<div dir="ltr"><div>first of all reinstall freeswitch from scratch, usiing the instruction in <a href="https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie">https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie</a> , without the minimum change</div><div><br></div><div>then check if calls are working between two softphones, on two different machines</div><div><br></div><div>after all is working then do the fancy things.</div><div><br></div><div>you may want to give attention at external_rtp_ip and external_sip_ip values in "external" sip profile</div><div><br></div><div>anyway, first you must have calls in local LAN working</div><div><br></div><div>then calls from other, more complex, scenarios.</div><div><br></div><div>you may find the freeswitch books, easily available through the internet, useful as well as confluence documentation.</div><div><br></div><div>-giovanni</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 18 June 2018 at 07:14, Hossein Gholizadeh <span dir="ltr"><<a href="mailto:hggh96.hg@gmail.com" target="_blank">hggh96.hg@gmail.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">hi all ...
<span style="color:rgb(51,51,51);font-family:"Helvetica Neue",Helvetica,Arial,sans-serif;font-size:14px;background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline">i'm using sip.js for my IVR project ... it could connect to freeswitch correctly but there is no audio between call legs ... first of all i have to say i really cannot use Verto . this is my function of when call button pressed </span><div><span style="color:rgb(51,51,51);font-family:"Helvetica Neue",Helvetica,Arial,sans-serif;font-size:14px;background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline"><br></span></div><div><span style="background-color:rgb(224,234,243);text-decoration-style:initial;text-decoration-color:initial;float:none;display:inline;font-size:14px"><font face="Helvetica Neue, Helvetica, Arial, sans-serif"><div><font color="#333333"> </font><b><font color="#333333"> phoneCallButtonPressed : </font><font color="#0000ff">function</font><font color="#333333">(</font><font color="#666666">sessionid</font><font color="#333333">) {</font></b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> var s = ctxSip.Sessions[sessionid],</b></div><div><b><font color="#333333"> target = $(</font><font color="#e69138">"#numDisplay"</font><font color="#333333">).val();</font></b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> if (!s) {</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div><b><font color="#333333"> $(</font><font color="#ff9900">"#numDisplay"</font><font color="#333333">).val(</font><font color="#ff9900">""</font><font color="#333333">);</font></b></div><div style="color:rgb(51,51,51)"><b> ctxSip.sipCall(target);</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> } else if (s.accept && !s.startTime) {</b></div><div style="color:rgb(51,51,51)"><b><br></b></div><div style="color:rgb(51,51,51)"><b> s.accept({</b></div><div style="color:rgb(51,51,51)"><b> media : {</b></div><div style="color:rgb(51,51,51)"><b> stream : ctxSip.Stream,</b></div><div><b><font color="#333333"> constraints : { audio : </font><font color="#0000ff">true</font><font color="#333333">, video : </font><font color="#0000ff">false </font><font color="#333333">},</font></b></div><div style="color:rgb(51,51,51)"><b> render : {</b></div><div><b><font color="#333333"> remote : { audio: $(</font><font color="#ff9900">'#audioRemote'</font><font color="#333333">).get()[0] }</font></b></div><div style="color:rgb(51,51,51)"><b> },</b></div><div><b><font color="#333333"> RTCConstraints : { </font><font color="#ff9900">"optional"</font><font color="#333333">: [{ </font><font color="#ff9900">'DtlsSrtpKeyAgreement'</font><font color="#333333">: </font><font color="#ff9900">'true'</font><font color="#333333">} ]}</font></b></div><div style="color:rgb(51,51,51)"><b> }</b></div><div style="color:rgb(51,51,51)"><b> });</b></div><div style="color:rgb(51,51,51)"><b> }</b></div><div style="color:rgb(51,51,51)"><b> },</b></div><div style="color:rgb(51,51,51)"><br></div><div style="color:rgb(51,51,51)">------------------------------<wbr>------------------------------<wbr>-----------------------------</div><div style="color:rgb(51,51,51)">also i have to say that i tested it between my phone and x-lite ... X-lite could receive the voice but my phone could'nt ... i was using Zoiper in my phone.</div><div style="color:rgb(51,51,51)"><br></div></font></span></div></div>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a><br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><span><span><br>Sincerely,<br></span><span><font color="#888888"><br>
Giovanni Maruzzelli<br>
OpenTelecom.IT</font></span><br><span>cell: +39 347 266 56 18<span><font color="#888888"><br></font></span></span></span></div></div></div>
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