Aastra Models 6700i and 9000i Series SIP IP Phones. SIP Service Pack 1 Release Notes

Transcription

1 Aastra Models 6700i and 9000i Series SIP IP Phones SIP Service Pack 1 Release Notes RN REV

2 Content SIP IP Phone Models 6700i and 9000i Series Phones Release Notes Service Pack (SP) About this document Release notes topics General information Release content information Hardware supported Bootloader requirements Before you upgrade New Features in Release SP Configuration Features SIP Features Additional Information Configuration Features Dynamic Whitelist Option to Include/Remove Silence Suppression Attribute from SDP Offer SIP Features Multiple Voic Registration Active Voice-over-IP (VoIP) Recording Issues Resolved in Release SP Contacting Aastra Telecom Support II RN REV

3 SIP IP Phone Models 6700i and 9000i Series Phones Release Notes Service Pack (SP)1 About this document This document provides details on new features and/or issues resolved for the 6700i Series SIP IP Phones (6730i, 6731i, 6739i, 6753i, 6755i, 6757i, 6757i CT) and the 9000i Series SIP IP Phones (9143i, 9480i, 9480i CT) for Release SP1. Notes: 1. This release applies to the phone models mentioned above only i and 6737i are not supported in this release. For more detailed information about features associated with each phone, and for information on how to use the phones, see your model-specific SIP IP Phone Installation Guide and the SIP IP Phone User Guide. For detailed information about more advanced features, see the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide and/or the Development Guide XML API For Aastra SIP Phones Firmware Release notes topics Topics in these release notes include: General information New Features in Release SP1 Additional Information Issues Resolved in Release SP1 Contacting Aastra Telecom Support RN REV

4 SIP IP Phone Models 6700i and 9000i Series Phones Release Notes General information Release content information This document provides release content information on the Aastra 9000i and 6700i Series SIP IP Phone firmware. Model Release name Release version Release filename Release date 6730i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i CT Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i Generic SIP SP1 FC REV07 January i CT Generic SIP SP1 FC REV07 January 2013 Hardware supported This release of firmware is compatible with the following Aastra IP portfolio products: 673xi Models 675xi Models 9xxxi Models 6730i 6753i 9143i 6731i 6755i 9480i 6739i 6757i 9480i CT 6757i CT 2 RN REV

5 SIP IP Phone Models 6700i and 9000i Series Phones Release Notes Bootloader requirements This release of firmware is compatible with the following Aastra IP portfolio product bootloader versions: 673xi Models 675xi Models 9xxxi Models 6730i: Bootloader or higher 6731i: Bootloader or higher 6739i: Bootloader or higher 6753i: Bootloader or higher 6755i: Bootloader or higher 6757i: Bootloader or higher 6757i CT: Bootloader or higher 9143i: Bootloader or higher 9480i: Bootloader or higher 9480i CT: Bootloader or higher RN REV

6 SIP IP Phone Models 6700i and 9000i Series Phones Release Notes Before you upgrade Please read before upgrading your phone Note: The following information applies to all IP phones EXCEPT the 6739i. If you have a firmware version on your phone prior to 2.3, please read the following IMPORTANT information before upgrading the phones: LLDP is enabled by default If LLDP is enabled on your network, the phones may come up with different network settings. For more information about LLDP, see the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide. Support for DHCP Options 159 and 160 If the DHCP server supplies Options 159 and 160, the phones will attempt to contact the configuration server given in these options. For more information about Options 159 and 160, see the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide. HTTPS validation If you are using HTTPS and the certificates are not valid or are not signed by Verisign, Thawte, or GeoTrust, the phones fail to download configuration files. For more information about HTTPS validation, see the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide. WatchDog task feature If the phone detects a failure (for example, a crash), the phone automatically reboots. For more information about the WatchDog feature, see the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide. Note: If you factory default a phone with Release 2.3 and above software, when the phone reboots, it attempts to connect to rcs.aastra.com. There is no personal information transmitted from the phone and the phone continues to boot up as normal. Warning: Applicable to IP Phone Models 6730i and 6731i only: The default negotiation setting for the Ethernet ports on the phones is auto-negotiation. If you have changed this default setting to another value (i.e. Half Duplex or Full Duplex), you MUST set the negotiation value back to the default value of autonegotiation before upgrading to Release SP1. Failure to do so may cause the phone to fail to connect to the network. In addition, downgrade will not be possible. 4 RN REV

7 New Features in Release SP1 This section provides the new features in SIP IP Phone Release SP1. The following table summarizes each new feature and provides a link to more information within this release note. Each feature also specifies whether it affects the Administrator, the User, or the XML Developer. This table may also provide the documentation location of features that have already been documented in Aastra s documentation suite. Refer to those documents for more information about the applicable feature. Feature Description Configuration Features Dynamic Whitelist (For Administrators) Option to Include/Remove Silence Suppression Attribute from SDP Offer (For Administrators) Previously, if the Whitelist Proxy feature was enabled, the whitelist would be loaded only during the phone s boot up process. With Release SP1, dynamic whitelist support has been implemented, which allows the phones to refresh and resync the whitelist (without a reboot) if the IP addresses corresponding to the proxy FQDNs are changed on the DNS server. The parameter sip remove silence suppression offer has been introduced in Release SP1 allowing administrators the ability to control whether or not the silence suppression attribute should be included in the Session Description Protocol (SDP) offer. SIP Features Multiple Voic Registration (For Administrators and Users) Active Voice-over-IP (VoIP) Recording (For Administrators and Users) Multiple voic registration is now supported on the IP phones in Release SP1. By configuring a softkey or programmable key as Speeddial/MWI and defining call and voic URIs, users can monitor and listen to pending messages on multiple voic accounts. Active VoIP recording is now supported by the IP phones in Release SP1. When using the IP phones with the Aastra MX-ONE call manager (v4.1 or 5.0) and ASC s EVOip 10.0 voice recording system, administrators can configure the phones to send duplicate copies of the transmit and receive RTP or SRTP voice packets to the voice recording system. RN REV

8 Additional Information Configuration Features Dynamic Whitelist To protect your IP phone network, you can configure a Whitelist Proxy feature that screens incoming call requests received by the IP phones. When this feature is enabled, an IP phone only accepts call requests from a trusted proxy server and rejects any call requests from an untrusted proxy server. Previously, if the Whitelist Proxy feature was enabled, the whitelist would be loaded only during the phone s boot up process. With Release SP1, dynamic whitelist support has been implemented, which allows the phones to refresh and resync the whitelist (without a reboot) if the IP addresses corresponding to the proxy FQDNs are changed on the DNS server. The IP phone monitors the following events and an update is triggered if required: 200 OK responses to REGISTER requests if the peer s IP address is not currently in the whitelist. 200 OK responses to INVITE requests if the peer s IP address is not currently in the whitelist. INVITE requests from untrusted proxy servers. Note: An update will not be triggered for the following events: 200 OK responses to INVITE requests for an IP call. When more than one INVITE requests are received from the same untrusted proxy server. Administrators can enable the Whitelist Proxy feature through the Aastra Web UI or by defining the sip whitelist parameter in the configuration files. The Whitelist Proxy feature is disabled by default. 6 RN REV

9 Additional Information Enabling the Whitelist Proxy Feature Using the Aastra Web UI Use the following procedure to configure the Whitelist Proxy feature using the Aastra Web UI. Aastra Web UI 1. Click on Advanced Settings -> Global SIP -> Advanced SIP Settings. 2. The "Whitelist Proxy" field is disabled by default. To enable this field, check the box. When this feature is enabled, the IP phone only accepts call requests from a trusted proxy server. The IP phone rejects any call requests from an untrusted proxy server. 3. Click Save Settings to save your changes. RN REV

10 Additional Information Configuring the Whitelist Proxy Feature Using the Configuration Files Use the following parameter to enable/disable the Whitelist Proxy feature: Parameter sip whitelist Description Format Default Value Range Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg This parameter enables/disables the whitelist proxy feature, as follows: Set to 0 to disable the feature. Set to 1 to enable the feature. When this feature is enabled, the IP phone only accepts call requests from a trusted proxy server. The IP phone rejects any call requests from an untrusted proxy server. Boolean 0 (Disabled) 0 (Disabled) 1 (Enabled) Example sip whitelist: 1 8 RN REV

11 Additional Information Option to Include/Remove Silence Suppression Attribute from SDP Offer The parameter sip remove silence suppression offer has been introduced in Release SP1 allowing administrators the ability to control whether or not the silence suppression attribute should be included in the Session Description Protocol (SDP) offer. If enabled (1), the silence suppression attribute will be removed from the SDP offer. If disabled (0), the attribute will not be removed from the SDP offer. This parameter is disabled by default and requires a reboot if the value of the parameter has changed. Including/Removing the Silence Suppression Attribute from the SDP Offer Use the following parameter to include/remove the silence suppression attribute from the SDP offer: Parameter sip remove silence suppression offer Description Format Default Value Range Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg Specifies whether or not the silence suppression attribute should be included in the Session Description Protocol (SDP) offer. If enabled, the silence suppression attribute will be removed from the SDP offer. If disabled, the attribute will not be removed from the SDP offer. Note: If the value of this parameter has changed, a reboot will be required for the change to take effect. Boolean 0 (Disabled) 0 (Disabled) 1 (Enabled) Example sip remove silence suppression offer: 1 RN REV

12 Additional Information SIP Features Multiple Voic Registration Multiple voic registration is now supported on the IP phones in Release SP1. This feature can be useful in scenarios where a user needs to monitor the voic accounts of his/her team members or an assistant requires access to his/her manager s voic messages. By configuring a softkey, programmable key, or expansion module softkey as Speeddial/MWI and defining call and voic URIs, users can monitor and listen to pending messages on multiple voic accounts. When new messages are pending on a monitored voic account the corresponding Speeddial/MWI key s LED will blink and the UI (for softkeys) will display an envelope icon and the number of pending messages beside the defined label. When a user presses the softkey, the phone will send an INVITE to the configured call URI whereby the user will be able to listen to the new messages. Users can configure the Speeddial/MWI key through the Aastra Web UI while Administrators can configure the key through the Aastra Web UI as well as the configuration files. Note: For the 6739i IP phone, the Speeddial/MWI key can be configured directly using the IP phone s UI. 10 RN REV

13 Additional Information Configuring a Speeddial/MWI Key Using the Aastra Web UI Use the following procedure to configure a Speeddial/MWI key using the Aastra Web UI: Aastra Web UI 1. Click on Click on Operation->Softkeys and XML. or Click on Operation-> Programmable Keys. or Click on Operation->Expansion Module Keys. 2. In the Type field, select Speeddial/MWI from the list of options. 3. If applicable, in the Label field, enter a key label to assign to the Speeddial/MWI key. When messages are pending, the IP phone UI will display an envelope icon, the number of pending messages, and then the defined label (e.g. 3 Peter). 4. In the Value field, enter in the call URI and voic URI separated by a semi-colon, as per the following syntax: [call URI];[voic URI]. For example, ,,,3456#0000#@domain;sip:voic _peter@domain Notes: As the example above illustrates, pauses and DTMF are supported for the call URI. Ensure that no spaces are added between the call URI and the voic URI when defining the key value. If only one URI is provided, the value will be used for the voic URI and the call URI will be left as undefined. 5. In the Line field, select the line for which you want to use the key functionality. 6. For phones with softkeys: In the States field, select the state(s) (idle, connected, incoming, outgoing, busy) for which you want to use on the key. Note: States are not applicable to programmable keys. 7. Click Save Settings to save your settings. RN REV

14 Additional Information Configuring a Speeddial/MWI Key Using the IP Phone UI (6739i only) Use the following procedure to configure a Speeddial/MWI key using the 6739i IP phone s UI: IP Phone UI 1. Press on the phone to enter the Options menu. 2. Select Softkeys. By default, all of the softkeys that display are configured as None. 3. Press a <None> key. A softkey configuration screen displays. 4. In the Type field, press the <None> key. A screen displays with softkey keys. 5. Press the <Speeddial/MWI> function key to apply to the softkey. 6. In the Label field, enter a key label to assign to the Speeddial/MWI key. When messages are pending, the IP phone UI will display an envelope icon of pending messages, and then the defined label (e.g. 3 Peter)., the number 7. In the Value field, enter in the call URI and voic URI separated by a semi-colon, as per the following syntax: [call URI];[voic URI]. For example, ,,,3456#0000#@domain;sip:voic _peter@domain Notes: As the example above illustrates, pauses and DTMF are supported for the call URI. Ensure that no spaces are added between the call URI and the voic URI when defining the key value. If only one URI is provided, the value will be used for the voic URI and the call URI will be left as undefined. 8. In the Line field, press the or to select a line for which to apply to the softkey. Valid values are 1 through Press <Save> to save the softkey. The softkey applies to your phone immediately and displays on your idle screen. 10.Press the to return to the previous menu or press the key to return to the idle screen. 12 RN REV

15 Additional Information Configuring the Speeddial/MWI Key Using the Configuration Files To configure the Speeddial/MWI key using the configuration files, you must enter speeddialmwi for the key type. For the label, enter a key label to assign to the Speeddial/MWI key (e.g. Peter). For the value, enter in the call URI and voic URI separated by a semi-colon, as per the following syntax: [call URI];[voic URI]. For example: ,,,3456#0000#@domain;sip:voic _peter@domain Notes: As the example above illustrates, pauses and DTMF are supported for the call URI. Ensure that no spaces are added between the call URI and the voic URI when defining the key value. If only one URI is provided, the value will be used for the voic URI and the call URI will be left as undefined. For the line, enter the line for which you want to use the key functionality (e.g. 1 through 9). The following parameters are examples you can use to configure the Speeddial/MWI key using the configuration files: For Softkeys softkey1 type: speeddialmwi softkey1 label: Peter softkey1 value: ,,,3456#0000#@domain;sip:voic _peter@domain softkey1 line: 3 For Top Softkeys topsoftkey1 type: speeddialmwi topsoftkey1 label: Peter topsoftkey1 value: ,,,3456#0000#@domain;sip:voic _peter@domain topsoftkey1 line: 3 For Programmable Keys prgkey1 type: speeddialmwi prgkey1 value: ,,,3456#0000#@domain;sip:voic _peter@domain prgkey1 line: 3 RN REV

16 Additional Information For Expansion Module Softkeys expmod1 key1 type: speeddialmwi expmod1 key1 label: Peter expmod1 key1 value: expmod1 key1 line: 3 Note: For more information on how to configure softkeys, programmable keys, and expansion module softkeys using the configuration files, please refer to the Aastra Models 9000i and 6700i Series SIP IP Phones Administrator Guide. 14 RN REV

17 Additional Information Active Voice-over-IP (VoIP) Recording Active VoIP recording is now supported by the IP phones in Release SP1. When using the IP phones with an Aastra call manager supporting voice recording and a recording system with the predefined subset of the SIP interface, administrators can configure the phones to send duplicate copies of the transmit and receive RTP or SRTP voice packets to the voice recording system. Notes: Currently, for Release SP1, the active VoIP recording feature is only supported when using the Aastra MX-ONE call manager (v4.1 or 5.0) in conjunction with ASC s EVOip 10.0 voice recording system. Support for additional call managers and voice recording systems will be implemented in future releases. The active VoIP recording feature is disabled by default. Both dynamic (i.e. per call) and static (i.e. per the duration that the phone is registered) recording sessions are supported by the IP phones. Additionally, administrators have the option of enabling a Record-On-Demand feature allowing users to initiate and terminate a call recording session at their discretion. The call recording sessions are initiated by the voice recording system and when the session is established, the IP phone will duplicate all of its incoming and outgoing RTP/SRTP packets and send them to the voice recording system where they can be archived and analyzed as required. Notes: Please contact your Aastra MX-ONE account manager for details on how to configure and utilize the Record-On-Demand feature. As the RTP/SRTP packets sent to the voice recording system are duplicate copies, the codec used for the original call as well as the recording are identical as well. If active VoIP recording is required, ensure that the IP phone is configured to use the G.711 or G.729 codec as these are currently the only two codecs supported by ASC s EVOip 10.0 voice recording system. Administrators must configure a whitelist for voice recording system authentication using the recorder addressn parameters (where N is a number from 1 to 6). These parameters are used to specify trusted IP addresses corresponding to the voice recording system. The IP phone will check and respond to SIP messages coming from these IP addresses. If all of these parameters are left undefined, the active VoIP recording feature is disabled. RN REV

18 Additional Information A whitelist can also be configured for RTP/SRTP packet destination authentication using the recording destinationn parameters (where N is a number from 1 to 6). These parameters are used to specify trusted IP addresses corresponding to the destination where the RTP/SRTP packets should be sent. The IP phone will check to see if the destination IP addresses are trusted before sending the duplicated RTP/SRTP packets. If all of these parameters are left undefined, no authentication checks will be performed. When a recording session is in progress, the IP phones display a recording icon on screen. Moreover, the phone will, by default, play a periodic audible beep tone through the selected audio path notifying users that their call is being recorded. Playback of the beep tone is optional and if required, administrators can disable the beep tone by defining the recording periodic beep parameter as 0 in the configuration files. Note: In addition to the aforementioned parameters corresponding to the active IP voice recording feature, the transport protocol parameters for SIP services (i.e. sip services transport protocol and sip services port ) must also be defined in the configuration files. 16 RN REV

19 Additional Information Configuring Active VoIP Recording Support Use the following parameters to configure active VoIP recording support: Voice Recording System Authentication Whitelist Settings Parameter recorder addressn Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg (N is a number from 1 to 6) Description Format Default Value Range Specifies a trusted IP address (maximum of six) corresponding to the voice recording system. The IP phone will check and respond to SIP messages coming from these IP addresses on the port defined by the sip services port parameter. Note: If all of the recorder addressn parameters are left undefined, the active IP voice recording feature is disabled. IP Address N/A N/A Example recorder address1: recorder address2: recorder address3: recorder address4: recorder address5: recorder address6: RN REV

20 Additional Information RTP/SRTP Packet Destination Authentication Whitelist Settings Parameter recording destinationn Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg (N is a number from 1 to 6) Description Format Default Value Range Specifies trusted IP addresses (maximum of six) corresponding to the destination where the RTP/SRTP packets should be sent. The IP phone will check to see if the destination IP addresses are trusted before sending the duplicated RTP/SRTP packets. Note: If all of these parameters are left undefined, no authentication checks will be performed. IP Address N/A N/A Example recording destination1: recording destination2: recording destination3: recording destination4: recording destination5: recording destination6: Recording Indicator (Periodic Beep Tone) Setting Parameter recording periodic beep Description Format Default Value Range Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg Enables or disables the periodic audible recording indicator (beep tone) notifying users that their call is being recorded. Integer 1 (Enabled) 0 (Disabled) 1 (Enabled) Example recording periodic beep: 0 18 RN REV

21 Additional Information Transport Protocol for SIP Services Settings Parameter sip services transport protocol Description Format Default Value Range Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg Specifies the transport protocol used for SIP services. By default, this parameter is set to -1 (invalid) whereby SIP services use the same transport protocol as defined in the sip transport protocol parameter. Numeric -1 (Invalid) -1 (Invalid) 0 (TCP/UDP) 1 (UDP) 2 (TCP) Example sip services transport protocol: 1 Parameter sip services port Description Format Configuration Files aastra.cfg, <model>.cfg, <mac>.cfg Specifies the port used for SIP services. Numeric Default Value 5060 Range Greater than 1024 and less than Example sip services port: 7300 RN REV

22 Issues Resolved in Release SP1 This section describes the issues resolved on the IP Phones in Release SP1. The following table provides the issue number and a brief description of each fix. Note: Unless specifically indicated, these resolved issues apply to all phone models. Issues Resolved Issue Number Description of Fix Configuration DEF21828/CLN26163 DEF24145/CLN25510/ CLN30826 DEF29383 DEF29920 DEF29956 DEF30173/CLN30178/ CLN30179 DEF30349/CLN30556 An issue was observed whereby changes made to the Caller ID parameter in the Web UI were not being dynamically updated on the phone. This issue has been fixed. Previously, when a programmed key was removed (via Web UI or the IP Phone UI s Speed Dial Edit Menu), the key would not be completely removed from the local.cfg file. Consequently, if the same softkey was defined in the aastra.cfg or MAC.cfg files, the key would not show up (as the remaining parameters in the local.cfg file override the server.cfg file). This issue has been corrected and now when a programmed key is removed, the whole content of the respective key is removed from the local.cfg file. When the phone s network settings were initially configured through DHCP, if DHCP was disabled through the phone s UI and the phone was restarted, the phone did not retain the previous DHCP-configured IP address and gateway as was expected. This issue has been resolved. 9143i: The Preferred Line and Preferred Line Timeout settings were not available to be configured through the phone s Web UI in a previous firmware release. This issue has been corrected and the settings are now available in the Basic Settings > Preferences > General section of the phone s Web UI. An issue was observed whereby setting the map conf key to parameter in the configuration files did not map the Conference key to the defined value as expected. This issue has been resolved. Telnet port access was open on the phone. Access has now been disabled as the phone does not support configuration or administration via Telnet. 6755i & 6757i: When the parameter sip keepalive timer was defined in the configuration files, the phone did not send out the UDP keep alive messages at the configured intervals as expected. This issue has been fixed. 20 RN REV

23 Issues Resolved in Release SP1 Issue Number Description of Fix SIP DEF22863 DEF23080 DEF26996/CLN26997 DEF27586 DEF28614/CLN29658 DEF28863/DEF30352/ CLN29112/CLN30625 DEF29019 DEF29024 DEF29398 DEF29877/CLN30118 An intermittent stability issue was observed when 802.1x and TLS were enabled on the phone and the phone was connected to a 802.1x enabled port on a switch with re-authentication set at 60 seconds. This issue has been fixed. When in the process of transferring an active call, using the handset to speak with the transfer target and then pressing the Speakerphone button and placing the handset on hook would incorrectly initiate a transfer. This issue has been resolved. 9143i, 9480i, & 6757i: When two 802.1x enabled phones were connected to the same hub on a non-802.1x network and one of the phones was restarted, intermittent stability issues could be observed on the remaining phone. This issue has been fixed. 6739i: A stability issue was observed whereby if DHCP was disabled on the phone and the DNS server was unreachable the phone would repeatedly reboot itself until the DNS server was reachable once again. This issue has been resolved. 6739i: When configured to use TLS and SRTP and used in conjunction with the Aastra MX-ONE call manager, if a user answered an incoming call and then terminated the call, no BYE message was sent from the user s phone to the server causing the remote party s phone to remain in a connected state. This issue has been corrected. 6757i: With the live dialpad and confxfer live dial parameters enabled, attempting to transfer an active call by pressing the Xfer softkey, a BLF-configured softkey, and then the Goodbye key very quickly would at times not send out the expected Refer message (i.e. the transfer would at times fail) or alternatively cause intermittent stability issues to occur. These issues have been fixed. When the backup outbound proxy was defined and the primary outbound proxy was not accessible, the phone did not consider the backup outbound proxy as trusted and therefore would not allow calls to be received from the backup outbound proxy. This issue has been corrected. When in a held state, it was observed that the phone would incorrectly send a re-invite containing an SDP sendrecv when refreshing the session timer. This issue has been resolved and the phone now sends a re-invite containing the correct SDP recvonly attribute in such situations. If the phone received an INVITE message with unknown parameters contained in the P-Asserted-Identity (PAI) header, the phone would respond with a 400 Too many or erroneous P-Asserted-Identity header(s) message. This issue has been fixed and the phone now ignores unknown parameters contained in the PAI header and processes the INVITE correctly. An issue was observed whereby if the transfer of an active call failed, the call on hold (due to the attempted transfer) could not be retrieved. This issue has been corrected. RN REV

24 Issues Resolved in Release SP1 Issue Number DEF30174/CLN30192 DEF30414/CLN30680/ CLN30681 Description of Fix A stability issue was observed when the phone was acting as a REFER notifier and, in a certain scenario, received an un-hold request. This issue has been fixed. 6739i: When TLS/SRTP and Dual Homing mode was enabled on the phone and the phone was being used in a multi-site environment with two Aastra 5000 call managers, if the SIP service of the main call manager was restarted, the phone did not send a REGISTER to the main site as expected. This caused the phone to lose service. This issue has been resolved. User Interface DEF23702 DEF26012 DEF26013 DEF26151 DEF26269/CLN26270 DEF26511 DEF26574 DEF26617 DEF26780 An issue was observed whereby the LED corresponding to the Speakerphone/Headset audio mode would incorrectly remain lit if a call on Line 1 (answered by speakerphone) was successfully transferred at the same time a call was incoming on Line 2. This issue has been corrected. 9143i, 6730i, 6731i, & 6753i: Users were unable to save a programmable key while in an active call or when navigating the Callers or Redial Lists. This issue has been fixed. 6730i, 6731i, & 6753i: An issue was observed whereby users were not able to configure a key for speeddial usage using a Directory entry (i.e. by pressing the Save key while viewing a Directory entry). This issue has been resolved. 6755i, 6757i, & 6757i CT: When saving a new Directory entry or editing and saving a previous Directory entry, no indication was displayed on the phone notifying the user that the entry was saved. This issue has been fixed and a Entry Saved message is now displayed on screen in such situations. 6755i: The phone did not allow users to create more than 10 Speeddial softkeys using the phone s UI. This issue has been resolved and users can now press and hold the More softkey to create additional Speeddial softkeys if required. An issue was observed whereby the LEDs corresponding to keys configured with Line functionality did not display the line states correctly when performing a transfer or creating a conference call. This issue has been fixed. With the Play Call Waiting Tone setting enabled and the Call Waiting Tone Period configured, if two simultaneous calls were incoming and one call was answered, no call waiting tone could be heard. This issue has been corrected. 6757i & 6757i CT: With the handset off hook and while in the process of creating a conference call, placing the handset on hook after the second leg of the conference call was connected did not terminate the second leg as expected. This issue has been resolved. 9480i & 9480i CT: An issue was observed whereby the LED for a BLF softkey would not flash slowly when the corresponding line was placed on hold. This issue has been corrected. 22 RN REV

25 Issues Resolved in Release SP1 Issue Number DEF26902/DEF27689 DEF27098 DEF27335/CLN28473 DEF27419 DEF27497 DEF27500 DEF27571 DEF27579 DEF27687 DEF27735 DEF27736 Description of Fix 6757i: When an incoming call was answered, pressing the Xfer or Conf softkey and then a keypad key programmed with speeddial functionality, or alternatively, pressing a Speeddial/Xfer or Speeddial/Conf softkey did not dial the speeddial number as expected. This issue has been fixed. 6739i: Previously, if a softkey label containing the Euro symbol (i.e. ) was defined, the symbol would not be displayed on the phone s screen. This issue has been fixed. An issue was observed whereby the MWI LED did not stay permanently lit as expected when the phone was in a No Service state. This issue has been resolved. When two phones were configured for SCA and AFE functionality, and DND key mode was set to Phone, pressing the DND key once did not disable the DND feature on the phones. This issue has been corrected. When using the phone in conjunction with the Aastra MX-ONE call manager, an issue was observed whereby the playback of a continuous ring splash alert pattern (indicating an incoming call on a BLF line) would not continue as expected if another ring splash alert pattern (for an incoming call on another BLF line) was played and then ended. This issue has been resolved. When the phone was locked and a call was incoming, pressing the Answer softkey or button did not answer the incoming call in Handsfree Speakerphone mode. This issue has been fixed. 6755i: When configuring Call Forward settings through the phone s UI, in certain scenarios the Call Forward softkey s LED would unexpectedly turn on immediately after the setting was applied. This issue has been corrected and the LED now correctly turns on only after completely exiting the Call Forward menu. When the Call Waiting feature was enabled on the phone, but the Play Call Waiting Tone setting was disabled, if the phone was in an active call and a secondary call was incoming, the MWI LED would not flash (i.e. notifying users of the incoming call) as expected. This issue has been fixed. 6739i: Audio issues were observed when adding an additional party to an active centralized conference call. Moreover, the Conf keys on the phone became unresponsive; as a result users were unable to add any more additional parties to the conference call. These issues have been corrected. With SCA lines configured on the phone and while in a three-way conference call, if a user dropped the first leg of the conference call and added a new party, the phone would display the same Caller ID (corresponding to the new party) for both legs of the newly created conference. This issue has been resolved. 6731i & 6739i: An issue was observed whereby the Conference hardkey was still active/responsive when in the process of transferring an active call. This issue has been corrected and the Conference hardkey is now disabled when a transfer is being performed. RN REV

26 Issues Resolved in Release SP1 Issue Number DEF27770 DEF28293 DEF28515/DEF29342 DEF28551 DEF28660 DEF29295 DEF29678/CLN30801 DEF29715/DEF29716/ DEF29717 DEF29790 DEF29824 DEF30021/CLN30404 DEF30047/CLN30074 Description of Fix 6731i: When attempting to transfer an active call, pressing a volume button to increase or decrease the speaker volume before entering the transfer target s number, caused the transfer to fail. This issue has been fixed. If a Speeddial key was defined as a string using lowercase characters, the phone would incorrectly change the dialstring to all uppercase characters when the Speeddial key was used to dial out. This issue has been resolved. Pressing the left navigation key was not functioning as expected when in various Options List submenus. This issue has been fixed and pressing the left navigation key in such situations now correctly causes the phone to navigate back to the previous menu. A duplicate entry issue caused an error whereby users were not able to view all the Directory entries defined in a.csv file through the phone s Directory List. This issue has been corrected. The Backspace softkey would at times not function as expected when performing a search in the Directory. This issue has been fixed. 6739i: When a user tried to add a new entry to the Directory but the Directory already contained the maximum number of entries (i.e. 200), no indication was made to the user that the Directory was full. This issue has been corrected and a Directory is full message is now displayed on screen in such situations. An issue was observed whereby the phone would lose Directory entries that were initially downloaded from a configuration server if the configuration server was unreachable when the phone was restarted. This issue has been resolved and the phone now retains all previously downloaded Directory entries in such situations. 6731i: When in an active call on Line 1, users could not dial out on the secondary line by using the L2 button in conjunction with the Redial, Callers, or Directory lists. This issue has been corrected. 6755i & M675i: With multiple Line softkeys configured on the expansion module and the phone configured so that line icons were not displayed, the labels of the Line softkeys on the expansion module would unexpectedly refresh multiple times when seizing a line on the phone or releasing a call. This issue has been resolved. 6731i: When the phone had more than one call on hold, the phone did not show the Calls held message on screen. This issue has been fixed. An issue was observed whereby changing a softkey with a user-configured label to a softkey type with an unconfigurable label (e.g. Do Not Disturb, Last Call Return, Empty, etc...) would not update the label as expected. This issue has been resolved. With a call on hold on Line 1, a UI issue was observed whereby the phone s screen would not refresh if a transfer was attempted on Line 2 but the transfer target dropped the call before the transfer was completed. This issue has been fixed. 24 RN REV

27 Issues Resolved in Release SP1 Issue Number DEF30080 DEF30096/CLN30117 DEF30112/DEF30125/ CLN30123 DEF30181 DEF30185/CLN30201 DEF30212 DEF30213 DEF30302 DEF30351/CLN30362 DEF30370 DEF30766/CLN30806 Description of Fix 6739i, 6755i, 6757i, 6757i CT, & M675i: Pressing any one of the three page selection keys on the expansion module would not automatically turn on the phone s and expansion module s backlight as expected. This issue has been resolved. When in the process of transferring a call, if a secondary call was incoming and the initial call was transferred before answering the secondary call, the phone would display the Call Transferred message even after answering the secondary call. This issue has been corrected and the Caller ID is now displayed as expected in such scenarios. 9143i, 6730i, 6731i, & 6753i: During an active call, if a user pressed the Save key and then the Goodbye key, the phone would incorrectly terminate the active call and continue to display the Save to message on screen. Additionally, pressing the Goodbye key again would cause the phone to display unwanted characters on screen. These issues have been fixed. An issue was observed whereby the MWI LED did not flash as expected when the phone had an incoming call. This issue has been resolved. 9143i, 6730i, 6731i, & 6753i: An issue was observed whereby users could not add the plus (i.e. + ) sign to a number corresponding to a Directory entry using the phone s keypad. This issue has been corrected. 6739i: When the phone was locked, entering an incorrect password while attempting to unlock the phone would cause the keypad to become unresponsive. This issue has been resolved. When configuring the phone using the TR-069 protocol, the MAC address was displayed incorrectly (i.e. each pair of hexadecimal digits were not separated by the required colon character). This issue has been fixed. 6739i: With the Display DTMF Digits option enabled on the phone, if the phone was in a three-way conference call, pressing a digit on the phone s keypad would cause the phone s screen to briefly flicker. This issue has been resolved. 6739i: When transferring an active call or creating a conference call, if a user entered the number of the second leg but used the Backspace key to correct the number at any time, the transfer/conference would unexpectedly fail. This issue has been corrected. An issue was observed whereby certain versions of Microsoft s Internet Explorer were not able to access the phone s Web UI via the HTTPS protocol. This issue has been resolved. If a Directory entry that was loaded from a.csv file was renamed through the phone s UI, restarting the phone and then accessing the renamed Directory entry caused a stability issue to occur. This issue has been fixed. RN REV

28 Issues Resolved in Release SP1 Issue Number DEF22076/DEF26645/ DEF26651/DEF26657/ DEF26662/DEF29706/ DEF30153/DEF30336/ DEF30372/DEF30624/ CLN26646/CLN26652/ CLN26658/CLN26663/ CLN30340 DEF22938/DEF29761/ DEF30830/CLN30246/ CLN30835 DEF26571/DEF26790/ DEF28452/DEF30541 DEF27530/DEF29323/ DEF30090 Description of Fix Various user interface issues related to the paging feature have been corrected in Release SP1. Various language/translation issues have been corrected in Release SP1. Various issues with regards to audio mode switching/handling in certain scenarios have been corrected in Release SP1. Various issues with regards to unexpected line focus in certain scenarios have been corrected in Release SP1. XML DEF22617/DEF25536/ DEF28597/DEF28608/ DEF29694/DEF29927/ DEF29931/DEF30035/ CLN30082/CLN30403 Various XML-related issues have been corrected in Release SP1. 26 RN REV

29 Contacting Aastra Telecom Support If you have read this release note, and consulted the Troubleshooting section of your phone model s manual and still have problems, please contact Aastra Telecom Support via one of these methods: North America Toll Free Direct Online at click on Contact Technical Support Outside North America Please contact your regional Aastra Technical Support. RN REV

30 Disclaimer Aastra Telecom, Inc. will not accept liability for any damages and/or long distance charges, which result from unauthorized and/or unlawful use. While every effort has been made to ensure accuracy, Aastra Telecom, Inc. will not be liable for technical or editorial errors or omissions contained within this documentation. The information contained in this documentation is subject to change without notice. Copyright 2013 Aastra Technologies Limited,