Avaya Aura Communication Manager and Avaya Aura Session Manager Compatibility

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1 Avaya one-x Deskphone SIP Release Readme This file is the Readme for the Avaya one-x Deskphone SIP Release software for the 9601 SIP Deskphone. This file describes the contents of the October 2012 maintenance release software distribution package. SIP 6.1 software is supported on the 9601 SIP deskphone only and when used with Avaya Aura Communication Manager with Avaya Aura Session Manager. The SIP 6.1 software will not load or operate on any other phone models. This release supersedes all previous Avaya one-x Deskphone SIP 6.1 releases. Avaya recommends that all customers using Avaya one-x Deskphone SIP 6.1 software upgrade to this version at their earliest convenience. Avaya Aura Communication Manager and Avaya Aura Session Manager Compatibility The 9601 SIP Deskphone using one-x Deskphone SIP Release 6.1 software is supported on Avaya Aura Communication Manager SP 6 patch or later with Avaya Aura Session Manager 6.1 SP5 or later. It is also supported on Avaya Aura Communication Manager 6.2 with Avaya Aura Session Manager 6.2. The 9601 SIP Deskphone using one-x Deskphone SIP Release 6.1 software is not compatible with SIP Enablement Services or Avaya Aura Application Enablement Services. Page 1 of 12

2 New Features in SIP There are no new features in this maintenance release. New Features in SIP Support for Hearing Aid Compatibility (HAC) Option Hearing Aid Compatibility (HAC) allows for audio performance when IP Deskphones are used in conjunction with hearing aids. Providing an acceptable level of audio quality for users without a hearing aid may conflict with meeting the stricter HAC criteria. With this service pack, a configurable option has been introduced which allows the choice of tuning the handset to meet either TIA 810/920 audio standards or FCC Part 68 HAC audio standards. This option can be enabled either via 1. A-Menu: Menu -> Options & Settings -> Advanced Options -> Handset Equalization followed by choice of Default, Audio Opt. (compliant with TIA 810/920), or HAC Opt. (compliant with FCC Part 68 HAC) 2. The 46xxsettings.txt file with the following parameter. ADMIN_HSEQUAL Specifies handset audio equalization standards compliance. Valid values: 1 - (default) Use handset equalization which is compliant with TIA 810/920 2 Use handset equalization which is compliant with FCC Part 68 HAC requirements. Page 2 of 12

3 SIP Package Content The SIP package (9601-IPT-SIP-R6_1_ zip) contains all the files necessary to upgrade Avaya new or previously installed Avaya 9601 SIP Deskphones to the SIP software. The following files are included in each package: S9601_SALBR6_1_5r12_V4r52.tar- The SIP application tar file S9601_UKR_V11r2087_V11r2087.tar- The SIP platform system tar file 96x1Supgrade.txt This file is downloaded by the 9601 SIP Deskphones and instructs the deskphone on how to upgrade. NOTE Refer to the Advisements section later in this document for specific instructions on upgrading from previous versions of SIP 6.1 software or for upgrading in a mixed environment of 9608/9611G/9621G/9641G IP Deskphones and 9601 SIP Deskphones. This package does not contain platform/application files for the 9608/9611G/9621G/9641G IP Deskphones. The included 96x1Supgrade.txt file has commented references to the SIP 6.2 platform/application files. Sixteen predefined language files for phone display: o Mlf_Arabic_6_1.xml o Mlf_BrazilianPortuguese_6_1.xml o Mlf_CanadianFrench_6_1.xml o Mlf_CastilianSpanish_6_1.xml o Mlf_Chinese_6_1.xml o Mlf_Dutch_6_1.xml o Mlf_Englarge_6_1.xml o Mlf_English_6_1.xml o Mlf_German_6_1.xml o Mlf_Hebrew_6_1.xml o Mlf_Italian_6_1.xml o Mlf_Japanese_6_1.xml o Mlf_Korean_6_1.xml o Mlf_LatinAmericanSpanish_6_1.xml o Mlf_ParisianFrench_6_1.xml o Mlf_Russian_6_1.xml Eight extended Korean ring tone files: o KoreanRT1.xml o KoreanRT2.xml o KoreanRT3.xml o KoreanRT4.xml o KoreanRT5.xml o KoreanRT6.xml o KoreanRT7.xml o KoreanRT8.xml Four certificate files: o av_csca_pem_2032.txt Avaya Call Server CA certificate with an expiration date of o av_prca_pem_2033.txt Avaya Product Root CA certificate with an expiration date of o av_sipca_pem_2027.txt Avaya SIP Root CA certificate with an expiration date of release.xml Page 3 of 12

4 The signatures in the signatures subdirectory of the.zip distribution packages are only intended to be used by the file server, and the file server that is on the CM6.x Utility Server is the only file server that currently supports this. System specific parameters should be entered into the 46xxsettings.txt file which is available for separate download at SIP Resolved Issues The SIP release provides continued improvement and optimization to key audio functions as well as stability, robustness and reliability. The following table includes issues which are resolved with this release of software: External ID Internal Summary ID None SIP Audio - Voice clipping for 1-2 seconds when user join Aura Conferencing SIP Audio Additional 1 second cut-thru delay when RTCP is enabled in multiple call scenarios SIP CID - Phones display incorrect number while making an outgoing call SIP (598) Audio getting stuttering message when calling announcement None SIP Failover issue phone should fail to secondary SM upon receiving 503 service unavailable without retry after header None SIP Crash phone lockup in an active call when deny all services on the SIP proxy None SIP Failover issue Call is not dropped from other end in failover failback scenario None SIP Crash - Async connector on startup under certain scenarios SIP Upgrade - robustness issue None SIP Crash - Phone reboots after network outage under certain scenarios None SIP Upgrade - Rename Language files for SIP 6.1 to be different than SIP 6.0 and SIP 6.2 None SIP Upgrade - Supgrade file comment out 96x1 references from file None SIP Crash upgrade in LCD driver None SIP Config - Phone incorrectly responding to Reload Config 5 None SIP Network - Phone 9601 does not send DHCP release packet to release IP addr of the old VLan Page 4 of 12

5 External ID Internal ID Summary None SIP Upgrade occasionally after an upgrade, the UI displays the screens incorrectly None SIP Config - Phone incorrectly responding to Reload Config 0 None SIP Config - Phone cannot login after login with wrong username or password None SIP (712) Crash phone lockup and restart after failover from primary SM to secondary SM SIP /711 Connectivity 9061 has! on the display screen and shows as active on both primary and secondary controller 1-1RY7PJP SIP Config Send corrupted NOTIFY when receive Proxy Authorization SIP /732 Audio 9601 is not correctly setting audio packet size SIP Connectivity PPM download fails if access to the SNTP server is blocked by a firewall None SIP Audio Adjustment of Automatic Gain Control (AGC). None SIP Audio Adjustment of headset volume and sidetone levels. None SIP Audio 9601 is not correctly displaying audio packetization delay SIP Resolved Issues (also in 6.1.5) The SIP release provides continued improvement and optimization to key audio functions as well as stability, robustness and reliability. The following table includes issues which are resolved with this release of software: External ID Internal Summary ID None SIP Audio Timestamp issue causing audio quality issue (Patch d) , , Various Reliability issues - Crash, lockup, hang, reboot conditions, Memory leaks (Patch d) None SIP Reliability Phone performance degrades following 4+ days of normal usage (Patch d) None SIP Audio buffer issue resolved (Patch d) None SIP Ringtone is continued when receiving incoming call and selecting another CA to make an outgoing call None SIP Incorrect public dial tone is generated for COUNTRY Italy (Patch d) Page 5 of 12

6 External ID Internal Summary ID None SIP Incoming call from PSTN has no audio until a Hold/Resume is pressed None SIP Hearing Aid Compatibility (HAC) compliant audio tuning. None SIP sends wrong RTP packet for 300ms after holdunhold to the call None SIP HardwareVersion.txt in release distribution incorrectly indicates support for all 96x1 phone types None SIP BFCP-enabled call to video-enabled 96x1 results in call drop None SIP Using SET COUNTRY UK plays wrong ring back tone affects early media (Patch d) None SIP Audio roll over issue (Patch d) None SIP Software download increase transfer timeout to 15 minutes to accommodate many phone downloads SIP Phone ignores the SIP INVITE request when SIP Display info is (Patch ) SIP Phone ignores the SIP INVITE request when request from Nortel or Polycom phone. (Patch ) None SIP Audio Hot fix from Broadcom , SIP Update audio tuning values (Patch c) None SIP MWI Message waiting indicator fix (Patch ) SIP MWI and a DTMF fix (Patch ) None SIP Audio AGC and audio tuning (Patch c) SIP SIP APP memory leak when receive certain LLDP packet None SIP Memory leak related to ticket 581 (incremental fix) SIP does not support Half screen SIP Update upgrade script file None SIP SIP Signaling thread being killed resulting in ignoring of INVITE/BYE None SIP PUSH: Cannot push Top line and Audio None SIP phone goes in to loop of installing backup application and reboot None SIP Phone query wrong DNS name after phones gets DNS domain name (Option 15) from DHCP None SIP SIP parsing of HTTPS messages issue None SIP New LLDP integration Incorrect Typical Power and Max Power displayed None SIP Application crashed, phone is not rebooting wrong config file used Page 6 of 12

7 External ID Internal Summary ID , SIP , SIP Top line is frozen If screen saver is activated and phone goes off hook None SIP Phone goes stuck and reboot when login with sign is entered None SIP , Call is established with one voice speech path SIP None SIP Contacts not visible if not connected to PPM. Phone clears its local cache in case of 500 server errors. None SIP Can t move up/down in the contact list when the SM is unavailable Unresolved issues with SIP The following table includes unresolved issues with this release of software: External ID Defect ID Summary None SIP Some items in Log categories are placed in wrong alphabetic order None SIP Phone can t get device cert from Microsoft SCEP server None SIP Phone displays language as English but right-aligned as Hebrew format None SIP PPM overrides some manually set fields on SIP Global Settings. them None SIP Phone displays contact name instead of phone number under certain scenarios None SIP Phone incorrectly displays the option to set the Ethernet speed to 1000Mbps. Note: 1Gig speed is not available on this phone. None SIP Phone fails to display contacts added via SMGR (H.323, Digital, Analog) None SIP Phone does not register to AudioCode after failover/failback with third party gateway alternate registration policy None SIP Interchanging primary and secondary SM s for users causes registration issues and may require time to adapt (menu may be sluggish) None SIP Phone can get stuck and needs to be reboot after calls lasting 2+ hours None SIP Phone fails to add contact in Chinese language None SIP Phone display the location of ringing icon wrongly in Hebrew language None SIP Phone can incorrectly get PPM data from Server update transport type to TLS. None SIP Limited phone service when double press Enter soft key quickly Page 7 of 12

8 External ID Defect ID Summary None SIP Can t move up/down in the contact list when SM is unavailable None SIP Remove clearing of contacts cache in the phone in case of 500 server errors users will then be able to see their contacts None SIP Phone does not store both server s addresses in its cache memory after removing the SIP_CONGROLLER_LIST from settings file and reboot (RESTART) the phone through craft option None SIP Phone does not work properly with CALL_LOG_BRIDGED parameter None SIP Phone failed to update according to SIP proxy list from settings in SMGR None SIP Unhold softkey not working during conference call in OneX Communicator shared control mode None SIP Phone 9601 does not show any Arabic/korean characters if it is set to system language None SIP Primary Phone A displays missing continue soft key when phone B logs in the same Ext in vu_mode as primary. None SIP Phone clipping message when calling announcement. None SIP Phone displays missing VLAN ID when L2Q mode off. None SIP Phone sends DHCP Release after receiving new IP address from new VLAN None SIP Call is not established with G.726 codec in SRTP mode None SIP Phone does not send gethomeserver request when doing Reload Complete from SMGR. None SIP Phone gets stuck after deleting contact on Deskphone and OneX Communicator shared control mode inactive. None SIP Cannot save name and number on AUTODIAL feature from Menu option None SIP Phone ignores "NO_DIGITS_TIMEOUT" timer expiration in 46xxsetting file. Page 8 of 12

9 Advisements with SIP software 9601 Features Not Supported The 9601 SIP Deskphone utilizes Avaya one-x Deskphone SIP 6.1 software. The following features are specifically not supported with this version of software: Advanced Conferencing with Avaya Aura Conference 7.0 Avaya Audio Branding Call Center Elite features Call Pickup Alerting (CPA) DHCP Option 43 Dialing E.164 Numbers Enhanced Call Forwarding Group Paging Hotline Instant Messaging Key Repeat Delay Large Text Microsoft Exchange 2007 contact integration Microsoft OCS Early Media Per Line Ring Control Phone Lock Idle Timer Presence with Avaya Aura Presence Server Privacy Feature Enhanced Push API support Shared control with Avaya one-x Communicator. SLA Mon Software Agent SSH support Team Button Ten Emergency numbers Visual Indication of Encrypted Call WML Browser WML Push Software upgrades: 1. Make copies of the following files currently used on your system before upgrading: i. 46xxsettings.txt file ii. 96x1Supgrade.txt file 2. Unzip the SIP package file and place the contents in the root directory of your HTTP server. 3. If you have a mixed environment of 9608/9611G/9621G/9641G IP Deskphones and 9601 SIP Deskphones, you will need to manually merge the old and new 96x1Supgrade.txt files. 4. Make the adjustments required by your environment to your 46xxsettings.txt file. 5. Reset your Avaya 9600-Series IP Deskphone. Page 9 of 12

10 9608/9611G/9621G/9641G and 9601 Mixed Environment Language Files The language file names have been renamed in SIP so that they can co-exist with SIP 6.0.x and SIP 6.2.x installations. The 46xxsettings.txt file must include the new SIP 6.1 language file names (appended with _6_1 ) for the SIP 9601 to be able to download them. There are two ways to configure your http server in a mixed environment: The first one is to have two different 46xxsettings.txt files - one for 9601 and another one for the 9608/9611G/9621G/9641G. In that case you need to modify the 9601 version of settings file only. For example add the following string to enable two additional languages on your 9601 SIP Deskphones: SET LANGUAGES Mlf_German_6_1.xml,Mlf_ParisianFrench_6_1.xml Another approach is to have one 46xxsettings.txt file for all models. The following shows an example change in a 46xxsettings.txt file to accommodate both SIP 6.1 and SIP 6.2 language files. ## Languages IF $MODEL4 SEQ 9601 GOTO 9601LANG SET SYSTEM_LANGUAGE Mlf_English.xml SET LANGUAGES Mlf_German.xml,Mlf_ParisianFrench.xml GOTO NEXTSTEP # 9601LANG SET SYSTEM_LANGUAGE Mlf_English_6_1.xml SET LANGUAGES Mlf_German_6_1.xml,Mlf_ParisianFrench_6_1.xml GOTO NEXTSTEP # NEXTSTEP 9601 SIP Deskphone - Specific Upgrade Notes: Please follow these instructions carefully 1. All 9601 IP Deskphones in the network should be running software or newer before attempting to upgrade to software (PSN ) software simplifies the complex upgrade paths required by prior releases. To create customized upgrade scripts for your environment, it is recommended that you use the upgrade script provided with software as a guideline. 3. In the unlikely event that you need to restore your prior release to your phones, we highly recommend: Page 10 of 12

11 a. Edit your old 96x1Supgrade.txt script to directly specify your old target software load. You may safely remove all the release-based instructions ( IF $APPNAME_IN_USE SEQ statements) in the 9601 section. In this manner, a downgrade from is directly compatible with all prior releases and patches from and For example: IF $APPNAME_IN_USE SEQ S9601aSALB_R6_1_0r26418_V11r1271.tar GOTO 9601sp3 is no longer required. These statements should be deleted. b. If your old upgrade script contains other conditional statements, such as group-related instructions ( IF $GROUP ), you will continue to require these statements to maintain the configuration of your network. IMPORTANT NOTE: Upgrade files released prior to contain an error that will cause the phone to attempt to run an incompatible application/kernel binary pair. In these files, in section # 9601sp0 SET APPNAME S9601aSALB_R6_1_0r26418_V11r1271.tar SET APPSIGNAME S9601aSALB_R6_1_0r26418_V11r1271.tar.sig The kernel binary is not specified. If applied, this instruction will cause the phone to attempt to run the SP0 application binary on the current kernel binary. This mismatched kernel/application pair may result in a variety of problems, including the possible complete failure of the phone. These may not be recoverable. If it is necessary to downgrade, please ensure that this problematic clause is removed from any upgrade script. 4. Manually clearing or provisioning the HTTPSRVR address on the phone via the CRAFT menu can potentially prevent future DHCP HTTPSRVR address updates. 5. Releases prior to are no longer supported (PSN ). Please make sure that your upgrade or downgrade script files do not contain references to GA or SP1. Do not downgrade your 9601 SIP Deskphone further than Enabling Debug or using Syslog: As a general guide, it should be noted that response times could be impacted when debug or syslog is enabled. Please do not enable ALL logging at the same time or performance will be affected License Agreements Page 11 of 12

12 License agreements are available at Please select 96x1 SIP. Notice 2012 Avaya Inc. All Rights Reserved. While reasonable efforts were made to ensure that the information in this document was complete and accurate at the time of printing, Avaya Inc. can assume no liability for any errors. Changes and corrections to the information in this document may be incorporated in future releases. Documentation disclaimer. Avaya Inc. is not responsible for any modifications, additions, or deletions to the original published version of this documentation unless such modifications, additions, or deletions were performed by Avaya. Link disclaimer Avaya Inc. is not responsible for the contents or reliability of any linked Web sites referenced elsewhere within this Documentation, and Avaya does not necessarily endorse the products, services, or information described or offered within them. We cannot guarantee that these links will work all of the time and we have no control over the availability of the linked pages. Page 12 of 12