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<div>Here's the log with siptrace.</div>
<div><a href="http://pastebin.freeswitch.org/16424">http://pastebin.freeswitch.org/16424</a></div>
<div><font face="tahoma"></font> </div>
<div><font face="tahoma">Here's the content of transfer.js:</font></div>
<div> session.answer();</div>
<div> if (session.ready())<br>
{<br>
session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");</div>
<div> }<br>
</div>
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<font color="#000000" size="2" face="Tahoma"><b>From:</b> freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins [msc@freeswitch.org]<br>
<b>Sent:</b> Friday, May 27, 2011 6:21 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Transferring to a cell phone<br>
</font><br>
</div>
<div></div>
<div>
<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 style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div lang="EN-US">
<div>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">The whole picture looks like the following:</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"></span> </p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt">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="COLOR: #1f497d; FONT-SIZE: 11pt">Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.</span></p>
<p class="MsoNormal"><span style="COLOR: #1f497d; FONT-SIZE: 11pt"></span> </p>
<div>
<div style="BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt">
<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">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="COLOR: #1f497d; FONT-SIZE: 11pt">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="COLOR: #1f497d; FONT-SIZE: 11pt"></span> </p>
<div style="BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt">
<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">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 style="MARGIN-BOTTOM: 12pt" class="MsoNormal">-MC</p>
<div>
<p class="MsoNormal">On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <<a href="mailto:yungwei@resolvity.com">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>
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<p class="MsoNormal"><br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">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">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>
Sent from the freeswitch-users mailing list archive at Nabble.com.<br>
<br>
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