Single port with PSTNi VoIPi gateway supporting SIP/IAX2 protocol, support VPN, auto provisioning, VLAN. Via enterprise VPN and IP private line,it supports simultaneous access of IP audio and date service.Besides,it is gateway with PSTN suitable for VOIP service provider, can work with Asterisk IP PBXi perfectly.
Support SIP (RFC3261, RFC2543) Support IAX2 Support Reverse polarity Support voice code: G711A/u, G729, G726, iLBCi Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker Support Jitter Buffer, VAD, CNG, SIP/IAX2, domain name register, point to point communication. Support the voice communication through RTPi and RTCPi. Support the DTMFi Inbound/Outbound transmission; SIP info, DTMF Relay, RFC2833 Support standards of the ring in different countries and regions. NATi penetration, support STUN client, CITRON, AVS etc.. Support SIP domain, SIP Authentication (none basic, MD5), domain name parse. Support two SIP servers (Public Server / Private Server) synchronously, make a call by either proxy. Support SIP application, including SIP Call forward/transfer/holding/waiting. Support L2TP protocol
Support two models: Bridge and Router, and integrates the router functions of these two ports. Support basic NAT and NAPTi Support PPPoE for xDSL, and support redial automatically when its offline Support DHCP client for WAN port Support DHCP server for LAN port Support DNS relay for LAN port, and provide DNS service for LAN network devices. Use the advanced technology DSP to ensure high quality voice Use the advanced buffer technology to avoid the information package delay too long or lost. Support Network Tools that includes ping, trace route and telnet client Support 3 methods to configure the WAN IP: Static (static configuration for LAN), DHCP (Dynamic query through LAN) and PPPoE (Dynamic query through ADSLi) Provide firewall control for small size of LAN port Provide the communication PRI available for the small LAN of LAN port Support Qos (802.1p) for the second Layer
It can register to two SIP server synchronously,one IAX2 systems and a PSTN number, it has 3 VOIP number and 1 PSTN number, that means one phone has 4 numbers Call waiting, call transfer, 3 ways calling, many modes call forward Switch outgoing call freely between VOIP and PSTN Incoming display, forbid call out; setup avoid-disturb, auto answer, off hook auto dial, quick dial Setup blacklist number and limit number Support point-to-point call Setup the end number method Setup fixed call method Support Silence Suppression, VAD (Voice Activity Detection) Support CNG (Comfort Noise Generation) Support Echo Cancellation and AGCi (Automatic Gain Control) Support Polarity reversal Support VPN (L2TP), can be used in the VPN special network Support SIP register failure detection, offer busy tone, when the telephone is out of order Support line detection, busy tone when no line available
Configuration, management and maintenance:
Support post mode and can update gateway via post mode Support to operate and configure the gateway by keyboard and Http mode, and support the filtration restriction on user IP address Support to update the software and configuration via HTTP, FTP, TFTP Support multi-administrator management, user name, password; Support reversal Telnet through NAT/ Firewall to manage long-distance.
A newly developed hi-tech SIP phone for enterprise supporting 5 SIP servers synchronously,(POE) function and 128*64 LCD display.Besides, it supports 3 interactive soft key to play humanized operating prompt and an option of a pair of headsets for private calls.
Support SIP (RFC3261, RFC3262, RFC2543). Support Voice codec: G711A/u, G729, and G723.1 Support G.168 echo cancellation standard, compliant 96ms with speaker mode. Support Jitter Buffer, VAD, CNG, SIP Domain name register, point-to-point Call Support RTPi and RTCPi Support the Inbound/Outbound transmission; SIP info, DTMFi Relay, RFC2833 Support many countries' standard ring NATi transversal: Support STUN, CITRON, AVS Mode Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse Support 5 SIP servers synchronously, can call in and out by either proxy Support SIP application, including SIP call forward/transfer/holding/waiting
Support two models: Bridge and Router, integrate two ports router function. Support basic NAT and NAPTi. Support PPPoE for xDSL, and support off hook auto dial. Support DHCP Client for WAN; Support DHCP server for LAN; Support DNS relay for LAN and can provide DNS service for LAN Network equipment. Use advanced DSP tech to insure high quality voice Use advanced jitter buffer tech to prevent the delaying and losing for package information Support Network Tools, including ping, race route, and telnet client. Support three modes to configure WAN port IP, they are: static, DHCP, and PPPoE. Provide firewall control for small LAN. Provide optional communication priority level for small LAN. Support Secondly Layer QoS(802.1p) Support 12tp
Support headset Support 128*64 LCD Support Power over Ethernet (POE) function Support 5 SIP servers synchronously. 3 Interactive soft key to play humanized operating prompt. Support local voice record, message and server message record. Support message wait indication. Support user defined ring tone. Support L2TP client. Support call pickup, join call, auto-redial. Support 5 programmable keys, 5 PSTNi keys and 5 SIP keys, and it can be connected with the expansion board which can display more numbers' online status. Support dial switchboard and extension number at one time to get through extension directly. Support call list, and can set different rings according to different incoming callers. Dial waiting, call transfer, three ways call, and multi-dial forward Caller ID display, ban calling out, setting no-disturb, dial number automatically while picking up the telephone, set VIP numbers. Set the black name list and confine numbers Support point-point calling directly. Support flexible methods of receiving numbers. Support silence suppression and silence detection. Support noise background simulation. Support echoes suppression and auto gain.
An award-winning next generation IP phone supporting SIP/IAX2 and online status display.Featuring for VPN,VLAN,superb sound quality and rich functionalities at ultra-affordable price,it is a special phone for IP PBXi,can display presence status.
SIP (RFC3261, RFC2543) Support IAX2. Support codec: G.711A/u, G.723 high/low, G.729i A/B Support G.168 echo cancellation standard, compliant 96ms echo cancellation with speaker mode Support voice volume adjustment, including IN/OUT of handset and speaker Support Jitter Buffer, VAD, CNG, SIP, Domain name register, point-to-point Call Support RTPi and RTCPi Support the Inbound/Outbound transmission;SIP info,DTMFi Relay,RFC2833 Support many countries' standard ring Support NATi: Support STUN, CITRON, AVS Mode Support SIP domain, SIP Authentication (none, basic, MD5), Domain Name parse Support two SIP server synchronously, including Pubic Server/ Private server,can make a call by any proxy. You can back-up and select any above SIP server. Support SIP application, including SIP Call forward/transfer/holding/conference/pickup/redial/unredial/joincall Support BLF, server presence and Peer to peer presence negotiation Support VPN (L2TP) clients
Support two models: Bridge and Router, integrate two ports router function. Support basic NAT and NAPTi Support PPPoE for xDSL, and support auto redial when disconnect Support DHCP Client for WAN Support DHCP server for LAN Support DNS relay for LAN and provide DNS service for LAN Network equipment Support DNS domain name resolution in WAN port Support SNTP Client to get time from internet Support advanced DSP tech to ensure high quality voice Support advanced jitter buffer tech to prevent the info package delaying and losing Support network tool: ping，trace route，telnet client Support three modes to configure WAN port IP, they are: static, DHCP, and PPPoE Provide firewall for small LAN Provide optional priority level for small LAN Support second layer QoS (802.1p) Support VLAN Support VPN, L2TP protocol. (New hardware support openVPN)
Special and Advanced Function:
With IP PBX,support 10 group quick dial number, together with IP PBX presence subscribe, the IP phone can display directly the online state of the booking numbers by the indicator. If the indicator is green, means it is online; the indicator is green and twinkle, means it is in the course of the call; if red, means it is offline. Support the local voice message, play the message by one key, IVRi personality record the message, voice prompt. Caller ID display, ban calling out, avoid-disturb setting, auto-answering, auto dial while picking up the telephone, quick dial; Call waiting, call transfer, three ways call, and multi-dial forward Setting the black list and limit numbers Support point to point call Setting the ended number methods Setting the ended number add, delete and substitution Setting the fixed calling ways Support phone number Support Silence Suppression, VAD (Voice Activity Detection) Support CNG (Comfort Noise Generation) Support Echo Suppression and AGCi (Automatic Gain Control) Support DIGEST validate and MD5/MD5-sess encapsulation Support local/server Message-Waiting Indication Support auto long-distance configuration and edition auto-upgrade. Support call record checking and management. Support call pickup Support join Call Support redial and unredial Support directly dial IP+port to call SIP terminal device
The first GSMi / VOIP WIFI Mobile in the world. It's the first mobile wherein the GSM and VOIP via WIFI is working on parallel. It's basically a complete laptop in a mobile. NS Telecom, a manufacturing company in the Philippines, has developed a work on-parallel GSM / VOWNET (Voice Over Wireless Network) mobile device, which is known to be the first in the world. With free NS to NS, NS to Gtalk, US, Canada, Singapore, Hongkong, UK call anywhere in the world!
A 4-Part Step-by-Step Guide Integrating Openser/Asterisk for Beginners
This Step-by-Step Guide is intended for beginners who need assistance in setting up Bind 9.x.x, Installing and configuring an X100P card (easily upgradeable to TDM400P), Linux newbies, using the Linux syntax, configuring and setting up ATA units and sip hardphones, and configuring and integrating Openser/Asterisk - guide has complete configuration files, including connecting to the PSTNi, intro to sip debugging, etc.