| 904 IP Phone |
| 907 IP Phone |
| 908 IP Phone |
| VT 909 |
| CCS 900 |
| CCS 910 |
| CCS 920 |
| Analog PBX Solutions |
| SOHO Diagram |
| SMB Diagram |
| Large Diagram |
CCS / (Carrier Class Server) IP/(Internet Protocol) PBX is an affordable IP-PBX designed for small business users with less than 20 users. It provides FXS interfaces for on-site users and FXO interfaces for connecting to PSTN or traditional PBX system. CCS-IP is based on industry standard SIP, has various terminal solutions such as SIP phone, Soft phone and ATA. CCS IP PBX also can be used as a VoIP trunks, interconnecting with ITSP soft switch, routing all the subscribers to ITSP.
CCS
900 is a comprehensive, turn-key VoIP solusion, easy to install and manage. With
VoIPTREX's CCS-IP, small business users do not have to make huge upfront capital
investments to begin taking advantage of monthly telephony cost savings and begin
realizing productivity improvements.
Features
Nimble network ability
CCS 900 is based on standard SIP, any SIP phone, Soft phone, ATA and other kinds
of terminal devices can register to this IP-PBX. It can be used as a traditional
PBX, also can interconnect with PBX, soft switch and PSTN, provide multiple communication
solutions for customers.
Through IP load bearing, voice and data are converged to one network, voice
wiring is no longer needed that simplifies network architecture. And also
subscribers
can move to anywhere which provides more convenience to users.
Traverse NAT/Firewall without the need for a session border controller or
STUN
server to ensure subscriber mobility.
Audio enhancement
Advanced audio processing technology, superior VoIP voice quality
without dependency on Network QoS.
Sustain calls with up to 40% packet drop & 500ms of latency.
Ease of management
CCS-900
uses web based GUI, users can access management portals from anywhere inside IP
network. VoIPTREX also provides a set of administrator--AACP which integrated
call manager and real-time system monitor.
Scalable and resilient architecture
CCS-900
uses modularized user interface devices. Customers can choose desired type and
port number as necessary. It also can be interconnected with other devices and
application servers ( e.g. record server), so that easy to expand system size
and functionalities.
Specifications
Up to 16 users registration.
Up to 20 simultaneous users per system
(IP+TDM ports)
Up to 300 hours cumulative voicemail storage
Support 2 pieces of voice interface cards
4-port FXO card
16-ports FXS
card
8-ports FXO/4-ports FXS combo card
Physical Characteristics
System IO Console RS-232
Power supply External power adapter
Input: AC
100-240V
Output: DC 19V, 4.7A
Max power: 90W
Dimension Mini case:
322.58x68.58x254mm (WxHxD)
Weight 8.25lb (3.75kg)
Operating temperature
0°C to 40°C (32°F to 104°F)
Operating humidity 5% to 85%
Storage temperature 20°C to 75°C (-4°F to 167°F)
Storage humidity
5% to 95%
Certificate FCC, CE, CCC
Basic Features
Auto Attendant
Caller ID
Music on Hold
Call Waiting
Call Transfer
- Transfer with confirmation
- Direct transfer
Call Forwarding
- Call forward when busy
- Call forward when no
answer
- Unconditional Call forward
Call Pick Up
Call
Park
Hunt Groups
Ring Groups
DID Mapping
T.30 FAX
120 parties conference bridge
CDRs
Private Number Plans
Voice Mail
- Customized greeting
- Voicemail
retrieval via phone or e-mail
- Message waiting alerting Audio Features
Codec
- G.711 (A-Law, u-Law)
- GSM (21K)
- G.729
- G.726
DTMF
- RFC 1833
- In-band (by RTP)
- SIP INFO
Audio
processing
- Static sound suppression
- Comfortable noise production
- Echo cancellation G.165/ G.168 25ms
- Dynamic vibration cushion
- Programmable
AGC
Management Features
Layered user access level
Web-based management portal
English
and Chinese user interface
User based provisioning
Upgrade
IP Phone firmware through IP-PBX
Routing Features
IP trunking supports multiple connects
Least cost routing
Time of day routing
FXO trunks
Network Features
one 10/100M Ethernet connection
NAT/Firewall traversal
Upgrade
software behind firewall
Enhanced voice quality
Type of
service
Standards Supported
TCP/UDP/IP
RTP/RTCP
HTTP
ARP/RARP
MD5
ICMP
DNS
DHCP
NTP
TFTP
RFC 3261 - SIP
RFC 2617 - HTTP Digest Authentication
RFC 2976 - SIP INFO Method
RFC 3264 - Offer/Answer model with SDP
RFC 3265 - SIP-specific Event Notification
RFC 3515 - SIP REFER
Method
RFC 3842 - A Message Summary and Message Waiting Indication