Polycom SIP 3.1 manual Resource Files

Page 25

Overview

The following figure shows one possible layout of the central provisioning method.

Manual Configuration

When the manual configuration method is employed, any changes made are stored in a configuration override file. This file is stored on the phone, but a copy will also be uploaded to the central boot server if one is being used. When the phone boots, this file is loaded by the application after any centrally provisioned files have been read, and its settings will override those in the centrally provisioned files.

This can create a lot of confusion about where parameters are being set, and so it is best to avoid using the manual method unless you have good reason to do so.

Resource Files

In addition to the application and the configuration files, the phones may require resource files that are used by some of the advanced features. These files are optional, but if the particular feature is being employed, these files are required.

Some examples of resource files include:

Language dictionaries

Custom fonts

2 - 7

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Contents SIP Copyright Notice DisclaimerAbout This Guide Administrator’s Guide SoundPoint IP / SoundStation IP Contents Administrator’s Guide SoundPoint IP / SoundStation IP Contents Troubleshooting Your SoundPoint IP / SoundStation IP Phones Contents Administrator’s Guide SoundPoint IP / SoundStation IP SoundPoint IP Desktop Phones Introducing the SoundPoint IP / SoundStation IP FamilyAdministrator’s Guide SoundPoint IP / SoundStation IP SoundPoint IP SoundPoint IP 550/560 SoundPoint IP 600/601 SoundStation IP Conference Phones Currently supported conference phones are SoundStation IP Key Features of Your SoundPoint IP / SoundStation IP Phones SoundPoint IP 600 Administrator’s Guide SoundPoint IP / SoundStation IP Overview SoundPoint IP / SoundStation IP Phones on Where SoundPoint IP / SoundStation IP Phones FitSession Initiation Protocol Application Architecture BootROMApplication Master Configuration Files Application Configuration Files ConfigurationApplication Configuration Files Resource Files Ring tones Synthesized tones Contact directories Available FeaturesOverview Microsoft Live Communications Server Overview Administrator’s Guide SoundPoint IP / SoundStation IP New Features in SIP Administrator’s Guide SoundPoint IP / SoundStation IP Setting up Your System Dhcp or Manual TCP/IP Setup Setting Up the NetworkFor more information on Dhcp options, go to FTP Tftp Http Https Supported Provisioning ProtocolsCertificate Authority List on page C-1 Modifying the Network ConfigurationMain Menu Dhcp Menu Server Menu Ethernet Menu Syslog Menu Name Possible Values Description Dhcp Client EM PowerDhcp Menu Phone IP AddressPossible Name Values Description MenuSection, Server Menu Menu Server MenuName Possible Values Description Dhcp on page C-23Server. Refer to Supported Provisioning Protocols on Or later. Passive FTP is still supportedUsing the method specified in RFC Password these characters if they are correctly escapedPassword, this will be ignored This will be ignoredSetting unless you want to disable the PC port SettingEthernet menu Refer to Basic Logging level/change/ and render Setting Up the Boot ServerCreate account and home directory Information, contact your Certified Polycom ResellerEach phone may open multiple connections to the server These permissions, but will not be able to upload filesYou must decide on a boot server security policy Deploying Phones From the Boot ServerSip.cfg Phone1.cfg 000000000000.cfg Directory~.xml SoundPointIP-dictionary.xmlone of each supported languageProvisioning Phones Configuration Files on page C-4SIP/ on page A-10 Configuration on page A-4PhoneMACaddress.cfg 5EL@ Provisioning SoundStation IP 7000 Phones Using CLink Supporting SoundPoint IP and SoundStation IP Phones Upgrading SIP ApplicationSupporting SoundPoint IP 300 and 500 Phones To upgrade your SIP application Administrator’s Guide SoundPoint IP / SoundStation IP Configuring SoundPoint IP / SoundStation IP Phones Locally Setting Up Basic FeaturesThis chapter also provides instructions on Administrator’s Guide SoundPoint IP / SoundStation IP Call Log Call TimerCall Waiting Calling Party Identification Called Party IdentificationMissed Call Notification Central boot server Context Sensitive Volume ControlConnected Party Identification Customizable Audio Sound EffectsDistinctive Incoming Call Treatment Message Waiting IndicationMessages and voice messages are waiting Saf/ on page A-30 or Sound Effects se/ on page A-31Distinctive Call Waiting Distinctive RingingAddress-directory Xml LocalHandset, Headset, and Speakerphone Do Not Disturb116 Userpreferences/on page A-107 Local Contact DirectoryOry.xml DirectXml ?xml version=1.0 encoding=UTF-8 standalone=yes ? directoryElement Permitted Values Interpretation DirectoryUTF-8’s variable length encoding Space is added between first and last names7000, the maximum speed-dial index is Local Digit MapAuto-reject Soft Key Activated User Interface Microphone MuteSpeed Dial Time and Date Display Boot server EthernetIdle Display Animation Ethernet SwitchTheir phone Graphic Display BackgroundsYour choice Call Hold Automatic Off-Hook Call PlacementFor images, select a filename. For example AutoOffHook/ on page A-112Hold/localReminder/ on page A-67 Call TransferManage Conferences Local / Centralized ConferencingCall Forward Directed Call Pick-Up Group Call Pick-Up Setting Up Advanced FeaturesCall Park/Retrieve Last Call ReturnConfiguring Your System Feature Key Layouts on page C-12 Configurable Feature KeysMultiple Call Appearances Multiple Line Keys per RegistrationShared Call Appearances Bridged Line Appearance Busy Lamp Field EchnicalBulletinspub.html Customizable Fonts and IndicatorsLive Communications Server 2005 Integration on Attendant.uriInstant Messaging Central bootMultilingual User Interface Server Sip.cfgFonts, refer to Fonts font/ on page A-72 SwedishMicrobrowser Synthesized Call Progress TonesSaf/ on page A-30 Call Progress Patterns on page A-33Real-Time Transport Protocol Ports Corporate Directory Network Address TranslationNat/ on page A-120 Settings Basic Preferences Corporate Directory View This section contains the following information 670 have a functioning USB port Recording and Playback of Audio CallsDisplay Daisy-Chaining Phones Provisioning Phones Over CLink Efk Efklist Efkprompt Version Special Characters Enhanced Feature KeysEfk This element describes behavior of enhanced feature keyEfklist This element contains the following parametersThis element describes the behavior of the user prompts EfkpromptSpecial Characters VersionVersion efk.version=2 Macro Action Macro Action Prompt Macro Substitution Expanded MacrosPrompt Macro Substitution Using Invite if no active call or Dtmf if an activeCall. The use of refer method is call server Dependentand may require the addition of star codesPrompt is not required for every macro Collected. The macros are case sensitiveExpanded Macros Contact Directory File Format onExamples Enhanced Feature Key XML Files Configuration File ChangesAction String Example Action stringContact Directory Changes Using Call Park KeyWell as others mapped to Park Return and Call Pickup Configurable Soft Keys MyStatus and Buddies Hold, Transfer, and Conference New Call End Call Split Join ForwardSoftkey.feature.newcall = Update the sip.cfg configuration as follows103 Update sip.cfg as follows Voice Mail Integration Server server/ on page A-7 Multiple RegistrationsServer Redundancy Automatic Call DistributionServer/ on page A-7, and Registration reg/ on page A-107 DNS SIP Server Name Resolution For Outgoing Calls Invite Fallback Configured Phone ConfigurationPhone Operation for Registration Presence Microsoft Live Communications Server 2005 Integration Boot server Address-directoryImmediately with business contacts Examples onRoamingprivacy/ on page A-123 Roamingbuddies/ on page A-122Refer to Roaming Privacy roamingprivacy/ on page A-123 Refer to Roaming Buddies roamingbuddies/ on page A-122Set the reg.x.server.y.address to the LCS server name Set reg.x.auth.password to the LCS passwordLocate the roamingprivacy attribute Web Content Examples User Interface Signaling Changes Access URL in SIP MessageWeb Content Status Indication Settings Menu Static DNS Cache Example Dns.cache.A.1 , dns.cache.A.2 , and so on Set to null to force SRV lookups Display of Warnings from SIP Headers Setting Up Audio Features Jitter Buffer and Packet Error Concealment Low-Delay Audio Packet TransmissionVoice Activity Detection Dynamic Noise Reduction Treble/Bass ControlsDtmf Tone Generation Acoustic Echo CancellationDtmf Event RTP Payload DTMF/ on page A-28Following table summarizes the phone’s audio codec support Audio CodecsEffective Comfort Noise Fill Background Noise SuppressionOn page A-38 and Codec Profiles audioProfile/ on page A-41 Automatic Gain Control IP Type-of-ServiceIeee 802.1p/Q Periodic-Generated during a call at a configurable period Three types of quality reports can be enabledVoice Quality Monitoring ThresholdDynamic Noise Reduction Setting Up Security FeaturesTreble/Bass Controls Monitoring/ on page A-52Custom Certificates Local User and Administrator Privilege LevelsPwd/length/ on page A-89 Tinspub.htmlConfiguration File Encryption Incoming Signaling ValidationConfiguration changes can performed locally Secure Real-Time Transport ProtocolConfiguration on page A-124 Configuring SoundPoint IP / SoundStation IP Phones LocallyDevice.cfg Passwords Troubleshooting Your SoundPoint IP / SoundStation IP Phones BootROM Error Messages Error MessagesApplication Error Messages Status Menu Log Files Scheduled Logging Manual Log Upload Following figure shows a portion of a boot log file Reading a Boot LogReading an Application Log Testing Phone HardwareFollowing figure shows a portion of an application log file Symptom Problem Corrective Action Power and StartupControls To Rebooting the Phone on Access to Screens and SystemsCalling Phone on page C-10 DisplaysUpgrading AudioOice/soundpointip/VoIPTechnicalBullet Inspub.htmlAdministrator’s Guide SoundPoint IP / SoundStation IP Configuration Files Master Configuration Files One will cause a reboot loop CONFIGFILES=phone1MACADDRESS.cfg, sip.cfg MISCFILES= Application ConfigurationConfiguration Files Attribute Permitted Default Interpretation Values This configuration attribute is defined as followsProtocol voIpProt This attribute includesIf voIpProt.server.x.address is a Permitted Attribute Values Default InterpretationVoIpProt.server.x.transport is set to If voIpProt.server.x.transport is set toVoIpProt.server.x.address is an IP VoIpProt.SIP.lcs To 1 default is Parameter if set to 1 when the parameterPermitted Attribute Values Default Interpretation Reg.x.auth.optimizedInFailover takes Ept = 325,326,327,328,329,330 This attribute also includesLcl.ml.lang.tags.x in Multilingual ml Outbound Proxy outboundProxy Alert Information alertInfo Due to the additional signaling requiredRequest Validation requestValidation May have a negative performance impactSpecial Events specialEvent Conference Setup conferenceDial Plan dialplan Supported when configured with the valuesUDP, TCP, or TLS Dialplan.applyToCallListDialDigit Map digitmap Routing routing This attributes also includesConfiguration Files Server server Emergency emergency Attribute Permitted Values Default InterpretationServer server Localization lclEmergency emergency Multilingual ml Date and Time datetimeLcl.ml.lang.menu.1 Attribute Permitted Values InterpretationLcl.ml.lang.menu.2 Lcl.ml.lang.menu.3Lcl.datetime.date.dateTop Lcl.datetime.date.longFormatLcl.ml.lang.tags.1 = Zh-cn,zhq=0.9,enq=0.8Optional Set lcl.ml.lang to be the new languageregion string Permitted Attribute Values Interpretation User Preferences upOnIntensity value OnIntensity, it will be replaced withDual Tone Multi-Frequency Dtmf Chord-Sets chord Tones tonesDisabled Only be enabled when tone.dtmf.viaRtp isBe enabled when tone.dtmf.viaRtp is Sampled Audio for Sound Effects saf Following table, x is the sampled audio file number Sound Effects seTo SoundPointIPWelcome.wav Instruction Meaning Example Patterns pat Ring type rtCall Progress Patterns Miscellaneous PatternsCall progress Use within phone Pattern number Ringer Patterns Ringer pattern number Default descriptionCall progress Pattern number Use within phone Miscellaneous Pattern number Use within phone Miscellaneous PatternsPatterns on page A-34 SequentialDefined in Call Progress Patterns on page A-33 Voice Settings voice These codecs include Following voice codecs are supportedCodec Preferences codecPref Codec Profiles audioProfile Codec Preferences codecPrefPermitted Attribute Values Default Interpretation Voice.codecPref.IP7000.G722 Codec Profiles audioProfile Attribute Default Attribute Default Attribute Default Acoustic Echo Cancellation aec Acoustic Echo Suppression aes Background Noise Suppression ns Feature Receive Equalization rxEq Transmit Equalization txEq Attribute Default If voice.vadEnable is set to 0, add attribute line Voice.vadEnable parameterCentral Report Collector collector Alert Reports alert Central Report Collector collector Server server RTCP-XR rtcpxrNable.periodic is set 1, since Alert Reports alert Following settings control the 802.1p/Q userpriority field Quality of Service QOSRTCP-XR rtcpxr Ethernet Ieee 802.1p/Q ethernet IP TOS IPCall Control callControl These parameters apply to RTP packetsOther other RTP rtp Call Control callControlRTP rtp Qos.ip.callControl… Basic TCP/IP TcpipAttribute Permitted Default Values Permitted Attribute Values Default Interpretation Start.dayOfWeek If fixedDayEnable is set toStart.date is ignored Stop.dayOfWeekRTP rtp Must be enabled for this to workTcpIpApp.port.rtp.filterByIp TcpIpApp.port.rtp.filterByPortWeb Server httpd Default value is usedValue that is out of range, Configuration cfg Call Handling Configuration callIf call.stickyAutoLineSeize is set to 1, this Reg.x.callsPerLineKey. Refer to RegistrationSoundPoint IP 330/320 only Shared Calls shared Hold, Local Reminder hold/localReminderBroadWorks calls server only. You must change Value if your organization uses a different callIP 4000, 6000, and 7000 phones. For other ServersPhones a quick press and release of the line Key will resume a call whereas pressingLocal Directory local Corporate Directory corp Directory dirDir.local.volatile.4meg Dir.local.volatile.8meg, thisRead only, speed dial entry on SoundPoint IP 320/330 is disabledEnter the speed dial index followed By #Used for display purposes only 600, and 601 legacy phones, use Dir.corp.viewPersistenceLeg tagged parameter. This Prevents slow behavior after exitingFonts font Presence presSoundPoint IP 550, 560, 600, 601, 650, SoundPoint IP 320, 330, 430, 500IP330 font IP330 Keys key This configuration attribute is defined as followsFunctions Following table lists the functions that are availableBackgrounds bg Built-in default solid pattern is displayedSame to display correctly on grayscale Darkens the graphic during preview Individual phone when the user lightens orFollowing indicators are used by the phone Indicators indBitmaps bitmap IP500/, IP600/, IP4000/, Platform IP300/, IP 330/, IP400IP7000/ on page A-80 IP300/, IP330/, IP400/, IP500/, IP600 Attribute Permitted Interpretation ValuesIP4000/, and IP7000/ tag above Following table, x is the LED number LEDs ledLevel Interpretation Event Logging logTwo types of logging are supported Three formats are available for the event timestampType Example Syslog Menu on Log.render.level maps toYou do not change this value Support append mode unless Set starting with log.sched.x where x identifies the taskServer is set up for this Uploaded if no new events haveSecurity sec Encryption encryption Password Lengths pwd/lengthLicense license RAM Disk ramdisk Provisioning provRequest request Delay delayValue Feature feature Finder finder Quotas quotas Resource resValue. For the SoundStation IP 6000 Phones, this value is internally replaced by 2XPhones, this value is internally replaced by 4X SoundStation IP 4000, 6000, and 7000 phones Microbrowser mbThis value is internally replaced by 2X the value. For Replaced by 4X the valueSoundPoint IP 430, 501, 550, 560, 600 Miscellaneous XML errors can occur on650, and 670 and SoundStation IP 4000 7000 phonesFunction is selected If mb.main.idleTimeoutUsed. Refer to User Preferences up/ on Detrimental effect on performance of the phone Applications appsNon-Null values Apps.push.password must be set to non-NullValues DNS Cache dns Peer Networking pnetNaptr NAPTR/ attribute SRV SRV Http//tools.ietf.org/html/rfc2915 Http//tools.ietf.org/html/rfc2782 Macro Definition on Soft Keys softkeyPermitted Attribute Values Default Interpretation For this soft key to be displayed New Call and Callers soft keysParameters include Per-Phone ConfigurationUser Preferences userpreferences Registration regParameters will override the parameters Is non-Null, all of the reg.x.server.y.xxxSpecified in sip.cfg in Server server/ on A-7Shared line counts as a call for every phone Refer to Call Handling Configuration callSharing that registration If reg.x.serverFeatureControl.cf is not VoIpProt.SIP.strictLineSeize is Calls callMore information, refer to SIP SIP/ on Set to 1 enabled, this parameter is ignored. ForSylantro call server only If call.missedCallTracking.x.enabled is Parameter is enabled Forwarding is enabled, thisDiversion divert Divert.x.contact will be Calls can be automatically diverted when the phone is busyServer-base call forwarding is Enabled, this parameter isDialplan.x.digitmap is not Plan dialplan/ on page A-17Dialplan.x.applyToUserDial When present, and if Digit Map digitmap/ on Messaging msg Message Waiting Indicator mwiServer/ on page A-118 Chosen. Refer to Voice Mail Integration onNetwork Address Translation nat Attendant attendant VoIpProt.local.signalPort in sip.cfgValue 0 if the call server is Microsoft Live Roaming Buddies roamingbuddiesCommunications Server User Preferences userpreferences Roaming Privacy roamingprivacyVoIpProt.SIP.strictLineSeize VoIPProt.SIP.lineSeize.retries,Flash Parameter Configuration Setup on EnabledThis flash attributes are defined as follows For example, if device.net.ipAddress.set =Server address is preserved Render/ on page A-86 Refer to Basic Logging level/change/Menu on Administrator’s Guide SoundPoint IP / SoundStation IP Session Initiation Protocol SIP RFC and Internet Draft Support Following SIP request messages are supported Request SupportMethod Supported Following SIP request headers are supported Header SupportHeader Supported Header Supported Following SIP responses are supported Response SupportResponse Supported 3xx Responses Redirection 5xx Responses Server Failure Reliability of Provisional Responses Hold ImplementationTransfer Third Party Call ControlBridged Line Appearance Signaling Shared Call Appearance SignalingTrusted Certificate Authority List Miscellaneous Administrative TasksAdministrator’s Guide SoundPoint IP / SoundStation IP Miscellaneous Administrative Tasks Encrypting Configuration Files Option. This shows the digest field Changing the Key on the PhoneEncrypted and unencrypted file are the same Encryption/ on page A-89Model Width Height Color Depth Adding a Background LogoColor RGB Values Decimal Hexadecimal RGB ValuesModel Associate Parameter IP300 IP300 IP330 BitmapsIP330 IP400 IP500 IP500Animations Indicators BootROM/SIP Application DependenciesModel BootROM SIP Application Migration DependenciesMultiple Key Combinations IP 4000 and 6000 6, 8 and * dial pad keys BootROM until the password prompt appearsAbout three seconds IP 301 The two Line keys and the Up and Down arrow keysSoundStation IP 4000, 6000, and 7000 models Default Feature Key LayoutsSoundPoint IP Key ID FunctionSoundPoint IP 320/330 SoundPoint IP OPER0 14 # 12 11OPER Key ID SoundPoint IP 550/560/600/601/650/670 SoundStation IP Key ID Internal Key Functions Label Function LCR Label Function VLAN-A=10 VLAN-A=0x0a VLAN-A=012 Assigning a Vlan ID Using DhcpParsing Vendor ID Information Miscellaneous Administrative Tasks Product Name Model Name Product Part Number Product, Model, and Part Number MappingDisabling PC Ethernet Port Select Save ConfigPress Administrator’s Guide SoundPoint IP / SoundStation IP Third Party Software OpenSSL Third Party Software Zlib Copyright and Permission Notice Administrator’s Guide SoundPoint IP / SoundStation IP Numerics IndexAdministrator’s Guide SoundPoint IP / SoundStation IP IP TOS call control callControl A-58 IP400 font A-74 DhcpAdministrator’s Guide SoundPoint IP / SoundStation IP SDP SDP A-9 Sipsip A-10 POLYCOM, INC Application Programming Interface License API License Agreement for Development Purposes Support Services Export Controls Page Addendum to SIP 3.1 Administrator’s Guide Electronic Hookswitch Graphic Display Backgrounds New or Changed FeaturesDistribution Zip File Backlight Intensity Configuration File ChangesMetrics for listening and conversational quality Gains gain Transmit Equalization txEq Receive Equalization rxEqBackground bg Administrator’s Guide Addendum for the SoundPoint IP Multiple Key Combinations and Default Key Layout Key ID Administrator’s Guide Addendum for the SoundPoint IP
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SIP 3.1 specifications

Polycom SIP 3.1 is an advanced session initiation protocol designed to enhance voice and video communication in various business environments. As a pivotal component of Polycom’s telecommunication solutions, SIP 3.1 offers several features and characteristics that cater to the evolving needs of modern enterprises, particularly those that rely on seamless and efficient communication.

One of the standout features of Polycom SIP 3.1 is its robust interoperability. This protocol supports a wide range of endpoints and platforms, allowing organizations to integrate their existing systems with new technologies effortlessly. This flexibility ensures that businesses can leverage their previous investments while upgrading to the latest communication tools.

Another key aspect of Polycom SIP 3.1 is its enhanced call management capabilities. The protocol facilitates efficient call handling, enabling users to manage multiple calls seamlessly. Features like call hold, transfer, and conferencing are streamlined, which enhances productivity by allowing for efficient collaboration among team members. Additionally, it is optimized for low latency and high-quality audio, ensuring that conversations are clear and free from disruptions.

Security is paramount in today’s digital landscape, and Polycom SIP 3.1 addresses this concern with advanced encryption standards. By integrating security features such as Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP), it protects sensitive communication from unauthorized access and ensures that data remains confidential throughout the call.

Polycom SIP 3.1 also boasts compatibility with various video codecs, making it a versatile choice for video conferencing. This compatibility ensures high-quality video streams, which is essential for effective visual communication in remote meetings. Furthermore, the support for the H.264 codec provides efficient bandwidth usage, making high-definition video conferencing accessible, even in varying network conditions.

Moreover, the protocol provides strong support for presence and instant messaging, which enhances real-time communication among users. This integration of voice, video, and messaging capabilities fosters a more connected and collaborative work environment, allowing teams to engage effectively regardless of their geographical locations.

In summary, Polycom SIP 3.1 stands out as a sophisticated solution tailored to meet the demands of modern business communication. With its emphasis on interoperability, call management, security, video quality, and real-time collaboration, it caters to companies of all sizes seeking to optimize their communication infrastructure and enhance productivity in the workplace. As businesses continue to navigate the complexities of digital communication, Polycom SIP 3.1 remains a compelling choice in the market.