Yeastar - NeoGate

VoIP Hardware Solutions
Provider Solution Details
VoIP Hardware Zycoo UC Solutions
  • Modular Design IP PBX for SMB
  • Remote office Centralized Management solution
  • 3rd party app integration, Enterprise Billing, Android & iOS client
Details

Yeastar TG100 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG100 is a fully featured 1 port VoIP GSM/CDMA/UMTS gateway that provides GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It significantly reduces the costs of calls with two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. With friendly GUI, everything can be easily set up.

TG100侧面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 1
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 110x70x24mm
Power Supply: AC 100~240V 50/60Hz (DC 12V, 1A)

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG200 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG200 is a VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It supports two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM/CDMA/UMTS network.

TG200正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 2
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 213x160x44mm
Power Supply: AC 100-240V 50/60Hz 0.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG400 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG400 is a compact 4 channels VoIP GSM/CDMA/UMTS gateway that connects GSM/CDMA/UMTS network with VoIP. It is the ideal product for small and medium sized companies with heavy demands of calls to mobile networks. The cost-saving solution makes connection cheaper and efficient.

TG400正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 4
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 213x160x44mm
Power Supply: AC 100-240V 50/60Hz 0.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG800 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG800 is a powerful VoIP GSM/CDMA/UMTS gateway with 8 channels, bridging between GSM/CDMA/UMTS network and IP-based systems. Designed to slash the cost of telephone calls, it can find the cheapest route and use the most economical SIM card.

TG800正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 8
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 340x210x44mm
Power Supply: AC 100-240V 50/60Hz 1.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG1600 - VoIP GSM/CDMA Gateway

Reduce costs for SOHO and SMBs

Yeastar TG1600 is a powerful VoIP GSM/CDMA gateway with 16 channels, bridging between GSM/CDMA network and IP-based systems. Designed to slash the cost of telephone calls, it can find the cheapest route and use the most economical SIM card.

TG1600正面副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA channels (Max): 16
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440x280x44mm
Power Supply: AC 100-240V 50/60Hz

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA, GSM/CDMA to VoIP
GSM/CDMA Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TA400 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA400无WAN口+支架_20150312.2679.png


Highlights

1) 4 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 4
Phone Interface: 4 x RJ11 FXS ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration




Yeastar TA800 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA800无WAN口+支架_20150312.2696.png


Highlights

1) 8 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 8
Phone Interface: 8 x RJ11 FXS ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through

Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA1600 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA1600 正面副本.png


Highlights

1) 16 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 16
Phone Interface: 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA2400 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA2400 正面副本.png


Highlights

1) 24 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 24
Phone Interface: 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA3200 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA2400 正面副本.png


Highlights

1) 32 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 32
Phone Interface: 16 x RJ11 FXS ports plus 2 x 50 pin
Telco connectors
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA410 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA400无WAN口+支架_20150312.2679.png


Highlights

1) 4 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 4
Phone Interface: 4 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA810 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA800无WAN口+支架_20150312.2696.png


Highlights

1) 8 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 8
Phone Interface: 8 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA1610 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA1610正面副本.png


Highlights

1) 16 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 16
Phone Interface: 16 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm × 250 (W) mm × 44 (H) mm
Power Supply:AC100~240V(12V 5.5A)

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration


Yeastar TE100 - VoIP PRI Gateway(VoIP-E1/T1/J1)

Bridge the gap between E1/T1/J1 and VoIP networks

Yeastar TE100 is a single port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 30 concurrent calls. It’s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE100 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE100 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.

TE100正面副本副本.png


Benefits

1) Cost-effective Call Routing
2) Easy-to-navigate GUI
3) Simple Installation
4) Easy to integrate
5) Low power consumption for your green office

Specification:

E1/T1/J1 Port: 1 (Support PRI, MFC R2, SS7)
Protocol: SIP (RFC3261)
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM.
LAN: 10/100 Mbps
WAN: 10/100 Mbps

Size: 213x160x44mm
Power Supply: AC 100~240V,50~60Hz (DC 12V, 1A)

Features:

Trunk Support
Call Routing Rules
Automatic appending and stripping of digits to dialed numbers
Caller ID name and number support
FAX support
Blacklist
Firewall
DDNS Support
Backup And Restore
Easy to install

Applications

1) Connect legacy PBX systems to VoIP services
2) Connect legacy PBX systems to remote sites over private VoIP links
3) Connect IP PBX systems to legacy TDM services
4) Phased transition from Legacy PBX to IP PBX



Yeastar TE200 - VoIP PRI Gateway(VoIP-E1/T1/J1)

Bridge the gap between E1/T1/J1 and VoIP networks

Yeastar TE200 is a dual port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 60 concurrent calls. It’s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE200 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE200 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.

TE200正面副本副本.png


Benefits

1) Cost-effective Call Routing
2) Easy-to-navigate GUI
3) Simple Installation
4) Easy to integrate
5) Low power consumption for your green office

Specification:

E1/T1/J1 Port: 2 (Support PRI, MFC R2, SS7)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM.
LAN: 10/100 Mbps
WAN: 10/100 Mbps

Size: 213x160x44mm
Power Supply: AC 100~240V,50~60Hz (DC 12V, 1A)

Features:

Trunk Support
Call Routing Rules
Automatic appending and stripping of digits to dialed numbers
Caller ID name and number support
FAX support
Blacklist
Firewall
DDNS Support
Backup And Restore
Easy to install

Applications

1) Connect legacy PBX systems to VoIP services
2) Connect legacy PBX systems to remote sites over private VoIP links
3) Connect IP PBX systems to legacy TDM services
4) Phased transition from Legacy PBX to IP PBX



Yeastar TB200 - VoIP BRI Gateway(BRI-VoIP)

Flexible routing for your PBX

Yeastar TB200 - VoIP BRI Gateway(BRI-VoIP) is a full-featured VoIP gateway. It's used to integrate your PABX to
VoIP service or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

TB200正面副本.png


Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

Specification:

BRI Port: Up to 2 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 213x160x44mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install


Yeastar TB400 - VoIP BRI Gateway(BRI-VoIP)

Flexible routing for your PBX

Yeastar TB400 - VoIP BRI Gateway(BRI-VoIP) is a full-featured VoIP gateway. It's used to integrate your PABX to
VoIP service or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

TB400正面副本.png


Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

Specification:

BRI Port: Up to 4 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 213x160x44mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install

Yeastar TG100 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG100 is a fully featured 1 port VoIP GSM/CDMA/UMTS gateway that provides GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It significantly reduces the costs of calls with two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. With friendly GUI, everything can be easily set up.

TG100侧面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 1
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 110x70x24mm
Power Supply: AC 100~240V 50/60Hz (DC 12V, 1A)

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG200 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG200 is a VoIP GSM/CDMA/UMTS gateway with 2 channels providing GSM/CDMA/UMTS network connectivity for softswitch and IP PBX. It supports two-way communication: VoIP to GSM/CDMA/UMTS and GSM/CDMA/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM/CDMA/UMTS network.

TG200正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 2
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 213x160x44mm
Power Supply: AC 100-240V 50/60Hz 0.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG400 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG400 is a compact 4 channels VoIP GSM/CDMA/UMTS gateway that connects GSM/CDMA/UMTS network with VoIP. It is the ideal product for small and medium sized companies with heavy demands of calls to mobile networks. The cost-saving solution makes connection cheaper and efficient.

TG400正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 4
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 213x160x44mm
Power Supply: AC 100-240V 50/60Hz 0.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG800 - VoIP GSM/CDMA/UMTS Gateway

Reduce costs for SOHO and SMBs

Yeastar TG800 is a powerful VoIP GSM/CDMA/UMTS gateway with 8 channels, bridging between GSM/CDMA/UMTS network and IP-based systems. Designed to slash the cost of telephone calls, it can find the cheapest route and use the most economical SIM card.

TG800正面副本副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA/UMTS channels (Max): 8
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
UMTS Network type: 850/1900MHz, 850/2100MHz, 900/2100MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 340x210x44mm
Power Supply: AC 100-240V 50/60Hz 1.5A MAX

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA/UMTS, GSM/CDMA/UMTS to VoIP
GSM/CDMA/UMTS Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TG1600 - VoIP GSM/CDMA Gateway

Reduce costs for SOHO and SMBs

Yeastar TG1600 is a powerful VoIP GSM/CDMA gateway with 16 channels, bridging between GSM/CDMA network and IP-based systems. Designed to slash the cost of telephone calls, it can find the cheapest route and use the most economical SIM card.

TG1600正面副本.png


Benefits

1) Cost Savings - Save phone bills tremendously with mobile-to-mobile calls and LCR.
2) Back up - Work as a cost-effective backup when the landline network goes down.
3) Easy to install - Everything can be easily set up in the Web based management interface.
4) Easy to integrate - High compatibility with major IP PBX and softswitch brands.

Specification:

Number of GSM/CDMA channels (Max): 16
GSM Network type: 850/900/1800/1900MHz
CDMA Network type: 800MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711A/U law, G.722, G.723.1, G.726, G.729a.
Echo Cancellation: ITU-T G.168 LEC

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440x280x44mm
Power Supply: AC 100-240V 50/60Hz

Operation Range: 0° to 40°C, 32° to 104° F
Storage Range: -20° to 65°C, -4° to 149° F
Humidity: 10-90% non-condensing

Features:

SIP Server and SIP Trunk supported
SIP Peer Mode supported
Calling Type: VoIP to GSM/CDMA, GSM/CDMA to VoIP
GSM/CDMA Ports Group Manage, VoIP Trunk Group
Incoming /Outgoing Routing rules
SMS Sending and Receiving
Send Bulk SMS
Gain Adjustment
USSD
PIN Modify
Carrier Selection: Auto/Manual
Balance Alarm
Caller ID/CLIR
Black List
Hotline
Call Duration Limitation
Call Transfer
Call Back
Call Status Display
Call Detail Record (CDR)
Call Progress Tone Generation
Call Duration Limitation for SIM Card/Single Call
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
SIP Response Code Switch
Open API for SMS and USSD
Real Open API Protocol (Based on Asterisk)
IP Blacklist
Network Attack Alert
System Logs
Web based configuration



Yeastar TA400 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA400无WAN口+支架_20150312.2679.png


Highlights

1) 4 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 4
Phone Interface: 4 x RJ11 FXS ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration




Yeastar TA800 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA800无WAN口+支架_20150312.2696.png


Highlights

1) 8 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 8
Phone Interface: 8 x RJ11 FXS ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through

Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA1600 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA1600 正面副本.png


Highlights

1) 16 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 16
Phone Interface: 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA2400 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA2400 正面副本.png


Highlights

1) 24 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 24
Phone Interface: 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal

LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA3200 - Analog VoIP Gateway

Multi-Port FXS Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA2400 正面副本.png


Highlights

1) 32 FXS ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 32
Phone Interface: 16 x RJ11 FXS ports plus 2 x 50 pin
Telco connectors
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm x 250 (W) mm x 24 (H) mm
Power Supply: AC 100~240V/50~60Hz

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All
Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA410 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA400无WAN口+支架_20150312.2679.png


Highlights

1) 4 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 4
Phone Interface: 4 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA810 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA800无WAN口+支架_20150312.2696.png


Highlights

1) 8 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 8
Phone Interface: 8 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 200 (L) mm × 137 (W) mm × 25.6 (H) mm
Power Supply:12V 1.5A

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration



Yeastar TA1610 - Analog VoIP Gateway

Multi-Port FXO Gateway

Yeastar TA Analog VoIP Gateways are cutting-edge products that connect legacy telephones, fax machines and PBX systems with IP telephony networks and IP-based PBX systems. Featuring rich functionalities and easy configuration, Yeastar TA is ideal for small and medium enterprises that wish to integrate a traditional phone system into IP-based system. Yeastar TA helps them to preserve previous investment on legacy telephone system and reduce communication costs significantly with the true benefits of VoIP.

TA1610正面副本.png


Highlights

1) 16 FXO ports
2) Fully compliant with SIP and IAX2
3) Flexible calling rules
4) Reliable fax performance with T.38
5) Provides high-quality voice compression with industry standard codecs
6) Line echo cancellation for 32, 64 or 128 ms echo delays
7) Web-based GUI for easy configuration and management
8) Excellent interoperability with a wide range of legacy and IP equipment

Specification:

Number of Telephony Ports: 16
Phone Interface: 16 x RJ11 FXO ports
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation: ITU-T G.168 LEC
FAX over IP: T.38 and pass-through
Signalling: Kewl Start, Loop Start
Caller ID: BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
Disconnect Method: Busy Tone, Polarity Reversal
LAN: 1 (10/100Mbps)
Network: Static IP, DHCP Client, Firewall, VLAN, DDNS,
QoS, OpenVPN
NAT Traversal: Static NAT, STUN

Size: 440 (L) mm × 250 (W) mm × 44 (H) mm
Power Supply:AC100~240V(12V 5.5A)

Operation Range: 0° to 50°C, 32° to 122° F
Storage Range: -20° to 65°C, 4° to 149° F
Humidity: 10-90% non-condensing

Features:

3-way Conference
Attended Transfer
Black List
Blind Transfer
Call Detail Record (CDR)
Caller ID Display
Call Forward: No Answer, Busy, All Call Waiting
Call Status Display
Do Not Disturb
IP Blacklist
Network Attack Alert
Paging
RADIUS Accounting & Login
Real Open API Protocol (Based on Asterisk)
Voice Prompt
SIP Server and SIP Trunk supported
SIP Peer Mode supported
SNMP
TR069
System Logs
Web based configuration


Yeastar TE100 - VoIP PRI Gateway(VoIP-E1/T1/J1)

Bridge the gap between E1/T1/J1 and VoIP networks

Yeastar TE100 is a single port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 30 concurrent calls. It’s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE100 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE100 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.

TE100正面副本副本.png


Benefits

1) Cost-effective Call Routing
2) Easy-to-navigate GUI
3) Simple Installation
4) Easy to integrate
5) Low power consumption for your green office

Specification:

E1/T1/J1 Port: 1 (Support PRI, MFC R2, SS7)
Protocol: SIP (RFC3261)
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM.
LAN: 10/100 Mbps
WAN: 10/100 Mbps

Size: 213x160x44mm
Power Supply: AC 100~240V,50~60Hz (DC 12V, 1A)

Features:

Trunk Support
Call Routing Rules
Automatic appending and stripping of digits to dialed numbers
Caller ID name and number support
FAX support
Blacklist
Firewall
DDNS Support
Backup And Restore
Easy to install

Applications

1) Connect legacy PBX systems to VoIP services
2) Connect legacy PBX systems to remote sites over private VoIP links
3) Connect IP PBX systems to legacy TDM services
4) Phased transition from Legacy PBX to IP PBX



Yeastar TE200 - VoIP PRI Gateway(VoIP-E1/T1/J1)

Bridge the gap between E1/T1/J1 and VoIP networks

Yeastar TE200 is a dual port VoIP E1/ T1 /J1 gateway (VoIP to E1/ T1 /J1, and E1/ T1 /J1 to VoIP) that supports up to 60 concurrent calls. It’s designed to bridge the gap between E1/ T1 /J1 and VoIP networks. TE200 offers SMBs cost effective additions to an legacy telephone system to bring the true benefits of VoIP. Integrating TE200 into an existing network will allow for inexpensive communication via SIP trunking. Also, it could connect VoIP systems with E1/T1/J1 service from legacy carriers.

TE200正面副本副本.png


Benefits

1) Cost-effective Call Routing
2) Easy-to-navigate GUI
3) Simple Installation
4) Easy to integrate
5) Low power consumption for your green office

Specification:

E1/T1/J1 Port: 2 (Support PRI, MFC R2, SS7)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.722, G.726, G.729 A, GSM.
LAN: 10/100 Mbps
WAN: 10/100 Mbps

Size: 213x160x44mm
Power Supply: AC 100~240V,50~60Hz (DC 12V, 1A)

Features:

Trunk Support
Call Routing Rules
Automatic appending and stripping of digits to dialed numbers
Caller ID name and number support
FAX support
Blacklist
Firewall
DDNS Support
Backup And Restore
Easy to install

Applications

1) Connect legacy PBX systems to VoIP services
2) Connect legacy PBX systems to remote sites over private VoIP links
3) Connect IP PBX systems to legacy TDM services
4) Phased transition from Legacy PBX to IP PBX



Yeastar TB200 - VoIP BRI Gateway(BRI-VoIP)

Flexible routing for your PBX

Yeastar TB200 - VoIP BRI Gateway(BRI-VoIP) is a full-featured VoIP gateway. It's used to integrate your PABX to
VoIP service or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

TB200正面副本.png


Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

Specification:

BRI Port: Up to 2 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 213x160x44mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install


Yeastar TB400 - VoIP BRI Gateway(BRI-VoIP)

Flexible routing for your PBX

Yeastar TB400 - VoIP BRI Gateway(BRI-VoIP) is a full-featured VoIP gateway. It's used to integrate your PABX to
VoIP service or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

TB400正面副本.png


Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

Specification:

BRI Port: Up to 4 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 213x160x44mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install

Created by: michaelchan, Last modification: Fri 29 of May, 2015 (06:46 UTC)
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