2/4/8 Analog Lines VoIP Gateway Integrated H.323 Gatekeeper
This EAL-SK200/EAL-SK400/EAL-SK800 series Gatekeeper is designed as an embedded-based platform. It performs internet telephony services for status of registration, call detail record and security registration policy. EAL-SK200/EAL-SK400/EAL-SK800 series Gatekeeper provides centralized address translation and controls access to the H.323 terminal Networks compatible with all H.323 v2/v3/v4 IP telephony products, including the entire Internet Voice Gateway line. Beside of Gatekeeper function, EAL-SK200/EAL-SK400/EAL-SK800 series also build in 2/4/8 voice channels which supports H.323 signal protocols.
| 200 H.3232 endpoints scale: SK400 providing 250 H.323 endpoints to register. | |
| Register Security Policy: SK400 providing security setting on your H.323 VoIP network. This provides protection for VoIP calls and insures proper endpoint identification. | |
| Pre-Granted Endpoints: letting other gateways or H.323 endpoints which were not register to this embedded gatekeeper. And the registered VoIP Gateways can make an Off-Net call to these pre-granted endpoints. | |
| Real time Call Detail Record and Post Call Detail Record Report : Support Real Time CDR to monitor VoIP calls, including caller ip, called ip , call date , call duration and other information. Also providing a CDR report to look up VoIP call record. | |
| Top 20 list: SK400 Gatekeeper can lists top 20 calls by call duration, caller number, calling number, caller IP or callee IP address. | |
| Syslog Client: Providing CDR information to Syslog Server. |
The following application was use in Multi-Office as an example:
| Voice Codec: G.711(A-law /¦Ì-law),G.729 AB, G.723 (6.3 Kbps / 5.3Kbps) | |
| FAX support : T.30 / T.38 | |
| Echo Cancellation: G.165/G168 | |
| FXO Caller ID detection : DTMF and FSK (Optional) | |
| FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional) | |
| Answer supervision: Support Battery Reverse Detection and Voice Detection. | |
| FXO answer delay time: Support delay 0 ¨C 8000 ms to answer. | |
| Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance | |
| Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional)) | |
| 12K Hz and 16K Hz Metering (Customized) |
| H.323 v2/v3/v4 | |
| LAN :NAT, Virtual Server, DHCP Server | |
| WAN: PPPoE client, DHCP client, Fix IP Address, DDNS client | |
| Network Address Translation: Providing build-in NAT router function | |
| Smart QoS: Guarantee the voice bandwidth |
| Voice channels status display | |
| Adjustable volume : - 9 db ~ 9 db | |
| Silence Compression | |
| Auto Dial for speed | |
| Dynamic Jitter Buffer |
| web-based Graphical User Interface | |
| Remote management over the IP Network | |
| FTP firmware upgrade | |
| Backup and Restore Configuration file | |
| Syslog client support |
| AC power : AC100V-240V, DC12V/1.5A,50/60 Hz | |
| Temperature: 0¡ãC ~ 40¡ãC (Operation) | |
| Humidity: up to 90% non-condensing | |
| Emission: FCC Part 15 Class B, CE Mark | |
| Dimension : 260 x 130 x 35 mm | |
| Weight: 900g (Aluminum) |
SK200/SK400/SK800 VoIP Gateway Catalogue
|
SK400 |
SK401 |
SK402 |
SK404 |
SK200 |
SK201 |
SK202 |
Ethernet Port |
4 LAN + 1 WAN |
||||||
H.323 Gatekeeper |
Embedded H.323 Gatekeeper |
||||||
Telephony Interface |
Quad RJ-11 |
Dual RJ-11 |
|||||
FXS/FXO |
4FXS |
3FXS+1FXO |
2FXS+2FXO |
4FXO |
2FXS |
1FXS+1FXO |
2 FXO |
H.323 v2/v3/v4 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
Smart-Qos |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
Echo Cancellation |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
G.711 a/ฆฬ law G.723, G729AB |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
T.38/T.30 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
NAT Router |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
DHCP server for Private Network |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|
SK800 |
SK802 |
SK804 |
SK808 |
Ethernet Port |
1 LAN + 1 WAN |
|||
H.323 Gatekeeper |
Embedded H.323 Gatekeeper |
|||
Telephony Interface |
Octal RJ-11 |
|||
FXS/FXO |
8FXS |
6FXS+2FXO |
4FXS+4FXO |
8FXO |
H.323 v2/v3/v4 |
จน |
จน |
จน |
จน |
Smart-Qos |
จน |
จน |
จน |
จน |
Echo Cancellation |
จน |
จน |
จน |
จน |
G.711 a/ฆฬ law G.723, G729AB |
จน |
จน |
จน |
จน |
T.38/T.30 |
จน |
จน |
จน |
จน |
2 - 4 Analog Lines FXS/FXO VoIP Gateway
The EAL-S200/EAL-S400 Series VoIP Gateway is fully both SIP and H.323 standard compliant residential gateway that provides a total solution for integrating voice-data network and PSTN. By simple installation, this revolutionary compact voice over IP (VoIP) gateway could be configured as a 2/4 port FXS/FXO VoIP Gateway which provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN). The Gateway is equipped with a four port Ethernet switch and built-in NAT router function that provides Internet access using only one IP address.
Besides, it provides high voice quality and optimized packet voice streaming over managed and public (Internet) IP networks.
| Both support SIP and H.323 protocols: SIP Registration and Digest Authentication; H.323 Gatekeeper Registration. | |
| Single Number / Account for multiple ports. | |
| Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO supports DTMF&FSK Caller ID detection. (Optional) | |
| Flexible and Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting. | |
| AC termination Impedance : 600/900 OHM and complex impedance | |
| Answer Supervision for Polarity Reversal Detection and Voice detection | |
| NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server. | |
| Smart-QoS: This feature provides good voice quality when user place a VoIP call and also access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds. | |
| Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway. | |
| Voice channels status display: This function display each port status like as on-hook, off-hook, calling number called number, talk duration, codec. | |
| Pulse Dial support: Support pulse dialing generation and detection. (Optional) | |
| Flash Detection and Generation Program: FXO support Flash Generation and FXS support Flash Detection. |
The following application was use in Multi-Office as an example:
| Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps) | |
| FAX support : T.30 / T.38 | |
| Echo Cancellation: G.165/G168 | |
| FXO Caller ID detection : DTMF and FSK (Optional) | |
| FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional) | |
| Answer supervision: Support Battery Reverse Detection and Voice Detection. | |
| FXO answer delay time: Support delay 0 ¨C 8000 ms to answer. | |
| Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance | |
| Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional)) | |
| 12K Hz and 16K Hz Metering (Customized) |
| H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support. | |
| LAN :Support Virtual Server, DHCP Server | |
| WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client | |
| Network Address Translation: Providing build-in NAT router function. | |
| Smart QoS: Guarantee the voice bandwidth | |
| IP TOS (IP Precedence) / DiffServ |
| Voice channels status display | |
| Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call) | |
| Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper | |
| Adjustable volume : - 9 db ~ 9 db | |
| Silence Compression / VAD | |
| Auto Dial for speed | |
| Dynamic Jitter Buffer | |
| Hot-Line Support |
| Web-based Graphical User Interface | |
| Remote management over the IP Network | |
| FTP firmware upgrade | |
| Backup and Restore Configuration file | |
| Syslog client support |
| AC power : AC100V-240V, DC12V/1.5A,50/60 Hz | |
| Temperature: 0¡ãC ~ 40¡ãC (Operation) | |
| Humidity: up to 90% non-condensing | |
| Emission: FCC Part 15 Class B, CE Mark | |
| Dimension : 260 x 130 x 35 mm | |
| Weight: 900g (Aluminum) |
S400/200 VoIP Gateway Catalogue
|
S400 |
S401 |
S402 |
S404 |
S200 |
S201 |
S202 |
Ethernet Port |
4 LAN + 1 WAN |
||||||
Telephony Interface |
Quad RJ-11 |
Dual RJ-11 |
|||||
FXS/FXO |
4FXS |
3FXS+1FXO |
2FXS+2FXO |
4FXO |
2FXS |
1FXS+1FXO |
2FXO |
H.323 v2/v3/v4 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
SIP(RFC3261) |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
Smart-Qos |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
NAT Traversal |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
Echo Cancellation |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
G.711 a /ฆฬ -law G.723, G729AB |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
T.38/T.30 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
NAT Router |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
DHCP server for Private Network |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
8/16/24 Modular High Density VoIP Gateway
By Modular hardware design EAL-SB800/EAL-S1600/EAL-S2400 series makes any combination of the telephone interface. The EAL-SB800/EAL-S1600/EAL-S2400 Series VoIP Gateway is fully SIP and /H.323 standard compliant high density gateway that provides a total solution for integrating voice-data network and PSTN. By modular installation, this revolutionary compact voice over IP (VoIP) gateway could be easily configured as 8/16/24 channels high density FXS/FXO VoIP Gateway which provides voice connectivity over the IP network and to the Public Switched Telephone Network (PSTN).
| Modular design to accommodate various types of telephony interfaces. | |
| Both support SIP and H.323 protocols: SIP Registration and Digest Authentication; H.323 Gatekeeper Registration. | |
| Single Number / Account for multiple ports | |
| Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO supports DTMF&FSK Caller ID detection. (Optional) | |
| Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting. | |
| AC termination Impedance : 600/900 OHM and complex impedance | |
| Answer Supervision for Polarity Reversal Detection and Voice detection | |
| NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server. | |
| Smart-QoS Guaranteed: This bandwidth management feature provide good voice quality when user place a VoIP call and access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds. | |
| Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway. | |
| Voice channels status display: This function display each port status like as onhook, offhook, calling number callee¡¯s number, talk duration, codec. | |
| Pulse Dial support: Support pulse dialing generation and detection. (Optional) | |
| Flash Detection and Generation Program: FXO support Flash Generation and FXS support Flash Detection. |
| Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps) | |
| FAX support : T.30 / T.38 | |
| Echo Cancellation: G.165/G168 | |
| FXO Caller ID detection : DTMF and FSK (Optional) | |
| FXO hang up detection / anti-seized : Tone Learning Automatically / Manual Tone Learning (Optional) | |
| Answer supervision: Support Battery Reverse Detection and Voice Detection. | |
| FXO answer delay time: Support delay 0 ¨C 8000 ms to answer. | |
| Adjustable AC Termination Impedance : 600 / 900 OHM and complex Impedance | |
| Failsafe Mechanism (FXS relay to FXO) : Power failed by pass support / Internet Failed by pass (Optional)) | |
| 12K Hz and 16K Hz Metering (Customized) |
| H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support. | |
| LAN :Support Virtual Server, DHCP Server | |
| WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client | |
| Network Address Translation: Providing build-in NAT router function. | |
| Smart QoS: Guarantee the voice bandwidth | |
| IP TOS (IP Precedence) / DiffServ |
| Voice channels status display | |
| Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call) | |
| Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper | |
| Adjustable volume : - 9 db ~ 9 db | |
| Silence Compression / VAD | |
| Auto Dial for speed | |
| Dynamic Jitter Buffer | |
| Hot-Line Support |
| Web-based Graphical User Interface | |
| RS232 for configuration | |
| Remote management over the IP Network | |
| FTP firmware upgrade | |
| Backup and Restore Configuration file | |
| Front LCD Panel for System Status and Management | |
| Syslog client support |
| Standard 50 pin RJ-21 Telco connectors | |
| AC power : AC100V-240V, DC12V/1.5A,50/60 Hz | |
| Temperature: 0¡ãC ~ 40¡ãC (Operation) | |
| Humidity: up to 90% non-condensing | |
| Emission: FCC Part 15 Class B, CE Mark | |
| Dimension : 440 x 250 x 45 mm | |
| Weight: 3500g (Aluminum) |
SB800/S1600/S2400 Series VoIP Gateway Catalogue
|
กก |
SB800 |
SB808 |
SB804 |
S1600 |
S1608 |
S1616 |
S2400 |
S2412 |
S2424 |
|||
|
Ethernet Interface |
1 WAN + 1 LAN |
|||||||||||
|
RS232 Console |
1 RS232 console for configuration |
|||||||||||
|
LCD Panel |
Front LCD Panel for System Status and Management |
|||||||||||
|
Telephony Interface |
Standard 50-pin Telco connector |
|||||||||||
|
FXS/FXO |
8 FXS |
8 FXO |
4FXS+4FXO |
16 FXS |
8FXS+8FXO |
16 FXO |
24 FXS |
12 FXS + 12 FXO |
24 FXO |
|||
|
Modules Combination |
SB800 |
SB808 |
SB804 |
1SB800+ 1SM800 |
1 SB804+ 1 SM804 |
1SB808 + 1 SM808 |
1SB800+2 SM800 |
1 SB804+ 2SM804 |
1 SB808 +2SM808 |
|||
|
H.323 v2/v3/v4 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
SIP (RFC3261) |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
Smart-Qos |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
NAT Traversal |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
Echo Cancellation |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
G.711 a / ฆฬlaw G.723, G729AB |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
|
T.38/T.30 |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
จน |
|||
SB800/1600/S2400 Series VoIP Modules Card Catalogue
|
Motherboard |
Description and Function |
|
SB800 |
Motherboard with 8 FXS Interface + 1U/19 inch Chassis |
|
SB804 |
Motherboard with 4 FXS + 4 FXO Interface+ 1U/19 inch Chassis |
|
SB808 |
Motherboard with 8 FXO Interface + 1U/19 inch Chassis |
|
Modular Card |
Description and Function |
|
SM800 |
8 FXS Interface Module |
|
SM804 |
4 FXS + 4 FXO Interface Module |
|
SM808 |
8 FXO Interface Module |
2 - 24 Pulse Metering FXS VoIP Gateway for Payphone
The EAL-S200/EAL-S400/EAL-S800/EAL-S1600/EAL-S2400 VoIP Gateway is fully both SIP and H.323 standard compliant residential gateway that provides a Payphone solution for integrating voice-data network and payphone. By simple installation, this revolutionary compact voice over IP (VoIP) gateway could be configured as a 2/4/8/16/24 port FXS VoIP Gateway which provides voice connectivity over the IP network and to Payphone system
Soundwin VoIP Payphone Solution
| User place a call | |
| The VoIP gateway send an Access-Request to the rate server. | |
| The rate server reply with an Access-Accept including the meter pulse pattern. | |
| The gateway decode the meter pulse pattern and make the outbound call on the H.323 / SIP Network. | |
| After receipt of the Connect message, the gateway begin to apply the meter pulse pattern to send pulse depend on the pulse pattern. |
| Both support SIP and H.323 protocols: SIP Registration and Digest Authentication and H.323 Gatekeeper Registration. | |
| 12K Hz and 16K Hz Metering (Customized) | |
| Radius AAA Client (Customized) | |
| Single Number / Account for multiple ports. | |
| Caller ID Delivery: FXS support DTMF&FSK Caller ID generation; | |
| Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name / IP address, PSTN or PBX) or hunting number setting. | |
| Answer Supervision for Polarity Reversal Detection and Voice detection | |
| NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. Need not to change any configuration of NAT/Firewall like setting virtual server. | |
| NAT traversal: This feature allow gateway to operate behind any NAT/Firewall device. There is no need to change any configuration of NAT/Firewall like setting virtual server. | |
| Smart-QoS: This feature provides good voice quality when user place a VoIP call and access internet at the same time. The gateway will automatically start to reserve bandwidth for voice traffic when VoIP call proceeds | |
| Call Hunting Facility: This function helps gateway to use the lines effectively. This facility automatically transfers your incoming call to a free line. Subscribers need not indicate numerous numbers of each port of gateway. | |
| Voice channels status display: This function display each port status like as on-hook, off-hook, calling number callee¡¯s number, talk duration, codec. |
| Voice Codec: G.711(A-law /¦Ì-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps) | |
| FAX support : T.30 / T.38 | |
| FXS Battery Revere Generation | |
| 12K Hz and 16K Hz Metering (Customized) |
| H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support. | |
| LAN :Support Virtual Server, DHCP Server | |
| WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client | |
| Network Address Translation: Providing build-in NAT router function. | |
| Smart QoS: Guarantee the voice bandwidth | |
| IP TOS (IP Precedence) / DiffServ |
| Voice channels status display | |
| Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call) | |
| Register Call Mode : register to SIP Proxy Server or H.323 Gatekeeper | |
| Adjustable volume : - 9 db ~ 9 db | |
| Silence Compression / VAD | |
| Auto Dial for speed | |
| Dynamic Jitter Buffer | |
| Hot-Line Support |
| Web-based Graphical User Interface | |
| Remote management over the IP Network | |
| FTP firmware upgrade | |
| Backup and Restore Configuration file | |
| Syslog client support |
| AC power : AC100V-240V, DC12V/1.5A,50/60 Hz | |
| Temperature: 0¡ãC ~ 40¡ãC (Operation) | |
| Humidity: up to 90% non-condensing | |
| Emission: FCC Part 15 Class B, CE Mark |
S200/S400/S800/S1600/S2400 VoIP Gateway Catalogue
|
S200 |
S400 |
S800 |
S1600 |
S2400 |
Ethernet Port |
4 LAN + 1 WAN |
4 LAN + 1 WAN |
1LAN + 1 WAN |
1LAN + 1 WAN |
1LAN + 1 WAN |
FXS/FXO |
2FXS |
4FXS |
8FXS |
16FXS |
24FXS |
H.323 v2/v3/v4 |
จน |
จน |
จน |
จน |
จน |
SIP(RFC3261) |
จน |
จน |
จน |
จน |
จน |
Smart-Qos |
จน |
จน |
จน |
จน |
จน |
NAT Traversal |
จน |
จน |
จน |
จน |
จน |
G.711 a/u law G.723, G729AB |
จน |
จน |
จน |
จน |
จน |
T.38/T.30 |
จน |
จน |
จน |
จน |
จน |
NAT Router |
จน |
จน |
จน |
จน |
จน |
DHCP server for Private Network |
จน |
จน |
จน |
จน |
จน |
Integrating PSTN call into VoIP Phone!
The EALCanada's VoIP Phone supported SIP protocol and equipped with an integrated FXO port for PSTN connectivity. This feature provides user can transfer to answer call from PSTN when using IP call. This unique feature lets user never missed any call from IP calls or PSTN calls especially for business call or emergency call. Otherwise the S102กฏs FXO interface supports operation of termination and origination by PSTN line. The IP Phone provides high voice quality and optimized packet voice streaming over IP networks.
S100 |
S102 |
Key Features |
|
* |
Answer PSTN Call: Holding VoIP call and answer the PSTN call. |
|
* |
Manual Select Call Route: User can select call route by VoIP or PSTN route. |
|
* |
Life Line support: IP Phone switch to PSTN call automatically when Power is off. |
* |
* |
Call History (Incoming calls / Outgoing calls / Missing calls), Phone Book |
The following application was use in Multi-Office as an example:
| SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC2916),RTP Payload for DTMF Digits(RFC2833), Outbound Proxy Support. | |
| WAN: PPPoE client, DHCP client, Fix IP Address, Dynamic DNS client |
| Provides Essential Features: Phone Book, Call History (Incoming calls / Outgoing calls / Missing calls) | |
| Hand free Speaker Phone | |
| Speed Dialing | |
| IP address Call / Domain Name Call | |
| Voice Codec : G.711(a-law /¦Ì -law ), G.723(6.3Kbps/5.3Kbps), G.729AB (8Kbps) |
| Key and LCD configuration (2x16 characters LCD) | |
| Web-based Graphical User Interface | |
| Remote management over the IP Network | |
| FTP firmware upgrade |
| AC power : AC100V-240V, DC12V/1.5A,50/60 Hz | |
| Temperature: 0¡ãC ~ 40¡ãC (Operation) | |
| Humidity: up to 90% non-condensing | |
| Emission: FCC Part 15 Class B, CE Mark | |
| Weight: 735g |
S100 Series VoIP Phone Catalogue
|
S100 |
S102 |
Ethernet Interface |
1 WAN + 1 PC |
|
FXO port |
|
1 |
SIP(RFC3261) |
* |
* |
STUN Client |
* |
* |
Symmetric RTP |
* |
* |
Outbound Proxy Server |
* |
* |
Call History |
|
* |
Phone Book |
* |
* |
Life Line |
|
* |
VOIP datasheet