Polycom SIP 3.1 manual Local Digit Map, 7000, the maximum speed-dial index is, Auto-reject

Page 66

Administrator’s Guide SoundPoint IP / SoundStation IP

Element

Permitted Values

Interpretation

 

 

 

sd

Null, 1 to 9999

speed-dial index

 

 

Associates a particular entry with a speed dial bin for one-touch

 

 

dialing or dialing from the speed dial menu.

 

 

Note: On the SoundPoint IP 330/320 and the SoundStation IP 6000

 

 

and 7000, the maximum speed-dial index is 99.

 

 

 

rt

Null, 1 to 21

ring type

 

 

When incoming calls can be associated with a directory entry by

 

 

matching the address fields, this field is used to specify ring type to

 

 

be used.

 

 

 

dc

UTF-8 encoded string

divert contact

 

containing digits (the

The forward-to address for the autodivert feature.

 

user part of a SIP

 

 

 

URL) or a string that

 

 

constitutes a valid SIP

 

 

URL

 

 

 

 

ad

0,1

auto divert

 

 

If set to 1, automatically diverts callers that match the directory entry

 

 

to the address specified in divert contact.

 

 

Note: If auto-divert is enabled, it has precedence over auto-reject.

 

 

 

ar

0,1

auto-reject

 

 

If set to 1, automatically rejects callers that match the directory entry.

 

 

Note: If auto-divert is also enabled, it has precedence over

 

 

auto-reject.

 

 

 

bw

0,1

buddy watching

 

 

If set to 1, add this contact to the list of watched phones.

 

 

 

bb

0,1

buddy block

 

 

If set to 1, block this contact from watching this phone.

 

 

 

Local Digit Map

The phone has a local digit map feature to automate the setup phase of number-only calls. When properly configured, this feature eliminates the need for using the Dial or Send soft key when making outgoing calls. As soon as a digit pattern matching the digit map is found, the call setup process will complete automatically. The configuration syntax is based on recommendations in 2.1.5 of RFC 3435. The phone behavior when the user dials digits that do not match the digit map is configurable. It is also possible to strip a trailing # from the digits sent or to replace certain matched digits (with the introduction of “R” to the digit map).

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Contents SIP Disclaimer Copyright NoticeAbout 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 Introducing the SoundPoint IP / SoundStation IP Family SoundPoint IP Desktop PhonesAdministrator’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 Where SoundPoint IP / SoundStation IP Phones Fit SoundPoint IP / SoundStation IP Phones onBootROM Session Initiation Protocol Application ArchitectureApplication Configuration Master Configuration Files Application Configuration FilesApplication Configuration Files Resource Files Available Features Ring tones Synthesized tones Contact directoriesOverview 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 Setting Up the Network Dhcp or Manual TCP/IP SetupFor more information on Dhcp options, go to Supported Provisioning Protocols FTP Tftp Http HttpsModifying the Network Configuration Certificate Authority List on page C-1Main Menu Dhcp Menu Server Menu Ethernet Menu Syslog Menu Dhcp Menu EM PowerName Possible Values Description Dhcp Client Phone IP AddressMenu Possible Name Values DescriptionSection, Server Menu Server Menu MenuServer. Refer to Supported Provisioning Protocols on Dhcp on page C-23Name Possible Values Description Or later. Passive FTP is still supportedPassword, this will be ignored Password these characters if they are correctly escapedUsing the method specified in RFC This will be ignoredSetting Setting unless you want to disable the PC portEthernet menu Setting Up the Boot Server Refer to Basic Logging level/change/ and renderEach phone may open multiple connections to the server Information, contact your Certified Polycom ResellerCreate account and home directory These permissions, but will not be able to upload filesSip.cfg Phone1.cfg 000000000000.cfg Directory~.xml Deploying Phones From the Boot ServerYou must decide on a boot server security policy SoundPointIP-dictionary.xmlone of each supported languageConfiguration Files on page C-4 Provisioning PhonesConfiguration on page A-4 SIP/ on page A-10PhoneMACaddress.cfg 5EL@ Provisioning SoundStation IP 7000 Phones Using CLink Upgrading SIP Application Supporting SoundPoint IP and SoundStation IP PhonesSupporting SoundPoint IP 300 and 500 Phones To upgrade your SIP application Administrator’s Guide SoundPoint IP / SoundStation IP Setting Up Basic Features Configuring SoundPoint IP / SoundStation IP Phones LocallyThis chapter also provides instructions on Administrator’s Guide SoundPoint IP / SoundStation IP Call Timer Call LogCall Waiting Called Party Identification Calling Party IdentificationMissed Call Notification Connected Party Identification Context Sensitive Volume ControlCentral boot server Customizable Audio Sound EffectsMessages and voice messages are waiting Message Waiting IndicationDistinctive Incoming Call Treatment Saf/ on page A-30 or Sound Effects se/ on page A-31Address-directory Distinctive RingingDistinctive Call Waiting Xml LocalDo Not Disturb Handset, Headset, and Speakerphone116 Local Contact Directory Userpreferences/on page A-107Xml DirectOry.xml ?xml version=1.0 encoding=UTF-8 standalone=yes ? directoryUTF-8’s variable length encoding DirectoryElement Permitted Values Interpretation Space is added between first and last namesLocal Digit Map 7000, the maximum speed-dial index isAuto-reject Microphone Mute Soft Key Activated User InterfaceSpeed Dial Boot server Ethernet Time and Date DisplayEthernet Switch Idle Display AnimationGraphic Display Backgrounds Their phoneYour choice For images, select a filename. For example Automatic Off-Hook Call PlacementCall Hold AutoOffHook/ on page A-112Call Transfer Hold/localReminder/ on page A-67Local / Centralized Conferencing Manage ConferencesCall Forward Directed Call Pick-Up Call Park/Retrieve Setting Up Advanced FeaturesGroup Call Pick-Up Last Call ReturnConfiguring Your System Configurable Feature Keys Feature Key Layouts on page C-12Multiple Line Keys per Registration Multiple Call AppearancesShared Call Appearances Bridged Line Appearance Busy Lamp Field Live Communications Server 2005 Integration on Customizable Fonts and IndicatorsEchnicalBulletinspub.html Attendant.uriMultilingual User Interface Central bootInstant Messaging Server Sip.cfgSwedish Fonts, refer to Fonts font/ on page A-72Saf/ on page A-30 Synthesized Call Progress TonesMicrobrowser Call Progress Patterns on page A-33Real-Time Transport Protocol Ports Network Address Translation Corporate DirectoryNat/ on page A-120 Settings Basic Preferences Corporate Directory View This section contains the following information Recording and Playback of Audio Calls 670 have a functioning USB portDisplay Daisy-Chaining Phones Provisioning Phones Over CLink Enhanced Feature Keys Efk Efklist Efkprompt Version Special CharactersEfklist This element describes behavior of enhanced feature keyEfk This element contains the following parametersEfkprompt This element describes the behavior of the user promptsVersion Special CharactersVersion efk.version=2 Macro Action Prompt Macro Substitution Expanded Macros Macro ActionCall. The use of refer method is call server Using Invite if no active call or Dtmf if an activePrompt Macro Substitution Dependentand may require the addition of star codesExpanded Macros Collected. The macros are case sensitivePrompt is not required for every macro Contact Directory File Format onExamples Action String Example Configuration File ChangesEnhanced Feature Key XML Files Action stringUsing Call Park Key Contact Directory ChangesWell as others mapped to Park Return and Call Pickup Configurable Soft Keys New Call End Call Split Join Forward MyStatus and Buddies Hold, Transfer, and ConferenceUpdate the sip.cfg configuration as follows Softkey.feature.newcall =103 Update sip.cfg as follows Voice Mail Integration Multiple Registrations Server server/ on page A-7Automatic Call Distribution Server RedundancyServer/ on page A-7, and Registration reg/ on page A-107 DNS SIP Server Name Resolution For Outgoing Calls Invite Fallback Phone Configuration ConfiguredPhone Operation for Registration Presence Immediately with business contacts Boot server Address-directoryMicrosoft Live Communications Server 2005 Integration Examples onRoamingbuddies/ on page A-122 Roamingprivacy/ on page A-123Refer to Roaming Buddies roamingbuddies/ on page A-122 Refer to Roaming Privacy roamingprivacy/ on page A-123Set reg.x.auth.password to the LCS password Set the reg.x.server.y.address to the LCS server nameLocate the roamingprivacy attribute Access URL in SIP Message Web Content Examples User Interface Signaling ChangesWeb 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 Voice Activity Detection Low-Delay Audio Packet TransmissionJitter Buffer and Packet Error Concealment Dynamic Noise Reduction Treble/Bass ControlsDtmf Event RTP Payload Acoustic Echo CancellationDtmf Tone Generation DTMF/ on page A-28Audio Codecs Following table summarizes the phone’s audio codec supportEffective Background Noise Suppression Comfort Noise FillOn page A-38 and Codec Profiles audioProfile/ on page A-41 IP Type-of-Service Automatic Gain ControlIeee 802.1p/Q Voice Quality Monitoring Three types of quality reports can be enabledPeriodic-Generated during a call at a configurable period ThresholdTreble/Bass Controls Setting Up Security FeaturesDynamic Noise Reduction Monitoring/ on page A-52Pwd/length/ on page A-89 Local User and Administrator Privilege LevelsCustom Certificates Tinspub.htmlConfiguration changes can performed locally Incoming Signaling ValidationConfiguration File Encryption Secure Real-Time Transport ProtocolConfiguring SoundPoint IP / SoundStation IP Phones Locally Configuration on page A-124Device.cfg Passwords Troubleshooting Your SoundPoint IP / SoundStation IP Phones Error Messages BootROM Error MessagesApplication Error Messages Status Menu Log Files Scheduled Logging Manual Log Upload Reading a Boot Log Following figure shows a portion of a boot log fileTesting Phone Hardware Reading an Application LogFollowing figure shows a portion of an application log file Power and Startup Symptom Problem Corrective ActionControls Access to Screens and Systems To Rebooting the Phone onCalling Displays Phone on page C-10Oice/soundpointip/VoIPTechnicalBullet AudioUpgrading Inspub.htmlAdministrator’s Guide SoundPoint IP / SoundStation IP Configuration Files Master Configuration Files One will cause a reboot loop Application Configuration CONFIGFILES=phone1MACADDRESS.cfg, sip.cfg MISCFILES=Configuration Files Protocol voIpProt This configuration attribute is defined as followsAttribute Permitted Default Interpretation Values This attribute includesVoIpProt.server.x.transport is set to Permitted Attribute Values Default InterpretationIf voIpProt.server.x.address is a If voIpProt.server.x.transport is set toVoIpProt.server.x.address is an IP VoIpProt.SIP.lcs Parameter if set to 1 when the parameter To 1 default isPermitted Attribute Values Default Interpretation Reg.x.auth.optimizedInFailover takes This attribute also includes Ept = 325,326,327,328,329,330Lcl.ml.lang.tags.x in Multilingual ml Outbound Proxy outboundProxy Request Validation requestValidation Due to the additional signaling requiredAlert Information alertInfo May have a negative performance impactConference Setup conference Special Events specialEventUDP, TCP, or TLS Supported when configured with the valuesDial Plan dialplan Dialplan.applyToCallListDialThis attributes also includes Digit Map digitmap Routing routingConfiguration Files Attribute Permitted Values Default Interpretation Server server Emergency emergencyEmergency emergency Localization lclServer server Multilingual ml Date and Time datetimeLcl.ml.lang.menu.2 Attribute Permitted Values InterpretationLcl.ml.lang.menu.1 Lcl.ml.lang.menu.3Lcl.ml.lang.tags.1 = Lcl.datetime.date.longFormatLcl.datetime.date.dateTop Zh-cn,zhq=0.9,enq=0.8Optional Set lcl.ml.lang to be the new languageregion string User Preferences up Permitted Attribute Values InterpretationOnIntensity, it will be replaced with OnIntensity valueTones tones Dual Tone Multi-Frequency Dtmf Chord-Sets chordOnly be enabled when tone.dtmf.viaRtp is DisabledBe enabled when tone.dtmf.viaRtp is Sampled Audio for Sound Effects saf Sound Effects se Following table, x is the sampled audio file numberTo SoundPointIPWelcome.wav Patterns pat Ring type rt Instruction Meaning ExampleMiscellaneous Patterns Call Progress PatternsCall progress Use within phone Pattern number Ringer pattern number Default description Ringer PatternsCall progress Pattern number Use within phone Miscellaneous Patterns Miscellaneous Pattern number Use within phoneSequential Patterns on page A-34Defined in Call Progress Patterns on page A-33 Voice Settings voice Codec Preferences codecPref Codec Profiles audioProfile Following voice codecs are supportedThese codecs include 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 Voice.vadEnable parameter If voice.vadEnable is set to 0, add attribute lineCentral Report Collector collector Alert Reports alert Server server RTCP-XR rtcpxr Central Report Collector collectorNable.periodic is set 1, since Alert Reports alert RTCP-XR rtcpxr Quality of Service QOSFollowing settings control the 802.1p/Q userpriority field Ethernet Ieee 802.1p/Q ethernet IP TOS IPOther other These parameters apply to RTP packetsCall Control callControl RTP rtp Call Control callControlRTP rtp Basic TCP/IP Tcpip Qos.ip.callControl…Attribute Permitted Default Values Permitted Attribute Values Default Interpretation Start.date is ignored If fixedDayEnable is set toStart.dayOfWeek Stop.dayOfWeekTcpIpApp.port.rtp.filterByIp Must be enabled for this to workRTP rtp TcpIpApp.port.rtp.filterByPortDefault value is used Web Server httpdValue that is out of range, Call Handling Configuration call Configuration cfgReg.x.callsPerLineKey. Refer to Registration If call.stickyAutoLineSeize is set to 1, thisBroadWorks calls server only. You must change Shared Calls shared Hold, Local Reminder hold/localReminderSoundPoint IP 330/320 only Value if your organization uses a different callPhones a quick press and release of the line ServersIP 4000, 6000, and 7000 phones. For other Key will resume a call whereas pressingDir.local.volatile.4meg Directory dirLocal Directory local Corporate Directory corp Dir.local.volatile.8meg, thisEnter the speed dial index followed SoundPoint IP 320/330 is disabledRead only, speed dial entry on By #Used for display purposes only Leg tagged parameter. This Dir.corp.viewPersistence600, and 601 legacy phones, use Prevents slow behavior after exitingPresence pres Fonts fontSoundPoint IP 320, 330, 430, 500 SoundPoint IP 550, 560, 600, 601, 650,IP330 font IP330 This configuration attribute is defined as follows Keys keyFollowing table lists the functions that are available FunctionsBuilt-in default solid pattern is displayed Backgrounds bgSame to display correctly on grayscale Individual phone when the user lightens or Darkens the graphic during previewIndicators ind Following indicators are used by the phoneBitmaps bitmap Platform IP300/, IP 330/, IP400 IP500/, IP600/, IP4000/,IP7000/ on page A-80 Attribute Permitted Interpretation Values IP300/, IP330/, IP400/, IP500/, IP600IP4000/, and IP7000/ tag above LEDs led Following table, x is the LED numberEvent Logging log Level InterpretationThree formats are available for the event timestamp Two types of logging are supportedType Example Log.render.level maps to Syslog Menu onYou do not change this value Server is set up for this Set starting with log.sched.x where x identifies the taskSupport append mode unless Uploaded if no new events haveEncryption encryption Password Lengths pwd/length Security secLicense license Provisioning prov RAM Disk ramdiskDelay delay Request requestValue Feature feature Resource res Finder finder Quotas quotasPhones, this value is internally replaced by 2X Value. For the SoundStation IP 6000Phones, this value is internally replaced by 4X This value is internally replaced by 2X the value. For Microbrowser mbSoundStation IP 4000, 6000, and 7000 phones Replaced by 4X the value650, and 670 and SoundStation IP 4000 Miscellaneous XML errors can occur onSoundPoint IP 430, 501, 550, 560, 600 7000 phonesIf mb.main.idleTimeout Function is selectedUsed. Refer to User Preferences up/ on Applications apps Detrimental effect on performance of the phoneApps.push.password must be set to non-Null Non-Null valuesValues Peer Networking pnet DNS Cache dnsNaptr NAPTR/ attribute SRV SRV Http//tools.ietf.org/html/rfc2915 Http//tools.ietf.org/html/rfc2782 Soft Keys softkey Macro Definition onPermitted Attribute Values Default Interpretation New Call and Callers soft keys For this soft key to be displayedPer-Phone Configuration Parameters includeRegistration reg User Preferences userpreferencesSpecified in sip.cfg in Server server/ on Is non-Null, all of the reg.x.server.y.xxxParameters will override the parameters A-7Refer to Call Handling Configuration call Shared line counts as a call for every phoneSharing that registration If reg.x.serverFeatureControl.cf is not Calls call VoIpProt.SIP.strictLineSeize isSet to 1 enabled, this parameter is ignored. For More information, refer to SIP SIP/ onSylantro call server only If call.missedCallTracking.x.enabled is Forwarding is enabled, this Parameter is enabledDiversion divert Calls can be automatically diverted when the phone is busy Divert.x.contact will beDialplan.x.digitmap is not Enabled, this parameter isServer-base call forwarding is Plan dialplan/ on page A-17Dialplan.x.applyToUserDial When present, and if Digit Map digitmap/ on Server/ on page A-118 Message Waiting Indicator mwiMessaging msg Chosen. Refer to Voice Mail Integration onNetwork Address Translation nat VoIpProt.local.signalPort in sip.cfg Attendant attendantRoaming Buddies roamingbuddies Value 0 if the call server is Microsoft LiveCommunications Server VoIpProt.SIP.strictLineSeize Roaming Privacy roamingprivacyUser Preferences userpreferences VoIPProt.SIP.lineSeize.retries,Flash Parameter Configuration This flash attributes are defined as follows EnabledSetup on For example, if device.net.ipAddress.set =Server address is preserved Refer to Basic Logging level/change/ Render/ on page A-86Menu on Administrator’s Guide SoundPoint IP / SoundStation IP Session Initiation Protocol SIP RFC and Internet Draft Support Request Support Following SIP request messages are supportedMethod Supported Header Support Following SIP request headers are supportedHeader Supported Header Supported Response Support Following SIP responses are supportedResponse Supported 3xx Responses Redirection 5xx Responses Server Failure Transfer Hold ImplementationReliability of Provisional Responses Third Party Call ControlShared Call Appearance Signaling Bridged Line Appearance SignalingMiscellaneous Administrative Tasks Trusted Certificate Authority ListAdministrator’s Guide SoundPoint IP / SoundStation IP Miscellaneous Administrative Tasks Encrypting Configuration Files Encrypted and unencrypted file are the same Changing the Key on the PhoneOption. This shows the digest field Encryption/ on page A-89Adding a Background Logo Model Width Height Color DepthRGB Values Color RGB Values Decimal HexadecimalModel Associate Parameter IP330 IP400 BitmapsIP300 IP300 IP330 IP500 IP500Model BootROM SIP Application BootROM/SIP Application DependenciesAnimations Indicators Migration DependenciesMultiple Key Combinations About three seconds BootROM until the password prompt appearsIP 4000 and 6000 6, 8 and * dial pad keys IP 301 The two Line keys and the Up and Down arrow keysSoundPoint IP Default Feature Key LayoutsSoundStation IP 4000, 6000, and 7000 models 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 Assigning a Vlan ID Using Dhcp VLAN-A=10 VLAN-A=0x0a VLAN-A=012Parsing Vendor ID Information Miscellaneous Administrative Tasks Product, Model, and Part Number Mapping Product Name Model Name Product Part NumberSelect Save Config Disabling PC Ethernet PortPress 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 Index NumericsAdministrator’s Guide SoundPoint IP / SoundStation IP Dhcp IP TOS call control callControl A-58 IP400 font A-74Administrator’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 New or Changed Features Electronic Hookswitch Graphic Display BackgroundsDistribution Zip File Configuration File Changes Backlight IntensityMetrics for listening and conversational quality Gains gain Receive Equalization rxEq Transmit Equalization txEqBackground 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.