You can configure a Virtual Meeting Room or Virtual Auditorium to automatically dial out to one or more participants whenever a conference using that service starts.
You can also manually dial out to participants from a Virtual Meeting Room or Virtual Auditorium, on an ad-hoc basis. For more information, see Manually dialing out to a participant from a conference.
When an Automatically Dialed Participant (ADP) has been added to a Virtual Meeting Room or Virtual Auditorium's configuration, a call is placed from the conference to the ADP's endpoint as follows:
|Meeting type||Participant automatically dialed...|
|No PIN||when the first participant joins the conference|
|Guests not allowed; participants must enter a PIN||when the first participant has entered a valid PIN|
|Guests allowed; Hosts must enter a PIN but Guests do not||when the first participant joins the conference|
|Guests allowed; Hosts and Guests must enter a PIN||when the first participant has entered a valid PIN|
The role of the first participant does not matter - both Hosts and Guests can trigger a call to an ADP.
A call is not placed to an ADP if that participant has already joined the conference.
When the call is answered, the Automatically Dialed Participant will join the conference as either a Host or Guest, depending on which role you selected when configuring them as an ADP. They will not be required to enter a PIN even if one is required for that role.
Usually, the call to the ADP is placed from the same Conferencing Node that is being used by the participant who initiated the conference.
However, you can optionally configure the call to be placed from a Conferencing Node in a specific location. The behavior varies according to whether it is a manually routed ADP (where Route this call is set to Manually) or the ADP is called according to Call Routing Rules (where Route this call is set to Automatically).
- Manually routed ADPs: the nominated location is the location of the node that will dial the ADP.
Automatically routed ADPs: the nominated location is the notional source location used when considering if a Call Routing Rule applies or not — but the rules themselves will determine the location of the node that dials the ADP. In this case, the configured Outgoing location for the ADP is deemed to be the location handling the call (when matched against the Calls being handled in location setting for the Call Routing Rule) and the rule's Outgoing location setting is used to determine the actual location of the node that dials the ADP.
This behavior is useful if, for example, you have configured lowest-cost-routing rules that only allow calls to be placed from "internal" Conferencing Nodes but you have a conference that is triggered by a call to an "external" Conferencing Node. In this case you still want the call to the ADP to be initiated, but you don't want, in general, to allow calls to arbitrary ISDN destinations to be placed from Conferencing Nodes in the external location.
When the endpoint is called, the Automatically Dialed Participant will see the call as coming from one of the Virtual Meeting Room or Virtual Auditorium's aliases. This means that if the participant misses the call, they can easily return it by dialing the alias that appears on their endpoint.
When deciding which alias to use to identify the call, Pexip Infinity will select the first in the list that is valid for the selected call protocol. So if your Virtual Meeting Room has two aliases, the first meet.sales and the second email@example.com, when you place a call to an ADP over SIP it will show as coming from meet.sales because that is the first valid SIP address in the list. However, endpoints may not always be able to return a call to this alias because it is missing the domain. Therefore, if you want ADPs to be able to return calls to Virtual Meeting Rooms and Virtual Auditoriums, we recommend that you list the most routable aliases first when assigning them to these services. Note that you can also use external or local policy (see policy profiles) to specify the source alias for calls to ADPs.
Automatically Dialed Participants with a role of Host may or may not prevent a conference from being automatically terminated, depending on their Keep conference alive setting and whether any other ADPs remain in the conference. For more information, see Automatically ending a conference.
If you have a large number of ADPs to configure, you can import them using a CSV file. For more information, see Bulk import/export of Automatically dialed participants (ADPs).
To configure a Virtual Meeting Room or Virtual Auditorium to automatically dial out to a participant when a conference starts:
- Go to .
You will be taken to thepage.
Complete the following fields:
Field Description Participant alias The alias of the participant to dial when a conference starts. If you select a Protocol of SIP, this must be a valid SIP alias. Creation time
This read-only field shows the date and time at which this record was first configured.
Participant display name
An optional user-facing display name for this participant, which may be used in participant lists and as the overlaid speaker name (if enabled).
If this name is not specified then the Participant alias is used as the display name instead.
An optional description of the Automatically Dialed Participant.
Route this call
Select how to route the call:
- Manually: uses the requested Protocol and the defaults for the specified System location.
- Automatically: routes the call according to the configured Call Routing Rules— this means that the dialed alias must match an outgoing Call Routing Rule for the call to be placed (using the protocols and call control systems etc. as configured for that rule).
The signaling protocol to use when dialing the participant. Select either SIP, H.323, or if the endpoint is a Lync / Skype for Business client, select Lync / Skype for Business (MS-SIP). The RTMP protocol is typically used when adding a streaming participant. Note that if the call is to a registered device, Pexip Infinity will instead use the protocol that the device used to make the registration.
This field only applies when Route this call is set to Manually.
An optional DTMF sequence to be transmitted after the call to the Automatically Dialed Participant starts.
A DTMF sequence can include:
- the digits 0-9
- * (asterisk)
- # (hash)
- , (comma).
The DTMF tones will be sent 3 seconds after the call connects, one at a time, every 0.5 seconds. A comma is a special digit that represents a 2 second pause (multiple commas can be used if a longer pause is needed).
For example, if you need your Automatically Dialed Participant to dial an audio bridge and then enter conference number 666 followed by #, pause for six seconds and then supply conference PIN 1234 followed by #, you would configure 666#,,,1234# as your DTMF sequence.
The level of privileges the participant will have in the conference. For more information, see About PINs, Hosts and Guests.
Conference Select the names of the Virtual Meeting Rooms and Virtual Auditoriums from which this participant will be dialed automatically whenever a conference using that service starts. Outgoing location
For Manually routed ADPs, this is the location of the Conferencing Node from which the call to the ADP will be initiated.
For Automatically routed ADPs, this is the notional source location used when considering if a routing rule applies or not - however the routing rule itself determines the location of the node that dials the ADP. For more information, see Choosing the calling location.
To allow Pexip Infinity to automatically select the Conferencing Node to use to place the outgoing call, select Automatic.
As with all calls, signaling and media may be handled by different Conferencing Nodes in that location.
Identifies the dialed participant as a streaming or recording device. See Streaming a conference over YouTube for more information.
Keep conference alive
Determines whether the conference will continue when all other non-ADP participants have disconnected.
Yes: the conference will continue to run until this participant has disconnected (applies to Hosts only).
If multiple: the conference will continue to run as long as there are two or more If multiple participants and at least one of them is a Host.
No: the conference will be terminated automatically if this is the only remaining participant.
Default: If multiple.
For streaming participants, we recommend that this option is set to No.
For more information, see Automatically ending a conference.
Allows you to limit the media content of the call. The participant being called will not be able to escalate beyond the selected capability. For more information, see Controlling media capability.
Default: Main video + presentation.
This field only applies when Route this call is set to Manually.
This field does not apply to RTMP calls.
Dual stream (presentation) URL
When adding a dual streaming RTMP participant, this specifies the RTMP URL for the second (presentation) stream.
Leave this field blank when adding a single streaming participant.