6.1.2.46. MeetMeCount (dialplan application)
1. MeetMeCount() - this application gives you information how many participant there are in a conference
NOTE: This application is valid for Asterisk version 1.0.9 and above.
Syntax:
Purpose and usage
This application gives you information how many participant there are in a conference room.
Prerequisites
Asterisk PBX configurations
iax.conf Configurations
sip.conf Configurations
meetme.conf Configurations
In order create conference room we have to configure it in the meetme.conf (look the picture below):
For this tutorial we are going to use meetme room number 1234, it will have also and PIN code for access (1111), and PIN code for the conference administrator (2222). We just follow the standard template:
conf => conference number,PIN,Administrator PIN
Now we have only to add some extensions in the extensions.conf
extensions.conf Configurations
2. Screenshots of what you can see on the CLI of the Asterisk PBX
3. Additional information
4. Uploaded files
5. Similar dialplan applications
NOTE: This application is valid for Asterisk version 1.0.9 and above.
Syntax:
MeetMeCount(<confno>[|<var>])
Purpose and usage
This application gives you information how many participant there are in a conference room.
Prerequisites
To use this application you need a working Asterisk PBX with registered users in iax.conf, sip.conf or mgcp.conf(It depends on which protocol you would like to use) and made extensions.
To see how the application works we recommend to use our IAX softphone Idefisk. You can download it from here. Please also read our tutorial to learn how to configure it to work with Asterisk PBX.
To see how the application works we recommend to use our IAX softphone Idefisk. You can download it from here. Please also read our tutorial to learn how to configure it to work with Asterisk PBX.
Asterisk PBX configurations
NOTE: This is only an example of what for you can use this application. Of course you can use it and for other things.
iax.conf Configurations
We need two registered users in iax.conf file. This is because we are going to use the IAX2 protocol. If you want to use other protocol such as SIP or MGCP, you have to do the configurations below respectively in sip.conf or mgcp.conf.
So, we have registered the users:userA and userB.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
Now lets add and one sip user.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
Now lets add and one sip user.
sip.conf Configurations
We need one registered users in sip.conf file. This is because we want to test and the SIP protocol.
So, we have registered the user sip_user.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.
meetme.conf Configurations
In order create conference room we have to configure it in the meetme.conf (look the picture below):
For this tutorial we are going to use meetme room number 1234, it will have also and PIN code for access (1111), and PIN code for the conference administrator (2222). We just follow the standard template:
conf => conference number,PIN,Administrator PIN
Now we have only to add some extensions in the extensions.conf
extensions.conf Configurations
Now lets take a look at the extensions.conf file.
On the picture above you can see our extensions.conf file.
In the example above we are using the variable users to keep the number of the participant in the conference, and after that to show this value on the CLI. To hear how many users are in conference you have to use MeetMeCount(1234), instead of MeetMeCount(1234|USERS).
Finally, in order to be sure that the Asterisk PBX will hang up the line after the conversation is over, it is a good idea to make an extension for hanging up the line.
[training]
exten => 1234,1,Goto(conf,1)
exten => conf,1,Set(MEETME_RECORDINGFILE=/tmp/Tutorial-${TIMESTAMP})
exten => conf,2,Meetme(1234|sr)
exten => conf,3,Hangup()
exten => 1337,1,Goto(count,1)
exten => count,1,MeetMeCount(1234|USERS)
exten => count,2,NoOp(Total: ${USERS} users)
exten => 1234,1,Goto(conf,1)
exten => conf,1,Set(MEETME_RECORDINGFILE=/tmp/Tutorial-${TIMESTAMP})
exten => conf,2,Meetme(1234|sr)
exten => conf,3,Hangup()
exten => 1337,1,Goto(count,1)
exten => count,1,MeetMeCount(1234|USERS)
exten => count,2,NoOp(Total: ${USERS} users)
In the example above we are using the variable users to keep the number of the participant in the conference, and after that to show this value on the CLI. To hear how many users are in conference you have to use MeetMeCount(1234), instead of MeetMeCount(1234|USERS).
Finally, in order to be sure that the Asterisk PBX will hang up the line after the conversation is over, it is a good idea to make an extension for hanging up the line.
2. Screenshots of what you can see on the CLI of the Asterisk PBX
3. Additional information
For more information about extensions.conf you can check here.
For more information about iax.conf you can check here.
This application is tested with our IAX softphone Idefisk. You can download it from here. For more information about this softphone please read our tutorial.
If you would like to test this application with the SIP channel you can read our tutorials about the SIP Softphones to learn how to configure them to work with Asterisk PBX
For more information about iax.conf you can check here.
This application is tested with our IAX softphone Idefisk. You can download it from here. For more information about this softphone please read our tutorial.
If you would like to test this application with the SIP channel you can read our tutorials about the SIP Softphones to learn how to configure them to work with Asterisk PBX
4. Uploaded files
5. Similar dialplan applications
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