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Conferences (MeetMe)

in

Information

Conferences is a standard multi-party conferencing facility that is available as a destination.

Conference Details

  • Conference Number
    • This is a number that local users can dial to join the conference
  • Conference Name
    • This is used as an Identifier, along with the number, when picking a conference as a destination
  • User PIN and Admin PIN
    • If either of these options are set, anyone calling into the
      conference will be prompted for a PIN. If 'user' is left blank, they
      can just push '#' to enter. The only use of 'Admin' is to not actually
      open the conference until the admin user has arrived. If 'Music On
      Hold' is enabled, users will be placed on hold with the 'default' Music
      On Hold class.

Conference Options

  • Join Message
    • This is a sound that is played to all users upon entering.
  • Leader Wait
    • When there is an Admin PIN set, the conference won't start until the 'Admin' user joins. See above.
  • Quiet Mode
    • Usually a 'bing' noise is played when a user enter or leaves
      the conference, alerting other members to the fact that someone has
      joined or left. You can disable that by selecting 'Yes' here.
  • User Count
    • When someone joins, the conference will say 'There are (number) people in this conference"
  • User Join/Leave
    • When someone connects to the conference, it will ask them to
      record their name. The conference will then announce when they join and
      leave, by name.
  • Music On Hold
    • Totally enables or disables Music on Hold in this conference.
  • Allow Menu
    • Enables the user or admin to enter an the management mode by pushing '*'. The commands whilst in management mode are:
      • 1: Mute yourself
      • 4 or 6: Decrease or Increase the Conference Volume (eg, the sound you hear)
      • 7 or 9: Decrease or Increase your Volume (eg, the sound other people hear)
    • Additionally, Admin users have the added features of:
      • 2: Lock or Unlock the conference
      • 3: Eject the last person that called
  • Record Conference:
      Toggle this to yes if you would like to record the conference.

‹ Caller ID Lookup SourcesupCore Module ›

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When i dial the conference

shahab.sabir's picture
On September 20th, 2007 shahab.sabir (tadpole) said:

When i dial the conference number it says that it is not available to you.

Its not working properly


invalid pin

ip-rob's picture
On December 11th, 2007 ip-rob (tadpole) said:

We always receive an invalid pin on whatever is entered. We're running Asterisk 1.4.14 and FreePBX 2.3.1.1. I have double checked and triple checked the PIN's. I've also verified by monitoring the CLI that the proper PIN is being passed to the MeetMe function.

I've also noticed that only the participant PIN's are stored in the meetme_additional.conf.

Any ideas on what the issue may be? Right now conference calling is not working at all for us.

Rob


Resolution to invalid PIN

ip-rob's picture
On December 20th, 2007 ip-rob (tadpole) said:

Fixed by entering chown -R asterisk.asterisk /dev/zap

Rob


The conference number is not valid

tingwang's picture
On September 4th, 2008 tingwang (tadpole) said:

Hi Rob,

Thanks very much in advanced.
I just got knewn FreePBX days ago. I used SIPFoundy before. Even they are similiar, but I was failed to add a conference.

Once I added a conference number 7777 with no User and Admin PIN, the server would return invalid conference number.
Once I added a conference number 8888 with Admin PIN, it will return me invalid PIN number.
Once I added a conference number 9999 with User and Amin PIN, it also return me invalid PIN number.

In addition, I can not enable the 'join message' item. It always show 'none' and can not choose other option.

Best regards,
TingWang

Thanks very much for helping me.


Please don't post new

fskrotzki's picture
On September 4th, 2008 fskrotzki (Contributor) said:

Please don't post new questions here.

The proper place to post questions that people will see and respond to are in the forum. Posting questions under the documentation sections should be restricted to to the documentation provided at the top of the page (or a expansion/correction of the documentation provided) otherwise your question will get missed by 98% of the helping population on this site.


meet-me Feature Suggestions

jacknab's picture
On January 9th, 2008 jacknab (tadpole) said:

Hello,

I am not sure if FreePBX are the developers for the meet-me feature, if anyone could point me in the right direction be great.

Would like to offer some great suggestions, many great features I have seen other conference services offering and would like to see if some or all of these features could be implemented in to our meet-me software.

Caller count - *2 key (Host only)
Allows the host to get a count of how many callers are on the code.

Exit conference - *3 key
Pressing *3 Takes the individual user out of the conference call and back into the lobby.

Record Conference - *9 key (Host only)

Recording Instructions
During the free live conference, the host can start recording at any time by pressing *9. The system disconnects him/her from the free live conference and prompts for the Subscriber PIN. If the Subscriber PIN is valid, the host is returned to the conference and all attendees are notified that the recording option has been started. To stop the recording, the host will press *9 again.

Play Back Instructions
To listen to the recorded conference, the host must provide attendees with the play back number and access code. Please note, this number is different from the conference dial-in number. When users call into the recording playback number, they will be prompted to enter the access code. Once confirmed, the system will play back the recorded conference. During play back, the user can scroll forward or backwards through the recorded conference.
Playback Feature Keys
*1 Rewind 30 seconds
*2 Fast forward 30 seconds
*5 Pause/resume playback


conference not working

riddhesh's picture
On August 29th, 2008 riddhesh (tadpole) said:

hello all
thank for your help in advance

I am using freepbx 2.4.1 with asterisk 1.4.20.1. I m facing problem regarding conferencing
i had created conference room as 8200 and then try to call that room but the call is getting established the and in 2 second it got hangup. I changed codec to ulaw as i fond in one the forum but result was same. also i checked with zaptel settings also but there was no help having same result
i m sending u CLI wht i m getting on asterisk prompt

-- Executing [8200@from-internal:1] Macro("SIP/3001-b7d220f8", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/3001-b7d220f8", "user-callerid: device 3001") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/3001-b7d220f8", "AMPUSER=3001") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/3001-b7d220f8", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/3001-b7d220f8", "1|Set|REALCALLERIDNUM=3001") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/3001-b7d220f8", "REALCALLERIDNUM is 3001") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/3001-b7d220f8", "AMPUSER=3001") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/3001-b7d220f8", "AMPUSERCIDNAME=Alldean") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/3001-b7d220f8", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/3001-b7d220f8", "AMPUSERCID=3001") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/3001-b7d220f8", "CALLERID(all)="Alldean" <3001>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/3001-b7d220f8", "REALCALLERIDNUM=3001") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/3001-b7d220f8", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/3001-b7d220f8", "TTL: ARG1: ") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/3001-b7d220f8", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/3001-b7d220f8", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/3001-b7d220f8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/3001-b7d220f8", "Using CallerID "Alldean" <3001>") in new stack
-- Executing [8200@from-internal:2] Set("SIP/3001-b7d220f8", "MEETME_ROOMNUM=8200") in new stack
-- Executing [8200@from-internal:3] GotoIf("SIP/3001-b7d220f8", "0?READPIN") in new stack
-- Executing [8200@from-internal:4] Answer("SIP/3001-b7d220f8", "") in new stack
-- Executing [8200@from-internal:5] Wait("SIP/3001-b7d220f8", "1") in new stack
== Manager 'admin' logged off from 127.0.0.1
-- Executing [8200@from-internal:6] Set("SIP/3001-b7d220f8", "PINCOUNT=0") in new stack
-- Executing [8200@from-internal:7] Read("SIP/3001-b7d220f8", "PIN|enter-conf-pin-number||||") in new stack
-- User disconnected
== Spawn extension (from-internal, 8200, 7) exited non-zero on 'SIP/3001-b7d220f8'
-- Executing [h@from-internal:1] Macro("SIP/3001-b7d220f8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/3001-b7d220f8", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/3001-b7d220f8", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/3001-b7d220f8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/3001-b7d220f8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/3001-b7d220f8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/3001-b7d220f8", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/3001-b7d220f8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/3001-b7d220f8'

please help me out to resolve this problem


Hi!

sadzas's picture
On September 22nd, 2008 sadzas (tadpole) said:

I'm sure that your problem is a SOUNDS problem...

-- Executing [8200@from-internal:7] Read("SIP/3001-b7d220f8", "PIN|enter-conf-pin-number||||") in new stack
-- User disconnected

Just after ask the PIN number, the system hang up... That's because doesn't find the initial sound.