Menu for MeetMe conference system


#1

After years of working with Nortel and other PBX systems, Asterisk is truly a wonderful breath of fresh air.

I’d like to contribute to this wonderful open-source community by working on creating a really nice menu system for the MeetMe feature in Asterisk. However, I have a few questions that I’ve been unable to answer.

  1. How do you change the tones played when someone joins/leaves a MeetMe conference? I’m referring to the little “blip” that is made when you join a MeetMe conference which does not have the option enabled to announce callers (there is still a little noise made though, instead of “Bob has joined the conference”)

  2. This may be a really novice question, so I apologize, but is it possible to change the configuration files while asterisk is running and reload them? I found “asterisk -r -x reload” but I’m thinking it’s not safe, especially if the files have changed and a caller is already in a call queue. Is the only other option to call AGI scripts that pull information from elsewhere (database, etc.)?

Thanks for your time! Asterisk is really very cool.


#2

Hi !

Asterisk is loading all configs into the RAM at start.

There are diff. ways to start asterisk.

  1. Daemon (via init.d eg - set daemon to respawn)
  2. Via Console (“Asterisk”)
  3. Via console with CLI (commandline interf.)

Esp. #3 is the common use.
Issue a “reload” (without “”) to reload all configs, valid for every next call.

Current calls are NOT affected by the new config, since their “threads” started with the config which were present at the time the call spawned.

Sometimes, a simple reload wont do the job, esp. when ZAP was changed.
Then you issue a “restart when convenient” which restarts asterisk as soon as there is no load/call present.

This restart takes not even a second, so no1 will realize that the PBX was just restarted !

To have an “standard” asterisk CLI running, you normally have verbose set to 5, to see the most common outputs and warnings.

My console:
asterisk -rvvvvvT

So:
“remote” (my asterisk is a daemon with respawn option)
vvvvv=Verbose level 5
T= (Note the uppercase) Timestamps on important outputs

You can even colorize the CLI:
asterisk -rvvvvTc

Colorizing does NOT work, when asterisk is a daemon and your CLI is a remote.

To archive this, you need to start the deamon with a colorized CLI already.


#3

FYI, I have had user calls DROPPED in last week when doing a reload (using Asterisk 1.2.7.1 and all latest stable releases). This was with SIP clients and SIP gateway for PSTN access.

For now, before I do a “reload” I do a “show channels” to make sure there are no active calls. Fortunately it’s not too busy yet on the system I can actually find idle times.