SIP Software Release 3.0 for IP Deskphones
Contents
1. P Ve olaola O orio PEC Description Software File NTYSO3xxxxxx 1120E IP Deskphone SIP1120e03 02 16 00 bin NTYSO5xxxxxx 1140E IP Deskphone SIP1140e03 02 16 00 bin NTYSO7xxxxxx 1165E IP Deskphone SIP1165e03 02 16 00 bin NTYS19xxxxxx 1220 IP Deskphone SIP12x0e03 02 16 00 bin NTYS20xxxxxx 1230 IP Deskphone SIP12x0e03 02 16 00 bin Call Server Compatibility and Requirements SIP software release 3 2 is compatible with the Call Servers listed below Avaya CS1000 Release 7 0 Avaya CS2100 Avaya Aura Communication Manager 6 0 Avaya Aura Session Manager 6 0 Genband Nortel CS2000 CVAM 13 C20 CVAM 14 Genband Nortel AAE 7 0 7 1 8 0 SIP Release 3 2 Readme 11 AVAYA References and Related Documents SIP Release 3 2 for IP Deskphones related documents are available on http www nortel com support using the following product path Changes and Enhancements in SIP Release 3 2 have been documented in this product bulletin For SIP on 1120E IP Deskphone http support nortel com go main jsp cscat DOCUMENTATION amp poid 15741 For SIP on 1140E IP Deskphone http support nortel com go main jsp cscat DOCUMENTATION amp poid 15721 For SIP on 1165E IP Deskphone http support nortel com go main jsp cscat DOCUMENTATION amp poid 22201 For SIP on 1220 IP Deskphone http support nortel com go main jsp cscat DOCUMENTATION amp poid 19422 For SIP on 1230 IP Deskphone http supp2. FNE Transfer to Voice Mail Via Feature Name Extension FNE Whisper Page Via Feature Name Extension FNE Note Some supported features shown in the above Table can be invoked by dialing a Feature Name Extension FNE Or a speed dial button on the deskphone can be programmed to an FNE Communication Manager automatically handles many other standard features such as call coverage trunk selection using Automatic Alternate Routing AAR and Automatic Route Selection ARS Class Of Service Class Of Restriction COS COR and voice messaging Quality Improvements The SIP software Release 3 2 for IP Deskphones also continues to improve the overall quality of the IP Deskphone software through the delivery of ongoing resolution of CRs Numerous quality improvements have been delivered and N customer cases have been closed in SIP 3 2 SIP software Release 3 2 for IP Deskphones closes the following customer cases Closed Cases Description Case Id 30 Minute Delay for MADN Subscription after Registration 100519 47736 1100 IP Deskphone Does not Recognize 2nd DNS Server 100520 48372 Display issue MADN key labeling limited to 10 digits customer using 1 1 digit 10051 1 48242 number plan 100505 40578 1120 IP Deskphone DSCP values are not set correctly for media or control payloads 100521 49010 IP Deskphone Compatibility SIP software Release 3 2 for IP Deskphones is compatible with the following IP Deskphones
3. applications detect adds and removals provide Layer 3 information and group the attached devices into IP subnets IEEE has developed 802 1ab Link Layer Discovery Protocol LLDP a standard for discovering the physical topology between neighboring devices 802 1ab LLDP defines a standard method for Ethernet network devices such as switches routers and IP Deskphones to advertise information about themselves to other nodes on the network and store the information they discover in a MIB This feature implements 802 1ab and its IEEE 802 1 802 3 and LLDP MED specific extensions Auto Login and Per Phone Configuration SIP Release 3 2 supports downloading a user s login credentials from a central repository The login credentials are associated with the MAC address of the particular IP Deskphone This feature simplifies installation of the IP Deskphone by allowing it to automatically log into the SIP proxy server without end user intervention Please refer to the Administration Guide for details on this feature and provisioning this capability Before the user first receives the IP Deskphone the system administrator may configure the IP Deskphone with the user s login id and password If the IP Deskphone is configured with automatic login the user cannot use the logout function and may need a password to access some features and functions of the SIP software In addition the administrators can provision specific features for users using the configuration
4. 00 call servers Internet Protocol version 6 IPv6 is a network layer for packet switched internetworks and is the successor of IPv4 IPv6 provides larger address space which allows greater flexibility in assigning addresses The extended address length used within IPv6 eliminates the need to use Network Address Translation to avoid address exhaustion to simplify the aspects of address assignment and renumbering when changing providers The IP Deskphones can be configured to support IPv4 and IPv6 protocols IP Deskphones use IPv4 mechanisms for example DHCP to acquire their IPv4 addresses and IPv6 mechanisms for example Stateless auto configuration to acquire their IPv6 addresses IPv6 uses a hierarchical method to allocate IP addresses which provides simplified routing and renumbering IPv6 provides the following e 128 bits for address space compared to 32 bits for IPv4 e well defined Quality of Service QoS mechanism e simplified configuration stateless auto configuration e SIP IP Deskphones provide complete support for IPv4 and IPv6 Internet protocols as follows e provides transition mechanism to IPv6 e enables SIP IP Deskphones to interoperate with IPv4 hosts and utilize IPv4 routing SIP Release 3 2 Readme AVAYA e ability to send and receive both IPv4 and IPv6 packets e __interoperates directly with IPv4 nodes using IPv4 packets e __interoperates directly with IPv6 nodes using IPv6 packets IPv6 and IPv4 IP Des
5. AVAYA avaya com SIP Software Release 3 2 for IP Deskphones Date Revision Summary of Changes 23 August 2010 Original bulletin This is the original publication Introduction Avaya is pleased to announce the availability of SIP software Release 3 2 for IP Deskphones SIP software release 3 2 expands the number of supported IP Deskphone devices and makes available the following software versions for the following IP Deskphones SIP Software Release 3 22 IP Deskphone Software 1120E IP Deskphone 03 02 16 1140E IP Deskphone 03 02 16 1165E IP Deskphone 03 02 16 1220 IP Deskphone 03 02 16 1230 IP Deskphone 03 02 16 Avaya recommends an upgrade to this release of software for all applicable IP Deskphones and Call Servers at the earliest convenience SIP Software Release 3 2 includes re branding changes both in the product as well as in supporting documentation and is the minimum release for re branding SIP software Release 3 2 for IP Deskphones is available for download from the Software Download link under Support and Training on the product support website located at http support nortel com The software is available by phone model under Phones Clients and Accessories Note These SIP software loads have not been introduced as the default loads for the IP Deskphones shipped from Avaya SIP Release 3 2 Readme AVAYA Enhancements Several enhancements have been inc
6. E_ALTERN_CODEC LIST is set to YES the IP Phone uses the list of codecs as below High ptime 20 ms G711 G729 Med ptime 20 ms G729 G711 Low ptime 40 ms G729 G711 Thus G729 20ms or 40ms can be selected as default value and may be suitable for certain networks requiring bandwidth optimization Note that when using Audio _Codecx these settings are ignored This is primarily for legacy support SIP Release 3 2 Readme 13 AVAYA Security NAT Traversal amp STUN protocol The SIP Release for IP Deskphones supports two methods for NAT traversal of the signaling path e SIP_PING Simple Traversal of User Datagram Protocol through Network Address Translators STUN SIP PING is an Avaya protocol for NAT traversal for SIP signaling only The STUN protocol lets a client discover the presence and type of NATs between the client and the public Internet In addition a client can discover the mapping between the private IP address and port number and the public IP address and port number Typically an Enterprise or a service provider operates a STUN server in the public Internet or internet domain A STUN server can be located using DNS SRV records using the domain as the lookup STUN typically uses the well known port number 3478 The NAT traversal method can be selected manually through the Device Settings menu or configured through the device configuration file SIP Release 3 2 for IP Deskphones also supports NA
7. P Deskphones there are 3 methods to configure QoS for media and signaling packets 1 Through device configuration file 2 LLDP MED and 3 Through service package If none of the above 3 methods are used system default DSCP values will be used To implement this feature a new checkbox to enable and disable Avaya Automatic QOS is added to the device settings UI Also 2 more fields to configure DSCP values for media and signaling are introduced in the device settings Ul The precedence of taking QoS values will be as follows Service package Avaya Automatic QOS LLDP Device settings UI and finally device configuration file If none of them are configured default DSCP values will be used User configured DSCP values and Avaya Automatic QOS feature will work in conjunction with user configured control and media priority bits as is since later is in layer 2 VLAN ToS and the former will be in layer 3 IP The Device configurations file change is given below The following flag is added to toggle the Avaya on Avaya feature e NORTEL _Automatic_QOS YES or NO YES Phone will use private DSCP values unless overridden NO Phone will either use one of the configured DSCP values or the system default values Secure Digits while in call When users try to access voicemail by dialing the voice mail number it is desirable the typed digits be not displayed as a plain text it should rather be displayed as an asterisk or a dot An administrator defin
8. T 5061 Specify TCP keep alive mechanism o KEEP_ALIVE_TYPE lt CRLF or OS gt o CONN_KEEP_ALIVE lt time gt Certificate based Authentication Certificate based authentication allows the administrator to ensure that the IP Deskphone is authorized to access the enterprise LAN environment and to connect securely to SIP proxy and provisioning servers Certificates bind an identity to a pair of electronic keys that are used to encrypt and sign digital information and make it possible to verify someone s claim that they have the right to use a given key Certificates provide a complete security solution assuring the identity of all parties involved in a transaction Certificates are issued by a Certification Authority CA and are signed with the CA s private key A Certificate Authority issues certificates to users and devices such as IP Deskphones A CA is a trusted third party The certificate issued by a CA contains a variety of data This data includes21 the identity of the issuing CA Certificate Usage and expiry date for the certificate Certificate based authentication is provided on the IP Deskphone by installing trusted root certificates device certificates SIP Release 3 2 Readme 7 AVAYA and Certificate Trust Lists CTL Device Certificates are installed by importing a password protected PKCS 12 file A PKCS 12 file contains both private and public key pair of the certificate CTL is a predefined list of trusted server
9. T traversal for the Media path However STUN protocol cannot coexist with Application Layer Gateway ALG Media Portals or RTP Proxy servers If STUN is configured on the SIP Release ensure none of these devices are configured in the SIP proxy server 802 1X and Extensible Authentication Protocol EAP SIP Release 3 2 for IP Deskphones supports 802 1 X EAP device authentications The authentication protocol currently supported is EAP MD5 Customers need to procure appropriate RADIUS authentication servers both HW SW Certificate based authentication is planned for a future release Shared Call Appearance SIP Release 3 2 for IP Deskphones supports Shared Call Appearance Feature The Shared Call Appearance feature allows a given line to be configured with multiple locations essentially allowing multiple endpoints to login to the system while using the same external number Any one of these locations can be used to originate or receive calls Any call appears to the other party to be originating from or terminating to the same number regardless of the location initiating or receiving the call The Shared Call Appearance feature allows a user to pick up a call that was put on hold by another user of the same group and it allows a user to join an active call of another user in the group The Shared Call Appearance feature is supported on Broadsoft BroadWorks server SIP Release 1 1 and above for IP Deskphones supports Multiple Appearance Directory Nu
10. able screensavers in the Display Screensaver gt Images menu The following uploaded screensaver types are available e Simple background The screensaver just displays one image on activating e SLIDE SHOW The screensaver displays all images presented on the set one after another with a delay between displays e SLIDE GROUP lt xs Similar to the SLIDE SHOW mode however users are able to determine which collection of images will be displayed The collection is being predefined by special template of the file name of the image file Note The SIP 3 2 IP Deskphone will not automatically resize image files Therefore the images must be scaled prior to loading them to the phone The size of the files depends upon the IP Deskphone model e 1120E 240 x 88 e 1140E 320 x 160 e 1165E 320 x 240 SIP Release 3 2 Readme 9 AVAYA Support for Avaya Aura Communication Manager Session Manager SIP Release 3 2 expands support for Avaya 1100 and 1200 Series IP Deskphones with Avaya Aura Communication Manager R6 Avaya Aura Session Manager R6 and Avaya Modular Messaging R5 2 Detailed information is available in an Application Note at the following address https enterpriseportal avaya com ptlWeb internal products P0533 ApplicationTechnicalNotes Note that the 11xx 12xx deskphones were originally developed for use with Nortel call servers and as such do not currently support the Avaya Advanced SIP Telephon
11. ach user will be able to maintain his her list of programmed feature keys You can program the feature keys as Speed Dial Presence Send IM Call Forward or Do Not Disturb The user has the ability to automatically populate the unused Expansion Module feature keys using the friends list or the address book as sources There is no specific dependency on call server for the Expansion Module to be functional However friends list and presence notification are call server dependant and may require changes to the user package of your IP Deskphone Distinctive ringing and Call waiting The Distinctive ringing feature permits users to distinguish between different types of call actions by playing a different ringing pattern The request for a specific ringing pattern comes from the server at the time the call is being established The Avaya 1140E IP Deskphone with SIP software does not request the playing of distinctive ringing from other parties The ringing patterns to be used follow the North American standards which also include call waiting tones for the times when the receiving end is already engaged ina call session The predefined ringing pattern identifiers available on the phone accommodate usage by the Communication server IEEE 802 1ab Link Layer Discovery Protocol Discovery protocols provide a mechanism to identify devices attached to a network Popular network management systems use automated discovery to obtain the topology of a network These
12. certificates which the IP Deskphone views as trusted endpoints It is used a mechanism to provide connection to only trusted servers IP Deskphones enable the administrator to manage view and delete trusted certificates device certificates and CTLs through user interface Events are logged to Security Logs to mark events such as Certificate Addition and Deletion The administrator is to view security and error logs from the user interface as well Certificates are used in various security protocols including e TLS e HTTPS e EAP Refer to the SIP Software for Avaya 1100 Series IP Deskphones Administration guide for details on loading certificates onto the Deskphones Note The phone cannot be secured prior to configuring it to be secured cart before the horse SIP 3 2 also adds support for EAP PEAP in addition to EAP MD5 EAP TLS EAP PEAP and EAP TLS These require that certificates be loaded on the IP Desk phone SIP Release 3 2 Readme 8 AVAYA Embedded Screensavers 1165E sets onl SIP 3 2 for the 1165E includes several embedded screensavers These are always present on the phone regardless of whether images were uploaded to the phone or not These screensavers may be activated at any time The following embedded screensavers are provided e Sun ball e Magic Lines e Radio Waves Pre loaded Screensavers Users may upload custom screensavers to the Deskphone Once uploaded the screensavers are presented in a list of avail
13. e 7 This feature allows a single DN to have shared appearances on several different deskphones Ideal for environments where users are moving between many different phones or where a single person such as an administrative assistant may be answering calls on behalf of others This feature may operate in one of two ways Single DN shared appearances Single DN multiple appearances With shared appearances the DN is shared between the deskphones such that there may only be one call active on the DN at a time server enforced The status of DN is shown on all appearances including idle ringing busy hold Call pickup or barge in are possible depending on status of call This feature is automatically enabled during registration when the deskphone is provisioned on CS1000 call server With multiple appearances each appearance of the DN can independently make receive calls The SCA feature requires the use of Authentication IDs Separate User ID and authentication ID If the configuration file contains the line PROMPT_AUTHNAME_ENABLE Yes default is disabled the user will then be prompted to enter the authentication ID Ifthe authentication ID is not specified it will default to the login id The authentication ID can be specified via AUTOLOGIN_AUTHID_KEYxx IPv6 Support SIP Release 3 2 adds support for IPV4 IPv6 dual mode on 1120E 1140E 1165E and 1220 30 SIP deskphones Currently IPv6 is only supported on AS5300 and CS10
14. e Administration Guide listed in the reference documents section RTP port configuration The SIP Release 3 2 for IP Deskphones provides option to set or change RTP ports RTP port configuration is available only through the provisioning server Voice Quality Monitoring VQMon SIP Release 3 2 for IP Deskphones provides support for monitoring voice quality in conjunction with an Avaya Communication Server and a Telchemy 3rd party server When Voice Quality Monitoring VQMon is enabled in the SIP software the SIP software gathers statistics regarding the quality metrics of the current call and sends reports to the Avaya Communication Server Telchemy 3rd party server at regular intervals The voice quality related statistics include jitter packet loss delay burst gap loss listening R factor R LQ R CQ MOS LQ and MOS C Q This report can be used for QoS monitoring A Telchemy server is required to collect and organize the data SIP Release 3 2 for IP Deskphones can interwork with Telchemy SQmediator 1 1 Please note customers are required to procure necessary Telchemy hardware and Telchemy SQmediator Software directly from Telchemy http telchemy com Avaya Automatic QoS When SIP Release 3 2 for IP Deskphones is deployed with Avaya switches a better treatment for signaling and media packets is provided Avaya devices will use private DSCP values to give better treatment to the traffic coming from a peer Avaya device With SIP Release 3 2 for I
15. ed configuration flag GECURE_INCALL_DIGITS is provided for setting the feature on or off The feature will come in effect only when the call is active Most recently pressed key will be displayed but overwritten when next key is pressed The user has the option to hide or unhide the digits typed and next digits which he she is going to press if flag is set to YES otherwise by default digits will be displayed SIP Release 3 2 Readme 16 AVAYA DHCP avaya com When a DHCP server has to issue a new IP address to the IP Deskphone after a lease expires the IP Phone may go through a reboot and recover process and this process may take up to five minutes for the IP Phone to get back to its ready state to make receive calls In addition if a user is on a call the call may be dropped and the IP Phone may go through the reboot and recover process If the DHCP lease expires the IP Phone must initiate a new negotiation of an IP address from the server s pool of addresses DHCP lease may expire so the network does not run out of IP addresses or due to instability in the network DHCP lease expiration is not a common event About Avaya Avaya is a global leader in enterprise communications systems The company provides unified communications contact centers and related services directly and through its channel partners to leading businesses and organizations around the world Enterprises of all sizes depend on Avaya for state of the art communicat
16. files based on the MAC address of the IP Deskphone RFC support enhancements RFC 3262 RFC 3311 RFC 3581 RFC 3262 amp 3211 The combined support of PRACK and UPDATE allows the IP Deskphone with SIP software to provide reliability to provisional responses and the ability to update session parameters during call setup as well as after the initial invite has received a final response The combination of reliable provisional responses PRACK and the ability to change session before call establishment using UPDATE will improve the IP Deskphone interactions with some PSTN networks where the parameters of a session may need updating before the call is established To provision this feature please refer to the Administration Guide listed in the reference documents section SIP Release 3 2 Readme 15 AVAYA RFC 3581 This feature implements an extension to SIP for Symmetric Response Routing for the IP Deskphone with SIP software This extension permits the conduction of SIP dialogs through a Symmetric Network Address Translator NAT using UDP This allows the phone to work from behind and or in front of a symmetrical NAT with servers and or clients that support RFC 3581 In particular it enhances the capability of the IP Deskphone with SIP software to interoperate with other proxies and to work in any network configuration For this feature to work properly the receiving end device must support RFC 3581 To provision this feature please refer to th
17. ftware are prefixed with NEW Null Screensaver Q02160800 Set hangs after selecting Screensaver If there are no images available for a slide show and the user sets SLIDE SHOW as screen saver then the IP deskphone can hang when the user attempts to select one Analog Terminal Adapter Support Q02036636 Noise is heard on Analog phone connected through ATA 4900 When an analog phone is connected to an ATA on an 11xx IP Deskphone users may hear crackling sounds in the earpiece of the analog set while in 2 way conversations This will be fixed in a future release Audio Codec Enhancements SIP Release 3 2 for IP Deskphones expands support of the existing G729a codec with Annex B G729b codec comfort noise generation and adds support for G723 1 high compression codec codec SIP Release 3 2 for IP Deskphones supports selection of audio codec list as below via the device configuration file For details on the configuration files please refer to the Administration guides SIP Release 3 2 for IP Deskphones supports existing Administrative setting of DEF_AUDIO_QULITY high med low The device configuration flag shown below for codec list selection continues to be supported in SIP Release 2 2 for IP Deskphones ENABLE_ALTERN_CODEC_LIST YES NO When ENABLE_ALTERN_CODEC LIST is set to NO the IP Phone uses the list of codecs below High ptime 20 ms G711 G729 Med ptime 30 ms G711 G729 Low ptime 30 ms G729 G711 When ENABL
18. ing Security via Transport Layer Security TLS per RFC 5246 TLS is a protocol for establishing a secure connection between two end points After a connection is established using TCP TLS negotiates the cryptographic parameters used to secure the traffic that is sent over that connection TLS Public Key Cryptography and X 509 certificates provide either mutual or server authentication Mutual authentication occurs when both the client and the server have public key certificates assigned that are used during the TLS handshake to validate the identity of both communicating parties Both the server and the end point device certificates are signed by well known trusted certificate authorities Server authentication occurs when a server has a certificate signed by a certificate authority The certificate is only used for the client to validate the identity of the server it is connected to After the TLS connection is established the server can identify the IP Deskphone through a user name and password Both mutual authentication and server authentication are supported Note that TLS is supported on TCP only not UDP New configuration parameters to support TLS include Specify TCP and TLS ports typically 5060 and 5061 o SERVER_TCP_PORT lt X gt _ lt Y gt lt Port number gt o SERVER_TLS_PORT lt X gt _ lt Y gt lt Port number gt Specify listening ports o SIP_TCP_PORT lt port number gt e g SIP_TCP_PORT 5060 o SIP_TLS_PORT 5061 e g SIP_TLS_POR
19. ions that improve efficiency collaboration customer service and competitiveness For more information please visit www avaya com 2010 Avaya Inc All Rights Reserved Avaya and the Avaya Logo are trademarks of Avaya Inc and are registered in the United States and other countries All trademarks identified by TM or SM are registered marks trademarks and service marks respectively of Avaya Inc All other trademarks are the property of their respective owners Avaya may also have trademark rights in other terms used herein References to Avaya include the Nortel Enterprise business which was acquired as of December 18 2009 02 10 SIP Release 3 2 Readme AVAYA INTELLIGENT COMMUNICATIONS 17
20. kphones operate in one of two modes e _ Pv4 enabled and IPv6 stack disabled default e both IPv4 and IPv6 stacks enabled Three new device Configuration flags have been added to support IPv6 e IPV6_ENABLE Y N enables IPv6 e PREFER_IPV6 orders media lines e IPV6_STATELESS Y N enables stateless auto configuration Media security SRTP With SIP 3 2 media security is provided using SRTP RFC 3711 The deskphone may operate in one of two modes Secure Only or Best Effort With secure only a secure path must be setup before the call will complete otherwise the call will fail With best effort an attempt will be made to initiate a secure call however if it is not possible the call will continue in an unsecure mode 3 different SRTP modes are available e BE Cap Neg Best Effort Capabilities negotiation e BE 2M Lines Best Effort 2M lines negotiation e SecureOnly If there is no SRTP on the endpoint then call fails These are controlled by the following configuration parameters e SRTP_ENABLED lt Yes No gt e SRTP_MODE lt BE 2MLines BE CapNeg SecureOnly gt Cipher support e SRTP_CIPHER_ lt X gt lt AES_128 SHA1_32 AES_128 SHA1_80 gt e MKI Master Key Identifier e Support of MKI and non MKI modes e MKI_ENABLE lt Yes No gt Note SRTP is not supported on CS1000 Release 7 0 It is planned for CS1000 Release 7 5 SIP Release 3 2 Readme 6 AVAYA Signaling Security TLS SIP 3 2 also provides Signal
21. luded in SIP Release 3 2 for the 11xx and 12xx Series deskphones including e Improved Licensing e SIP Support for 1220 1230 and 1165E IP Deskphones e Shared Call Appearances CS1000 e Pv6 Support e SRTP Media Security e TLS Signaling Security e Certificate based Authentication e Enhanced Screensavers e Background images e Support for Avaya Aura Communication Manager Session Manager A description of each feature is provided in the following sections SIP Release 3 2 Readme 2 avaya com AVAYA Improved Licensing Licensing was introduced in the SIP 3 0 release With SIP 3 2 the following changes are made to the licensing mechanism e The Standard feature set is now available on all desksets without a token This provides a basic set of SIP features conforming to RFC 3261 SIPPING 19 at no additional cost e Now when the phone is registered to a recognized Avaya call server Avaya Aura AS 5300 CS1000 or CS2100 the Extended feature set is available as well without a token The Advanced feature set is reserved for Federal and DoD features on the AS 5300 call server only The feature packages have been re organized o Wideband is part of Standard feature set o _IPv6 and Broadworks SCA are part of Extended feature set o Security is now part of the Extended feature set The feature packages are now as follows SIP Core Features RFC3261 SIPPING 19 Standard Feature Set Standard Feature Set Extended Fea
22. mber MADN Single Call Appearance SCA feature for Genband Nortel Communication Servers CS2000 and CS2100 SIP URI Dialing SIP Release 3 2 for IP Deskphones supports URI dialing Some Communication Servers may not support URI dialing and in such cases attempt to make calls from the Call softkey menu of SIP may not complete and users may get an error message In addition as a valid URI is to be entered for monitoring a user s presence state the presence state updates may not be reflected accurately with Communication Servers that may not support URI SIP Release 3 2 Readme 14 AVAYA Expansion Module SIP Release 3 2 for IP Deskphones supports the Expansion Module The Expansion Module is a hardware accessory that connects to the IP Phone and provides additional line appearances and feature keys Up to three expansion modules are supported The 1120E 1140E 1165E 1220 and 1230 IP Deskphones can have up to 54 additional line feature keys with three Expansion Modules The Expansion Module is equipped with a graphical pixel based grayscale LCD display area beside the 18 line feature keys Each of the 18 physical keys on the Expansion Module has a 10 character display label This label is set automatically however the user can edit the label using the controls on the IP Deskphone The Expansion Module will display similar graphics for the key label and icons as displayed on the IP phone Programmed keys are stored in the user profile E
23. ort nortel com go main jsp cscat DOCUMENTATION amp poid 19423 SIP Release 3 2 for IP Deskphones documentation includes the following materials e NN43112 101 Avaya 1120E IP Deskphone with SIP Software User Guide e NN43113 101 Avaya 1140E IP Deskphone with SIP Software User Guide e NN43170 100 Avaya 1165E IP Deskphone with SIP Software User Guide e NN43110 301 Avaya 1100 Series Expansion Module for SIP Software User Guide e NN43170 600 SIP Software for Avaya 1100 Series IP Deskphones Administration e NN43170 101 Avaya 1220 IP Deskphone with SIP Software User Guide e NN43170 102 Avaya 1230 IP Deskphone with SIP Software User Guide e NN43139 100 Avaya 1200 Series Expansion Module for SIP Software User Guide e NN43170 601 SIP Software for Avaya 1200 Series IP Deskphones Administration Note that beginning with SIP 3 2 the individual Administration Guides for each phone are now combined into a common Administration Guide for the series Hence only 2 administration guides are now available one for the 1100 series and one for the 1200 series phones SIP Release 3 2 Readme 12 AVAYA Product Advisements The following is a list of advisements associated with SIP software release 3 2 Some advisements remain from previous releases of software whereas other advisements reflect new or changed behavior introduced with SIP software release 3 2 Advisements that are new to SIP software release 3 2 or have changed since previous releases of SIP so
24. s e Fast Ethernet and switch in all models e 2 position footstand wallmount e Wideband enabled to be supported in a future release e Both Key Expansion Modules are supported e 18 Key with LED indicators and paper labels e 12 Key with self labeling LCD shown o Cascadable up to 7 LCD modules e Does not support Bluetooth wireless technology USB graphics or screen savers Support for 1165E Deskphone SIP Release 3 2 also provides support for the Avaya 1165E Deskphone The 1165E is an advanced design with the following features e High resolution 240 x 360 pixels QVGA color Liquid Crystal Display e Integrated 10 100 1000 Base T Auto Sensing Ethernet switch for shared PC access one LAN port and one PC port e Superior quality audio experience including wideband ready speakerphone and handset e Power efficiency gains with support for IEEE 802 3af PoE standard as a Class 2 device e 8 programmable line feature keys e User selectable background e Digital picture slideshow e Bluetooth 2 1 support o Headset profile e Integrated USB port for keyboard mouse or headset connectivity e USB flash drive support e Avaya 1100 Series Expansion Module Support o upto 54 additional line feature keys with three Expansion Modules SIP Release 3 2 Readme 4 AVAYA Shared Call Appearances CS1000 Onl SIP Release 3 2 provides support for Shared Call Appearances SCA in CS1000 environments beginning with CS1000 Releas
25. ture Set 3 way call call conference Authentication Security Multi Level Precedence and Preemption MLPP Audio Codecs Standard Wideband Bluetooth Headset support 1140E 1165E only Call Origination Busy Auto Login Logout Call Server Service Package DoD Network Background Image Expansion Module support FIPS Certified Busy Lamp Field BLF Instant Messaging Distinctive Ringing Media Security SRTP Downloadable Ringtones Multi user Login support Image Screensaver amp Lock NAT Traversal STUN Standard Font Languages Proactive Voice Quality Mgt Multiple calls per user PC Client Control Server Failover Redundancy Signaling Security TLS Session Timers USB headset support for audio SNTP Time Server IPv6 Support Speed Dial List Broadsoft Broadworks SCA Transfer to VM softkey USB memory stick Hotline One token waived when connected to an Avaya call server SIP Release 3 2 Readme AVAYA Support for 1220 and 1230 IP Deskphones SIP Release 3 2 expands support for the Avaya 1220 and 1230 Deskphones The 1200 series is a similar industrial design to the 1100 Series but is mechanically simpler and lower cost with a smaller processor It is ideal for SMB and cost sensitive applications Key features e Delivers High Quality Audio Experience same full duplex algorithm as 1100 serie
26. y AST protocol implemented in Avaya 9600 Series IP deskphones SIP Nevertheless Communication Manager and Session Manager have the capability to extend some advanced telephony features to non AST telephones A summary of the features supported follows Basic Calling features Extension to extension call Basic call to non SIP phones Intercept tones displays Reorder with Mute Redial Call Waiting Do Not Disturb Speed Dial buttons Redial from call logs Compressed codecs G 729A G 729AB G 722 64k Message Waiting Support SIPPING RFC 5359 Features Call Hold Consultation Hold Music on Hold Unattended Transfer Attended Transfer Call Forward Unconditional Call Forward Busy Via Feature Name Extension FNE Call Forward No Answer Via Feature Name Extension FNE 3 way conference 3rd party added 3 way conference 3rd party joins Find Me Modular Messaging Find Me Incoming Call Screening Via Class of Restriction Outgoing Call Screening Via Class of Restriction Call Park Unpark Via Feature Name Extension FNE Call Pickup Via FNE Via Feature Name Extension FNE Additional Station Side Features Calling Name Number Block Via Feature Name Extension FNE SIP Release 3 2 Readme 10 AVAYA Directed Call Pick Up Via Feature Name Extension FNE Priority Call Via Feature Name Extension
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