[Freeswitch-users] Transferring to a cell phone

Yungwei Chen yungwei at resolvity.com
Thu Jun 2 04:17:53 MSD 2011


Using bridge in the dialplan directly works.

I tried putting only the following line in transfer.js, but it doesn't work.
       session.execute("bridge", "sofia/gateway/broadvoice/2223334444");

Putting the following instead doesn't work either.
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");

Using the following instead doesn't work either.
       session.answer();
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");

Still don't understand why.

From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org]
Sent: Wednesday, June 01, 2011 4:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone


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.  


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:
<extension name="route DID">
  <condition field="destination_number" expression="^(INCOMING_DID_NUMBER)$">
    <action application="bridge" data="sofia/gateway/broadvoice/2223334444"/>
  </condition>
</extension>



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. 


-MC


On Wed, Jun 1, 2011 at 9:52 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

Here's the log with siptrace.
http://pastebin.freeswitch.org/16424

Here's the content of transfer.js:
   session.answer();
   if (session.ready())
   {
       session.execute("bridge", "{ignore_early_media=true}sofia/gateway/broadvoice/2223334444");
   }




From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org]
Sent: Friday, May 27, 2011 6:21 PM 

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone



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.


-MC


On Fri, May 27, 2011 at 2:27 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

The whole picture looks like the following:
A land-line<-->SIP provider A<-->FS<-->SIP provider A<-->a cell phone

Is there any configuration setting that needs to be enabled or set differently so that those 2 endpoints can talk to each other?
Btw, the same scenario works with Asterisk so my SIP provider shouldn't be the problem.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yungwei Chen
Sent: Friday, May 27, 2011 11:34 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

Here's the debug log, http://pastebin.freeswitch.org/16398

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, May 27, 2011 10:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Transferring to a cell phone

You need to put a debug log on pastebin.freeswitch.org. Be sure to use "FreeSWITCH Log" for the syntax highlighting.
-MC
On Fri, May 27, 2011 at 7:41 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
No.

I just verified that making an outbound call to my cell phone still works.
I even recorded the session just to make sure, and in the recording I hear things both ways.
I first dial 9911 from my SIP client (behind freeswitch), and this leads to the javascript program below.
  session.answer();
  var new_session = new Session("{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  session.hangup(16); // disconnects the session between the SIP client and freeswitch
  new_session.execute("record_session", "/tmp/call-from-js-to-cell.wav");
  while (new_session.ready())
  {
      new_session.streamFile("/path/to/local/wav/file");
  }

Now back to the case I'm having problem with.
In this case, I first make a call from a landline to freeswitch through my sip provider, and then a javascript program takes over.
I want to transfer the call to a cell phone so that the landline and the cell phone can communicate with each other.
Here's the javascript program:
  session.answer();

  if (session.ready())
  {
      session.execute("bridge", "{ignore_early_media=true}sofia/gateway/sip_provider/1231231234");
  }

So what am I missing here?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of mazilo
Sent: Friday, May 27, 2011 7:28 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Transferring to a cell phone


Yungwei Chen wrote:
>
> Thanks for your reply. Using bridge fixed the problem. But I cannot hear
> anything both ways. Any idea?
After switching to using bridge function, does this also happen when you
make an outbound call to your cell phone using your javascript?

-----
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View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
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