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.

Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.

When i dial the conference

shahab.sabir's picture

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

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

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

Rob


The conference number is not valid

tingwang's picture

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

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.


Resolution to invalid PIN

iptouch's picture

Hi

Ihave the same problem iam tring to use the command but it says

[root@localhost ~]# chown -R asterisk.asterisk /dev/zap
chown: cannot access `/dev/zap': No such file or directory
[root@localhost ~]#

Please help


you don't have any zap

fskrotzki's picture

you don't have any zap devices. it only exists if you have a zap device and driver loaded. You are also following directions from 2 years ago and possibly two versions ago.

[B]Please don't hijack a howto section with question.[\b] We allow postings in the howto's for corrections, updates, etc. Please take your whole problem and post it as a new message in the forum with all the details on your system (versions of things, distro/hand build, who's distro or directions you used to hand build and ANY variations from the published directions and WHY.

Almost nobody looks in the howto sections with the intent of helping answer questions, that's what the forum area is for.


meet-me Feature Suggestions

jacknab's picture

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

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

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.


Conference Management Question

mvabown's picture

Can you have multiple user pins for the same conference? I was hoping to have each person in the conference obtain there own pin for security reasons, but so far i have 1 for admin , and 1 for users. Basically how much control can we have in the conference at the creation of the bridge. Can we give out a pin that as soon as they come online they are muted or does that remain a function for the administrator to take care of once they have joined?

Thank you in advance


How to add Conference call in IVR

amandhally's picture

Hello Sir,

I need to know one thing.
I need announcing conference call on IVR menu.
How can I dedicate a number like 500 for conference call.\

Thanks
aman


Please post questions in the

fskrotzki's picture

Please post questions in the forum as that's where people helping go and look to help. This is the documentation section and comments are intended for corrections and updates to the documentation.

amandhally it's simple, create a conference room 500, then in the IVR make 500 point to it.


Thanks

amandhally's picture

Sorry Sir, I will take care of it.

Thanks


Comment about this "documentation"

Derrick32's picture

Granted this is NOT the place for peoples questions, the forums are.. And its somewhat annoying that people continue to post questions, BUT when i clicked on this i was expecting more details about the module, setting it up, and how to use it. Is there somewhere else i can get that info, rather than posting in a forum "How does Meetme conferencing work? FAQ's about it? Is it usable with the FOP?" you get the point..

i dont mean to sound negative, i love this product and want to learn everything i can about it. Its just a pain to search between freepbx forums, asterisk forums, the occasional trixbox forum, and PIAF without tears to basic info about this stuff..

Thanks again for the awesome product..

D