Polycom SIP 3.1 manual Connected Party Identification, Context Sensitive Volume Control

Page 59

 

 

Configuring Your System

 

 

 

 

Configuration changes can performed centrally at the boot server:

 

 

 

Central

Configuration file:

Turn this feature on or off.

(boot server)

sip.cfg

For more information, refer to Feature <feature/> on page A-92.

 

 

 

 

Configuration file:

Specify per-registration whether all missed-call events or only

 

phone1.cfg

remote/server-generated missed-call events will be displayed.

 

 

For more information, refer to Missed Call Configuration

 

 

<serverMissedCall/> on page A-112.

 

 

 

Connected Party Identification

The identity of the remote party to which the user has connected is displayed and logged, if the name and ID is provided by the call server. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion.

There are no related configuration changes.

Context Sensitive Volume Control

The volume of user interface sound effects, such as the ringer, and the receive volume of call audio is adjustable. While transmit levels are fixed according to the TIA/EIA-810-A standard, receive volume is adjustable. For SoundPoint IP and phones, if using the default configuration parameters, the receive handset/headset volume resets to nominal after each call to comply with regulatory requirements. Handsfree volume persists with subsequent calls.

Configuration changes can performed centrally at the boot server:

Central (boot server)

Configuration file:

sip.cfg

Adjust receive and handset/headset volume.

For more information, refer to Volume Persistence <volume/> on page A-42.

Customizable Audio Sound Effects

Audio sound effects used for incoming call alerting and other indications are customizable. Sound effects can be composed of patterns of synthesized tones or sample audio files. The default sample audio files may be replaced with alternates in .wav file format. Supported .wav formats include:

mono G.711 (13-bit dynamic range, 8-khz sample rate)

mono L16/16000 (16-bit dynamic range, 16-kHz sample rate)

mono L16/32000 (16-bit dynamic range, 32-kHz sample rate)

mono L16/48000 (16-bit dynamic range, 48-kHz sample rate)

4 - 5

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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 ProtocolsMain Menu Dhcp Menu Server Menu Ethernet Menu Syslog Menu Modifying the Network ConfigurationCertificate Authority List on page C-1 Phone IP Address EM PowerName Possible Values Description Dhcp Client Dhcp MenuSection, Server Menu MenuPossible Name Values Description Menu Server MenuOr later. Passive FTP is still supported Dhcp on page C-23Name Possible Values Description Server. Refer to Supported Provisioning Protocols onThis will be ignored Password these characters if they are correctly escapedUsing the method specified in RFC Password, this will be ignoredEthernet menu SettingSetting unless you want to disable the PC port Refer to Basic Logging level/change/ and render Setting Up the Boot ServerThese permissions, but will not be able to upload files Information, contact your Certified Polycom ResellerCreate account and home directory Each phone may open multiple connections to the serverSoundPointIP-dictionary.xmlone of each supported language Deploying Phones From the Boot ServerYou must decide on a boot server security policy Sip.cfg Phone1.cfg 000000000000.cfg Directory~.xmlProvisioning Phones Configuration Files on page C-4PhoneMACaddress.cfg Configuration on page A-4SIP/ on page A-10 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 This chapter also provides instructions on Setting Up Basic FeaturesConfiguring SoundPoint IP / SoundStation IP Phones Locally Administrator’s Guide SoundPoint IP / SoundStation IP Call Waiting Call TimerCall Log Missed Call Notification Called Party IdentificationCalling Party Identification Customizable Audio Sound Effects Context Sensitive Volume ControlCentral boot server Connected Party IdentificationSaf/ on page A-30 or Sound Effects se/ on page A-31 Message Waiting IndicationDistinctive Incoming Call Treatment Messages and voice messages are waitingXml Local Distinctive RingingDistinctive Call Waiting Address-directory116 Do Not DisturbHandset, Headset, and Speakerphone Userpreferences/on page A-107 Local Contact Directory?xml version=1.0 encoding=UTF-8 standalone=yes ? directory DirectOry.xml XmlSpace is added between first and last names DirectoryElement Permitted Values Interpretation UTF-8’s variable length encodingAuto-reject Local Digit Map7000, the maximum speed-dial index is Speed Dial Microphone MuteSoft Key Activated User Interface Time and Date Display Boot server EthernetIdle Display Animation Ethernet SwitchYour choice Graphic Display BackgroundsTheir phone AutoOffHook/ on page A-112 Automatic Off-Hook Call PlacementCall Hold For images, select a filename. For exampleHold/localReminder/ on page A-67 Call TransferManage Conferences Local / Centralized ConferencingCall Forward Directed Call Pick-Up Last Call Return Setting Up Advanced FeaturesGroup Call Pick-Up Call Park/RetrieveConfiguring 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 Attendant.uri Customizable Fonts and IndicatorsEchnicalBulletinspub.html Live Communications Server 2005 Integration onServer Sip.cfg Central bootInstant Messaging Multilingual User InterfaceFonts, refer to Fonts font/ on page A-72 SwedishCall Progress Patterns on page A-33 Synthesized Call Progress TonesMicrobrowser Saf/ on page A-30Real-Time Transport Protocol Ports Nat/ on page A-120 Network Address TranslationCorporate Directory Settings Basic Preferences Corporate Directory View This section contains the following information Display Recording and Playback of Audio Calls670 have a functioning USB port Daisy-Chaining Phones Provisioning Phones Over CLink Efk Efklist Efkprompt Version Special Characters Enhanced Feature KeysThis element contains the following parameters This element describes behavior of enhanced feature keyEfk EfklistThis element describes the behavior of the user prompts EfkpromptVersion efk.version=2 VersionSpecial Characters Macro Action Macro Action Prompt Macro Substitution Expanded MacrosDependentand may require the addition of star codes Using Invite if no active call or Dtmf if an activePrompt Macro Substitution Call. The use of refer method is call serverContact Directory File Format on Collected. The macros are case sensitivePrompt is not required for every macro Expanded MacrosExamples Action string Configuration File ChangesEnhanced Feature Key XML Files Action String ExampleWell as others mapped to Park Return and Call Pickup Using Call Park KeyContact Directory Changes Configurable Soft Keys MyStatus and Buddies Hold, Transfer, and Conference New Call End Call Split Join Forward103 Update the sip.cfg configuration as followsSoftkey.feature.newcall = Update sip.cfg as follows Voice Mail Integration Server server/ on page A-7 Multiple RegistrationsServer/ on page A-7, and Registration reg/ on page A-107 Automatic Call DistributionServer Redundancy DNS SIP Server Name Resolution For Outgoing Calls Invite Fallback Phone Operation for Registration Phone ConfigurationConfigured Presence Examples on Boot server Address-directoryMicrosoft Live Communications Server 2005 Integration Immediately with business contactsRoamingprivacy/ 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-122Locate the roamingprivacy attribute Set reg.x.auth.password to the LCS passwordSet the reg.x.server.y.address to the LCS server name 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 Dynamic Noise Reduction Treble/Bass Controls Low-Delay Audio Packet TransmissionJitter Buffer and Packet Error Concealment Voice Activity DetectionDTMF/ on page A-28 Acoustic Echo CancellationDtmf Tone Generation Dtmf Event RTP PayloadEffective Audio CodecsFollowing table summarizes the phone’s audio codec support On page A-38 and Codec Profiles audioProfile/ on page A-41 Background Noise SuppressionComfort Noise Fill Ieee 802.1p/Q IP Type-of-ServiceAutomatic Gain Control Threshold Three types of quality reports can be enabledPeriodic-Generated during a call at a configurable period Voice Quality MonitoringMonitoring/ on page A-52 Setting Up Security FeaturesDynamic Noise Reduction Treble/Bass ControlsTinspub.html Local User and Administrator Privilege LevelsCustom Certificates Pwd/length/ on page A-89Secure Real-Time Transport Protocol Incoming Signaling ValidationConfiguration File Encryption Configuration changes can performed locallyDevice.cfg Configuring SoundPoint IP / SoundStation IP Phones LocallyConfiguration on page A-124 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 LogFollowing figure shows a portion of an application log file Testing Phone HardwareReading an Application Log Symptom Problem Corrective Action Power and StartupControls To Rebooting the Phone on Access to Screens and SystemsCalling Phone on page C-10 DisplaysInspub.html AudioUpgrading Oice/soundpointip/VoIPTechnicalBulletAdministrator’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 This attribute includes This configuration attribute is defined as followsAttribute Permitted Default Interpretation Values Protocol voIpProtIf voIpProt.server.x.transport is set to Permitted Attribute Values Default InterpretationIf voIpProt.server.x.address is a 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 Lcl.ml.lang.tags.x in Multilingual ml This attribute also includesEpt = 325,326,327,328,329,330 Outbound Proxy outboundProxy May have a negative performance impact Due to the additional signaling requiredAlert Information alertInfo Request Validation requestValidationSpecial Events specialEvent Conference Setup conferenceDialplan.applyToCallListDial Supported when configured with the valuesDial Plan dialplan UDP, TCP, or TLSDigit Map digitmap Routing routing This attributes also includesConfiguration Files Server server Emergency emergency Attribute Permitted Values Default InterpretationMultilingual ml Date and Time datetime Localization lclServer server Emergency emergencyLcl.ml.lang.menu.3 Attribute Permitted Values InterpretationLcl.ml.lang.menu.1 Lcl.ml.lang.menu.2Zh-cn,zhq=0.9,enq=0.8 Lcl.datetime.date.longFormatLcl.datetime.date.dateTop Lcl.ml.lang.tags.1 =Optional 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 To SoundPointIPWelcome.wav Sound Effects seFollowing table, x is the sampled audio file number Instruction Meaning Example Patterns pat Ring type rtCall progress Use within phone Pattern number Miscellaneous PatternsCall Progress Patterns Call progress Pattern number Use within phone Ringer pattern number Default descriptionRinger Patterns Miscellaneous Pattern number Use within phone Miscellaneous PatternsDefined in Call Progress Patterns on page A-33 SequentialPatterns on page A-34 Voice Settings voice Codec Preferences codecPref Following voice codecs are supportedThese codecs include Codec Preferences codecPref Codec Profiles audioProfilePermitted 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 Central Report Collector collector Alert Reports alert Voice.vadEnable parameterIf voice.vadEnable is set to 0, add attribute line Nable.periodic is set 1, since Server server RTCP-XR rtcpxrCentral Report Collector collector Alert Reports alert Ethernet Ieee 802.1p/Q ethernet IP TOS IP Quality of Service QOSFollowing settings control the 802.1p/Q userpriority field RTCP-XR rtcpxrRTP rtp Call Control callControl These parameters apply to RTP packetsCall Control callControl Other otherRTP rtp Qos.ip.callControl… Basic TCP/IP TcpipAttribute Permitted Default Values Permitted Attribute Values Default Interpretation Stop.dayOfWeek If fixedDayEnable is set toStart.dayOfWeek Start.date is ignoredTcpIpApp.port.rtp.filterByPort Must be enabled for this to workRTP rtp TcpIpApp.port.rtp.filterByIpValue that is out of range, Default value is usedWeb Server httpd Configuration cfg Call Handling Configuration callIf call.stickyAutoLineSeize is set to 1, this Reg.x.callsPerLineKey. Refer to RegistrationValue if your organization uses a different call Shared Calls shared Hold, Local Reminder hold/localReminderSoundPoint IP 330/320 only BroadWorks calls server only. You must changeKey will resume a call whereas pressing ServersIP 4000, 6000, and 7000 phones. For other Phones a quick press and release of the lineDir.local.volatile.8meg, this Directory dirLocal Directory local Corporate Directory corp Dir.local.volatile.4megBy # SoundPoint IP 320/330 is disabledRead only, speed dial entry on Enter the speed dial index followedUsed for display purposes only Prevents slow behavior after exiting Dir.corp.viewPersistence600, and 601 legacy phones, use Leg tagged parameter. ThisFonts 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 orBitmaps bitmap Indicators indFollowing indicators are used by the phone IP7000/ on page A-80 Platform IP300/, IP 330/, IP400IP500/, IP600/, IP4000/, IP4000/, and IP7000/ tag above Attribute Permitted Interpretation ValuesIP300/, IP330/, IP400/, IP500/, IP600 Following table, x is the LED number LEDs ledLevel Interpretation Event Logging logType Example Three formats are available for the event timestampTwo types of logging are supported You do not change this value Log.render.level maps toSyslog Menu on Uploaded if no new events have Set starting with log.sched.x where x identifies the taskSupport append mode unless Server is set up for thisSecurity sec Encryption encryption Password Lengths pwd/lengthLicense license RAM Disk ramdisk Provisioning provValue Delay delayRequest request Feature feature Finder finder Quotas quotas Resource resPhones, this value is internally replaced by 4X Phones, this value is internally replaced by 2XValue. For the SoundStation IP 6000 Replaced by 4X the value Microbrowser mbSoundStation IP 4000, 6000, and 7000 phones This value is internally replaced by 2X the value. For7000 phones Miscellaneous XML errors can occur onSoundPoint IP 430, 501, 550, 560, 600 650, and 670 and SoundStation IP 4000Used. Refer to User Preferences up/ on If mb.main.idleTimeoutFunction is selected Detrimental effect on performance of the phone Applications appsValues Apps.push.password must be set to non-NullNon-Null values Naptr NAPTR/ attribute SRV SRV Peer Networking pnetDNS Cache dns 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 regA-7 Is non-Null, all of the reg.x.server.y.xxxParameters will override the parameters Specified in sip.cfg in Server server/ onSharing that registration Refer to Call Handling Configuration callShared line counts as a call for every phone If reg.x.serverFeatureControl.cf is not VoIpProt.SIP.strictLineSeize is Calls callSylantro call server only Set to 1 enabled, this parameter is ignored. ForMore information, refer to SIP SIP/ on If call.missedCallTracking.x.enabled is Diversion divert Forwarding is enabled, thisParameter is enabled Divert.x.contact will be Calls can be automatically diverted when the phone is busyPlan dialplan/ on page A-17 Enabled, this parameter isServer-base call forwarding is Dialplan.x.digitmap is notDialplan.x.applyToUserDial When present, and if Digit Map digitmap/ on Chosen. Refer to Voice Mail Integration on Message Waiting Indicator mwiMessaging msg Server/ on page A-118Network Address Translation nat Attendant attendant VoIpProt.local.signalPort in sip.cfgCommunications Server Roaming Buddies roamingbuddiesValue 0 if the call server is Microsoft Live VoIPProt.SIP.lineSeize.retries, Roaming Privacy roamingprivacyUser Preferences userpreferences VoIpProt.SIP.strictLineSeizeFlash Parameter Configuration For example, if device.net.ipAddress.set = EnabledSetup on This flash attributes are defined as followsServer address is preserved Menu on Refer to Basic Logging level/change/Render/ on page A-86 Administrator’s Guide SoundPoint IP / SoundStation IP Session Initiation Protocol SIP RFC and Internet Draft Support Method Supported Request SupportFollowing SIP request messages are supported Header Supported Header SupportFollowing SIP request headers are supported Header Supported Response Supported Response SupportFollowing SIP responses are supported 3xx Responses Redirection 5xx Responses Server Failure Third Party Call Control Hold ImplementationReliability of Provisional Responses TransferBridged 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 Encryption/ on page A-89 Changing the Key on the PhoneOption. This shows the digest field Encrypted and unencrypted file are the sameModel Width Height Color Depth Adding a Background LogoModel Associate Parameter RGB ValuesColor RGB Values Decimal Hexadecimal IP500 IP500 BitmapsIP300 IP300 IP330 IP330 IP400Migration Dependencies BootROM/SIP Application DependenciesAnimations Indicators Model BootROM SIP ApplicationMultiple Key Combinations IP 301 The two Line keys and the Up and Down arrow keys BootROM until the password prompt appearsIP 4000 and 6000 6, 8 and * dial pad keys About three secondsKey ID Function Default Feature Key LayoutsSoundStation IP 4000, 6000, and 7000 models SoundPoint IPSoundPoint 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 MappingPress Select Save ConfigDisabling PC Ethernet Port 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 Distribution Zip File New or Changed FeaturesElectronic Hookswitch Graphic Display Backgrounds Metrics for listening and conversational quality Configuration File ChangesBacklight Intensity 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.