Prepare to install the integrated telephony adaptors

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:

• InterCall telephony adaptor
• MeetingOne NA or EMEA telephony adaptor
• PGi (formerly Premiere Global) NA or EMEA telephony adaptor
• Arkadin telephony adapter
• LoopUp telephony adapter

Information needed when installing

Items marked with an asterisk (*) are required.

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.

Permits the meeting host to dial out.

Permits the presenter to dial out.

Permits participants to dial out.

If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.

The host name or address of the Meeting Exchange server.

The ID of the operator channel used to associate with the Meeting Exchange server.

The Login ID used to establish a connection with the Meeting Exchange Server.

The password used with the Login ID to connect to the Meeting Exchange server.

The FTP directory for audio files on the Bridge.

User name for FTP login.

Password for FTP login.

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.

Planning for deployment

To deploy the InterCall adaptor, certain ports must be available, as shown in the following table:

Information needed when installing

Items marked with an asterisk (*) are required.

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.

Permit the meeting host to dial out.

Permit the presenter to dial out.

Permit participants to dial out.

If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.

URL for the InterCall CCAPI service

URL for the InterCall CCAPI authorization service.

Callback URL used the InterCall service to call back to Adobe Connect. This URL must be publicly accessible.

Value used to identify your connection with the InterCall audio service.

List of country codes for which Adobe Connect displays available conference service numbers.

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.

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.

Permit the meeting host to dial out.

Select this option to permit the presenter to dial out.

Select this option to permit participants to dial out.

If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.

URL for the MeetingOne audio conference API service.

Specifies whether SSH downloading of recordings is enabled.

The ID you use to for the MeetingOne audio conference API service.

Password for the administrator account.

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.

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.

Select this option to permit the meeting host to dial out.

Select this option to permit the presenter to dial out.

Select this option to permit participants to dial out.

If dial out is enabled, select this option to display the "Call Me" dialog box to participants when they join a meeting.

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.

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.

The ID you use when you connect to the PGi audio conference service. This value is provided by PGi. 

The password you use when you connect to the PGi audio conference service. This value is provided by PGi. 

The login used to download audio recordings from the PGi audio conference service.

The password used with the Recording Download Login to retrieve recordings from the PGi audio conference service.

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/.

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

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.

Select this option to permit the meeting host to dial out.

Select this option to permit the presenter to dial out.

Select this option to permit participants to dial out.

Arkadin client application identifier (provided by Arkadin, no default value).

Arkadin server URL (provided by Arkadin, no default value).

Adobe Connect external host name. The default value is https://[external-hostname]/servlet/bamboo/.

Arkadin access number server URL (provided by Arkadin, no default value).

Arkadin authentication server URL (provided by Arkadin, no default value.

LoopUp telephony adapter

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>