[Freeswitch-users] Transferring to a cell phone
Michael Collins
msc at freeswitch.org
Sat May 28 03:21:00 MSD 2011
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?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Transferring-to-a-cell-phone-tp6408246p6410887.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
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