Posted: Tue Nov 10, 2009 10:17 am Post subject: SIP,IAX2 VOIP Phone,Gateway with lowest price for sale
Hello everybody,
This is carol from YUXIN,We are the manufacturer of VOIP Phone,Gateway,IP Camera in China.Following is our products brief introduction.If you are interested in it.please contact us for more information.E-mail:sales@yntx.com
VOIP Phone YWH201 VOIP Phone
Performance and features
Support SIP,IAX2 protocol
DHCP support for LAN or Cable modem
Audio codec G.711,G.729,GSM,iLBC
Blue back light
CALL WAIT,HOLD,TRANSFER supported
Set by HTTP web browser (IE6.0) or Telnet
Upgrade by FTP,TFTP or HTTP
VAD(Voice active detect)
CNG (Comfort noise generation)
G.168/165 compliant 16ms echo cancellation
Dynamic voice jitter buffer
Network interface:1/2 RJ-45 Ethernet connectors
YWH602 VOIP Phone
Performance and features
Support SIP,IAX2,H323 protocol
NAT, Firewall.
It can register two SIP account and one IAX2 account in the same time.
All there 3 account can be used as called anytime,and for the caller,it can be slelected
by setting different dial prefix.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC. VAD,CNG.
G.711 u/a; G729, G7231 5.3/6.3 audio Codec
G.168 compliant 32ms echo cancellation
Tone generation and Local DTMF re-generation according with ITU-T
E.164 dial plan and customi ed dial rules
Hotline.
Speed Dial
Call Forward, Call Transfer, 3-way conference calls
Record
Support POE [optional]
Caller ID display
DND(Do Not Disturb),Black List,Limit List
Upgrade firmware through FTP, TFTP or HTTP,.
Web management.
Telnet remote management.
adjustable user password and super password
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
YWH202 VOIP Phone
Performance and features
Support SIP,IAX2,H323 protocol
NAT, Firewall.
It can register two SIP account and one IAX2 account in the same time.
All there 3 account can be used as called anytime,and for the caller,it can be slelected
by setting different dial prefix.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC. VAD,CNG.
G.711 u/a; G729, G7231 5.3/6.3 audio Codec
G.168 compliant 32ms echo cancellation
Tone generation and Local DTMF re-generation according with ITU-T
E.164 dial plan and customi ed dial rules
Hotline.
Speed Dial
Call Forward, Call Transfer, 3-way conference calls
Record
Support POE [optional]
Caller ID display
DND(Do Not Disturb),Black List,Limit List
Upgrade firmware through FTP, TFTP or HTTP,.
Web management.
Telnet remote management.
adjustable user password and super password
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
YWH600 VOIP Phone
Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP
IP/ICMP/ARP/RARP/SNTP
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
2RJ45 Ports,Built-in Router,conference.Auto-provision or updating by HTTP,FTP or TFTP.
For each YWH600A,It can be register 3-SIP account,all can used as callee at anytime.It
also can be slected among these 3-SIP account as caller by dial different dial prefix.
For each YWH600B,it can have 3-SIP account and one PSTN phone number,that means each phone
own 4 phone numbers,all can be used as callee at anytime.Fo the caller,these 4 account can be
selected by dial different relevant prefix(including switch to PSTN as an ordinary PSTN phone)
YWH600B has a real FXO port to support router call from PSTN to VOIP or VOIP to PSTN.
YWH300C VOIP Phone
Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP/CMP/ARP/RARP/SNTP
2RJ45 Ports,Built-in Router,conference,3 SIP account,Auto-Provison or updating by HTTP,FTP
or TFTP.
It can be register 3-SIP account in the same time and can be selected by setting different
dail prefix to using the different 3 account.
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
VOIP Gateway/ATA YGW30 1FXS,1FXO SIP ATA
Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP
IP/ICMP/ARP/RARP/SNTP
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
2RJ45 Ports,Built-in Router,conference,Auto-provison or updating by HTTP,FTP or TFTP.
For each YGW30A,it can have 3-SIP account,all can be used as caller at anytime.It can be
selected among these 3-SIP account as caller by dial different dial prefix.It can be used as
PSTN ordinary phone when power off.
For each YGW30B,it can have 3-SIP account and one PSTN phone number,that means each ATA
own 4 phone numbers,all can be used as callee at anytime.For the caller,these 4 account can be
selected by dial different relevant prefix(including switch to PSTN as an ordinary PSTN phone)
YGW30B has a real FXO port to support router call from PSTN to VOIP or from VOIP to PSTN.
YGW50 SIP,IAX2 1FXS ATA
Performance and features
SIP and IAX2
Support Bridge and Router(NAT&NAPT) models.
TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS
Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
Voice Codecs: G.711(A-law/U-law),G.729A/B,ILBC
NAT penetration: STUN client, AVS and Citron etc. Can modify SIP register port,
HTTP server port, Telnet server port and RTP port.
Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold,
Transfer, Do-Not-disturb, Forward, in-band and out-of-band
DTMF, Hotline (off hook autodial), ban outgoing.
Support standard encryption and authentication (DIGEST using MD5, MD5- sess).
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Provide easy configuration thru manual operation (Web interface and Telnet) or automated
centralized configuration file via TFTP or HTTP.
Support firmware upgrade via TFTP/FTP and HTTP.
Support syslog, can send event of gateway to syslog server.
It have 2-SIP account,one IAX2 account and one PSTN phone number,that means each ATA own 4
phone number,all can be used as callee at anytime.
For the caller,these 4 account can be selected by dial different relevant prefix(including
switch to PSTN as an ordinary PSTN phone),It can be used as PSTN phone when power off.
2FXS YGW60 SIP ATA
Performance and features
Support SIP protocol
Support Bridge and Router(NAT&NAPT) models.
TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS
Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
Voice Codecs: G.711(A-law/U-law), G.723.1, G.729A/B,G.726,ILBC
NAT penetration: STUN client, AVS and Citron etc. Can modify SIP register port, HTTP
server port, Telnet server port and RTP port.
Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold,
Transfer, Do-Not-disturb, Forward, in-band and out-of-band
DTMF, Hotline (off hook autodial), ban outgoing.
Support standard encryption and authentication (DIGEST using MD5, MD5- sess).
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Provide easy configuration thru manual operation (Web interface and Telnet) or automated
centralized configuration file via TFTP or HTTP.
Support firmware upgrade via TFTP/FTP and HTTP.
Support syslog, can send event of gateway to syslog server.
Each FXS port have one SIP account and one PSTN phone number.
It can use the same SIP account for 2FXS port,but at that time,only one of the port can be
used as callee,however,both port can be used as caller simultaneously.
But when they used the different SIP account,it can be used as caller or callee anytime.
For the lifeline port,the default is 1FXS port to be used as PSTN phone when power off.
You can select any FXS port to be switched to PSTN as caller through lifeline port by dial
different dial prefix.
4FXS YGW80 SIP ATA
Performance and features
Support SIP
Support Bridge and Router(NAT&NAPT) models.
TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS
Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
Voice Codecs: G.711,G.729A/B
NAT penetration: STUN client, AVS and Citron etc. Can modify SIP register port, HTTP
server port, Telnet server port and RTP port.
Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold,
ban outgoing.
Support standard encryption and authentication (DIGEST using MD5, MD5- sess).
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Provide easy configuration thru manual operation (Web interface and Telnet) or automated
centralized configuration file via TFTP or HTTP.
Support firmware upgrade via TFTP/FTP and HTTP.
Support syslog, can send event of gateway to syslog server.
Each FXS port can have one SIP account.It can be use the same SIP account for 4 FXS
ports,but at that time,only one of the port can be used as callee,however,all port can be used
as caller simultaneously.
But when they used the different SIP account,it can be used as caller or callee anytime.
Contact Info:
Carol Chang
Sales Manager
ZHENGZHOU YUNENG COMMUNICATION CO.,LTD(YUXIN)
URL:http://www.yntx.com
E-mail:sales@yntx.com ync@vip.163.com
Tel:86-371 67657240 Mobile:86-15803816798
MSN:carol_chang8009@hotmail.com
Last edited by yuxin on Mon Aug 16, 2010 8:58 am; edited 3 times in total
Posted: Mon Apr 26, 2010 7:33 am Post subject: YWH300C also called IP956
YWH300C VOIP Phone
• Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP/CMP/ARP/RARP/SNTP
2RJ45 Ports,Built-in Router,conference,3 SIP account,Auto-Provison or updating by HTTP,FTP or TFTP.
It can be register 3-SIP account in the same time and can be selected by setting different dail prefix to using the different 3 account.
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
• Main technical index
Main chip: 32-bit RISC CPU with 125MHz clock rate
Data storage:2MB SDRAM
Program memory: 1MB Flash memory
Application Network environment: Two 10/100Mbps Fast Ethernet MAC
Echo cancellation:G165 16ms
Store quick dial number: 100
Record phone number of missed call:80
Power loss:2.7W(max)
Power adapter: input AC 220V,output DC 9V 500mA
Employing condition:
Ambience temperature 0-40℃（32-104°F）
Relative humidity 10-95%
Atmosphere pressure 86-106Kpa
Overall size:215×190×70mm(L×W×H)
YWH202 VOIP Phone
• Performance and features
Support SIP,IAX2,H323 protocol
NAT, Firewall.
It can register two SIP account and one IAX2 account in the same time.
All there 3 account can be used as called anytime,and for the caller,it can be slelected by setting different dial prefix.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC. VAD,CNG.
G.711 u/a; G729, G7231 5.3/6.3 audio Codec
G.168 compliant 32ms echo cancellation
Tone generation and Local DTMF re-generation according with ITU-T
E.164 dial plan and customi ed dial rules
Hotline.
Speed Dial
Call Forward, Call Transfer, 3-way conference calls
Record
Support POE [optional]
Caller ID display
DND(Do Not Disturb),Black List,Limit List
Upgrade firmware through FTP, TFTP or HTTP,.
Web management.
Telnet remote management.
adjustable user password and super password
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
• Main technical index
Main chip: 150MHZ
DSP: 100MHZ
Data storage: 16MB SDRAM
Program memory: 2MB Flash memory
Application Network environment: 10 Base-T/100 Base-T
Echo cancellation:G165 16ms
Power loss: 2.7 W (max)
Power adapter: input AC 220V,output DC 9V 500mA
Employing condition:
Ambience temperature 0-40℃（32-104°F）
Relative humidity 10-95%
Atmosphere pressure 86-106Kpa
Overall size:215×190×70mm(L×W×H)
Performance and features
Support SIP,IAX2,H323 protocol
NAT, Firewall.
It can register two SIP account and one IAX2 account in the same time.
All there 3 account can be used as called anytime,and for the caller,it can be slelected by setting different dial prefix.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC. VAD,CNG.
G.711 u/a; G729, G7231 5.3/6.3 audio Codec
G.168 compliant 32ms echo cancellation
Tone generation and Local DTMF re-generation according with ITU-T
E.164 dial plan and customi ed dial rules
Hotline.
Speed Dial
Call Forward, Call Transfer, 3-way conference calls
Record
Support POE [optional]
Caller ID display
DND(Do Not Disturb),Black List,Limit List
Upgrade firmware through FTP, TFTP or HTTP,.
Web management.
Telnet remote management.
adjustable user password and super password
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
YWH300C VOIP Phone
Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP/CMP/ARP/RARP/SNTP
2RJ45 Ports,Built-in Router,conference,3 SIP account,Auto-Provison or updating by HTTP,FTP or TFTP.
It can be register 3-SIP account in the same time and can be selected by setting different dail prefix to using the different 3 account.
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
Posted: Wed Jun 09, 2010 9:40 am Post subject: YWH602 VOIP Phone
YWH602 VOIP Phone
Performance and features
Support SIP,IAX2,H323 protocol
NAT, Firewall.
It can register two SIP account and one IAX2 account in the same time.
All there 3 account can be used as called anytime,and for the caller,it can be slelected by setting different dial prefix.
DHCP client and server.
Support PPPoE, (used for ADSL, cable modem connecting).
Support major G7.xxx CODEC. VAD,CNG.
G.711 u/a; G729, G7231 5.3/6.3 audio Codec
G.168 compliant 32ms echo cancellation
Tone generation and Local DTMF re-generation according with ITU-T
E.164 dial plan and customi ed dial rules
Hotline.
Speed Dial
Call Forward, Call Transfer, 3-way conference calls
Record
Support POE [optional]
Caller ID display
DND(Do Not Disturb),Black List,Limit List
Upgrade firmware through FTP, TFTP or HTTP,.
Web management.
Telnet remote management.
adjustable user password and super password
IEEE 802.3 /802.3 u 10 Base T / 100Base TX
Posted: Tue Jun 22, 2010 9:02 am Post subject: YWH2010 SIP Phone
YWH2010 VOIP Phone
Key Features
Support 5 navigation/menu/volume keys, 13 dedicated function keys for: HOLD, SPEAKERPHONE, SEND/REDIAL, MUTE, TRANSFER, CONFERENCE, and MESSAGE(with message indicator),LINE1,LINE2,DND,PLAY,PHONE BOOK,ANSWER ON.
Support SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, ICMP, ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN, UPNP, etc.
Support various codecs including G.711 (PCM a-law and u-law), G.723.1, G.722, G.729A, G.726and iLbc(Pending)
Support standard encryption and authentication (DIGEST using MD5, MD5-sess)
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Support Silence Suppression, VAD, CNG, G.168, and AGC
Support automated provisioning for mass deployment ,RTP and TLS (pending)for security protection
Support automated NAT traversal without manual manipulation of firewall/NAT
Support numeric Caller ID Display, Hold, Transfer, Forward, 3-way Conference, in-band and out-of-band DTMF, Call Waiting, Call Log, Off-hook Audio Dial, Audio Answer, Downloadable Ringtones, Sever Redundancy and Fail-over Support.
Support syslog, full duplex hands-free speakerphone with advanced acoustic echo cancellation, redial, volume control, voice mail with indicator, downloadable ring tones.
YWH800A
http://www.yntx.com/english/products/show2.asp?ID=149
Key Features:
Support 5 navigation/menu/volume keys, 13 dedicated function keys for: HOLD, SPEAKERPHONE, MUTE, TRANSFER, CONFERENCE,and MESSAGE(with message indicator),6 speedkeys.
Support SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, ICMP,ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN, UPNP, etc.
Support various codecs including G.711 (PCM a-law and ulaw),G.723.1 , G.729A, G.726.
Support standard encryption and authentication (DIGEST usingMD5, MD5-sess)
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS,DiffServ, MPLS)
Support Silence Suppression, VAD, CNG, G.168, and AGC
Support automated provisioning for mass deployment ,RTP and TLS(pending)for security protection
Support automated NAT traversal without manual manipulation of firewall/NAT
Support numeric Caller ID Display, Hold, Transfer, Forward, 3-way Conference, in-band and out-of-band DTMF, Call Waiting, Call Log, Off-hook Auto Dial, Auto Answer, Downloadable Ringtones,SMS(these function need SIP server supported).
Support direct IP call. Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy.
Support syslog, full duplex hands-free speakerphone with advanced acoustic echo cancellation, redial, volume control, voice mail with indicator, downloadable ring tones.
Feature Highlights:
1 SIP accounts and 6 speed keys
80*39mm LCD with blacklight, support 4 lines with 16 characters every line
Support Download Phone Book(XML, up to 120 items)
dual switched audio-sensing 10/100Mbps network ports
Support headset jack
Key Features
Support 5 navigation/menu/volume keys, 13 dedicated function keys for: HOLD, SPEAKERPHONE, SEND/REDIAL, MUTE, TRANSFER,CONFERENCE, and MESSAGE(with message indicator),LINE1,LINE2,DND,INTERCOM,PHONE BOOK,PICK UP.
Support SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, ICMP,ARP/RARP, DNS, DHCP, NTP, PPPoE, STUN, UPNP, etc.
Support various codecs including G.711 (PCM a-law and ulaw),G.723.1 , G.729A, G.726.
Support standard encryption and authentication (DIGEST using MD5, MD5-sess)
Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
Support Silence Suppression, VAD, CNG, G.168, and AGC
Support automated provisioning for mass deployment ,RTP and TLS(pending)for security protection
Support automated NAT traversal without manual manipulation of firewall/NAT
Support numeric Caller ID Display, Hold, Transfer, Forward, 3-way Conference, in-band and out-of-band DTMF, Call Waiting, Call Log, Off-hook Auto Dial, Auto Answer, Downloadable Ringtones, and Intercom, paging,pick up, SMS(these function need SIP server supported).
Support direct IP call. Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy.
Support syslog, full duplex hands-free speakerphone with advanced acoustic echo cancellation, redial, volume control, voice mail with indicator, downloadable ring tones.
Feature Highlights
2 line appearances with LED and 2 independent SIP accounts
80*39mm LCD with blacklight,support 4 lines with 16 characters every line
Support Download Phone Book(XML, up to 120 items)
dual switched audio-sensing 10/100Mbps network ports with integrated POE(option)
Support headset jack
Contact Info:
Carol Chang
Sales Manager
ZHENGZHOU YUNENG COMMUNICATION CO.,LTD(YUXIN)
URL:http://www.yntx.com
E-mail:sales@yntx.com ync@vip.163.com
Tel:86-371 67657240 Mobile:86-15803816798
MSN:carol_chang8009@hotmail.com
Posted: Mon Aug 09, 2010 3:52 am Post subject: 4FXS YGW80 SIP ATA
4FXS YGW80 SIP ATA
[img]http://www.yntx.com/english/products/show2.asp?id=119[/img]
Performance and features
Support SIP
Support Bridge and Router(NAT&NAPT) models.
TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
Voice Codecs: G.711,G.729A/B
NAT penetration: STUN client, AVS and Citron etc. Can modify SIP register port, HTTP server port, Telnet server port and RTP port.
Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold, Transfer, Do-Not-disturb, Forward, in-band and out-of-band DTMF, Hotline (off hook autodial), ban outgoing.
Support standard encryption and authentication (DIGEST using MD5, MD5- sess).
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Provide easy configuration thru manual operation (Web interface and Telnet) or automated centralized configuration file via TFTP or HTTP.
Support firmware upgrade via TFTP/FTP and HTTP.
Support syslog, can send event of gateway to syslog server.
Each FXS port can have one SIP account.It can be use the same SIP account for 4 FXS ports,but at that time,only one of the port can be used as callee,however,all port can be used as caller simultaneously.
But when they used the different SIP account,it can be used as caller or callee anytime.
[/img]
YGW30 1FXS,1FXO SIP ATA
Performance and features
SIP v1 (RFC2543), v2(RFC3261)
Support Route,Two 10/100Mbps MACs
Support T.38(Doing)
IP/TCP/UDP/RTP/RTCP
IP/ICMP/ARP/RARP/SNTP
TFTP Client/DHCP Client/ PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
Support G.723.1 5.3k/6.3k，G.729，G.711 A-Law、μ-Law audio codec algorithm
Dynamic voice detection Echo cancellation Comfort noise generation
Tone generation and Local DTMF generation and detection according with ITU-T
Settings by HTTP web browser (IE6.0)
Advanced settings by Telnet
Voice prompt
Upgrade by TFTP
2RJ45 Ports,Built-in Router,conference,Auto-provison or updating by HTTP,FTP or TFTP.
For each YGW30A,it can have 3-SIP account,all can be used as caller at anytime.It can be
selected among these 3-SIP account as caller by dial different dial prefix.It can be used as
PSTN ordinary phone when power off.
For each YGW30B,it can have 3-SIP account and one PSTN phone number,that means each ATA
own 4 phone numbers,all can be used as callee at anytime.For the caller,these 4 account can be
selected by dial different relevant prefix(including switch to PSTN as an ordinary PSTN phone)
YGW30B has a real FXO port to support router call from PSTN to VOIP or from VOIP to PSTN.
Performance and features
SIP and IAX2
Support Bridge and Router(NAT&NAPT) models.
TCP/UDP/IP, ICMP, HTTP, DHCP Client (WAN Interface), DHCP Server(LAN Interface), DNS
Client, DNS Relay, SNTP, PPPoE, FTP, TFTP.
Voice Codecs: G.711(A-law/U-law),G.729A/B,ILBC
NAT penetration: STUN client, AVS and Citron etc. Can modify SIP register port,
HTTP server port, Telnet server port and RTP port.
Support standard voice features such as numeric Caller ID Display, Call Waiting, Hold,
Transfer, Do-Not-disturb, Forward, in-band and out-of-band
DTMF, Hotline (off hook autodial), ban outgoing.
Support standard encryption and authentication (DIGEST using MD5, MD5- sess).
Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise
Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control).
Provide easy configuration thru manual operation (Web interface and Telnet) or automated
centralized configuration file via TFTP or HTTP.
Support firmware upgrade via TFTP/FTP and HTTP.
Support syslog, can send event of gateway to syslog server.
It have 2-SIP account,one IAX2 account and one PSTN phone number,that means each ATA own 4
phone number,all can be used as callee at anytime.
For the caller,these 4 account can be selected by dial different relevant prefix(including
switch to PSTN as an ordinary PSTN phone),It can be used as PSTN phone when power off.
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