Polycom SIP 3.1 Localization lcl, Server server, Emergency emergency, Server/ on page A-21 that

Page 175

 

 

 

 

 

 

Configuration Files

 

 

 

 

 

 

 

 

Server <server/>

 

 

 

 

 

This configuration attribute is defined as follows:

 

 

 

 

 

 

 

Attribute

 

Permitted Values

Default

Interpretation

 

 

 

 

 

 

dialplan.routing.server.x.

 

dotted-decimal IP address

Null

IP address or host name and port of

address

 

or host name

 

a SIP server that will be used for

 

 

 

 

routing calls. Multiple servers can

dialplan.routing.server.x.port

 

1 to 65535

5060

 

be listed starting with x=1, 2, ... for

 

 

 

 

fault tolerance.

 

 

 

 

 

 

 

 

Emergency <emergency/>

 

 

 

 

 

In the following attributes, x is the index of the emergency entry description

 

and y is the index of the server associated with emergency entry x. For each

 

emergency entry (index x), one or more server entries (indexes (x,y)) can be

 

configured. x and y must both use sequential numbering starting at 1.

 

 

 

 

 

 

 

Attribute

 

Permitted Values

Default

 

 

Interpretation

 

 

 

 

 

 

 

dialplan.routing.emergency.x.

 

Single entry representing

for x =1,

 

 

This determines the URLs

value

 

a SIP URL

value = “911”, Null

 

that should be watched for.

 

 

 

for all others

 

 

When one of these defined

 

 

 

 

 

 

URLs is detected as having

 

 

 

 

 

 

been dialed by the user, the

 

 

 

 

 

 

call will automatically be

 

 

 

 

 

 

directed to the defined

 

 

 

 

 

 

emergency server.

 

 

 

 

 

 

dialplan.routing.emergency.x.

 

positive integer

for x=1, y =1, Null

 

Index representing the

server.y

 

 

for all others

 

 

server defined in Server

 

 

 

 

 

 

<server/> on page A-21 that

 

 

 

 

 

 

will be used for emergency

 

 

 

 

 

 

routing.

 

 

 

 

 

 

 

Localization <lcl/>

The phone has a multilingual user interface. It supports both North American and international time and date formats. The call progress tones can also be customized. For more information, refer to Chord-Sets <chord/> on page A-29, and Call Progress Patterns on page A-33.

This attribute includes:

Multilingual <ml/>

Date and Time <datetime/>

A - 21

Image 175
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 Phone IP Address EM PowerName Possible Values Description Dhcp Client Dhcp MenuPossible Name Values Description MenuSection, Server Menu 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 ignoredSetting unless you want to disable the PC port SettingEthernet menu 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-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 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-directoryHandset, Headset, and Speakerphone Do Not Disturb116 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 encoding7000, 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 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 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 KeysThis element contains the following parameters This element describes behavior of enhanced feature keyEfk EfklistThis element describes the behavior of the user prompts EfkpromptSpecial Characters VersionVersion efk.version=2 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 ExampleContact 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 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-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 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 PayloadFollowing 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 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 locallyConfiguration 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 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 Ept = 325,326,327,328,329,330 This attribute also includesLcl.ml.lang.tags.x in Multilingual ml 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 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 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 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 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.filterByIpWeb 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 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 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 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 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 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 4000Function 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 regA-7 Is non-Null, all of the reg.x.server.y.xxxParameters will override the parameters Specified in sip.cfg in Server server/ onShared 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 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.cfgValue 0 if the call server is Microsoft Live Roaming Buddies roamingbuddiesCommunications Server 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 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 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 LogoColor RGB Values Decimal Hexadecimal RGB ValuesModel Associate Parameter 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 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.