<div>Something is not right with the outbound leg of the call. It looks like there is an immediate hangup after the b leg answers. Get a siptrace of that traffic and look to see what is causing the hangup.</div><div><br></div>
<div>-MC<br><br><div class="gmail_quote">On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <span dir="ltr"><<a href="mailto:yungwei@resolvity.com">yungwei@resolvity.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">The whole picture looks like the following:</span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Is there any configuration setting that needs to be enabled or set differently so that those 2 endpoints can talk to each other?</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.</span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p>
<div><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Yungwei Chen<br>
<b>Sent:</b> Friday, May 27, 2011 11:34 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Transferring to a cell phone</span></p></div></div><p class="MsoNormal"> </p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Here's the debug log, <a href="http://pastebin.freeswitch.org/16398" target="_blank">http://pastebin.freeswitch.org/16398</a></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Michael Collins<br>
<b>Sent:</b> Friday, May 27, 2011 10:08 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Transferring to a cell phone</span></p></div><p class="MsoNormal"> </p><p class="MsoNormal">You need to put a debug log on <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>. Be sure to use "FreeSWITCH Log" for the syntax highlighting.</p>
<div><p class="MsoNormal" style="margin-bottom:12.0pt">-MC</p><div><p class="MsoNormal">On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <<a href="mailto:yungwei@resolvity.com" target="_blank">yungwei@resolvity.com</a>> wrote:</p>
<p class="MsoNormal">No.<br><br>I just verified that making an outbound call to my cell phone still works.<br>I even recorded the session just to make sure, and in the recording I hear things both ways.<br>I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.<br>
session.answer();</p><div><p class="MsoNormal"> var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");</p></div><p class="MsoNormal"> session.hangup(16); // disconnects the session between the SIP client and freeswitch<br>
new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");<br> while (new_session.ready())<br> {<br> new_session.streamFile("/path/to/local/wav/file");<br> }<br><br>Now back to the case I'm having problem with.<br>
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.<br>I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.<br>
Here's the javascript program:<br> session.answer();<br><br> if (session.ready())<br> {<br> session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");<br> }<br>
<br>So what am I missing here?</p><div><div><p class="MsoNormal"><br>-----Original Message-----<br>From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of mazilo<br>
Sent: Friday, May 27, 2011 7:28 AM<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Subject: Re: [Freeswitch-users] Transferring to a cell phone<br>
<br><br>Yungwei Chen wrote:<br>><br>> Thanks for your reply. Using bridge fixed the problem. But I cannot hear<br>> anything both ways. Any idea?<br>After switching to using bridge function, does this also happen when you<br>
make an outbound call to your cell phone using your javascript?<br><br>-----<br>FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.<br>--<br>View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html</a><br>
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