[Freeswitch-users] Session Question

[Brian West]

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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