Know how to install the telephony adaptors that integrate with Adobe Connect to provide audio conferencing capabilities.
Integrated telephony adaptors provide communication between Adobe Connect and specific audio conferencing providers. Integrated adaptors have advanced call capabilities, allowing hosts and presenters to control the audio conference from the meeting.
To install integrated telephony adaptors, run the Adobe Connect installer.
Each adaptor requires you to supply specific pieces of information when you install it. For more information, see:
You can enable multiple audio bridges for Adobe Connect Server. Meeting hosts choose which audio bridge to use when they create a meeting in Adobe Connect Central. Each meeting can have only one audio bridge.
Information needed when installing
Items marked with an asterisk (*) are required.
Enable Dial Out
Enables a system-wide dial out. If you don’t select this option, any selections you make for the following four entries are ignored. If you do select this option, use the following four entries to specify how dial out is implemented.
Enable Dial Out for Host
Permits the meeting host to dial out.
Enable Dial Out for Presenter
Permits the presenter to dial out.
Enable Dial Out for Participant
Permits participants to dial out.
Enable “Call Me” Dialog
If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.
Meeting Exchange Host Name*
The host name or address of the Meeting Exchange server.
Phone Operator ID*
The ID of the operator channel used to associate with the Meeting Exchange server.
Login ID*
The Login ID used to establish a connection with the Meeting Exchange Server.
Password*
The password used with the Login ID to connect to the Meeting Exchange server.
FTP Directory*
The FTP directory for audio files on the Bridge.
FTP Login*
User name for FTP login.
FTP Password*
Password for FTP login.
Meeting Exchange Dial In Number*
A valid phone number dialed by Adobe Connect to reach the Meeting Exchange server.
InterCall telephony adaptor
The InterCall telephony adaptor allows meeting hosts, presenters, and participants to control audio conference features from Adobe Connect meeting rooms. This adaptor requires a VoIP or SIP provider and Adobe Media Gateway (Universal Voice) for recording meetings. Complete the following workflow to enable the telephony adaptor.
InterCall recommends using API instead of CCAPI.
Planning for deployment
To deploy the InterCall adaptor, certain ports must be available, as shown in the following table:
Port |
Description |
|---|---|
80 |
InterCall uses port 80 to communicate with Adobe Connect over HTTP. This port must be open for incoming communication, to be able to receive callbacks from InterCall to Adobe Connect. |
443 |
InterCall uses port 443 to communicate with Adobe Connect over HTTPS (SSL). This port must be open for incoming communication, to be able to receive callbacks from InterCall to Adobe Connect. |
8443 |
Adobe Connect uses port 8443 to communicate with InterCall over HTTPS (SSL). Adobe Connect uses this port for CCAPI/API and authorization services. This port must be open so that outgoing messages can be sent from Adobe Connect to InterCall. |
9080 |
As mentioned earlier, this port is required for telephony in general. For InterCall, however, open this port on the firewall for each node in a cluster. |
Information needed when installing
Items marked with an asterisk (*) are required.
Enable Dial Out
Enables a system-wide dial out. If you don’t select this option, any selections you make for the following four entries are ignored. If you do select this option, use the following four entries to specify how dial out is implemented.
Enable Dial Out for Host
Permit the meeting host to dial out.
Enable Dial Out for Presenter
Permit the presenter to dial out.
Enable Dial Out for Participant
Permit participants to dial out.
Enable “Call Me” Dialog
If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.
CCAPI Host*
URL for the InterCall CCAPI service
CCAPI Auth Host*
URL for the InterCall CCAPI authorization service.
Client Callback URL*
Callback URL used the InterCall service to call back to Adobe Connect. This URL must be publicly accessible.
Application Token*
Value used to identify your connection with the InterCall audio service.
Country Codes*
List of country codes for which Adobe Connect displays available conference service numbers.
Toll Free Number Country Code
The country code whose conference number is toll-free; for example, US.
MeetingOne NA or EMEA telephony adaptor
The MeetingOne telephony adaptor allows meeting hosts, presenters, and participants to control audio conference features from Adobe Connect meeting rooms.
Information needed when installing
Items marked with an asterisk (*) are required.
Enable Dial Out
Select this option to enable system-wide dial out. If you don’t select this option, any selections you make for the following four entries are ignored. If you do select this option, use the following four entries to specify how dial out is implemented.
Enable Dial Out for Host
Permit the meeting host to dial out.
Enable Dial Out for Presenter
Select this option to permit the presenter to dial out.
Enable Dial Out for Participant
Select this option to permit participants to dial out.
Enable “Call Me” Dialog
If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.
MeetingOne API URL*
URL for the MeetingOne audio conference API service.
SSH
Specifies whether SSH downloading of recordings is enabled.
Telephony API Server Login*
The ID you use to for the MeetingOne audio conference API service.
Telephony API Server Password*
Password for the administrator account.
Confirm Password
Retype the password for the MeetingPlace administrator account.
PGi (formerly Premiere Global) NA telephony adaptor
The PGi telephony adaptor allows meeting hosts, presenters, and participants to control audio conference features from Adobe Connect meeting rooms. This information applies to both the PGi NA adaptors.
Information needed when installing
Items marked with an asterisk (*) are required.
Enable Dial Out
Select this option to enable system-wide dial out. If you don’t select this option, any selections you make for the following four entries are ignored. If you do select this option, use the following four entries to specify how dial out is implemented.
Enable Dial Out for Host
Select this option to permit the meeting host to dial out.
Enable Dial Out for Presenter
Select this option to permit the presenter to dial out.
Enable Dial Out for Participant
Select this option to permit participants to dial out.
Enable “Call Me” Dialog
If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.
PGi Hostname*
The host name or IP address of the PGi audio conference service. This value is provided by PGi. For example, for PGi NA, this value can be csaxis.premconf.com.
PGi Port Number*
The port number that Adobe Connect uses to connect to the PGi audio conference service. This value is provided by PGi and usually it is 443.
PGi Web ID*
The ID you use when you connect to the PGi audio conference service. This value is provided by PGi.
PGi Password*
The password you use when you connect to the PGi audio conference service. This value is provided by PGi.
Recording Download Login*
The login used to download audio recordings from the PGi audio conference service.
Download Password*
The password used with the Recording Download Login to retrieve recordings from the PGi audio conference service.
Download URL
The URL that Adobe Connect uses to download recordings from the PGi audio conference service. The default value for PGi NA is https://ww5.premconf.com/audio/.
Country Code for Universal Line Number*
The corresponding country code for a Universal line number.
Arkadin telephony adapter
The Arkadin telephony adaptor allows meeting hosts, presenters, and participants to control audio conference features from Adobe Connect meeting rooms. This adaptor requires a VoIP or SIP provider and Adobe Media Gateway (Universal Voice) for recording meetings. To enable the telephony adaptor, complete the following workflow.
Planning for deployment
For Arkadin telephony adapter to work, you must get the host IP address of your Adobe Connect Server running telephony services, whitelisted on the firewall of Arkadin. Since the IP address is allowed on Arkadin’s firewall, the Adobe Connect Server hosting Arkadin services, must have a public IP address.
For more information, see http://www.arkadin.com/.
Information needed when installing
Enable Dial Out
Select this option to enable system-wide dial out. If you don’t select this option, any selections you make for the following four entries are ignored. If you select this option, use the following four entries to specify how dial out is implemented.
Enable Dial Out for Host
Select this option to permit the meeting host to dial out.
Enable Dial Out for Presenter
Select this option to permit the presenter to dial out.
Enable Dial Out for Participant
Select this option to permit participants to dial out.
Arkadin client application identifier*
Arkadin client application identifier (provided by Arkadin, no default value).
Arkadin server URL*
Arkadin server URL (provided by Arkadin, no default value).
Adobe Connect external hostname*
Adobe Connect external host name. The default value is https://[external-hostname]/servlet/bamboo/.
Arkadin Access Number server URL*
Arkadin access number server URL (provided by Arkadin, no default value).
Arkadin authentication server URL*
Arkadin authentication server URL (provided by Arkadin, no default value.
The LoopUp telephony adaptor allows meeting hosts, presenters, and participants to control audio conference features from Adobe Connect meeting rooms. Configure the adaptor using sample files attached here.
To enable or disable dialout feature for host, presenter or participant, toggle below settings in telephony-capabilities file:
<capability id="dial-out" enabled="true">
<host enabled="true" />
<presenter enabled="true" />
<participant enabled="true" />
</capability>
This is to avoid any extra information to be shared with adaptor endpoint unless needed specifically. To suppress extra logging on server and keep only local logs, add the below setting in telephony-settings file:
<setting id="LogSuppressAllByContract">true</setting>