rPhone 8 User Guide


Version 1.0, Revised August 2011

DIALING IN TO YOUR rPHONE/ZipDX CONFERENCE

 

HOST INSTRUCTIONS
Dial 1-888-947-3988

If ZipDX does not recognize your phone number please enter your 7-Digit PIN
Press “1” to start a Conference and then enter your 6-Digit Conference Code
(You may also start the Conference using the rPhone pod in which case you will enter the meeting room automatically as soon as you enter your PIN)

ATTENDEE INSTRUCTIONS
Dial 1-888-947-3989

Enter the 6-Digit Conference Code

Attendees outside of North America also have the option of dialing in
using one of over 25 International Local Dial-In Numbers
(Note: Callers are responsible for any local charges to dial these numbers)

You may prefer to have meeting attendees dial out to themselves instead. In this case, do not publish the dial-in information prior to the event but allow users to dial the phone number where they can be reached, directly in the rPhone pod once they have entered your meeting room*.

*ZipDX charges a single flat rate fee per user per minute which allows you to dial out to any landline in the following countries: USA, Canada, China, Denmark, France, Germany, Hungary, Ireland, Israel, Italy, Malaysia, Netherlands, New Zealand, Norway, Singapore, South Korea, Spain, Sweden, Switzerland, Taiwan, Thailand, United Kingdom. Cellphones in the following countries are included at the same flat rate: USA, Canada, China, Hong Kong, India, Singapore, South Korea, Thailand. Rates to landlines and cellphones in other destinations may incur a surcharge based on the carrier. ZipDX Admins can cap the surcharge rate they are willing to pay based on the countries they wish to access. Dial in calls to the Toll-Free number or our local country dial-in numbers never incur surcharges.

 

WHAT IS THE rPHONE?

The rPhoneTM is an Adobe ConnectTM add-in pod for organizations that utilize telephony in ConnectTM Meeting Rooms. It is an audio bridge that fully integrates with Adobe Connect’s Universal Voice feature (available in Adobe Connect 7.5TM) to broadcast PTSN, mobile, and SIP-based audio communication over the computer speakers of attendees in a meeting room.

FEATURES

The rPhone’s Features include:

  • Runs as a simple add-in pod in any Adobe Connect meeting (hosted or licensed)
  • The attractive, intuitive interface mimics the Adobe Attendee Pod
  • Toll-Free Dial In for North America and local dial in numbers in over 25 Countries worldwide
  • Users and Hosts can dial-out to any phone in any country
  • Supports wideband audio for better sounding live audio and higher quality recordings
  • Dial Out to SIP Based connections such as TandbergTM, PolycomTM or SIP-based soft phones
  • Dial strings can include Extension Numbers, pauses & special characters to navigate IVR systems
  • No hidden fees! One low, flat, all-in (taxes included) rate for most land lines worldwide*
  • No monthly minimum or annual commitment required
  • We’ll match any documented Toll-Free conference rate on an all-in (taxes included) basis
  • Cap surcharges on outbound calls to cell phones & countries not included in the flat rate
  • Real-time, on-demand transcription services (identified by speaker) inside of Adobe Connect
  • Supports sub-conferencing for audio breakouts.
  • Supports synchronization with breakouts in Connect!
  • Two-Way Universal Voice support.
  • Support for real-time, multi-language translation of live events (client supplies translators)
  • Instantaneous availability for Adobe Connect recordings – no need to wait hours anymore!
  • Auto-merging of web and audio connections – see who’s speaking in the meeting room
  • Hosts can control the volume of any user inside of Adobe Connect
  • Hosts can mute one or multiple lines – support for Soft Mute, Hard Mute and Call On features
  • Rebroadcast conference audio over user’s computer speakers – zero cost for passive attendees
  • Configure your conference to automatically dial out to attendees before the meeting starts
  • Hosts can dial out to any number worldwide – all users have to do is answer the phone!

* A single flat rate fee allows you to dial out to any landline in the following countries: USA, Canada, China, Denmark, France, Germany, Hungary, Ireland, Israel, Italy, Malaysia, Netherlands, New Zealand, Norway, Singapore, South Korea, Spain, Sweden, Switzerland, Taiwan, Thailand, United Kingdom. Cellphones in the following countries are included at the same flat rate: USA, Canada, China, Hong Kong, India, Singapore, South Korea, Thailand. Rates to landlines and cellphones in other destinations may incur a surcharge based on the carrier. Admins can cap the surcharge rate they are willing to pay based on the countries they wish to access. Dial in calls to the Toll-Free number or our local country dial-in numbers never incur surcharges.

GETTING STARTED

Congratulations on downloading the rPhone for Adobe Connect 8. This revolutionary add-in pod will enhance your meetings and reduce your telephony costs. This document will get you started. Time to complete: 5-10 Minutes.

The rPhone is a telephony integration created through a partnership between Refined DataTM Solutions and ZipDXTM. It works inside Adobe Connect Meeting Rooms and takes maximum advantage of the Universal Voice features in Connect Pro 8. You can also use your ZipDX account to conduct Phone Conference Meetings independently of Adobe Connect.

STEP 1 – OPENING A NEW ZipDXTM CONFERENCING ACCOUNT

Please contact Refined Data Solutions at 1-877-643-6439 to set up a new rPhone account with an initial credit balance for your trial or sign up directly at ZipDX.com. You will receive a welcome email from ZipDX similar to the one below. Just click the link to activate your new account.

 

STEP 2 – CREATE YOUR PERSONAL ZipDX PROFILE

Create your personal rPhone /ZipDX profile by entering your name and choosing a unique 7-Digit PIN, which will identify you when starting your conference calls from a phone. You can use the temporary PIN assigned to you in your email (see above) or you can pick something that you will find easier to remember (e.g. the last seven digits of your parent’s phone number).

Your PIN acts as your password and should be treated with care. Anyone who knows your PIN can start meetings as if they were you and the resulting costs will be charged to your account.
Treat your PIN as you would your Credit Card Number.

For greater security, we recommend that you provide an optional alpha-numeric web password which will be used for logging in to your account online. Your 7-Digit PIN will continue to be used when starting conferences from your phone.


The Advanced Preferences section can safely be ignored as you can set any of the options available there at a later time.

Just click on the Next button when you are ready to continue.

STEP 3 – PROVIDE THE PHONE NUMBERS YOU REGULARLY USE (OPTIONAL)

One of the great things about ZipDX is that it uses Caller-ID to automatically recognize you when you dial in. This way you won’t even need to use your PIN when calling from your cell phone, your home or your direct dial number at the office. It’s a great time-saver.

Add all of the phone numbers that you frequently call in from, on the screen below. These can include your cell phone, home phone and direct-dial office phone if you are the only person likely to dial in to a conference from them.

Note: if your organization uses a central switchboard number with extensions for each employee, then your office number does not uniquely identify you and should not be used.


The Advanced Preferences on the above screen are optional. When you have completed this process, just click the “Save” button.

STEP 4 – SET YOUR CONFERENCE PREFERENCES

Click the “Home” tab and then the blue “Meeting” link under “Conference Templates”
This is where you can customize your Meeting Conference Code and the options for your spontaneous conference calls. If you are happy with the Conference Code assigned to you automatically by the system (indicated by the arrow below) you can skip this step and use the default conference settings, which work for most people.


Clicking on the Meeting link will take you to the default meeting template, which you can configure for optimum use with the Universal Voice features of Adobe Connect.

Simply choose a 6-Digit Unit Conference Code if you wish to change the Code that was chosen for you automatically by the system (shown by the blue arrow above).

This is not your personal PIN but is the Code you will give out to users who would like to dial in to your events.

This Code will be used by Adobe Connect to dial in to the conference via the Universal Voice feature in Adobe Connect 8.

You have a number of choices that are configurable based on your personal preferences. E.g. should users hear Chimes and have their names announced as they enter your events. These are mostly a matter of personal preference.

To learn more about all of the features of your ZipDX account you can view the following links:
ZipDX Users Guide ZipDX Deployment Guide ZipDX FAQ


Note: The rPhone pod will start your Spontaneous Conference automatically for you. Just decide if callers should be able to chat amongst themselves before you arrive (we recommend that they hear music until you are on the call) and how long they should be able to talk after you leave if you don’t explicitly terminate the conference.

From the Accounts tab you can add additional users to your account, each with their own unique Conference Code.

STEP 5 – CONFIGURING A NEW ADOBE CONNECT AUDIO PROVIDER

This is the final part of the setup process and assumes that you are using Connect 8 with Universal Voice.

Once you have set up Conference Codes for yourself and optionally for your other ACP Admins, Named Hosts and Seminar Hosts in your organization, you need to provide this information to your Adobe Connect Account.

Setting up a new Audio Provider in Connect 8 can be quite a daunting task if you have to do it from scratch or if you have a large number of Meeting Hosts. Thankfully, the entire process has been completely automated by the nice folks at Refined Data Solutions.

Just use the Automated rPhone Configuration Utility on the Refined Data Site – the process takes about 2-3 minutes.

The Configuration Utility can be accessed at http://refineddata.com/products/rphone/configure.php
To use the tool you will need:
The URL of your Adobe Connect 8 Account (i.e. http://admin.na5.acrobat.com)
The Account ID of your ACP account (this value appears as part of the URL whenever you log in)
An Admin Username and Password on your Adobe Connect Account.
The Name you would like to assign to the new Audio Provider (just use the default value in most cases)

Just press the “Next” button to continue.

Please Note: You can use the rPhone Configuration Utility as often as you would like to assign conference codes to new hosts on your account. Please ensure that you use the same Audio Provider Name each time you use the tool.

STEP 6 – ASSIGN CONFERENCE CODES TO YOUR MEETING AND SEMINAR HOSTS

Each Admin, Named Host or Seminar Host on the system should have their own login and Conference Code on ZipDX. You should strongly discourage users from sharing a single Conference Code as only one call can ever be in progress at one time using a single Conference Code. Sharing Conference Codes among users can create scheduling problems and invites abuse of the system – it becomes impossible to know who is responsible for the charges incurred.

Remember, you can easily create additional logins under your ZipDX account at any time just by clicking on the “Accounts” tab in ZipDX and entering the names of all of your users as Organizers. They will receive an invitation, exactly like the one you received where they can configure their own conference as you did in Steps 1-4 above.

Once you know the Conference Codes for one or more of your users, simply log in to the Configuration Utility (described in Step 5 above).

The system will show you a list of all users on the account to which you can assign a Conference Code.

Enter one or more Conference Codes and click “Create” to add the rPhone Audio Profile to your Host Accounts

Note: Be sure to enter the Conference Codes assigned to each user in their ZipDX Meeting Template in Step 4.

You may include spaces in the Conference Code to aid in readability e.g. “544 278” or “54 42 78.”

That’s all there is to it. You may return as often as you would like to add new codes for other Meeting Hosts.

Once this step has been completed, your Admins and Hosts can select their new Audio Profile when creating any new meeting in Adobe Connect. This is extremely important if you wish to record your meetings and need to include the phone audio or if you would like to give users the choice of listening on their computer speakers.

 

rPHONE INTERFACE

 

 

ADDING THE rPHONE POD TO YOUR MEETINGS

The rPhone Audio Bridge for Adobe Connect is supplied as a SWF file contained in a zip file plus this documentation.

There are three ways to include the rPhone in your meetings:

  1. Upload the rPhone pod in a meeting as a regular “Share”
  2. Upload the rPhone to your Content Library so it can always be accessed from there – “Share” from Library
  3. Add the rPhone to your Meeting Templates so it will be included automatically in any new event you create.


We recommend using both options 2 & 3 for maximum flexibility but the choice is yours.

To install the rPhone, download the most recent version of the pod from Refined Data’s website.  Browse to the ‘All Pods Download’ section of the website and select rPhone from the Connect 8 downloads section. Once the file has been saved to your computer, open an Adobe Connect Meeting room.

Then:

  1. Select ‘Pods’ in the Adobe Connect menu
  2. Select ‘Share’
  3. Select ‘Add New Share’
  4. Click the dropdown next to the ‘Share My Screen’ and select ‘Share Document’
  5. Select ‘Browse My Computer’
  6. Locate and select the rPhone file (SWF or ZIP format)
  7. Once the pod has loaded and installed, click to activate it
  8. Complete the form that opens in your browser (ensure your pop-up blocker is not activated)

The first time you load the rPhone, you will be asked to register with Refined Data Solutions to create an account with us. This is similar to creating an iTunes App Store Account. You will not be asked to do this again for your account.

Once complete the pod is ready for use! (Note: this assumes you have already set up your ZipDX account.)

For convenience, when you load the rPhone into a Connect meeting room for the first time, a prompt will ask if you would automatically like to load it onto all layouts.

Please do not load multiple instances of the rPhone into the same meeting room as this can cause problems. If you inadvertently load more than one instance of the pod in a room, just use Pods > Organize Pods > Delete to remove unwanted copies.

As soon as the rPhone pod is loaded in a meeting room it will automatically load the sign-in screen presented below.

Fill in the three empty fields with the values you used when setting up your ZipDX account.

You can use either your 7-Digit PIN or your ZipDX Web Password (case sensitive) in the password field, depending on your personal preferences. This information needs to be entered in every meeting room where you are using the pod the first time. If you reuse your meeting room, the rPhone pod will remember your credentials between meetings.

Please Note: Do not share your ZipDX User Name and Password with others as they will be able to use the system posing as you, and you are completely responsible for any and all charges incurred on your account as a result of abuse.

To activate Universal Voice, you must do the following:

  1. Click ‘Audio’ from the Adobe Connect Menu
  2. Select ‘Start Meeting Audio’
  3. Ensure the ‘Using Phone’ and ‘Start Broadcasting Telephony Audio’ buttons are checked.

 

BROADCASTING AUDIO

The ‘Start Broadcasting Telephony Audio’ button begins the transmission of the phone audio over the computer speakers of all attendees in the meeting room, including the Host that initiated the phone conference.

(Note: While you can run audio conferences without initiating this connection, you will not be able to re-broadcast phone audio over attendee’s speakers unless you complete this step.)

As the Host that initiated the conference, your conference audio will automatically be muted so that you are not distracted by hearing the audio from the phone conference coming over your speakers. (After all, you are already on the call.) Other Hosts will need to do this manually, however. To mute the phone audio you hear over your computer speakers, select the green ‘speaker’ icon from the Connect menu, and select ‘Mute Conference Audio Only’ as per the image below.

 

Generally speaking, only a handful of people will need to manually mute the Conference Audio (most often, the lecturers) as the majority of your users will likely elect to listen via VOIP.

Alternatively, you may only want to use the broadcast audio capability to instruct attendees on how to connect (i.e. via VOIP or phone) after which you can disable the audio broadcast by selecting ‘Audio’ from the Connect menu, and clicking ‘Stop Audio Broadcast’, as per the image below.

 

(Note: The rPhone supports two-way Universal Voice, new to Connect 8. This means that users that are using Connect’s VOIP can be heard by phone-only users, and vice versa. This is discussed in greater length later on.)

We recommend double-clicking on the pod title bar to change the pod name to something more descriptive such as ‘rPhone Audio Conference Bridge’. (You can use this trick on any Adobe Connect pod.)

 

CONFERENCE CALL OPTIONS

As soon as an Attendee enters a meeting room in which the rPhone has been loaded and a conference has been started, all users will automatically be presented with the following dialing options:  

 

I’m already on the call

This option is for users who are already on the call. In most cases we cannot determine which inbound phone call is associated with which meeting attendee so we ask the users to identify themselves by pressing a short sequence of keys on their phone’s dial pad to help us sort out who is who. If you press the ‘I’m already on the call’ button you will see the following window:

 

Listen

If you elect to listen through your computer, or to participate in the phone conference via Connect’s VOIP, select this option.

Close

This option will return the user to the main view of the rPhone pod.

Dial in to the conference

Users who choose this option will be presented with conference dial-in numbers and codes. Use the tabs on the window that opens (image below) to find the appropriate dial in number. You will notice that there’s also a SIP dial-in option.

 

Dial out to me

Users who choose this option can enter their phone number and the system will dial them directly. The rPhone supports dialing extension numbers (preceded by “x” or using the special button on the rPhone dial pad), entering pauses (space, comma or “p”) and other special characters (#*).


To return to the dialing options menu at any time, click of the ‘Connection Options’ icon at the bottom right of the rPhone pod as highlighted in the image below.

 

TWO WAY UNIVERSAL VOICE

The rPhone and ZipDX support two-way Universal Voice integration, a feature new to Connect 8. While in previous versions of Adobe Connect, users on the phone could be heard by those strictly using VOIP, the reverse was not possible: namely, that attendees using VOIP could speak through their headsets and have the audio transmitted back over the phone conference and listened to by phone-only users.

This is now possible thanks for Connect 8.

To activate two-way Universal Voice, after starting the audio by clicking ‘Audio’ -> ‘Start Meeting Audio’ in meeting room’s top menu, the pop-up below will appear. Ensure that you have all options enabled as presented in the image below.


Phone and VOIP-only users will now be able to communicate with one another using their preferred means.

POD OPTIONS BUTTON

The rPhone’s ‘Pod Options’ settings can be accessed through the ‘Pod Options’ button located in the bottom right portion of the pod as per the graphic below.

Once accessed, three tabs will open up.

USERS
Participants Can See

Determine whom Participants are able to see in the rPhone’s attendee list – everyone or just Hosts/Presenters. This option is ideal for one-to-many lectures, or situations in which it is best if Participants remain unaware of whom the other attendees are.

Lecture Mode

When activated, all Participants are muted until called upon by a Host or Presenter. A prompt indicates to the person that they have been called upon, providing notification that his or her attention is sought. The audio broadcast continues to work, as well. Ideal for one-to-many broadcasts with many attendees as this feature allows you to instantly mute all non-Hosts with a single operation.

Enable List Sorting For

Determine who gets access to the rPhone’s ‘Sort’ button that is located on the pod’s main interface.

 

RECORDING
Record Meeting

This feature creates an audio-only recording of the meeting that is stored on ZipDX’s dashboard. This feature is not a substitute for Connect’s built-in recording mechanism that records both the audio (assuming UV has been set up) and the video.

Transcribe Meeting

Coming soon, the rPhone for Connect 8 will feature a transcription functionality that will transcribe the phone conversation into text, live, in a Connect meeting room. This toggle turns that feature on and off.

 

Enable Transcription For

Determines to whom the transcription feature is visible. When set to ‘Hosts & Presenters’ and ‘None’, the transcription button and pod become hidden to Participants and all attendees, respectively.

ROOM
Lock Conference

When activated, the audio conference is locked and new participants are barred entry.

Play Entrance/Exit Chimes

When activated, chimes are heard when individuals enter and leave the audio conference.

Announce Participant Names

When activated, the names of all newcomers are announced if previously configured through the ZipDX website. This feature will only work for registered users of ZipDX.

Renaming Sub Conference Rooms

Double click on the name of any sub conference room to rename it.

 

VOLUME CONTROL

The rPhone enables hosts to adjust the volume of phone-only users relative to one another, to mute individual or all users, and to give the floor to a specific user by ‘calling’ upon them.

To adjust the volume of a user, highlight his/her name in the rPhone pod, and select the ‘Audio Options’ button.


When complete, the volume bar will open up that permits the host to adjust the volume of users relative to one another, and to hard mute or soft mute the user.

 

Soft Mute

In this mode, the user’s microphone is muted and a prompt is heard by the user, indicating that she is soft-muted and that she can un-mute herself by pressing *6 on her phone’s keypad.

Hard Mute

In this mode, the user’s microphone is muted and a prompt is heading by the user, indicating that he has been hard-muted. Only a host can un-mute a user whose microphone has been hard-muted.

Lecture Mode

When activated (through Pod Options -> Users), all Participants on the phone conference are automatically hard-muted. They may individually be un-muted by the Host as described above, or when ‘called upon’ using the ‘Call On User’ button. Lecture mode is ideal for one-to-many broadcasts.

'Call On User' Toggle

The ‘Call On User’ toggle, when activated, prompts a user that their attention is sought by the host. If the user is muted when called upon, they will automatically be un-muted so that they are able to speak. If the Host proceeds to give another user the floor, the prior user is then automatically muted. If the user is un-muted prior to being called upon, he or she will automatically be muted after another user has been called upon. This renders this option most appropriate for use in Lecture Mode.

To call on a user, highlight their name and select the ‘Call On User’ button as per the image below.

 

DISCONNECTING USERS & REMOTE DIAL-IN

Using the rPhone interface, Hosts can manually dial out to individuals (used, for example, in instances in which the input of a person not in the meeting is required) or to disconnect users (if someone has forgotten to hand up, for instance). Both functions are performed through the same button located in the upper-left corner of the pod (image below).

 

Dialing Out to Remote Users

Us the rPhone to dial out to phones of individuals either in the meeting room or not. To do so, press the phone button that appears in the upper-left corner of the rPhone pod that has the ‘Dial New User’ pop-up appear when moused-over.


When activated, a prompt appears (image below) that allows the Host to enter a phone number to dial out to.

 

When the user becomes connected on the call, their audio instance will appear in the attendee list located in the rPhone pod. If the individual being dialed out to is present in the meeting room, the Host may elect to merge the presence of the individual in the Connect meeting room with their audio presence so that it is known whom the individual is. To do so, hold down ‘control’ on your keyboard, highlight the audio and Connect meeting room presence, and select the ‘Merge Users’ button. View the image below for clarification.

 

Disconnecting Users

To disconnect a user that’s on the call, highlight their name by left-clicking on it, and select the ‘Disconnect User’ button that appears in the top-left corner of the pod as per the image below.

 

 

SORTING THE rPHONE LIST

You may sort the rPhone’s list using the ‘Sort List’ button located in the upper right hand corner of the pod. Please note that the status of the individual in the meeting room – Host, Presenter or Participant – remains the primary filter by which these secondary sort options are organized. Hosts will always remain at the top of rPhone List, followed by Presenters and Participants, respectively.


When activated, a menu appears (view image below) that allows the rPhone List to be ordered by name, who is speaking, and hand raise status.

 

SUB-CONFERENCE ROOMS

The rPhone provides support for up to eight (8) sub conference rooms. When in one of these rooms, attendees only hear the audio coming from others originating in the same room. A useful application of this may be meetings in which multiple languages are used, or independent discussions are being held simultaneously.

Sub-conference rooms are active as long as the audio conference is active. Participants are unable to assign themselves to sub-conference rooms, but a host can manually assign any attendee to any sub-conference room at his or her discretion.

To manually assign an attendee to a particular sub-conference room, highlight their name in the rPhone pod and select the ‘Change Conference Audio’ button.


A pop-up will appear (view image below) that will allow you to move the user into the room of your choosing.

 

BREAKOUT ROOMS INTEGRATION

The rPhone integrates with the breakout rooms functionality in Connect such that when breakouts are activated, users are moved into the audio sub-conference room that corresponds with the Connect breakout room they have been placed in. For instance, if user 1 is moved into breakout room 1, they will automatically be placed by the rPhone into the audio sub-conference room 1, as well. When breakouts end, users are automatically moved back into the main audio conference room.

Note: This functionality only works with users that are on the phone conference.

 

TRANSCRIPTION

The rPhone and ZipDX offer a first-of-its-kind live voice-to-text transcription service! When activated, the conversation on the phone is transcribed into text that appears in the Transcription Box of the rPhone.

Transcription is an additional service that will be charged for in addition to the cost of your phone connections. While the cost is extremely reasonable and significantly lower than other transcription services, it is not a free service.

Transcription is available in two offerings:
Scribble is a fully automated machine transcription service that is extremely inexpensive. It is however not terribly accurate and often misinterprets what is said. While it is extremely cheap it is typically of only limited value at this time. We hope to see improvements in the quality of this service over the coming months but we suggest using this in some trial meetings to see what value it might have for you.

Scribe is a human-based transcription service and does an excellent job of converting what is said in your meetings to readable English text. The phone audio is broken down in to small “snippets” of audio that are then transcribed by a large pool of human operators. The audio from each speaker in the room is analyzed by a different operator and there is no guarantee that consecutive snippets of audio by the same speaker will be transcribed by the same operator.

As the transcriptions are returned, the rPhone reassembles them back to the rPhone Transcription pod for viewing inside of the meeting room. This is a unique feature not available on any other platform.

To activate transcription, mouse over and select the ‘Transcriptions’ button in the bottom right hand corner of the pod, as per the image below.


When activated, the Transcription Box will launch (image below). This is where text appears. You have the option of choosing which transcription method you would like to use – scribble (automated) or scribe (human transcription).

 

HIDING THE rPHONE

You may wish to hide the rPhone on your layout for a variety of reasons: so as not to reveal the attendance of participants to one another; real estate on the layout is hard to find; or you do not use the pod’s features much.

The rPhone comes with a hide option that is embedded in the Connect title bar in your meeting room.

The hide option allows Hosts to show the pod to all meeting room attendees (‘All’ option); just to Hosts (‘Hosts’ option); or to render it invisible to everyone (‘None’ option). Regardless of whether you elect to hide the pod or not, it will continue to be located on the layout, and all attendees that enter the meeting room after the audio conference has started will still be prompted with dial-in options.

 

INTERNATIONAL DIAL-IN NUMBERS

For the most updated list of international dial-in numbers, please go here.

 

FREQUENTLY ASKED QUESTIONS

Q: What is the rPhone and who is Refined Data?
A: The rPhone – designed by Refined Data – is add-in pod for organizations that utilize telephony in Connect Meeting Rooms. It is an audio bridge that fully integrates with Adobe Connect’s Universal Voice feature (available in Adobe Connect 7.5 and 8.0) to broadcast PTSN, mobile, and SIP-based audio communication over the computer speakers of attendees in a meeting room.. Refined Data is an official Adobe Reseller and worldwide leading developer of custom applications for Adobe Connect. We design products that extend Connect’s underlying capabilities. To learn more about Refined Data please visit http://www.RefinedData.com

Q: Are demonstrations of the rPhone available?
A: Yes. Please contact a sales representative at This e-mail address is being protected from spambots. You need JavaScript enabled to view it for a demo.

Q: What versions of Adobe Connect is the rPhone compatible with?
A: rPhone is compatible with Adobe Connect 8, 7.5 and 7.0.

Q: Can the rPhone be used with both Hosted and Licensed versions of Adobe Connect?
A: Yes.

Q: How much does the rPhone cost?
A: The rPhone is free. Your only expense is your per-minute rate plus any additional surcharges for long distance calling or use of on-demand services such as transcription. There is no additional fee to record your audio conference. Please contact us at This e-mail address is being protected from spambots. You need JavaScript enabled to view it for a quote.

Q: Is the rPhone audio picked up when the meeting is recorded using Adobe Connect’s recording mechanism?
A: Yes, and it is instantaneously available! Best of all, there are no additional recording fees.

Q: Am I able to limit per-minute surcharges on my account?
A: Yes. You may do so through the ZipDX website. Calls that incur charges above your determined maximum surcharge will not be completed.

Q: How do I find the list of surcharge rates for calls to international destinations?
A: Login to your ZipDX account and click the ‘Surcharge Lookup’ button in the top right corner of the screen.

Q: How do I find the list of international dial-in numbers??
A: For the most updated list of international dial-in numbers, please go here.

Q: Does the rPhone’s sub conference rooms feature automatically sync with Adobe Connect’s breakout rooms?
A: Yes, however only for those participants on the phone.

Q: Does the rPhone support two-way Universal Voice?
A: Yes.

Q: What is the difference between the ‘Scribble’ and ‘Scribe’ transcription options?
A: Scribble provides fully automated transcription by a computer, while Scribe delivers transcription by human operators. Scribe transcription provides a far more accurate record of what was said by participants.

Q: What is the typical lag time between activating the transcription function and having the text appear in the Meeting Room?
A: The typical interval is 30-60 seconds for Scribble and 120-240 seconds for Scribe, respectively. In most cases, the actual delays are towards the lower end of ranges for both services.

Q: Whom do I contact for support and troubleshooting?
A: Please contact us at This e-mail address is being protected from spambots. You need JavaScript enabled to view it if you have any questions.

 

 

 

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