Manually dialing out to a participant from a conference

The Pexip Infinity platform allows you to dial out to participants from an ongoing conference, on an ad hoc basis. When you dial out to a participant in this way, a call is placed to their endpoint from the Virtual Meeting Room or Virtual Auditorium. If they answer the call, they join the conference as either a Host or Guest, depending on the option that you selected. Participants joining the conference in this way do not go through the IVR screen and do not have to enter a PIN.

The participant could also take the form of a dedicated multimedia stream to enterprise CDN (Content Delivery Network) streaming and recording services such as Wowza, Quickchannel, Qumu, VideoTool, Microsoft Stream and Azure Media Services, and to public streaming services such as YouTube, Facebook and Periscope. Any Pexip conference can be streamed as a live event to an unlimited number of viewers, and can automatically be recorded and stored for later consumption. For more information, see Streaming and recording a conference.

Dialing out from a VMR to an externally-hosted conference, such as a Microsoft Teams or Skype for Business meeting, or Google Meet is not supported.

Administrators and conference hosts can add participants to a conference. Administrators can do so via the Administrator interface, and hosts can do so via the Connect app.

You can also configure a Virtual Meeting Room or Virtual Auditorium so that one or more participants are dialed out to automatically whenever a conference starts. See Automatically dialing out to a participant from a conference for more information.

If your environment includes a PSTN gateway or uses an ITSP (Internet telephony service provider), consider the potential for toll fraud if you have Call Routing Rules that can route calls to the PSTN gateway or ITSP, or if you allow conference participants to dial out to other participants via the PSTN gateway or ITSP. See PSTN gateways and toll fraud for more information.

When dialing out from a conference, the outbound call bandwidth limit is inherited from the VMR's bandwidth settings. (If a Call Routing Rule is applied, the rule's bandwidth settings are not used.)

Call Routing Rule requirements and controlling dial-out capabilities

The ability to dial out to a participant from a conference is enabled by default. Call Routing Rules may be required when dialing out from a conference to a new participant, depending on the method used:

  • Using the Connect app or client API: in most cases, you need to configure appropriate Call Routing Rules to enable end-users using the Connect app or the client API to place an outbound call. Rules may not be required for the client API or legacy (Webapp1) clients if the Enable legacy dialout API setting (Platform > Global settings > Connectivity) is selected — see the Connectivity options in global settings for more information.
  • Using administrator tools: when dialing out via the Administrator interface, management API or when using Automatically Dialed Participants (ADPs), you have the option to choose automatic routing (where a rule is required) or to manually specify the call details (where a rule is not required).

This means that when dialing out from an ongoing conference, any calls made via a Connect app (including RTMP calls to a streaming or recording service) always use automatic routing and thus must match an appropriate Call Routing Rule for the call to be placed.

Dialing out via the Administrator interface

You can use the Pexip Infinity Administrator interface to dial out to a participant from a Virtual Meeting Room or Virtual Auditorium. If and when the call is answered, that endpoint will join the conference.

To dial out to a participant using the Administrator interface:

  1. Select the Virtual Meeting Room or Virtual Auditorium to dial the participant from. You can do this in any of the following ways:

    • Go to Services > Virtual Meeting Rooms and select the name of the Virtual Meeting Room.
    • Go to Services > Virtual Auditoriums and select the name of the Virtual Auditorium.
    • If the conference that you want to dial out from is already in progress, go to Status > Conferences and select the required Virtual Meeting Room or Virtual Auditorium.
  2. At the bottom left of the screen, select Dial out to participant.
  3. Complete the following fields:

    Field Description
    System location

    Select the system location from which the call will be placed. If there is more than one Conferencing Node in that location, Pexip Infinity will choose the most appropriate.

    The system location determines which H.323 gatekeeper or SIP proxy to use to route the call.

    Service alias This lists all the aliases that have been configured for the selected Virtual Meeting Room or Virtual Auditorium. The participant will see the incoming call as coming from the selected alias.
    Participant alias The alias of the endpoint/participant that you want to dial.
    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, outgoing location and call control systems etc. as configured for that rule).

    Default: Manually.

    Participant display name

    An optional user-facing display name for this participant, which may be used in participant lists and as the overlaid participant name (if enabled).

    If this name is not specified then the Participant alias is used as the display name instead.


    The signaling protocol to use when dialing the participant. Select either SIP, H.323, or if the endpoint is a Skype for Business / Lync 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.

    Call capability

    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.

    Role Select whether you want the participant to join the conference as a Host or Guest.
    DTMF sequence

    An optional DTMF sequence to be transmitted after the call to the dialed participant starts.

    A DTMF sequence can include: the digits 0-9, "*" (asterisk), "#" (hash) or "," (comma).

    The DTMF tones are 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 are dialing an audio bridge and want to enter conference number 777 followed by #, pause for six seconds and then supply conference PIN 1234 followed by #, you would configure 777#,,,1234# as your DTMF sequence.


    Identifies the dialed participant as a streaming or recording device. When a conference participant is flagged as a streaming/recording participant, it is treated as a receive-only participant and is not included in the video stage layout seen by other participants. See Streaming and recording a conference for more information.

    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.

    Keep conference alive

    Determines whether the conference will continue when all other 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: Yes.

    For streaming participants, we recommend that this option is set to No.

    For more information, see Automatically ending a conference.

  4. Select Dial out to participant.

A message Initiated dial out to participant appears at the top of the screen.

To confirm whether the participant has joined the conference you can go to Status > Conferences, select the conference, and then select the Participants tab. The new participant should appear in the list.

Dialing out via the Connect app

If you have Host privileges, you can use the Connect app to dial out to participants from the conference you are in. A call is placed to the participant and if they answer the call they will join the conference directly (they do not go through an IVR screen or have to enter a PIN).

Automatic routing is used when a Connect app user adds a new participant to a conference. This means that the dialed alias must match an appropriate Call Routing Rule that applies to Outgoing calls from a conference for the call to be placed (using the protocols and call control systems etc. as configured for that rule).

To dial out to a participant using the Connect app, from within a conference:

  1. From the toolbar at the bottom of the screen, select Add participant:

  2. At the prompt, enter the address of the person you want to dial:

  3. Select whether you want the participant to have Host or Guest privileges.
  4. Select Call in.

The call is placed from the conference to the participant and they appear in the participant list with a green line under their name while their endpoint is ringing. If and when the participant answers the call they will join the conference; if they do not answer, or do not accept the call, they will disappear from the participant list.

For legacy clients, there is the option to select a protocol of Automatic (the default), SIP, H.323, Lync/Skype or RTMP. To successfully place calls via the Automatic protocol option, suitable Call Routing Rules must be configured. To enable calls to be placed via the other protocols you must select Enable legacy dialout API (via Platform > Global settings > Connectivity).