[Freeswitch-users] Session Question

Fri May 2 14:04:25 PDT 2008

Still testing, but it seems to be working now! -- thx.

On Fri, May 2, 2008 at 2:02 PM, Brian West <brian at freeswitch.org> wrote:

> Yes please update.
> /b
>
> On May 2, 2008, at 12:27 PM, Nick Temple wrote:
>
> rc3 tarball - I can upgrade to latest svn if that could help (looking
> forward to rc4).
>
> On Fri, May 2, 2008 at 12:57 PM, Brian West <brian at freeswitch.org> wrote:
>
> > Nick, Before we go any further what svn rev are you running?
> >
> > /b
> >
> > On May 2, 2008, at 11:23 AM, Nick Temple wrote:
> >
> > Thanks,
> >
> > Theat was my attempt to sanitize the data,  and I did a really poor job
> > of it :->
> >
> > The waves & numbers I'm using in the script are correct, though.
> >
> > I've narrowed it down somewhat, it's not the type of phone ... it's
> > something else.
> > (though I have a better chance of hearing a message on land line ...
> > maybe.)
> >
> > Sometimes the messages play, sometimes they start then die 1/2 way
> > through ...
> > usually they don't start.  If I loop and wait for a tone, then replay,
> > sometimes the
> > second attempt does actually play.
> >
> > I've tried using both streamFile (with and without the callback) and
> > session.execute('playback', media);
> >
> > All with the same inconsistent results.
> >
> > Any idea where I should be looking?
> >
> > Nick
> >
> >
> > On Fri, May 2, 2008 at 11:57 AM, Michael S Collins <
> > mercutio.viz at gmail.com> wrote:
> >
> > > Nick,
> > >
> > > I don't know if it was meant to be this way but your first script has
> > > /
> > > tmp/test.wav and your second script has /mnt/test.wav - was that
> > > intentional?
> > >
> > > -MC
> > >
> > > Sent from my iPhone
> > >
> > > On May 1, 2008, at 9:28 AM, "Nick Temple" <nick at intellispire.com>
> > > wrote:
> > >
> > > > Hello,
> > > >
> > > > I am extremely new to VOIP in general and have been using freeswitch
> > > > only a couple of days,
> > > > so if I have missed something in the docs please point me in the
> > > > right direction.
> > > >
> > > > I am attempting to create an outbound call system -- mostly research
> > > > and playing at
> > > > this point.
> > > >
> > > > The initial goal is to have the system ring a number, then deliver a
> > > > wave file. I have
> > > > it (mostly) working in two different flavors.
> > > >
> > > > The first seems to work the best right now:
> > > >
> > > > (works fine via XMLRPC or the CLI):
> > > > originate sofia/gateway/lesnet/1502NNNNNNN '&javascript(call1.js)'
> > > >
> > > > call1.js:
> > > > session.waitForAnswer(10000);
> > > > session.streamFile("/tmp/test.wav");
> > > > exit();
> > > >
> > > >
> > > > The second way does everything in Javascript:
> > > > jsrun call2.js
> > > >
> > > > where call2.js is nearly as trivial:
> > > > sessionx = new Session();
> > > > if (sessionx.originate(session, "sofia/gateway/lesnet/1502NNNNNNN",
> > > > 30)) {
> > > >   sessionx.waitForAnswer(10000); // Wait 30 seconds
> > > >   sessionx.streamFile("/mnt/test.wav");
> > > > };
> > > > exit();
> > > >
> > > >
> > > > The difference is that while both methods work on my land line, only
> > > > #1 works when calling my cell phone ...
> > > > for #2, the call is made, but I can't hear the media file playing.
> > > >
> > > > After lots of experimentation, the only thing I can figure at this
> > > > point is that call1.js has some
> > > > sort of defaults setup that I fail to do in call2.js
> > > >
> > > > Thanks for a great product!
> > > >
> > > > Nick
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > _______________________________________________
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> >
> >  Brian West
> > sip:brian at freeswitch.org
> >
> >
> >
> >
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>
> Brian West
> sip:brian at freeswitch.org
>
>
>
>
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