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Kexint Come-X1e&X1ep Entry Level IP VoIP Phone 2 SIP Lines Ipv6 Ipv4 Fast Ethernet IP Phone

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Min Order	10 Pieces
Availability	In Stock
Shipping From	Guangdong, China
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Shenzhen kexint Technology Co., Ltd

Audited Supplier 2 years

Profile Certified by SGS/BV

Guangdong, China
Masha (Ms.)

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Model NO.:

COME-X1E&X1EP

Type:

IP Phone

Protocol:

PPPoE

Support:

SMS

Number of Channels:

48

Wire or Not:

Wireless

Color:

Black

LCD X1:

128X48 DOT-Matrix Screen(No Backlight)

RJ45 Port X2:

Network X1, PC X1 (Bridged to Network)

Network:

IP

Rj9 Port X2:

Handset X1, Headset X1

Power:

5V/0.6A or Poe (X1ep)

Trademark:

KEXINT Comelink

Transport Package:

Inner Carton+Outer Carton

Origin:

Guangdong, China

HS Code:

8517623300

Product Description

COME-X1E/X1EP

An entry-level IP Phone,

128x48 dot matrix screen(No backlight).

Supports 2 SIP accounts,

PoE power supply (X1EP),

Three-party conference calls and other rich functions can meet different enterprise application scenarios.

And provide a quality user experience

Model	Screen	Network	PoE	Power Adapter
COME-X1E	128x48 Dot-matrix	10/100Mbps	×	√
COME-X1EP	128x48 Dot-matrix	10/100Mbps	√	-

2 SIP lines, 3-way conference Fast Ethernet - 10/100Mbps Ethernet with PoE(X1EP) enabled Opus voice: can handle a variety of audio applications, including IP voice, multiparty conference, etc Economic and Environmental Friendly design with Elegant housing Compatible with major platforms: Asterisk, Broadsoft, 3CX, Metaswitch, Elastix, Avaya etc.

LCD Lens Logo

Web Logo and Model

Packaging (Label and Gift box)

2 SIP Lines

PoE Enable (X1EP)

Dot-matrix Screen(No backlight)

Handset / Hands-free / Headset mode

Desktop Stand

Optional External Power Supply

Physical: 10/100Mbps Ethernet, dual bridged port for PC bypass IP Mode:IPv4/IPv6/IPv4&IPv6 IP Configuration: Static IP / DHCP / PPPoE Network Access Control: 802.1x VPN: OpenVPN (Requires third-party app support) VLAN LLDP CDP QoS

Local Phonebook (500 entries) Remote Phonebook (XML/LDAP, 500 entries) Call logs (In/out/missed, 600 entries) Black/White List Call Filter Voice Message Waiting Indication (VMWI) Programmable Soft keys Network Time Synchronization Support Recording (on Server) Action URL / Active URI UaCSTA

♦ Audio

Wideband ADC/DAC 16KHz Sampling Narrowband Codec: G.711a/u, G.726-32K, G.729A,,iLBC Wideband Codec: G.722, Opus Full-duplex Acoustic Echo Canceller (AEC) Voice Activity Detection (VAD) / Comfort Noise Generation (CNG) / Background Noise Estimation (BNE) / Noise Reduction (NR) Packet Loss Concealment (PLC) Dynamic Adaptive Jitter Buffer DTMF: In-band, Out-of-Band - DTMF-Relay(RFC2833) / SIP INFO.

♦ Deployment & Management

Auto-Provisioning via FTP/TFTP/HTTP/HTTPS/DHCP OPT66/SIP PNP/TR-069 Web Management Portal Web-based Packet Dump Supports Web upgrade Supports factory reset datas.

♦ Call Features

Call out / Answer / Reject Mute / Unmute (Microphone) Call Hold / Resume Call Waiting Intercom Caller ID Display Speed Dial Anonymous Call (Hide Caller ID) Call Forwarding (Always/Busy/No Answer) Call Transfer (Attended/Unattended) Call Parking/Pick-up (Depending on server) Redial Do-Not-Disturb Auto-Answering Voice Message (On server)

3-way Audio Conference Hot Line Hot desking