ConfBridge

A few questions relating to ConfBridge use with FreePBX. I’m using the distro Centos/FreePBX package.

We’re a 10 person small business. No POTS anything - completely VOIP and I want to ditch meetme for confbridge for improved quality. What I’ve been able to find has been rather sparse - but from I’ve read is that FreePBX will default to ConfBridge only if Meetme is not installed. I tried to force, clumsily, Asterisk to use confbridge and it did not work. Anyway, I’m a “seasoned newbie” and here are my questions:

  1. Does anyone have any suggestions on how to get confbridge working with a FreePBX system?
  2. How can I confirm if the “app” confbridge is working and installed?
  3. Is the general consensus that if you don’t need DAHDI confbridge is the way to go (especially relating to issues of voice quality)?

Move the file app_meetme.so out of /usr/lib/asterisk/modules
Restart Asterisk
In FreePBX, save one extension without changing anything, then Apply Changes.

If you now check Conferences you will see less options, but it will use ConfBridge.

Please report your findings so that we know that it work as it should.

Edit: I forgot, you need to save any Conference that you have created to update MeetMe to ConfBridge

What you suggest works perfectly (and simply!). One last question, in order to fine-tune the ‘mix_sample_rate’ setting where would be the best place to do that without hampering any of the FreePBX settings?

Create a file called confbridge.conf and put it in /etc/asterisk
Modify it to something like this:

[general]
mix_interval=20
mix_sample_rate=dynamic ;adjust to best quality sample rate for the channels joined.
denoise=yes ; denoise every channel that joins the bridge

Read more here:
http://www.venturevoip.com/news.php?rssid=2509

I just noticed that there are no confbridge.conf.sample in Asterisk source.

I’m sorry. My problem is I was playing a join announcement that was the same as the IVR announcement and it was taking forever to be joined in. Works perfectly with SIP the way it is.

To continue my original post (and leave this behind for posterity since it was hard to find info online about it) I’m posting what I had to do in order to get the enter/leave sounds working AND provide a workaround for recording confbridge conferences. This is using Asterisk 1.8. I’m really excited about the changes to Confbridge in 1.10 but am not ready to make the jump just as yet.

Basically I built my conference in freepbx then copied the [ext-meetme] into extensions_override_freepbx.conf and made changes.

[ext-meetme]
include => ext-meetme-custom
exten => STARTMEETME,1,ExecIf($["${MEETME_MUSIC}" != ""]?Set(CHANNEL(musicclass)=${MEETME_MUSIC}))
exten => STARTMEETME,n,Set(GROUP(meetme)=${MEETME_ROOMNUM})
exten => STARTMEETME,n,GotoIf($[${MAX_PARTICIPANTS} > 0 && ${GROUP_COUNT(${MEETME_ROOMNUM}@meetme)}>${MAX_PARTICIPANTS}]?MEETMEFULL,1)
exten => STARTMEETME,n,ConfBridge(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})
exten => STARTMEETME,n,Hangup
exten => MEETMEFULL,1,Playback(im-sorry&conf-full&goodbye)
exten => MEETMEFULL,n,Hangup
exten => h,1,Hangup
exten => 99,1,Macro(user-callerid,)
exten => 99,n,Set(MEETME_ROOMNUM=99)
exten => 99,n,Set(MAX_PARTICIPANTS=0)
exten => 99,n,Set(MEETME_MUSIC=default)
exten => 99,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?READPIN)
exten => 99,n,Set(CONFBRIDGE_JOIN_SOUND=beep)
exten => 99,n,Set(CONFBRIDGE_LEAVE_SOUND=beeperr)
exten => 99,n,Answer
exten => 99,n,Wait(1)
exten => 99,n,Set(PINCOUNT=0)
exten => 99,n(READPIN),Read(PIN,enter-conf-pin-number,,,,)
exten => 99,n,GotoIf($[x${PIN} = x113]?USER)
exten => 99,n,GotoIf($[x${PIN} = x1114]?ADMIN)
exten => 99,n,Set(PINCOUNT=$[${PINCOUNT}+1])
exten => 99,n,GotoIf($[${PINCOUNT}>3]?h)
exten => 99,n,Playback(conf-invalidpin)
exten => 99,n,Goto(READPIN)
exten => 99,n(ADMIN),Set(MEETME_OPTS=aAcM)
exten => 99,n,Goto(STARTMEETME,1)
exten => 99,n(USER),Set(MEETME_OPTS=cM)
exten => 99,n,GotoIf($[${GROUP_COUNT(${MEETME_ROOMNUM}@meetme)}=1]?REDBUTTON)
exten => 99,n,Goto(STARTMEETME,1)
exten => 99,n(REDBUTTON) ,MixMonitor(/var/www/html/conferences/conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.wav,,/myscripts/mp3process.sh conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.wav conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.mp3)
exten => 99,n,Goto(STARTMEETME,1)

exten => 99,hint,MeetMe:99

; end of [ext-meetme]

Really the following lines of code were added. . . .

Setting the join/leave sounds.

exten => 99,n,Set(CONFBRIDGE_JOIN_SOUND=beep)
exten => 99,n,Set(CONFBRIDGE_LEAVE_SOUND=beeperr)

I guess this is due to the way confbridge works with audio (and perhaps my ignorance) but mixmonitor should only instantiate on the joining of the second caller and start recording then. We don’t use leaders so this code works fine (we could add the same code while checks the leader as well if we wanted)

exten => 99,n,GotoIf($[${GROUP_COUNT(${MEETME_ROOMNUM}@meetme)}=1]?REDBUTTON)
exten => 99,n,Goto(STARTMEETME,1)
exten => 99,n(REDBUTTON) ,MixMonitor(/var/www/html/confcalls/conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.wav,,/myscripts/mp3process.sh conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.wav conference-${STRFTIME(${EPOCH},,%m_%d_%y)}.mp3)
exten => 99,n,Goto(STARTMEETME,1)

I’m passing the command at the end of the MixMonitor off to a script that converts the files to mp3 and sends them off to a google group email address.

#!/bin/bash
/usr/bin/lame --silent -s 8 -V 3 -B 96 -c /var/www/html/confcalls/$1 /var/www/html/confcalls/$2
wait
rm -f /var/www/html/confcalls/$1
echo "Company Conference" | mutt -a /var/www/html/confcalls/$2 -s "Company Conference Recorded $(date)" [email protected]
wait
rm -f /var/www/html/confcalls/$2

In FreePBX 2.10 go to advanced settings and set the default conference engine to be app_conf instead of meetme.

I’m having trouble getting the join/leave sounds to work with either ConfBridge or MeetMe. Is there a particular method or need to modify configuration files? Whenever I reload through FreePBX 2.10, it generates a new meetme_additional.conf even though I have ConfBridge selected.

EDIT:

Solved my issue. Turns out I’m not as careful with using Hyphens and Underscores as I thought I was. Makes a huge difference (Freezing up the server huge), so debugging really paid off.

Now I’m curious if there’s a method to have FreePBX make it work. Perhaps with confbridge.conf general settings modification? Need thoughts.

My company wants to have all conferences recorded and either FTP’d or Emailed to a location once they are complete… I have been searching for a couple days now and I can’t find a good solution.

Thanks!

-Mark

I have an ftp solution for /recordings, but it is not free. post an email or phone if you would like more info.

I am interested. sqlguy(at)yahoo(dot)com

thanks!