I'm not really sure what's happening here. The first BYE message that I see comes in at line #548 from the provider side of what I assume is the A leg. <div><br></div><div>For now I would try using a very simple bridge app right in the public context and avoid all the javascript so that you can get the simplest scenario. SImply route the A leg to the bridge app and bridge it to the broadvoice gateway:</div>
<div><font class="Apple-style-span" face="'courier new', monospace"><extension name="route DID"></font></div><div><font class="Apple-style-span" face="'courier new', monospace"> <condition field="destination_number" expression="^(INCOMING_DID_NUMBER)$"></font></div>
<div><font class="Apple-style-span" face="'courier new', monospace"> <action application="bridge" data="sofia/gateway/broadvoice/2223334444"/></font></div><div><font class="Apple-style-span" face="'courier new', monospace"> </condition></font></div>
<div><font class="Apple-style-span" face="'courier new', monospace"></extension></font><br><div><br></div><div>A call coming in from one SIP endpoint and being bridged to another SIP endpoint is pretty much the core and fundamental functionality of FreeSWITCH, so I'm thinking that the javascript is doing something unintended. </div>
<div><br></div><div>-MC<br><br><div class="gmail_quote">On Wed, Jun 1, 2011 at 9:52 AM, 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>
<div style="font-family:Tahoma;direction:ltr;color:#000000;font-size:x-small">
<div>Here's the log with siptrace.</div>
<div><a href="http://pastebin.freeswitch.org/16424" target="_blank">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>
<div dir="ltr"><font color="#000000" size="2" face="Tahoma"></font> </div>
<div style="direction:ltr">
<hr>
<font color="#000000" size="2" face="Tahoma"><b>From:</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>] On Behalf Of Michael Collins [<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>]<br>
<b>Sent:</b> Friday, May 27, 2011 6:21 PM<div><div></div><div class="h5"><br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Transferring to a cell phone<br>
</div></div></font><br>
</div><div><div></div><div class="h5">
<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" target="_blank">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" 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="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" 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 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" 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>
Sent from the freeswitch-users mailing list archive at Nabble.com.<br>
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