Dial-in conferencing enables users who want or need to use a public switched telephone network (PSTN) phone to join the audio portion of an on-premises conference. By default, each conference that is enabled for PSTN dial-in includes the following information in the conference invitation:

In addition, the conference invitation might include a numeric-only pass code for the conference that can be used to authenticate anonymous users. The global policy for meetings dictates whether or not pass codes are required for anonymous users.

Dial-in support is available to both enterprise and anonymous users. Enterprise users have Active Directory Domain Services (AD DS) credentials and Office Communications Server accounts. Anonymous users do not have enterprise credentials, but they have the conference pass code. A user in a federated partner’s organization who uses the PSTN to connect to a conference is considered to be an anonymous user in this context.

Enterprise users or conference leaders who join a PSTN conference dial one of the conference access numbers, and then they enter their PSTN conference ID, followed by their unified communications (UC) extensions and PINs. The combination of extension and PIN enable the Front End Server to map enterprise users to their Active Directory credentials. As a result, enterprise users can be authenticated and identified by name in the conference. They can also assume a conference role predefined by the organizer.

Anonymous users who join a PSTN conference dial one of the conference access numbers, and then they enter a PSTN conference ID and a pass code for the conference if the conference requires one. Anonymous users are not identified by name and cannot be assigned a predefined role.

Adding a PSTN User to a Conference

A conference access number maps to the SIP URI of the Conferencing Attendant. When a PSTN user calls a conference access number, the Mediation Server directs the call to the Front End Server, where the Front End Server initiates phone-number lookup to obtain the SIP URI of the allocated Conferencing Attendant. Mediation Server directs the call to the allocated Conferencing Attendant, which accepts the conference ID and passes it in a SIP SERVICE request to the Front End Server.

The Front End Server looks up the conference URI by using the conference ID. The Front End Server then sends the conference URL to the Conferencing Attendant in the response to the SERVICE request. If the user is an enterprise user, Conferencing Attendant uses the user’s UC extension and PIN to send a request to the Front End Server to obtain the user’s identity. The PIN is encrypted. If the user is an anonymous user and a pass code is required, Conferencing Attendant sends a request to the Front End Server to verify the pass code. When user verification is complete, the Conferencing Attendant then uses the conference URI to join the Focus, and then it transfers the call to the A/V Conferencing Server. The A/V Conferencing Server adds the caller to the conference, at which point the initial connection between the Mediation Server and the Conferencing Attendant is dropped.

Adding Conferencing Announcement Service to a Conference

Conferencing Announcement Service plays entry and exit tones to the PSTN participants in an audio conference.

When a PSTN user dials in to a conference, the A/V Conferencing Server determines that the connection is from the PSTN and whether a Conferencing Announcement Service is already active for the conference. If not, the conferencing server requests a new instance of Conferencing Announcement Service and invites it to join the conference. Conferencing Announcement Service plays the entry and exit tone for each PSTN participant. Conferencing Announcement Service also announces to PSTN callers when they have been muted or unmuted.

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