Abbreviation and Terms in Telecom
Abbreviation and Terms in Telecom
Abbreviation and Terms in Telecom
Support Any SIP Provider Contivio.com has the ability to work with any
SIP service provider.
Support Any SIP Gateway Contivio.com has the ability to work with any
SIP gateway to connect to the PSTN.
Support Any SIP PBX Contivio.com has the ability to work with any SIP
PBX device.
Support 3rd Party Call Controlling (3PCC) Allows one entity to set up
and manage a communications relationship or telephone call between two or
more other parties.
Support Any SIP Device The unique ability to use any SIP device (SIP
phone or SIP PBX).
Agent Designated Trunk Agents may use their own personal SIP trunk.
Identity Designated Trunk Agents have the ability to select and choose a
Direct Inbound Dial Number (DID).
Dynamic Call ID Allows you to present any phone number on your account
as your Outbound Caller ID.
Reduce costs and simplify your telephony needs by having Contivio.com as your PBX solution.
If you dont need all the bells and whistles of our contact center solution, use our PBX offering
to connect your organization.
Direct Inward Dial (DID) Calls DID numbers from SIP enabled VoIP
providers may be configured for individual extensions/users.
Stutter Dial Tone Message waiting indicator that alerts you to an unheard
voicemail.
Video Calls (Coming Soon) Agents using the Contivio.com App with an
active webcam can place video calls to other Agents or VoIP Systems allowing
video transport.
Lower VoIP costs With the freedom of choice to select any SIP-compliant
VoIP service, you can select any provider of your choice to help reduce the
total cost of ownership. With Contivio.com you dont have to worry about
hidden, by-the-minute fees.
Improve Agent Productivity: Agents can manage all of their needs without having to
navigate through different systems.
Reduce Agent Fatigue: An easy-to-use solution is critical to keep your agents happy and
their spirits high so their energy is directed towards working with your customers rather
than searching for records and fighting data entry.
Increase Agent Retention: Contact centers, especially, telemarketing firms, are known
for high employee turnover. Simplifying and streamlining workflow will help maintain
your workforce and prevent additional onboarding costs.
Interactive Toolbar Browser The power of Contivio.com located within your web
browser will full functionality and media handling capabilities.
Telephone Soft Pad On-screen dialing pad used to perform outbound calls.
Call Transfer Hold Allows a person to transfer a call to a designated agent where the
call remains on hold until the agent is available.
Find Me, Follow Me External phones will be dialed to find Agents if the initial line is
not answered.
On-demand Call Recording The ability to manually record calls and store those calls
as a hyperlink in your CRM.
Agent Instant Messaging The ability to message internally using the Contivio.com
agent instant messaging app.
Voicemail Management Agents can create their own personal Voicemail message
when calls are not answered.
Call Dispositions Classify calls with a predefined result (e.g., call back customer, etc.).
Call Scripts Scribe exactly what your Agents should say and/or ask when engaged with
a Customer.
Agent Notepad Built in notepad that will route to additional agents when calls are
transferred.
Agent Chat Agents may engage with customers or internally with other
Agents/Supervisors using the web chat feature.
Recent Activity Up to date activity list for quick call back abilities.
Call Wrap Up Allocate a call wrap up period once the call is concluded to close out
your notes or send follow up emails.
History Search The ability to review your prior communication with a customer by
identifiers such as customer account ID.
Online Help 24/7 online help documentation for configuration of your Contivio.com
contact center.
Contivio.coms call & media blending is a mix of multi-channel inbound and outbound
capabilities that work together seamlessly to improve agent productivity and increase customer
satisfaction. Agents can balance and prioritize their workflow to manage their customers by
sending emails, taking incoming chats or placing calls theres no limitation to how you want to
route or manage all communication traffic exactly what you need from contact center
management apps.
Automatic Call Back The ability to request a call back at a later time while
waiting on hold in the IVR
Workforce Management
Overview
Managing your contact center is a balancing act between great customer satisfaction and low
costs. By prioritizing your agents schedules and leveraging their skill sets, youre already at the
forefront of improving the customer experience.
Multi-level Administration Ability to have Administrators and SubAdministrators with different privileges.
Operating Hours Select your daily operational hours dictating the IVR flow.
Staff Scheduling Create staff schedules for Agent availability and move
into creating shifts.
Lower Costs By having the right people at the right time and place for the
right price, you minimize overtime, lower call abandon rates, and retain your
workforce.
Contact center managers can easily assess the quality of the customer experience by reviewing
conversations between agents and callers.
As consumers become more accustomed to self-service, and less-frequent personal interactions
with the companies they trust with their business, the stakes with each interaction become much
higher. Many contact centers are now considered the face of the company and executives are
eager to ensure that every customer is greeted and serviced in a friendly and accommodating
manner. The Contivio.com cloud contact center software allows you to variable record the
communication between agents and customers, and review them afterwards. By saving the
recordings as activities in your CRM, they can be stored indefinitely for compliance or agent
training purposes.
Real-time monitoring and reporting provides critical contact center metrics and gives supervisors
the ability to manage their agent teams effectively. Authorized supervisors can monitor live agent
and customer interactions from any location, at any time.
Authorized supervisors are able to see in real time the status of their contact center through a
web browser. Available statistics such as Service Levels versus Goals, the number of calls in
progress, the number waiting, and the longest waiting interaction per queue/skill are available to
the supervisor, as well as real-time agent status graphs indicating the percentage of time in
various states (e.g., on call, waiting, wrap up, on break, etc.). The intuitive web based report
writer allows you to tailor any type of report for your specific needs, and export them to your inhouse reporting solution.
Call Recording With full automated call recording, you maintain a record of
all the calls. Easy integration with your CRM enables exporting and
forwarding of recorded calls to facilitate quality control, archival, and
transcription.
Call Scripting Scribe exactly what your Agents should say and/or ask when
engaged with a Customer.
Audio Injection Inject Audio into calls (Useful for playback of Terms and
Conditions, or repetitive voice-mails).
Agent Status View the current status of all Agents, e.g., who is ready,
busy, and/or on break.
Analyze historical KPI reports and trends for productivity and areas of
improvement.
Call Center
What Is A Call Center?
Call centers special offices that are purpose-built to handle a large volume of phone calls. Call
centers typically handle customer service, support, telemarketing, telesales and collections
functions. The employees who staff call centers are referred to as agents or customer service
representatives (frequently abbreviated as CSRs). Call centers range from very small informal
operations to massive, highly optimized sites with hundreds or even thousands of agents.
Outbound call centers frequently use a dialer application to connect agents with
targets. Dialers can be simple desktop applications that implement a basic click-to-call
function, or much more advanced systems
Desktop Dialer
This pattern continues as long as the agent remains logged into the system. Power dialers allow
an outbound call center to place far more calls than could be accomplished manually. However,
there is still some agent time wasted handling calls that do not connect, reach answering
machines or otherwise fail to reach the target.
Predictive Dialer
Predictive dialers are essentially smarter power dialers. They carefully monitor the
average handle time for each agent and attempt to predict when an agent will become available.
Rather than placing calls on a one-agent-to-one-call basis, they place more calls than there are
agents available. When a call is answered the system uses various methods to determine if the
answering party is a human or an answering machine. Answering machines are either dropped or
fed a pre-recorded message while live answers are handed off to agents.
Robo Dialer
Sometimes dialer systems are completely automated. These place calls then play
messages to the answering party or machine. These systems are commonly used for notification
purposes (i.e. reminder of a doctors appointment, notice that school has been cancelled) as well
as marketing and political messaging. Some automatic dialer systems support additional features
like surveys or transfers to live agents.
Dialer Considerations
In most jurisdictions there are many laws governing the use of dialers. Before
deploying a dialer be certain to review these to avoid significant penalties or even criminal
charges.
Adjunct Technologies
While ACDs and dialer systems are quite powerful on their own, interconnecting
them with a number of related technologies has the potential to increase the efficiency and in
some cases improve customer experience.
Automated Attendant
Automated attendant systems have long been paired with ACDs, allowing callers to
route themselves into the appropriate call queue. Automated attendants are simply menu systems
that prompt callers to indicate their preference using the keys on their phone or, in some cases,
by speaking keywords. Callers are generally willing to accept up to two levels of menu before
reaching a live agent. More than two levels tends to annoy most callers and can result in an
increase in abandoned calls.
Interactive Voice Response
Call Recording
Call centers frequently record calls either to monitor the performance of their
agents or for regulatory compliance. Call recording systems handle the process of capturing the
audio from all participants in the call, mixing it, storing it and producing an index that allows
administrators or regulators to locate and review recordings. A properly built recording system
makes it easy to pinpoint conversations using common keys including Caller ID, date, time and
agent ID.
Conference Bridge
What Is A Conference Bridge?
A conference bridge allows a group of people to participate in phone call. The most common
form of bridge allows participants dial into a virtual meeting room from their own phone.
Meeting rooms can hold dozens or even hundreds of participants. This is in contrast to threeway calling, a standard feature of most phone systems which only allows a total of three
participants. For most phone systems, conference bridging is an add-on feature that costs
thousands of dollars.
Key Features
Conferencing systems typically support multiple conference rooms, each of which can contain
multiple participants. The total number of rooms and participants varies depending on the
model, hardware capabilities and licensing terms.
Most conference bridge systems allow the administrator to assign DID numbers to conference
rooms. In some cases as single DID number connects callers with an IVR application that
prompts for a room number.
Conference rooms can optionally be secured by a PIN number. Some systems use a common
PIN for all participants while others use custom PINs for each.
Advanced conferencing systems include a graphical user interface that allows all participants to
see who is currently speaking and optionally who has joined the conference. Admnistrators and
moderators generally have a more comprehensive view that includes advanced controls.
Some conference bridge systems include dynamic meeting rooms: rooms that are created on a
scheduled basis. This is particularly common for larger systems where capacity planning is an
issue.
Key Benefits
Conferencing allows companies to save significantly on travel costs. In-person meetings are
costly and time consuming. A conference bridge system can pay for itself in just one avoided
"on-site" meeting.
Conferencing is the core of collaboration and enables distributed or virtual teams. Combined
with VoIP connectivity for remote workers, conferencing makes it simple and affordable for a
team to function across a diverse geography.
Voicemail
What Is Voice Messaging?
Also known as "voicemail," voice messaging lets callers leave messages for subscribers (users)
of the system. Voice messaging systems are frequently used in conjunction with PBX systems,
mobile phones and residential phone services.
Voice messaging includes several core components. The message collection process is activated
when a caller is unable to reach a system user. The message collection application receives data
from the phone system indicating which subscriber the caller was attempting to reach. The
application plays a greeting then records the message. The greeting may be a standard system
greeting or a custom outgoing message recorded by the subscriber.
Once the message has been recorded, the notification component of the voicemail system takes
over and lets the subscriber know that a new message is available. This is handled in different
ways depending on the type of phone system with which the voice messaging platform is
integrated. In most cases, the voicemail system will send a command to the upstream system
(PBX, mobile switching platform, etc.), telling it to turn on the message waiting indicator (MWI)
for the subscriber's phone. The notification system may also send an email which may include an
audio file attachement of the message.
When the subscriber receives the notification they will access the message using one of several
methods. Legacy voice messaging systems require the subscribe to call into an application,
authenticate using their extension number and password, and listen to their store messages
sequentially. More modern systems allow the subscriber to review their messages on their
desktop or mobile phone directly using "visual voicemail." If the message was delivered in an
email, the subscriber can listen using their computer as well.
Once the message(s) have been reviewed, the messaging system sends another command to the
upstream phone system, instructing it to turn off the message waiting indictator and/or decrease
the message count.
Advanced voice messaging is a key component of "unified messaging" platforms -- systems that
combine multiple messaging formats into a single point of access for the subscriber. In a unified
messaging system, the inbox will contains voice, email, fax and sometimes text (IM and/or SMS)
messages. Unified messaging is frequently bundled into an even more comprehensive platform
called Unified Communications or UC.
Key Facts & Features
All voice messaging systems are able to record messages, notfiy subscribers of waiting messages
and provide secured access to those messages. Voice messaging has become a standard feature
on most phone systems, but most vendors still sell it as an add-on or "adjunct" product.
Advanced voice messaging systems can forward messages directly to subscribers as an email. In
some cases the subscriber will still need to delete old messages from the voice messaging
system. More advanced, unified systems will synchronize message status automatically.
There are several standards for integrating voice messaging systems with PBXs and other
telephony platforms. These include simple "in-band" integrations that use touch-tones and more
complex "out-of-band" integrations that use various technolgies. One of the more common
standards is "System Message Desk Interface" or SMDI.
Key Benefits
If properly implemented, voice messaging can increase productivity and enhance customer
service.
What Is A VoIP Gateway?
A VoIP gateway is used to build a bridge between the worlds of legacy telephony and the VoIP.
Gateways are typically used to connect legacy phone systems (PBXs or ACDs) with VoIP
resources, or to connect modern VoIP phone systems with legacy phone lines.
Adding VoIP to a legacy PBX system is a great way to add features and reduce costs. The
gateway connects to the legacy system through either analog or digital trunk ports. The PBX
sees the gateway as either the phone company or as another networked PBX. Calls from the
PBX to the outside world are converted into VoIP calls and sent over the Internet to a VoIP
service provider or other VoIP peer. Calls coming from VoIP sources are converted into the
appropriate legacy protocol and delivered to the PBX.
Using a gateway to connect a VoIP phone system to traditional phone lines makes sense in
situations where SIP trunks are not available or where your application requires the reliability of
the PSTN. It also makes it easy to build redundant systems. The gateway normally
communicates with a primary IP PBX. In the event of a failure on the primary the gateway can
communicate with a backup system.
Other uses for VoIP gateways include staged migrations, where the gateway acts as a bridge
between the PSTN, a legacy PBX and a new IP PBX. In this case, the PSTN trunks are
connected to one interface on the gateway. Another interface connects to the trunk port on the
legacy PBX. The new IP PBX is integrated over a VoIP protocol (generally SIP). The gateway
directs some incoming calls to the legacy PBX and others to the IP PBX. It also passes calls
between the two PBXs. This allows some department or other subdivisions of the company to
remain on the legacy system while others move to the IP system.
Key Facts & Features
Gateways come in several formats. Analog gateways convert between VoIP protocols and
traditional analog phone lines and/or phones. Digital gateways convert between VoIP and various
kinds of digital phone services: T1, E1, PRI and/or BRI, depending on the gateway.
Gateways can typically connect to multiple VoIP endpoints (devices or services), which allows
PSTN resources to be shared between multiple IP communications systems.
The routing functions built into a gateway allow it to intelligently adapt calls from one medium
to another. For example, they can add or remove digits in the dial string when passing a call.
They can also recognize various patterns and route them across different interfaces.
Gateways can be used to convert VoIP calls from one code to another, a process known as
transcoding. This can be useful when bridging multiple VoIP solutions together.
Key Benefits
Gateways can extend the life of legacy equipment by "VoIP enabling" it. This can include
replacing traditional trunk lines with SIP trunks or routing some subset of traffic over VoIP to a
remote PBX or gateway -- a process known as toll bypass.
Gateways enable flexibility. Rather than moving from a traditional PBX to an IP PBX in one
step, a company can stage the migration using the gateway as a bridge between the two systems.
IP PBX
What Is A PBX?
Also known as an IP PBX, Unified Communications System or business phone system, a PBX
acts as the central switching system for phone calls within a business. PBX systems handle
internal traffic between stations and act as the gatekeeper to the outside world. The initials PBX
stand for Private Branch Exchange, a very old fashioned term for a system that has evolved
significantly over the past century.
A traditional PBX is made up of two key elements: lines and stations. The lines, sometimes
called trunks, are connections to the global public switched telephony network (PSTN) by way
of a telephone company. Stations are simply telephones or other endpoint devices like fax
machines, modems and credit card terminals.
The original mission of the PBX was to provide shared access to limited resources. Rather than
having a separate phone line for each phone, a business could share a small pool of lines across a
much larger pool of stations. When a call came it was answered by an operator who then
connected it with the appropriate person or department. When someone inside needed to make a
call, the operator connected them with an available line. Frequently these early systems were
simply called switchboards.
Over time, operators were replaced by electromechanical and later electronic systems for
managing access to lines. Additional features were added to automatically route incoming calls,
to allow active calls to be transferred between stations and to permit or deny calls based on
various rules. Adjunct systems were added for voice messaging, call queuing and other value
added services.
Today, a business phone system is much more than just a simple switch. Adjunct technologies
like automated attendant, voice messaging, call queuing and multi-party conferencing have
become standard features. Basic analog and proprietary digital phones are giving way to
standards-based IP phones. Outside connectivity is now available over the Internet in the form of
SIP trunks or other VoIP services.
When PBXs were originally developed, wireline phone calls were the only type of electronic
communication available. Today, the communications landscape has expanded to include email,
instant messaging, video conferencing, desktop sharing, SMS and mobile telephony. Unified
Communications is a catch-all term that describes the process of merging all of these
technologies and integrating them with business processes. Unified Communications aims to
increase efficiency while simplifying management.
Key PBX Features
If youre looking for a PBX, here are some of the features you should be sure are included:
VoIP Ready: The world is moving away from legacy PSTN lines and towards VoIP. Make
sure your PBX can support IP stations (phones) and IP trunks (service). SIP is the current
de facto standard, so dont buy a phone system that doesnt support it.
Voice Messaging: Once upon a time, voicemail was an optional add-on. Today, its table
stakes. Look for PBXs that can forward voicemail messages to your email as
attachments. If possible, look for IP phones that support visual voicemail.
Mobility: Most businesses have at least some road warriors who spend much of their time
out of the office. Make sure your PBX supports mobility features like Find Me / Follow
Me, remote IP extensions and fixed / mobile convergence.
Conferencing: One of the best ways to cut down on travel costs is teleconferencing. Make
sure your phone system has native support for true multi-party conferences (not just basic
three-way calling).
Reporting: If you cant measure it, you cant manage it. Make sure that the PBX you pick
includes basic call history reporting features.
Product menu
Call Features
ADSI On-Screen Menu System
Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number Discovery (DUNDi)
Do Not Disturb
E911
ENUM
Fax Transmit and Receive
Flexible Extension Logic
Interactive Directory Listing
Volume Control
Privacy
Open Settlement Protocol (OSP)
Overhead Paging
Protocol Conversion
Remote Call Pickup
Remote Office Support
Roaming Extensions
Route by Caller ID
Call Features
SMS Messaging
Spell / Say
Streaming Hold Music
Supervised Transfer
Talk Detection
Text-to-Speech (via Festival)
Three-way Calling
Time and Date
Transcoding
Trunking
VoIP Gateways
Voicemail:
Voicemail to email
Voicemail Groups
Zapateller
Computer-Telephony Integration
Asterisk Gateway Interface (AGI)
Asterisk Manager Interface (AMI)
Asterisk REST Interface (ARI)
Outbound Call Spooling
Scalability
TDMoE (Time Division Multiplex over Ethernet)
Allows direct connection of Asterisk PBX
Zero latency
Uses commodity Ethernet hardware
Voice-over IP
Allows for integration of physically separate installations
Uses commonly deployed data connections
Allows a unified dialplan across multiple offices
Speech
Cepstral TTS
Lumenvox ASR
Codecs
ADPCM
CELT (pass through)
G.711 (A-Law & -Law)
G.719 (pass through)
G.722
G.722.1 licensed from Polycom
G.722.1 Annex C licensed from Polycom
G.723.1 (pass through)
G.726
G.729a
GSM
iLBC
Linear
LPC-10
Speex
SILK
VoIP Protocols
Google Talk
H.323
IAX (Inter-Asterisk eXchange)
Jingle/XMPP
MGCP (Media Gateway Control Protocol
SCCP (Cisco Skinny)
SIP (Session Initiation Protocol)
UNIStim
ISDN Protocols
AT&T 4ESS
EuroISDN PRI and BRI
Lucent 5ESS
National ISDN 1
National ISDN 2
NFAS
Nortel DMS100
Q.SIG