Electronics

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Electronics Suppliers and Manufacturers

Telecommunication

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Telecommunication Suppliers and Manufacturers

Phones

elec

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Phones Suppliers and Manufacturers

Avantec Mfg Ltd

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

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

Primatronix Nanho Technology Ltd.

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

Shenzhen Chengfenghao Electronics Co., Ltd

10.1' Tablet Phone Android 10.0 PSTN+ 4G+SIP With handset (RJ9) 4G RAM+32G ROM (8G+64G optional) HD quality camera Support WiFi 2.4G /5G Desk / Wall mount Support GMS Support Google play store  

Smart Android Tablet Phone SIP Phone support 4G network

Tecom Co., Ltd.

This is an Internet phone which specializes in making free Skype-to-Skype calls without using a computer. You can use this phone to call landlines and cell phones at great rates. The phonebook can store up to 500 sets of phone numbers. Featured an easy-to-use interface, this Internet phone showcases a user-friendly design.
 
  Automatic signing into Skype account. 
  With call hold/release. 
  With call pickup/park. 
  With caller ID. 
  With call-timer display. 
 

Internet Phone

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. 
 

IP Phone

Shenzhen Taifeng Network Equipment Co Ltd

The WCDMA video phone supports traditional phone calls and video calls enabled by the WCDMA network. It is equipped with a built-in Wi-Fi router that supports Internet browsing. The contemporary design easily matches your home decor.
 
  Supports WCDMA and GSM network with auto switching. 
  Excellent workmanship. 
 

WCDMA Video Phone

Suntra Int'l Development Ltd

 This high-performance SIP phone is widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol. It features&nbsp;multiple lines that register to a single server on a multiple-server mode. It is equipped with an extra-large, easy-to-read LCD display and a headset jack. 
 
  With SIP standardization deployed throughout. 
  Optional feature: PoE. 
 

SIP Telephone

The WCDMA video informational terminal supports both the WCDMA and GSM network with auto switching. The 3.2-inch LCD color screen shows calling information, date and time. This video informational terminal also supports Internet browsing as well as sending and receiving E-mail.
 
  Supports traditional phone call and video call by WCDMA network. 
  Camera function with self timer. 
  Receiving and sending MMS automatically or manually, and the MMS can be edited, forwarded, deleted. 
  Shows the battery power and strength of the signal. 
  Friendly interface using both text and images displaying. 
  Chinese and English menu switching. 
  Big memory of phone book (can storing 500 phone numbers and the numbers can be edited, deleted or sought). 
  Low-power design (connecting to outer power supply or using backup battery). 
  With USB port. 
 

WCDMA Video Information Terminal

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 

SaferZone Co., Ltd.

This Wi-Fi phone series supports SIP standard and IEEE 802.11b/g networks. It provides the best solution for home and enterprises users to enjoy toll-quality voice communication service by using WLAN network or Internet access. You will experience anywhere portability and wonderful affordability. 	 Wi-Fi phone series based on international standards which have adopted as a standard for the wireless AP and SIP system. 	 It supports WEP, Wi-Fi Protected Access (WPA) and WPA2 with pre-shared keys (PSKs) to keep your conversation secure and 802.1X. 	 You can send and receive calls at home, office, caf&eacute;, airport, bus, taxi and any Wi-Fi environments. 	 Standby time: 160 hours, talking time: 9 hours. 	 Outdoor call coverage max.: 280m. 	 Excellent voice quality. 	 User friendly buttons and an intuitive UI. 	 Optimized Wi-Fi phone chipset. 	 Excellent talking time and simple design.  

Security Wi-Fi Phone

   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.   Due toits excellent cost-performance ratio and high performance, it is absolutely a good choice for you.   Two RJ45 ports with router built-in, one for internet, one for PC.   POE (power over Ethernet) optional.  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.   Certificate: CE, FCC  

IP Phone with POE and PSTN Port

Shenzhen Voptel Techonology Co Ltd

   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.  

   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

Guangdong Telepower Communication Technology Co., Ltd.

The 4-line phone billing meter features a large LCD&nbsp;screen to show incoming and outgoing call records and durations. You can input various tariff rates for international, internal, free and forbidden calls. This advanced phone billing meter is ideal for telecommunication service providers and Internet bars.
 
  LCD screen with backlight. 
  Anti-lightning and anti-jamming design. 
  With rechargeable batteries. 
 

4-Line Phone Billing Meter

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.

The IP phone&nbsp;displays incoming and missed calls with date and time. The intuitive control panel ensures easy operation. This IP phone is a great choice for your home and office.

 
  Call log. 
  Hold function. 
  Hot keys. 
 

Eware Networks Limited

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T23P Enterprise HD IP Phone

Auraland Electronics Co Ltd

Feature is a Internet Telephone.   3 VolP accounts, hotline   Call waiting, call transfer, call forward, Hold, mute, flash, auto-answer, redial, 3-way conference, DND, speed dial   Phonebook with 140 groups memory   Volumn adjustment, ring tone selection, tone scheme, System log, DDNS function  

SIP/VoIP Internet Telephone

 Displayed is an IP phone that comes with a user-friendly color screen. Supporting up to three VoIP accounts, this phone has 14 programmable keys and an inclusive HK voice system, including a&nbsp;handset, speaker, codec and wideband codec. With a fast ethernet port, this IP phone is&nbsp;a nice choice for those living far away from one another. 
 
  Enterprise color IP phone. 
  3.5&rdquo; TFT LCD&nbsp;with&nbsp;480 x 320 pixels and&nbsp;262K colors. 
  Dual ports. 
  10/100Mbps ethernet port. 
  3 VoIP accounts. 
  Multiple languages. 
  Wall-mounted. 
  14 programmable keys. 
  Support BLF and BLA. 
  HD voice: HD handset, HD speaker, HD codec and&nbsp;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. 
 

MSQ VoIP gateway series are feature-rich, cost-effective SIP gatway 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.   1FXS Port  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.  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.  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: 100-240VAC, output power: DC-12V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.  Certificate:
   CE.   FCC.  

VoIP SIP Gateway

Neo Telecom Corporation

This wireless phone enables the high-quality VoIP service through a wireless-G network. It boasts unique features, such as 500 phone book memories and&nbsp;50 call histories. This functional cordless phone is perfect for busy families or offices.
 
  Call time display. 
  Dialed number display. 
  Redial function. 
  Call transfer, holding, waiting and forwarding functions. 
  SMS. 
  Language: English and Korean. 
  1.8&quot; color TFT LCD screen. 
  Resolution: 128 x 160. 
 

Wireless VoIP Phone

MSQ VoIP gateway series are feature-rich, cost-effective SIP gatway 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.   2 FXS + 1FXO port.  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.  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.  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-12V, 1A.   WAN RJ45, 10/100M port.   LAN/PC RJ45, 10/100M port.  Certificate:
   CE.   FCC.  

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