ASTRI

iSIP Phone is a SIP-compliant IP phone developed by ASTRI. Equipped with a two-line display, it supports a variety of advanced call features which are suitable for both enterprise and residential use. As a standard-based IP phone, it can interoperate with a number of 3rd party SIP servers and phones.

I. Standard Compliance
Being compliant with the latest Internet standards, iSIP Phone supports the SIP call signaling protocol (RFC 3261) and its extensions. Besides, to tackle with the NAT traversal problem, iSIP Phone is capable of using various solutions such as STUN and rport for connecting to the different NAT networks. iSIP Phone can also interoperate with a number of dominating SIP service providers and products.

II. Toll-Quality Voice
Featuring multiple voice codecs like G.711A/µ, G.729A and G.723.1, iSIP Phone delivers excellent voice quality. Plus, its adaptive jitter buffering and echo cancellation algorithms enable callers to enjoy uninterrupted conversation even when delay and jitter are experienced over the IP network. The fullduplex speakerphone ensures the voice quality is optimal.

III. Advanced Functions
Based on the latest VoIP technology, iSIP Phone supports a variety of call features including caller ID display, call waiting, call transfer, call forwarding, 3-way conferencing and phonebook, which are necessary in residential and business use.

IV. Key Features

•	SIP-Compliant (RFC3261)
•	Standard PBX Functions
•	Two Call Lines
•	3-Way Conferencing
•	Call History
•	Phonebook
•	Two 10/100 Ethernet Ports
•	2 x 20 Character LCD

•	24 Buttons For Keypad & Other Functions
•	4 LEDs For Status Indication
•	Handset
•	Speakerphone
•	NAT Traversal
•	Phone Menu And Web Configurations
•	Multiple User Settings
•	Network Upgrade

Specifications

Call Features

•	2 Call Lines
•	Call Hold
•	Call Waiting
•	Call Transfer (Attended & Unattended)
•	Call Forwarding (Unconditional, Busy & No Answer)
•	3-Way Conferencing
•	Outgoing Call Blocking
•	Call Mute

•	DND
•	In-call DTMF: Inband, RFC2833
•	Caller ID Display
•	Call History
•	Phonebook
•	Redialing
•	Speed Dialing
•	Direct IP Call

Audio Features

•	G.711 A Law and μ Law (64 Kbps)
•	G.723.1 (5.3 Kbps)
•	G.729A (8 Kbps)
•	Voice Activity Detection (VAD)
•	Comfort Noise Generation (CNG)

•	Adaptive Jitter Buffering
•	Acoustic Echo Cancellation (AEC)
•	Volume Control (Ringer, Handset, Speakerphone, Hands-free)

Protocol Standards

•	SIP (RFC3261)
•	SDP (RFC2327)
•	RTP
•	Auto Media Negotiation (RFC3264)
•	SIP REFER (RFC3515)
•	STUN (RFC3489)

•	rport (RFC3581)
•	Digest Authentication (INVITE, BYE)
•	Network Time (NTP)
•	TFTP (Firmware Upgrade)
•	HTTP (For Web Administration)

Configurations

•	Network: Static IP, DHCP, PPPoE
•	Date/Time: Local, NTP (With Daylight Saving & Timezone Settings)
•	Phone Menu Display
•	Remote Web Configuration
•	Web Password Protection

•	Easy Setup Wizard
•	Multiple User Settings
•	SIP Registration Tuning
•	Diagnostic Tools: PING, TRACEROUTE, SIP Log
•	Network Upgrade (TFTP)

Hardware Specifications

•	2 x 20 Character LCD Display
•	24 Buttons For Keypad & Other Functions
•	4 LEDs For Status Indication
•	Handset

•	Speakerphone
•	Hands-free Support
•	Two-port 10/100 Ethernet Switch
•	Power Jack (9V DC @ 500 mA)