进达电子制品有限公司

&bull; 3&quot; monochrome LCD
&bull; Multi-sip account
&bull; PoE 802.11af
&bull; VLAN, SRTP, HTTPS
&bull; 8 one-touch speed-dial keys
&nbsp;

VoIP Phone

General
-&nbsp;VoIP (SIP) Business Phone
-&nbsp; Maximum of 4 VoIP accounts
-&nbsp; 128 x 64 dot matrix LCD with greyish backlight
-&nbsp; Wideband G.722 Codec
-&nbsp; IPv4 / IPv6
-&nbsp; WAN port and LAN port
-&nbsp; PoE (This option must be specified at the time of order placement)
Server Compliance
-&nbsp; Asterisk / Broadsoft / 3CX / Alcatel / NEC / Avaya / Aastra / Mitel
Call features
-&nbsp; Call Hold
-&nbsp; Call Flash
-&nbsp; Call Forward
-&nbsp; Call Waiting Indication
-&nbsp; Call transfer (blind/attended)
-&nbsp; Caller ID display
- &nbsp;Redial List and redial
-&nbsp; Do not disturb DND
-&nbsp; Mute-&nbsp; Auto-answer
-&nbsp; Auto-complete dialling
-&nbsp; Smart redialling
-&nbsp; 3-way conferencing
-&nbsp; Configurable hotkeys for speedial / voicemail
-&nbsp; Message Waiting Indication (MWI) LED
-&nbsp; Volume control
-&nbsp; Direct IP call without SIP proxy
-&nbsp; Ring tone selection
-&nbsp; Phonebook (15+ entries), Black list
-&nbsp; Call history: dialled/received/missed/forwarded
-&nbsp; LCD menu user interface
-&nbsp; Web Portal for administrators
Physical Features
-&nbsp; 128 x 64 dot matrix display
-&nbsp; Four softkeys
-&nbsp; Eight function keys including one programmable memory button
-&nbsp; LED for MWI
-&nbsp; RJ45 10/100M Ethernet WAN port
n Gigabit Ethernet WAN port (August 2015)
-&nbsp; RJ45 10/100M Ethernet LAN port
n Gigabit Ethernet LAN port (August 2015)
-&nbsp; Power adapter: AC 100~240V input DC 12V/1.2A output (This option must be specified at
the time of order placement)
-&nbsp; Power over Ethernet IEEE 802.3af (This option must be specified at the time of order
placement)
Codecs and Voice Features
-&nbsp; Wideband codec: G.722
-&nbsp; Narrowband codec: G.711&micro;/A, ILBC
-&nbsp; G.729
-&nbsp; VAD, CNG, AEC, PLC, AJB, AGC
-&nbsp; Full-duplex speakerphone with AEC
Network Features
-&nbsp; SIP v1 (RFC2543), v2 (RFC3261)
-&nbsp; IPv4 / IPv6
-&nbsp; VLAN QoS (802.1pq)
-&nbsp; NAT Traversal: STUN mode
-&nbsp; DTMF: In-Band, RFC2833, SIP Info
-&nbsp; Proxy mode and peer-to-peer SIP link mode
-&nbsp; IP Assignment: Static/DHCP/PPPoE
-&nbsp; Bridge/router mode for PC port
-&nbsp; TFTP/DHCP/PPPoE client
-&nbsp; Telnet/HTTP server
-&nbsp; DNS client
-&nbsp; NAT/DHCP server
Management and Provisioning
-&nbsp; F/W provision via TFTP/HTTP
-&nbsp; Auto-provisioning: Option66
-&nbsp; Mass-configuration tool for system integration (check with your local supplier / distributor)
-&nbsp; Telnet

VoIP (SIP) Business Phone

This is a VoIP handset set that comes with one base and six handsets. Using the&nbsp;advanced DECT technology, this handset set supports IPv4 and PoE. This professional VoIP handset set is designed with an LCD screen with backlight for easy reading. 	 For business and home use. 	 DECT, GAP, IPv4 and PoE. 	 1 base&nbsp;and 6 handsets. 	 Colors: white gloss, black gloss, red gloss and&nbsp;yellow gloss. 	 Lighted keypad. 	 Mono LCD with backlight.  

VoIP DECT Handset Set

 	 Maximum of 6 VoIP accounts - IPv4 - WAN port and LAN port - POE - Wideband G.722 Codec 	 Server Compliance Asterisk / Broadsoft / Alcatel / NEC / Avaya / Aastra&nbsp; 	 Dual power supplied Support 1.8GHz (EU) and DECT 6.0 (US) frequency band  

VoIP Dect Phone

General 	 SIP Accounts : 3 	 Wideband G.722 Codec 	 Narrowband Codec : G.729, iLBC 	 IPv4  
Server Compliance 	 Asterisk&nbsp;/&nbsp;Broadsoft&nbsp; /&nbsp;Alcatel&nbsp;/&nbsp;NEC&nbsp;/ Avaya basic call features supported  Call Features 	 Call Hold 	 Call Mute&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 	 Off-hook Emergency Call 	 Headset : RJ9 	 Hands-free  
Physical Features 	 Trimline form factor 	 Magnetic Hook Switch 	 Wall Mountable&nbsp;&nbsp; 	 RJ45 10/100M Ethernet WAN port 	 7 function keys 	 Hotline 	 line keys (can be defined as function keys at the time of purchase) 	 LCD Soft Keys : 7 	 memory button 	 Power over Ethernet 802.11AF 	 LCD Backlint  Codecs and Voice Features 	 Wideband G.722 Codec 	 Narrowband Codec : G.729, iLBC&nbsp;&nbsp; 	 VAD, CNG, AEC, PLC, AJB, AGC&nbsp;&nbsp;&nbsp; 	 Full-duplex speakerphone with AEC  
Other Features 	 Phonebook : CVS, XML, LDAP, VCF 	 Intercom 	 Paging : By KTS-100 Support 	 Call Park : By KTS-100 Support 	 Call Pick up : By KTS-100 Support 	 Follow-me : By KTS-100 Support 	 Do-not-Disturb : Support 	 Redial List 	 System Log 	 Call Log 	 Dist. Ringtone 	 Ringtone : WAV  

Color LCD Business VoIP Phone

风青实业股份有限公司

This SIP phone combines the reliability of traditional telephones with the efficiency of internet-based technology, delivering a modern, flexible, and cost-effective solution for business communication. With minimal hardware and maintenance needs, it lowers long-distance and international call costs while enabling easy scaling by allowing businesses to add or remove users as needed. SIP Phone [Lower Communication Costs] Minimal hardware and maintenance requirements.[Scalable &amp; Efficient] Easily add or remove users as business needs evolve.[Wireless convenience] Perfect for spacious workplaces or settings where traditional wired connections are impractical.

SIP Phone

YEASTAR TECHNOLOGY CO., LIMITED

 Specification 
 Number of Telephony Ports: &nbsp;4
 Phone Interface:&nbsp; 4 x RJ11 FXS ports
 Protocol: &nbsp;SIP (RFC3261), IAX2
 Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP
 DTMF:&nbsp; RFC2833, SIP INFO, In-band
 Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

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

 Size: &nbsp;200 (L) mm &times; 137 (W) mm &times; 25.6 (H) mm
 Power Supply: 12V 1.5A

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

 Features
-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

Analog VoIP Gateway

 Specification 
 Number of Telephony Ports: &nbsp;8
 Phone Interface:&nbsp; 8 x RJ11 FXS ports
 Protocol: &nbsp;SIP (RFC3261), IAX2
 Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP
 DTMF:&nbsp; RFC2833, SIP INFO, In-band
 Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through
 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

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

 Size: &nbsp;200 (L) mm &times; 137 (W) mm &times; 25.6 (H) mm
 Power Supply: 12V 1.5A

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

 Specification 
 Number of Telephony Ports: &nbsp;16
 Phone Interface:&nbsp; 16 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
 Protocol: &nbsp;SIP (RFC3261), IAX2
 Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP
 DTMF:&nbsp; RFC2833, SIP INFO, In-band
 Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

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

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

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 

 Specification 
 Number of Telephony Ports: &nbsp;24
 Phone Interface:&nbsp; 24 x RJ11 FXS ports plus 1 x 50 pin
Telco connector
 Protocol: &nbsp;SIP (RFC3261), IAX2
 Transport Protocol:&nbsp; UDP,TCP, TLS, SRTP
 DTMF:&nbsp; RFC2833, SIP INFO, In-band
 Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
 Echo Cancellation:&nbsp; ITU-T G.168 LEC
 FAX over IP:&nbsp; T.38 and pass-through

 Signaling : Kewl Start, Loop Start
 Caller ID : BELL202, ETSI (V23), NTT (V23-Japan), and
DTMF-based CID
 Disconnect Method : Busy Tone, Polarity Reversal

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

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

 Operation Range: &nbsp;0&deg; to 50&deg;C, 32&deg; to 104&deg; F
 Storage Range: &nbsp;-20&deg; to 65&deg;C, -4&deg; to 149&deg; F
 Humidity: &nbsp;10-90% non-condensing

 Features
3-way Conference 
 Attended Transfer 
 Black List 
 Blind Transfer 
 Call Detail Record (CDR) 
 Caller ID Display or Block 
 Call Forward: No Answer, Busy, All 
 Call Waiting 
 Call Status Display 
 Do Not Disturb 
 IP Blacklist 
 Network Attack Alert 
 Paging 
 RADIUS Accounting &amp; Login 
 Real Open API Protocol (Based on Asterisk) 
 Voice Prompt 
 SIP Server and SIP Trunk supported 
 SIP Peer Mode supported 
 SNMP* 
 TR-069* 
 System Logs 
 Web based configuration 
&nbsp;

Eware Networks Limited

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T21P E2 Entry-Level IP Phone with 2 Lines & HD Voice

CEM Solutions Pvt Ltd.

   Stand Alone, powerful voice gateway supports up to 50 simultaneous calls on PSTN/IP link and 5 VoIP registration &amp; VoIP Trunks. No additional licenses required for codec transcoding or web interface   Our Scalable VoIP PRI Gateways ranging from 1 to 2 PRI ports developed to enable direct connection of VoIP traffic to the PSTN/fixed line ISDN for extending communications and reducing call charges  Key Features 
   E1/T1 PRI Gateway; up to 2 SPAN interface   Built In Firewall   Flexible Call Routing   Transcoding (Voice coding G.711 A-law/u-law, G.722 G.729AB) licenses included.   Fax Pass-through.   Stand Alone with Web User Interface.   128 ms DSP based Echo Cancellation  Additional Features 
   Up to 50 simultaneous VoIP to ISDN PRI calls   2 T1/E1/J1 PRI ports with programmable NET/CPE modes   Support SIP v2 (RFC 3261 complaint)   Register with multiple SIP service providers   ISDN network clock synchronization for error-free communication   Interoperable with all the major-brand soft switches and IP-PBXs   PIN authentication for accessing Trunks to prevent unauthorized usage   Call detail records   Web-based configuration and management  

ISDN PRI VoIP Gateway

&nbsp;Specification
Number of GSM Channels (Max):&nbsp;16
Antenna Splitter:&nbsp;4 in 1
Network Type:&nbsp;850/900/1800/1900MHz&nbsp;
Protocol:&nbsp;SIP (RFC3261), IAX2
Transport Protocol:&nbsp;UDP, TCP, TLS, SRTP
DTMF:&nbsp;RFC2833, SIP INFO, In-band
Code:&nbsp;G.711 a/u-law, G.722, G.726, G.729a, GSM,
ADPCM, Speex
Echo Cancellation:&nbsp;ITU-T G.168 LEC

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

Size:&nbsp;440x280x44 mm
Power Supply:&nbsp;AC 100-240V 50/60Hz

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

VoIP-GSM Gateway

    NeoGate 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&rsquo;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.   Specification:   E1/T1/J1 Port:&nbsp; 2 (Support PRI, MFC R2, SS7)    Protocol: &nbsp;SIP (RFC3261)    Transport Protocol:&nbsp; UDP, TCP, TLS, SRTP    DTMF:&nbsp; RFC2833, SIP INFO, In-band    Codec:&nbsp; G.711 a/u-law, G.722, G.726, G.729 A, GSM, &nbsp; ADPCM, Speex     LAN:&nbsp; 10/100/1000 Mbps     WAN:&nbsp; 10/100/1000 Mbps    Size: &nbsp;213x160x44mm    Power Supply: &nbsp;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  Benefits:   Cost-effective Call Routing   Easy-to-navigate GUI   Simple Installation   Easy to integrate   Low power consumption for your green office  Applications:   Connect legacy PBX systems to VoIP services   Connect legacy PBX systems to remote sites over private VoIP links   Connect IP PBX systems to legacy TDM services   Phased transition from Legacy PBX to IP PBX  

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

   The revolutionary FXS/FXO to VOIP card will be detected as a network card in all Windows and Linux systems. The OS will install the driver (network card) then you can use it with your favorite opensource PBX as a regular VOIP channel. The card itself is configured with a script. Hardware DSP for LEC (carrier grade 128ms) and transcoding.  Additional Features: 
   Compatible with 3CX, Elastix, Asterisk, Trixbox, FreePBX.   Windows/Linux compatible (all versions).   On board 128ms Line Echo Cancellation DSP.   On board Transcoding (G711, G729) DSP.   4 analog ports (FXO or FXS).   T.38 fax relay.   Pre configured for almost plug and play experience.   Clean and easy to understand GUI with many options.   Caller ID.   DTMF Digit Detection and Generation.   SIP User Agent IETF RFC3261 compliant.  

FXS/FXO To VOIP Card - Analog Card

Fanvil Technology Co., Ltd.

The IP phone is a voice communication terminal. It comes with high voice quality. This versatile VoIP phone is perfect for executives, office managers or administrative assistants who make and receive a large amount of calls.
 
  Flexible dial map and&nbsp;hotline. 
  Empty calling reject and&nbsp;black list. 
  Limit call, no-disturb function and&nbsp;caller ID display. 
  Supports phonebook of 500 records. 
  Incoming calls/outgoing calls/missing calls, each supports 100 records. 
  Supports conference and voice record on SIP server. 
  Supports 10 kinds of ring type and 1 user ring. 
  Supports IAX2. 
  Supports message waiting indicator. 
  Supports SMS. 
  Supports speed dial. 
  Supports memo. 
  Multi-language realizes localization. 
 

IP Phone

Huizhou City Huiyang District Meisiqi Industry Development Co., Ltd.

MSQ VoIP phone series are feature-rich, cost-effective SIP phones specially designed for ITSPs, small to medium businesses and residential application. It supports 5 user accounts, auto-provision as well as zero configuration and easy maintenance. It offers high quality audio, a broad range of voice codes and phone features to general level users. 
   Two RJ45 ports with router built-in, one for Internet, one for PC.   One RJ11 port FXO port.   POE (power over Ethernet).  Network protocol: 
   SIP V1(RFC2543), V2(RFC3261).   IP/TCP/UDP/RTP/RTCP.   IP/ICMP/ARP/RARP/SNTP.   TFTP client/DHCP client/PPPoE client.   Telnet/HTTP server.   DNS client.   NAT.  SIP server:
   Registrar server: 5 SIP user accounts.   Outbound proxy mode.   Power over Ethernet (optional).  Code:
   G.711.   G.723.1/iLBC.   G.726.   G.729A.   G.729B adds VAD, CNG to G.729.  Configuration: 
   Web browser.   Console/telnet.   IVR/keypad.   Auto-provision.  IP assignment: 
   Static IP.   DHCP.   PPPoE.  NAT traversal: 
   STUN.   QoS.   ToS field.  Dial method: 
   Direct IP call without SIP proxy.   Dial number via SIP server.   Dial URL via SIP server.  DTMF function: 
   In-band DTMF.   Out-of-band DTMF.   SIP info.  Phone features: 
   Caller ID.   Call hold.   Call waiting.   Call forward.   Speaker phone.   3-way conference.   Redial and flash.   Speed dial.   Phone book.   Volume adjustment.   Ring tone selection.  Voice quality: 
   VAD.   CNG.   LEC.   Packet loss compensation.   Adaptive jitter buffer.  Firmware upgrade: 
   TFTP.   Console.   HTTP.   Auto-provision.  Security: 
   HTTP 1.1 basic/digest authentication for web setup.   MD5 for SIP authentication (RFC 2069/RFC 2617).  Interface: 
   Power input: 100-240V AC, output power: DC-5.5V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.   TEL IN (FXO) RJ11 port.  Certificate:
   CE.   FCC.  

VoIP SIP Phone POE with PSTN 

Suzhou International Science Park

This IP phone can register up to 4 SIP accounts. It features a large LCD screen and functional keys for added performance. This IP phone is convenient to use.
 
  SIP 2.0 (RFC3261). 
  HD voice on all lines. 
  Echo cancellation: G.168 compatible with the LEC, including hands-free (AEC). 
  VGS, VAD, CNG. 
  Speakerphone full-duplex. 
  Phonebook up to 500 entries. 
  Customized logos are welcome. 
 

深圳市沃宇科技有限公司

Asterisk, Elastix, 3CX compatible.
VoIP phone with 3 SIP lines &amp; 1 IAX2 line.
Support VLAN, VPN (L2TP), POE.
RJ9 headset.
2RJ 45 ports.
Large dot matrix LCD display.
 
Basic features: 
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Support IA x 2.   SIP supports 3 SIP servers. Can connect to 3 SIP servers at the same time.   Codec: G.711A/u, G.723, G.729a/b, G.722.   Echo cancellation: G.168 compliance in LEC, additional acoustic echo cancellation.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   SIP support SIP domain, SIP authentication (none, basic, MD5).   DTMF mode: support DTMF_RELAY, SIP_INFO, RFC_2833.   Network features: 
   WAN/LAN: 10/100M Ethernet ports, supports Route and Switch mode.   Support PPPoE for xDSL.   Support basic NAT and NAPT.   NAT transverse: support STUN client.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support main DNS and secondary DNS server.   Support VLAN.   Support DMZ.   Support DNS relay, SNTP client, firewall.   Support VPN (L2TP).   QoS with DiffServ.   Network tools in telnet server: including ping, trace route, telnet client.   Advanced application features: 
   Large dot matrix LCD display and softkeys make user easier to use.   2 headset interface (3.5mm Jack or RJ9 connector).   6 memory keys.   Support POE.   Multi language realizes localization.   SIP application: support call forward/transfer/holding/waiting/conference/.   Call control features: flexible dial map, hotline, empty calling no. reject service.   Incoming calls/outgoing calls/missing calls each support 100 records.   Support phone book 500 records.   Support conference and voice record on SIP server.   Support SMS.   Maintenance and management.   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through POST mode.   Telnet remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware through HTTP, FTP or TFTP).   Support syslog.  

VoIP Phone Compatible with Asterisk

   VI2007 is SIP phone ideal for both the small office and general enterprise users. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.   VI2007 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.   VI2007 supports IAX2 &amp; SIP protocols, support bridge and router model. Support IA x 2 and dual public server, offers two Ethernet interfaces and is compatible with various soft switch systems and VoIP voice gateways to provide broadband IP voice service.   VI2007 supports traditional telephony features such as flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.  Specifications:
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Codec: G.711 A/U Law, G.723.1, G.729a/b, G.722, G.722.1, G.726.   Echo cancellation: support G.168, and hands-free can support 96m, hand free speaker phone.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   NAT transverse: support STUN client.   SIP support SIP domain, SIP authentication (none, basic, MD5), Peer to Peer/IP call.   SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time.   DTMF: support SIP info, DTMF Relay, RFC2833.   SIP application: support call forward/transfer/holding/waiting/3 way talking.   Call control features: flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support SMS.   Support speed dial.  Network features:
   Support basic NAT and NAPT.   Support PPPoE for xDSL.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support VLAN (optional: voice VLAN/data VLAN) QoS with DiffServ.   Support DMZ.   Support main DNS and secondary DNS server.   Support DNS relay, SNTP client, firewall.   Network tools in telnet server: including ping, trace route, telnet client.  Maintenance &amp; management:
   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through HTTP, FTP or TFTP. Telnet.   Remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

The SIP-T46G is Yealink latest revolutionary IP Phone for executive users and busy professionals. New designs appears the commerce, with highresolution TFT color display, delivering a rich visual experience. Yealink Optima HD technology enables rich, clear, life-like voice communications. Supports Gigabit Ethernet, a variety of device connections, including EHS headset and USB. With programmable keys, the IP Phone supports vast productivity-enhancing features.


 Features  	 Revolutionarily new design 	 Yealink Optima HD voice 	 Dual-port Gigabit Ethernet 	 4.3&quot; 480 x 272-pixel color display with backlight 	 Built-in a USB port, support Bluetooth headset (Through USB Dongle) 	 Up to 16 SIP accounts 	 Paper label free design 	 PoE(802.3af) class support 	 Headset, EHS support 	 Supports expansion modules 	 Stand with 2 adjustable angles 	 Wall mountable 	 Simple, flexible and secure provisioning options  
 Specification 

Audio Features 	 HD voice: HD handset, HD speaker 	 Codecs: G.722, G.711(A/&mu;), G.723, G.729AB, G.726, iLBC 	 DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO 	 Full-duplex hands-free speakerphone with AEC 	 VAD, CNG, AEC, PLC, AJB, AGC  
Phone Features 	 16 VoIP accounts 	 Call hold, mute, DND 	 One-touch speed dial, hotline 	 Call forward, call waiting, call transfer 	 Group listening, SMS, emergency call 	 Redial, call return, auto answer 	 3-way conferencing 	 Direct IP call without SIP proxy 	 Ring tone selection/import/delete 	 Set date time manually or automatically 	 Dial plan 	 XML Browser 	 Action URL/URI 	 RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)  

Yealink SIP-T46G Ultra-Elegant Gigabit IP Phone

   VI2006 is IP phone ideal for service provider and small offices. Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.   VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.   VI2006 supports IA x 2 &amp; SIP protocols, support bridge and router model. Support IA x 2 and dual public server, offers two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.   VI2006 supports traditional telephony features such as flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing.  Specifications:
   Support SIP 2.0 (RFC3261) and correlative RFCs.   Codec: G.711 A/U law, G.723.1, G.729a/b, G.722, G.722.1, G.726.   Echo cancellation: support G.168, and hands-free can support 96ms, hand free speaker phone.   Support voice gain setting, VAD, CNG.   Full duplex hands-free speaker phone.   NAT transverse: support STUN client.   SIP support SIP domain, SIP authentication (none, basic, MD5), peer to peer/IP call.   SIP support 2 SIP lines. Can connect to SIP1 and SIP2 server at the same time.   DTMF: support SIP info, DTMF relay, RFC2833.   SIP application: support call forward/transfer/holding/waiting/3 way talking.   Call control features: flexible dial map, hotline, empty calling reject, black list for reject authenticated call, limit call, no disturb, caller ID.   Support phonebook 500 records incoming calls/outgoing calls/missing calls.   Each support 100 records.   Support 10 kind of ring type and 1 user ring.   Support SRTP.   Support SMS.   Support speed dial.  Network features:
   Support basic NAT and NAPT.   Support PPPoE for xDSL.   Support DHCP client on WAN.   Support DHCP server on LAN.   Support VLAN (optional: voice VLAN/data VLAN) QoS with DiffServ.   Support DMZ.   Support main DNS and secondary DNS server.   Support DNS relay, SNTP Client, Firewall.   Network tools in telnet server: Including ping, trace route, telnet client.  Maintenance &amp; management:
   Web, telnet and keypad management.   Management with different account right.   Upgrade firmware through HTTP, FTP or TFTP. Telnet.   Remote management/upload/download setting file.   Safe mode provide reliability.   Support auto provisioning (upgrade firmware or configuration file).   Support syslog.  

Low Cost VoIP Phone in Promotio

VI2006 is ideal for both the small office and general enterprise users.Enterprise SIP phone's new design and enhanced features address the need for an elegant IP handset solution for the executive office at a highly competitive price.


VI2006 is an Internet based voice network phone terminal, which adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


VI2006 supports IAX2&amp;SIP protocols,support Bridge and Router model.Support IAX2 and dual public server, offers Two Ethernet interfaces and is compatible with various softswitch systems and VoIP voice gateways to provide broadband IP voice service.


VI2006 supports traditional telephony features such as Flexible dial map, Hotline, Empty calling reject, Black list for reject authenticated call, limit call, No disturb, Caller ID as well as advanced enterprise features such as 2 independent SIP accounts, 3-way and conferencing. 

Specifications

Support SIP 2.0 (RFC3261) and correlative RFCs
Codec:G.711 A/U Law, G.723.1, G.729a/b,G.722,G.722.1, G.726
Echo cancellation: Support G.168, and Hands-free can support 96ms, Hand free Speaker Phone
Support Voice Gain Setting, VAD, CNG
Full duplex hands-free speakerphone
NAT transverse:support STUN client
SIP support SIP domain, SIP authentication (none,basic, MD5), Peer to Peer/ IP call
SIP support 2 SIP lines. Can connect to SIP1 and SIP2server at the same time
DTMF:Support SIP info, DTMF Relay, RFC2833
SIP application: support Call forward/ transfer/holding/ waiting / 3 way talking
Call control features: Flexible dial map, Hotline,Empty calling reject, Black list for reject authenticatedcall, limit call, No disturb, Caller ID
Support Phonebook 500 recordsIncoming calls / Outgoing calls /Missing calls. 
Each support 100 records
Support 10 kind of ring type and 1 user ring
Support SRTPSupport SMS
Support Speed Dial

Network Features

Support basic NAT and NAPT
Support PPPoE for xDSL
Support DHCP Client on WAN
Support DHCP Server on LAN
Support VLAN (optional: voice vlan/data vlan) QoS with Diff Serv
Support DMZSupport main DNS and secondary DNS server
Support DNS Relay, SNTP Client, Firewall
Network tools in telnet server: Including ping, trace route,telnet client

Maintenance&amp;Management

Web ,telnet and keypad management
Management with different account right
Upgrade firmware through HTTP, FTP or TFTP. Telnet 
Remote management/Upload/download setting file
Safe mode provide reliability
Supoort Auto Provisioning (upgrade firmware orconfiguration file)
Support Syslog

 

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