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厦门亿联 Yealink SIP-T9CM 普通级IP网络电话机



厦门亿联 Yealink SIP-T9CM 普通级IP网络电话机

产品编号: SIP-78
厂商品牌: Yealink
计量单位:
产品简介: 零配置,即插即用,自動更新,高音質通話 ;Web 集中管理;LCD藍光大螢幕顯示;支持语言设置。
货存状况: 现货供应
本店价格: 455.00
积分点数: 4

购买 收藏
企业客户、团购批发及大宗订单折扣,请拨打021-61021281向王先生咨询。

详细介绍

  产品名称 普通级网络电话  
  产品型号 SIP-T9CM

* 采用CM語音晶片。
* 支持3 SIP帳號,可註冊到3個SIP伺服器。
* 2X16 文字型 LCD螢幕。
* 支持语言设置,短信息(SMS),语言信箱,线路供电。
* 支持远程维护。
* 零配置:支持电话自动配置,软件升级。


IP Phone
MODEL: SIP-T9CM

 2x16 characters LCD with backlight.
 Voice codec: G.711, G.729AB, G.726, G.723.1 and GSM.
 Supports SIP 2.0 and NAT transverse: STUN mode.
 2xRJ45 ports with router built-in, one for internet, one for PC.

Product Description:

Yealink SIP-T9CM is a feature-rich, cost-effective SIP phone specially designed for ITSPs, small to medium businesses and residential application. It supports 3 user accounts, auto-provision and PoE as well as zero configuration and easy maintenance. As the basic model of the Yealink SIP phones, SIP-T9CM offers high quality audio, a broad range of voice codes and rich telephone features to general level users. Due to its 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.
• Voice codec: G.711, G.729AB, G.726, G.723.1/iLBC and GSM.
• Supports SIP 2.0 and NAT transverse: STUN mode.
• 2X16 characters LCD with backlight.

Network Features

SIP v1(RFC2543), v2(RFC3261)
Registrar Server: 3 SIP user accounts
Power over Ethernet optional
Outbound Proxy mode
Peer-to-peer SIP link mode
TFTP/DHCP/PPPoE Client
Telnet/HTTP Server
DNS Client
NAT/DHCP Server
QoS

Voice Features

G.711, G.723.1/iLBC, G.726, G.729AB and GSM
VAD, CNG, AEC
Packet Loss Compensation
Adaptive Jitter Buffer
 

Telephony Features

Call Hold
Call Waiting
Call Transfer
Call Forward
Caller ID
Call List
Redial and Flash
Speed Dial
Phone Book
Volume Adjustment
Ring Tone Selection
3-way Conference
Speakerphone
 

Configuration

Web Browser
Console/Telnet
Keypad
Auto-provision
 

Firmware Upgrade

TFTP
Console
HTTP
Auto-provision


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


IP Assignment

Static IP
DHCP
PPPoE
 

NAT Traversal

STUN
UPnP


Security

HTTP 1.1 basic/digest authentication for Web setup
MD5 for SIP authentication (RFC 2069/RFC 2617)