<div dir="ltr"><div class="gmail_default" style="color:#660000">It's definitely possible. Many moons ago (before mixmonitor) I once hacked the mixing script in FreePBX to use sox to build a stero wav file from the two recordings - so I understand the usefulness of what you're trying to do.</div>
<div class="gmail_default" style="color:#660000"><br></div><div class="gmail_default" style="color:#660000">Be aware that both the Asterisk project and the FreePBX project are open source, and you can contribute to either or both.</div>
<div class="gmail_default" style="color:#660000"><br></div><div class="gmail_default" style="color:#660000">If I'm not mistaken (willing to be corrected on the internet if I'm wrong ;-) the WAV handling code is fixed in mono and 8khz (or at least used to be) and it may be tricky to add in stereo support. That being said, it's certainly possible to do it within Asterisk. It would require changes to both MixMonitor() or creation of StereoMixMonitor and the WAV handling. The part that I'm fuzzy on is how best to pass the two channels between them.</div>
<div class="gmail_default" style="color:#660000"><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Nov 14, 2013 at 3:57 AM, Alex Barnes <span dir="ltr"><<a href="mailto:Alex.Barnes@completeautomotivesolutions.co.uk" target="_blank">Alex.Barnes@completeautomotivesolutions.co.uk</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-GB" link="blue" vlink="purple">
<div>
<p class="MsoNormal">Hi all,</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I was hoping somebody knowledgeable about the inner workings of app_mixmonitor could give me an idea of how feasible our change might be?</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">User Story:</p>
<p class="MsoNormal">As a FreePBX user I would like MixMonitor to save recordings in stereo wav format where inbound calls are one channel and outbound are another.</p>
<p class="MsoNormal">This will allow me to more clearly hear who is speaking as well as more easily extract the audio of just one speaker.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Note:</p>
<p class="MsoNormal">The “FreePBX” part is important as we cannot make it use Monitor and an external bash script without hacking apart a lot of the dial plan, hence we are stuck using MixMonitor.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">We are a dev company but with little C experience and zero Asterisk application development knowledge HOWEVER we’re very happy to give it a whirl.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">I’m literally just starting on researching this and no doubt have lots of reading to do regarding general Asterisk app development, research WAV format issues in C and MixMonitor, how to submit changes to the Asterisk project etc etc. I
was hoping somebody could let me know if they think this will definitely not work or if they happen to know of some of the issues we may face and possibly where to start looking.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Thanks in advance</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Alex</p>
<p class="MsoNormal"> </p>
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