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Telecommunication

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Telecommunication Suppliers and Manufacturers

Avantec Mfg Ltd

&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

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

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

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 

   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

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

Eware Networks Limited

 	 Features Yealink HD Voice 	 Wall mountable  

Yealink SIP-T23P Enterprise HD IP Phone

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

APN Merchandising Limited

Featured here is a touch screen fixed PSTN and VoIP smart desktop telephone with Android operating system. It can be perfect integrated with mobile phones and offers an easy dialing experience. This smart desktop telephone has a large screen for making video calls and is excellent for conferencing use. 	 Touch screen fixed PSTN and VoIP smart desktop telephone with Android operating system. 	 USB connection for charging, also can connect with computers, support OTG. 	 Support WI-FI, WI-FI reset, WLAN connector. 	 5V USB charging output. 	 Ringer volume control, reset function. 	 Support headset calls and dial and hands-free calls and dial. 	 Hot spot sharing. 	 Perfect integration with mobile phones. 	 Easy dialing experience. 	 Unlimited intelligent application for making calls easily. 	 Directly call via contacts. 	 Contact information inquiry. 	 Easily set the black and white list, automatically identify and mark strange calls and refuse crank calls. 	 One key to import contacts. 	 Show caller ID. 	 Telephone message, call notes, telephone recording. 	 Historical call record. 	 Electronic photo album. 	 Video call. 	 Call recording. 	 Modern and functional design. 	 Great for video conferencing use. 	 Suitable for home and offices.  

Video 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 

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

 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;

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

   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

    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)

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. 
 

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