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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Hi,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Unfortunately using MixMonitor and the options ‘r’ and ‘t’ didn’t work. The two files created are not synchronised (different lengths) depending on various
scenarios. For example who initiated the call and then who hung-up first.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">After reading through the source code I’m not convinced that our own dev team is the most viable resource to extend the MixMonitor app and associated WAV areas.
Minimal C dev experience partly but mostly it feels like quite a lot of Asterisk source and audio processing skills are required.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Is there a recommended way to offer a bounty for feature requests (I have read the wiki on Bug Bounties). I’m sure I can find budget to sponsor this feature,
although it may not be enough if this is a really big bit of work :) I don’t want to just post up a random figure which may be insulting to the amount of work required, no more than wasting money.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I did post a message on the Asterisk Forums but haven’t yet had a reply, so apologies for the duplication if you have already read that.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Many Thanks<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Alex<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Alex Barnes
<br>
<b>Sent:</b> 14 November 2013 16:19<br>
<b>To:</b> Asterisk Developers Mailing List<br>
<b>Subject:</b> RE: [asterisk-dev] Extend MixMonitor to record stereo files<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Thanks for your reply Scott.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">We hadn’t noticed the new MixMonitor options ‘r’ and ‘t’; I must stop using the voip-info wiki rather than the new docs :)
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">We are currently testing to see if setting both options will produce two separate (or three maybe looking at the app_monitor.c file) wav files that we can merge
together like we do when using Monitor. If this does work then extending FreePBX to pass these extra options should be trivial in comparison to adding stereo WAV support to Asterisk itself.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">I’ll continue reading around the Asterisk source in the meantime though so thanks again for pointing out the WAV handling issue.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Kind Regards<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Alex<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a> [<a href="mailto:asterisk-dev-bounces@lists.digium.com">mailto:asterisk-dev-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Scott Griepentrog<br>
<b>Sent:</b> 14 November 2013 15:16<br>
<b>To:</b> Asterisk Developers Mailing List<br>
<b>Subject:</b> Re: [asterisk-dev] Extend MixMonitor to record stereo files<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span 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.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#660000"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span 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.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#660000"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span 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.<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="color:#660000"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><o:p> </o:p></p>
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<p class="MsoNormal">On Thu, Nov 14, 2013 at 3:57 AM, Alex Barnes <<a href="mailto:Alex.Barnes@completeautomotivesolutions.co.uk" target="_blank">Alex.Barnes@completeautomotivesolutions.co.uk</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hi all,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I was hoping somebody knowledgeable about the inner workings of app_mixmonitor could give me an idea of how feasible our change might be?<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">User Story:<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">This will allow me to more clearly hear who is speaking as well as more easily extract the audio of just one speaker.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Note:<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">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.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thanks in advance<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Alex<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p> <o:p></o:p></p>
<p><span style="font-size:10.0pt;font-family:"Calibri","sans-serif"">The Manheim group of companies within the UK comprises: Manheim Europe Limited (registered number: 03183918), Manheim Auctions Limited (registered number: 00448761), Manheim Retail Services
Limited (registered number: 02838588), <a href="http://Motors.co.uk" target="_blank">
Motors.co.uk</a> Limited (registered number: 05975777), Real Time Communications Limited (registered number: 04277845) and Complete Automotive Solutions Limited (registered number: 05302535). Each of these companies is registered in England and Wales with the
registered office address of Central House, Leeds Road, Rothwell, Leeds LS26 0JE. The Manheim group of companies operates under various brand/trading names including Manheim Inspection Services, Manheim Auctions, Manheim Direct, Manheim De-fleet and Manheim
Aftersales Solutions.</span> <o:p></o:p></p>
<p><span style="font-size:6.0pt;font-family:"Calibri","sans-serif"">V:0CF72C13B2AC<o:p></o:p></span></p>
<p> <o:p></o:p></p>
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<p class="MsoNormal"><img border="0" width="288" height="50" id="_x0000_i1025" src="https://my.digium.com/images/graphics/digium_RGB_signature.gif" alt="Digium logo"><span style="font-size:9.0pt;font-family:"Arial","sans-serif";color:black"><br>
<strong><span style="font-family:"Arial","sans-serif"">Scott Griepentrog</span></strong><br>
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