and shows 'Connected' ?<br>
        If yes, the username that you are trying to call, is it also
        logged in on the website, in the 'web calls' section ? If it's
        using a regular soft/hard phone, does the phone have webRTC
        capabilities ?<br>
        <br>
        Regards,<br>
        <pre class="moz-signature" cols="72">Vlad Paiu
OpenSIPS Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
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        On 07/04/2013 07:35 PM, Stefano Pisani wrote:<br>
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      <blockquote cite="mid:51D5A43E.1050909@omnianet.it" type="cite">
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        I tried to call a SIP URI but it do not seems to be working.<br>
        I used crome. The connection works but it cannot place the call.<br>
        <br>
        s<br>
        <br>
        <div class="moz-cite-prefix">Il 04/07/2013 15.55, Vlad Paiu ha
          scritto:<br>
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          Hello,<br>
          <br>
          The free VoIP service offered by opensips.org has now been
          enhanced in order to support WebRTC calls.<br>
          In order to test it, you can login to your account at [1] and
          go to 'web calls' in the left menu. The integrated client
          supports both audio and video calls between two parties. <br>
          <br>
          Also, we have added a new tutorial, available at [2], which
          shows how to add WebRTC capabilities to any existing
          OpenSIPS-based deployment.<br>
          The tutorial makes use of an OpenSIPS deployment with NAT
          support, and adds WebRTC capabilities on top of that by using
          OverSIP as a WS to SIP gateway and sipML5 as the web client.<br>
          <br>
          [1]
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            href="https://www.opensips.org/account/">https://www.opensips.org/account/</a><br>
          [2]
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          <a moz-do-not-send="true"
            href="http://www.opensips.org/Documentation/Tutorials-WebSocket">http://www.opensips.org/Documentation/Tutorials-WebSocket</a><br>
          <br>
          Best Regards,<br>
          <pre class="moz-signature" cols="72">-- 
Vlad Paiu
OpenSIPS Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
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          <pre wrap="">_______________________________________________
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        <pre wrap="">_______________________________________________
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      <pre wrap="">_______________________________________________
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