[Freeswitch-users] Session Question

[Nick Temple]

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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