The documentation set for this product strives to use bias-free language. For the purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product. Learn more about how Cisco is using Inclusive Language.
The following table lists the latest Cisco IP Phone firmware version supported for Cisco Unified Communications Manager 10.0. Phones in this table that are identified with an asterisk (*) are new since Cisco Unified Communications Manager 9.0.
Phone family | Firmware release number |
---|---|
Cisco Unified SIP Phone 3905 |
9.4(1) |
Cisco Unified IP Phones 6901 and 6911 |
9.3(1)SR1 |
Cisco Unified IP Phones 6921, 6941, 6945, and 6961 |
9.4(1) |
* Cisco IP Phone 7800 Series |
10.1(1) |
Cisco Unified IP Phone 7900 Series |
9.3(1)SR2 |
Cisco Unified Wireless IP Phone 792x Series |
1.4(5) |
* Cisco Unified IP Conference Phone 8831 |
9.3(3) |
Cisco Unified IP Phones 8941 and 8945 |
9.3(2)SR1 |
Cisco Unified IP Phones 8961, 9951, and 9971 |
9.4(1) |
The following table lists the features added to the Cisco Unified SIP Phone 3905 for firmware release 9.4(1). For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/ps7193/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
IPv6 |
9.4(1) |
IPv6 Ready Logo (SIP) |
9.4(1) |
Multiple Date Display Formats |
9.4(1) |
Paging Support on Cisco Unified Communications Manager |
9.4(1) |
Paging Support on Cisco Unified Communications Manager Express |
9.4(1) |
SIP MD5 Digest Authentication |
9.4(1) |
The following table lists the features added to the Cisco Unified IP Phone 6900 Series for firmware releases 9.3(1)SR1, 9.3(3), 9.3(3)SR1, and 9.4(1). For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/ps10326/prod_release_notes_list.html
Feature name | Supported on Cisco Unified IP Phones 6901 and 6911 | Supported on Cisco Unified IP Phones 6921, 6941, 6945, and 6961 |
---|---|---|
Configurable Maximum Number of Calls and Busy Trigger |
9.3(1)SR1 |
No |
Web Access Disabled by Default |
9.3(1)SR1 |
9.3(3) |
Call Waiting Ring |
No |
9.3(3) |
Debug Phone |
No |
9.3(3) |
HTTPS |
No |
9.3(3) |
Show Calling ID and Calling Number |
No |
9.3(3) |
Electronic Hookswitch |
No |
9.3(3) |
Minimum Ring Volume |
No |
9.3(3) |
Secure EMCC |
No |
9.3(3) |
TVS and Security by Default |
No |
9.3(3) |
E-SRST Service Improvements |
No |
9.4(1) |
IPv6 Support for SIP |
No |
9.4(1) |
Peer Firmware Sharing |
No |
9.4(1) |
PSTN Mode |
No |
9.4(1) |
Save Volume Change |
No |
9.4(1) |
Serviceability for SIP Endpoints |
No |
9.4(1) |
Show Call Duration in Call History |
No |
9.4(1) |
The Cisco® IP Phone 7800 Series is a high-fidelity voice communications portfolio designed for people- centric collaboration. It combines always-on reliability and security, full-featured easy-to-use IP telephony, and wideband audio to increase productivity, with an earth-friendly design for reduced costs.
The Cisco IP Phone 7800 Series brings a higher quality standard, with full wideband audio support for handset, headset and speaker, to our voice-centric portfolio. A new ergonomic design includes support for larger grayscale, graphical backlit displays. The Cisco IP Phone 7800 Series also offers customers very low power consumption, as the phones are IEEE Class 1 devices. Combined with support for Cisco EnergyWise™, this delivers greater economies of scale in customers wiring closets as well as helping to reduce operating expenditures with energy savings. Other key differences include Electronic Hook-switch capability, for call control while using third-party headsets, encrypted communications, and a field replaceable bezel option.
For more information on the Cisco IP Phone 7800 Series, see http://www.cisco.com/en/US/products/ps13220/tsd_products_support_series_home.html
The following table lists the features added to the Cisco Unified IP Phone 7900 Series for firmware releases 9.3(1)SR1 and 9.3(2)SR2. For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
Hardware Updates |
9.3(1)SR1 |
Firmware Release 9.3(1)SR2 Security Enhancements |
9.3(1)SR2 |
The following table lists the features added to the Cisco Unified Wireless IP Phone 792x Series for firmware releases 1.4(3), 1.4(4), and 1.4(5). For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
Clear Call History Confirmation |
1.4(3) |
7926G J2ME Memory Increase |
1.4(4) |
792x USB Driver Support for Microsoft Windows 7 |
1.4(4) |
Dock Icon Support for Cisco Unified Wireless IP Phone 7925G Desktop Charger |
1.4(4) |
Timezone Support |
1.4(4) |
XSI Audio Path Control |
1.4(4) |
MIDlet Minimize to Background When Power On |
1.4(5) |
The Cisco Unified IP Conference Phone 8831 is a full-featured single line conference station that provides voice communication over an IP network. It functions much like a digital business phone, allowing users to place and receive calls and to access features such as mute, hold, transfer, speed dial, call forward, and more. In addition, because conference stations connect to the data network, they offer enhanced IP telephony features, including access to network information and services, and customizable features and services.
The conference phone provides a backlit LCD screen, support for up to ten speed-dial numbers, and a variety of other sophisticated functions. Optional microphone extension kits provide enhanced room coverage that can be further expanded by linking two units together.
In addition to basic call handling features, the conference phone can provide enhanced productivity features that extend call handling capabilities. Depending on configuration, the phone supports:
Access to network data, XML applications, and web-based services.
Online customizing of conference station features and services from the User Options web pages.
The phone supports seamless firmware upgrade. If two devices are connected in Linked Mode, firmware upgrades are pushed automatically from the primary unit to the secondary unit.
For more information on the Cisco Unified IP Conference Phone 8831, see http://www.cisco.com/en/US/products/ps12965/tsd_products_support_series_home.html.
The following table lists the features added to the Cisco Unified IP Phones 8941 and 8945 for firmware release 9.3(2). For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/ps10451/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
Audio Only Lock Icon |
9.3(2) |
Enhanced Message Waiting Indicator |
9.3(2) |
HTTPS Support |
9.3(2) |
One Click to Home Screen |
9.3(2) |
User Experience Enhancements |
9.3(2) |
VPN Client Support |
9.3(2) |
XSI Component API Support |
9.3(2) |
The following table lists the features added to the Cisco Unified IP Phones 8961, 9951, and 9971 for firmware releases 9.3(2), 9.3(4), and 9.4(1). For more information, see the Release Notes at the following location:
http://www.cisco.com/en/US/products/ps10453/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
Actionable Incoming Call Alert |
9.3(2) |
Call History Display Enhancement for Call Window |
9.3(2) |
Custom Line filters |
9.3(2) |
New Hardware Models |
9.3(2) |
Prompt for Barge |
9.3(2) |
Audio-Only Lock Icon |
9.3(2) |
Configurable DF Bit |
9.3(2) |
CGI CallInfo and LineInfo |
9.3(4) |
Conference and Transfer Enhancement |
9.3(4) |
Configurable Font Size |
9.3(4) |
Hide Softkeys in Full Screen Video Mode |
9.3(4) |
Hold or Resume Toggle from Hard Key |
9.3(4) |
One Button to Access Call History |
9.3(4) |
Unique cBarge Call Instance ID |
9.3(4) |
Cisco IP Manager Assistant Support |
9.3(4) |
Separate Audio and Video Mute |
9.3(4) |
Softkey Template |
9.3(4) |
URI Dialing Enhancement |
9.3(4) |
CGI ModeInfo |
9.4(1) |
Confidential Access Level |
9.4(1) |
Configurable RTP/SRTP Port Range |
9.4(1) |
Configurable TLS Resumption Timer |
9.4(1) |
CTL and ITL Status Display and Report |
9.4(1) |
CTL/ITL Signature |
9.4(1) |
E-SRST Service Improvements |
9.4(1) |
FIPS 140-2 Level 1 Compliance |
9.4(1) |
Gateway Recording For SIP |
9.4(1) |
Hide Wi-Fi UI Setting |
9.4(1) |
IPv6 Support |
9.4(1) |
Line State Display Enhancement |
9.4(1) |
RTCP Always On |
9.4(1) |
Separate Video and Audio Port Range configuration |
9.4(1) |
Serviceability for SIP Endpoints |
9.4(1) |
Unified Font Size Enhancement |
9.4(1) |
The following table lists the latest Cisco Desktop Collaboration Experience DX600 Series firmware version supported for Cisco Unified Communications Manager 10.0. Phones in this table that are marked with an asterisk (*) are new since Cisco Unified Communications Manager 9.0.
Phone family | Firmware release number |
---|---|
* Cisco Desktop Collaboration Experience DX650 | 10.1(1) |
The Cisco Desktop Collaboration Experience DX650 (Cisco DX650) is built to deliver integrated, always-on and secure, high-definition (HD) voice and video communications; conferencing with Cisco WebEx meeting applications; presence and instant messaging with the Cisco Jabber messaging integration platform; and on-demand access to cloud services. Cisco DX650 meets the demands of people who must collaborate effectively with experts even if separated by long distances.
The following table lists the features added to the Cisco DX650 for firmware releases 10.0(1) and 10.1(1). For more information on the Cisco DX650, see the Release Notes at the following location: http://www.cisco.com/en/US/products/ps12956/prod_release_notes_list.html
Feature name | Firmware release |
---|---|
+ Dialing (ITU E.164) | 10.0(1) |
Abbreviated dialing | 10.0(1) |
Adjustable ringing and volume levels | 10.0(1) |
Adjustable display brightness | 10.0(1) |
Android bundled applications and widgets | 10.0(1) |
Android core features | 10.0(1) |
Application dial rule | 10.1(1) |
Auto-answer | 10.0(1) |
Auto-detection of headset | 10.0(1) |
Barge (cBarge) | 10.0(1) |
Bluetooth Hands-Free Profile | 10.1(1) |
Bluetooth Phone Book Access Profile | 10.1(1) |
Callback | 10.0(1) |
Call Chaperone | 10.0(1) |
Call forward | 10.0(1) |
Call forward notification | 10.0(1) |
Call history lists | 10.0(1) |
Call park (including Directed Call Park and Assisted Directed Call Park) | 10.0(1) |
Call pickup | 10.0(1) |
Call timer | 10.0(1) |
Call waiting | 10.0(1) |
Caller ID | 10.0(1) |
Cisco AnyConnect Secure Mobility Client (VPN) Version 3.0 | 10.0(1) |
Cisco Jabber IM | 10.0(1) |
Cisco WebEx Version 2.5 | 10.0(1) |
Corporate directory | 10.0(1) |
Conference (ad hoc) | 10.0(1) |
Data migration | 10.1(1) |
Default wallpaper control | 10.1(1) |
Direct transfer | 10.0(1) |
Divert (iDivert) | 10.0(1) |
Do Not Disturb (DND) | 10.0(1) |
Extension Mobility service | 10.0(1) |
Fast-dial service | 10.0(1) |
Flexible DSCP | 10.1(1) |
Forced access codes and client matter codes | 10.0(1) |
Google bundled applications | 10.0(1) |
Group call pickup | 10.0(1) |
Hold (and Resume) | 10.0(1) |
Intercom | 10.0(1) |
International call logging | 10.0(1) |
Join (ad hoc) | 10.0(1) |
Last-number redial (LNR) | 10.0(1) |
Malicious-caller ID | 10.0(1) |
Media Net _End of session | 10.1(1) |
Message-waiting indicator (MWI) | 10.0(1) |
Meet-me conference | 10.0(1) |
Mobility (Mobile Connect and Mobile Voice Access) | 10.0(1) |
Music on Hold (MoH) | 10.0(1) |
Mute (audio and video) | 10.0(1) |
Network profiles (automatic) | 10.0(1) |
On- and off-network distinctive ringing | 10.0(1) |
Personal directory | 10.0(1) |
Phone-Only Mode | 10.1(1) |
PickUp | 10.0(1) |
Predialing before sending | 10.0(1) |
Privacy | 10.0(1) |
Private Line Automated Ringdown (PLAR) | 10.0(1) |
Quick Contact Badge | 10.0(1) |
Remote Wipe/Lock | 10.1(1) |
Remotely check CTL/ITL file | 10.1(1) |
Ring tone per line appearance | 10.0(1) |
Self-provisioning | 10.1(1) |
Self-View (video call) | 10.0(1) |
Service URL | 10.0(1) |
Shared line(s) | 10.0(1) |
SIP Gateway Recording | 10.1(1) |
Time and date display | 10.0(1) |
Transfer (ad hoc) | 10.0(1) |
Visual Voicemail | 10.0(1) |
Wireless profiles | 10.1(1) |
Directory number settings
Field | Description |
---|---|
Directory Number Information |
|
Urgent Priority |
If the dial plan contains overlapping patterns, Cisco Unified Communications Manager does not route the call to the device associated with the directory number until the interdigit timer expires (even if the directory number is a better match for the sequence of digits dialed as compared to the overlapping pattern). Check this check box to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately to the device associated with the directory number. By default, the Urgent Priority check box is unchecked. |
Translation pattern settings
Field | Description |
---|---|
Pattern Definition |
|
Use Originator's Calling Search Space |
To use the originator's calling search space for routing a call, check the Use Originator's Calling Search Space check box. When you check this check box, it disables the Calling Search Space drop-down list box. When you save the page, the Calling Search Space box is grayed out and set to <None>. If the originating device is a phone, the originator's calling search space results from the device calling search space (configured on the Phone Configuration window) and line calling search space (configured on the Directory Number Configuration window). Whenever a translation pattern chain is encountered, for subsequent lookups Calling Search Space is selected depending upon the value of this check box at current translation pattern. If you check the Use Originator's Calling Search Space check box at current translation pattern, then originator's Calling Search Space is used and not the Calling Search Space for the previous lookup. If you uncheck the Use Originator's Calling Search Space check box at current translation pattern, then Calling Search Space configured at current translation pattern is used. |
Do Not Wait For Interdigit Timeout On Subsequent Hops |
When you check this check box along with the Urgent Priority check box and the translation pattern matches with a sequence of dialed digits (or whenever the translation pattern is the only matching pattern), Cisco Unified Communications Manager does not start the interdigit timer after it matches any of the subsequent patterns. Note: Cisco Unified Communications Manager does not start the interdigit timer even if subsequent patterns are of variable length or if overlapping patterns exist for subsequent matches. Whenever you check the Do Not Wait For Interdigit Timeout On Subsequent Hops check box that is associated with a translation pattern in a translation pattern chain, Cisco Unified Communications Manager does not start the interdigit timer after it matches any of the subsequent patterns. Note: Cisco Unified Communications Manager does not start interdigit timer even if subsequent translation patterns in a chain have Do Not Wait For Interdigit Timeout On Subsequent Hops unchecked. |
Call Control Discovery feature parameters
To access the feature parameters that support the call control discovery feature, choose . For additional information, you can click the question mark help in the Feature Configuration window.
Feature Parameter |
Description |
---|---|
Set Urgent Priority for Fixed-Length CCD Learned Patterns |
This parameter determines whether Cisco Unified Communications Manager waits for interdigit timer before routing the call to the destination that is associated with the fixed-length learned pattern (when the fixed-length learned pattern is a better match for the sequence of digits dialed as compared to the overlapping route pattern configured). If the parameter is set to True, Cisco Unified Communications Manager does not wait for the interdigit timer before routing the call to the destination that is associated with the fixed-length learned pattern. If the parameter is set to False, Cisco Unified Communications Manager waits for interdigit timer before routing the call to the destination that is associated with the fixed-length learned pattern. The default equals False. Example: Cisco Unified Communications Manager learns the pattern +44987XXX for routing the calls to another Cisco Unified Communications Manager and there is also a route pattern configured as \+44! for routing the calls to the PSTN destination. If this parameter is set to False and +44987127 is dialed, Cisco Unified Communications Manager waits for interdigit timer before routing the call to another Cisco Unified Communications Manager (this interdigit timer allows user to dial more digits after +44987127 to reach the PSTN destination). If this parameter is set to True and +44987127 is dialed, then Cisco Unified Communications Manager immediately routes the call to another Cisco Unified Communications Manager. |
Set Urgent Priority for Variable-Length CCD Learned Patterns |
This parameter determines whether Cisco Unified Communications Manager waits for interdigit timer before routing the call to the destination that is associated with the variable-length learned pattern. If the parameter is set to True, Cisco Unified Communications Manager does not wait for interdigit timer before routing the call to the destination that is associated with the variable-length learned pattern. If the parameter is set to False, Cisco Unified Communications Manager waits for interdigit timer before routing the call to the destination that is associated with the variable-length learned pattern. The default equals False. Example: Cisco Unified Communications Manager has translation pattern 9011.!# configured. This translation pattern strips predot digits and the trailing # character and adds the prefix +55 to the dialed digits. Cisco Unified Communications Manager also learns pattern \+55.! for routing the calls to another Cisco Unified Communications Manager. If this parameter is set to False and 9011234567# (resultant digits = +55234567) is dialed, Cisco Unified Communications Manager waits for interdigit timer before routing the call to another Cisco Unified Communications Manager. If this parameter is set to True and 9011234567# (resultant digits = +55234567) is dialed, then Cisco Unified Communications Manager immediately routes the call to another Cisco Unified Communications Manager. |
Field | Description | ||
---|---|---|---|
Connected Party Settings |
|||
Connected Party Transformation CSS |
This setting is applicable only for inbound Calls. This setting allows you to transform the connected party number that Cisco Unified Communications Manager sends in another format, such as a DID or E.164 number. This setting is applicable while sending connected number for basic call as well as sending connected number after inbound call is redirected. Cisco Unified Communications Manager includes the transformed number in the Connected Number Information Element (IE) of CONNECT and NOTIFY messages. Make sure that the Connected Party Transformation CSS that you choose contains the connected party transformation pattern that you want to assign to this device.
|
||
Use Device Pool Connected Party Transformation CSS |
To use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Connected Party Transformation CSS that you configured for this device in the Trunk Configuration window. |
||
Outbound Calls |
|||
Called Party Transformation CSS |
This setting allows you to send transformed called party number in SETUP message for outgoing calls. Make sure that the Called Party Transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device.
|
||
Calling Party Transformation CSS |
This setting allows you to send transformed calling party number in SETUP message for outgoing calls. Also when redirection occurs for outbound calls, this CSS will be used to transform the connected number sent from Cisco Unified Communications Manager side in outgoing NOTIFY messages. Make sure that the Calling Party Transformation CSS that you choose contains the calling party transformation pattern that you want to assign to this device.
|
Field | Description | ||
---|---|---|---|
Connected Party Settings |
|||
Connected Party Transformation CSS |
This setting is applicable only for inbound Calls. This setting allows you to transform the connected party number that Cisco Unified Communications Manager sends in another format, such as a DID or E.164 number. This setting is applicable while sending connected number for basic call as well as sending connected number after inbound call is redirected. Cisco Unified Communications Manager includes the transformed number in the Connected Number Information Element (IE) of CONNECT and NOTIFY messages. Make sure that the Connected Party Transformation CSS that you choose contains the connected party transformation pattern that you want to assign to this device.
|
||
Use Device Pool Connected Party Transformation CSS |
To use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Connected Party Transformation CSS that you configured for this device in the Trunk Configuration window. |
||
Call Routing Information - Outbound Calls |
|||
Called Party Transformation CSS |
This setting allows you to send transformed called party number in SETUP message for outgoing calls. Make sure that the Called Party Transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device.
|
||
Calling Party Transformation CSS |
This setting allows you to send transformed calling party number in SETUP message for outgoing calls. Also when redirection occurs for outbound calls, this CSS will be used to transform the connected number sent from Cisco Unified Communications Manager side in outgoing NOTIFY messages. Make sure that the Calling Party Transformation CSS that you choose contains the calling party transformation pattern that you want to assign to this device.
|
Field | Description | ||||
---|---|---|---|---|---|
Call Routing Information - Outbound Calls |
|||||
Called Party Transformation CSS |
This setting allows you to send transformed called party number in SETUP message for outgoing calls. Make sure that the Called Party Transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device.
|
||||
Calling Party Transformation CSS |
This setting allows you to send transformed calling party number in SETUP message for outgoing calls. Also when redirection occurs for outbound calls, this CSS will be used to transform the connected number sent from Cisco Unified Communications Manager side in outgoing NOTIFY messages. [ For PRI DMS - 100 and DMS - 200 ]. Make sure that the Calling Party Transformation CSS that you choose contains the calling party transformation pattern that you want to assign to this device.
|
||||
Connected Party Settings |
|||||
Connected Party Transformation CSS |
This setting is applicable only for inbound Calls. This setting allows you to transform the connected party number sent from Cisco Unified Communications Manager in another format, such as a DID or E.164 number.
Using this setting, Cisco Unified Communications Manager includes transformed number in Connected Number Information Element ( IE) of CONNECT message for basic call. For PRI DMS - 100 and DMS - 250 protocols , Cisco Unified Communications Manager includes transformed number in Connected Number Information Element ( IE) of NOTIFY message for inbound calls after redirection. Make sure that the Connected Party Transformation CSS that you choose contains the connected party transformation pattern that you want to assign to this device.
|
||||
Use Device Pool Connected Party Transformation CSS |
To use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Connected Party Transformation CSS that you configured for this device in the Trunk Configuration window. |
||||
Incoming Called Party Settings |
|||||
Clear Prefix Settings |
To delete all prefixes for all called party number types, click Clear Prefix Settings. |
||||
Default Prefix Settings |
To enter the default value for all prefix fields at the same time, click Default Prefix Settings. |
||||
National Number |
Configure the following settings to transform incoming called party numbers that use National for the Called Party Number Type.
|
||||
International Number |
Configure the following settings to transform incoming called party numbers that use International for the Called Party Number Type.
|
||||
Unknown Number |
Configure the following settings to transform incoming called party numbers that use Unknown for the Called Party Number Type.
|
||||
Subscriber Number |
Configure the following settings to transform incoming called party numbers that use Subscriber for the Called Party Number Type.
|
Field | Description | ||||
---|---|---|---|---|---|
Incoming Called Party Settings |
|||||
Clear Prefix Settings |
To delete all prefixes for all called party number types, click Clear Prefix Settings. |
||||
Default Prefix Settings |
To enter the default value for all prefix fields at the same time, click Default Prefix Settings. |
||||
National Number |
Configure the following settings to transform incoming called party numbers that use National for the Called Party Number Type.
|
||||
International Number |
Configure the following settings to transform incoming called party numbers that use International for the Called Party Number Type.
|
||||
Unknown Number |
Configure the following settings to transform incoming called party numbers that use Unknown for the Called Party Number Type.
|
||||
Subscriber Number |
Configure the following settings to transform incoming called party numbers that use Subscriber for the Called Party Number Type.
|
Note | When you configure the Local Route Group feature, add the route groups to the route list by selecting those local route group names that are appended with the Local Route Group tag that appears in the drop-down list box. |
No changes.
No changes.
No changes.
No changes.
No changes.
No changes.