A number of the new features in Release 2.6 and later are available only on recent CM and SES versions as shown in the table below.

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1 Avaya one-x Deskphone SIP Readme Avaya one-x Deskphone SIP Release Readme This file is the Readme for the Avaya One-X Deskphone SIP Release software which is applicable for the following 9600 Series IP Deskphones: 9620, 9620C, 9620L 9630, 9630G 9640, 9640G 9650 and 9650C. This file describes the contents of the December 2013 software distribution package. This release supersedes all previous Avaya one-x Deskphone SIP 2.6.x releases. Avaya recommends that all customers using Avaya one-x Deskphone SIP 2.6.x software upgrade to this version at their earliest convenience. Avaya Aura Compatibility This release is recommended for installations with Avaya Aura Session Manager 6.0 and later, with an Avaya Aura Survivable Remote, Avaya Aura Communication Manager 6.0 and later and also with installations with SIP Enablement Services 5.x with Communication Manager 5.x. A number of the new features in Release 2.6 and later are available only on recent CM and SES versions as shown in the table below. Minimum CM/SM Version Required Minimum SES Version Required SBM-24 button module Feature button assignments and operation SBM-24 button module Autodial assignments and operation SBM-24 button module Bridged Call Appearance (BCA) and Call Appearance (CA) operation 5.0/5.x /5.x /5.x 5.1 Auto Answer 5.0/5.x 5.0 Deactivation of Call Forward On Busy Do Not Answer 5.2/5.x 5.2 Support for up to ten emergency numbers 6.2/6.2 N/A (only a single emergency number is supported) Support for Multipart MIME 6.2.3/6.2 N/A Improved Recovery during failover 6.2.4/6.3.0 N/A Issued 2 December 2013 Page 1 of 10

2 Avaya one-x Deskphone SIP Readme New Features in SIP Avaya one-x Deskphone SIP Release contains the following new feature. New with this release Improved recovery during failover. Description Following a failover from one Session Manager to the backup Session Manager, this enhancement reduces the time required to re-synchronize the state of the IP Deskphone with Communication Manager. This feature also requires the use of Avaya Aura Session Manager 6.3 or later. Documentation for SIP There are no updates to documentation for this release of software. Administering Avaya one-x Deskphone SIP Release 2.6 Avaya one-x Deskphone SIP 9620, 9620C, 9620L IP Telephone User Guide Avaya one-x Deskphone SIP for 9640/9640G IP Telephone User Guide Avaya one-x Deskphone SIP for 9650/9650C IP Telephone User Guide Avaya one-x Deskphone SIP for 9620/9620C/9620L IP Telephones Quick Reference Guide Avaya one-x Deskphone SIP for 9640/9640G IP Telephones Quick Reference Guide Avaya one-x Deskphone SIP for 9600 Series IP Telephones API Guide This documentation is available on under 9600 Series IP Deskphones -> SIP 2.6.x -> Documents. Issued 2 December 2013 Page 2 of 10

3 SIP Package Content Avaya one-x Deskphone SIP Readme The SIP package (96xx-IPT-SIP-R2_6_ zip) contains all the files necessary to upgrade new or previously installed IP Deskphones to the SIP software. The following files are included in this package: hb96xxua3_00.bin The SIP boot application file SIP96xx_2_6_11_4.bin The SIP phone application file 96xxupgrade.txt This file is downloaded by the Avaya 9600 IP telephones and instructs the telephones on how to upgrade. DO NOT EDIT this file. You MUST USE the 96xxupgrade.txt file included in this package to upgrade to this release of software. o Note that this file is designed for environments in which all 9620, 9620C, 9620L 9630, 9630G, 9640, 9640G 9650 and 9650C endpoints will be upgraded to the SIP software. o Alternative 96xxupgrade file for mixed H.323 and SIP 96xx deployments - Available at support.avaya.com the alternate_96xxupgrade.txt file is designed for environments that will support 96xx upgrades for both the H323 and SIP modes of operation. See the Avaya support website for additional information for how to edit this file and setup the SIG setting (the Signaling Procedure) in the 96xx phones. release.xml Fifteen predefined language files for phone display: o Mlf_Arabic.xml o Mlf_BrazilianPortuguese.xml o Mlf_CanadianFrench.xml o Mlf_CastilianSpanish.xml o Mlf_Chinese.xml o Mlf_Dutch.xml o Mlf_English.xml o Mlf_German.xml o Mlf_Hebrew.xml o Mlf_Italian.xml o Mlf_Japanese.xml o Mlf_Korean.xml o Mlf_LatinAmericanSpanish.xml o Mlf_ParisianFrench.xml o Mlf_Russian.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 Three certificate files: o av_csca_pem_2032.txt - Avaya Call Server CA certificate with an expiration date of 2032 Issued 2 December 2013 Page 3 of 10

4 Avaya one-x Deskphone SIP Readme o o av_prca_pem_2033.txt- Avaya Product Root CA certificate with an expiration date of 2033 av_sipca_pem_2027.txt- Avaya SIP Root CA certificate with an expiration date of 2027 System specific parameters should be entered into the 46xxsettings.txt file, which is available for separate download at One provisioning example may be helpful. Assumptions: SIP IP Deskphone - IP address: SIP IP Deskphone - network mask: SIP IP Deskphone - gateway: HTTP file server: SES IP address (primary controller): o TLS transport protocol, default so not explicitly specified in setting AudioCodes/Cisco IP address(secondary controller ): o TCP transport protocol, explicitly set in setting file SNTP Server IP address: The SIP domain: mynetwork.com Time zone where IP telephone is provisioned: Eastern Daylight Savings Time (following the current rules where Daylight Savings Time begins on the first Sunday of April and ends on the last Sunday of October for more information on provisioning DST see the 9600 Series IP Deskphone Administration Guide) The following are the parameters that must be defined in your 46xxsettings.txt file to make the 9600 Series IP Deskphone operational: SET SNTPSRVR SET GMTOFFST -5:00 SET DSTOFFSET 1 SET DSTSTART SET DSTSTOP 1SunApr2L LSunOct2L SET SIP_CONTROLLER_LIST :5061;transport=tls, :5060;transport=tcp SET SIP_MODE 0 SET SIPDOMAIN mynetwork.com If the IP Deskphone is not acquiring its address from a DHCP server, it must be provisioned on the phone, itself, through the use of the phone administration menu. Once in the Admin Procedures menu, the ADDR submenu will allow the administrator to provision the following fields: Phone: Router: Mask: HTTP File Server: Issued 2 December 2013 Page 4 of 10

5 Avaya one-x Deskphone SIP Readme Advisements with SIP software Recommended SIP Transport Protocols SIP Proxy / Phone Connection Type UDP TCP TLS SES (as primary controller ) Not Recommended Recommended Recommended Avaya Session Manager (as primary controller) Not Recommended Recommended Recommended Avaya Secure Router 2330 and 4134 Not Recommended Recommended Recommended AudioCodes MP-series analog and BRI gateways (as secondary controller) Not Recommended Recommended Not Recommended Cisco 2811 ISR (as secondary controller) Not Recommended Recommended Not supported I55 (as secondary controller) Not Recommended Recommended Not supported Teldat Vyda gateway Not Recommended Recommended Not Recommended Table of Phone Connection Types for Survivability Configurations Recommended means minimal latency in the detection of a Failover condition The Audiocodes SIP gateways tested for interoperability are the MP114 and MP118; the minimum firmware version is 5.60A or later. The Cisco gateways tested for interoperability are the Cisco [ISR] ; the minimum firmware revision is c2800nm-ipbasek9-mz ya2.bin When the phone is failed over to a non-avaya secondary controller (AudioCodes/Cisco): We expect all the phones in the environment are operating in same transport protocol. There may be a few call-based issues if the phones are using different transport protocols. Configuration with 3 BMs and few dozen Busy Indicators \ Bridged Appearances administrated Release supports 3 x Button Modules with up to 72 features configured on them. The only exception is the number of notifications per second that the phone can process without performance degradation and significant delays visible for the user. For the incoming \ outgoing calls the number of notifications per second for this release should not be more than: o o 3 BM with 72 Busy indicators configured: Incoming call 12 BI NOTIFYs / 1 sec Outgoing call 8 BI NOTIFYs / 1 sec 3 BM with 72 Bridged Line Appearances configured: Incoming call 6 BA NOTIFYs / 1 sec Outgoing call 4 BA NOTIFYs / 1 sec Issued 2 December 2013 Page 5 of 10

6 NAT Avaya one-x Deskphone SIP Readme 96xx SIP telephones should not be provisioned behind a NAT with private network addresses when the CM and SES or SM switching fabric are provisioned on a different network. Recommended Failback Policy Admin failback not recommended in R2.6 and Auto failback policy is recommended. SRTP The Avaya Gateway firmware version must support SDES SRTP Crypto suites must match Requires CM or greater for SM and CM 5.1 or greater for SES Always provision None encryption (setting 9) as one of the encryption selections in the settings file and on CM Microsoft Exchange 2010 Calendar Integration Calendar integration is not supported with Microsoft Exchange Server 2010 or later. Customers who are using this functionality on Microsoft Exchange Server 2007 and upgrade to Microsoft Exchange Server 2010 are advised to notify their users and disable the functionality in the 46xxsettings.txt file. Issued 2 December 2013 Page 6 of 10

7 SIP Resolved Issues Avaya one-x Deskphone SIP Readme The SIP release provides continued improvement to many areas. The following table includes issues which are resolved with this release of software. External ID Internal ID Issue Description User Interface SIP96X When the Power Management error screen is displayed, it is not possible to log in because the Log In screen is not displayed. SIP96X Outgoing call failure for calls which require authorization code. SIP96X If a call is made to a deskphone and ended very quickly before answering, the History button will light but the call will not appear in the Call Log. SIP96X A contact (originally added via SMGR) is added as a local contact. If that local contact is edited to add another work number, then the additional number is not saved. SIP96X If a user has a very large number of contacts (200 or more), then there is a multiple-second delay when using the Assign Speed Dial Entries procedure SIP96X Phone has two bridge appearances of another extension. If a call is made on one bridge appearance, and then a second call is made on the other bridge appearance during a transfer operation, then cannot complete the transfer. SIP96X Phone has two bridge appearances of two different extensions. If a call is made on one bridge appearance, and then a second call is made on the other bridge appearance during a transfer operation, then cannot complete the transfer. SIP96X Phone has two call appearances. A call is made on one appearance, another call is made on the other appearance, and the user presses the Conf softkey. The Join softkey is not displayed SIP96X Phone incorrectly counts + as a digit when applying call display formatting rules. Failover SIP96X Wrong Call Appearance state during a conference following a failover to a 3 rd -party gateway. SIP96X User cannot press the Hold soft-key and voice path is lost during a conference following a failover to a 3 rd -party gateway. SIP96X Direct IP-IP Audio Connections? and Initial IP-IP Direct Media? parameters are set to y, and phones are configured with RTCPMON parameter. When two phones are in a failover mode, pressing Hold on one phone drops the call on the second phone SIP96X Incorrect public dial tone is generated when COUNTRY is set to ITALY during non-ast failover. Data Networking SIP96X If a list of HTTP file servers are configured via LLDP with the third (or later) being valid, and the SIP proxy is configured via a settings file, then the phone will reboot on the log-in screen before successfully retrieving the configuration file. QoS Monitoring Issued 2 December 2013 Page 7 of 10

8 Avaya one-x Deskphone SIP Readme External ID Internal ID Issue Description SIP96X RTCP IPv4 Subtype 5 packet contents malformed Subtype 5 o IPv4/IPv6 Traceroute Hop Count o IPv4 Traceroute information SIP96X When using TLS/SRTP, phones are incorrectly reporting encryption as Unknown instead of AES. PPM SIP96X After CLEARing a phone, and logging in, not all information is retrieved from PPM. Interworking SIP96X Hold via 3 rd -party Call Control is rejected if the phone is in a conference call. SIP96X DTMF commands for a conference bridge are intermittently not recognized. SIP96X In-band DTMF commands are not recognized when Initial IP-IP Direct Media is turned on. Administration SIP96X When trying to Restore Default Button Labels Screen, pressing the OK button does not restore to the default labels. SIP96X AUDIOENV settings are sometimes not applied until after a second reboot of the phone. Presence SIP96X Deskphone is incorrectly shown as Available (instead of Away ) when the AES Out of Office timer expires. Security and Authentication SIP96X Deskphone is not able to authenticate successfully against Session Manager that has custom server certificates. SIP96X Deskphone does not transfer software files using HTTPS when set to only use HTTPS. NOTE: Customers will still need to use http to upgrade to SIP Any subsequent upgrades will be able to use https. Issued 2 December 2013 Page 8 of 10

9 Unresolved Issues in SIP Avaya one-x Deskphone SIP Readme The following table includes unresolved issues with this release of software which were known as of the issue date of this document. External ID Internal ID Issue Description SIP96X SIP96X SIP96X SIP96X SIP96X SIP96X Configuration Any configuration parameter that is set through the CRAFT menu to its default value will be ignored and the value set via DHCP or 46xxsettings file will be used. Workaround: To explicitly set default values for parameters, set them in DHCP or the 46xxsettings file. Phone incorrectly reads and uses more than two encryption parameters (SET MEDIAENCRYPTION) from the 46xxsettings.txt file. Workaround: Only set two encryption parameters. Presence Presence status does not change after Presence Services restart. Workaround: Logout and login the user. Note: This issue is resolved with Presence Services Appendix 1 Release History The following table provides a history of the SIP 2.6 software releases. Release Date Link to Readme file 2.6 June 2010 ftp://ftp.avaya.com/incoming/up1cku9/tsoweb/9600/ /96xxSIP_ReadmeR2_6.pdf August October November September October March August November May December Issued 2 December 2013 Page 9 of 10

10 License Agreements Avaya one-x Deskphone SIP Readme License agreements are available at Please select 96x0 SIP. Notice 2013 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. Issued 2 December 2013 Page 10 of 10