Avaya 9600 manual Httpexception, Domains, Httpproxy, Interdigittimeout, L2QAUD

Page 81

Administering Options for the 9600 Series SIP IP Telephones

Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name

Default

Description and Value Range

 

Value

 

HTTPEXCEPTION

" " (Null)

Domains to be excluded for SCEP. String representing

DOMAINS

 

zero or one domains in a URL of 0 to 255 characters in

 

 

dotted decimal or DNS name format with multiple

 

 

domains delimited by commas.

HTTPPORT

80

Destination TCP port used for requests to the HTTP

 

 

server during initialization. Range is 0 - 65535.

 

 

Note: For SIP Release 1.0, there should be no need to

 

 

set this parameter to values other than default value.

HTTPPROXY

" " (Null)

Zero or one IP or DNS address of the HTTP server for

 

 

SCEP. 0 to 255 characters in dotted decimal or DNS

 

 

name format followed by a colon and port number. The

 

 

colon and port number are optional. If this parameter is

 

 

not null, this (proxy) transport address is used to set up

 

 

the HTTP connection as the transport protocol for SCEP.

HTTPSRVR

0.0.0.0

List of IP Address(es) or DNS Name(s) of HTTP file

 

 

server(s) used to download telephone files. HTTP server

 

 

addresses can be in dotted decimal or DNS format, and

 

 

must be separated by commas (0-255 ASCII characters,

 

 

including commas).

ICMPDU

1

Controls whether ICMP Destination Unreachable

 

 

messages will be processed. Values are:

 

 

0=DU messages not transmitted

 

 

1= DU messages not transmitted in response to specific

 

 

events

 

 

2= DU message with code 2 will be transmitted in case of

 

 

specific events

ICMPRED

0

Controls whether ICMP Redirect messages will be

 

 

processed. Values are:

 

 

0 = Redirect messages will neither be transmitted nor

 

 

received Redirect messages will be supported

 

 

1 = Redirect messages will not be transmitted, but

 

 

received Redirect messages will be supported per RFC

 

 

1122

INTER_DIGIT_TIMEOUT

5

This is the timeout that takes place when user stops

 

 

inputting digits. The timeout is treated as digit collection

 

 

completion, and when it occurs, the application sends out

 

 

an invite. Range in seconds of 1 to 10.

IPADD

0.0.0.0

IP Address of the telephone. Range is 7 to 15 ASCII

 

 

characters (less than the default string length) defining

 

 

one IP Address in dotted-decimal format.

L2Q

0

Requests 802.1Q tagging mode (auto/on/off). Values are:

 

 

0 = auto

 

 

1 = on

 

 

2 = off

L2QAUD

6

Layer 2 audio priority value. Range from 0 to 7.

L2QSIG

6

Layer 2 signaling priority value. Range from 0 to 7.

 

 

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Issue 2 December 2007 81

Page 80 Page 82
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Contents Issue 2 December Security Introduction SIP Enablement Services SES Administration Telephone Software and Binary Files Appendix C Sample Station Forms 121 Index 135 About This Guide IntroductionIntroduction SIP Software Release Change HistoryRsvp VmonWhat’s New in SIP Software Release For more information, see Visiting User AdministrationWhat’s New in SIP Software Release PHY1DUPLEX PHY2DUPLEX Interdigitdialingtimeoutduration UseexchangecontactsUseexchangecalendar Other Documentation Document OrganizationPage Series IP Telephones Administration Overview and RequirementsAdministration Overview and Requirements Quality of Service AdministrationIP Addresses Tagging and VlanLldp Dhcp Parameter Data PrecedenceAdministrative Checklist Administrative ProcessLAN Network RequirementsTelephone to LLDP-Enabled Network Telephone Initialization ProcessTelephone to Network Telephone to Dhcp Server Telephone and File ServerTelephone and the SES Server Error Conditions Page Network Requirements Network AssessmentHardware Requirements Dhcp Server Server RequirementsRequired Network Information HTTP/HTTPS ServerNetwork Time Protocol NTP Server Other Network Considerations Registration and AuthenticationReliability and Performance QoSIeee 802.1D and 802.1Q Parameters in Real-Time Possible Values SIP Station Number PortabilityUDP TCP/UDP Port UtilizationTransmitted Packets Source = SIP IP Telephone UDP/TCPCNA RTP RangeSecurity Registration and Authentication Page Communication Manager Administrative Requirements Communication Manager AdministrationSwitch Compatibility Call Server RequirementsSystem-Level Preparation Tasks SIP Trunk AdministrationCall Routing Administration IP Interface and AddressesUDP Port Selection Rsvp and RTCP/SRTCP L2QVLANL2QSIG Voice Mail Integration Auto HoldCall Transfer Considerations Telephone Administration Conferencing Call ConsiderationsFNU Call Forward CM/SIP IP Telephone Configuration Requirements CM/SIP IP Telephone Configuration Requirements Services SES Button Assignments Xxxxxx where Administering StationsAdministering Features Communication Manager Administration Using the Web Browser to Configure SES SIP Enablement Services SES AdministrationClick on Launch Administration Web Interface IntroductionClick Continue Server Administration Software ChecklistDhcp and File Servers Configuring Dhcp for 9600 Series SIP IP Telephones Dhcp Server AdministrationParameters Set by Dhcp Description Dhcp Generic Setup Option 51 Dhcp lease time Option 42 Sntp ServerOption 52 Overload Option, if desired Option 53 Dhcp message typeOption 59 Dhcp lease rebind time Dhcpack Setting of System Values Set to Select Start--Settings--Control PanelWindows NT 4.0 Dhcp Server Define the Telephone IP Address Range Set the Subnet Mask Select Start--Programs--Admin Tools--DHCP ManagerClick Edit Array Select Scope under Dhcp OptionsSelect the Global parameter under Dhcp Options Activating the Leases Select Defaults from the DHCPOptions menuWindows 2000 Dhcp Server Select Start--Programs--Administrative Tools--DHCPCompleting the New Scope Wizard dialog box displays Change the Data Type to String Under Predefined Options and Values, click AddHttp Generic Setup Telephone Software and Binary Files General Download ProcessSoftware Series SIP IP Telephone Scripts and Binary Files Choosing the Right Binary File and Upgrade Script FileSettings File Upgrade Script FileContents of the Settings File SET Sntpsrvr 192.168.1.111 SET Gmtoffset -500 SET Dstoffset # END Group System ValueAdministering Options for the 9600 Series SIP IP Telephones Administering Telephone OptionsAgchead AgchandAgcspkr AudasysConfig Server CallfwddelayConfigserver CalltransferCountry USA SettingmodeCoverageaddr CurrentskinDOT1XSTAT DOT1XEAPS MD5Dscpaud DscpsigDefault ENABLEG722 ENABLEG711UENABLEG726 ENABLEG729Removaltimer EnablerediallistEnhdialstat ExchangeserverHttpexception InterdigittimeoutDomains HttpproxyLANG0STAT LldpenabledLanguages LocalloglevelLogos LogcategoryMediaencryption MsgnumMwisrvr MusicsrvrMycertcaid Mycertcn $SERIALNOMycertwait NodigitstimeoutOutbound SubscriptionPHY2PRIO PhnnumofsaPHY2VLAN PoeconssupportPPM PresenceserverProvideedited DialingProvideexchange QkloginstatCalendar Provide LogoutRtcpmon RtcpcontRtcpmonport RtpportlowSipportsecure SipsignalSipconferenceconti NUESnmpadd SkinsSnmpstring SpeakerstatTimeformat TcpkeepalivetimeTlssrvrid TrustcertsRegistrationtimer VutimerVumode WaitforVlan Considerations Vlan TaggingWmlproxy Vlan Detection Vlan Default Value and Priority TaggingVlan Separation Vlan Separation Rules Then Ieee DNS Addressing802.1X Supplicant Operation 802.1X Pass-Through and Proxy LogoffAdministering Telephone Options Link Layer Discovery Protocol Lldp MAC / PHY BootnameTIA Lldp MED LLDP-MED TIA Lldp MEDSubtype = Parameter Name Impact Impact of TLVs on System Parameter ValuesVisiting User Administration PoeconsSupport Emergency Number Administration Language Selection Enhanced Local Dialing Enhanced Local Dialing Enhanced Local Dialing Requirements Customizing Telephone Applications and Options Administering Applications and OptionsAvaya a Menu Administration Administering Standard Avaya Menu EntriesAdministering the WML Browser Avaya a Menu Administration Page Appendix a Glossary of Terms Channel MediaEncryption ProxyUnnamed SignalingRegistration VoIPPage Ietf Documents ITU Documents ISO/IEC, ANSI/IEEE Documents Appendix B Related DocumentationPage Appendix C Sample Station Forms StationStation Options Enhanced Call Forwarding Station Feature OptionsStation Site Data Station Feature Button AssignmentsAbbreviated Dialing Button AssignmentsService Observing FEATURE-RELATED System ParametersCall Center System Parameters EAS VectoringVlan IP Address MappingIP Network Region Media ParametersTCP Signaling Link Establishment for Avaya H.323 Endpoints Backup Servers in Priority Order Security ProceduresINTER-GATEWAY Alternate ROUTING/DIAL Plan Transparency Stations with OFF-PBX Telephone IntegrationTDD/TTY FAXCAC Igar WANNumbering PUBLIC/UNKNOWN Format CPNRtcp Monitor Server IP Dtmf Transmission ModeAutomatic Trace Route on Media Gateway IP EndpointFRL Class of RestrictionClass Restriction COR Optional Features IP Port Capacities Used Optional FeaturesID SID UsedARS? DS1 MSP?ATMS? System Parameters Customer-Options Optional Features screen Numerical Index108 Snmp
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9600 specifications

The Avaya 9600 series is a robust line of desktop phones designed to meet the needs of modern business communication. Tailored for users who require advanced functionalities, the 9600 series offers a rich set of features that enhance productivity and collaboration in the workplace.

One of the standout features of the Avaya 9600 series is its high-resolution graphical display. The intuitive interface with customizable menu options makes it easy for users to navigate through various functions and settings. The color screen enhances visibility and allows for clearer visual communication, essential for today’s fast-paced business environments.

In terms of audio quality, the Avaya 9600 series excels with high-definition voice technology. Users can benefit from crystal-clear audio, which minimizes misunderstandings during calls and enhances remote collaboration. The phones are equipped with full-duplex speakerphone functionality, ensuring that all parties can be heard clearly during conference calls.

Another significant characteristic of the Avaya 9600 series is its support for a wide range of communication protocols, including Voice over IP (VoIP) and SIP (Session Initiation Protocol). This versatility enables seamless integration with various communication systems, making the 9600 series suitable for companies of all sizes. The phones can connect to both cloud-based and on-premise solutions, allowing businesses to choose the best communication strategy for their needs.

User personalization is a key aspect of the 9600 series. The phones come with programmable function keys that allow users to customize their setup according to individual preferences and frequently used features. Additionally, the series supports Bluetooth and USB connectivity, enabling users to connect a variety of headsets and accessories for enhanced audio options.

Security is paramount in business communications, and the Avaya 9600 series addresses this with advanced security features. The phones support encryption protocols to protect sensitive information during calls, ensuring that businesses can communicate confidentially without the risk of eavesdropping.

Lastly, the Avaya 9600 series is designed for scalability. As organizations grow, the phones can easily be integrated into existing systems or expanded to accommodate additional users without requiring significant changes to the infrastructure.

In summary, the Avaya 9600 series is a powerful communication tool that combines advanced features, superior audio quality, and robust security to empower businesses. Its adaptability and user-friendly design make it a preferred choice for organizations aiming to enhance their communication efficiency and productivity.
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