 Feedback
|
Mobile Downloads
Contents
- Cisco Unified IP Phone 8961, 9951, and 9971 Release Notes for Firmware Release 9.3(1)
- Introduction
- Related documentation
- Cisco Unified IP Phones 8900 Series documentation
- Cisco Unified IP Phone 9900 Series documentation
- Cisco Unified Communications Manager documentation
- Cisco Business Edition 5000 documentation
- Cisco Virtualization Experience Client 2000 Series documentation
- New and Changed Features
- Features available with firmware release
- Assured Services SIP
- Display Survivable Remote Site Telephony Message
- Device Invoked Recording
- Edit Speed Dial Without Restart
- Extension Mobility Cross Cluster Enhancement
- Handset Bass Adjustment
- Pause In Speed Dial
- PLK Support for Queue Statistics
- RTCP Hold On SIP
- Secure Extension Mobility Cross Cluster
- SIP Phone No Alert Name in Placed Calls History
- Uniform Resource Identifier Dialing
- Unique Call ID Display
- Features available with latest Cisco Unified Communications Manager Device Pack
- Default Wallpaper Control
- RTCP Control for Video
- sRTP Secure Video
- Simplified New Call Bubble
- Single Tunnel for Cisco VXC VPN
- Minimum Cisco VXC Firmware Release Required
- Network guidelines
- Static IP Fallback for Cisco VXC 2111
- View Call Logs From Shared Line
- Installation
- Install latest Cisco Unified Communications Manager firmware
- Install firmware release on Cisco Unified Communications Manager
- Install firmware zip files
- Cisco Unified Video Camera firmware
- Important Notes
- Plantronics Audio 615M Headset and Cisco Unified IP Phone 8961
- Plantronics CS50 USB Headset and Cisco Unified IP Color Key Expansion Module
- Cisco Unified IP Phones 9951 and 9971 one-way video calls
- Cisco Virtualization Experience Client 2100
- Multiple Text Messages
- No UDP for SIP support
- Secure Video Bandwidth When Calling Over VPN
- Turn Off VPN Before Downgrade
- Caveats
- Access Cisco Software Bug Toolkit
- Open caveats
- Resolved caveats
- Documentation, service requests, and additional information
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.
Related documentation
- Cisco Unified IP Phones 8900 Series documentation
- Cisco Unified IP Phone 9900 Series documentation
- Cisco Unified Communications Manager documentation
- Cisco Business Edition 5000 documentation
- Cisco Virtualization Experience Client 2000 Series documentation
Cisco Unified IP Phones 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
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
- Features available with firmware release
- Features available with latest Cisco Unified Communications Manager Device Pack
Features available with firmware release
- Assured Services SIP
- Display Survivable Remote Site Telephony Message
- Device Invoked Recording
- Edit Speed Dial Without Restart
- Extension Mobility Cross Cluster Enhancement
- Handset Bass Adjustment
- Pause In Speed Dial
- PLK Support for Queue Statistics
- RTCP Hold On SIP
- Secure Extension Mobility Cross Cluster
- SIP Phone No Alert Name in Placed Calls History
- Uniform Resource Identifier Dialing
- Unique Call ID Display
Assured Services 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:
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:
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:
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:
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:
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:
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:
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:
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:
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):
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:
NoteWait for the ABC softkey to appear before you proceed with URI dialing.
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:
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
- RTCP Control for Video
- sRTP Secure Video
- Simplified New Call Bubble
- Single Tunnel for Cisco VXC VPN
- View Call Logs From Shared Line
Default Wallpaper Control
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:
The feature is supported on the following SIP phones:
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:
The feature is supported on the following SIP phones:
This feature has no user impact.
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:
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 menu path. In addition, you must populate the Enable VXC VPN for MAC field, using the Phone Configuration Window ().
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:
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).
NoteIf the client is factory reset, the setting will revert back to the original default setting listed.
Installation
- Install latest Cisco Unified Communications Manager firmware
- Install firmware release on Cisco Unified Communications Manager
Install latest Cisco Unified Communications Manager firmware
ProcedureBefore using the Cisco Unified IP Phone with Cisco Unified Communications Manager, you must install the latest firmware on all Cisco Unified Communications Manager servers in the cluster. For more information, see the Cisco Unified Communications Manager installation and upgrade guides.
To download and install the Cisco Unified Communications Manager version, perform these steps.
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
ProcedureBefore 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.
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:
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: Step 8 Follow the instructions in the readme file to install the firmware.
Install firmware zip files
ProcedureIf a Cisco Unified Communications Manager is not available to load the installer program, the following .zip files are available to load the firmware.
NoteFirmware 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.
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.
Important Notes
- Plantronics Audio 615M Headset and Cisco Unified IP Phone 8961
- Plantronics CS50 USB Headset and Cisco Unified IP Color Key Expansion Module
- Cisco Unified IP Phones 9951 and 9971 one-way video calls
- Cisco Virtualization Experience Client 2100
- Multiple Text Messages
- No UDP for SIP support
- Secure Video Bandwidth When Calling Over VPN
- Turn Off VPN Before Downgrade
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.
Caveats
Access Cisco Software Bug Toolkit
ProcedureKnown problems (bugs) are graded according to severity level. These release notes contain descriptions of the following:
You can search for problems by using the Cisco Software Bug Toolkit.
To use the Software Bug Toolkit, follow these steps.
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 Call session label is inconsistent under various locale
Tone is played incorrectly for digits pressing in whisper intercom call
LED is still lighted after change phone button template
Conference list is still shown when it is disabled in FCP
Video will flash when call ended in some situation
EnergyWise will not work if IP address of phone is changed
IP addresses between 128.x.x.x-223.x.x.x are not supported in startMedia
Rings chirp1 and chirp2 have pop sounds
No standby server info when bring Sub up first then SRST
four direction arrow on TFTP input box and list item
Call history is listed for all lines but not the selected line
Call from call history list with one touch of list item
Still show CUCM domain name as active server when stop DNS and TFTP
Cursor is displayed in a wrong location in the intercom dial window
DHCP Release turn Yes first then it turn No when setting it to No
Encrypted phone will not play zip tone after CFwdAll is input
SenderReportsSent of "show stream active video" is incorrect
UI mess under 7.1(5) CCM when alerting name long
RT: Can't handle rtprx and rtpmrx with volume value
DN is logged in the remote shareline phone when offhook pickup call
Appear once press speaker key can't end the call
Password Persistence not effect on RT phones
RT can't handle senddigits event
VXC VPN alert stays in screen while loses focus and softkeys
[double shot ] No prompt for 4th "double shot" incoming call
WIFI toast an alert window overlapping with other UI menu
Redial does not work when there's an incoming intercom call
UI filter will focus on two items when intercom
ConcurrentModificationException when switching between the two calls
UI mess when offhook call from admin page
UI display incorrect after "Apply Config" followed by restart
Mute tone instead of digit DTMF tone playing when mute caller
Call time will count from -3 if lots of speed dial BLF
remote placed call doesn't log DN in cross cluster if callee no answer
Phone in trusted net try establish VPN when power on w/ auto net detect
6911 SIP: No Alarm(rc=18) when secure phone fallback from SRST to CUCM
KEM default logging needs to detect crashes
3rd KEM reset/crashed while on active call
8961 incorrect time after NTP server time is changed
ETSGJ-CH: Conference message is showing in ENGLISH instead of JAPANESE
the loading of image fail of phone 9951/9971 in option66 case
ALL-LANG: SRST: 99xx: Directory is blank when restored from fallback.
SRST: ALL-LANG: RT Phone: "Abbreviated Dial" softkey will disappear.
Impossible to transfer a call from SCCP phone to SIP phone.
Phone Caches a Service Page Despite Expired Header Set as Page Expired
RT: experimental changes
Can't answer call through BT headset after hold revert
Phone keep alive timer issue in 9.2.(2) phone load
EM login logout time performance
Bluetooth Device Does Not Reconnect
Phone cant enter into debush when no active debugsh
Savi 7xx can't reconnect automatically after enable side USB on CUCM
phone not response after enter debugsh
Blackwire C220 "accessory not supported"
Group pickup call in call history do not have call duration time
Touch screen issue with call history
No UI feedback when dialing external speed dial using softkey in onhook
A triangle is displayed on title bar of call history window
Call duration time string is cut off in multi-leg call log detail
PD displays abnormal when we press return button after log out.
Softkey displayed as pressed after restart
Dialed digits lost while making conf or xfer call quickly in video call
9971 not transmitting RTP stream over the air
Pip is messed display when swapping pip and remote video
99xx become abnormal or crash after long period of network impairments
Joggling fullscreen selview during VGA video call to CSF softphone
OpenSSL SSL_CTX_new Uninitialized Buffer Remote Information Disclosure
99xx/8961 takes long to to register after EM login in certain stuation
SRST ip is still displayed on "Stand-by Server" even if srst is down
A calls B cannot get any re-order tone but silence for 500
Xfer call bounces back when picked up @ ~same time as 2nd xfer keypress
Memory leak during SIP Codenomicon run
8961/99XX firmware does not verify source address on Rx unicast RTP
Phone does not send REFER alarm when it fallback to primary CUCM
RT phone lost wireless connectivity during call
9971 does not connect to wireless when wired in (using PoE)
Call be disconnected when reauthen happen for phone
No toast prompt up when switch between wired/wireless network
Sometimes dial window will lose focus and user can't input digit
Phone may have a white screen when video call is quickly dropped
Recording Failed toast appears twice in various fonts
XSI API on CP-9971
Directed call park timer start from -1 or -2 sometimes
Call Quality is Reduced When Upgrade Phone Load
RTCP port will keep open when posting RTPRx/RTPTx to phone then end call
UI mess when end the directed call park
Top row of pixels not updated when modify SD label of LKEM
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 Phone syslogs prints NOT[ice] level always
VXC 2111 and 2112 can't get power from phone and lkem
9971 May Intermittently Drop from WLAN and Begin Scanning
Video call with TP, PiP is transparent or all green
Phone may not switch to new load after upgrade
PiP window flashes once or twice sometimes at beginning of video call
Bandwidth calcuation not correct when multiple sessions exist
Pressing the Back button has no effect when on Edit Dial UI
Low performance if configure BLF on RT KEM module registered to CME
Lock icon may be displayed on non-secure idle UI
TFTP Server inputbox is not activated while enable alt-tftp
Green strip is displayed when switching from full screen to window
Unable to show video of video call made with SIPp
99XX phone reboot due to chinese locale
Same jad or jar file is always downloaded twice by phone
no toast when calls fwd to 2nd line
Phone retries using same bad password if it contains space
Disable Speakerphone Parameter Allows Audio Output on 9971
FAC and CMC only in speed dial not work
Phone may freeze if restart it while pressing Release to get new call UI
9971 phone - one-way video when called MCU
RT phones cannot display low-res video from Jabber for Windows
Incorrect DSCP marking on IP Phone 9971 during recording
Non-normal used IP can't be displayed in TFTP address
CR/LF in QED XML cause could not save device configuration
Group pickuped call will be displayed "Unknown Caller" in call history
vpn tunnel can't be setup for route info is wrong in kernel
Intercom call can't be ended by line triggered by speed dial
the function of 'Display on when Incoming Call' can't work
RT: All Calls enabled, CID disappears w/ xfer to final called party
PiP and call statistics lost in RT video phones
Connect USB headset on usb hub 'Max num exceeded' display and not work
Mute LED showed lit, but device was unmuted
TVS Client should support same size certificate as CUCM
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.
