<div dir="ltr">Hi Mr Alex ,<div>But in dialplan/default.xml i configure a dtmf ,when a user 1000 call the extension 1001 we will get rtp event in wireshark .</div><div>but with this one originate user/1000 &bridge(user/1001) i don't get the dtmf ? can you help me please ? </div><div><br></div><div>Best regards </div><div>Amani </div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">Le mer. 11 mai 2016 à 14:39, Alex Pierry <<a href="mailto:alex@teclan.com.br">alex@teclan.com.br</a>> a écrit :<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p><span>originate user/1000<context> &bridge(user/1001@<context>)</span><br>
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<div dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>De:</b> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> <<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>> em nome de amani mansour <<a href="mailto:amani.mansour2@gmail.com" target="_blank">amani.mansour2@gmail.com</a>><br>
<b>Enviado:</b> quarta-feira, 11 de maio de 2016 10:17<br>
<b>Para:</b> FreeSWITCH Users Help<br>
<b>Assunto:</b> [Freeswitch-users] make a call between two sip phones using fs_cli ?</font>
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<div dir="ltr">Hi all ,
<div>Can i make a call between 2 sip phones from fs_cli interface ? if yes ,and it is using originate haw can i do it please ? </div>
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<div>2 extensions 1000 and 1001 .</div>
<div>Best regards </div>
<div>Amani</div>
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