TIPS Name announce in conference?

[bigunk]

PBX in a Flash PURPLE Status Program

SYSTEM INFORMATION *VERIFIED
x Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE x
x SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE x
x Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = OFFLINE x
x Disk Free = ADEQUATE| Mem Free = ADEQUATE| NTPD = ONLINE x
x Postfix = ONLINE | Samba = OFFLINE | Webmin = ONLINE x
x Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A x
x x
x PIAF Installed Version = 2.0.6.5 under *HARDWARE* x
x FreePBX Version = 12.0.47 1 x
x Running Asterisk Version = 1.8.28.0 x
x Asterisk Source Version = 1.8.28.0 x
x Dahdi Source Version = 2.9.0 x
x Libpri Source Version = 1.4.14 x
x IP Address = 192.168.40.2 on eth0 x
x Operating System = CentOS release 6.5 (Final) x
x Kernel Version = 2.6.32-431.1.2.0.1.el6.i686 - 32 Bit x
x x


Okay, so here's my setup. Seems when people dial into conference rooms, they're asked for the password, and then the welcome prompt is played to them. They get placed into the conference and a quick tone is played to let existing participants know someone has joined. So the question is how to get PIAF to require the caller to record their name and play that to the conference as that person enters?

 

[bigunk]

This is what I see when I try to edit a conference.

 

[bigunk]

So from what I'm reading, Asterisk 1.8 hasn't got this functionality. Is there a way to install a different version with the things I need without having to redo the whole system from scratch?

 

[randy7376]

bigunk

If I remember this correctly (and I probably don't!), I think you're experiencing the differences in functionality between app_meetme and app_confbridge. It looks like you're running the same version of FreePBX that I am on the home system. If you'll go into Settings -> Advanced Settings -> Dialplan and Operational -> Conference Room App, it should show you which application you're currently using (probably app_confbridge from the sound of it). You can switch the application over to app_meetme, But, from what I remember, it requires that you delete the existing conference and recreate it. After that, the recorded name should function. Be aware that app_meetme is considered deprecated in later versions of Asterisk.

Lastly, you might want to check /usr/lib/asterisk/modules for app_meetme.so and app_confbridge.so to verify that both applications exist before making any changes.

My two cents...

 

[bigunk]

Hmm... Is it just as simple as deleting the conferences (I have 6 of them), making the change in that drop-down box, and then rebuilding the conferences?

 

[bigunk]

I hear from Ward Mundy that we can do an upgrade to a newer version of FreePBX from within the GUI. Interesting and scary as hell. My questions on this are 1) will my database and all voice recordings, etc., survive the upgrade? How to do it in the GUI? Any gotchas or procedures to do pre-flight? Does the fact that this is a PIAF install cause anything to break when doing such an upgrade? Lastly, I've heard you cannot do a backup and restore it to a major upgrade. Thoughts?

 

[wardmundy]

From your first post, it appears you're already on FreePBX 12 so this sounds more like an Asterisk version deficiency. What you could do is use a tool such as Incredible Backup to make a backup of your existing setup. Then build a new server with Asterisk 11 and FreePBX 12. Then restore the backup. Should catch just about everything. Make a full image backup before you begin... just in case. Plenty of existing threads on how best to do that.

 

[bigunk]

Okay, so there's a rebuild in my future I guess. Man, am I scared to death of this. So will IB get everything? Including voice messages and prompts? Trust me when I tell you this client does not want to re-record stuff. If I'm running FreePBX 12 and I just upgrade the asterisk, is that considered an upgrade that might not restore? I guess I'm a bit confused between upgrades of FreePBX and asterisk. Be gentle with me, as this is my very first, and so far, only, install of anything asterisk so far. I decided I was going to do it the hard way. No lab time, no small test systems, nothing. Just 51 phones, PRI, copper, and SIP trunks in a box. It's been quite a ride, and it ain't over yet.

 

[bigunk]

Then there's this:

Could it really be that easy, or is there something in PIAF/FreePBX/asterisk that will break things?

http://wiki.freepbx.org/display/FD/Changing Major Asterisk Versions on the Fly