|
System Level Features
|
Spectel CS7000
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Wyde Voice Softbridge
|
|
Uses Off-the-shelf hardware
|
N
|
Y
|
|
Operating System
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Proprietary
|
Linux
|
|
Each conference call can share a DID/DDI but have its own language, message, and DTMF set
|
N
|
Y
|
|
DTMF commands can be programmed to emulate any bridge call flow including Spectel
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Spectel only
|
Y
|
|
Billing adapters to ease integration of bridge and billing application
|
N
|
Y
|
|
Authentication adapters to ease integration of bridge and authentication systems
|
N
|
Y
|
|
SIP based
|
N
|
Y
|
|
TDM support
|
Y
|
Y (VoIP gateway required)
|
|
Softphone support
|
N
|
Y
|
|
SNMP monitoring
|
N
|
Y
|
|
Easy to deploy and manage at a remote datacenter
|
N
|
Y
|
|
Real-time Data Replication
|
N
|
Y
|
|
LDAP to MS Active Directory
|
N
|
Y
|
|
Dial-in and Dial-Out Conferencing
|
Y
|
Y
|
|
Active Speaker Notification (to identify who is talking or which line is generating noise)
|
N
|
Y
|
|
Question and Answer Queue Management Tool
|
N
|
Y
|
|
Broadcaster Feature (Broadcast any discrete audio file into a live
conference to reduce need to repeat training or other content)
|
N
|
Y
|
|
Flash-based User Interface that allows operators and hosts to manage conference calls
|
N
|
Y
|
|
Access codes of any length
|
N
|
Y
|
|
Ability for each call host to control and manage multiple access codes
associated with their email address
|
N
|
Y
|
|
Platform can have static IPs, take IPs from external DHCP server or work in a separate
dedicated subnet and run internal DHCP server for local components
|
N
|
Y
|
|
Scheduled and Instant (Reservationless) Conferencing
|
Y
|
Y
|
|
Attended and Unattended Services
|
Y
|
Y
|
|
Callflow Flexibility - More than 30 parameters related to call flow can be changed through web interface
|
N
|
Y
|
|
Each phone line recorded individually in the conference allowing ability to search
for time when specific person talked during call
|
N
|
Y
|
|
Sub conferencing (ability have create/manage breakout rooms in a conference)
|
N
|
Y
|
|
Flash Based Dashboard that allows operators and call hosts to manage Question and Answer Sessions
|
N
|
Y
|
|
Web service API and RT API that allows for integration with external web conferencing solutions
|
N
|
Y
|
|
Blast Dial-out and Dial-Out to Streaming available through API
|
N
|
Y
|
|
Active Speaker Notification (tell who is talking, isolate noisy lines)
|
N
|
Y
|
|
Custom greeting available per DNIS and per conference
|
N
|
Y
|
|
Access to Call playback via phone or UI
|
N
|
Y
|
Flexible Mute Control
Individual/Group mute is available via DTMF/Web/API.
Group mute allows moderator to mute/unmute all participants (Open-Mute-Lecture);
Open - all can speak
Mute - all participants muted but can unmute themselves
Lecture - all muted can’t unmute themselves
|
N
|
Y
|
|
High Definition Codec support: G.722, G.722.1, G.722.2, iSAC
|
N
|
Y
|
|
Waiting Room Feature for incoming participants. Moderator can review the queue
through the UI or interview each participant, and then drop selected calls.
|
N
|
Y
|
|
Automatic Gain Control
|
N
|
Y
|
|
Support for local and remote software upgrade, monitoring, diagnosis, configuration,
and troubleshooting
|
N
|
Y
|
