<div dir="auto">Set the channel caller id variables before joining the conference<br><br><div data-smartmail="gmail_signature">sent from mobile<br>cell: +39 347 266 56 18<br>Giovanni Maruzzelli<br>OpenTelecom.IT</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Jul 7, 2017 18:35, "Mark Melling" <<a href="mailto:mark.melling@savageminds.com">mark.melling@savageminds.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Based on this I implemented a solution using lua:</div><div><br></div><div>The key bits being:</div><div><br></div><div><div>    session = freeswitch.Session(<call url>)</div></div><div>    ...</div><div>    session:execute("sleep", 1000)</div><div dir="ltr"><div><div>    session:execute("conference", <conference-name>)</div></div><div><br></div><div>The only problem is that when the verto user is added to the conference they appear (in the conference) with the originating caller id and name (as set in the freeswitch.Session call), rather than their own.</div><div><br></div><div>Is there anyway around this?</div><div><br></div><div>Thanks</div><div><br></div><div>Mark </div><div><br></div><br><div class="gmail_quote"><div dir="ltr">On Wed, 5 Jul 2017 at 22:21 Mark Melling <<a href="mailto:mark.melling@savageminds.com" target="_blank">mark.melling@savageminds.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div>Your suggestion does work, I did the following manually from fs_cli.</div><div><br></div><div>So with a user dialled into a conference room I was able to do:</div><div>





<p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">> originate  <call-url> &park</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">Where the call-url referring to a verto client, then</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">





</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">> uuid_transfer <verto-call-uuid> conference:<conf-name> inline</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">And that worked, I could hear audio from both clients connected to the conference.</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1">Thanks for your help.</span></p><p class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-p1"><span class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571inbox-inbox-s1"><br></span></p></div></div><br><div class="gmail_quote"><div dir="ltr">On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com" target="_blank">gmaruzz@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote">On 5 July 2017 at 18:08, Mark Melling <span dir="ltr"><<a href="mailto:mark.melling@savageminds.com" target="_blank">mark.melling@savageminds.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Thanks Giovanni for the suggestion.<div><br></div><div>I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).</div><div><br></div><div>But if they dial into the conference then everything appears fine and they do hear audio. </div><div><br></div><div>Specifically from fs_cli I entered:</div><div><br></div>





<div>originate <call-url> &conference(<conf-name>@<wbr>default)</div><div><br></div><div>If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking). </div><div><br></div><div>Whereas if you dialled into a conference from a verto client then you would hear the conference music. </div><div><br></div><div>So I'm not sure how I can work around this.</div><div> <div><br></div></div></div></blockquote><div><br></div></div></div></div><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><div>Have you tried what I suggested?<br><br><br></div></div></div></div><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571m_-6781179818359036472HOEnZb"><div class="m_-7982853431385651740m_6458714403571550325m_5237246327901979884m_-6668851994448783571m_-6781179818359036472h5"><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><div><br><br><div><div><div>Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc<br><br></div>Try
 this: instead of generating autocall from inside conference (eg instead
 of using autocall),originate call to user, wait for her to answer, then
 (after she answer) sleep for 2 seconds, then transfer her to the conf<br><br></div><div>-giovanni<br></div></div></div></div></div></div></blockquote></div></div></div></blockquote><div><br><br></div></div></div></div>
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