I told you what to do if you want to have the answer.<div>do I need to repost it or did you get the last email?</div><div><br><br><div class="gmail_quote">On Wed, Apr 28, 2010 at 5:08 AM, Frank @ Impact <span dir="ltr"><<a href="mailto:frank@impactfax.com">frank@impactfax.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="blue">
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<p class="MsoNormal"><font size="3" color="blue" face="Arial"><span style="font-size:12.0pt;font-family:Arial;color:blue">Yes. I understand your position of trying
to help people using old code. <span> </span>That is
why I tried to download the latest code to try to stay current.</span></font></p>
<p class="MsoNormal"><font size="3" color="blue" face="Arial"><span style="font-size:12.0pt;font-family:Arial;color:blue"> </span></font></p>
<p class="MsoNormal"><font size="3" color="blue" face="Arial"><span style="font-size:12.0pt;font-family:Arial;color:blue">But that is also when I found this problem
with the newer code.</span></font></p><div><div></div><div class="h5">
<p class="MsoNormal"><font size="3" color="blue" face="Arial"><span style="font-size:12.0pt;font-family:Arial;color:blue"> </span></font></p>
<p class="MsoNormal" style="margin-left:.5in"><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma">-----Original Message-----<br>
<b><span style="font-weight:bold">From:</span></b>
<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>
[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b><span style="font-weight:bold">On Behalf Of </span></b>Anthony Minessale<br>
<b><span style="font-weight:bold">Sent:</span></b> Tuesday, April 27, 2010
11:40 AM<br>
<b><span style="font-weight:bold">To:</span></b>
<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [Freeswitch-users]
session.steamFile misses DTMF event forfirst few seconds</span></font></p>
<p class="MsoNormal" style="margin-left:.5in"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal" style="margin-right:0in;margin-bottom:12.0pt;margin-left:.5in"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">12XXX is so many years old, i wish users who want free
help would at least stay up to date with the code.<br>
<br>
FYI, our repo is on git now and the svn mirror is not updating at the moment.<br>
<br>
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.<br>
<br>
Open an issue on <a href="http://jira.freeswitch.org" target="_blank">http://jira.freeswitch.org</a>
and attach the script.<br>
<br>
<br>
</span></font></p>
<div>
<p class="MsoNormal" style="margin-left:.5in"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <<a href="mailto:frank@impactfax.com" target="_blank">frank@impactfax.com</a>> wrote:</span></font></p>
<p class="MsoNormal" style="margin-left:.5in"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">The calls are coming from land based lines.
Traditional POTS (not cable<br>
company). Caller is calling into FS and providing those DTMF. I can<br>
reproduce on my POTS line (3000' from CO) and there is no discernable<br>
noise on the line.<br>
<br>
The call comes into the media gateway and then goes sip to FS.<br>
<br>
I can reproduce the problem just by starting FS version from the latest<br>
trunk. And then I can eliminate the problem by restarting the FS<br>
version 12790. In both test cases, the media gateway remains constant.</span></font></p>
<div>
<div>
<p class="MsoNormal" style="margin-left:.5in"><font size="3" face="Times New Roman"><span style="font-size:12.0pt"><br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a><br>
[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
On Behalf Of<br>
Alberto Escudero<br>
Sent: Tuesday, April 27, 2010 9:08 AM<br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for<br>
first few seconds<br>
<br>
How are you receiving those DTMFs, inbound or outbound? Are those calls<br>
coming from a mobile network (GSM). We have experienced lots of problem<br>
with DTMF detection in noisy lines.<br>
<br>
/aep<br>
--<br>
Stopping junk mailers is good for the environment<br>
<br>
> I recently upgraded from FS 12790M to svn 17188. When I did, I<br>
noticed<br>
> that session.streamFile behaved differently and I started having<br>
> problems with my IVR app.<br>
><br>
> With the upgraded FS, I have a problem with streamFile no firing on<br>
the<br>
> DTMF and calling the callback function for the first few seconds of<br>
the<br>
> wav file playback. It behaves as though it does not hear the DTMFs.<br>
If<br>
> I wait for 2 seconds or so of the wav file and then DTMF, streamFile<br>
> catches the DTMF and all is well. If I key as soon as I hear the wav<br>
> file start, streamFile just keeps playing the wav and does not call<br>
the<br>
> callback function.<br>
><br>
> When I revert back to the previous version of FS, streamFile always<br>
> fires the callback right away no matter how quickly I press the first<br>
> DTMF as the wav file starts to stream out.<br>
><br>
> The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM,<br>
> 16 bit, mono 8000 Hz<br>
><br>
> The snippet of js code I am using is as follows.<br>
><br>
> if(session.ready()) {<br>
> session.answer();<br>
> session.sleep(750);<br>
> while(session.ready()) {<br>
> session.sleep(500);<br>
> session.flushDigits(); // clear out input
buffers<br>
><br>
><br>
><br>
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi<br>
> ts_cb,""))===false) {<br>
>
pin=session.getDigits(pinmax,pinterm,pinwait);<br>
> } else {<br>
>
pin+=session.getDigits(pinmax-1,pinterm,pinwait);<br>
> }<br>
> // more code here..<br>
> }<br>
><br>
> Do I need to change the way I use streamFile in the later release?<br>
> _______________________________________________<br>
> FreeSWITCH-users mailing list<br>
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-- <br>
Anthony Minessale II<br>
<br>
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