Cisco Unified IP Phone 8961, 9951, and 9971 Release Notes for Firmware Release 9.3(1)
Cisco Unified IP Phone 8961, 9951, and 9971 Release Notes for Firmware Release 9.3(1)
Downloads: This chapterpdf (PDF - 1.34MB) The complete bookPDF (PDF - 2.06MB) | Feedback

Cisco Unified IP Phone 8961, 9951, and 9971 Release Notes for Firmware Release 9.3(1)

Contents

Cisco Unified IP Phone 8961, 9951, and 9971 Release Notes for Firmware Release 9.3(1)

Introduction

These release notes support the Cisco Unified IP Phones 8961, 9951, and 9971 running SIP firmware release 9.3(1).

The following table lists the Cisco Unified Communications Manager release and protocol compatibility for the Cisco Unified IP Phones.

Table 1 Cisco Unified IP Phones, Cisco Unified Communications Manager, and firmware release compatibility
Cisco Unified IP Phone Protocol Cisco Unified Communications Manager

Cisco Unified IP Phones 8961, 9951, and 9971

SIP

Cisco Unified Communications Manager release 7.1(5) and later

Related documentation

Use the following sections to obtain related information.

Cisco Unified IP Phone 8900 Series documentation

Refer to publications that are specific to your language, phone model, and Cisco Unified Communications Manager release. Navigate from the following documentation URL:

http:/​/​www.cisco.com/​en/​US/​products/​ps10451/​tsd_​products_​support_​series_​home.html

Cisco Unified IP Phone 9900 Series documentation

Refer to publications that are specific to your language, phone model, and Cisco Unified Communications Manager release. Navigate from the following documentation URL:

http:/​/​www.cisco.com/​en/​US/​products/​ps10453/​tsd_​products_​support_​series_​home.html

Cisco Unified Communications Manager documentation

See the Cisco Unified Communications Manager Documentation Guide and other publications that are specific to your Cisco Unified Communications Manager release. Navigate from the following documentation URL:

http:/​/​www.cisco.com/​en/​US/​products/​sw/​voicesw/​ps556/​tsd_​products_​support_​series_​home.html

Cisco Business Edition 5000 documentation

See the Cisco Business Edition 5000 Documentation Guide and other publications that are specific to your Cisco Business Edition 5000 release. Navigate from the following URL:

http:/​/​www.cisco.com/​en/​US/​products/​ps7273/​tsd_​products_​support_​series_​home.html

Cisco Virtualization Experience Client 2000 Series documentation

Refer to publications that are specific to your language. Navigate from the following documentation URL:

http:/​/​www.cisco.com/​en/​US/​products/​ps11499/​tsd_​products_​support_​series_​home.html

New and Changed Features

The following sections describes the features that are new or have changed.

Features available with firmware release

The following sections describe the features available with the Firmware Release.

Assured Services SIP

The Assured Services for SIP Lines (AS-SIP) feature provides the ability for users to place priority calls and, if necessary, preempt lower-priority phone calls.

These AS-SIP enhancements are supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971

This feature is also supported on third-party telephones.

Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

Display Survivable Remote Site Telephony Message

The Display Survivable Remote Site Telephony (SRST) Message feature displays a message to the users on the phone screen when communication with the Cisco Unified Communications Manager fails. This message alerts users that some of the features of their phones are no longer available.

This feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Device Invoked Recording

The Device Invoked Recording feature enables users to control the recording of phone calls using the Record button on the phone.

Users see a status indicator on the phone display, showing when a conversation is being recorded.

The Device Invoked Recording feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SCCP and SIP)

Edit Speed Dial Without Restart

The Speed Dial Without a Restart feature makes it easier maintain an updated collection of Speed Dial numbers by:

  • Administrators can add, modify, or delete a Speed Dial number from the Cisco Unified Communications Manager Administration page.
  • Users can add, modify, or delete a Speed Dial number from the Cisco Unified Communications Manager User Options web pages.

The phone is not required to restart in order to accept these changes.

This feature does not require any specific configuration.

This enhancement is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Extension Mobility Cross Cluster Enhancement

The Extension Mobility Cross Cluster (EMCC) Enhancement feature preserves the Product Specific Configuration settings for the phone. By so doing, security policies are maintained, network bandwidth is preserved and network failure is avoided within the visiting cluster (VC).

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information

Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

Handset Bass Adjustment

The Handset Bass Adjustment feature allows the phone to operate with a reduced bass tone rather than with full bass. Reduced bass removes low frequencies, which can improve muffled voices or insufficient volume on handsets. There are no administrator or user controlled settings for this feature.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Pause In Speed Dial

The Pause in Speed Dial feature enables users to set up the speed dial feature to reach destinations that require a Forced Authorization Code (FAC), Client Matter Code (CMC), dialing pauses, and additional digits (such as a user extension, a meeting access code, or a voicemail password) without manual intervention. When the user presses the speed dial, the phone establishes the call to the specified DN and sends the specified FAC, CMC, and DTMF digits to the destination with dialing pauses inserted.

To include dialing pauses in the speed dial, the user must specify a comma (,) in the speed dial string. Each comma indicates a pause of 2 seconds. The comma also acts as a delimiter between destination digits, the FAC, CMC, and additional DTMF digits. The comma as delimiter is useful in the following cases:

  • Differentiates overlapping dial patterns (for example 9.xxx from 9.xxxxx)
  • Differentiates overlapping FAC or CMC (for example, 8787 from 87879)
  • Identifies the destination number when using variable-length dial patterns (for example 9.!)

Be aware of the following requirements when you include FAC and CMC in the speed dial string:

  • FAC must always precede CMC in the speed dial string.
  • A speed dial label is required for speed dials containing FAC and DTMF digits.
  • Only one comma is allowed between FAC and CMC digits in the string.

For any additional DTMF digits specified after the FAC and CMC, the phone dials these additional digits (with pauses) after the call is connected.

This feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

PLK Support for Queue Statistics

The PLK Support for Queue Statistics feature enables the users to query the call queue statistics for hunt pilots and the statistics display on the phone screen.

The programmable line key Queue Status can be configured by the administrator. When the user presses Queue Status, the phone displays the Queue Status screen. The Queue Status screen includes hunt pilot directory number, number of callers in queue, and the longest call waiting time in queue.

The statistics information does not update automatically. The user must press the Update softkey to view updated statistics. To exit from the queue display screen, the user presses the Exit softkey.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

RTCP Hold On SIP

The RTCP Hold For SIP feature ensures that held calls are not dropped by the gateway. The gateway checks the status of the RTCP port to determine if a call is active or not. By keeping the phone port open, the gateway will not end held calls.

This feature has no administration or user impacts.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971

Secure Extension Mobility Cross Cluster

Secure Extension Mobility Cross Cluster (EMCC) enables a user in one cluster (using an encrypted/authenticated Cisco Unified IP Phone with TFTP Encrypted Config/Digest Authentication enabled) to log in to another cluster when two cluster are both in mixed mode.

Configure Cisco Extension Mobility on Cisco Unified IP Phones before you configure EMCC.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information

Cisco Unified Communications Manager Features and Services Guide, chapter "Cisco Extension Mobility Cross Cluster"

SIP Phone No Alert Name in Placed Calls History

The SIP Phone No Alert Name in Placed Calls History feature displays the alert name in the Placed Calls history when the phone is in a translation pattern or call redirection state. Currently these calls appear on the calling party's call history as Unknown. With this enhancement these calls appear as the callee's Alert Name.

This enhancement does not require any specific configuration.

The feature is supported on following phones (SIP):

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information

Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Uniform Resource Identifier Dialing

The Uniform Resource Identifier (URI) Dialing feature enables the user to place calls using alphanumeric URI address as a directory number, for example, bob@cisco.com. The user must enter the URI address to select the contact.

The phone screen displays the call information for the URI call. The call history record the URI call information in the Call History and the Details page.

The user cannot place calls by URI address using the soft keypad.

URI Dialing has the following feature requirements:

Onhook call initiation

The user must press the ABC softkey to switch the input method to URI Dialing mode using the keypad.

Off-hook call initiation

The user can place calls using URI Dialing if the URI address is stored in the speed dial list or call history.

Redial

Press the Redial button to call the most recently dialed URI address.

Speed Dial

The user can configure a URI address as a speed dial entry to place a call.

Session bubble

When the user dials or receives a call through URI Dialing, the call bubble displays the complete URI address.

Incoming call notification

The incoming call alert notification supports the URI address display.

Missed, Placed, and Received call history

The URI Dialing logs are saved in the call history.

Dial URI from call history

The user can select the URI address from the call list to place a call. The user can navigate to the URI call history or enter the URI Dial mode to place a call.

Default domain

The user can enter the complete domain name and override the default domain.

Call History filter

While the user enters the URI address to place a call through URI Dialing, the call history appears based on the characters entered.

Call Forward All

The user can configure the Call Forward All destination using the speed dial or call history entries.

Transfer

The user can initiate a Transfer call using URI dialing if the URI address is stored in the Speed Dial list or Call History.

Ad Hoc Conference

The user can initiate a conference call and add multiple parties using URI Dialing if the URI address is stored in the speed dial list or call history.

Privacy

The user can hide the display of the URI address information. For more information on Privacy, see Cisco Unified Communications Manager Features and Services Guide 9.0 and Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP).

Busy Lamp Field Speed Dial

The user can monitor the state (in-use or idle) of a call using URI Dialing associated with speed dial or call history.

Call back

The user can initiate a directory number call when the URI Dialing target becomes available.

Features compatibility

The URI address speed dial or redial is disabled under Meet Me conference and Group Call Pickup features.

Cisco Unified Communications Manager Express and Survivable Remote Site Telephony

When the phones are connected to the Cisco Unified Communications Manager Express and Survivable Remote Site Telephony (CME/SRST), the URI Dialing functionalities are disabled. The ABC softkey does not appear on the phone screen.

This feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971

Note


Wait for the ABC softkey to appear before you proceed with URI dialing.


Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9851, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Unique Call ID Display

The Unique Call ID Display feature ensures that all calls with same group call ID display the same call ID on all the phones with the same shared line DN. Displaying the same call ID on all phones ensures that all users with the same shared line DN can identify the correct active call.

There is no administrator impact to this feature.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

Features available with latest Cisco Unified Communications Manager Device Pack

The following sections describe features in the release which require the new firmware and the latest Cisco Unified Communications Manager Device Pack.

For information about the Cisco Unified IP Phones and the required Cisco Unified Communications Manager device packs, see the following URL:

http:/​/​www.cisco.com/​en/​US/​docs/​voice_ip_comm/​cucm/​compat/​devpack_​comp_​mtx.html

Default Wallpaper Control

The Default Wallpaper Control feature allows the phone administrator to enable or disable the user preference to change the wallpaper on the phone. The phone administrator can also specify the default wallpaper image for all end users.

This feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

RTCP Control for Video

The RTCP Control for Video feature gives the administrator the flexibility to enable the phones to transmit and receive Real Time Control Protocol (RTCP) packets for audio and video streams in a video call. Using RTCP, instead of Real Time Transport Protocol (RTP), provides feedback and statistics that assist in phone system support. The choice of protocol does not impact the users. By default, the feature is disabled.

The administrator enables or disables the RTCP for video field from one of the following Cisco Unified Communications Manager windows:

  • Device > Phone
  • Device > Device Settings > Common Phone Profile

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951,and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

sRTP Secure Video

The sRTP Secure Video feature adds more media (audio and video) encryption capabilities and gives the administrator the flexibility to choose RTCP authentication tag length between 32 bit (default) and 80 bit from Cisco Unified CM Administration.

The administrator enables or disables (default) the 80-bit SRTCP field from one of the following Cisco Unified Communications Manager windows:

  • Device > Phone
  • System > Enterprise Phone Configuration
  • Device > Device Settings > Common Phone Profile

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971

This feature has no user impact.

Where to find more information

Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

Simplified New Call Bubble

The Simplified New Call Bubble feature provides a simplified window for the user to place an off-hook call. The administrator enables or disables the feature in the Phone Configuration window using the Simplified New Call UI field. By default, the feature is disabled.

When the user start dialing a call with the Simplified New Call Window, the phone does not display possible phone number matches from the call history.

The feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phone 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager 9.0 (SIP)

Single Tunnel for Cisco VXC VPN

The Cisco VXC VPN feature provides integrated VPN functionality for Cisco Virtualization Experience Clients (Cisco VXC) 2111 and 2112. The feature enables VPN tunneling for the Cisco VXC 2111 and Cisco VXC 2112 clients when they are attached to a Cisco Unified IP Phone 8961, 9951, or 9971.

You can configure the Cisco Unified IP Phone and the attached Cisco VXC client to share one VPN tunnel or to use two separate tunnels. To enable the VXC VPN feature, you must set up the VPN feature for the attached IP Phone in Cisco Unified Communications Manager Administration, using the submenus under the Advanced Features > VPN menu path. In addition, you must populate the Enable VXC VPN for MAC field, using the Phone Configuration Window (Device > Phone > Phone Configuration).

You can set up the Cisco VXC VPN and the phone VPN to use the same tunnel or separate tunnels, in the following configurations:

  • Dual Tunnel: Cisco VXC VPN traffic and phone VPN traffic use separate tunnels with the same access credentials. To ensure the highest quality of service for the phone voice and video services, Cisco recommends the Dual Tunnel setting, which is the default setting. With two VPN tunnels, the host Cisco Unified IP Phone can provide prioritization of CPU and memory resources to the data that associates with the phone voice and with video functions over the data that associates with the Cisco VXC VPN tunnel.
  • Single Tunnel: Cisco VXC VPN traffic and phone VPN traffic share one tunnel. All data travels over a single VPN tunnel by sharing the available phone processor and memory resources across the voice, video, and Cisco VXC services. The IP phone does not prioritize data handing of one service over another. As a result, possible performance degradation of the IP phone voice and video media handling and UI functions may occur due to IP phone CPU loading.

You can configure the feature to prompt the user only once for access credentials (using the Phone VPN Sign In window), or once each for the phone VPN (using the Phone VPN Sign In window) and the Cisco VXC VPN (using the VXC VPN Sign In window).

This feature is supported on the following SIP phones:

  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information

Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)

Minimum Cisco VXC Firmware Release Required

To support the VXC VPN feature, the Cisco VXC clients must be running the following minimum firmware releases:

  • Cisco VXC 2112: ICA Firmware Release 7.1_118
  • Cisco VXC 2111: PCoIP Firmware Release 4.0 (Q3CY12)
Network guidelines

The following are network guidelines for the Cisco VXC VPN feature implementation:

  • The MTU size in the phone VPN profile is a configurable value. The default value is 1290.
  • The maximum MTU value on the phone itself is hardcoded at 1406.
  • The MTU value must be less than or equal to 1406, but it should not be less than 576, because some IIS and virtualization servers do not accept values less than 576.
  • You must set up the firewall to allow the MTU value that you specify in the phone VPN profile.
  • If the phone cannot download the certificate file or the phone configuration file, check for the allowed packet size in the network.
  • If the Cisco VXC VPN cannot establish a tunnel, then ping the VPN concentrator IP address with a packet size (load) to match the MTU value that the VPN profile specifies.
  • If the ping fails, try another ping that specifies no load. If the ping still fails without the load, check the routing configuration.
  • If the ping fails only with the load included, check the firewall to ensure that it is configured to allow the required MTU.
  • Perform a traceroute to the VPN concentrator IP address, and then ping each route with the load to determine the source of the issue.
  • Ensure the Don’t Fragment (DF) bit is not set on the server, network, or IP phone VPN tunnel.
Static IP Fallback for Cisco VXC 2111

At power on, the Cisco VXC 2111 attempts to obtain an IP address using DHCP. The Static IP Fallback setting on the Cisco VXC 2111 determines the client behavior if it cannot obtain an IP address using DHCP.

The following table describes how the Static IP Fallback setting affects the Cisco VXC 2111 operations. The default setting is determined by the version of firmware that was originally running when the client was shipped.

Table 2 Cisco VXC 2111 Static IP Fallback behavior
Firmware on shipped client Default Static IP Fallback setting Cisco VXC 2111 behavior if DHCP unsuccesful at power on
Firmware release prior to 3.5.1 On After 2 minutes, the client stops sending DHCP requests and uses a self-defined static IP.
3.5.1 firmware or later Off The client keeps sending DHCP requests indefinitely.

The Static IP Fallback setting will be configurable with Cisco VXC Manager release 4.9.1 (available Q3CY12).


Note


If the client is factory reset, the setting will revert back to the original default setting listed.


View Call Logs From Shared Line

The Call History for Shared Line feature offers enhanced viewing of shared line activity in the Cisco Unified IP Phone call history. In addition to logging missed calls for a shared line, this feature will log all answered and placed calls on a shared line.

This feature is supported on the following SIP phones:
  • Cisco Unified IP Phone 8961
  • Cisco Unified IP Phone 9951
  • Cisco Unified IP Phone 9971
Where to find more information
  • Cisco Unified IP Phones 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 9.0 (SIP)
  • Cisco Unified IP Phones 8961, 9951, and 9971 User Guide for Cisco Unified Communications Manager Guide 9.0 (SIP)

Installation

Install latest Cisco Unified Communications Manager release

Before using the Cisco Unified IP Phone with Cisco Unified Communications Manager, your Cisco Unified Communications Manager servers must be running a version of the server software that supports the phones. All Cisco Unified Communications Manager servers in the cluster must support the phones. For information about the minimum Cisco Unified Communications Manager software version that the phone requires, see the introductory sections of these release notes.

For more information on Cisco Unified Communications Manager installations and upgrades, see the documents for your Cisco Unified Communications Manager version at the following location: http:/​/​www.cisco.com/​en/​US/​products/​sw/​voicesw/​ps556/​prod_​installation_​guides_​list.html

To download and install the Cisco Unified Communications Manager version, perform these steps.

Procedure
    Step 1   Go to the following URL:

    http:/​/​www.cisco.com/​cisco/​software/​navigator.html?mdfid=268439621&catid=278875240

    Step 2   Choose your Cisco Unified Communications Manager version.
    Step 3   Choose the appropriate software type.
    Step 4   Hover over the desired file. When the popup window displays, click the Readme link to open the readme file.
    Step 5   Choose Download or Add to cart for the desired file.
    Step 6   Use the instructions in the readme file to install the updated file on the Cisco Unified Communications Manager.

    Install firmware release on Cisco Unified Communications Manager

    Before using the Cisco Unified IP Phone firmware Release 9.3(1) with Cisco Unified Communications Manager, you must install the latest firmware on all Cisco Unified Communications Manager servers in the cluster.

    Procedure
      Step 1   Go to the following URL:

      http:/​/​www.cisco.com/​cisco/​software/​navigator.html?mdfid=268437892&flowid=5293

      Step 2   Depending on your phone model, choose Cisco Unified IP Phones 8900 Series or Cisco Unified IP Phones 9900 Series.
      Step 3   Choose your phone type.
      Step 4   Choose Session Initiation Protocol (SIP) Software.
      Step 5   In the Latest Releases folder, choose 9.3(1).
      Step 6   Select one of the following firmware files, click the Download Now or Add to cart button, and follow the prompts:
      • cmterm-8961.9-3-1-33.cop.sgn
      • cmterm-9951.9-3-1-33.cop.sgn
      • cmterm-9971.9-3-1-33.cop.sgn
      Note   

      If you added the firmware file to the cart, click the Download Cart link when you are ready to download the file.

      Step 7   Click the + next to the firmware file name in the Download Cart section to access additional information about this file. The hyperlink for the readme file is in the Additional Information section, which contains installation instructions for the corresponding firmware:
      • cmterm-8961.9-3-1-33-readme.html
      • cmterm-9951.9-3-1-33-readme.html
      • cmterm-9971.9-3-1-33-readme.html
      Step 8   Follow the instructions in the readme file to install the firmware.

      Install firmware zip files

      If a Cisco Unified Communications Manager is not available to load the installer program, the following .zip files are available to load the firmware.

      • cmterm-8961.9-3-1-33.zip
      • cmterm-9951.9-3-1-33.zip
      • cmterm-9971.9-3-1-33.zip

      Note


      Firmware upgrades over the WLAN interface may take longer than upgrades using a wired connection. Upgrade times over the WLAN interface may take more than an hour, depending on the quality and bandwidth of the wireless connection.


      Procedure
        Step 1   Go to the following URL: http:/​/​www.cisco.com/​cisco/​software/​navigator.html?mdfid=268437892&flowid=5293
        Step 2   Depending on your phone model, choose Cisco Unified IP Phones 8900 Series or Cisco Unified IP Phones 9900 Series.
        Step 3   Choose your phone type.
        Step 4   Choose Session Initiation Protocol (SIP) Software.
        Step 5   In the Latest Releases folder, choose 9.3(1).
        Step 6   Download the relevant zip files.
        Step 7   Unzip the files.
        Step 8   Manually copy the unzipped files to the directory on the TFTP server. See Cisco Unified Communications Operating System Administration Guide for information about how to manually copy the firmware files to the server.

        Cisco Unified Video Camera firmware

        The Cisco Unified Video Camera is supported on Cisco Unified Communications Manager Release 7.1(5) and later.

        Important Notes

        This section contains important information to consider with the Cisco Unified IP Phone 8961, 9951, and 9971.

        Plantronics Audio 615M Headset and Cisco Unified IP Phone 8961

        The Plantronics Audio 615M headset is not compatible with the Cisco Unified IP Phone 8961. You must use an alternate headset type for this IP phone. For more information, see CSCth71104.

        Plantronics CS50 USB Headset and Cisco Unified IP Color Key Expansion Module

        The Plantronics CS50 USB headset causes the phone to request power from the switch even though the headset is self powered. In this case, if a device such as a camera or expansion module is connected and active on the phone, the switch will reject the power request for the headset because the power budget has been exceeded. In this case, the headset cannot be used.

        Cisco Unified IP Phones 9951 and 9971 one-way video calls

        Because of limitations in the H.264 video signaling standards, Cisco Unified IP Phones 9951 and 9971 may not correctly display video that is received from devices supporting resolutions greater than 640 x 480. In this case, the user sees a black video screen.

        To ensure that video from such devices displays properly on the IP phone, configure high definition phones and Cisco Unified IP Phones 8961, 9951, and 9971 into different call regions and limit the video bandwidth to 384 kb/s when calling between regions.

        Cisco Virtualization Experience Client 2100

        The Cisco Virtualization Experience Client (VXC) 2100 Series are zero clients designed to deliver a user desktop from a centralized host server, providing access to desktop applications as if they were available locally. The Cisco VXC 2100 series attaches to the Cisco Unified IP Phone 8961, 9951, and 9971 through a spine connector cable.

        When running VXC with a single-tunnel option, high traffic to or from VXC may affect the phone's performance. Cisco Unified IP Phones 8961, 9951 and 9971 support 384kbps throughput bandwidth for VXC.

        Set VPN MTU to 1406 to reduce packet reassemble in Cisco Unified IP Phones 8961, 9951 and 9971, to improve bandwidth in limited packet rate.

        Video and audio applications may not play smoothly through VXC, even on a device plugged directly into a LAN.

        USB CD-ROMs and other USB storage devices have limited support. In the current VXC environment, the expected performance is a low bit/transfer rate for USB storage devices and USB CD-ROMs connected to the VXC device.

        For more information, see http:/​/​www.cisco.com/​en/​US/​products/​ps11499/​tsd_​products_​support_​series_​home.html.

        Multiple Text Messages

        If you send multiple text messages to 8961, 9951, and 9971 phones within a few seconds using CiscoIPPhoneExecute.xml, the phone may reply with an HTTP/1.1 503 Service Unavailable message, and the user may not receive the text message on their phone screen.

        No UDP for SIP support

        Cisco Unified IP Phones 8961, 9951, and 9971 do not support UDP for SIP. Do not configure the UDP protocol for SIP signaling on 8961, 9951, and 9971 phones.

        Secure Video Bandwidth When Calling Over VPN

        When using Cisco Unified IP Phones 8961, 9951, and 9971 for a secure video call over VPN and VXC VPN, the maximum supported bandwidth is 320 kbps.

        Turn Off VPN Before Downgrade

        The phone will go into an unregistered state if it is downgraded from a 9.3(1) load to a pre-9.2(3) load while VPN is turned on. Turn off VPN before downgrading to a pre-9.2(3) load.

        Caveats

        The following sections list the open and resolved caveats, and describe how to get further information on the caveats.

        Access Cisco Software Bug Toolkit

        Known problems (bugs) are graded according to severity level. These release notes contain descriptions of the following:

        • All severity level 1 or 2 bugs
        • Significant severity level 3 bugs

        You can search for problems by using the Cisco Software Bug Toolkit.

        To use the Software Bug Toolkit, follow these steps.

        Before You Begin

        To access Bug Toolkit, you need the following items:

        • Internet connection
        • Web browser
        • Cisco.com user ID and password
        Procedure
          Step 1   To access the Bug Toolkit, go to:

          http:/​/​tools.cisco.com/​Support/​BugToolKit/​action.do?hdnAction=searchBugs

          Step 2   Log in with your Cisco.com user ID and password.
          Step 3   To look for information about a specific problem, enter the bug ID number in the Search for Bug ID field, then click Go.

          Open caveats

          The following table lists severity 1, 2, and 3 defects that are open for the Cisco Unified IP Phones that use Firmware Release 9.3(1).

          For more information about an individual defect, you can access the online record for the defect by clicking the Identifier or going to the URL that is shown. You must be a registered Cisco.com user to access this online information.

          Because defect status continually changes, the table reflects a snapshot of the defects that were open at the time this report was compiled. For an updated view of open defects, access Bug Toolkit as described in Access Cisco Software Bug Toolkit

          Table 3 Open caveats for Firmware Release 9.3(1)
          Identifier Headline

          CSCty96858

          Call session label is inconsistent under various locale

          CSCty90733

          Tone is played incorrectly for digits pressing in whisper intercom call

          CSCty44400

          LED is still lighted after change phone button template

          CSCtv04593

          Conference list is still shown when it is disabled in FCP

          CSCtz98106

          Video will flash when call ended in some situation

          CSCtz90586

          EnergyWise will not work if IP address of phone is changed

          CSCtz85424

          IP addresses between 128.x.x.x-223.x.x.x are not supported in startMedia

          CSCtz82754

          Rings chirp1 and chirp2 have pop sounds

          CSCtz77224

          No standby server info when bring Sub up first then SRST

          CSCtz68986

          four direction arrow on TFTP input box and list item

          CSCtz54715

          Call history is listed for all lines but not the selected line

          CSCtz54317

          Call from call history list with one touch of list item

          CSCtz46348

          Still show CUCM domain name as active server when stop DNS and TFTP

          CSCtz34504

          Cursor is displayed in a wrong location in the intercom dial window

          CSCtz30127

          DHCP Release turn Yes first then it turn No when setting it to No

          CSCty76206

          Encrypted phone will not play zip tone after CFwdAll is input

          CSCty31537

          SenderReportsSent of "show stream active video" is incorrect

          CSCtz98112

          UI mess under 7.1(5) CCM when alerting name long

          CSCtz93333

          RT: Can't handle rtprx and rtpmrx with volume value

          CSCtz90864

          DN is logged in the remote shareline phone when offhook pickup call

          CSCtz90108

          Appear once press speaker key can't end the call

          CSCtz88254

          Password Persistence not effect on RT phones

          CSCtz85354

          RT can't handle senddigits event

          CSCtz79477

          VXC VPN alert stays in screen while loses focus and softkeys

          CSCtz76680

          [double shot ] No prompt for 4th "double shot" incoming call

          CSCtz71381

          WIFI toast an alert window overlapping with other UI menu

          CSCtz66049

          Redial does not work when there's an incoming intercom call

          CSCtz43088

          UI filter will focus on two items when intercom

          CSCtz22269

          ConcurrentModificationException when switching between the two calls

          CSCtz17167

          UI mess when offhook call from admin page

          CSCtz11498

          UI display incorrect after "Apply Config" followed by restart

          CSCty93030

          Mute tone instead of digit DTMF tone playing when mute caller

          CSCty89938

          Call time will count from -3 if lots of speed dial BLF

          CSCty69131

          remote placed call doesn't log DN in cross cluster if callee no answer

          CSCty56877

          Phone in trusted net try establish VPN when power on w/ auto net detect

          CSCti98208

          6911 SIP: No Alarm(rc=18) when secure phone fallback from SRST to CUCM

          CSCtz86079

          KEM default logging needs to detect crashes

          CSCtz86075

          3rd KEM reset/crashed while on active call

          CSCtz59235

          8961 incorrect time after NTP server time is changed

          CSCtr51513

          ETSGJ-CH: Conference message is showing in ENGLISH instead of JAPANESE

          CSCtx90826

          the loading of image fail of phone 9951/9971 in option66 case

          CSCtz65871

          ALL-LANG: SRST: 99xx: Directory is blank when restored from fallback.

          CSCti77356

          SRST: ALL-LANG: RT Phone: "Abbreviated Dial" softkey will disappear.

          CSCti77929

          Impossible to transfer a call from SCCP phone to SIP phone.

          CSCtz72610

          Phone Caches a Service Page Despite Expired Header Set as Page Expired

          CSCtz83811

          RT: experimental changes

          CSCtr93019

          Can't answer call through BT headset after hold revert

          CSCtr13418

          Phone keep alive timer issue in 9.2.(2) phone load

          CSCtz98079

          EM login logout time performance

          CSCtz82195

          Bluetooth Device Does Not Reconnect

          CSCtz68472

          Phone cant enter into debush when no active debugsh

          CSCtz71926

          Savi 7xx can't reconnect automatically after enable side USB on CUCM

          CSCua06066

          phone not response after enter debugsh

          CSCtw97451

          Blackwire C220 "accessory not supported"

          CSCtz51849

          Group pickup call in call history do not have call duration time

          CSCtz93469

          Touch screen issue with call history

          CSCtt05778

          No UI feedback when dialing external speed dial using softkey in onhook

          CSCtz45639

          A triangle is displayed on title bar of call history window

          CSCua08465

          Call duration time string is cut off in multi-leg call log detail

          CSCua11010

          PD displays abnormal when we press return button after log out.

          CSCua14036

          Softkey displayed as pressed after restart

          CSCty31023

          Dialed digits lost while making conf or xfer call quickly in video call

          CSCua00229

          9971 not transmitting RTP stream over the air

          CSCua13836

          Pip is messed display when swapping pip and remote video

          CSCts01615

          99xx become abnormal or crash after long period of network impairments

          CSCtn89145

          Joggling fullscreen selview during VGA video call to CSF softphone

          CSCty01167

          OpenSSL SSL_CTX_new Uninitialized Buffer Remote Information Disclosure

          CSCtz51332

          99xx/8961 takes long to to register after EM login in certain stuation

          CSCty57481

          SRST ip is still displayed on "Stand-by Server" even if srst is down

          CSCtz65557

          A calls B cannot get any re-order tone but silence for 500

          CSCty35673

          Xfer call bounces back when picked up @ ~same time as 2nd xfer keypress

          CSCti79116

          Memory leak during SIP Codenomicon run

          CSCtz82603

          8961/99XX firmware does not verify source address on Rx unicast RTP

          CSCua09003

          Phone does not send REFER alarm when it fallback to primary CUCM

          CSCtx48830

          RT phone lost wireless connectivity during call

          CSCty75302

          9971 does not connect to wireless when wired in (using PoE)

          CSCtz40373

          Call be disconnected when reauthen happen for phone

          CSCtz23376

          No toast prompt up when switch between wired/wireless network

          CSCua15471

          Sometimes dial window will lose focus and user can't input digit

          CSCua17545

          Phone may have a white screen when video call is quickly dropped

          CSCua25787

          Recording Failed toast appears twice in various fonts

          CSCua26548

          XSI API on CP-9971

          CSCua27527

          Directed call park timer start from -1 or -2 sometimes

          CSCua27553

          Call Quality is Reduced When Upgrade Phone Load

          CSCua30357

          RTCP port will keep open when posting RTPRx/RTPTx to phone then end call

          CSCua33239

          UI mess when end the directed call park

          CSCua22427

          Top row of pixels not updated when modify SD label of LKEM

          CSCts51576

          9971 differ from 7945 on rtp sequence jump behavior

          Resolved caveats

          The following table lists severity 1, 2, and 3 defects that are resolved for the Cisco Unified IP Phones that use Firmware Release 9.3(1).

          For more information about an individual defect, you can access the online record for the defect by clicking the Identifier or going to the URL that is shown. You must be a registered Cisco.com user to access this online information.

          Because defect status continually changes, the table reflects a snapshot of the defects that were open at the time this report was compiled. For an updated view of open defects, access Bug Toolkit as described in Access Cisco Software Bug Toolkit

          Table 4 Resolved caveats for Firmware Release 9.3(1)
          Identifier Headline

          CSCtt18467

          Phone syslogs prints NOT[ice] level always

          CSCtu49998

          VXC 2111 and 2112 can't get power from phone and lkem

          CSCtu53630

          9971 May Intermittently Drop from WLAN and Begin Scanning

          CSCtw65250

          Video call with TP, PiP is transparent or all green

          CSCtw99539

          Phone may not switch to new load after upgrade

          CSCtx20913

          PiP window flashes once or twice sometimes at beginning of video call

          CSCtx66216

          Bandwidth calcuation not correct when multiple sessions exist

          CSCtx82992

          Pressing the Back button has no effect when on Edit Dial UI

          CSCtx84153

          Low performance if configure BLF on RT KEM module registered to CME

          CSCtx85612

          Lock icon may be displayed on non-secure idle UI

          CSCty03846

          TFTP Server inputbox is not activated while enable alt-tftp

          CSCty10161

          Green strip is displayed when switching from full screen to window

          CSCty25822

          Unable to show video of video call made with SIPp

          CSCty34439

          99XX phone reboot due to chinese locale

          CSCty35299

          Same jad or jar file is always downloaded twice by phone

          CSCty77693

          no toast when calls fwd to 2nd line

          CSCty60735

          Phone retries using same bad password if it contains space

          CSCty83995

          Disable Speakerphone Parameter Allows Audio Output on 9971

          CSCty91081

          FAC and CMC only in speed dial not work

          CSCty96383

          Phone may freeze if restart it while pressing Release to get new call UI

          CSCtz19102

          9971 phone - one-way video when called MCU

          CSCtz20164

          RT phones cannot display low-res video from Jabber for Windows

          CSCtz22348

          Incorrect DSCP marking on IP Phone 9971 during recording

          CSCtz23146

          Non-normal used IP can't be displayed in TFTP address

          CSCtz25519

          CR/LF in QED XML cause could not save device configuration

          CSCtz31444

          Group pickuped call will be displayed "Unknown Caller" in call history

          CSCtz43376

          vpn tunnel can't be setup for route info is wrong in kernel

          CSCtz45859

          Intercom call can't be ended by line triggered by speed dial

          CSCtz60278

          the function of 'Display on when Incoming Call' can't work

          CSCtz64608

          RT: All Calls enabled, CID disappears w/ xfer to final called party

          CSCtz74624

          PiP and call statistics lost in RT video phones

          CSCtz77357

          Connect USB headset on usb hub 'Max num exceeded' display and not work

          CSCtz84937

          Mute LED showed lit, but device was unmuted

          CSCtz87589

          TVS Client should support same size certificate as CUCM

          CSCtz96238

          Phone still send out video when mute is on

          Documentation, service requests, and additional information

          For information on obtaining documentation, submitting a service request, and gathering additional information, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:

          http:/​/​www.cisco.com/​en/​US/​docs/​general/​whatsnew/​whatsnew.html

          Subscribe to the What’s New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free service and Cisco currently supports RSS Version 2.0.