The file is located in the /etc/asterisk/ directory.
Meetme is used for conference calls. In this configuration file a conference room that you want to use has to be declared.
This configuration file is read every time you call app meetme() and Asterisk should not be restarted or reloaded in order to see the changes made in this configuration file.
[rooms]
;Context where conference rooms have to be declared.
Syntax is like: conf => confno[,pin][,adminpin].
conf is declaration for conference room. confno is number for the conference room – dial this number to enter the room. pin is optional pin number to authenticate when entering the conference room. adminpin is optional pin for administrating the conference.
Example:
conf => 1357
;This will create conference room 1357 open for everybody who can enter the context in the dialplan it will be accessible.
conf => 1111,9938
;This is conference room 1111 with pasword for authentication at log in 9938. No admin password.
conf => 2222,1821,191871
;This room number 2222 prompts for password on login 1821 and password for administration 191871.
In the dialplan the application looks like MeetMe(confno, [options]) where the available options are one or more of the following (do not use separator if you need more than one option):
'm' -- set monitor only mode (user can only hear the audio, not participate)
'p' -- allow user to exit the conference by pressing '#'
't' -- set talk only mode, user won't be able to hear
'v' -- video mode
'q' -- quiet mode (don't play enter/leave sounds)
'd' -- dynamically add conference
'M' -- enable music on hold when the conference has a single caller
'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}.
Default is conf-background.agi (Zap channels only, does not work with non-Zap channels in the same conference)
Note: In order to have a working MeetMe conference you need a timing source (ztdummy).
User Comments
Arvind (arvindsandilya24 at gmail dot com) 26 August 2009 12:38:20 Hi Friends,
Can you please tell me how to communicate with meetme.conf of another server by SIP trunk.
I want to configure meetme in only listen mode, ok no problem but i want to do this remotely. Means, when we call from xlite which is registered on one server and we should listen conversation of another server.
Is it possible with meetme, If possible. Please reply me at arvind.kumar@buzzworks.com
Thanks in advance.
Thanks & regards
Arvind
feir8510 (feir8510 at 163 dot com) 06 February 2009 03:21:19 if I want sb to talk with more than one people simultaneously(he setup the talk,others only hang),how can i do?
Arvind (arvindsandilya24 at gmail dot com) 28 July 2008 09:02:16 Hello friends,
when we enter into meetme with vicidial, after 1 to 2 hours, noise is coming. When I goto asterisk CLI and operate one command 'show channels', I see that on four to five zap channels, meetme is going on, while all agents are logout. How to hangup focefully meetme from that channels and plz. tell me any solution for this.
srinivas antarvedi (srinivas dot antarvedi at gmail dot com) 14 May 2008 12:13:09 Hello i have following setup
above 3 dtmf's working fine for the overall conference functionality management
The below is created to Mute a user
who entered first in to conference
exten => 4,1,MeetMeAdmin(1234|M|1)
The below is created to UnMute user
who entered first in to conference
exten => 5,1,MeetMeAdmin(1234|m|1)
The below is created to kick out user
who entered first into conference
exten => 6,1,MeetMeAdmin(1234|k|1)
the third argument is the usernumber as they
entered into the confrence
first enter number=1
2nd enter number=2
last enter number =n.
but to kick a specific user using MeetMeAdmin
i need to pass this usernum variable which
i was unable to do as i have only 9 dtmf allowed and can anybody help me out finding how to send this usernumber dynamically ????
thanks in advance
regards
srinivas antarvedi
athan (athan262002 at gmail dot com) 22 August 2007 11:05:42 Hi,
any friendly advice on how to work my meetme... I have meetme.conf on my asterisk but it is not working with this tagged of conf=>12347878 , is there i missed to make it function? I am using the older version of asterisk...please help
robert (hipersenil at hotmail dot com) 09 August 2007 21:48:44 I have a problem.. I had already configured asterisk with meetme and everything was working fine until yesterday when I tried to make a conference but i get no audio... I have my conference configured so it prompts for a password dependeing on the room number and although i get no voice when i dial a conference number it appears to be executing the dialplan.... any ideas why the audio may be failing?... pd: it isn't the soundcarr
robert (hipersenil at hotmail dot com) 09 August 2007 21:47:30 I have a problem.. I had already configured asterisk with meetme and everything was working fine until yesterday when I tried to make a conference but i get no audio... I have my conference configured so it prompts for a password dependeing on the room number and although i get no voice when i dial a conference number it appears to be executing the dialplan.... any ideas why the audio may be failing?... pd: it isn't the soundcarr
bt617 (rzhumagulova at cepr dot org) 08 June 2007 13:31:52 I have created a conference room that functions perfectly. Now, I would like to enable record conference. Is that possible to do on Asterisk 1.2.13?
rupam halder (halderrupam at yahoo dot co dot in) 31 May 2007 10:55:44 can you explain me the DID with sample dial plan
Ravi Sajjan (ipbx dot support at gmail dot com) 26 March 2007 13:35:04 Hi Everyone,
I have configured one Asterisk based Trixbox. Now i want to make conference call acording to call center process.
With the xlite phone i am able to make conference call but if i have one FXS Audio code than ho can i make thirdparty conference for international call.
Can anyone help me out for this.
Thanks
If you have any comments or solution for that. please send me your replay in my mail address: ipbx.support@gmail.com
Thanks & Regards,
Ravi Sajjan
ipbx.support@gmail.com
ravi (ravimalpani2003 at yahoo dot co dot in) 08 February 2007 15:39:03 Hii asterisk users,
i m unable to use conference facility.CLI console shows "No application 'Meetme' for extension (default, conf, 2)" for meet(ext|sr).
i m using asterisk-1.2.10
plz help...
John (bmek at deltamobile dot com) 16 December 2006 19:08:46 The ztdummy module does work, if you are trying to create a conference call, using SIP. Now, I am trying to use the option "w- wait until marked user enters the conference." How do you set a user to be "marked"?
sunkara_voip (ravi dot sunkara at hyperion-tech dot com) 24 October 2006 07:27:24 Hi Users ...
Asterisk is New Revolution With New Technology in Telephony .
Zorro (azzoumarou at hotmail dot com) 30 August 2006 13:44:15 I have no audio Carte, and i want to configure conférence with Asterisk ,but i can't.
Help!!!!!!!!
Zorro (azzoumarou at hotmail dot com) 30 August 2006 13:43:31 I have no audio Carte, and i want to configure conférence with Asterisk ,but i can't.
Help!!!!!!!!
dhiraj (mundadadhiraj at gmail dot com) 16 August 2006 13:04:33 I want to create a conference and allow people to participate and resign from it.
Can you tel me stepwise details of how to do it with asterisk...
I have X-Lite as well as IaxComm softphones...
redcc (red_ex at tom dot com) 18 March 2006 11:06:07 i want to know if the length of meetme number can
exceed 5, eg: if i set up a extension with number 7211111122,
then when i use this device to dial 87211111122, meetme can't
working!!!
thanks in advance!
ogwang charles (cogwang at tech dot mak dot ac dot ug) 04 January 2006 14:53:17 this sounds interesting
steve (sbenavides at unicauca dot edu dot co) 21 November 2005 15:01:49 does meetme work with SIP?
Zowwie (foo at bla dot com) 20 September 2005 23:03:26 Added conf => 888 to meetme_additional.conf and although the system knows that conf room 888 is present, it hangs up on me.