[Freeswitch-users] session.steamFile misses DTMF event forfirst few seconds

[Frank @ Impact]

Yes. I understand your position of trying to help people using old code.
That is why I tried to download the latest code to try to stay current.
 
But that is also when I found this problem with the newer code.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Tuesday, April 27, 2010 11:40 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event
forfirst few seconds
 
12XXX is so many years old, i wish users who want free help would at
least stay up to date with the code.

FYI, our repo is on git now and the svn mirror is not updating at the
moment.

produce a complete minimal script that reproduces your problem and can
be run on git HEAD (see download instructions to learn how to build with
git)  Use existing sound files from the FS install so we can just run it
in our lab to reproduce the issue.

Open an issue on http://jira.freeswitch.org and attach the script.



On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank at impactfax.com>
wrote:
The calls are coming from land based lines.  Traditional POTS (not cable
company).  Caller is calling into FS and providing those DTMF.  I can
reproduce on my POTS line (3000' from CO) and there is no discernable
noise on the line.

The call comes into the media gateway and then goes sip to FS.

I can reproduce the problem just by starting FS version from the latest
trunk.  And then I can eliminate the problem by restarting the FS
version 12790.  In both test cases, the media gateway remains constant.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Alberto Escudero
Sent: Tuesday, April 27, 2010 9:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for
first few seconds

How are you  receiving those DTMFs, inbound or outbound? Are those calls
coming from a mobile network (GSM). We have experienced lots of problem
with DTMF detection in noisy lines.

/aep
--
Stopping junk mailers is good for the environment

> I recently upgraded from FS 12790M to svn 17188.  When I did, I
noticed
> that session.streamFile behaved differently and I started having
> problems with my IVR app.
>
> With the upgraded FS, I have a problem with streamFile no firing on
the
> DTMF and calling the callback function for the first few seconds of
the
> wav file playback.  It behaves as though it does not hear the DTMFs.
If
> I wait for 2 seconds or so of the wav file and then DTMF, streamFile
> catches the DTMF and all is well.  If I key as soon as I hear the wav
> file start, streamFile just keeps playing the wav and does not call
the
> callback function.
>
> When I revert back to the previous version of FS, streamFile always
> fires the callback right away no matter how quickly I press the first
> DTMF as the wav file starts to stream out.
>
> The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,
> 16 bit, mono 8000 Hz
>
> The snippet of js code I am using is as follows.
>
> if(session.ready()) {
>     session.answer();
>     session.sleep(750);
>     while(session.ready()) {
>         session.sleep(500);
>         session.flushDigits(); // clear out input buffers
>
>
>
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
> ts_cb,""))===false) {
>             pin=session.getDigits(pinmax,pinterm,pinwait);
>         } else {
>             pin+=session.getDigits(pinmax-1,pinterm,pinwait);
>         }
>         // more code here..
>     }
>
> Do I need to change the way I use streamFile in the later release?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100428/3511776b/attachment-0001.html