SUNCOMM IP PBX

12FXO VoIP PBX and 3G/GSM VoiP Terminal for VOIP - GSM connection SC-495 & SC-895:

1.IP PBX SC-1030 with 1WAN, 1LAN, 100 extension, 30 concurrent call, 25 SIP trunk
2.IP PBX SC-7024 with 1WAN, 1LAN, 18 SIP trunk, 70 extension, 24 concurrent call
3.IP PBX SC-2050 with 1WAN, 1LAN, 200 ext, 50 concurrent call, 50 SIP trunk,
4.SC-495N GSM VoIP Terminal with 4SIM, 1WAN for 4 channels GSM-VoIP connection, Quad band, SMS, Support NAT, Auto Hunting
5.SC-895N GSM VoIP Terminal with 8SIM, 1WAN for 8 channels use, SMS, Quad band(Support NAT, Auto Hunting)

1. Suncomm IP PBX SC-1030 SC-7024 SC-2050
System Highlight:
• Highly integrated, embedded system with more stability
• 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 Email 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)
Supported SIP 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:
• 100 users and extensions with Voice Mail account
• 30 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
• 2 expandable slots for telephony interfaces
• 25 SIP trunks for ITSP account or private trunking shared by extensions
• 100 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 phone5 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:
• 18 conference rooms with configurable number and PIN
• Up to 18 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:
• 16 queues with 16 agents among all queues.
• 16 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 mounting
System Power Requirement: Power input 100~240V AC, 50~60Hz 40W (max)
Environment:
• Operating temperature 0~50℃; Storage temperature -10~70℃;
• Humidity (RH) 10~80% non-condensing
Regulatory and Safety: FCC, CE/EMC/LVD/TBR21; VCCI, JATE, ROHS。

PBX.jpg
The attachment id given is not valid.

Back and Front Picture

2. 3G / GSM VoIP Terminal
System high light:
SC-495 with 4SIM, 1WAN for 4 channels 3G/ GSM - VoIP connection, quad band,
SMS
SC-895 with 8SIM, 1WAN for 8 channels 3G/ GSM - VoIP connection, quad band,
SMS
Both model support NAT & Auto Hunting.



Contact:
Suncomm Technology Co.,Ltd
Website : www.suncommtech.com.tw
www.suncomm.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison6.suncomm.com.tw

12FXO VoIP PBX and 3G/GSM VoiP Terminal for VOIP - GSM connection SC-495 & SC-895:

1.IP PBX SC-1030 with 1WAN, 1LAN, 100 extension, 30 concurrent call, 25 SIP trunk
2.IP PBX SC-7024 with 1WAN, 1LAN, 18 SIP trunk, 70 extension, 24 concurrent call
3.IP PBX SC-2050 with 1WAN, 1LAN, 200 ext, 50 concurrent call, 50 SIP trunk,
4.SC-495N GSM VoIP Terminal with 4SIM, 1WAN for 4 channels GSM-VoIP connection, Quad band, SMS, Support NAT, Auto Hunting
5.SC-895N GSM VoIP Terminal with 8SIM, 1WAN for 8 channels use, SMS, Quad band(Support NAT, Auto Hunting)

1. Suncomm IP PBX SC-1030 SC-7024 SC-2050
System Highlight:
• Highly integrated, embedded system with more stability
• 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 Email 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)
Supported SIP 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:
• 100 users and extensions with Voice Mail account
• 30 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
• 2 expandable slots for telephony interfaces
• 25 SIP trunks for ITSP account or private trunking shared by extensions
• 100 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 phone5 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:
• 18 conference rooms with configurable number and PIN
• Up to 18 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:
• 16 queues with 16 agents among all queues.
• 16 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 mounting
System Power Requirement: Power input 100~240V AC, 50~60Hz 40W (max)
Environment:
• Operating temperature 0~50℃; Storage temperature -10~70℃;
• Humidity (RH) 10~80% non-condensing
Regulatory and Safety: FCC, CE/EMC/LVD/TBR21; VCCI, JATE, ROHS。

PBX.jpg
The attachment id given is not valid.

Back and Front Picture

2. 3G / GSM VoIP Terminal
System high light:
SC-495 with 4SIM, 1WAN for 4 channels 3G/ GSM - VoIP connection, quad band,
SMS
SC-895 with 8SIM, 1WAN for 8 channels 3G/ GSM - VoIP connection, quad band,
SMS
Both model support NAT & Auto Hunting.



Contact:
Suncomm Technology Co.,Ltd
Website : www.suncommtech.com.tw
www.suncomm.com.tw
E-mail: liaison6@suncomm.com.tw
MSN:liaison6@suncomm.com.tw
Skype: liaison6.suncomm.com.tw

Created by: joansuncomm, Last modification: Mon 18 of Jul, 2011 (13:02 UTC)
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