Avantec Mfg Ltd

 	 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

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

&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

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

Suntra Int'l Development Ltd


   This is a business class (SIP) IP telephone.  The phone features handsfree dialing.  Alphanumeric LCD  Lan bridge build in 2 port ethernet hub  Phonebook memory dialing  SIP sandardization deployed throughout  
 

Business Class (SIP) IP Telephone


   This is a business class (SIP) IP telephone.  It is equipped with headset jack.  Multiple lines (up to 4 lines)   Register to a single server or multiple service mode   Extra large LCD display (128*64 dot,v/a 74*40)  SIP sandardization deployed throughout  PO (no adaptor required)  
 

Suzhou International Science Park

This outlined IP video phone is designed with advanced features. The large LCD touch screen and keypads ensures enhanced user friendliness. The phonebook can save up to 2,000 records. This IP video phone is ideal for office use.
 
  Telephone features: forward and transfer calls, conference call, paging and intercom, etc. 
  Blacklist, whitelist, DND. 
  Call history: incoming, outgoing and missed (each 1,000 entries). 
  Import and export phonebook. 
  External phone book: XML. 
  Photo frame. 
 

IP Video Phone

Wahkow Science & Technologies Limited

This is an IP phone with an elegant and practical design. It features a black housing with a sophisticated control panel and a 3.5-inch LCD display. Well engineered, this telephone delivers excellent performance with high stability.
 
  Gigabit color. 
  3.5&rdquo; TFT-LCD screen. 
  480 x 320 pixels. 
  262K color dual-port gigabit Ethernet ports. 
  4 VoIP accounts. 
  Multi-language 14 programmable keys. 
  Support BLF/BLA HD voice: HD handset, HD speaker, and HD codec. 
  HD wideband codec: G.722. 
  Call waiting, call transfer, call forward, voicemail, intercom, and SMS LDAP/Xml phonebook. 
  Open VPN. 
  TLS-SRTP Auto-provision via TFTP/FTP/HTTP/HTTPS/PNP. 
 

IP Phone

Shenzhen Voptel Techonology Co Ltd

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

 

Allodotcom Inc.

  - As shown is a telephone adapter.  - The telephone adapter is equipped with a 2x ethernet LAN interface - RJ45.  - Out-of-band DTMF (RFC 2833)   - 2 FXS ports & 1 FXO port  - Adaptive jitter buffer  - SIP Phone Number per FXS port  - User-friendly web panel  - DTMF out-of-band (RFC 2833)  - DHCP server for LAN  - Operating System: Linux 2.4.26  - IP Address Reservation by MAC Address  - Simultaneous active calls on FXS1 and FXS2  - PSTN pass-through  - Firmware upgrade via RFC 1867 upload  - SIP compatible  

VoIP Analog Telephone Adapter

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 

HK Retail Technology Industry Association Ltd/GS1 HK

The wafer UC manager provides call traffic statistics, call billing and QoS analysis functions. It also provides a management platform of the enterprise IP telephone network. The UC manager offers analysis solutions of investment evaluations for the enterprises based on the voice communication network.
 
  High quality. 
  Professional and reliable. 
 

Wafer UC Manager

 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;

 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 

The wafer VoizMate is a daily Cisco IP phone application platform. It combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises including more value-added services. It can satisfy more requirements of corporation for service integration and information bulletin, finally improving the corporate efficiency and competitiveness.
 
  Combines with the Cisco IP telephony solutions to make full use of IP phone communications for enterprises. 
  High quality. 
 

Wafer VoizMate

CEM Solutions Pvt Ltd.

IP telephony applications commonly require the use of multiple voice codecs, used to digitally compress voice signals and save on bandwidth. Voice signals from the PSTN come in the form of the G.711 codec, but the VoIP terminal equipment and networks can support a variety of different voice codecs, such as G.729. The VoIP infrastructure needs the capability to mediate between endpoints supporting different codecs. 

Our Transcoding Cards are designed to handle complex codec translation, using dedicated DSP resources, which would otherwise be processed by host CPU in software. This card greatly reduces the MIPS or CPU consumption, so that it can be used for handling other tasks. 

Our Transcoding Cards are available in PCI and PCI Express form factors, converts simultaneous channels of transcoding from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. 

 Key Features  
   This card can handle 64, 128, 256 or 400 Trascoding Sessions   Supported codecs: G711u/a G.723 G.726 G.722 G729 AB   PCI 2.2/PCIe compliant   Integrates with Open Source platforms such as Asterisk, Freeswitch, Elastix , YATE and more  

Transcoding Card for Asterisk PCI/PCIe

Nanjing Hanlong Technology Co., ltd

 Displayed is an IP phone that comes with a user-friendly graphic LCD screen. With two ethernet ports of different speeds, this telephone can easily help you make internet calls of satisfactory quality.&nbsp;This wall-mounted VoIP phone also has an HD voice system so that you can hear better.&nbsp; 
 
  HD IP phone with 128 x 96 graphic LCD screen. 
  Dual ports: 10Mbps and 100Mbps ethernet ports. 
  3 VoIP accounts. 
  Multiple languages. 
  Wall mounted. 
  14 programmable keys. 
  Support BLF and BLA. 
  HD voice: HD handset, HD speaker, HD codec and HD wideband codec (G.722). 
  Call waiting, call transfer, call forward, voicemail, intercom, SMS LDAP or Xml phonebook. 
  Open VPN and&nbsp;TLS-SRTP. 
  Auto-provision via TFTP, FTP, HTTP, HTTPS and PNP. 
 

   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

Vogtec (H.K.) Co., Limited

VoIP Desktop T12 	 Generous 128x64 pixel back light graphical LCD display or 2.8&#39; TFT color LCD 	 Multi Languages 	 SIP (RFC3261),TCP/IP/UDP, RTP, HTTP, ARP, ICMP, DNS (A record and SRV),DHCP(both client and server), PPPoE, TFTP, NTP 	 NAT traversal (STUN, etc), server fail-over, SIP presence (SIMPLE), and more 	 2 lines indicators with individual SIP account profiles 	 Soft keys, navigation keys and Line Keys can defined by user 	 Call waiting, hold, mute, transfer (blind or attended), forward, 3-way conferencing and more 	 Full duplex speaker phone w/advanced audio echo cancellation 	 Headset jack RJ9 for broad headset compatibility 	 Large phonebook (up to 500 contacts) and call history (up to 100 records) 	 Automated provisioning using TR-069 or encrypted XML configuration file, SRTP and TLS for advanced security protection, 802.1x for media access control 	 And many more enterprise grade features  

VoIP Desktop

 	 Deeply customized Android system with best compatibility, stability, and extensibility 	 2.8&quot; 320 x 240LCD with touch panel options 	 TDD LTE / FDD / LTE, CDMA2000, WCDMA fix line 	 Inbuilt mobile antenna, ext. WiFi&nbsp;antenna 	 Support both voice and data via WiFi hots-pot or WAN port (optional) 	 Specialized for Tel co fix line with network lock / cell lock / pin lock 	 Support both Bluetooth and RJ9 headset 	 VoIP over LTE / WCDMA / CDMA2000 data network 	 FoTA 	 Wire Type: Cordless 	 Voice Mail Function: Yes  

VoIP Over LTE and WiFi 2.4 / 5G Desktop Phone 

   VG1X-8 gateway is a perfect product for the telecom carriers to provide the residential telephony and fax services through the broadband networks such as FTTB, HFC, and ADSL. VG series gateway can be connected to the traditional PBXs to provide the voice solutions for enterprises, government, and schools. VG Series Gateway can also serve as a remote SIP terminal for IP-PBX solutions. VG Series Gateway is powerful hardware platform provides sufficient power for IP packetization, voice compression and echo cancellation even when the voice traffic is at the peak.   VG series SIP gateway is based on the embedded Linux operating system. With its powerful hardware platform and flexible software design VG Series Gateway can meet various voice application requirements.  Key feature:
   Line configuration 4 or 8 FXS analog ports; 4 FXO analog ports, mixed 4FXS/FXO analog port.   Voice processing voice codec: G.711, G.729A, G.723.1, GSM, iLBC.   End user features caller ID, call forwarding, call waiting, hotline, prefix dialing, DND.   Fax processing T.30, T.38 fax relay.   Billing radius interface; polarity reverse.   Configuration methods web based user interface; text based configuration.   Remote provisioning HTTP/web, auto provision, remote configuration, software upgrade, collects log files.  Signaling:
   Off hook detectio FXS loop start, DDI (Hong Kong).   VoIP protocol SIP (RFC3261), MGCP (RFC2705) 3GPP TS 24.228, TS 24.229.   DTMF in-band: RFC2833, INFO (SIP).  Other special feature:
   Auto dial dtMF mode.   Dual telephone number FXS port can have two number for incoming calls (PSTN and extension).   Support primary and backup the gateway can be configured and controlled by multiple soft switches.   Fashion ringing tones Can download and store 20 music files.  Hardware:
   Maximum user line length 3000 meters.   Ringer equivalence number 5 for short loop (300 meters), 3 for medium loop (1500 meters).   Ringing voltage 60V RMS.   Line interface RJ11.   Ethernet interface connector RJ-45 10/100 Base-T.   Internal memory 32MB.   Flash memory 8MB.   Input voltage 110～240V AC, 50～60Hz, input current 1A max.   CPU Motorola Power Quicc 852.   DSP TI 5509A.   Power consumption: 5V/3A DC (A AC/DC convector is provided).   Dimension (H x W x D): 195 x 118 x 24 millimeters.   Weight: 0.65kg.  

8 FXS/FXO Port VoIP Gateway

   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.  

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