Phone Features
- 6 VoIP accounts, hotline, emergency call
- Call waiting, call transfer, call forward
- Hold, mute, flash, auto-answer, redial
- 3-way conference, DND, speed dial
- XML Phonebook search/import/export
- Black list, call history (100 entries)
- Volume adjustment, ring tone selection
- Tone scheme, System log
- Multi-language (more than 20)
- Supports up to 6 expansion modules
IP PBX System Integration
- Dial plan, dial-now
- SMS, Voicemail, MWI
- BLF/BLA, intercom, paging
- Call park, call pickup
- Distinctive ringtone
Codecs and Voice Features
- Wideband codec: G.722
- Narrowband codec: G.711, G.723.1, G.726, G.729AB
- VAD, CNG, AEC, PLC, AJB, AGC
- Full-duplex speakerphone with AEC
Network Features
- SIP v1 (RFC2543), v2 (RFC3261)
- NAT Traversal: STUN mode
- DTMF: In-Band, RFC2833, SIP Info
- Proxy mode and peer-to-peer SIP link mode
- IP Assignment: Static/DHCP/PPPoE
- Bridge/router mode
- TFTP/DHCP/PPPoE client
- Telnet/HTTP/HTTPS server
- DNS client
- NAT/DHCP server
Management
- FTP/TFTP/HTTP/PnP auto-provision
- Configuration: browser/phone/auto-provision
- Direct IP call without SIP proxy
- Dial number via SIP server
- Dial URL via SIP server
Security
- HTTPS (server/client)
- SRTP (RFC3711)
- Transport Layer Security (TLS)
- VLAN (802.1 pq), QoS
- Digest authentication using MD5/MD5-sess
- Secure configuration file via AES encryption
- Phone lock for personal privacy protection
- Admin/user configuration mode
Physical Features
- TI TITAN chipset
- 320x160 graphic LCD with 4-level grayscales
- 48 keys including 16 programmable keys
- 1xRJ9 (4P4C) handset port
- 1xRJ9 (4P4C) headset port
- 2xRJ45 10/100M Ethernet ports
- 1XRJ12 (6P6C) expansion module port
- Power adapter: AC 100~240V input and DC 5V/1.2A output
- Power over Ethernet (IEEE 802.3af)
- Power consumption: 1.6-2.6W
- Net weight: 1.05KG
- Dimension: 273x204x42MM
- Operating humidity: 10~95%
- Storage temperature: up to 60°C
IP (Internet Protocol)
A protocol that specifies the way data is broken into packets and the way those packets are addressed for transmission. Unlike traditional telephone lines which create a straight path from caller to recipient; IP uses the best available path at any given moment to move data packets between point A & point B. The beauty of IP is that the sending end and receiving end communicate success in sending and receiving to insure all data packets are delivered. A really great concept of IP is that of self healing. If a route is broken and the data packets cannot be delivered an alternate route is automatically selected. LCD
The acronym for "Liquid Crystal Display". The most commonly used technology for numeric/text or graphical displays. Park
Similar to hold except the call is held by the Server or KSU and to retrieve the call you need to know the Park Number or have Park buttons on the phone. Typically used in a larger PBX system when a receptionist does not know where to find an individual, in this case the receptionist would Park and then Page the individual. PoE (Power Over Ethernet)
Instead of being powered by a conventional AC plug-in type, the power comes through the ethernet cable connection. PoE reduces the amount of wires in an office. Data and power in a single cord. QOS (Quality of Service)
A feature of routers that prioritize packets for VoIP traffic or other specific applications. This is an essential piece of a VoIP system as the timing of VoIP packets being received is critical to the voice quality. Click here for our preferred routers. Server
The central processor in a network. In our case the server supports the telephony functions in a VoIP network. It manages the connections between the phone lines and the telephones, voicemail, etc. SIP (Session Initiation Protocol)
Internationally recognized IP telephony signaling protocol used for VoIP. This is the most widely used protocol in the market and is a standard meaning anybody who builds a SIP 2.0 compliant product should work with any other SIP 2.0 compliant device. For instance our Talkswitch is compliant so any off the shelf SIP phone will work as an endpoint for a home phone or a remote worker. The trick with SIP 2.0 is that the combined devices will work to the feature set of the lowest common denominator meaning a SIP 2.0 telephone only support 80% of the SIP feature set then even though the telephone system supports a higher set of features, you will only have the base features of the phone. SIP Proxy
Acts as the gatekeeper in a SIP connection. For instance the SIP Proxy for connection to a SIP trunk holds the translation information necessary to make the connection between your telephone system and the SIP trunk provider. Speakerphone
A phone that has a microphone and a speaker that allows you to talk to your caller without lifting the handset. This is best used in private offices and when multi party conferencing is required. Never use a speakerphone in a noisy environment. Transfer
This business feature allows an employee to send a call to another phone without having to announce the call first. This is the typical means for moving a call through a PBX phone system. Voicemail
The business feature that gives every employee a voice mailbox used to take messages while they are busy or away from their office. They typically also feature an Automated Attendant which answers acts as a receptionist to answer company calls and distribute them to the correct individual or department. Voicemail is an essential piece to a professional office, but must be managed to assure proper phone etiquette is maintained. VoIP (Voice over Internet Protocol)
This is simply the carrying of voice via IP meaning a standard voice stream is broken into packets at the transmit end, sent over a standard data network (such as the Internet) and reassembled into a voice stream at the receiving end. VoIP traffic is trickier than normal data traffic because the timing of packets on the receiving end is critical. For more detail see Basics of the Internet.
|
