GSM VoiP Terminal

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SunComm GSM VoIP Terminal


GSM VoIP Terminal SC-495N with 4SIM, 1WAN for 4 channels GSM-VoIP connection, Quad band, SMS, Support NAT, Auto Hunting

SC-495N with antenna (new photo) July 25 2009.jpg

SC-495N GSM VoIP Terminal has 4 channels for call termination (VoIP to GSM) and origination (GSM to VoIP) simultaneously. It is SIP based, compatible with Asterisk, SIP Proxy Server, VoipBuster. Which enable to make 4 calls from IP phones to GSM networks and from GSM networks to IP Phone at the same time. SC-495N support Build in Dial peer Server, Stun Server & NAT Function
It is like a traditional PBX. 5060 is a delegate NUMBER
Which has 4 lines: 5062, 5064, 5066, 5068
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/ IP PBX
      • 5060 Port can be changed to any port as user’s need***

Major Function:
1. VoIP (SIP) – GSM conversion
2. 50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting.
- Support one stage dialing:
  • When LAN phone and SC-495N both register SIP proxy Server or Asterisk or VoipBuster, Users can dial any destination number from LAN phone directly.
      • Please note, SIP proxy Server, Asterisk need to have the route of destination number. VoipBuster must have credit in deposit ***
- Support free mode - 2 stages dialing & assigned mode - 1 stage dialing
3. Voice response for setting and status (dial in from mobile).
4. For call termination (VoIP to GSM) and origination (GSM to VoIP).
5. Standard SIP (RFC2543,RFC3261) protocol, Communicates with other gateway or PC
6. Receive SMS and Send SMS (CDMA VoIP Terminal: SMS feature is not available)
7. Allows your program to Send/receive SMS with AT Command
8. Support Call Back feature
9. All functions can be set on web.

Specification:

Protocols: SIP (RFC2543, RFC3261)

TCP/IP: IP/TCP/UDP/RTP/RTCP/, CMP/ARP/RARP/SNTP, DHCP/DNS Client, IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE

Codec: G.711 u-Law, G.711 a-Law, G.729A,G.729A/B
Voice Quality, VAD, CNG, AEC, LEC, Packet loss

Frequency: Quad Band: 900/1800/1900/850MHZ

SC-495N series models:
1.SC-495N GSM VoIP Terminal: for GSM – VoIP connection
2.SC-495G 3G VoIP Terminal: for 3G/UMTS - VoIP connection, tri band 850/1900/2100MHz (Voice only)
3.SC-495cdma VoIP Terminal: for CDMA - VoIP connection
Other Option:
Remote SIM Switch SC-0032RS with 32 SIM: To be installed with SC-495N, SC-895N at same site or at remote site or foreign country for ISP use.

Joan Hu
Suncomm Technology Co.,Ltd


Website : www.suncommtech.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison3.suncomm.com.tw
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021



GSM VoIP Terminal SC-895N with 8SIM, 1 WAN for 8 channels GSM-VoIP connection, SMS, quad band, (Support NAT, Auto Hunting)

SC-895N with antenna (nw photo) July 25 2009.jpg

SC-895N GSM VoIP Terminal has 8 channels for call termination (VoIP to GSM) and origination (GSM to VoIP) simultaneously. It is SIP based, compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster compatible. Which enable to make 8 calls from IP phones to GSM networks and from GSM networks to IP Phone at the same time. SC-895N support built in Dial peer Server, Stun Server & NAT Function.
It is like a traditional PBX. 5060 is a delegate NUMBER
Which has 8 lines: 5064, 5066, 5068, 5070, 5072, 5074, 5076, 5078
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/ IP PBX
      • 5060 Port can be changed to any port as user’s need***
Major Function:
1. VoIP (SIP) – GSM conversion
2. 50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting.
- Support one stage dialing:
  • When LAN phone and SC-895 both register SIP proxy Server or Asterisk or VoipBuster, Users can dial any destination number from LAN phone directly.
      • Please note, SIP proxy Server, Asterisk need to have the route of destination number. VoipBuster must have credit in deposit ***
- Support free mode - 2 stages dialing & assigned mode - 1 stage dialing
3. Voice response for setting and status (dial in from mobile).
4. For call termination (VoIP to GSM) and origination (GSM to VoIP).
5. Standard SIP (RFC2543,RFC3261) protocol, Communicates with other gateway or PC
6. Receive SMS and Send SMS (CDMA VoIP Terminal: SMS feature is not available)
7. Allows your program to Send/receive SMS with AT Command
8. Support Call Back feature
9. All functions can be set on web.


Specification:

Protocols: SIP (RFC2543, RFC3261)

TCP/IP: IP/TCP/UDP/RTP/RTCP/, CMP/ARP/RARP/SNTP, DHCP/DNS Client, IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE

Codec: G.711 u-Law, G.711 a-Law, G.729A,G.729A/B
Voice Quality, VAD, CNG, AEC, LEC, Packet loss

GSM Frequency: Quad band 900/1800/850/1900MHz

SC-895 series models:
1.SC-895N GSM VoIP Terminal: for GSM – VoIP connection
2.SC-895G 3G VoIP Terminal: for 3G/UMTS VoIP connection,
UMTS tri band 850/1900/2100MHz (Voice only)
3.SC-895cdma VoIP Terminal: for CDMA VoIP connection

Other Option:
1.Remote GSM Sim Switch SC-0032RS with 32 SIM (hardware): Installed with SC-495N with 4SIM, SC-895N with 8SIM at same installation site or remote site or foreign country for centralized SIM card management for ISP use.
2.SIM Switch Server (software): SC-4X32 for 128 channels use; SC-8X32 for 256 channels use; SC-16X32 for 512 channels use. (Must be installed with SC-495N or SC-895N (hardware) and SC-0032RS Remote Switch with 32SIM (hardware) together)

Joan Hu
Suncomm Technology Co.,Ltd


Website : www.suncommtech.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison3.suncomm.com.tw
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021


SunComm IP PBX


IP PBX SC-1030: 100 ext. 30 concurrent call, 25 SIP trunks; (1WAN, 1LAN)

SC-1030-2 + Logo.jpg

IP PBX SC-7024: 70 ext. 24 concurrent call, 18 SIP trunks (1WAN, 1LAN)

PBX.jpg

IP PBX SC-2050 with 1WAN, 1LAN, 200 ext, 50 concurrent call,50 SIP trunk;support BRI: 4-port ISDN BRI TE card,4FXS+4FXO, 8FXO, 4FXO+8FXO optional

PBX.jpg


System Highlight:
• Highly integrated, embedded system for more stability and worth of your investment
• Immediately provides VoIP connection to ITSP and to any remote site
• Analog interface to PSTN
• Seamless integration with legacy PBX
• Flexible dialing plan settings for applying to a virtual IP-PBX system
• Off-ramp and on-ramp call for VoIP and PSTN
• Remote management capability
• Quick-batch configuration by Command Line Interface (CLI)
Features Highlight:
• Multi-lingual voice prompts for international business
• Customizable 3-layer IVR for creating your own IVR scenarios
• ACD ( Automatic Call Distribution ) to form basic Call Centers for facilitating small-medium enterprise’s needs
• Various automatic schemes including auto provision and auto firmware upgrade
• Stackable design for scalability and for preserving your pervious investment
• UMS ( Unified Messaging feature ) supported
• Meet-me conference service to expend you meeting room virtually
• Newly released wizard function for easier configuration
• Function-rich Voice Mail System to manage your voice message efficiently
• Voice Mail System with E-mail Notification
• IP Intercom/ IP Paging
• D-Auth (Dialer Authentication)
• Time-based Memo Call
• Scheduled Broadcast Event
• Group Call
• Caller ID/DISA
• CAC ( Call Admission Control )
• Call keep alive scheme
• Most legacy PBX features supported
• CDR (Call Detail Record) with .CSV file format
• FAX relay and pass through (T.38 and T.30)
SIP Compliance
Supported Standards:
RFC 3261, RFC 3311, RFC 3515, RFC 3265, RFC 3892, RFC 3361
RFC 3842, RFC 3389, RFC 3489
RFC 3428, RFC 2327, RFC 2833
RFC 2976, RFC 3263, RFC 3264
RFC 3362, RFC 4612,


SIP Registrar:
• Static/Dynamic registration
• Configurable Expiry Time
• MD5 authentication
• Handle loose RFC-compliant SIP devices
• Resilient message retry mechanism
• Cache client registrations
SIP Proxy:
• Stateful proxy server
• NAT traversal for clients
• Inter-proxy call hand-off
• Outbound Proxy behind NAT Device
PBX System
Call Features:
• 200 users and extensions with Voice Mail account
• 50 concurrent sessions2
• Codec G.711 (μ/A-law), G.723.1 (6.3k/5.3k bit/s), G.729A, and G.726 (16k/24k/32k/40k bit/s) supported
• Transcoding channel 0~16, subject to add-on card
• In-band/RFC2833/SIP-INFO DTMF translation
• Two expandable slots for telephony interfaces
• 50 SIP trunks for ITSP account or private trunking shared by extensions
• 200 DID SIP trunks to extensions
• Support gateway trunk mode per SIP trunk
• Enable/Disable NAT Traversal per SIP trunk
• Call admission control of call count or bandwidth per SIP trunk
• Long call audit
• Support Call keep alive
• Support Registered keep alive
• NAT session keep alive
• Configurable RFC 2833 payload type per SIP trunk
• FXS/FXO analog trunking
• FXO disconnection tone detection
• FXO disconnection tone parameter setting
• FXS hot line
• FXS warm line
• Caller ID detection
• Trunk hunting
• Digits manipulation during hunting
• Life-line priority call
• Support SIP Call Hold, Call Waiting
• Support SIP phone 3-way conference
• Support Blind/ Attended Transfer
• In-line Call Transfer
• Unconditional, Unavailable, Busy Call forward
• Call Back on Busy between extensions
• Per calling number forward and rejection
• Blacklist of number patterns
• 32 call pick-up groups
• Call Park and Retrieve
• Recording on demand with Essence IP phones
• Remote extension registration via Internet
• Direct line to extension (DID to Extension)
• Direct line by called number (DID by Number)
• Direct line by privilege (DID by Privilege)
• Echo Cancellation (G.168)
• Flexible numbering plan
• Call privilege grouping
• Configurable Music on Hold
• Memo Call for extension
• Schedule-based Broadcast
• Support T.38 FAX over IP
• Support T.30, T.38 FAX pass through

• ENUM resolution
• Auth. Dial passcode
• Group Call
• Support H.261, H.263, H.264, MPEG-4 and MJPEG Video codec pass through
• Peer to Peer (Invite/Update)
• Fast Bridging for expressing media forwarding
NAT:
• Auto NAT discovery and traversal
• Built-in STUN client
• RTP proxy
• RTP port range designation
IVR:
• 50 configurations of 3-layer IVR
• Work time/ Holiday setting for different IVR
• Configurable greeting prompts
• Music on Ringing extensions
• Forward to Voice Mail on No-answer
• Support 3 languages in IVR tree
• Hot key to operator
Voice Mail:
• User Authentication by PIN
• Multilingual, 3 languages
• Multi-folder Archive
• Fast-forward /Rewind /Undelete
• MWI notification
• VMWI notification
• E-mail notification and attachment (Unified messaging)
• Personal greeting on unavailability and busy
• Record personal greeting through phone
• Voicemail Forwarding
• Reply call or new call after logged in Voicemail menu
• Built-in 40GB hard disk drive for Voicemail
• Support USB 2.0 interface for Voicemail, CDR, and system configuration backup
• Support NFS remote backup for Voicemail, CDR, and system configuration
Meet-me Conference:
• 24 conference rooms with configurable number and PIN
• Up to 24 parties4,1 among all conference rooms
• Lock/Mute/Join/Drop control for administrator
• Music on First Dial-in Party
• Hot key to leave the conference
• Hot key for administrator to manage the conference
Automatic Call Distribution:
• 32 queues with 32 agents among all queues.
• 32 inbound call among all queues
• Configurable waiting length for individual queue
• Support five distribution policies including round robin, ring all, least recent, fewest call, and random
• Configurable waiting time for each queue
• Allow agent remotely log-in
• Agent can participate multiple queues
• Agent phones also allow extension calls
Stackable:
• Support LAN stacking up to 4 units in the same model
• Automatic intra-trunking creation among stacking units
• Automatic configuration publishing from Master to Slaves
• Automatic load balancing in hosting feature phones3
Administration
System Management:
• Web-based configuration with session control
• User and administrator configuration mode
• Automatic expiring the idle sessions

• Support firmware upgrade through the Internet
• Configuration Wizard for mass extensions and users creation
• Step-by-Step Wizard for adding users, extensions and trunks
• Built in online help in wizard
• Command Line Interface (CLI) for configuration
• System event Syslog
• Downloadable Call Detail Record (CDR)
• Extension registration status
• Active call status
• TFTP server and TFTP repository maintenance
• NTP synchronization
• Real Time Clock setting
• DHCP Server with multiple partitions, Per-MAC IP binding, list of options
• Configurable Time Zone
• Firmware Upgrade through web interface and console
Network Management:
• DHCP/PPPoE/Static IP on WAN
• Support MAC Clone on WAN
• Allow WAN to Respond PING
• Allow LAN use only
• Static LAN routing
• Firewall on predefined services
• Virtual Server for client device
• NAT for outbound traffic from LAN
• WAN QoS queuing mechanism for VoIP and data traffic
• Support TOS setting
• DNS forwarder and dynamic DNS
• SNMPv2 with standard MIB format
• Adaptive WAN bandwidth and DSP channel saving
Optional Feature cards:
• FXO: 4-port FXO card (with 4 transcoding channels)
• FXS: 4-port FXS card (with 4 transcoding channels)
• BRI: 4-port ISDN BRI TE card
• DSP:, DSP card with 8 transcoding channels
• IPSec:
Hardware Specification
Hardware Interfaces:
• One RJ-45 10/100 base-T WAN Ethernet port
• One RJ-45 10/100 base-T LAN Ethernet port
• Two USB 2.0 ports
• One RS-232 serial port
• Two expandable PCI interface slots
System Dimension:
• 443 x 315 x 44 (mm), 1U rack mount
System Power Requirement:
• Power input 100~240V AC, 50~60 Hz
• 40 W (max)
Environment:
• Operating temperature 0~50℃
• Storage temperature -10~70℃
• Humidity (RH) 10~80% non-condensing
Regulatory and Safety:
• FCC Class A certified, FCC part 68, CE/EMC/LVD/TBR21, VCCI, JATE, ROHS




Gary Tseng
Suncomm Technology Co.,Ltd
Website : www.suncomm.com.tw
E-mail: sales2@suncomm.com.tw
MSN:gary@suncomm.com.tw
Skype: gary-suncomm
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021





SunComm GSM VoIP Terminal


GSM VoIP Terminal SC-495N with 4SIM, 1WAN for 4 channels GSM-VoIP connection, Quad band, SMS, Support NAT, Auto Hunting

SC-495N with antenna (new photo) July 25 2009.jpg

SC-495N GSM VoIP Terminal has 4 channels for call termination (VoIP to GSM) and origination (GSM to VoIP) simultaneously. It is SIP based, compatible with Asterisk, SIP Proxy Server, VoipBuster. Which enable to make 4 calls from IP phones to GSM networks and from GSM networks to IP Phone at the same time. SC-495N support Build in Dial peer Server, Stun Server & NAT Function
It is like a traditional PBX. 5060 is a delegate NUMBER
Which has 4 lines: 5062, 5064, 5066, 5068
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/ IP PBX
      • 5060 Port can be changed to any port as user’s need***

Major Function:
1. VoIP (SIP) – GSM conversion
2. 50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting.
- Support one stage dialing:
  • When LAN phone and SC-495N both register SIP proxy Server or Asterisk or VoipBuster, Users can dial any destination number from LAN phone directly.
      • Please note, SIP proxy Server, Asterisk need to have the route of destination number. VoipBuster must have credit in deposit ***
- Support free mode - 2 stages dialing & assigned mode - 1 stage dialing
3. Voice response for setting and status (dial in from mobile).
4. For call termination (VoIP to GSM) and origination (GSM to VoIP).
5. Standard SIP (RFC2543,RFC3261) protocol, Communicates with other gateway or PC
6. Receive SMS and Send SMS (CDMA VoIP Terminal: SMS feature is not available)
7. Allows your program to Send/receive SMS with AT Command
8. Support Call Back feature
9. All functions can be set on web.

Specification:

Protocols: SIP (RFC2543, RFC3261)

TCP/IP: IP/TCP/UDP/RTP/RTCP/, CMP/ARP/RARP/SNTP, DHCP/DNS Client, IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE

Codec: G.711 u-Law, G.711 a-Law, G.729A,G.729A/B
Voice Quality, VAD, CNG, AEC, LEC, Packet loss

Frequency: Quad Band: 900/1800/1900/850MHZ

SC-495N series models:
1.SC-495N GSM VoIP Terminal: for GSM – VoIP connection
2.SC-495G 3G VoIP Terminal: for 3G/UMTS - VoIP connection, tri band 850/1900/2100MHz (Voice only)
3.SC-495cdma VoIP Terminal: for CDMA - VoIP connection
Other Option:
Remote SIM Switch SC-0032RS with 32 SIM: To be installed with SC-495N, SC-895N at same site or at remote site or foreign country for ISP use.

Joan Hu
Suncomm Technology Co.,Ltd


Website : www.suncommtech.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison3.suncomm.com.tw
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021



GSM VoIP Terminal SC-895N with 8SIM, 1 WAN for 8 channels GSM-VoIP connection, SMS, quad band, (Support NAT, Auto Hunting)

SC-895N with antenna (nw photo) July 25 2009.jpg

SC-895N GSM VoIP Terminal has 8 channels for call termination (VoIP to GSM) and origination (GSM to VoIP) simultaneously. It is SIP based, compatible with Asterisk, Trixbox, 3CX, SIP Proxy Server, VoipBuster compatible. Which enable to make 8 calls from IP phones to GSM networks and from GSM networks to IP Phone at the same time. SC-895N support built in Dial peer Server, Stun Server & NAT Function.
It is like a traditional PBX. 5060 is a delegate NUMBER
Which has 8 lines: 5064, 5066, 5068, 5070, 5072, 5074, 5076, 5078
The call automatically switches from a busy line to available line.
So user just send call to 5060 port from Asterisk/ IP PBX
      • 5060 Port can be changed to any port as user’s need***
Major Function:
1. VoIP (SIP) – GSM conversion
2. 50 sets of LAN --> MOBILE routes setting; 50 sets of MOBILE --> LAN routes setting.
- Support one stage dialing:
  • When LAN phone and SC-895 both register SIP proxy Server or Asterisk or VoipBuster, Users can dial any destination number from LAN phone directly.
      • Please note, SIP proxy Server, Asterisk need to have the route of destination number. VoipBuster must have credit in deposit ***
- Support free mode - 2 stages dialing & assigned mode - 1 stage dialing
3. Voice response for setting and status (dial in from mobile).
4. For call termination (VoIP to GSM) and origination (GSM to VoIP).
5. Standard SIP (RFC2543,RFC3261) protocol, Communicates with other gateway or PC
6. Receive SMS and Send SMS (CDMA VoIP Terminal: SMS feature is not available)
7. Allows your program to Send/receive SMS with AT Command
8. Support Call Back feature
9. All functions can be set on web.


Specification:

Protocols: SIP (RFC2543, RFC3261)

TCP/IP: IP/TCP/UDP/RTP/RTCP/, CMP/ARP/RARP/SNTP, DHCP/DNS Client, IEEE802.1P/Q, ToS/DiffServ, NAT Traversal, STUN, uPnP, IP Assignment, Static IP, DHCP, PPPoE

Codec: G.711 u-Law, G.711 a-Law, G.729A,G.729A/B
Voice Quality, VAD, CNG, AEC, LEC, Packet loss

GSM Frequency: Quad band 900/1800/850/1900MHz

SC-895 series models:
1.SC-895N GSM VoIP Terminal: for GSM – VoIP connection
2.SC-895G 3G VoIP Terminal: for 3G/UMTS VoIP connection,
UMTS tri band 850/1900/2100MHz (Voice only)
3.SC-895cdma VoIP Terminal: for CDMA VoIP connection

Other Option:
1.Remote GSM Sim Switch SC-0032RS with 32 SIM (hardware): Installed with SC-495N with 4SIM, SC-895N with 8SIM at same installation site or remote site or foreign country for centralized SIM card management for ISP use.
2.SIM Switch Server (software): SC-4X32 for 128 channels use; SC-8X32 for 256 channels use; SC-16X32 for 512 channels use. (Must be installed with SC-495N or SC-895N (hardware) and SC-0032RS Remote Switch with 32SIM (hardware) together)

Joan Hu
Suncomm Technology Co.,Ltd


Website : www.suncommtech.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison3.suncomm.com.tw
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021


SunComm IP PBX


IP PBX SC-1030: 100 ext. 30 concurrent call, 25 SIP trunks; (1WAN, 1LAN)

SC-1030-2 + Logo.jpg

IP PBX SC-7024: 70 ext. 24 concurrent call, 18 SIP trunks (1WAN, 1LAN)

PBX.jpg

IP PBX SC-2050 with 1WAN, 1LAN, 200 ext, 50 concurrent call,50 SIP trunk;support BRI: 4-port ISDN BRI TE card,4FXS+4FXO, 8FXO, 4FXO+8FXO optional

PBX.jpg


System Highlight:
• Highly integrated, embedded system for more stability and worth of your investment
• Immediately provides VoIP connection to ITSP and to any remote site
• Analog interface to PSTN
• Seamless integration with legacy PBX
• Flexible dialing plan settings for applying to a virtual IP-PBX system
• Off-ramp and on-ramp call for VoIP and PSTN
• Remote management capability
• Quick-batch configuration by Command Line Interface (CLI)
Features Highlight:
• Multi-lingual voice prompts for international business
• Customizable 3-layer IVR for creating your own IVR scenarios
• ACD ( Automatic Call Distribution ) to form basic Call Centers for facilitating small-medium enterprise’s needs
• Various automatic schemes including auto provision and auto firmware upgrade
• Stackable design for scalability and for preserving your pervious investment
• UMS ( Unified Messaging feature ) supported
• Meet-me conference service to expend you meeting room virtually
• Newly released wizard function for easier configuration
• Function-rich Voice Mail System to manage your voice message efficiently
• Voice Mail System with E-mail Notification
• IP Intercom/ IP Paging
• D-Auth (Dialer Authentication)
• Time-based Memo Call
• Scheduled Broadcast Event
• Group Call
• Caller ID/DISA
• CAC ( Call Admission Control )
• Call keep alive scheme
• Most legacy PBX features supported
• CDR (Call Detail Record) with .CSV file format
• FAX relay and pass through (T.38 and T.30)
SIP Compliance
Supported Standards:
RFC 3261, RFC 3311, RFC 3515, RFC 3265, RFC 3892, RFC 3361
RFC 3842, RFC 3389, RFC 3489
RFC 3428, RFC 2327, RFC 2833
RFC 2976, RFC 3263, RFC 3264
RFC 3362, RFC 4612,


SIP Registrar:
• Static/Dynamic registration
• Configurable Expiry Time
• MD5 authentication
• Handle loose RFC-compliant SIP devices
• Resilient message retry mechanism
• Cache client registrations
SIP Proxy:
• Stateful proxy server
• NAT traversal for clients
• Inter-proxy call hand-off
• Outbound Proxy behind NAT Device
PBX System
Call Features:
• 200 users and extensions with Voice Mail account
• 50 concurrent sessions2
• Codec G.711 (μ/A-law), G.723.1 (6.3k/5.3k bit/s), G.729A, and G.726 (16k/24k/32k/40k bit/s) supported
• Transcoding channel 0~16, subject to add-on card
• In-band/RFC2833/SIP-INFO DTMF translation
• Two expandable slots for telephony interfaces
• 50 SIP trunks for ITSP account or private trunking shared by extensions
• 200 DID SIP trunks to extensions
• Support gateway trunk mode per SIP trunk
• Enable/Disable NAT Traversal per SIP trunk
• Call admission control of call count or bandwidth per SIP trunk
• Long call audit
• Support Call keep alive
• Support Registered keep alive
• NAT session keep alive
• Configurable RFC 2833 payload type per SIP trunk
• FXS/FXO analog trunking
• FXO disconnection tone detection
• FXO disconnection tone parameter setting
• FXS hot line
• FXS warm line
• Caller ID detection
• Trunk hunting
• Digits manipulation during hunting
• Life-line priority call
• Support SIP Call Hold, Call Waiting
• Support SIP phone 3-way conference
• Support Blind/ Attended Transfer
• In-line Call Transfer
• Unconditional, Unavailable, Busy Call forward
• Call Back on Busy between extensions
• Per calling number forward and rejection
• Blacklist of number patterns
• 32 call pick-up groups
• Call Park and Retrieve
• Recording on demand with Essence IP phones
• Remote extension registration via Internet
• Direct line to extension (DID to Extension)
• Direct line by called number (DID by Number)
• Direct line by privilege (DID by Privilege)
• Echo Cancellation (G.168)
• Flexible numbering plan
• Call privilege grouping
• Configurable Music on Hold
• Memo Call for extension
• Schedule-based Broadcast
• Support T.38 FAX over IP
• Support T.30, T.38 FAX pass through

• ENUM resolution
• Auth. Dial passcode
• Group Call
• Support H.261, H.263, H.264, MPEG-4 and MJPEG Video codec pass through
• Peer to Peer (Invite/Update)
• Fast Bridging for expressing media forwarding
NAT:
• Auto NAT discovery and traversal
• Built-in STUN client
• RTP proxy
• RTP port range designation
IVR:
• 50 configurations of 3-layer IVR
• Work time/ Holiday setting for different IVR
• Configurable greeting prompts
• Music on Ringing extensions
• Forward to Voice Mail on No-answer
• Support 3 languages in IVR tree
• Hot key to operator
Voice Mail:
• User Authentication by PIN
• Multilingual, 3 languages
• Multi-folder Archive
• Fast-forward /Rewind /Undelete
• MWI notification
• VMWI notification
• E-mail notification and attachment (Unified messaging)
• Personal greeting on unavailability and busy
• Record personal greeting through phone
• Voicemail Forwarding
• Reply call or new call after logged in Voicemail menu
• Built-in 40GB hard disk drive for Voicemail
• Support USB 2.0 interface for Voicemail, CDR, and system configuration backup
• Support NFS remote backup for Voicemail, CDR, and system configuration
Meet-me Conference:
• 24 conference rooms with configurable number and PIN
• Up to 24 parties4,1 among all conference rooms
• Lock/Mute/Join/Drop control for administrator
• Music on First Dial-in Party
• Hot key to leave the conference
• Hot key for administrator to manage the conference
Automatic Call Distribution:
• 32 queues with 32 agents among all queues.
• 32 inbound call among all queues
• Configurable waiting length for individual queue
• Support five distribution policies including round robin, ring all, least recent, fewest call, and random
• Configurable waiting time for each queue
• Allow agent remotely log-in
• Agent can participate multiple queues
• Agent phones also allow extension calls
Stackable:
• Support LAN stacking up to 4 units in the same model
• Automatic intra-trunking creation among stacking units
• Automatic configuration publishing from Master to Slaves
• Automatic load balancing in hosting feature phones3
Administration
System Management:
• Web-based configuration with session control
• User and administrator configuration mode
• Automatic expiring the idle sessions

• Support firmware upgrade through the Internet
• Configuration Wizard for mass extensions and users creation
• Step-by-Step Wizard for adding users, extensions and trunks
• Built in online help in wizard
• Command Line Interface (CLI) for configuration
• System event Syslog
• Downloadable Call Detail Record (CDR)
• Extension registration status
• Active call status
• TFTP server and TFTP repository maintenance
• NTP synchronization
• Real Time Clock setting
• DHCP Server with multiple partitions, Per-MAC IP binding, list of options
• Configurable Time Zone
• Firmware Upgrade through web interface and console
Network Management:
• DHCP/PPPoE/Static IP on WAN
• Support MAC Clone on WAN
• Allow WAN to Respond PING
• Allow LAN use only
• Static LAN routing
• Firewall on predefined services
• Virtual Server for client device
• NAT for outbound traffic from LAN
• WAN QoS queuing mechanism for VoIP and data traffic
• Support TOS setting
• DNS forwarder and dynamic DNS
• SNMPv2 with standard MIB format
• Adaptive WAN bandwidth and DSP channel saving
Optional Feature cards:
• FXO: 4-port FXO card (with 4 transcoding channels)
• FXS: 4-port FXS card (with 4 transcoding channels)
• BRI: 4-port ISDN BRI TE card
• DSP:, DSP card with 8 transcoding channels
• IPSec:
Hardware Specification
Hardware Interfaces:
• One RJ-45 10/100 base-T WAN Ethernet port
• One RJ-45 10/100 base-T LAN Ethernet port
• Two USB 2.0 ports
• One RS-232 serial port
• Two expandable PCI interface slots
System Dimension:
• 443 x 315 x 44 (mm), 1U rack mount
System Power Requirement:
• Power input 100~240V AC, 50~60 Hz
• 40 W (max)
Environment:
• Operating temperature 0~50℃
• Storage temperature -10~70℃
• Humidity (RH) 10~80% non-condensing
Regulatory and Safety:
• FCC Class A certified, FCC part 68, CE/EMC/LVD/TBR21, VCCI, JATE, ROHS




Gary Tseng
Suncomm Technology Co.,Ltd
Website : www.suncomm.com.tw
E-mail: sales2@suncomm.com.tw
MSN:gary@suncomm.com.tw
Skype: gary-suncomm
Taiwan Tel:886-2-32341496
Taiwan Fax:886-2-32341393
China Tel:86-769-82036350
China Fax:86-769-87868021





Created by: joansuncomm, Last modification: Mon 20 of Feb, 2012 (01:46 UTC)
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