Hi,

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.

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.

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.

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.

Many Thanks

Alex

From: Alex Barnes
Sent: 14 November 2013 16:19
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] Extend MixMonitor to record stereo files

Thanks for your reply Scott.

We hadn't noticed the new MixMonitor options 'r' and 't'; I must stop using the 
voip-info wiki rather than the new docs :)

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.

I'll continue reading around the Asterisk source in the meantime though so 
thanks again for pointing out the WAV handling issue.

Kind Regards

Alex

From: 
[email protected]<mailto:[email protected]>
 [mailto:[email protected]] On Behalf Of Scott Griepentrog
Sent: 14 November 2013 15:16
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Extend MixMonitor to record stereo files

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.

Be aware that both the Asterisk project and the FreePBX project are open 
source, and you can contribute to either or both.

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.


On Thu, Nov 14, 2013 at 3:57 AM, Alex Barnes 
<[email protected]<mailto:[email protected]>>
 wrote:
Hi all,

I was hoping somebody knowledgeable about the inner workings of app_mixmonitor 
could give me an idea of how feasible our change might be?

User Story:
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.
This will allow me to more clearly hear who is speaking as well as more easily 
extract the audio of just one speaker.

Note:
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.

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.

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.

Thanks in advance

Alex




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