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Key
Features
- Dual mode operation PSTN as well as SIP VoIP agents
- Built-in SIP VoIP Gateway & Proxy functions
- Pick-up and transfer of VoIP and PSTN calls with IVR assisted vocal
assistance plus sophisticated routing capability
- Network based AA & VM enabled (Support traditional PBX as well)
- Personal Assistant & Follow me services
- Remote extension via SIP agent (With PABX features as local extension)
- Easy setup of administrator and Users using intuitive Web-based
interface
- High reliability in distributed architecture
- Licensed-Provision to allow special function customization (Via CallURL
or S.I designers)
Analog
- Extension to extension call Transfer using Flash key or Key sequence
defined by PBX maker Call Pick up based on PBX usage; for external call,
provide Least Cost routing based on rules defined by system administer(If
via PCX’s IVR engine)
- Access Group and external access are under system control achieving
“Intelligent” dialing (User just dial the number he want, no need to add
prefix or remember access digit(s)
SIP VoIP
- SIP VoIP Extension to extension call Transfer using Flash key
- Call Pick up using “Predefined keys”; for external call, provide Least
Cost routing based on rules defined by system administer.
- Access Group and external access are under system control achieving
“Intelligent” dialing (User just dial the number he want, no need to add
prefix or remember access digit(s)
- Combined SIP Proxy and PCX Call Routing to make SIP VoIP calling easy
and transfer seamlessly
- Monitor and rescue scheme has been included to take care bad network
traffic situation
- Peer to peer VoIP phone calling can still be logged under PCX system
Voice Portal Web
Services
- Complete Auto-Attendant features support multi-language greeting and
multi-operators
- Provide different Operator mode and forwarding selection for work hour
and off-hour response
- Vocal prompts to inform calling line status during call or transfer for
friendly calling experience
- Yellow Book Service Caller ID display; Detail call log for real-time
monitor and later reference
- System verification services including “Circuit Check”; Voice Portal
Quality test, Recording and Replay tests, and “what is my extension”, … etc.
- Easy Customization capability, can do special function per customer’s
request
Personal
Services
- Leave or Retrieve Voice Mail (Via phone or Web UI) at user’s wish
- Voice message (and other Media) can be forwarded, saved, deleted and
e-mailed, etc.
- Send / Receive FAX in personal message box with Fax Server functions
(Optional)
- Call Forwarding (Multiple conditions and levels of setting) to reach
people all the time
- Call Recording (Selectable option depending on PSTN or VoIP)
- Personalize prompts with conditional play selections (Set by users)
- Personal Message auto-play, manipulation via Web UI. (Not bounded by
location and time)
- Phone and Web-based Personal Settings features
OSS and
Administration
- Web-based interface for administer and Users (break time and location
limits)
- Real-time channel and extension Monitor for Analog, VoIP as well as FAX
lines
- Routing rules implementation for different kinds of calls
- Channels configuration for IVR, FAX, SMS and Mail Devices
- User account / extension table Setting (for Analog and VoIP extensions)
- Setup user accounts for enabling web-access to their mail boxes
- Setup Voice Portal Services table and access rules
- Setup Call Routing tables for PSTN, VoIP, FAX and SMS calls
- TUI and GUI Multi-language Production Studio
- Monitor on-line VoIP agents status with searchable logs
- Perform call-progress configuration for PSTN and / or SIP VoIP ports
connected to IVR
- System-wise and misc. settings
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