Cisco Unified IP Phone 7970G SIP Firmware
Manufacturer Part Number :CP-7970G
Warranty :15 Months
CISCO Unified IP Phone 7970 SIP CP7970G is loaded with SIP firmware for compatibility with HOSTED Platforms that support this model LIKE OURS
New Curly Cord
New Patch Lead
As this phone Supports POE a power supply can be purchased Separately
No Quibble Returns
15 month RTB Warranty
On Site Testing
CISCO Unified IP Phone 7970 with SCCP Firmware
Messages Button: The message key provides direct access to voicemail.
Directories Button: The phone identifies incoming messages and categorizes them on the screen, allowing users to quickly and effectively return calls using direct dial-back capability. The corporate directory integrates with the Lightweight Directory Access Protocol Version 3 (LDAPv3) standard directory.
Settings Button: The settings feature key allows the user to adjust display contrast and select from many ringer sounds and volume settings for all audio such as ringer, handset, headset, and speaker. Network configuration preferences also can be set up (usually by the system administrator). Configuration can either be automatic or manually set up for Dynamic Host Control Protocol (DHCP), Trivial File Transfer Protocol (TFTP), Cisco Unified CallManager software, and backup Cisco Unified CallManager software.
Services Button: The Cisco Unified IP Phone 7971G-GE allows users to quickly access diverse information such as weather, stocks, quote of the day, or any Web-based information. The phone uses XML to provide a portal to an ever-growing world of features and information.
Help: The online help feature gives users information about the phone keys, buttons, and features. The pixel display allows for greater flexibility of features and significantly expands the information viewed when using features such as Services, Information, Messages, and Directory. For example, the directory button can show local and server-based directory information.
Speakerphone, Mute and Headset Button: The phone features high-quality speakerphone technology. It also includes an easy-to-use speaker on/off button and microphone mute buttons. These buttons are lit when active. The convenient volume-control button provides for easy decibel-level adjustments for the speakerphone, handset, headset, and ringer.
Display: The display key provides easy access to previous pages or applications still open on the LCD
You will have to configure this telephone via the handset as there is no web browser for configuration
Power supply not included as these phones can run off Power over Ethernet (POE)
It’s important to ask what SIP is used for before deploying it.
Session Initiation Protocol (SIP) is a communications protocol that is widely used for managing multimedia communication sessions such as voice and video calls. SIP, therefore is one of the specific protocols that enable VoIP. It defines the messages that are sent between endpoints and it governs establishment, termination and other essential elements of a call. SIP can be used to transmit information between just two endpoints or many. In addition to voice, SIP can be used for video conferencing, instant messaging, media distribution and other applications. SIP has been developed and standardized under the auspices of the Internet engineering task force (IETF).
In short, if you want to an all-inclusive solution to your business communication needs, SIP trunking it is your best bet. Employing further tools, such as Asterisk, will make your SIP communications platform even better because you can customize it to your needs.