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FireBrick Call Recording: Difference between revisions

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=Overview=
 
The FireBrick supports call recording by teeing off the two way audio from a call leg and sending to a SIP endpoint. The SIP endpoint will then record the call and handle it in any way you wish.
 
 
There are two elements to the A&A/FireBrick call recording software:
* a C application, ''voip-recordanswer'', which acts as a SIP endpoint and handles the call recording; and
* a python script
 
==Setting up ''voip-recordanswer''==
 
The ''voip-recordanswer'' application comes with a Makefile, but I found I needed to make a couple of tweaks to get it to work:
 
1.) the Makefile is dependent on deps/siptools.c. If you do not have access to the A&A svn, you'll need to:
a.) comment out the following lines in the Makefile:
 
'' svn export --force -r 28442 svn://changeless.aa.net.uk/fb6000/trunk/apps/siptools.c $@''
'' touch $@ # SVN sets the timestamp to the committed one (which triggers "building" every time)''
 
b.) make sure that you have siptools.c in dep/
 
2.) depending on your version of voip-answer.c, you may need to set the email address parameter (i.e. the ''from'' address). Change line 803 from:
 
'' sprintf (temp, "sip:%c@voiceless.aa.net.uk", done);''
 
to whatever email address you want to the recordings to be sent from. (%c is the unique ID of the call recording)
 
(As of 9 August 2018, this has been replaced in voip-answer.c with "sip:%s%.*s", so no modification is necessary.)
 
3.) By default, the (temporary) path for the recordings is /tmp/. If you want to change this, amend line 62:
 
'' char template[] = "/tmp/voip-answer-XXXXXX";''
 
 
Once you have configured the script to your preferences, install it by running ''make''.
 
(If you do not have it installed already, you might need to install libpopt-dev.)
 
 
=Running the A&A / FireBrick call recording and emailing software=
 
The A&A/Firebrick call recording software has seven possible arguments / switches:
Coming soon...
 
* ''rec-script'' / ''r'': path to your recording script
Describe variables/parameters
* ''save-script'' / ''s'': path to your saved file script
* ''bind-host'' / ''h'': binds host to the hostname you provide
* ''bind-port'' / ''p'': binds to the port you provide
* ''directory'' / ''d'': sets the directory path for wav files to be used in audio playback / announcements (see below)
* ''debug'' / ''v'': enables verbose / debug mode
* ''dump'' / ''V'': outputs a packet dump
 
So, for example, if you wanted to run ''voip-answer'' with a call recording script at ''/voipuser/voip-answer/python/voip-rec-email/voip_rec_email.py'', serving wav files for announcements from ''/voipuser/voip-answer/sounds/'', you would run it with:
 
./voip-answer -r /voipuser/voip-answer/python/voip-rec-email/voip_rec_email.py -d /voipuser/voip-answer/sounds/
 
If you want to run it as a background process (which you probably do...), append an ampersand:
 
./voip-answer -r /voipuser/voip-answer/python/voip-rec-email/voip_rec_email.py -d /voipuser/voip-answer/sounds/ &
 
=Using the A&A / FireBrick call recording software for announcements (e.g. out of hours announcement)=
 
As well as acting as a SIP endpoint for recording calls, ''voip-answer'' doubles as a playback system, which you can use to play announcements.
Holding notes:
 
== Creating the announcement / playback files ==
 
The sound files used for announcements / playback must meet all the following:
* .wav
* A-LAW
* 8000 Hz sample rate
* mono only
 
You can easily use Audacity to create suitable files; just remember to set the sample rate to 8000 Hz, and to set the track to mono only, first.
 
If, when you play the files back through ''voip-answer'', they sound distorted, chances are you have either got the sample rate wrong, or else accidentally recorded them in stereo.
 
If you have a recording at a different sample rate, using Audacity, change the project's sample rate to 8000, then change the recording's sample rate to 8000, and then use the "Change Speed" effect on the recording — if the original rate was 44100 Hz, apply a speed multiple of 5.50.
 
== Hosting the announcement / playback files ==
 
Files to be played back can be put in any directory readable by ''voip-answer''. You tell it which directory to look in by using the ''directory'' or ''-d'' switch. For example:
 
./voip-answer -d /voipuser/voip-answer/sounds/
 
== Triggering the announcement / playback files ==
 
You send traffic to the announcement / playback files by configuring a new extension on the FireBrick, and giving it a SIP URI containing the relevant parameters.
 
The parameters you can choose from are:
 
=== Prefixes (in this order) ===
XXX= This call is not to be answered, but call progress. XXX is final status
= This call is not to be answered, but call progress. End with constant ringing
- Add a ring, may be repeated
! Add a SIT, may be repeated
N* The playback sequence is to be repeated N times
 
You can download aai.wav and sit.wav [https://support.aa.net.uk/images/9/98/Firebrick_tones.zip here].
 
=== Filenames ===
Following the prefix are dot separated filenames, assumed to be wav files, to play.
 
A filename can have a ? after it followed by another filename. This skips the second file if the first exists.
 
Each dot is also a small time delay, so several dots can be used for a pause.
=== Suffixes, one applies ===
=filename Record to file
* Silence
# Refer to #
 
So if you wanted two rings, followed by playback of a file called "onholiday.wav", you would specify the following SIP URI:
 
''--onholiday@hostname/address''
 
=Pre-call announcements (e.g. "this call will be recorded")=
 
You can combine these features with the REFER function ("#"), to create an extension which will first play back an announcement (such as "This call will be recorded"), before referring the call to your chosen destination.
 
You do not need to do anything with the voip-answer.c script (other than make sure your chosen recordings are in the right place).
 
Configure your FireBrick:
 
# As at 10 August 2018, make sure you have alphas enabled, and have at least alpha V1.48.123
# Create a new telephone extension (or amend an existing one)
# Give it the username "Voicemail" and do not give it a password
# Add the URI: <your-pre-call-sound-file>#<extension you want to refer to>@hostname/address
 
So if you wanted to first play an announcement of called "callsarerecorded.wav", followed by a refer to your extension or hunt group number 500, you would use:
Audacity, 8000Hz sample rate, mono only, output as uncompressed .wav A-LAW. If playback is distorted, chances are you've got the sample rate wrong, or you are using a stereo-encoded track.
 
callsarerecorded#500@hostname/address
Save to voip-record/c/bin/
 
Reference in SIP URI in FireBrick config (e.g. recording-name@hostname/address)
 
 
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