I'm not sure but I think PIKA cards in openzap use hardware dtmf to avoid this issue, maybe we can figure out how to do the same thing with Sangoma.<br><br><br><div class="gmail_quote">On Wed, Jun 25, 2008 at 10:40 AM, Steve Laroche <<a href="mailto:slaroche@ip5.com">slaroche@ip5.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-US">
<div>
<p><span style="font-size: 10pt; color: blue;">Anthony, </span></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<p><span style="font-size: 10pt; color: blue;">Yes, it does work with echo canceller off. But without echo
canceller, when we bridge 2 calls together, we have echo on the line for the
voice and</span></p>
<p><span style="font-size: 10pt; color: blue;">the DTMF. For the DTMF, its crazy, one DTMF can loop
forever. I'm genna contact sangoma to do a follow up on that, but
do you have any T1</span></p>
<p><span style="font-size: 10pt; color: blue;">card provider that works without any problem that you guys work
with? </span></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<p><span style="font-size: 10pt; color: blue;">Thanks.</span></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<p><span style="font-size: 10pt; color: blue;">SL</span></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<p style="margin-bottom: 6pt;"><b><span style="font-size: 10pt; color: navy;"> </span></b></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Anthony Minessale<br>
<b>Sent:</b> Wednesday, June 25, 2008 10:30 AM<div><div></div><div class="Wj3C7c"><br>
<b>To:</b> <a href="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank">freeswitch-dev@lists.freeswitch.org</a><br>
<b>Subject:</b> Re: [Freeswitch-dev] DTMF issue on playback command with
openzap</div></div></span></p>
</div><div><div></div><div class="Wj3C7c">
<p> </p>
<p style="margin-bottom: 12pt;">just for fun try it with the
echo canceler off.<br>
I know for a fact that sangoma's echo canceler messes up the soft dtmf.<br>
<br>
see the power of details? ;)<br>
<br>
<br>
</p>
<div>
<p>On Tue, Jun 24, 2008 at 6:43 PM, Steve Laroche <<a href="mailto:slaroche@ip5.com" target="_blank">slaroche@ip5.com</a>> wrote:</p>
<div>
<div>
<p>I'm driving the call with an IVR outside of the freeswitch using
event_socket. I'm using</p>
<p>digital lines. In fact, they are T1 with DMS100 protocol connected to
a sangoma card.</p>
<p>The driver I'm using for the sangoma card is wanpipe-3.2.5. This card
have echo </p>
<p>canceller activated on.</p>
<p> </p>
<p>IVR originate a call through event_socket :</p>
<p> </p>
<p>bgapi originate openzap/1/a/5147884250 &park()</p>
<p> </p>
<p>The call is originated to a PSTN phone.</p>
<p> </p>
<p>When IVR receive the events from event_socket confirming the call is
connected, IVR</p>
<p>send a playback command :</p>
<p> </p>
<p>SendMsg uuid</p>
<p>call-command: execute</p>
<p>execute-app-name: playback</p>
<p>execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav</p>
<p> </p>
<p>Audio file is about 25sec long, so when the playback is executed on line, I
enter DTMF</p>
<p>on my PSTN phone.</p>
<p> </p>
<p>I'm collecting the DTMF events with my system that drives the event_socket.
It's </p>
<p>basically my test.</p>
<p> </p>
<p>Thanks.</p>
<p> </p>
<p>SL</p>
<p> </p>
<div style="border-style: solid none none; border-color: -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Anthony Minessale<br>
<b>Sent:</b> Tuesday, June 24, 2008 6:39 PM</span></p>
<div>
<div>
<p><span style="font-size: 10pt;"><br>
<b>To:</b> <a href="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank">freeswitch-dev@lists.freeswitch.org</a><br>
<b>Subject:</b> Re: [Freeswitch-dev] DTMF issue on playback command with
openzap</span></p>
</div>
</div>
</div>
<div>
<div>
<p> </p>
<p style="margin-bottom: 12pt;"><br>
I understand your problem so you need not repeat anything you have already posted.<br>
<br>
please describe in detail exactly what you are doing and how <br>
when you say openzap specify if it's analog or digital etc<br>
<br>
I still cannot tell how you are using openzap in this scheme <br>
<br>
are you saying you are doing something like:<br>
<br>
originate openzap/1/a/12121231234 1000<br>
<br>
to call your pots phone and xfer to 1000<br>
<br>
where an ivr is collecting dtmf?<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
</p>
<div>
<p>On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <<a href="mailto:slaroche@ip5.com" target="_blank">slaroche@ip5.com</a>> wrote:</p>
<div>
<div>
<p><span style="font-size: 10pt; color: blue;">Anthony, do you need more info ? </span></p>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
<div style="border-style: solid none none; border-color: -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">freeswitch-dev-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Anthony Minessale<br>
<b>Sent:</b> Monday, June 23, 2008 2:33 PM</span></p>
<div>
<p><span style="font-size: 10pt;"><br>
<b>To:</b> <a href="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank">freeswitch-dev@lists.freeswitch.org</a><br>
<b>Subject:</b> Re: [Freeswitch-dev] DTMF issue on playback command with
openzap</span></p>
</div>
</div>
<p> </p>
<p style="margin-bottom: 12pt;">please describe the entire path of the call and
what protocol is involved on the example that does not work.</p>
<div>
<div>
<p style="margin-bottom: 12pt;"><br>
<br>
starting with the telephone that placed the call all the way to the telephone
or ivr that revived it.<br>
<br>
</p>
</div>
</div>
<div>
<div>
<div>
<p>On Mon, Jun 23, 2008 at 1:20 PM, Brian West <<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>>
wrote:</p>
<div>
<p>You still didn't answer what was receiving the audio. If its OpenZAP
that means its inband. But can you elaborate more on the path? If
the audio has ANY gaps in it the DTMF will be double detected but I need to
know more about that before I can tell what is going on.</p>
<div>
<p> </p>
</div>
<div>
<p><span style="color: rgb(136, 136, 136);">/b</span></p>
</div>
<div>
<div>
<p> </p>
<div>
<div>
<p>On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:</p>
</div>
<p style="margin-bottom: 12pt;"> </p>
<div>
<p><span style="font-size: 10pt; color: blue;">Instead of
123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*.
While for all the other test that I did I do receive
123456789012345678901234567890*. I did this test many time and always the
same result, I have an issue with DTMF when playback is running and the call is
originate through openzap.</span></p>
</div>
<div>
<p><span style="font-size: 10pt; color: blue;"> </span></p>
</div>
</div>
<p> </p>
</div>
</div>
</div>
<p style="margin-bottom: 12pt;"><br>
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<br>
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<br>
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