Dcare Technologies India Private Limited

 One voice line with two call appearances
 Backlit pixel-based display: 128 x 64 monochrome graphical liquid crystal display (LCD)

Line status: active line indication, name andnumber

 Menu-driven user interface

 Shared line appearance*

 Speakerphone

 Call hold

 Music on hold*

 Call waiting

 Caller ID name and number

 Outbound caller ID blocking

 Call transfer: attended and blind

 Three-way call conferencing with local mixing

 Multiparty conferencing via external conference ridge

 Automatic redial of last calling and last called umbers

 On-hook dialing

 Call pickup: selective and group*

 Call park and unpark*

 Call swap

 Call back on busy

 Call blocking: anonymous and selective

 Call forwarding: unconditional, no answer, on usy

 Hot line and warm line automatic calling

 Call logs (100 entries each): made, answered, nd missed calls

 Redial from call logs

 Personal directory with auto-dial (150 entries)

Do not disturb (callers hear line busy tone)

 Digits dialed with number auto-completion

 Anonymous caller blocking

 Uniform Resource Identifier (URI) (IP) dialing upport (vanity numbers)

 On-hook default audio configuration (speakerphone and headset)

 Multiple ring tones with selectable ring tone per ine

Called number with directory name matching

 Ability to call number using name: directory tching or via caller ID

 Subsequent incoming calls show calling name nd number

 Date and time with support for intelligent aylight vings

 Call duration and start time stored in call logs

 Call timer

 Name and identity (text) displayed at startup

tinctive ringing based on calling and called umber

 12 user-customizable ring tones

 Speed dialing, eight entries

 Configurable dial/numbering plan support

 Intercom*

 Group paging

 NAT Traversal, including STUN support

 DNS SRV and multiple A records for proxy ookup nd proxy redundancy

 Advanced Port Mirroring between PC Port and SW Port Debug, syslog, report generation, and event logging

 Secure call encrypted voice communication support

 Built-in web server for administration and configuration with multiple security levels

 Automated remote and secure provisioning via TFTP, HTTP or HTTPS

 Option to require administrator password to reset unit to factory defaults

 Four voice lines

 Four Independent SIP Registrations*

 Line status: active line indication, with name nd umber

Menu-driven user interface

 Shared line appearance**

 Speakerphone

 Call hold

Music on hold**

 Call waiting

 Caller ID name and number

Outbound caller ID blocking

 Call transfer: attended and blind

 Three-way call conferencing with local mixing

 Multiparty conferencing via external conference ridge

 Automatic redial of last calling and last called umbers

 On-hook dialing

 Call pickup: selective and group**

 Call park and unpark**

 Call swap

 Call back on busy

Call blocking: anonymous and selective

 Call forwarding: unconditional, no answer, on sy

 Hot line and warm line automatic calling

 Call logs (60 entries each): made, answered, d missed calls
Redial from call logs

 Personal directory with auto-dial (100 entries)

 Do not disturb
 Digits dialed with number auto-completion
 Anonymous caller blocking

 Uniform Resource Identifier (URI) (IP) alingupport (vanity numbers)

On-hook default audio onfiguration(speakerphone and headset)

 Multiple ring tones with selectable ring tone per ine

 Called number with directory name matching

 Ability to call number using name: directory atching or via caller ID

 Subsequent incoming calls show calling name nd umber

Date and time with support for intelligent daylight savings

 Call start time stored in call logs

 Call timer

 Name and identity (text) displayed at startup

Distinctive ringing based on calling and called number

 10 user-downloadable ring tones

 Speed dialing, eight entries

 Configurable dial/numbering plan support

 Intercom**

 Group paging

 Network Address Translation (NAT) Traversal, ncluding Simple Traversal of UDP Through NATs TUN) support

 DNS SRV and multiple A records for proxy ookup and proxy redundancy

 Syslog, debug, report generation, and event gging

Highly secure call encrypted voice communications support

 Built-in web server for administration and onfiguration with multiple security levels

 Automated remote provisioning, multiple ethods; up to 256-bit encryption (HTTP, HTTPS, rivial File Transfer Protocol [TFTP])

 One voice line

 One SIP registration

 Line status: active line indication, with name and number

 Menu-driven user interface

 Shared line appearance**

 Speakerphone

 Call hold

 Music on hold**

 Call waiting

 Caller ID name and number

 Outbound caller ID blocking

 Call transfer: attended and blind

 Three-way call conferencing with local mixing

 Multiparty conferencing via external conference ridge

 Automatic redial of last calling and last called umbers

 On-hook dialing

 Call pickup: selective and group**

 Call park and unpark**

 Call swap

 Call back on busy

 Call blocking: anonymous and selective

 Call forwarding: unconditional, no answer, on usy

 Hot line and warm line automatic calling

 Call logs (60 entries each): made, answered, nd missed calls

 Redial from call logs

Personal directory with auto-dial (100 entries)

 Do not disturb

igits dialed with number auto-completion

 Anonymous caller blocking

 Uniform Resource Identifier (URI) (IP) dialing upport (vanity numbers)

 On-hook default audio configuration speakerphone and headset)

 Multiple ring tones

 Called number with directory name matching

 Ability to call number using name: directory atching or via caller ID

 Subsequent incoming calls show calling name nd umber

 Date and time with support for intelligent daylight savings

 Call start time stored in call logs

 Call timer

 Name and identity (text) displayed at startup

 Distinctive ringing based on calling and called umber

 10 user-downloadable ring tones

 Speed dialing, eight entries

 Configurable dial/numbering plan support

 Intercom**

 Group paging

Network Address Translation (NAT) Traversal, including Simple Traversal of UDP Through NATs STUN) support

 DNS SRV and multiple A records for proxy ookup nd proxy redundancy

 Syslog, debug, report generation, and event ogging

 Highly secure call encrypted voice ommunications support

 Built-in web server for administration and onfiguration with multiple security levels

 Automated remote provisioning, multiple ethods; up to 256-bit encryption (HTTP, HTTPS, rivial File Transfer Protocol [TFTP])

• Ability to make high-quality feature-rich business-quality VoIP calls with cordless handset convenience

• High-resolution color display with hands-free speakerphone and four-way menu navigation button

• Superior performance and range along with highly efficient battery power usage based on DECT technology

• Essential telephony feature support such as caller ID, call transfer, call waiting, call forwarding, three-way conferencing, call history, and private and shared phonebooks

• Cost-effective, multiline on-premises mobility solution capable of registering up to five handsets to a Cisco SPA232D Multi-Line DECT ATA supporting four simultaneous active calls

• Highly configurable and secure remote provisioning capabilities to enable mass-scale service provider activation and deployment

24+ user-adjustable ring tones

A hearing-aid-compatible handset (meets American Disabilities Act [ADA] requirements) and HAC compliance for magnetic coupling to approved HAC hearing aids

G.711 and G.729a audio compression

H.323 compatible and Microsoft NetMeeting compatibility

An IP address assignment - DHCP client or statically configured

Comfort noise generation and voice activity detection (VAD) programming on a system basis

• For business or home office use
• Full-featured 4-line business-class IP phone supporting Power over Ethernet (PoE)
• Monochrome backlit display for ease of use, aesthetics, and on-screen applications
• Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
• Dual switched Ethernet ports for connecting a computer behind the phone, reducing cabling costs
• Wideband audio for unsurpassed voice clarity and enhanced speaker quality
• Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
• Supports up to two Cisco® SPA500S Expansion Module, adding up to 64 additional buttons*
• Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco Unified Communications 500 Series for Small Business

Cisco Unified IP Phone 7906G and 7911G are full-feature telephones that provide voice communication over the same data network that your computer uses, allowing you to place and receive calls, put calls on hold, transfer calls, make conference calls, and so on.

In addition to basic call-handling features, your Cisco Unified IP Phone 7906G and 7911G can provide specialized or advanced telephony features that extend your call-handling capabilities. Depending on the configuration, your phone supports:

You can use the following figures and table to identify the buttons and hardware on your phone.

• 3-line business-class IP phone
• Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
• Dual switched Ethernet ports, speakerphone, caller ID, call hold, conferencing, and more
• Easy installation and highly secure remote provisioning, as well as menu-based and web-based configuration
• Supports both Session Initiation Protocol (SIP) and Smart Phone Control Protocol (SPCP) with the Cisco® Unified Communications 500 Series

Additional Information:

• Delivery Time: 1 week

Advanced, Feature-Rich, Multiline IP Phone for SIP-Based VoIP Service

Highlights

• Industry-leading VoIP technology from Cisco that delivers a high-quality IP phone that is unparalleled in features, value, and support.

• Standard features include four active lines, dual-switched Ethernet ports, Power over Ethernet, a high-resolution graphical display, full-duplex speakerphone, and a headset port

• Business-grade reliability with advanced voice quality of service

• Comprehensive interoperability with the Cisco^® Small Business SPA9000 Voice System, SIP-based phone systems, and SIP phone services from your Internet telephony service provider

Refer to these sections for information about the impact of power outages and other devices on your Cisco Unified IP Phone.

Power Outage

Your accessibility to emergency service through the phone is dependent on the phone being powered. If there is an interruption in the power supply, Service and Emergency Calling Service dialing will not function until power is restored. In the case of a power failure or disruption, you may need to reset or reconfigure equipment before using the Service or Emergency Calling Service dialing.

Using External Devices

When you use external devices with the Cisco Unified IP Phone, Cisco recommends the use of good-quality external devices, such as headsets, that are shielded against unwanted radio frequency (RF) and audio frequency (AF) signals.

Depending on the quality of these devices and their proximity to other devices such as mobile phones or two-way radios, some audio noise may still occur. In these cases, Cisco recommends that you take one or more of these actions:

•Move the external device and the cables away from the source of the RF or AF signals.

•Use shielded cables for the external device, or use cables with a better shield and connector.

•Shorten the length of the external device cable.

•Apply ferrites or other such devices on the cables for the external device.

Cisco cannot guarantee the performance of the system because Cisco has no control over the quality of external devices, cables, and connectors. The system will perform adequately when suitable devices are attached using good quality cables and connectors.

 One voice line

 Music on hold*

Call waiting

 Outbound caller ID blocking**

 Call transfer: attended and blind** ( SIP Only )

 Three-way call conferencing with local mixing

 Multiparty conferencing via external conference bridge

 Automatic redial of last calling and last called numbers

 Call pickup: selective and group*

 Call swap

 Call back on busy**

 Call blocking: anonymous and selective**

 Call forwarding: unconditional, no answer, and on busy**

 Hot line and warm line automatic calling

 Digits dialed with number auto-completion

 Anonymous caller blocking**

 Support for Uniform Resource Identifier (URI) (IP) dialing (vanity numbers)

 Multiple ring tones with selectable ring tone per line

 Call duration and start time stored in call logs in web GUI

 Distinctive ringing based on calling and called number

 10 user-downloadable ring tones

 Speed dialing, eight entries Configurable dial/numbering plan support

 Intercom*

 Group paging

 Network Address Translation (NAT) Traversal, including Serial Tunnel (STUN) support

 DNS SRV and multiple A records for proxy lookup and proxy redundancy

 Syslog, debug, report generation, and event logging

 Support for highly secure encrypted voice ommunications

 Built-in web server for administration and configuration, with multiple security levels

 Automated remote provisioning, multiple methods; up to 256 bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP])

 Option to require administrator password to reset unit to factory defaults

• Easy to install, and designed exclusively for the Cisco® SPA 500 Series IP Phones
• 32 programmable buttons simplify dialing and call transfers
• Multicolored LEDs monitor the status of each configured voice line via Busy Lamp Field (BLF)
• Two attendant consoles can be linked, for a total of 64 buttons

• Call transfer
• Speed dial
• DSS
• Busy lamp field
• One-touch transfer
• Call pickup