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Polycom SIP 3.0.2 User's Manual

1. E E 9 E 15 on 2D 1 Ss 4 5 6 Ss za U Go 6 L_ 22 eg 7 7 6 ol dp L 3 y SO mn E ie Ae D eA Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 SoftKey1 12 DialpadStar 23 Dialpad9 34 n a 2 ArrowUp 13 SoftKey3 24 DialpadPound 35 n a 3 Menu 14 ArrowLeft 25 n a 36 n a 4 Conference 15 Dialpad2 26 Select 37 n a 5 Redial 16 Dialpad5 27 VolUp 38 n a 6 Handsfree 17 Dialpad8 28 VolDown 39 n a 7 SoftKey2 18 DialpadO 29 MicMute 40 n a 8 ArrowDown 19 SoftKey4 30 Release 41 n a 9 Dialpad1 20 ArrowRight 31 n a 42 n a 10 Dialpad4 21 Dialpad3 32 n a 11 Dialpad7 22 Dialpad6 33 n a 1 13 Administrator s Guide Addendum for the SoundStation IP 7000 Addendum to SIP 3 0 Administrator s Guide Note This addendum addresses changes to the SoundPoint IP SoundStation IP SIP 3 0 Administrator s Guide specific to the release of the SoundStation IP 6000 conference phone The SoundPoint IP 6000 conference phone is a next generation phone with a modern industrial design and the latest advancements in Polycom voice technology Some of the features include e Excellent wideband audio performance e Aconsole microphone with a range of up to ten feet e Expansion microphones to accommodate large room sizes The new or changed features include e Dis
2. Permitted Attribute Values Default Interpretation volpProt SIP alertInfo x value string to Null Alert Info fields from INVITE requests will be compare compared against as many of these against the parameters as are specified x 1 2 N value of and if a match is found the behavior Alert Info described in the corresponding ring class headers in refer to Ring type lt rt gt on page A 33 will INVITE be applied requests volpProt SIP alertInfo x class positive Null integer Request Validation lt requestValidation gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP requestValidation x req One of Null Sets the name of the method for which uest INVITE ACK validation will be applied BYE WARNING Intensive request validation REGISTER may have a negative performance impact CANCEL due to the additional signaling required in a some cases therefore use it wisely MESSAGE SUBSCRIBE NOTIFY REFER PRACK or UPDATE volpProt SIP requestValidation x me Null or Null If Null no validation is done Otherwise this thod one of source sets the type of validation performed for the digest or request both all source ensure request is received from an IP address of a server belonging to the set of target registration servers digest challen
3. O IP Address 1 String When the Boot Server parameter is set to Custom this parameter specifies the type of the DHCP option in which the phone will look for its boot server The IP Address must specify the boot server The String must match one of the formats described for Server Address in the following section Server Menu Setting up Your System Possible Name Values Description VLAN Discovery 0 Disabled No VLAN discovery through DHCP default 1 Fixed Use predefined DHCP vendor specific option values of 128 144 157 and 191 If this is used the VLAN ID Option field will be ignored 2 Custom Use the number specified in the VLAN ID Option field as the DHCP private option value VLAN ID Option 128 through 254 Cannot be the same as Boot Server Option default is 129 The DHCP private option value when VLAN Discovery is set to Custom For more information refer to Assigning a VLAN ID Using DHCP on page C 16 Note If multiple alternate DHCP servers respond e The phone should gather the responses from alternate DHCP servers e If configured for Custom Option6 6 the phone will select the first response that contains a valid custom option value e f none of the responses contain a custom option value the phone will select the first response that contains a valid option66 value Server Menu The following server configuration parameters can be modified
4. Attribute Default voice gain rx analog handset 0 voice gain rx analog headset 0 voice gain rx analog chassis 0 voice gain rx analog chassis IP_300 6 voice gain rx analog chassis IP_330 0 voice gain rx analog chassis IP_430 0 voice gain rx analog chassis IP_601 6 voice gain rx analog chassis IP_650 0 voice gain rx analog ringer 0 voice gain rx analog ringer IP_300 6 voice gain rx analog ringer IP_330 0 voice gain rx analog ringer IP_430 0 voice gain rx analog ringer IP_601 6 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice gain rx analog ringer IP_650 0 voice gain rx digital handset 15 voice gain rx digital headset 21 voice gain rx digital chassis 0 voice gain rx digital chassis IP_330 6 voice gain rx digital chassis IP_430 6 voice gain rx digital chassis IP_4000 0 voice gain rx digital chassis IP_601 0 voice gain rx digital chassis IP_650 6 voice gain rx digital ringer 21 voice gain rx digital ringer IP_330 12 voice gain rx digital ringer IP_430 12 voice gain rx digital ringer IP_4000 21 voice gain rx digital ringer IP_601 21 voice gain rx digital ringer IP_650 12 voice gain rx analog handset sidetone 20 voice gain rx analog headset sidetone 24 voice gain tx analog handset 6 voice gain tx analog headset 3 voice gain tx analog chassis 3 voice gain tx analog chassis IP_300 0 voice gain tx
5. Administrator s Guide Addendum for the SoundStation IP 7000 Treble Bass Controls The treble and bass controls equalize the tone of the high and low frequency sound from the speakers The SoundStation IP 7000 phone s treble and bass controls can be modified by the user through Menu gt Settings gt Basic gt Audio gt Treble EQ or Bass EQ There are no related configuration changes Voice Quality Monitoring Voice Quality Monitoring is not supported on the SoundStation IP 7000 conference phone at this time Daisy Chaining Phones You can join two SoundStation IP 7000 conference phones together through the use of a CLink cable and the Multi Interface Module The graphic display of each phone shows the same user interface and phone numbers The SoundStation IP 7000 phone that has the Ethernet connection is referred to as the primary The SoundStation IP 7000 phone that does not have the Ethernet connection is referred to as the secondary The primary secondary relationship of the phones is determined by their MAC address registration status and the configuration files Multi Interface Module 12 foot Ethernet Cable 25 foot lt Network Cable O Instructions for daisy chaining SoundStation IP 7000 conference phones are available in the SoundStation IP 7000 User Guide Administrator s Guide Addendum for the SoundStation IP 7000 Provisioning Phones Over CLink Normally the SoundSt
6. For more information refer to refer to Flash Parameter Configuration on page A 105 Configuring SoundPoint IP SoundStation IP Phones Locally A local phone based configuration web server is available unless it is disabled through sip cfg It can be used as the only method of modifying phone configuration or as a distributed method of augmenting a centralized provisioning model For more information refer to Web Server lt httpd gt on page A 57 Administrator s Guide SoundPoint IP SoundStation IP The phone s local user interface also permits many application settings to be modified such as SIP server address ring type or regional settings such as time date format and language Local Web Point your web browser to http lt phonelPAddress gt Server Access Configuration pages are accessible from the menu along the top banner The web server will issue an authentication challenge to all pages except for the home page Credentials are case sensitive User Name Polycom Password The administrator password is used for this Local Settings Some items in the Settings menu are locked to prevent accidental changes Menu Access To unlock these menus enter the user or administrator passwords The administrator password can be used anywhere that the user password is used Factory default passwords are User password 123 Administrator password 456 Passwords Administrator Network Config
7. Administrator s Guide Addendum for the SoundStation IP 6000 Gains lt gain gt The default gain settings have been carefully adjusted to comply with the TIA 810 A digital telephony standard Polycom recommends that you do not change these values POLYCOM Attribute Default voice gain rx analog chassis IP_6000 0 voice gain rx analog ringer IP_6000 0 voice gain rx digital chassis IP_6000 6 voice gain rx digital ringer IP_6000 21 voice gain tx analog chassis IP_6000 0 voice gain tx digital chassis IP_6000 6 Receive Equalization lt rxEq gt These settings control the performance of the receive equalization feature Polycom recommends that you do not change these values we POLYCOM Attribute Default voice rxEq hf IP_6000 preFilter enable 0 voice rxEq hf IP_6000 postFilter enable 0 Administrator s Guide Addendum for the SoundStation IP 6000 Transmit Equalization lt txEq gt These settings control the performance of the hands free transmit equalization feature Polycom recommends that you do not change these values POLYCOM Attribute Default voice txEq hf IP_6000 preFilter enable 0 voice txEq hf IP_6000 postFilter enable 0 Feature lt feature gt The call list feature cannot be disabled on the SoundStation IP 6000 phone Multiple Key Combinations and Default Key Layout The multiple key combinations on the SoundSta
8. Permitted Attribute Values Interpretation ind pattern x step y state On or Off Turn LED on or off for this step ind pattern x step y duration positive integer Duration in milliseconds for this step O infinite ind pattern x step y colour Red or Green default is Red if not specified For bi color LEDs specify color Classes lt class gt This section defines the available classes for the LED and graphical icon indicator types In the following table x is the class number y is the identifier of the state number for that class Attribute Permitted Values Interpretation ind class x state y index positive integer For LED type indicators index refers to the pattern index such as index x in the Patterns lt pattern gt tag above For Graphic Icon type indicators index refers to the animation index such as index y in the Animations lt anim gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt tag above Assignments This attribute assigns a type and a class to an indicator In the case of the Graphic Icon type it also assigns a physical location and size in pixels on the LCD display refer to the following section In the case of the LED type it assigns a physical LED number refer to Graphic Icons lt gi gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_40
9. Setting up Your System Note Your SoundPoint IP SoundStation IP SIP phone is designed to be used like a regular phone on a public switched telephone network PSTN This chapter provides basic instructions for setting up your SoundPoint IP SoundStation IP phones This chapter contains information on Setting Up the Network Setting Up the Boot Server e Deploying Phones From the Boot Server Upgrading SIP Application Because of the large number of optional installations and configurations that are available this chapter focuses on one particular way that the SIP application and the required external systems might initially be installed and configured in your network For more information on configuring your system refer to Configuring Your System on page 4 1 For more information on the configuration files required for setting up your system refer to Configuration Files on page A 1 For installation and maintenance of Polycom SoundPoint IP phones the use of a boot server is strongly recommended This allows for flexibility in installing upgrading maintaining and configuring the phone Configuration log and directory files are normally located on this server Allowing the phone write access to the server is encouraged The phone is designed such that if it cannot locate a boot server when it boots up it will operate with internally saved parameters This is useful for occasions when the boot server is not availab
10. PSTN Gateway a Ethernet 1 Switches Voice Bridge Local Local 40 100 Application Boot Server Ethernet Server Overview Session Initiation Protocol Application Architecture BootROM The software architecture of SIP application is made of 4 basic components BootROM loads first when the phone is powered on Application software that makes the device a phone Configuration configuration parameters stored in separate files Resource Files optional needed by some of the advanced features Configuration Resource Files The bootROM is a small application that resides in the flash memory on the phone All phones come from the factory with a bootROM pre loaded The bootROM performs the following tasks in order l 2 Performs a power on self test POST Optional Allows you to enter the setup menu where various network on provisioning options can be set The bootROM software controls the user interface when the setup menu is accessed Requests IP settings and accesses the boot server to look for any updates to the bootROM application If updates are found they are downloaded and saves to flash memory eventually overwriting itself after verifying the integrity of the download If a new bootROM is downloaded format the file system clearing out any application software or configuration files that may have been present Administrator s Guide SoundPoint IP
11. The SoundStation IP 7000 phone supports a single registration Administrator s Guide Addendum for the SoundStation IP 7000 When the phone is unable to register with the call control server the icon gt is shown outline Once the phone is able to register the icon is shown solid Audio Codecs The following table summarizes the SoundStation IP 7000 phone s new audio codec support Effective Sample audio Algorithm MIME Type Bit Rate Rate Frame Size bandwidth G 722 1C G7221 24 Kbps 32 Ksps 20ms 80ms 14 KHz 32000 32 Kbps 48 Kbps Lin16 L16 16000 25 6 Kbps 16 Ksps 10ms 7 KHz L16 32000 51 2 Kbps 32 Ksps 14 KHz L16 48000 76 8 Kbps 48 Ksps 22 KHz Siren14 SIREN14 24 Kbps 32 Ksps 20ms 80ms 14 KHz 16000 32 Kbps 48 Kbps Siren22 SIREN22 32 Kbps 32 Ksps 20ms 80ms 14 KHz 48000 48 Kbps 64 Kbps Note The network bandwidth necessary to send the encoded voice is typically 5 10 confirm number higher than the encoded bit rate due to packetization overhead For example a G 722 1C call at 48kbps consumes 5xkbps of network bandwidth Configuration changes can performed centrally at the boot server Central Configuration file Specify codec priority preferred payload sizes and jitter buffer tuning boot server sip cfg parameters For more information refer to Codec Preferences lt codecPref gt on page 1 8 and Codec Profiles lt audioProfile gt on page 1 9
12. e Multiple language support Set on screen language to your preference Select from Chinese Danish Dutch English French German Italian Japanese Korean Norwegian Portuguese Russian Spanish and Swedish Note In SIP 3 0 default support for Chinese Japanese and Korean was removed from the SoundPoint IP 600 and 601 Overview This chapter provides an overview of the Session Initiation Protocol SIP application and how the phones fit into the network configuration SIP is the Internet Engineering Task Force IETF standard for multimedia conferencing over IP It is an ASCII based application layer control protocol defined in RFC 3261 that can be used to establish maintain and terminate calls between two or more endpoints Like other voice over IP VoIP protocols SIP is designed to address the functions of signaling and session management within a packet telephony network Signaling allows call information to be carried across network boundaries Session management provides the ability to control the attributes of an end to end call For the SoundPoint IP SoundStation IP phones to successfully operate as a SIP endpoint in your network it must meet the following requirements e A working IP network is established e Routers are configured for VoIP e VoIP gateways are configured for SIP e The latest or compatible SoundPoint IP SoundStation IP phone SIP application image is available e Acall server is active a
13. e Using T in the left part of RRR syntax is not recommended For example ROTR322R should be avoided This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan digitmap string compatible with the 2 9 11 OT When this attribute is digit map feature of 011Xxx T present number only dialing MGCP described in 2 1 5 during the setup phase of O 2 9 xxxxxxxxx of RFC 3435 String is new calls will be compared limited to 768 bytes and 1 2 9 xxxxxxxx against the patterns therein 30 segments a comma is 2 9 xxxxxxxxx and if a match is found the also allowed when 2 9 xxxT call will be initiated reached in the digit map automatically eliminating the a comma will turn dial need to press Send tone back on is allowed Attributes as a valid digit extension dialplan applyToCallLis letter R is used as tDial defined above dialplan applyToDirecto ryDial dialplan applyToUserDia 1 and dialplan applyToUserSen d control the use of match and replace in the dialed number in the different scenarios dialplan digitmap timeOut string of positive integers 3 3 3 3 3 3 Timeout in seconds for each separated by segment of digit map Note If there are more digit maps than timeout values the default value of 3 will be used If there are more timeout values than digit maps the extra timeout values are ignored Routing lt routing gt
14. fnt file or Uxx00_UyyFF fon file For more information refer to Multilingual User Interface on page 4 29 e lt fontExtension gt describes the file type Either fnt for single 256 characters font or fon for multiple fnt files If it is necessary to overwrite an existing font use these lt fontName gt _ lt fontHeightInPixels gt SoundPoint IP 320 330 430 500 and 501 fontProp_10 This is the font used widely in the current implementation fontPropSoftkey_10 This is the soft key specific font SoundPoint IP 550 600 601 and 650 fontProp_ 19 fontProp_26 fontProp_x This is the font used widely in the current implementation including for soft keys This is the font used to display time but not date This is a small font used for the CPU Load Net utilization graphs this is the same as the fontProp_10 for the SoundPoint IP 500 If the lt fontName gt _ lt fontHeightInPixels gt does not match any of the names above then the downloaded font will be applied against all fonts defined in the phone which means that you may lose the benefit of fonts being calibrated differently depending on their usage For example the font used to display the Configuration Files time on the SoundPoint IP 600 is a large font larger than the one used to display the date and if you overwrite this default font with a unique font you lose this size aspect For example e to ove
15. lt ad gt 0 lt ad gt lt ar gt 0 lt ar gt lt bw gt 0 lt bw gt lt bb gt 0 lt bb gt lt item gt lt item gt lt ln gt Smith lt 1n gt lt fn gt Bill lt fn gt lt ct gt 1003 lt ct gt lt sd gt 3 lt sd gt lt rt gt 3 lt rt gt lt dc gt lt ad gt 0 lt ad gt lt ar gt 0 lt ar gt lt bw gt 0 lt bw gt lt bb gt 0 lt bb gt lt item gt lt item_list gt lt directory gt Element Permitted Values Interpretation fn UTF 8 encoded string first name of up to 40 bytes Note In some cases this will be less than 40 characters due to UTF 8 s variable length encoding In UTF 8 encoded string last name of up to 40 bytes ct UTF 8 encoded string contact containing digits the Used by the phone to address a remote party in the same way that a user part of a SIP string of digits or a SIP URL are dialed manually by the user This URL or a string that element is also used to associate incoming callers with a particular constitutes a valid SIP directory entry URL ee Note This field cannot be null or duplicated 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 the maximum speed dial index is 99 Administrator s Guide SoundPoint IP SoundStation IP Element Permitted Values Interpretation r
16. 0 eee eee 4 36 Multiple Registrations 000 e eee eee eee 4 37 Automatic Call Distribution 0 0 002 eee eee 4 38 Server Redundancy 0 6 c cece cece eens 4 39 Presne 25 Soews dak eee eee eee eee pE ENEE bt ee ee ee hee 4 43 Microsoft Live Communications Server 2005 Integration 4 43 Setting Up Audio Features 00 0 0222 4 47 Low Delay Audio Packet Transmission 00004 4 47 Jitter Buffer and Packet Error Concealment 0 4 48 Voice Activity Detection 0000 0 eects 4 48 DTMF Tone Generation 6 0 0 4 49 DTMF Event RTP Payload 0c cece eee eee 4 49 Acoustic Echo Cancellation 0 0 4 49 Audio Codecs siriene a e E E baebite cheese 4 50 Background Noise Suppression 0 00 eee eee 4 51 Comfort Noise Fill 0 ees 4 51 Automatic Gain Control 0 0 0 4 51 IP Type Of Service 4c ine icwhaens Yee ENNE ERE eek aha eee 4 51 TREE 8021p O erruis nenien kEi Nita pone etal i 4 52 Voice Quality Monitoring 0 00 eee eee eee 4 52 Setting Up Security Features 0 0 0 cece eee 4 53 Local User and Administrator Privilege Levels 4 53 Custom Certificates sscersssinth dose tease a hee aiediresle Mere a tad nanan ae 4 54 Incoming Signaling Validation 6 c cece eee eee eee 4 54 Configuration File Encryption 0 0c ee
17. Administrator s Guide for the SoundPoint IP SoundStation IP Family SIP 3 0 2 April 2008 Edition 1725 1 1530 300 Rev A2 ha POLYCOM Trademark Information Polycom the Polycom logo design SoundPoint IP SoundStation SoundStation VTX 1000 ViaVideo ViewStation and Vortex are registered trademarks of Polycom Inc Conference Composer Global Management System ImageShare Instructor RP iPower MGC PathNavigator People Content PowerCam Pro Motion QSX ReadiManager Siren StereoSurround V2IU Visual Concert VS4000 VSX and the industrial design of SoundStation are trademarks of Polycom Inc in the United States and various other countries All other trademarks are the property of their respective owners Patent Information The accompanying product is protected by one or more U S and foreign patents and or pending patent applications held by Polycom Inc Disclaimer Some countries states or provinces do not allow the exclusion or limitation of implied warranties or the limitation of incidental or consequential damages for certain products supplied to consumers or the limitation of liability for personal injury so the above limitations and exclusions may be limited in their application to you When the implied warranties are not allowed to be excluded in their entirety they will be limited to the duration of the applicable written warranty This
18. For more information refer to Microbrowser lt mb gt on page A 86 Local Web Server if enabled Specify the Applications browser home page and proxy to use Navigate to http lt phonelPAddress gt coreConf htm mb Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Real Time Transport Protocol Ports The phone is compatible with RFC 1889 RTP A Transport Protocol for Real Time Applications and the updated RFCs 3550 and 3551 Consistent with RFC 1889 the phone treats all RTP streams as bi directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports This allows real time transport control protocol RTCP to operate correctly even with RTP media flowing in only a single direction or not at all It also allows greater security packets from unauthorized sources can be rejected The phone can filter incoming RTP packets arriving on a particular port by IP address Packets arriving from a non negotiated IP address can be discarded The phone can also enforce symmetric port operation for RTP packets packets arriving with the source port set to other than the negotiated remote sink port can be rejected
19. Session Initiation Protocol SIP Basic Call Flow Examples draft ietf sip cc transfer 05 txt SIP Call Control Transfer RFC 3725 Best Current Practices for Third Party Call Control 3pec in the Session Initiation Protocol SIP RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol SIP RFC 3856 A Presence Event Package for Session Initiation Protocol SIP RFC 3891 The Session Initiation Protocol SIP Replaces Header RFC 3892 The Session Initiation Protocol SIP Referred By Mechanism RFC 3959 The Early Session Disposition Type for the Session Initiation Protocol SIP RFC 3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol SIP RFC 3968 The Internet Assigned Number Authority IANA Header Field Parameter Registry for the Session Initiation Protocol SIP Request Support Session Initiation Protocol SIP e RFC 3969 The Internet Assigned Number Authority IANA Uniform Resource Identifier URI Parameter Registry for the Session Initiation Protocol SIP e RFC 4028 Session Timers in the Session Initiation Protocol SIP e RFC 4235 An INVITE Initiated Dialog Event Package for the Session Initiation Protocol SIP e RFC 4662 Session Initiation Protocol SIP Event Notification Extension for Resource Lists e draft levy sip diversion 04 txt Diversion Indication in SIP e draft anil sippin
20. SoundStation IP Each event in the log contains the following fields separated by the character e time or time date stamp e 1 5 character component identifier such as so e event class e cumulative log events missed due to excessive CPU load e free form text the event description Example 011511 006 so 2 00 soCoreAudioTermChg chassis gt idle time stamp f ID event class missed events text Three formats are available for the event timestamp Type Example 0 seconds milliseconds 011511 006 1 hour 15 minutes 11 006 seconds since booting 1 absolute time with minute resolution 0210281716 2002 October 28 17 16 2 absolute time with seconds resolution 1028171642 October 28 17 16 42 Two types of logging are supported e Basic Logging lt level gt lt change gt and lt render gt e Scheduled Logging Parameters lt sched gt A 76 Configuration Files Basic Logging lt level gt lt change gt and lt render gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation log level change xxx 0 5 4 Control the logging detail level for individual components These are the input filters into the internal memory based log system Possible values for xxx are so app1 sip sspsc ssps pps net cfg cdp pmt ftp ares dns cxss httpd rdisk copy slog res key log curl rtos mb ib sotet
21. Specifies how to search the contact directory If set to 1 search by contact s first name If set to 0 search by contact s last name Corporate Directory lt corp gt A portion of the corporate directory is stored in flash memory on the phone The size is based on the amount of flash memory in the phone Different phone models have variable flash memory This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation dir corp address dotted decimal Null The IP address or host name of the IP address or LDAP server interface to the host name corporate directory dir corp port 0 Null 1 to 389 TCP This parameter is used to specify 65535 636 TLS the port to connect to on the server if a full URL is not provided dir corp transport TCP TLS Null TCP This parameter is used to specify whether a TCP or TLS connection is made with the server if a full URL is not provided dir corp baseDN UTF 8 encoded Null The base domain name is the string starting point for making queries on the LDAP server dir corp user UTF 8 encoded Null The username used to authenticate string to the LDAP server Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation dir corp password UTF 8 encoded Null The password used to authenticate string to the LDAP server dir corp filterPrefix UT
22. The phone can also jam the destination transport port to a specified value regardless of the negotiated port This can be useful for punching through firewalls When this is enabled all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well Incoming packets are sorted by the source IP address and port allowing multiple RTP streams to be multiplexed The RTP port range used by the phone can be specified Since conferencing and multiple RTP streams are supported several ports can be used concurrently Consistent with RFC 1889 the next higher odd port is used to send and receive RTCP Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify whether to filter incoming RTP packets by IP address boot server sip cfg whether to require symmetric port usage whether to jam the destination port and specify the local RTP port range start For more information refer to RTP lt rtp gt on page A 51 Local Web Server Specify whether to filter incoming RTP packets by IP address if enabled whether to require symmetric port usage whether to jam the destination port and specify the local RTP port range start Navigate to http lt phonelPAddress gt netConf htm rt Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Change
23. To replace the animation used for the idle display For more information refer to Animations lt anim gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt on page A 72 To change the position of the idle display animation For more information refer to Graphic Icons lt gi gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt on page A 74 Ethernet Switch The SoundPoint IP phones contain two Ethernet ports labeled LAN and PC and an embedded Ethernet switch that runs at full line rate The SoundStation IP phones conatin only one Ethernet port labeled LAN The Ethernet switch allows a personal computer and other Ethernet devices to connect to the office LAN by daisy chaining through the phone eliminating the need for a stand alone hub The SoundPoint IP switch gives higher transmit priority to packets originating in the phone The phone can be powered through a local Administrator s Guide SoundPoint IP SoundStation IP AC power adapter or can be line powered power supplied through the signaling or idle pairs of the LAN Ethernet cable Line powering typically requires that the phone plugs directly into a dedicated LAN jack Devices that do not require LAN power can then plug into the SoundPoint IP PC Ethernet port SoundPoint IP Switch Port Priorities To help ensure good voice quality the Ethernet switch
24. 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Permitted Attribute Values Default Interpretation call missedCallTracking x enabled 0 1 1 If set to 1 or Null missed call tracking is enabled If call missedCallTracking x enabled is set to 0 then missedCall counter is not updated regardless of what call serverMissedCalls x enabled is set to and regardless of how the server is configured There is no Missed Call List provided under Menu gt Features of the phone If call missedCallTracking x enabled is set to 1 and call serverMissedCalls x enabled is set to 0 then the number of missedCall counter is incremented regardless of how the server is configured If call missedCallTracking x enabledis set to 1 and call serverMissedCalls x enabled is set to 1 then the handling of missedCalls depends on how the server is configured Call Waiting lt callWaiting gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation call callWaiting ring beep ring beep Specifies the ring tone heard on an incoming silent call when another call is active If set to Null the default value is beep Administrator s Guide SoundPoint IP SoundStation IP Diversion lt divert gt The phone has a flexible call forward diversion feature for each registration In all cases a call will only be diverted if a non Null con
25. Increasing this value may have a detrimental effect on performance of the phone Configuration Files Per Phone Configuration wy POLYCOM Registration lt reg gt This section covers the parameters in the per phone example configuration file phonel cfg This file would normally be used as a template for the per phone configuration files For more information refer to Deploying Phones From the Boot Server on page 3 15 Polycom recommends that you create another file with your organization s modifications If you must change any Polycom templates back them up first For more information refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice The parameters include e Registration lt reg gt e Calls lt call gt e Diversion lt divert gt e Dial Plan lt dialplan gt e Messaging lt msg gt e Network Address Translation lt nat gt e Attendant lt attendant gt e Roaming Buddies lt roaming_buddies gt e Roaming Privacy lt roaming_privacy gt e User Preferences lt user_preferences gt SoundPoint IP 301 320 330 and 430 support a maximum of two unique registrations SoundPoint IP 501 supports three the SoundPoint IP 550 and 560 supports four and SoundPoint IP 600 601 and 650 support six Up to three SoundPoint IP Expansion Modules can be added to a single host SoundPoint IP 601 and 650 phone increasing the total num
26. The names OpenSSL Toolkit and OpenSSL Project must not be used to endorse or promote products derived from this software without prior written permission For written permission please contact openssl core openssl org 5 Products derived from this software may not be called OpenSSL nor may OpenSSL appear in their names without prior written permission of the OpenSSL Project 6 Redistributions of any form whatsoever must retain the following acknowledgment This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit http www openssl org THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT AS IS AND ANY EXPRESSED OR IMPLIED WARRANTIES INCLUDING BUT NOT LIMITED TO THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED IN NO EVENT SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT INDIRECT INCIDENTAL SPECIAL EXEMPLARY OR CONSEQUENTIAL DAMAGES INCLUDING BUT NOT LIMITED TO PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES LOSS OF USE DATA OR PROFITS OR BUSINESS INTERRUPTION HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY WHETHER IN CONTRACT STRICT LIABILITY OR TORT INCLUDING NEGLIGENCE OR OTHERWISE ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE This product includes cryptographic software written by Eric Young eay cryptsoft com This product includes software written by Tim
27. This attribute allows the user to create a specific routing path for outgoing SIP calls independent of other default configurations This attribute also includes e Server lt server gt e Emergency lt emergency gt Server lt server gt This configuration attribute is defined as follows Configuration Files Attribute Permitted Values Default Interpretation dialplan routing server x addre dotted decimal IP address Null IP address or host name and port of ss 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 lt emergency gt 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 value dialplan routing emergency x Single entry representing a SIP URL for x 1 value 911 Null for all others This determines the URLs that should be watched for 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 server y
28. U 303F Hiragana U 3040 U 309F Katakana U 30A0 U 30FF Bopomofo U 3100 U 312F Hangul Compatibility Jamo U 3130 U 318F Bopomofo Extended U 31A0 U 31BF Enclosed CJK Letters and Months U 3200 U 327F CJK Compatibility U 3300 U 33FF CJK Unified Ideographs U 4E00 U 9FFF Hangul Syllables U AC00 U D7A3 CJK Compatibility Ideographs U F900 U FAFF CJK Half width forms U FF00 U FFFF Note Within a Unicode range some characters may not be supported due to their infrequent usage A 22 Configuration Files Date and Time lt datetime gt This configuration attribute is defined as follows Permitted Attribute Values Interpretation Icl datetime time 24HourClock 0 1 If 1 display time in 24 hour clock mode rather than a m p m Icl datetime date format string which Controls format of date string includes D d and M and two optional commas D day of week d day M month Up to two commas may be included For example D dM Thursday 3 July or Md D July 3 Thursday The field may contain 0 1 or 2 commas which can occur only between characters and only one at a time For example D dM is illegal Icl datetime date longFormat 0 1 If 1 display the day and month in long format Friday November otherwise use abbreviations Fri Nov Icl datetime date dateTop 0 1 If 1 display da
29. attributes must be sequential 1cl ml lang menu 1 lcl ml lang menu 2 lcl ml lang menu 3 lcl ml lang menu N with no gaps and the strings must exactly match a folder name under the SoundPointlPLocalization folder on the boot server for the phone to be able to locate the dictionary file Icl ml lang clock x 24HourClock 0 1 If attribute present overrides lcl datetime time 24HourClock If 1 display time in 24 hour clock mode rather than am pm Icl ml lang clock x format string which includes D d and M and two optional commas If attribute present overrides Icl datetime date format D day of week d day M month Up to two commas may be included For example D dM Thursday 3 July or Md D July 3 Thursday The field may contain 0 1 or 2 commas which can occur only between characters and only one at a time For example D dM is illegal Configuration Files Attribute Permitted Values Interpretation Icl ml lang clock x longFormat 0 1 If attribute present overrides lcl datetime date longFormat If 1 display the day and month in long format Friday November otherwise use abbreviations Fri Nov Icl ml lang clock x dateTop 0 1 If attribute present overrides lcl datetime date dateTop If 1 display date above time otherwise display time above date Icl ml lang y list All or a A list of the languages supporte
30. check if the telephone is correctly registered to the server Press the Menu key followed by System Status and Network Statistics Scroll down to see if LAN port shows active or Inactive Check the termination at the switch or hub end of the network LAN cable Ensure that the switch hub port connected to the telephone is operational if not accessible contact your system administrator Before restarting your phone contact your system administrator since this may allow more detailed troubleshooting to occur before losing any current status information 5 10 Troubleshooting Your SoundPoint IP SoundStation IP Phones Access to Screens and Systems Symptom Problem Corrective Action There is no response from feature key presses The SoundPoint IP SoundStation IP family SIP phone is not in active state Do one of the following Press the keys more slowly Check to see whether or not the key has been mapped to a different function or disabled Make a call to the phone to check for inbound call display and ringing as normal If successful try to press feature keys within the call to access Directory or Buddy Status for example Press Menu followed by Status gt Lines to confirm line is actively registered to the call server Reboot the phone to attempt re to the call server refer to Rebooting the Phone on page C 9 The display shows Network Link is Down The LA
31. dialplan routing emergency x positive integer for x 1 y 1 Null for all others Index representing the server defined in Server lt server gt on page A 19 that will be used for emergency routing Localization lt lcl gt 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 lt chord gt on page A 26 and Call Progress Patterns on page A 30 This attribute includes e Multilingual lt ml gt e Date and Time lt datetime gt A 19 Administrator s Guide SoundPoint IP SoundStation IP Multilingual lt ml gt The multilingual feature is based on string dictionary files downloaded from the boot server These files are encoded in standalone XML format Several western European and Asian languages are included with the distribution An exact match for one of the folder names under the SoundPointlPLocalizat ion folder on the boot server Attribute Permitted Values Interpretation Icl ml lang Null If Null the default internal language US OR English will be used otherwise the language to be used may be specified in the format language region Icl ml lang menu x String in the format language_region Multiple 1cl ml lang menu x attributes are supported as many languages as are desired However the 1cl ml lang menu x
32. maintained for each distinct call in progress Call Transfer Call transfer allows the user to transfer a call in progress to some other destination Call Waiting When an incoming call arrives while the user is active on another call the incoming call is presented to the user visually on the display and a configurable sound effect will be mixed with the active call audio Called Party Identification The phone displays and logs the identity of the party specified for outgoing calls Calling Party Identification The phone displays the caller identity derived from the network signalling when an incoming call is presented if information is provided by the call server Overview Connected Party Identification The identity of the party to which the user has connected is displayed and logged if the name is provided by the call server Context Sensitive Volume Control The volume of user interface sound effects such as the ringer and the receive volume of call audio is adjustable Customizable Audio Sound Effects Audio sound effects used for incoming call alerting and other indications are customizable Directed Call Pick Up and Group Call Pick Up Calls to another phone can be picked up by dialing the extension of the other phone Calls to another phone within a pre defined group can be picked up without dialing the extension of the other phone Distinctive Call Waiting Calls can be mapped to distinc
33. on page A 100 that will be used for emergency routing Messaging lt msg gt Message waiting indication is supported on a per registration basis This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation msg bypassinstantMessage 0 1 0 If set to 1 the display offering a choice of Message Center and Instant Messages will be bypassed when pressing the Messages key The phone will act as if Message Center was chosen Refer to Voice Mail Integration on page 4 36 Instant Messages will still be accessible from the Main Menu This attribute also includes e Message Waiting Indicator lt mwi gt A 101 Administrator s Guide SoundPoint IP SoundStation IP Message Waiting Indicator lt mwi gt In the following table x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation msg mwi x subscribe ASCII encoded string containing Null If non Null the phone will send digits the user part of a SIP a SUBSCRIBE request to this URL or a string that constitutes contact after boot up a valid SIP URL 6416 or 6416 polycom com msg mwi x callBackMo contact or registration Configures message retrieval de registration or and
34. 0 0 outgoing hold signaling The alternative is RFC 3264 a sendonly or a inactive Call Transfer Call transfer enables the user party A to move an existing call party B into anew call between party B and another user party C selected by party A The phone offers three types of transfers Blind transfers The call is transferred immediately to party C after party A has finished dialing party C s number Party A does not hear ring back e Attended transfers Party A dials party C s number and hears ring back and decides to complete the transfer before party C answers This option can be disabled e Consultative transfers Party A dials party C s number and talks privately with party C after the call is answered and then completes the transfer or hangs up Configuration changes can performed centrally at the boot server Central boot server Configuration file Specify whether to allow a transfer during the proceeding state of a sip cfg consultation call For more information refer to SIP lt SIP gt on page A 10 Specify whether a transfer is blind or not For more information refer to Call Handling Configuration lt call gt on page A 58 Configuring Your System Local Centralized Conferencing The phone can conference together the local user with the remote parties of a configurable number of independent calls by using the phone s local audio processing
35. 1 e Locate the volpProt parameter Set the voIpProt server x transport attribute to TCPpreferred or TLS Your selection depends on the LCS configuration Note The TLS protocol is not supported on SoundPoint IP 300 and 500 phones f Set the volpProt server x address to the LCS address For example volpProt server l address lcs2005 local g Set the volpProt SIP 1cs attribute to 1 h Optional If SIP forking is desired set volpProt SIP ms forking attribute to 1 Refer to SIP lt SIP gt on page A 10 Save the modified sip cfg configuration file 2 Modify the phonel cfg configuration file as follows a Open phonel cfg in an XML editor Locate the registration parameter Set the reg 1 address to the LCS address For example reg 1 address 7778 Set the reg 1 server y address to the LCS server name Optional Set the reg 1 server y transport attribute to TCPpreferred or TLS Your selection depends on the LCS configuration Set reg 1 auth userId to the phone s LCS username For example reg 1 auth userId jbloggs Set reg 1 auth password to the LCS password For example reg 1 auth password Password2 Locate the roaming_buddies attribute Set the roaming_buddies reg element to 1 Refer to Roaming Buddies lt roaming_buddies gt on page A 104 Locate the roaming_privacy attribute Set the roaming_privacy reg element to 1 Refer to Roaming Privacy lt roaming_privacy gt on page A 104 Administrat
36. 16 Localization lt Icl gt 2 0 0 0 E eens A 19 User Preferences lt up gt 0 00 cee eee teen eens A 23 Tones lt toneS gt 2 0 ccc ccc een ene t een e eens A 24 Sampled Audio for Sound Effects lt saf gt 00 A 27 Sound Effects lt se gt 2 0 ccc cece cnet ene A 28 Voice Settings lt voice gt 0 eee eee eee eee A 34 Quality of Service lt QOS gt 0 000 A 50 Basic TCP IP lt TCP_IP gt 1 0 0 0 eee A 52 Web Server lt httpd gt 00 eee eee eee A 57 Call Handling Configuration lt call gt 00 e eee A 58 Directory lt i 455204 wnegeei eres he HO Awd ee Tenana A 61 Presence lt pres gt i c cieectohe stipends ees tbees tebe seeded A 65 Fonts lt font gt 2 cae de pede EAEEREN E pha ee suave E eae A 66 Keys key gt are rales Ge Se thee Cee eben ote de desea A 69 Backgrounds lt Do gt vcs cccctins cern cuatine veins eiei eiieeii s A 70 Bitmaps lt bitmap gt 0 0 cece eee eee A 71 Indicators ind gt ere eerren ob ket beak a e A 71 Event Logging lt log gt 0 A 75 Security lt se0 gt rs egos Sure eS 8 doe See bag es AOTEA EE aa sews A 79 License lt license gt iseer reretia reer bea EMERGE Sk eee thee oH A 80 Provisioning lt prov gt 0 06 A 81 RAM Disk lt ramdisk gt 1 0 0 00 cece eens A 81 Request lt request gt 0 0 t neute eee ee eect EESTE ees A 82 Contents Feature lt feature
37. 5 to 20 20 The time interval between successive periodic quality reports Alert Reports lt alert gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation voice qualityMonitoring collector alert moslq threshold warning Null 15 to 40 Null Threshold value of listening MOS score MOS LQ that causes phone to send a warning alert quality report Configure the desired MOS value multiplied by 10 If set to Null warning alerts are not generated due to MOS LQ For example a configured value of 35 corresponds to the MOS score 3 5 voice qualityMonitoring collector alert moslq threshold critical Null 15 to 40 Null Threshold value of listening MOS score MOS LQ that causes phone to send a critical alert quality report Configure the desired MOS value multiplied by 10 If set to Null critical alerts are not generated due to MOS LQ For example a configured value of 28 corresponds to the MOS score 2 8 Configuration Files Permitted Attribute Values Default Interpretation voice qualityMonitoring collector alert dela Null 10 to Null Threshold value of one way delay y threshold warning 2000 in ms that causes phone to senda critical alert quality report If set to Null warning alerts are not generated due to one way delay One way delay includes both network delay and end system delay voi
38. 550 560 max 6 on IP 600 max 48 on IP 601 650 without any Expansion Modules attached only 6 line keys are available The number of line keys on the phone to be associated with registration x Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Default Interpretation reg x callsPerLineKey 1 to 24 OR 1to8 24 OR 8 For the SoundPoint IP 600 601 and 650 the permitted range is 1 to 24 and the default is 24 For all other phones the permitted range is 1 to 8 and the default is 8 This is the number of calls or conferences which may be active or on hold per line key associated with this registration Note that this overrides call callsPerLineKey for this registration Refer to Call Handling Configuration lt call gt on page A 58 reg x bargelnEnabled 0 1 Null Allow remote user of SCA to interrupt call Works in a similar way to resume If set to 1 barge in is enabled for line x If set to 0 or Null barge in is disabled for line x reg x outboundProxy address dotted decimal IP address or host name Null reg x outboundProxy port 1 to 65535 5060 IP address or host name and port of a SIP server to which the phone shall send all requests reg x outboundProxy transport DNSnaptr or TCPpreferred or UDPOnly or TLS or TCPOnly DNSnap tr If set to Null or DNSnaptr If reg x outboundProxy address is
39. A 101 encryption lt encryption gt A 80 Ethernet IEEE 802 1p Q A 50 Ethernet menu 3 11 F feature lt feature gt A 83 features list of 1 5 finder lt finder gt A 85 flash parameter configuration A 105 Index flash parameter See also device fonts lt font gt A 66 forward all lt fwd gt A 96 G gains lt gain gt A 37 graphic display backgrounds 4 16 A 70 graphic icons lt gi gt A 74 group call pick up 4 21 H handset headset and speakerphone 4 9 hands free disabled A 24 hold lt hold gt A 61 I idle display lt idleDisplay gt A 86 idle display animation 4 15 incoming signaling validation 4 54 indicator classes lt class gt A 73 indicators A 71 assignments A 73 installing SIP application 3 15 instant messaging 4 28 IP TOS A 51 IP TOS call control lt callControl gt A 52 IP_400 font A 68 IP_500 font A 68 IP_600 font A 68 J jitter buffer 4 48 K keep alive lt keepalive gt A 57 key features 1 5 keys lt key gt A 69 L language support 1 6 4 29 languages adding new A 21 last call return 4 22 LEDs A 74 length lt length gt A 80 local centralized conferencing 4 19 local lt local gt A 6 local contact directory 4 9 Index 3 Administrator s Guide SoundPoint IP SoundStation IP local contact directory file format 4 10 local digit map 4 12 local reminder lt localReminder gt A 61 local user and administrator privilege levels 4 53 localization lt Icl gt
40. ARISING FROM OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE 2008 Polycom Inc All rights reserved Polycom Inc 4750 Willow Road Pleasanton CA 94588 2708 USA No part of this document may be reproduced or transmitted in any form or by any means electronic or mechanical for any purpose without the express written permission of Polycom Inc Under the law reproducing includes translating into another language or format As between the parties Polycom Inc retains title to and ownership of all proprietary rights with respect to the software contained within its products The software is protected by United States copyright laws and international treaty provision Therefore you must treat the software like any other copyrighted material e g a book or sound recording Every effort has been made to ensure that the information in this manual is accurate Polycom Inc is not responsible for printing or clerical errors Information in this document is subject to change without notice About This Guide The Administrator s Guide for the SoundPoint IP SoundStation IP family is for administrators who need to configure customize manage and troubleshoot SoundPoint IP SoundStation IP phone systems This guide covers the SoundPoint IP 301 320 330 430 501 550 560 600 601 and 650 desktop phones and the SoundStation IP 4000 conference phone The following related documents for Sou
41. CA American Express Global CA BelSign Object Publishing CA BelSign Secure Server CA Deutsche Telekom AG Root CA Digital Signature Trust Co Global CA 1 Digital Signature Trust Co Global CA 2 Administrator s Guide SoundPoint IP SoundStation IP e Digital Signature Trust Co Global CA 3 e Digital Signature Trust Co Global CA 4 e Entrust Worldwide by DST Entrust net Premium 2048 Secure Server CA e Entrust net Secure Personal CA Entrust net Secure Server CA e Equifax Premium CA e Equifax Secure CA e GTE CyberTrust Global Root e GTE CyberTrust Japan Root CA e GTE CyberTrust Japan Secure Server CA e GTE CyberTrust Root 2 e GTE CyberTrust Root 3 e GTE CyberTrust Root 4 e GTE CyberTrust Root 5 e GTE CyberTrust Root CA e GlobalSign Partners CA e GlobalSign Primary Class 1 CA e GlobalSign Primary Class 2 CA e GlobalSign Primary Class 3 CA e GlobalSign Root CA e National Retail Federation by DST TC TrustCenter Germany Class 1 CA TC TrustCenter Germany Class 2 CA TC TrustCenter Germany Class 3 CA TC TrustCenter Germany Class 4 CA Thawte Personal Basic CA e Thawte Personal Freemail CA e Thawte Personal Premium CA e Thawte Premium Server CA Miscellaneous Administrative Tasks e Thawte Server CA e Thawte Universal CA Root e UPS Document Exchange by DST e ValiCert Class 1 VA e ValiCert Class 2 VA e ValiCert Class 3 VA e VeriSign Class 4 Primary CA e Verisign Class 1 P
42. Changes on page 1 5 The default SIP key layout for the SoundStation IP 7000 conference phone is shown in Multiple Key Combinations and Default Key Layout on page 1 12 For more information refer to the Release Notes for the SIP Application Version 3 0 2 For more information on the SoundStation IP 7000 conference phone refer to the User Guide at http Avww polycom com support voip New or Changed Features Distribution Zip File As well as the sip Id file in the archive there is a separate file for the SoundStation IP 7000 conference phone called 3111 40000 001 sip 1d Managing Conferences Configuration changes can be 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 lt feature gt on page 1 12 Configurable Feature Keys No feature keys on the SoundStation IP 7000 can be remapped Multilingual User Interface Support for major western European languages is included and additional languages can be easily added Support for Asian languages Chinese Japanese and Korean is also included and will display on the SoundStation IP 7000 s higher resolution display A WGL4 character set is displayed the SoundStation IP 7000 For more information refer to http www microsoft com OpenType otspec WGL4E HTM Downloadable fonts are not supported on the SoundStation IP 7000 Single Registration
43. Each phone may open multiple connections to the server The phone will attempt to upload log files a configuration override file and a directory file to the server This requires that the phone s account has 3 13 Administrator s Guide SoundPoint IP SoundStation IP Note we POLYCOM delete write and read permissions The phone will still function without these permissions but will not be able to upload files The files downloaded from the server by the phone should be made read only Typically all phones are configured with the same server account but the server account provides a means of conveniently partitioning the configuration Give each account an unique home directory on the server and change the configuration on an account by account basis 3 Copy all files from the distribution zip file to the phone home directory Maintain the same folder hierarchy The distribution zip file contains sip ld including a separate one for every supported model sip cfg phonel cfg 000000000000 cfg 000000000000 directory xml SoundPointIP dictionary xml one of each supported language SoundPointIPWelcome wav Refer to the Release Notes for a detailed description of each file in the distribution Boot Server Security Policy You must decide on a boot server security policy Polycom recommends allowing file uploads to the boot server where the security environment permits This allows event
44. IP gt Ethernet IEEE 802 1p Q lt ethernet gt The following settings control the 802 1p Q user_priority field e RTP lt RTP gt e Call Control lt callControl gt e Other lt other gt RTP lt RTP gt These parameters apply to RTP packets Permitted Attribute Values Default Interpretation qos ethernet rtp user_priority 0 7 5 User priority used for RTP packets Call Control lt callControl gt These parameters apply to call control packets such as the network protocol signaling Permitted Attribute Values Default Interpretation qos ethernet callControl user_priority 0 7 5 User priority used for call control packets Other lt other gt These default parameter values are used for all packets which are not set explicitly Permitted Attribute Values Default Interpretation qos ethernet other user_priority 0 7 2 User priority used for packets that do not have a per protocol setting A 50 Configuration Files IP TOS lt IP gt The following settings control the type of service field in outgoing packets e RTP lt rtp gt e Call Control lt callControl gt RTP lt rtp gt These parameters apply to RTP packets Permitted Attribute Values Default Interpretation qos ip rtp dscp 0 to 63 or Null This parameter allows the DSCP of EF or packets to be specified If set to a any of value this will override th
45. If set to private use standard call signaling If set to shared augment call signaling with call state subscriptions and notifications and use access control for outgoing calls reg x thirdPartyName string in the same Null This field must match the reg x address format as value of the other registration which makes reg x address up the bridged line appearance BLA It must be Null in all other cases reg x auth userld string Null User ID to be used for authentication challenges for this registration If non Null will override the Reg User x parameter entered into the Authentication submenu off of the Settings menu on the phone reg x auth password string Null Password to be used for authentication challenges for this registration If non Null will override the Reg Password x parameter entered into the Authentication submenu off of the Settings menu on the phone Configuration Files Permitted Attribute Values Default Interpretation reg x server y address dotted decimal IP Null Optional IP address or host name port address or host transport registration period fail over name parameters and lineseize subscription period of a SIP server that accepts registrations reg x server y port 0 Null 1 to 65535 Null Multiple servers can be listed starting with reg x server y transport DNSnaptr or DNSnap y 1 2 for fault tolerance If specified these servers may override the
46. Incoming Call Treatment The phone can automatically apply distinctive treatment to calls containing specific attributes The distinctive treatment that can be applied includes customizable alerting sound effects and automatic call diversion or rejection Call attributes that can trigger distinctive treatment include the calling party name or SIP contact number or URL format For related configuration changes refer to Local Contact Directory on page 4 9 Distinctive Ringing There are three options for distinctive ringing Configuring Your System 1 The user can select the ring type for each line This option has the lowest priority 2 The ring type for specific callers can be assigned in the contact directory For more information refer to Distinctive Incoming Call Treatment the previous section This option has a higher priority than option 1 and a lower priority than option 3 3 The volpProt SIP alertInfo x value and volpProt SIP alertiInfo x class fields can be used to map calls to specific ring types This option has the highest priority Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Specify the mapping of Alert Info strings to ring types For more information refer to Alert Information lt alertInfo gt on page A 14 Configuration file phone1 cfg Specify the ring type to be used for each line For more informa
47. Name Possible Values Description Server Address dotted decimal IP address The syslog server IP address or host name OR The default value is NULL domain name string Server Type None 0 The protocol that the phone will use to write to the syslog UDP 1 server TCP 2 If set to None transmission is turned off but the server TLS 3 address is preserved Facility 0 to 23 A description of what generated the log message For more information refer to section 4 1 1 of RFC 3164 The default value is 16 which maps to local 0 Render Level 0 to6 Specifies the lowest class of event that will be rendered to syslog Itis based on log render level and can bea lower value Refer to Basic Logging lt level gt lt change gt and lt render gt on page A 77 Note Use left and right arrow keys to change values Prepend MAC Enabled Disabled If enabled the phone s MAC address is prepended to the Address log message sent to the syslog server Setting Up the Boot Server The boot server can be on the local LAN or anywhere on the Internet Note Note we POLYCOM Note Setting up Your System Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses The default number of boot servers is one and the maximum number is eight The following protocols are supported for redundant boot servers HTTPS HTTP and FTP For more information on the protocol used on each p
48. Note A key is generated by the utility and must be downloaded to the phone so that it can decrypt the files that were encrypted on the server The device sec configEncryption key configuration file parameter is used to set the key on the phone The utility generates a random key and the encryption is Advanced Encryption Standard AES 128 in Cipher Block Chaining CBC mode An example key would look like this Crypt 1 KeyDesc companyNameKey1 Key 06a9214036b8a15b512e03d534120006 If the phone doesn t have a key it must be downloaded to the phone in plain text a potential security hole if not using HTTPS If the phone already has a key a new key can be downloaded to the phone encrypted using the old key refer to Changing the Key on the Phone on page C 5 At a later date new phones from the factory will have a key pre loaded in them This key will be changed at regular intervals to enhance security It is recommended that all keys have unique descriptive strings in order to allow simple identification of which key was used to encrypt a file This makes boot server management easier After encrypting a configuration file it is useful to rename the file to avoid confusing it with the original version for example rename sip cfg to sip enc However the directory and override filenames cannot be changed in this manner You can check whether an encrypted file is the same as an unencrypted file by 1 Run the configFileEncrypt utility
49. Platform Bootrom 3 0 1 0026 29 Mar 05 10 29 0224000058 so 00 Application main Label SIP Version 1 6 5 0043 31 Jan 06 11 16 0224000058 so 00 Application main P N 3150 11530 165 0224000058 hw 00 Initial log entry Current logging level 4 0224000058 ares 00 Initial log entry Current logging level 4 0224000058 dns O0 Initial log entry Current logging level 4 0224000058 cfg O0 Initial log entry Current logging level 4 0224000058 dns 0224000058 log 0224000058 curl 0224000058 copy 0224000058 sec 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 00 Resolver initialized added 2 nameservers and set OO Initial log entry Current logging level 4 OO Initial log entry Current logging level 4 O0 Initial log entry Current logging level 3 O0 Initial log entry Current logging level 4 O0 UtilCopyC curl_easy_perform failed curlRes 23 respCode 150 00 UtilCopyC curl error Curl Error strings have been compiled out 00 UtilCopyC curl error buffer Failed writing body O0 UtilCopyC curl_easy_perform failed curlRes 23 respCode 150 00 UtilCopyC curl error Curl Error strings have been compiled out vancouver polycom c 0224000058 copy 0224000058 copy 0224000058 so 0224000058 dns 0223160058 log 0223160058 rtos 0223160058 rdisk 0223160058 res 0223160058 httpa 0223160058 cdp 0223160058 cdp 0223160058 sys 00 Initial log entry Current logging level
50. Size Bandwidth G 711p law PMCU G711mu 64 Kbps 8 Ksps 10ms 80ms 3 5 KHz G 711a law PCMA G711A 64 Kbps 8 Ksps 10ms 80ms 3 5 KHz G 722 G722 8000 G722 64 Kbps 16 Ksps 10ms 80ms 7 KHz G 722 1 G722 16000 G7221 16 Kbps 16 Ksps 20ms 80ms 7 KHz 24 Kbps 32 Kbps G 729AB G729 G729AB 8 Kbps 8 Ksps 10ms 80ms 3 5 KHz These codecs include e Codec Preferences lt codecPref gt e Codec Profiles lt audioProfile gt Codec Preferences lt codecPref gt Permitted Attribute Values Default Interpretation voice codecPref G711Mu Null 1 3 1 Specifies the codec preferences for SoundPoint IP 320 330 430 500 501 voice codecPref G711A 2 600 and 601 platforms voice codecPref G729AB 3 1 highest 3 lowest Null do not use Give each codec a unique priority this will dictate the order used in SDP negotiations voice codecPref IP_300 G711Mu Null 1 3 1 Specifies the codec preferences for SoundPoint IP 301 models Interpretation voice codecPref IP_300 G711A 2 as above voice codecPref IP_300 G729AB 3 voice codecPref IP_650 G711Mu Null 1 4 2 Specifies the codec preferences for the SoundPoint IP 550 560 and 650 platform voice codecPref IP_650 G711A 3 Interpretation as above voice codecPref IP_650 G729AB 4 voice codecPref IP_650 G722 1 Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation voice codecPref IP_4000 G711Mu Null 1 3 1 Specifies the c
51. System The phones can be configured to periodically poll the boot server to check for changed configuration files or application executable If a change is detected the phone will reboot to download the change Refer to Provisioning lt prov gt on page A 81 Supporting SoundPoint IP 300 and 500 Phones With enhancements in BootROM 4 0 0 and SIP 2 1 2 you can modify the 000000000000 cfg or lt Ethernet address gt cfg configuration file to direct phones to load the software image and configuration files based on the phone model number Refer to Master Configuration Files on page A 2 The SIP 2 2 0 or later software distributions contain only the new distribution files for the new release You must rename the sip ld sip cfg and phonel cfg from a previous 2 1 2 distribution that is compatible with SoundPoint IP 300 and 500 phones The following procedure must be used for upgrading to SIP 2 2 0 or later for installations that have SoundPoint IP 300 and 500 phones deployed It is also recommended that this same approach be followed even if SoundPoint IP 300 and 500 phones are not part of the deployment as it will simplify management of phone systems with future software releases To upgrade your SIP application 1 Do one of the following steps a Place the bootrom ld file corresponding to BootROM revision 4 0 0 or later onto the boot server b Ensure that all phones are running BootROM 4 0 0 or later code 2 Copy sip ld sip cfg
52. This will replace whatever phone specific contact directory is on the server even if it is unencrypted sec encryption upload ove rrides 0 1 If set to 0 the phone specific configuration override file lt Ethernet Address gt phone cfg is uploaded unencrypted regardless of how it was downloaded This will replace the override file on the server even if it is encrypted If set to 1 the phone specific configuration override file is uploaded encrypted regardless of how it was downloaded This will replace the override file on the server even if it is unencrypted Password Lengths lt pwd gt lt length gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation sec pwd length admin 0 32 1 Password changes will need to be at least this long Use 0 to allow null passwords sec pwd length user 0 32 2 License lt license gt This attribute s settings control aspects of the feature licensing system This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation license polling time 00 00 23 59 2 00am The time to check whether or not the license has expired Provisioning lt prov gt Configuration Files This attribute s settings control aspects of the phone s boot server provisioning system Attribute Permitted Values Default Interpre
53. Where SoundPoint IP SoundStation IP Phones Fit 2 2 Session Initiation Protocol Application Architecture 2 3 BootROM sep ieeepste ipnr nines erie eam Whole e aa Eoia e 2 3 AP PHCAtOM ssiri beast lh Pl i i E selec pad e eE R EE EEE 2 4 Configuration ergin ires on E E 2 5 Resource Piles serrr rert t IERA we Melee ENEE dae a RAN EL AANE REENE 2 7 Available Features 1 0 0 00 cece cee tee nnn nes 2 8 3 Setting up Your System 0 cece eee eee ee ee Ord Setting Up the Network 00 2 eee eee 3 2 DHCP or Manual TCP IP Setup 0 eee ee eee eee 3 2 Supported Provisioning Protocols 0008 3 4 Modifying the Network Configuration 000 3 5 Setting Up the Boot Server 0 00 3 12 Deploying Phones From the Boot Server 0000s eee ees 3 15 Upgrading SIP Application usuusu sunsun rnrn rrer 3 17 Supporting SoundPoint IP and SoundStation IP Phones 3 18 Supporting SoundPoint IP 300 and 500 Phones 3 19 4 Configuring Your System 2c cece ee e o ARI Setting Up Basic Features 0 ee eee eee 4 1 Call Eog viario i aii tele doit Bc EAEE aide etsy dee tace helt ca eiteiacate Meteid 4 3 Call TIMET ss0c054 vena teri hse een EEEE epee meshed eae menage 4 3 Administrator s Guide SoundPoint IP SoundStation IP vi Call Waitin ga 4 itachi cin ets
54. a lower value Refer to Basic Logging lt level gt lt change gt and lt render gt on page A 77 device syslog prependMac Enabled Disabled If enabled the phone s MAC address is prepended to the log message sent to the syslog server device em power Enabled Disabled Null Refer to the EM Power parameter in Main Menu on page 3 6 device net etherVlianFilter Enabled Disabled Refer to the VLAN Filtering parameter in Ethernet Menu on page 3 11 device net etherStormFilter Enabled Disabled Refer to the Storm Filtering parameter in Ethernet Menu on page 3 11 device sec SSL certList all custom default The type of certificate list device sec SSL customCert X 509 certificate The certificate value 108 Session Initiation Protocol SIP This chapter provides a description of the basic Session Initiation Protocol SIP and the protocol extensions that are supported by the current SIP application To find the applicable Request For Comments RFC document go to http www ietf org rfc html and enter the RFC number This chapter contains information on e Basic Protocols All the basic calling functionality described in the SIP specification is supported Transfer is included in the basic SIP support e Protocol Extensions Extensions add features to SIP that are applicable to a range of applications including reliable 1xx responses and session tim
55. and phonel cfg from the SIP2 2 0 or later release distribution onto the boot server These are the relevant files for all phones except the SoundPoint IP 300 and 500 phones 3 Rename sip Id sip cfg and phonel cfg from the previous distribution to sip_212 ld sip_212 cfg and phone1_212 cfg respectively on the boot server These are the relevant files for supporting the SoundPoint IP 300 and 500 phones 4 Modify the 000000000000 cfg file if required to match your configuration file structure For example lt APPLICATION APP_FILE_PATH sip 1d APP_FILE_PATH_SPIP500 sip_212 1d APP_FILE_PATH_SPIP300 sip_212 1d Administrator s Guide SoundPoint IP SoundStation IP CONFIG_FILES PHONE_MAC_ADDRESS user cfg phonel cfg sip cfg CONFIG_FILES_SPIP500 PHONE_MAC_ADDRESS user cfg phonel_212 cfg sip_212 cfg CONFIG_FILES_SPIP300 PHONE_MAC_ADDRESS user cfg phonel_212 cfg sip_212 cfg MISC_FILES LOG_FILE_DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY gt 5 Remove any lt Ethernet address gt cfg files that may have been used with earlier releases from the boot server Note This approach takes advantage of an enhancement that was added in SIP2 0 1 BootROM 3 2 1 that allows for the substitution of the phone specific MACADDRESS inside configuration files This avoids the need to create unique lt Ethernet address gt cfg files for each phone such that the default 000000000000 cfg file can be us
56. and whether diversion should be disabled on shared lines Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User Specify per registration line type private or shared using the SIP Interface Configuration menu Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations not a mixture of these options When the SIP Configuration menu is used it is assumed that all registrations use the same server Bridged Line Appearance Calls and lines on multiple phones can be logically related to each other A call that is active on one phone will be presented visually to phones that share that line Mutual exclusion features emulate traditional PBX or key system privacy for shared calls Incoming calls can be presented to multiple phones Configuring Your System simultaneously This feature is dependent on support from a SIP server that binds the appearances together logically and looks after the necessary state notifications and performs an access control function For more information refer to Bridged Line Appearance Signaling on page B 10 Note In t
57. at absolute time or relative to boot log sched x startTime positive Seconds since boot when startMode is rel or the start time in 24 hour integer OR clock format when startMode is abs hh mm log sched x startDay 1 7 When startMode is abs specifies the day of the week to start command execution 1 Sun 2 Mon 7 Sat Security lt sec gt This attribute s settings affect security aspects of the phone This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation sec tagSerialNo 0 1 Null If set to 1 the phone may advertise its serial number Ethernet address through protocol signaling If set to 0 or Null the phones does advertise its serial number This attribute also includes e Encryption lt encryption gt e Password Lengths lt pwd gt lt length gt Administrator s Guide SoundPoint IP SoundStation IP Encryption lt encryption gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation sec encryption upload dir 0 1 0 If set to 0 the phone specific contact directory is uploaded to the server unencrypted regardless of how it was downloaded This will replace whatever phone specific contact directory is on the server even if it is encrypted If set to 1 the phone specific contact directory is uploaded encrypted regardless of how it was downloaded
58. compatible it means an application file was downloaded from the boot server but it cannot be installed on this phone This issue can usually be resolved by finding a software image that is compatible with the hardware or the bootROM being used and installing this on the boot server There are various different hardware and software dependencies Refer to the latest Release Notes for details on the version you are using Could not contact boot server using existing configuration The phone could not contact the boot server but the causes may be numerous It may be cabling issue it may be related to DHCP configuration or it could be a problem with the boot server itself The phone can recover from this error so long as it previously downloaded a valid application bootROM image and all of the necessary configuration files Error application is not present There is no application stored in flash memory and the phone cannot boot A compatible SIP application must be downloaded into the phone using one of the supported provisioning protocols You need to resolve the issue of connecting to the boot server This error is typically a result one of the above errors This error is fatal Troubleshooting Your SoundPoint IP SoundStation IP Phones Not all configuration files were present on the server Similarly a message about configuration files not being present means that the phone was able to reach the boot server but that it was not
59. complete UDP Warning Configuring Your System timeout defined in RFC 3261 If it is not the last server in the list the maximum number of retries using the configurable retry timeout is used For more information refer to Server lt server gt on page A 7 and Registration lt reg gt on page A 89 If DNS is used to resolve the address for Servers the DNS server is unavailable and the TTL for the DNS records has expired the phone will attempt to contact the DNS server to resolve the address of all servers in its list before initiating a call These attempts will timeout but the timeout mechanism can cause long delays for example two minutes before the phone call proceeds using the working server To mitigate this issue long TTLs should be used It is strongly recommended that an on site DNS server is deployed as part of the redundancy solution Hosted VoIP Service Provider Ea Server 18 fess Gall Server 1A VolP SMB Customer Premise iso SIP Capable Router Server2 Phone Configuration The phones at the customer site are configured as follows Server 1 the primary server will be configured with the address of the service provider call server The IP address of the server s to be used will be provided by the DNS server For example reg 1 server 1 address voipserver serviceprovider com Server 2 the fallback server will b
60. dial tone DTMF tone while dialing and ringing back are heard by all existing participants in the conference If set to 0 or Null sound effects are only heard by conference initiator old behavior Only supported for SoundPoint IP 550 560 and 650 and SoundStation IP 7000 For all others set to 0 call localConferenceCallHold 0 1 0 If set to 0 a hold will happen for all legs when conference is put on hold old behavior If set to 1 only the host is out of the conference all other parties in conference continue to talk new behavior If set to Null the default value is 0 Only supported for SoundPoint IP 550 560 and 650 and SoundStation IP 7000 For all others set to 0 Administrator s Guide Addendum for the SoundStation IP 7000 Permitted Attribute Values Default Interpretation call cellPhoneAutoBridging Disabled Disabled If set to Enabled the cell phone audio is Enabled automatically bridged to active SIP call conference call If one SIP call is put on hold and another SIP call is activated the cell phone audio is automatically bridged to the newly active SIP call If set to Disabled the cell phone audio is available only if there is no active SIP calls If all SIP calls are placed on hold then the cell phone audio is available Bitmaps lt bitmap gt The IP_7000 parameters affect the SoundStation IP 7000 conference phone Indicators lt ind gt
61. embedded in the SoundPoint IP phones should be configured to give voice traffic emanating from the phone higher transmit priority than those from a device connected to the PC port If not using a VLAN VLAN set to blank in the setup menu this will automatically be the case If using a VLAN ensure that the 802 1p priorities for both default and real time transport protocol RTP packet types are set to 2 or greater Otherwise these packets will compete equally with those from the PC port For more information refer to Quality of Service lt QOS gt on page A 50 Graphic Display Backgrounds Note You can set up a picture or design to be displayed on the background of the graphic display of all SoundPoint IP 550 560 and 650 phones There are a number of default backgrounds both solid and bitmaps Users can select which background appear on their phone You can modify the solid backgrounds and bitmaps For example you can add a grey solid background or modify a bitmap to one of your choice When installing a background of your choice care needs to be taken to ensure that the background does not adversely affect the visibility of the text on the phone display As a general rule backgrounds should be light in shading for better usability To modify the backgrounds displayed on the supported SoundPoint IP phones 1 Modify the sip cfg configuration file as follows a Open sip cfg in an XML editor b Locate the background parameter c
62. for Ethernet TCP IP networks delivering excellent voice quality The high resolution graphic display supplies content for call information multiple languages directory access and system status The SoundPoint IP SoundStation IP family supports advanced functionality including multiple call and flexible line appearances HTTPS secure provisioning presence custom ring tones and local conferencing The SoundPoint IP SoundStation IP phones are end points in the overall network topology designed to interoperate with other compatible equipment including application servers media servers internet working gateways voice bridges and other end points The following models are described e SoundPoint IP Desktop Phones IP301 IP 320 330 IP 430 IP501 IP550 560 IP 600 601 IP650 e SoundStation IP Conference Phone IP 4000 This chapter also lists the key features available on the SoundPoint IP SoundStation IP phones running the latest software Administrator s Guide SoundPoint IP SoundStation IP SoundPoint IP Desktop Phones This section describes the current SoundPoint IP desktop phones For individual guides refer to the product literature available at http www polycom com support voice Additional options are also available For more information contact your Polycom distributor The currently supported desktop phones are e SoundPoint IP 301 e SoundPoint IP 320 330 Introducing
63. for the SIP application for known problems and possible workarounds For the latest Release Notes and the latest version of this Administrator s Guide go to Polycom Technical Support at http www polycom com support voice If your problems is not listed in this chapter nor described in the latest Release Notes contact your Certified Polycom Reseller for support Administrator s Guide SoundPoint IP SoundStation IP Error Messages There are several different error messages that can be displayed on the phone when it is booting Some of these errors are fatal meaning that the phone will not able to boot until this issue has been resolved and some are recoverable meaning the phone will continue booting after the error but the configuration of the phone may not be what you were expecting BootROM Error Messages Most of these errors are also logged on the phone s boot log however if you are having trouble connecting to the boot server the phone will likely not be able to upload the boot log for you to examine Failed to get boot parameters via DHCP The phone does not have an IP address and therefore cannot boot Check that all cables are connected the DHCP server is running and that the phone has not been put into a VLAN which is different from the DHCP server Check the DHCP configuration Application lt file name gt is not compatible with this phone When the bootROM displays an error like The application is not
64. gt 2 0 0 ect e etn eeas A 83 Resource lt teS gt 2 0 ccs ence ned be ehaw cede he eta kenga he eee Boe A 84 Microbrowser lt mb gt 0 cece cee cette tenes A 86 Per Phone Configuration 6006 sser rrr eee eee eee A 89 Registration lt feG gt ecese nie onde cede el we haw ad yew ERGERE A 89 Calls lt call gt 2 2 0 cee etian oe eee Wedd aes ee cede eta Ea A 93 Diversion lt divert gt reres KE cece cece ee eee eee A 96 Dial Plan lt dialplan gt 00 0 e eee eee eee eee A 98 Messaging lt msg gt 0000s A 101 Network Address Translation lt nat gt 00 00 eee A 102 Attendant lt attendant gt 0 cece ccc cence eens A 103 Roaming Buddies lt roaming_buddies gt 4 A 104 Roaming Privacy lt roaming_privacy gt 00 A 104 User Preferences lt user_preferences gt 00005 A 105 Flash Parameter Configuration 0 5 606 sssr rirse rnrn A 105 B Session Initiation Protocol SIP Brl RFC and Internet Draft Support 20 0 2 eee eee eee B 2 Request Support 35 0 sib0 Pig bbe rider ie bE EEEE A be baw ee eee wa B 3 Header Support 0 cece eee eee USEE B 4 Response SUP POL ee rrt treka EEEREN ROS HENS RAE ees B 6 Hold Implementation 00 000s B 9 Reliability of Provisional Responses 0 0000 eee eens B 9 Transfer 6 occ cee ne tee genre eet ee eve Se
65. gt A 36 comfort noise fill 4 51 conference setup lt conference gt A 15 configurable feature keys 4 23 configuration file encryption 4 55 configuration file example 4 44 configuring SoundPoint IP SoundStation IP phones locally 4 55 connected party identification 4 5 consultative transfers 4 18 context sensitive volume control 4 5 corporate directory 4 33 A 63 A 83 custom certificates 4 54 customizable audio sound effects 4 5 customizable fonts and indicators 4 28 D date and time lt datetime gt A 23 default feature key layouts C 10 default password 3 5 4 56 C 9 deploying phones from the boot server 3 15 device lt device gt A 105 DHCP secondary server 3 3 DHCP INFORM 3 3 3 7 3 8 DHCP menu 3 7 DHCP or manual TCP IP setup 3 2 dial plan lt dialplan gt A 16 digit map default A 18 examples A 17 match and replace A 17 digit map lt digitmap gt A 99 directed call pick up 4 21 directory lt dir gt A 61 distinctive call waiting 4 7 distinctive incoming call treatment 4 6 distinctive ringing 4 7 diversion A 96 DND See also do not disturb DNS SIP server name resolution 4 40 do not disturb 4 8 do not disturb lt dnd gt A 94 A 98 downloadable fonts 4 30 DTMF event RTP payload 4 49 DTMF tone generation 4 49 DTMF See also dual tone multi frequency dual tone multi frequency lt DMTF gt A 25 E electronic hookswitch 4 9 emergency lt emergency gt A 19 A 101 emergency routing A 19
66. is not Null this attribute overrides the global server based call forwarding flag in the sip cfg configuration file reg x auth optimizedinFailover 0 1 0 If set to 1 when failover occurs the first new SIP request is sent to the server that sent the proxy authentication request If set to 0 when failover occurs the first new SIP request is sent to the server with the highest priority in the server list If this parameter is Null voIpProt SIP authOptimizedInFailover is checked If both parameters are set this parameter takes precedence Calls lt call gt This attribute affects the call oriented per phone configuration This attribute includes Do Not Disturb lt donotdisturb gt Automatic Off Hook Call Placement lt autoOffHook gt Missed Call Configuration lt serverMissedCall gt Missed Call Tracking lt missedCallTracking gt Call Waiting lt callWaiting gt Administrator s Guide SoundPoint IP SoundStation IP Do Not Disturb lt donotdisturb gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation call donotdisturb perReg 0 1 0 If set to 1 the DND feature will allow selection of DND on a per registration basis Automatic Off Hook Call Placement lt autoOffHook gt An optional per registration feature is supported which allows automatic call placement when the phone goes off hook In the followi
67. is not a valid line registration number it is ignored pres idleSoftkeys 0 1 Null If set to Null or O the presence idle soft keys MyStat and Buddies do not appear If set to Null or 1 the presence idle soft keys appear Administrator s Guide SoundPoint IP SoundStation IP Fonts lt font gt A 66 Note This section does not apply to the SoundPoint IP 301 phones These settings control the phone s ability to dynamically load an external font file during boot up Loaded fonts can either overwrite pre existing fonts embedded within the software not recommended or can extend the phone s font support for Unicode ranges not already embedded The font file must be a Microsoft fnt or fon file format fon file format is a collection of fnt fonts grouped together within a single file The font file name must follow a specific pattern as described e Font filename lt fontName gt _ lt fontHeight InPixels gt _ lt fontRange gt lt fontExtension gt e lt fontName gt is a free string of characters that typically carries the meaning of the font Examples are fontFixedSize for a fixed size font or fontProportionalSize for a proportional size font e lt fontHeightInPixels gt describes the font height in number of screen pixels e lt fontRange gt describes the Unicode range covered by this font Since fnt or fon are 256 characters based blocks the lt fontRange gt is Uxx00_UxxFF
68. key functions can be changed from the factory defaults Corporate Directory The phone can be configured to access your corporate directory if it has a standard LDAP interface Customizable Fonts and Indicators The phone s user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns Downloadable Fonts New fonts can be loaded onto the phone Instant Messaging Supports sending and receiving instant text messages Microbrowser The SoundPoint IP 430 501 550 560 600 601 and 650 phones and the SoundStation IP 4000 phone support an XHTML microbrowser Microsoft Live Communications Server 2005 Integration SoundPoint IP and SoundStation IP phones can used with Microsoft Live Communications Server 2005 and Microsoft Office Communicator to help improve business efficiency and increase productivity and to share ideas and information immediately with business contacts Requires call server support Overview Multilingual User Interface All phones except SoundPoint IP 301 have multilingual user interfaces Multiple Call Appearances The phone supports multiple concurrent calls The hold feature can be used to pause activity on one call and switch to another call Multiple Line Keys per Registration More than one line key can be allocated to a single Multiple Registrations SoundPoint IP phones support multiple registrations per phon
69. locally Local Digit Map The phone has a local digit map to automate the setup phase of number only calls Administrator s Guide SoundPoint IP SoundStation IP Message Waiting Indication The phone will flash a message waiting indicator MWI LED when instant messages and voice messages are waiting Microphone Mute When the microphone mute feature is activated visual feedback is provided Missed Call Notification The phone can display the number of calls missed since the user last looked at the Missed Calls list Soft Key Activated User Interface The user interface makes extensive use of intuitive context sensitive soft key menus Speed Dial The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu Time and Date Display Time and date can be displayed in certain operating modes such as when the phone is idle and during a call Advanced Features Automatic Call Distribution Supports ACD agent available and unavailable and allows ACD login and logout Requires call server support Bridged Line Appearance Calls and lines on multiple phones can be logically related to each other Requires call server support Busy Lamp Field Allows monitoring the hook status and remote party information of users through the busy lamp field BLF LEDs and displays on an attendant console phone Requires call server support Configurable Feature Keys Certain
70. more information refer to Request Validation lt requestValidation gt on page A 14 Configuration File Encryption Note Configuration files excluding the master configuration file contact directories and configuration override files can all be encrypted The SoundPoint IP 300 and 500 phones will always fail at decrypting files These phones will recognize that a file is encrypted but cannot decrypt it and will display an error Encrypted configuration files can only be decrypted on the SoundPoint IP 301 320 330 430 501 550 560 600 601 and 650 and the SoundStation IP 4000 phones The master configuration file cannot be encrypted on the boot server This file is downloaded by the bootROM that does not recognize encrypted files For more information refer to Master Configuration Files on page A 2 For more information on encrypting configuration files including determining whether an encrypted file is the same as an unencrypted file and using the SDK to facilitate key generation refer to Encrypting Configuration Files on page C 3 Configuration changes can performed centrally at the boot server Central boot server Configuration File sip cfg Specify the phone specific contact directory and the phone specific configuration override file For more information refer to Encryption lt encryption gt on page A 80 Configuration file lt device gt cfg Change the encryption key
71. not upgrade past 2 4 e Only 2 6 can upgrade to 3 0 e 3 0 cannot be downgraded For example a two step upgrade would be necessary from bootROM 2 1 to bootROM 2 5 A direct upgrade is not supported but upgrading to bootROM 2 2 first then upgrading to 2 5 will work Downgrade restrictions are limited to major releases Going from 2 x to 1 x and from 3 x to 2 x are both impossible in the field Miscellaneous Administrative Tasks Multiple Key Combinations On SoundPoint IP and SoundStation IP phones certain multiple key combinations can be used to reboot the phone and restore factory defaults For other methods for resetting and rebooting your SoundPoint IP or SoundStation IP phones refer to Quick Tip 18298 Resetting and Rebooting SoundPoint IP Phones at http www polycom com support voice Rebooting the Phone For the key combination press and hold certain key combinations depending on the phone model simultaneously until a confirmation tone is heard or for about three seconds e IP 301 Volume Volumet Hold and Do Not Disturb e IP 320 and 330 Volume Volumet Hold and Hands free IP 430 and 501 Volume Volumet Hold and Messages e IP 550 560 600 601 and 650 Volume Volume Mute and Messages e IP 4000 Volume and Select Restoring Factory Defaults For the key combination press and hold certain key combinations depending on the phone model simultaneously during the countdown process i
72. occurs on the line which is currently in the call appearance Any new call scenario seizes the next available line Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation call singleKeyPressConference 0 1 0 If set to 1 the conference will be setup after a user presses the Conference soft key or Conference key the first time Also all sound effects dial tone DTMF tone while dialing and ringing back are heard by all existing participants in the conference If set to 0 or Null sound effects are only heard by conference initiator old behavior Only supported for SoundPoint IP 550 560 650 For all others set to 0 call localConferenceCallHold 0 1 0 If set to 0 a hold will happen for all legs when conference is put on hold old behavior If set to 1 only the host is out of the conference all other parties in conference continue to talk new behavior If set to Null the default value is 0 Only supported for SoundPoint IP 550 560 650 For all others set to 0 call transfer blindPreferred 0 1 Null If set to 1 the Blind soft key appears as a transfer type If set to O or Null the Normal soft key appears Note This parameter is supported on the SoundPoint IP 330 320 only This attribute also includes e Shared Calls lt shared gt e Hold Local Reminder lt hold gt lt localReminder gt Shared Calls lt shared gt This configu
73. of record managed by a call server The server allows multiple end points to register locations against the address of record The phone supports shared call appearances SCA using the SUBSCRIBE NOTIFY method in the SIP Specific Event Notification framework RFC 3265 The events used are e call info for call appearance state notification e line seize for the phone to ask to seize the line Bridged Line Appearance Signaling 10 A bridged line is an address of record managed by a server The server allows multiple end points to register locations against the address of record The phone supports bridged line appearances BLA using the SUBSCRIBE NOTIFY method in the SIP Specific Event Notification framework RFC 3265 The events used are e dialog for bridged line appearance subscribe and notify Miscellaneous Administrative Tasks This appendix provides information required by varied aspects of the Session Initiation Protocol SIP application This includes Trusted Certificate Authority List Encrypting Configuration Files Adding a Background Logo BootROM SIP Application Dependencies Multiple Key Combinations Default Feature Key Layouts Assigning a VLAN ID Using DHCP Parsing Vendor ID Information Trusted Certificate Authority List The following certificate authorities are trusted by the phone by default ABAecom sub Am Bankers Assn Root CA ANX Network CA by DST American Express
74. on the Server menu Name Possible Values Description Server Type O FTP 1 TFTP 2 HTTP The protocol that the phone will use to obtain 3 HTTPS 4 FTPS 5 Invalid configuration and phone application files from the boot server Refer to Supported Provisioning Protocols on page 3 4 Note Active FTP is not supported for bootROM version 3 0 or later Passive FTP is still supported Administrator s Guide SoundPoint IP SoundStation IP Name Possible Values Description Server Address dotted decimal IP address OR domain name string OR URL All addresses can be followed by an optional directory and optional file name The boot server to use if the DHCP client is disabled the DHCP server does not send a boot server option or the Boot Server parameter is set to Static The phone can contact multiple IP addresses per DNS name These redundant boot servers must all use the same protocol If a URL is used it can include a user name and password Refer to Supported Provisioning Protocols on page 3 4 A directory and the master configuration file can be specified Note or can be used in the user name or password these characters if they are correctly escaped using the method specified in RFC 1738 Server User any string The user name used when the phone logs into the server if required for the selected Server Type Note If the Server Address is a UR
75. on the unencrypted file with the d option This shows the digest field 2 Look at the encrypted file using WordPad and check the first line that shows a Digest field If the two fields are the same then the encrypted and unencrypted file are the same If a phone downloads an encrypted file that it cannot decrypt the action is logged an error message displays and the phone reboots The phone will continue to do this until the boot server provides an encrypted file that can be read an unencrypted file or the file is removed from the master configuration file list The SoundPoint IP 300 and 500 phones will always fail at decrypting files These phones will recognize that a file is encrypted but cannot decrypt it and will display an error This information is logged Encrypted configuration files can only be decrypted on the SoundPoint IP 301 320 330 430 501 550 560 600 601 and 650 and the SoundStation IP 4000phones The master configuration file cannot be encrypted on the boot server This file is downloaded by the bootROM that does not recognize encrypted files For more information refer to Master Configuration Files on page 2 5 Miscellaneous Administrative Tasks The following configuration file changes are required to modify this feature Central boot server Configuration File sip cfg Specify the phone specific contact directory and the phone specific configuration override file For more infor
76. or 6416 polycom com Null Forward to contact used for calls forwarded due to no answer if Null divert x contact will be used Administrator s Guide SoundPoint IP SoundStation IP Do Not Disturb lt dnd gt The phone can automatically divert calls when Do Not Disturb DND is enabled Attribute Permitted Values Default Interpretation divert dnd x enabled 0 1 0 If set to 1 calls will be forwarded on DND to the contact specified below Note If server based DND or server base call forwarding is enabled this parameter is disabled divert dnd x contact ASCII encoded string containing digits Null Forward to contact used for the user part of a SIP URL or a string calls forwarded due to DND that constitutes a valid SIP URL 6416 or status if Null 6416 polycom com divert x contact will be used Dial Plan lt dialplan gt Per registration dial plan configuration is supported In the following tables x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Permitted Attribute Values Default Interpretation dialplan x applyToCallListDial 0 1 0 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 dialplan x ap
77. or to replace certain matched digits with the introduction of R to the digit map Configuring Your System Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify impossible match behavior trailing behavior digit map boot server sip cfg matching strings and time out value For more information refer to Dial Plan lt dialplan gt on page A 16 Configuration file Specify per registration impossible match behavior trailing phonet cfg behavior digit map matching strings and time out values that override those in sip cfg For more information refer to Dial Plan lt dialplan gt on page A 98 Local Web Server Specify impossible match behavior trailing behavior digit map if enabled matching strings and time out value Navigate to http lt phonelPAddress gt appConf htm ls Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Microphone Mute A microphone mute feature is provided When activated visual feedback is provided This is a local function and cannot be overridden by the network There are no related configuration changes Soft Key Activated User Interface Speed Dial The user inter
78. packets such as SIP signaling packets with default settings configurable for all other packets Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify default and protocol specific 802 1p Q settings boot server sip cfg For more information refer to Ethernet IEEE 802 1p Q lt ethernet gt on page A 50 Local Web Server Specify 802 1p Q settings if enabled Navigate to http lt phonelPAddress gt netConf htm qo Local Phone User Specify whether CDP is to be used or manually set the VLAN ID or Interface configure DHCP VLAN Discovery Phase 1 bootRom Navigate to SETUP menu during auto boot countdown Phase 2 Application Navigate to Menu gt Settings gt Advanced gt Admin Settings gt Network Configuration For more information refer to Setting Up the Network on page 3 2 Voice Quality Monitoring Note This feature requires a license key for activation Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller The SoundPoint IP phones can be configured to generate various quality metrics for listening and conversational quality These metrics can be sent between the phones in RTCP XR packets The metrics can also be downloaded in SIP messages to a central voice quality report collector The collection of these metrics is supported on the SoundPoi
79. phone cfg on the boot server for security reasons Custom Certificates The phone trusts certificates issued by widely recognized certificate authorities when trying to establish a connection to a boot server for application provisioning Refer to Trusted Certificate Authority List on page C 1 In addition custom certificates can be added to the phone This is done by using the SSL Security menu on the phone to provide the URL of the custom certificate then select an option to use this custom certificate Note For more information on using custom certificates refer to Technical Bulletin 17877 Using Custom Certificates With SoundPoint IP Phones at www polycom com support voice Configuration changes can performed locally Local Local Phone User The custom certificate can be specified and the type of certificate to Interface trust can be set under the Settings menu Incoming Signaling Validation The three optional levels of security for validating incoming network signaling are e Source IP address validation e Digest authentication e Source IP address validation and digest authentication 4 54 Configuring Your System Configuration changes can performed centrally at the boot server Central Configuration File boot server sip cfg Specify the type of validation to perform on a request by request basis appropriate to specific event types in some cases For
80. resources for the audio bridging There is no dependency on network signaling for local conferences The phone also supports centralized conferences for which external resources are used such as a conference bridge This relies on network signaling Note Conferences are not available when the G 729 codec is enabled on the SoundStation IP 4000 conference phone Configuration changes can performed centrally at the boot server Central Configuration file Specify the conference hold behavior all parties on hold or only host boot server sip cfg is on hold For more information refer to Call Handling Configuration lt call gt on page A 58 Specify whether or not all parties hear sound effects while setting up a conference For more information refer to Call Handling Configuration lt call gt on page A 58 Specify which type of conference to establish and the address of the centralized conference resource For more information refer to Conference Setup lt conference gt on page A 15 Manage Conferences Note This feature is only supported on the SoundPoint IP 550 560 and 650 desktop phones This feature requires a license key for activation Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller The individual parties within a conference can be managed Configuration changes can be performed centrally at the boot
81. server Central Configuration file Turn this feature on or off boot server sip ctg e For more information refer to Feature lt feature gt on page A 83 Administrator s Guide SoundPoint IP SoundStation IP Call Forward Note The phone provides a flexible call forwarding feature to forward calls to another destination Call forwarding can be applied in the following cases e Automatically to all calls e Calls from a specific caller extension e When the phone is busy e When Do Not Disturb is active e After an extended period of alerting The user can elect to manually forward calls while they are in the alerting state to a predefined or manually specified destination The call forwarding feature works in conjunction with the distinctive incoming call treatment feature refer to Distinctive Incoming Call Treatment on page 4 6 The user s ability to originate calls is unaffected by all call forwarding options Each registration has its own forwarding properties Server based call forwarding is active if the feature is enabled on both the phone and the server and the phone is registered If server based call forwarding is enabled on any of the phone s registrations the other registrations are not affected Server based call forwarding will behave the same as per SIP2 1 feature with the following exceptions There is no indication on the phone s user interface whether or not server based call forwar
82. setting or volume is low Do one of the following Adjust the ringing level from the front panel using the volume up down keys Check same status of handset headset if connected and through the Hands Free Speakerphone Outbound or inbound calling is unsuccessful Do one of the following Place a call to the phone under investigation Check that the display indicates incoming call information Lift the handset Ensure dial tone is present and place a call to another extension or number Check that the display changes in response The line icon shows an unregistered line icon The phone line is unregistered Contact your system administrator 5 12 Displays Troubleshooting Your SoundPoint IP SoundStation IP Phones Symptom Problem Corrective Action There is no display The display is incorrect The display has bad contrast Power is not correctly applied to the SoundPoint IP family SIP phone Do one of the following e Check that the display is illuminated Make sure the LAN cable is inserted properly at the rear of the phone try unplugging and re inserting the cable e If using in line powering have your system administrator check that the switch is supplying power to the phone The contrast needs adjustment Do one of the following e Refer to the appropriate SoundPoint IP SoundStation IP SIP phone User Guide Reboot the phone to obt
83. specified port will be used Subsequent ports will be allocated from a pool starting with the specified port plus two up to a value of start port 46 after which the port number will wrap back to the starting value Configuration Files Keep Alive lt keepalive gt Allowing for the configuration of TCP keep alive on SIP TLS connections the phone can detect a failures quickly in minutes and attempt to re register with the SIP call server or its redundant pair This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation tcplpApp keepalive tcp idleTransmitlnterval 10 to 7200 Null After idle x seconds the keep alive message is sent to the call server If set to Null the default value is 30 seconds Note If this parameter is set to a value that is out of range the default value is used terval tcplpApp keepalive tcp noResponseTrasmitin 5 to 120 Null If no response is received to keep alive message another keep alive message is sent to the call server after x seconds If set to Null the default value to 20 seconds Note If this parameter is set to a value that is out of range the default value is used tcplpApp keepalive tcp sip tls enable 0 1 0 If set to 1 enable TCP keep alive for SIP signalling connections that use TLS transport If set to 0 disable TCP keep alive for SIP signalling connections that use TLS tran
84. the SoundPoint IP SoundStation IP Family e SoundPoint IP 430 e SoundPoint IP 501 e SoundPoint IP 550 560 Administrator s Guide SoundPoint IP SoundStation IP e SoundPoint IP 600 601 e SoundPoint IP 650 SoundStation IP Conference Phone This section describes the current SoundPoint IP conference phone For individual guides refer to the product literature available at http www polycom com support voice Additional options are also available For more information contact your Polycom distributor Introducing the SoundPoint IP SoundStation IP Family The currently supported conference phone is SoundStation IP 4000 Key Features of Your SoundPoint IP SoundStation IP Phones The key features of the SoundPoint IP SoundStation IP phones are Award winning sound quality and full duplex speakerphone or conference phone Permits natural high quality two way conversations one way monitor speaker in the SoundPoint IP 301 Uses Polycom s industry leading Acoustic Clarity Technology Easy to use Aneasy transition from traditional PBX systems into the world of IP Upto 18 dedicated hard keys for access to commonly used features Up to four context sensitive soft keys for further menu driven activities Platform independent Supports multiple protocols and platforms enabling standardization on one phone for multiple locations systems and vendors Polycom
85. throughout the life of the SoundPoint IP SoundStation IP phones where certain dependencies on specific bootROM and application versions have been necessitated Administrator s Guide SoundPoint IP SoundStation IP This table summarizes some the major dependences that you are likely to encounter Model BootROM SIP Application IP 301 501 2 6 1 or later 1 4 2 1 5 4 1 6 1 or later IP 320 330 3 2 3 B or later 2 1 1 or later IP 430 3 1 3 C or later 1 6 6 or later IP 550 3 2 2 B or later 2 1 or later IP 560 4 0 1 or later 2 2 2 or later IP 600 2 0 or later 1 0 or later IP 601 EM 3 1 or later 1 6 or later IP 650 EM 3 2 2 B or later 2 0 3 B or later IP 650 BEM 4 0 1 or later 2 2 2 or later IP 4000 2 6 or later 1 4 or later Migration Dependencies In addition to the bootROM and application dependencies there are certain restrictions with regard to upgrading or downgrading from one bootROM release to another bootROM release These restrictions are typically caused by the addition of features that change the way bootROM provisioning is done so the older version become incompatible There is always a way to move forward with bootROM releases although it may be a two or three step procedure sometimes but there are cases where it is impossible to move backward Make special note of these cases before upgrading Note that e 1 x cannot be upgraded to any 2 x automatically e 2 0 and 2 1 can
86. to which the phone directs Domain Name System queries device dns domain any string The phone s DNS domain device auth localAdminPas any string The phone s local administrator password sword device auth localUserPass any string The phone user s local password word A 107 Administrator s Guide SoundPoint IP SoundStation IP Name Possible Values Description device auth regUserx any string The SIP registration user name for registration x where x 1 to 12 device auth regPasswordx any string The SIP registration password for registration x where x 1 to 12 device sec configEncryptio any string Configuration encryption key that is used for n key encryption of configuration files device syslog serverName dotted decimal IP address OR domain name string The syslog server IP address or host name The default value is NULL device syslog transport None 0 The protocol that the phone will use to write to the UDP 1 syslog server TCP 2 If set to None transmission is turned off but the TLS 3 server address is preserved device syslog facility 0 to 23 A description of what generated the log message For more information refer to section 4 1 1 of RFC 3165 The default value is 16 which maps to local 0 device syslog renderLevel 0 to6 Specifies the lowest class of event that will be rendered to syslog It is based on log render level and can be
87. up for platforms with more SDRAM If set to O or Null the default value of 600 is used Note For the SoundPoint IP 650 platform this value is internally replaced by 2X the value Quotas lt quotas gt This configuration attribute is defined as follows Permitted Attribute Values Interpretation res quotas x name 1 tone The name of the sub application for which the particular quota 2 bitmap will apply 3 font 5 background tone relates to all downloaded tones and sound effects bitmap relates to all downloaded bitmaps font relates to all downloaded fonts background relates to all downloaded backgrounds res quotas x value positive integer When a particular resource one of category font bitmap or font is downloaded to the phone a quota equal to this value 1024 bytes of compound data size is applied for that category If downloading a resource would exceed the quota for that category the resource will not be downloaded and a predefined default will be used instead For res quotas x value the default is 600 KB for tones 10 KB for bitmaps and fonts and 600KB for backgrounds Note For the SoundPoint IP 650 platform this value is internally replaced by 2X the value Administrator s Guide SoundPoint IP SoundStation IP Microbrowser lt mb gt This attribute s settings control the home page proxy and
88. using the SIP Configuration menu Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations not a mixture of these options When the SIP Configuration menu is used it is assumed that all registrations use the same server Busy Lamp Field Note This feature is available only on SoundPoint IP 600 phones and SoundPoint IP 601 and 650 phones with an attached Expansion Module The Busy Lamp Field BLF feature enhances support for a phone based attendant console It allows monitoring the hook status and remote party information of users through the busy lamp fields and displays on an attendant console phone Administrator s Guide SoundPoint IP SoundStation IP we Polycom recommends that the BLF not be used in conjunction with the Microsoft Live Communications Server 2005 feature For more information refer to Microsoft POLYCOM Live Communications Server 2005 Integration on page 4 43 Note Use this feature with TCPpreferred transport refer to Server lt server gt on page A 7 Configuration changes can performed centrally at the boot server Central Configuration file Specify the list SIP URI and index of the registration which will be boot server phonet cfg used to send a SUBSCRIBE to the list SIP URI specified in attendant uri For more information refer to Attendant lt attendant gt on page A 103 Customizable Fonts an
89. when the protocol offers the option Note tone dtmf chassis masking should be enabled when tone dtmf viaRtp is disabled tone dtmf rfc2833Control 0 1 1 If set to 1 the phone will indicate a preference for encoding DTMF through RFC 2833 format in its Session Description Protocol SDP offers by showing support for the phone event payload type this does not affect SDP answers these will always honor the DTMF format present in the offer since the phone has native support for RFC 2833 tone dtmf rfc2833Payload 96 127 101 The phone event payload encoding in the dynamic range to be used in SDP offers A 26 Chord Sets lt chord gt Chord sets are the building blocks of sound effects that use synthesized rather than sampled audio most call progress and ringer sound effects A chord set is a multi frequency note with an optional on off cadence A chord set can contain up to four frequency components generated simultaneously each with its own level There are three blocks of chord sets e callProg used for call progress sound effect patterns e ringer e misc miscellaneous All three blocks use the same chord set specification format Configuration Files In the following table x is the chord set number and cat is one of callProg ringer or misc Permitted Attribute Values Interpretation tone chord cat x freq y 0 1600 Frequency for this component in Hertz up to fou
90. yes gt lt Default Master SIP Configuration File gt lt edit and rename this file to lt Ethernet address gt cfg for each phone gt lt SRCSfile 000000000000 cfg v SRevision gt lt APPLICATION APP_FILE_PATH sip ld CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG_FILE_DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY APP_FILE_PATH_SPIP300 SPIP300 sip 1d CONFIG_FILES_SPIP300 phonel_SPIP300 cfg sip_SPIP300 cfg APP_FILE_PATH_SPIP500 SPIP500 sip 1d CONFIG_FILES_SPIP500 phonel_SPIP500 cfg sip_SPIP500 cfg gt For more information refer to Technical Bulletin 35361 Overriding Parameters in Master Configuration File on SoundPoint IP Phones at http www polycom com support voice Application Configuration The configuration file sip cfg contains SIP protocol and core configuration settings that would typically apply to an entire installation and must be set before the phones will be operational unless changed through the local web Configuration Files server interface or local menu settings on the phone These settings include the local port used for SIP signaling the address and ports of a cluster of SIP application servers voice codecs gains and tones and other parameters These parameters include e Protocol lt volpProt gt e Dial Plan lt dialplan gt e Localization lt Icl gt e User Preferen
91. 0 there will be no background downloading from the LDAP server If set to 1 or Null there will be background downloading of data from the LDAP server Configuration Files Aitribute Permitted Values Default Interpretation dir corp backGroundSync period 60 to 65535 seconds 300 The corporate directory cache is refreshed after the corporate directory feature has not been used for this period of time dir corp viewPersistence 0 1 If set to 0 or Null the browse position in the data on the LDAP server and the attribute filters are reset for subsequent usage of the corporate directory If set to 1 the browse position in the data and the attribute filters are retained for subsequent usage of the corporate directory dir corp leg viewPersistence 0 1 This parameter is the same as dir corp viewPersistence Note For SoundPoint IP 301 501 600 and 601 legacy phones use leg tagged parameter This prevents slow behaviour after exiting from the corporate directory Presence lt pres gt The parameter pres reg is the line number used to send SUBSCRIBE If this parameter is missing the phone will use the primary line to send SUBSCRIBE Permitted Attribute Values Default Interpretation pres reg positive 1 Specifies the line registration integer number used to send SUBSCRIBE for presence Must be a valid line registration number If the number
92. 00 gt on page A 74 Administrator s Guide SoundPoint IP SoundStation IP LEDs lt led gt In the following table x is the LED number Permitted Attribute Values Interpretation ind led x index This is for internal usage only and should not be changed this is the logical index ind led x class positive integer Assigns the class defined in Classes lt class gt on page A 73 for this indicator ind led x physNum This maps the logical index to a specific physical LED Graphic Icons lt gi gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt In the following table x IP_300 IP_330 IP_400 IP_500 IP_600 IP_4000 y is the graphic icon number Note that IP_300 parameters affect SoundPoint IP 301 phones IP_330 parameters affect SoundPoint IP 320 and 330 phones IP_400 parameters affectSoundPoint IP 430 phones IP_500 parameters affect SoundPoint IP 501 phones and IP_600 parameters affect SoundPoint IP 550 560 600 601 and 650 phones IP_4000 parameters affect SoundStation IP 4000 phones Permitted Attribute Values Interpretation ind gi x y index This is for internal usage only and should not be changed this is the logical index ind gi x y class positive integer Assigns the class defined in Classes lt class gt on page A 73 for this indicator ind gi x y physX IP 300 0 19 For Graphic Icon type indicators this is the x axis lo
93. 0223160058 wdog 00 Initial log entry Current logging level 0223160118 sec O0 utilCryptoConfigFileEncrypted Could not read file ffs0 local local dir 0223160118 ssps O0 Initial log entry Current logging level 0223160118 net 0223160118 httpd 0223160118 key 0223160118 ssps 0223160118 ssps 00 UtilCopyc 00 Uti ican joseeonfiguration phonel cfg sip cfg 00 Log render Ievel set to 17 00 Initial log 00 Initial log 00 Initial log O0 Initial log O0 Initial log entry entry entry entry entry 00 CDP is DISABLED 00 Initial log O0 Initial log O0 Initial log 00 Application 00 Application entry entry entry comp comp Current logging level Current logging level Current logging level Current logging level Current logging level Current logging level Current logging level Current logging level 1 Label PolyDSP Orion Mem2 FS1 1 P N 3150 11580 133 a a a ee a a a a a curl error buffer Failed writing body Tia f ferenaoaa feundl Wedsatinew file ffe sin nfa con VYersion 1 3 3 0010 0 KKK KKK EK RK KKK KK RK KR OWWWWWW kK KR ROK OK 0223160118 pps 0223160118 sip 0223160118 app1 0223160121 so 0223160121 slog 0223160121 res 0223160121 so 0223160153 sip O0 Initial log entry ResFinderC Failed to download file SoundPoint IPielcone wav errno Oxdl SoToneC Failed to find tone SoundPoint PWelcone vay using default POU LASSE LHUDI YJ raracu VU UUWHLUOU LLLG JUUHU
94. 128 144 157 and 191 in that order for a valid DVD string When set to Custom the value set in VLAN ID Option will be examined for a valid DVD string DVD string in the DHCP option must meet the following conditions to be valid Must start with VLAN A case sensitive Must contain at least one valid ID VLAN IDs range from 0 to 4095 Each VLAN ID must be separated by a character The string must be terminated by a Allcharacters after the will be ignored There must be no white space before the VLAN IDs may be decimal hex or octal For example The following DVD strings will result in the phone using VLAN 10 VLAN A 10 VLAN A 0x0a VLAN A 012 If a VLAN tag is assigned by CDP DHCP VLAN tags will be ignored Miscellaneous Administrative Tasks The following figure shows the phone s processing to determine if the VLAN ID is valid DHCP Discover no VLAN tag Valid DVD string present in DHCP option Yes Release DHCP address For each VLAN listed in DVD string max 10 Response to DHCP Discover on VLAN X received More VLANs in DVD string No Boot process continues with VLAN tag assigned Phone Rebools Boot process continues without any VLAN assigned Parsing Vendor ID Information After the phone boot it sends a DHCP Discover packet to the DHCP server This is found in the Bootstrap Protocol opt
95. 6d sub option 1 company length Polycom 02 15 53 6f 75 6e 64 50 6f 69 6e 74 49 50 2d 53 50 49 50 5f 36 30 31 sub option 2 part length SoundPointIP SPIP_601 03 10 32 33 34 35 2d 31 31 36 30 35 2d 30 30 31 2c 32 sub option 3 part number length 2345 11605 001 2 04 1c 53 49 50 2f 54 69 70 2e 58 58 58 58 2f 30 38 2d 4a 75 6e 2d 30 37 20 31 30 3a 34 34 sub option 4 Application version length SIP Tip XXXX 08 Jun 07 10 44 05 ld 42 52 2f 33 2e 31 2e 30 2e 58 58 58 58 2f 32 38 2d 41 70 72 2d 30 35 20 31 33 3a 33 30 sub option 5 BootROM version length BR 3 1 0 XXXX 28 Apr 05 13 30 ff end of sub options For the BootROM sub option 4 and sub option 5 will contain the same string The string is formatted as follows lt apptype gt lt buildid gt lt date time gt where lt apptype gt can be BR BootROM or SIP SIP Application Third Party Software This appendix provides the copyright statements for third party software products that are part of the Session Initiation Protocol SIP application Ares Copyright 1998 by the Massachusetts Institute of Technology Permission to use copy modify and distribute this software and its documentation for any purpose and without fee is hereby granted provided that the above copyright notice appear in all copies and that both that copyright notice and this permission notice appear in supporting documentation and that the name of M I T not be used in adve
96. A 19 log files 5 4 logging lt log gt A 75 low delay audio packet transmission 4 47 M MAC address definition A 2 substitution 3 15 3 20 A 3 main browser lt main gt A 87 main menu 3 6 manage conferences 4 19 manual configuration overview 2 7 manual log upload 5 6 master configuration file model number version A 4 part number substitution A 4 master configuration files details A 2 overview 2 5 message waiting indication 4 6 message waiting indicator lt mwi gt A 102 messaging lt msg gt A 101 Microbrowser 4 30 Microbrowser lt mb gt A 86 microphone mute 4 13 Microsoft Live Communications Server 2005 Integration 4 43 migration dependencies C 8 miscellaneous patterns A 32 missed call configuration lt serverMissedCall gt A 94 missed call notification 4 4 model number substitution A 4 modifying network configuration 3 5 multilingual lt ml gt A 20 multilingual user interface 4 29 multiple call appearances 4 25 multiple line keys per registration 4 24 multiple registrations 4 37 N Network Address Translation lt nat gt A 102 network configuration modifying 3 5 Index 4 network monitoring lt netMon gt A 53 no answer lt noanswer gt A 97 O Option 66 3 7 outbound proxy lt outboundProxy gt A 13 P packet error concealment 4 48 password lt pwd gt A 80 patterns lt pat gt A 29 patterns lt pattern gt A 73 per phone configuration attendant A 103 automatic off hook call plac
97. A should be used Polycom recommends that you test the new configuration files on two phones before initializing all phones This should detect any errors including IP address conflicts This flash attributes are defined as follows Name Possible Values Description device set Oor1 If set to 0 do not use any device xxx yyy fields to default 0 set any parameters Set this to O after the initial installation If set to 1 use the device xxx yyy fields that have device xxx yyy set 1 Set this to 1 for the initial installation only device xxx yyy set Oor1 If set to 0 do not use the device xxx yyy value default 0 If set to 1 use the device xxx yyy value For example if device net ipAddress set 1 then use the contents of the device net ipAddress field device net ipAddress dotted decimal IP address Phone s IP address Note This field is not used when DHCP client is enabled device net subnetMask dotted decimal IP address Phone s subnet mask Note This field is not used when DHCP client is enabled device net IPgateway dotted decimal IP address Phone s default router IP gateway Note This field is not used when DHCP client is enabled device net vianld Null 0 to 4094 Phone s 802 1Q VLAN identifier Note Null no VLAN tagging device net cdpEnabled Oor1 If set to 1 the phone will attempt to determine its VLAN ID through the CDP device dhcp enabled O
98. D can be configured as a per registration feature Incoming calls received while DND is enabled are logged as missed For more information on forwarding calls while DND is enabled refer to Call Forward on page 4 20 Server based DND is active if the feature is enabled on both the phone and the server and the phone is registered The server based DND feature is applicable for all registrations on the phone no per registration mode and it disables local Call Forward and DND features Server based DND will behave the same as per SIP2 1 per registration feature with the following exceptions e There is no indication on the phone s user interface whether or not server based DND is active e If server based DND is enabled but inactive and the user presses the DND key or selects the DND option on the Feature menu the Do Not Disturb message does not appear on the user s phone incoming call alerting will continue Server based DND is disabled if Shared Call Appearance or Bridged Line Appearance is enabled Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Enable or disable server based DND sip cfg For more information refer to SIP lt SIP gt on page A 10 Specify whether or not DND results in incoming calls being given busy treatment e For more information refer to Call Handling Configuration lt call gt on page A 58 Configuration f
99. EE EEEE EE EEEE EEEE EEEE EE EEEE EEEE EEE EE EEEE Ed EREET EE EEEE EEEE EEE EEE EE EEE EEE EEEE E EE E EE EE E EE d Runnin blocks avg block max block 45 147367 6430376 23179 237 47391 221 alloc status bytes blocks avg block max block current free 6631528 45 147367 6430376 alloc 5498224 23179 237 cumulative alloc 10503508 47342 221 HHAHHSHHHHHHHHHHHHHHHHHHHHHHAAAASHHH HHH HHH HHS EERE E EE EE EEEE EEE EEEE EEEE EE EEE EE EEE EEE H HH HH HHS Running showCpuLoad Cpu load is 0 0 EEEE EEEE EEEE EEEE EEEE EEEE EEEE EE EEE EEEE EEEE EE Ea EEEE E E E EEEE E EE EE EEEE EEEE EE EEE EE EEE EEE EE EEE EE d Running memShow and the average is 88 5 g shov pulgad St Pnn If you want to look at the log files without having to wait for the phone to upload them which could take as long as 24 hours or more initiate an upload by pressing correct combination of keys on the phone a PN tr aaan g Troubleshooting Your SoundPoint IP SoundStation IP Phones For more information refer to Multiple Key Combinations on page C 9 When the log files are manually uploaded the word now is inserted into the name of the file for example 0004 200360b now boot log Reading a Boot Log The following figure shows a portion of a boot log file 0223214053 so ie Initial log entry 0223214053 so 4 00 4 0223218083 wdos lalol Note that bootrom log times are in GMT 022321405alefa_ 14 00 In_ 022321405 34 00 Init
100. F 8 encoded Null Predefined filter string string If set to Null or invalid objectclass person is used dir corp attribute x name UTF 8 encoded Null The name of the attribute to match string on the server Each name must be unique however an LDAP entry can have multiple attributes with the same name Up to eight attributes can be configured x 1 to 8 dir corp attribute x label UTF 8 encoded Null A UTF 8 encoded string that is used string as the label when data is displayed dir corp attribute x type first_name Null This parameter defines how the last_name attribute is interpreted by the phone phone_number Entries can have multiple attirbutes SIP_address of the same type Type other is URL used for display purposes only other If the user saves the entry to the local contact directory on the phone first_name last_name and phone_number are copied The user can place a cal to the phone_number and SIP_address from the corporate directory dir corp attribute x sticky 0 1 0 If set to O or Null the filter criteria for this attribute is reset after a reboot If set to 1 the filter criteria for this attribute is retained through a reboot Such attributes are denoted with a before the label when displayed on the phone dir corp attribute x filter UTF 8 encoded Null The filter string for this attribute string which is edited when searching dir corp backGroundSync 0 1 If set to
101. For the solid backgrounds set the name and RGB values For example bg gf gray pat solid 3 name Gray bg gf gray pat solid 3 red 128 bg gf gray pat solid 3 green 128 bg gf gray pat solid 3 blue 128 e Configuring Your System For the bitmaps enter a bitmap filename and put the file on the boot server For example bg gf gray 3 name polycom bmp The default size for bitmaps is 320 x 160 Smaller bitmaps will be centered and a white border added Larger bitmaps will be centered and cropped to fit Save the modified sip cfg configuration file Configuration changes can performed centrally at the boot server Central boot server Configuration file phonet cfg Specify which backgrounds will be displayed For more information refer to Backgrounds lt bg gt on page A 70 Automatic Off Hook Call Placement The phone supports an optional automatic off hook call placement feature for each registration Configuration changes can performed centrally at the boot server Central boot server Configuration file phonet cfg Specify which registrations have the feature and what contact to call when going off hook For more information refer to Automatic Off Hook Call Placement lt autoOffHook gt on page A 94 Call Hold The purpose of hold is to pause activity on one call so that the user may use the phone for another task such as to make or receive another ca
102. Hudson tjh cryptsoft com Original SSLeay License Copyright C 1995 1998 Eric Young eay cry ptsoft com All rights reserved This package is an SSL implementation written by Eric Young eay cryptsoft com The implementation was written so as to conform with Netscape s SSL This library is free for commercial and non commercial use as long as the following conditions are adhered to The following conditions apply to all code found in this distribution be it the RC4 RSA lhash DES etc code not just the SSL code The SSL documentation included with this distribution is covered by the same copyright terms except that the holder is Tim Hudson tjh cryptsoft com Copyright remains Eric Young s and as such any Copyright notices in the code are not to be removed If this package is used in a product Eric Young should be given attribution as the author of the parts of the library used This can be in the form of a textual message at program startup or in documentation online or textual provided with the package Redistribution and use in source and binary forms with or without modification are permitted provided that the following conditions are met 1 Redistributions of source code must retain the copyright notice this list of conditions and the following disclaimer 2 Redistributions in binary form must reproduce the above copyright notice this list of conditions and the following disclaimer in the documentation Third Pa
103. IP SoundStation IP Attribute Default voice rxEq hs IP_650 postFilter enable 0 voice rxEq hd IP_330 preFilter enable 0 voice rxEq hd IP_430 preFilter enable 0 voice rxEq hd IP_500 preFilter enable 0 voice rxEq hd IP_600 preFilter enable 0 voice rxEq hd IP_601 preFilter enable 0 voice rxEq hd IP_650 preFilter enable 1 voice rxEq hd IP_330 postFilter enable 0 voice rxEq hd IP_430 postFilter enable 0 voice rxEq hd IP_500 postFilter enable 0 voice rxEq hd IP_600 postFilter enable 0 voice rxEq hd IP_601 postFilter enable 0 voice rxEq hd IP_650 postFilter enable 0 voice rxEq hf IP_330 preFilter enable 1 voice rxEq hf IP_430 preFilter enable 1 voice rxEq hf IP_500 preFilter enable 1 voice rxEq hf IP_600 preFilter enable 1 voice rxEq hf IP_601 preFilter enable 1 voice rxEq hf IP_650 preFilter enable 1 voice rxEq hf IP_4000 preFilter enable 0 voice rxEq hf IP_330 postFilter enable 0 voice rxEq hf IP_430 postFilter enable 0 voice rxEq hf IP_500 postFilter enable 1 voice rxEq hf IP_600 postFilter enable 1 voice rxEq hf IP_601 postFilter enable 1 voice rxEq hf IP_650 postFilter enable 0 voice rxEq hf IP_4000 postFilter enable 0 Transmit Equalization lt txEq gt Configuration Files These settings control the performance of the hands free transmit equalization feature Polycom recommends that you do not change these values POLYCOM Attri
104. Ifthe secondary is configured for DHCP use the primary s boot server if the primary is configured for DHCP e Ifthe secondary is not configured for DHCP use the secondary s static boot server if it exists e Ifthe secondary s static boot server does not exists use the primary s boot server ignoring the source Configuration File Changes The following sip cfg configuration file changes were made to support the SoundStation IP 7000 conference phone e SDP lt SDP gt e User Preferences lt up gt e Sampled Audio for Sound Effects lt saf gt e Voice Coding Algorithms lt codecs gt Administrator s Guide Addendum for the SoundStation IP 7000 SDP lt SDP gt e Gains lt gain gt e Receive Equalization lt rxEq gt Transmit Equalization lt txEq gt e Call Handling Configuration lt call gt e Bitmaps lt bitmap gt e Feature lt feature gt This new configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SDP useLegacyPayloadNeg 0 1 Null If set to 1 the phone transmits and receives otiation RTP using the payload type identified by the first codec listed in the SDP of the codec negotiation answer If set to 0 or Null RFC 3264 is followed for transmit and receive RTP payload type values User Preferences lt up gt These configuration attributes are defined as follows Permitted Attribute Values Def
105. L with a user name this will be ignored Server Password any string The password used when the phone logs in to the server if required for the selected Server Type Note If the Server Address is a URL with user name and password this will be ignored File Transmit Tries 1 to 10 Default 3 The number of attempts to transfer a file An attempt is defined as trying to download the file from all IP addresses that map to a particular domain name Retry Wait 0 to 300 Default 1 The minimum amount of time that must elapse before retrying a file transfer in seconds The time is measured from the start of a transfer attempt which is defined as the set of upload download transactions made with the IP addresses that map to a given boot server s DNS host name If the set of transactions in an attempt is equal to or greater than the Retry Wait value then there will be no further delay before the next attempt is started For more information refer to Deploying Phones From the Boot Server on page 3 15 Provisioning Method Default or SAS VP If SAS VP is selected provisioning is done in addition to the normal process Network Cable DSL LAN Dial up The network environment the phone is operating in The default value is Cable DSL Tag SN to UA Disabled Enabled If enabled the phone s serial number MAC address is included in the User Agent header of the Microbrowser The default v
106. LWLHYLLWOLUUMG wav GALLU VAU 4 00 SoToneC Failed to find tone SoundPointIPWelcome wav using default 4 oolke Registration failed User 2125551212 Error Code 404 Not Fo Current logging level 4 Troubleshooting Your SoundPoint IP SoundStation IP Phones Power and Startup Symptom Problem Corrective Action There are power issues The SoundPoint IP Do one of the following SoundStation IP family SIP e Verify that no lights appear on the unit phone has no power when it is powered up e Check ifthe phone is properly plugged into a functional AC outlet e Make sure that the phone isn t plugged into a plug controlled by a light switch that is off f plugged into a power strip try plugging directly into a wall outlet instead e Try the phone in another room where the electricity is known to be working on a particular outlet Ifusing PoE the power supply voltage may be too high or too low Administrator s Guide SoundPoint IP SoundStation IP Controls Symptom Problem Corrective Action The dial pad does not work The dial pad on the SoundPoint IP SoundStation IP family SIP phone does not respond Do one of the following Check for a response from other feature keys or from the dial pad Place a call to the phone from a known working telephone Check for display updates Press the Menu key followed by System Status and Server Status to
107. N cable is not properly connected Do one of the following Check termination at the switch or hub furthest end of the cable from the phone Check that the switch or hub is operational flashing link status lights or contact your system administrator Press Menu followed by Status gt Network Scroll down to verify that the LAN is active Ping phone from another machine Reboot the phone to attempt re to the call server refer to Rebooting the Phone on page C 9 Administrator s Guide SoundPoint IP SoundStation IP Calling Symptom Problem Corrective Action There is no dial tone Power is not correctly applied to the SoundPoint IP family SIP phone Do one of the following Check that the display is illuminated Make sure the LAN cable is inserted properly at the rear of the phone try unplugging and re inserting the cable If using in line powering have your system administrator check that the switch is supplying power to the phone Dial tone is not present on one of audio paths Do one of the following Switch between Handset Headset if present or Hands Free Speakerphone to see if dial tone is present on another paths If dial tone exists on another path connect a different handset or headset to isolate the problem Check configuration for gain levels The phone is not registered Contact your system administrator The phone does not ring Ring
108. OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE Except as contained in this notice the name of a copyright holder shall not be used in advertising or otherwise to promote the sale use or other dealings in this Software without prior written authorization of the copyright holder Third Party Software Administrator s Guide SoundPoint IP SoundStation IP Index Numerics 802 10 VLAN header 4 52 A ACD See also automatic call distribution acoustic echo cancellation 4 49 acoustic echo cancellation lt aec gt A 40 acoustic echo suppression lt aes gt A 41 AEC See also acoustic echo cancellation AGC See also automatic gain control alert information A 14 animations lt anim gt A 72 application configuration acoustic echo cancellation A 40 acoustic echo suppression A 41 animations A 72 automatic gain control A 43 background noise suppression A 42 bitmaps A 71 call handling configuration A 58 call progress patterns A 30 chord sets A 26 codec preferences A 35 codec profiles A 36 conference setup A 15 date and time A 23 dial plan A 16 dial plan emergency A 19 directory A 61 dual tone multi frequency A 25 encryption A 80 Ethernet call control A 50 event logging A 75 feature A 83 finder A 85 fonts A 66 gains A 37 graphic icons A 74 hold local reminder A 61 idle display A 86 indicator classes A 73 indicator patterns A 73 indicators assignments A 73 IP TOS call control A 52
109. P 7000 platform voice codecPref IP_7000 G711A 7 Interpretation as above voice codecPref IP_7000 G722 4 voice codecPref IP_7000 G7221 16kbps Null voice codecPref IP_7000 G7221 24kbps Null voice codecPref IP_7000 G7221 32kbps 5 voice codecPref IP_7000 G7221C 24kbp Null s voice codecPref IP_7000 G7221C 32kbp Null s voice codecPref IP_7000 G7221C 48kbp 2 s voice codecPref IP_7000 G729AB 8 voice codecPref IP_7000 Lin16 16ksps Null voice codecPref IP_7000 Lin16 32ksps Null voice codecPref IP_7000 Lin16 48ksps Null voice codecPref IP_7000 Siren22 32kbp Null s voice codecPref IP_7000 Siren22 48kbp Null s voice codecPref IP_7000 Siren22 64kbp 1 s voice codecPref IP_7000 Siren14 24kbp Null s voice codecPref IP_7000 Siren14 32kbp Null s voice codecPref IP_7000 Siren14 48kbp 3 S Note Codecs with a default of Null are available for test purposes only and are not expected to be used in your deployment Administrator s Guide Addendum for the SoundStation IP 7000 Codec Profiles lt audioProfile gt The profile attributes can be adjusted for each of the new supported codecs namely x G7221C Lin16 Siren14 and Siren22 Permitted Attribute Values Interpretation voice audioProfile x payloadSize 10 20 30 80 Preferred Tx payload size in milliseconds to be provided in SDP offers and used in the absence of ptime negotiations This is also the range of supported Rx payload sizes The payload si
110. Server y address p arameter 65535 in Registration lt reg gt on page A 89 is non Null all of the reg x server y xxx parameters will override the volpProt server parameters If port is O or Null If volpProt server x address isa hostname and volpProt server x transport is set to DNSnaptr do NAPTR then SRV lookups If volpProt server x transport is set to TCPpreferred or UDPOnly then use 5060 and don t advertise the port number in signalling If volpProt server x address is an IP address there is no DNS lookup and 5060 is used for the port but it is not advertised in signaling If port is 1 to 65535 This value is used and it is advertised in signaling Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation volpProt server x transport DNSnaptr or DNSnapt If set to Null or DNSnaptr TCPpreferre r If volpProt server x addressisa dor hostname and volpProt server x port is 0 or UDPOnly or Null do NAPTR then SRV look ups to try to TLS or discover the transport ports and servers as TCPOnly per RFC 3263 If volpProt server x address is an IP address or a port is given then UDP is used If set to TCPpreferred TCP is the preferred transport UDP is used if TCP fails If set to UDPOnly Only UDP will be used If set to TLS If TLS fails transport fails Leave port field empty will default to 5061 or set to 5061 If se
111. SoundPoint IP SoundStation IP Status Menu Log Files Blinking Time If the phone has not been able to contact the SNTP server or if one has not been configured the date time display will flash until this is fixed If an SNTP is not available the data time display can be turned off so that the flashing display is not a distraction Debugging of single phone may be possible through an examination of the phone s status menu Press Menu select Status and then press the Select soft key Under the Platform selection you can get details on the phone s serial number or MAC address the current IP address the bootROM version the application version the name of the configuration files in use and the address of the boot server In the Network menu the phone will provide information about TCP IP setting Ethernet port speed connectivity status of the PC port and statistics on packets sent and received since last boot This would also be a good place to look and see how long it s been since the phone rebooted The Call Statistics screen shows packets sent and received on the last call The Lines menu will give you details about the status of each line that has been configured on the phone Finally the Diagnostics menu offers a series of hardware tests to verify correct operation of the microphone speaker handset and third party headset if present It will also let you test that each of the keys on the phone is working and
112. SoundPoint IP 330 switching text entry mode 3 7 SoundPoint IP 650 playback 4 35 A 83 recording 4 35 A 83 SoundPoint IP desktop phones 1 2 features list of 1 5 SoundStation IP conference phone 1 4 SoundStation IP conference phones features list of 1 5 speed dial 4 13 status menu 5 4 T text entry mode switching 3 7 time and date display 4 14 time synchronization A 53 transmit equalization lt txEq gt A 45 Index 5 Administrator s Guide SoundPoint IP SoundStation IP troubleshooting Application is not compatible 5 2 application error messages 5 3 application logging options 5 5 audio issues 5 14 blinking time 5 4 boot failure messages 5 7 bootROM error messages 5 2 calling issues 5 12 Config file error Error is 5 3 controls issues 5 10 Could not contact boot server 5 2 displays issues 5 13 Error loading 5 3 Error application is not present 5 2 Failed to get boot parameters via DHCP 5 2 log files 5 4 manual log upload 5 6 Network link is down 5 3 Not all configuration files were present 5 3 power and startup issues 5 9 reading a boot log 5 7 reading an application log 5 8 registration status 5 3 scheduled logging 5 6 screens and systems access issues 5 11 trusted certificate authority list C 1 type of service bits 4 51 U uaCSTA A 12 A 90 B 9 upgrading SIP application 3 17 USB device 4 35 user interface soft key activated 4 13 user preferences lt up gt A 23 V VAD See also voice activi
113. SoundStation IP Application Note Warning Note Download the master configuration file This file is either called lt MAC address gt cfg or 000000000000 cfg This file is used by the both the bootROM and the application for a list of other files that are needed for the operation of the phone Examine the master configuration file for the name of the application file and then look for this file on the boot server If the copy on the boot server is different than the one stored in flash memory or if there is no file stored in flash memory the application file is downloaded If the Application is any SIP version prior to 1 5 the bootROM will also download all the configuration files that are listed in the master configuration file 7 Extract the application from flash memory Install the application into RAM then upload a log file with events from the boot cycle The bootROM will then terminate and the application takes over The application manages the VoIP stack the digital signal processor DSP the user interface and the network interaction The application managed everything to do with the phone s operation The application is a single file binary image and as of SIP 1 5 contains a digital signature to prevent tampering or loading or rogue software images If your phones are using bootROM 3 0 or later the application must be signed All SIP 1 5 applications and later are signed but later patched versi
114. System DNS queries DNS Alternate Server dotted decimal IP address Secondary server to which the phone directs Domain Name System queries DNS Domain domain name string Phone s DNS domain Ethernet Refer to Ethernet Menu on page 3 11 EM Power Enabled Disabled This parameter is relevant if the phone gets Power over Ethernet PoE If enabled the phone will set power requirements in CDP to 12W so that up to three Expansion Modules EM can be powered If disabled the phone will set power requirements in CDP to 5W which means no Expansion Modules can be powered it will not work Syslog Refer to Syslog Menu on page 3 12 Setting up Your System Note A parameter value of indicates that the parameter has not yet been set and saved in the phone s configuration Any such parameter should have its value set before continuing The EM Power parameter is only available on SoundPoint IP 601 and 650 phones Note To switch the text entry mode on the SoundPoint IP 330 320 press the You may want to use URL or IP address modes when entering server addresses DHCP Menu The DHCP menu is accessible only when the DHCP client is enabled The following DHCP configuration parameters can be modified on the DHCP menu Possible Name Values Description Timeout 1 through 600 Number of seconds the phone waits for secondary DHCP Offer messages before selecting an offe
115. The IP_7000 parameters affect the SoundStation IP 7000 conference phone Animations lt anim gt The IP_7000 parameters affect the SoundStation IP 7000 conference phone Graphic Icons lt gi gt The IP_7000 parameters affect the SoundStation IP 7000 conference phone Feature lt feature gt The call list feature cannot be disabled on the SoundStation IP 7000 phone The manage conference feature is always enabled on the SoundStation IP 7000 phone Multiple Key Combinations and Default Key Layout The multiple key combinations on the SoundStation IP 7000 conference phone are as follows e Rebooting the Phone Press and hold Volume and Volumet simultaneously until a confirmation tone is heard or for about three seconds Administrator s Guide Addendum for the SoundStation IP 7000 e Restoring Factory Defaults Press and hold 1 3 5 and 7 dial pad keys simultaneously during the countdown process in the bootROM until the password prompt appears Enter the administrator password to initiate the reset Resetting to factory defaults will also reset the administrator password factory default password is 456 e Uploading Log Files Press and hold Up Down Left and Right arrow keys simultaneously until a confirmation tone is heard or for about three seconds The SoundStation IP 7000 conference phone default key layout is as follows GZ O amp A N A
116. Wha ame yWhta eine a 4 3 Called Party Identification usunn usune eee eee 4 4 Calling Party Identification 0 0 0 eee 4 4 Missed Call Notification usnu cece eee eee 4 4 Connected Party Identification 0 0 eee eee eee 4 5 Context Sensitive Volume Control 0 eee ee eee eee 4 5 Customizable Audio Sound Effects 0 00 cece eee eee 4 5 Message Waiting Indication 0 00 4 6 Distinctive Incoming Call Treatment 0 0 e eee ee 4 6 Distinctive Ringing zors ncnion ee en eee Meee ea ee Te 4 7 Distinctive Call Waiting 00 eee eee eee eee 4 7 Do Not Disturb ssie nei ieee Fen AAS eet bed atacand bse Sb a E 4 8 Handset Headset and Speakerphone 05 4 9 Local Contact Directory s 0 4 9 Local Digit Map ieie e ohare NN Geek eee wis Glades eee eo 4 12 Microphone Mute 0 aie cee eee eens 4 13 Soft Key Activated User Interface 0000000 4 13 Speed Dial siese wth eed ne kM eV EES eo EEE ERE 4 13 Time and Date Display 00 nren eee eee 4 14 Idle Display Animation 0 0 eee eee eee eee eee 4 15 Ethernet Switeh is ores seat E En E EEEIEE EEA 4 15 Graphic Display Backgrounds 0000000 4 16 Automatic Off Hook Call Placement 0000005 4 17 Call Holds i ous ee Gh dee tae ARES Nae Mi eee SOROS 4 17 Call Transfer co gs teri dena God
117. Yes 483 Too Many Hops No 484 Address Incomplete Yes 485 Ambiguous No 486 Busy Here Yes 487 Request Terminated Yes 488 Not Acceptable Here Yes 491 Request Pending No 493 Undecipherable No 5xx Responses Server Failure Response Supported Notes 500 Server Internal Error Yes 501 Not Implemented Yes 502 Bad Gateway No 503 Service Unavailable No 504 Server Time out No 505 Version Not Supported No 513 Message Too Large No 6xx Responses Global Failure Response Supported Notes 600 Busy Everywhere No 603 Decline Yes 604 Does Not Exist Anywhere No 606 Not Acceptable No Session Initiation Protocol SIP Hold Implementation Note The phone supports both currently accepted means of signaling hold The first method no longer recommended due in part to the RTCP problems associated with it is to set the c destination addresses for the media streams in the SDP to zero for example c 0 0 0 0 The second and preferred method is to signal the media directions with the a SDP media attributes sendonly recvonly inactive or sendrecv The hold signaling method used by the phone is configurable refer to SIP lt SIP gt on page A 10 but both methods are supported when signaled by the remote end point Even if the phone is set to use c 0 0 0 0 it will not do so if it gets any sendrecv sendonly or inactive from the server These fl
118. Za isshown outline Once the phone is registered the icon mis shown solid Audio Codecs The following table summarizes the SoundStation IP 6000 phone s audio codec support Effective Sample audio Algorithm MIME Type Ref Bit Rate Rate Frame Size bandwidth G 722 1C G7221 G7221C 24 Kbps 32 Ksps 20ms 80ms 14 KHz 32000 32 Kbps 48 Kbps Siren14 SIREN14 SIREN14 24 Kbps 32 Ksps 20ms 80ms 14 KHz 16000 32 Kbps 48 Kbps Administrator s Guide Addendum for the SoundStation IP 6000 Note The network bandwidth necessary to send the encoded voice is typically 5 10 higher than the encoded bit rate due to packetization overhead For example a G722 1C call at 48kbps consumes 5xkbps of network bandwidth one way audio Two way audio would take over 100kbps Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify codec priority preferred payload sizes and jitter buffer tuning boot server sip cfg parameters For more information refer to Codec Preferences lt codecPref gt on page 1 4 Voice Quality Monitoring Voice Quality Monitoring is not supported on the SoundStation IP 6000 conference phone at this time Configuration File Changes The following sip cfg configuration file changes were made to support the SoundStation IP 6000 conference phone e Sampled Audio for Sound Effects lt saf gt e Voice Coding Algorithms lt cod
119. _330 gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation font IP_330 x name fontName_height_Uxx00_U yyFF fon OR fontName_height_Uxx00_U xxFF fnt Null Defines the font file that will be loaded from boot server during boot up Note When several font IP_330 x name are defined the index x must follow consecutive increasing order Administrator s Guide SoundPoint IP SoundStation IP IP_400 font lt IP_400 gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation font IP_400 x name fontName_height_Uxx00_U yyFF fon OR fontName_height_Uxx00_U xxFF fnt Null Defines the font file that will be loaded from boot server during boot up Note When several font IP_430 x name are defined the index x must follow consecutive increasing order IP_500 font lt IP_500 gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation font IP_500 x name fontName_height_Uxx00_U yyFF fon OR fontName_height_Uxx00_U XxFF fnt Null Defines the font file that will be loaded from boot server during boot up Note When several font IP_500 x name are defined the index x must follow consecutive increasing order IP_600 font lt IP_600 gt This configuration attribut
120. a hostname and reg x outboundProxy port is 0 or Null do NAPTR then SRV look ups to try to discover the transport ports and servers as per RFC 3263 If reg x outboundProxy address is an IP address or a port is given then UDP is used If set to TCPpreferred TCP is the preferred transport UDP is used if TCP fails If set to UDPOnly Only UDP will be used If set to TLS If TLS fails transport fails Leave port field empty will default to 5061 or set to 5061 If set to TCPOnly Only TCP will be used NOTE TLS is not supported on SoundPoint IP 300 and 500 phones reg x proxyRequire string Null The string that needs to appear in the Proxy Require header If Null no Proxy Require will be sent 92 Configuration Files Attribute Permitted Values Default Interpretation reg x serverFeatureControl cf 0 1 0 If set to 1 server based call forwarding is enabled The call server has control of call forwarding If set to 0 server based call forwarding is not enabled This is the old behavior If reg x serverFeatureControl cf is not Null this attribute overrides the global server based call forwarding flag in the sip cfg configuration file reg x serverFeatureControl dnd 0 1 0 If set to 1 server based DND is enabled The call server has control of DND If set to 0 server based DND is not enabled This is the old behavior If reg x serverFeatureControl dnd
121. able to find all the necessary files So long as the files exist in flash memory the phone can boot following this error Note This error does not occur with the current BootROM Error loading lt file name gt When the required file does not exist in flash memory and cannot be found on the boot server the Error loading message will tell you which file could not be found This error only remains on the screen for a few seconds so you need to watch closely The phone reboots Note This error does not occur with the current BootROM Application Error Messages Config file error Error is lt Hex gt If there is an error in the configuration file you will not be able to reboot the phones You must review the boot server configuration make the correction and reapply the configuration file by restarting the phones Network link is down Since the SoundPoint IP SoundStation IP phones do not have an LED indicating network LINK status like many networking devices if a link failure is detected while the phone is running a message saying Network link is down will be displayed This message will be shown on the screen whenever the phone is not in the menu system and will remain on screen until the link problem is resolved Status When the phone is unable to register with the call control server the icon Za isshown outline Once the phone is able to register the icon mis shown solid Administrator s Guide
122. ad2 34 Line1 2 ArrowLeft 13 DialpadPound 24 Dialpad1 35 Line3 3 ArrowDown 14 DialpadO 25 SoftKey4 36 Redial 4 ArrowRight 15 DialpadStar 26 SoftKey3 37 Transfer 5 Select 16 Dialpadg 27 SoftKey2 38 Headset 6 Delete 17 Dialpad8 28 SoftKey1 39 Handsfree 7 Menu 18 Dialpad7 29 Services 40 Hold 8 Messages 19 Dialpad4 30 Directories 41 Line4 9 DoNotDisturb 20 Dialpad5 31 Line6 42 Lined 10 MicMute 21 Dialpad6 32 Conference 11 VolUp 22 Dialpad3 33 Line2 C 14 Miscellaneous Administrative Tasks SoundStation IP 4000 G4 ee S A AZ D O e 2 CONE ag Fy Fy Fre He NEP NO Key ID Function Key ID Function Key ID Function Key ID Function 1 Dialpad1 12 Select 23 n a 34 n a 2 Dialpad2 13 Dialpad7 24 n a 35 n a 3 Dialpad3 14 Dialpad8 25 Menu 36 n a 4 VolUp 15 Dialpad9 26 Exit 37 n a 5 Handsfree 16 MicMute 27 SoftKey1 38 n a 6 ArrowUp 17 n a 28 SoftKey2 39 n a 7 Dialpad4 18 ArrowDown 29 SoftKey3 40 n a 8 Dialpad5 19 DialpadStar 30 n a 41 n a 9 Dialpad6 20 DialpadO 31 n a 42 n a 10 VolDown 21 DialpadPound 32 n a 11 n a 22 Redial 33 n a Administrator s Guide SoundPoint IP SoundStation IP Assigning a VLAN ID Using DHCP Note To assign a VLAN ID to a phone using DHCP gt gt Inthe DHCP menu of the Main setup menu set VLAN Discovery to Fixed or Custom When set to Fixed the phone will examine DHCP options
123. ags will cause it to revert to the other hold method Reliability of Provisional Responses Transfer The phone fully supports RFC 3262 Reliability of Provisional Responses The phone supports transfer using the REFER method specified in draft ietf sip cc transfer 05 and RFC 3515 Third Party Call Control The phone supports the delayed media negotiations INVITE without SDP associated with third party call control applications When used with an appropriate server the User Agent Computer Supported Telecommunications Applications uaCSTA feature on the phone may be utilized for remote control of the phone from computer applications such as Microsoft Office Communicator The phone is compliant with Using CSTA for SIP Phone User Agents uaCSTA ECMA TR 087 for the Answer Call Hold Call and Retrieve Call functions and Services for Computer Supported Telecommunications Applications Phase III ECMA 269 for the Conference Call function This feature is enabled by configuration parameters described in SIP lt SIP gt on page A 10 and Registration lt reg gt on page A 89 and needs to be activated by a feature application key Administrator s Guide SoundPoint IP SoundStation IP SIP for Instant Messaging and Presence Leveraging Extensions e The phone is compatible with the Presence and Instant Messaging features of Microsoft Windows Messenger 5 1 Shared Call Appearance Signaling A shared line is an address
124. ain a default level of contrast refer to Rebooting the Phone on page C 9 Outbound or inbound calling is unsuccessful Do one of the following e Place a call to the phone under investigation Check that the display indicates incoming call information Lift the handset Ensure dial tone is present and place a call to another extension or number Check that the display changes in response The display is flickering Certain type of older fluorescent lighting causes the display to appear to flicker Do one of the following e Move the SoundPoint IP SoundStation IP SIP phone away from the lights e Replace the lights 13 Administrator s Guide SoundPoint IP SoundStation IP Audio Symptom Problem Corrective Action There is no audio on the headset The connections are not correct Do one of the following Ensure the headset is plugged into the jack marked Headset at the rear of the phone e Ensure the headset amplifier if present is turned on and or the volume is correctly adjusted There are audio and echo issues on the headset Possible issues include Echo on external calls through a gateway e Internal calls no gateway handsfree echo Internal calls no gateway handset to handset echo Refer to Technical Bulletin 16249 Troubleshooting Audio and Echo Issues on SoundPoint IP Phones o
125. allow recording of audio calls on a supported USB device Currently only the SoundPoint IP 650 has a USB port The filenames of the recorded wav files will include a date time stamp for example 20Apr2007_190012 wav was created on April 20 2007 at 19 00 12 An indication of the recording time remaining the space available of the attached USB storage media appears on the graphic display The user can browse through all recorded files through the menu shown on the graphic display Playback of recorded files can occur on the phone as well as on other devices such as a Windows or Apple based computer using an application like Windows Media Player or iTunes The user controls which calls are recorded and played back For more information refer to Technical Bulletin 38084 SountPoint IP 650 Supported USB Devices for Recording at http www polycom com support voip Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can be performed centrally at the boot server Central boot server Configuration file Turn this feature on or off sip cfg e For more information refer to Feature lt feature gt on page A 83 Voice Mail Integration The phone is compatible with voice mail servers The subscribe contact and callback mode can be configured per user registration on the phone The phone can be configured with a SIP URL to be called automatically by the phone when the use
126. als 914539400 the first 9 is removed when the call is placed RR604Rxxxxxxx Prepend 604 to all 7 digit numbers For example if a customer dials 4539400 604 is added to the front of the number so a call 6044539400 is placed e R9R6O4Rxxxxxxx Replaces 9 with 604 e R999R911R Convert 999 to 911 e xxR601R600Rxx When applied on 1160122 gives 1160022 e xR60xR600Rxxxxxxx Any 60x will be replaced with 600 in the middle of the dialed number that matches For example if a customer dials 16092345678 a call is placed to 16002345678 e 911xxx T A period which matches an arbitrary number including zero of occurrences of the preceding construct For example 91112 with waiting time to comply with T is a match 911123 with waiting time to comply with T is a match 9111234 with waiting time to comply with T is a match and the number can grow indefinitely given that pressing the next digit takes less than T The following guidelines should be noted e You must use only or 0 9 between second and third R e Ifadigit map does not comply it is not included in the digit plan as a valid one That is no matching is done against it Administrator s Guide SoundPoint IP SoundStation IP e There is no limitation on the number of R triplet sets in a digit map However a digit map that contains less than full number of triplet sets for example a total of 2Rs or 5Rs is considered an invalid digit map
127. alue is Disabled Provisioning String any string The URL used in XML post response transactions If empty the configured URL is used This field is disabled when Provisioning Method is Default 3 10 Note Setting up Your System The Server User and Server Password parameters should be changed from the default values Note that for insecure protocols the user chosen should have very few privileges on the server Ethernet Menu The following Ethernet configuration parameters can be modified on the Ethernet menu Name Possible Values Description CDP Enabled Disabled If enabled the phone will use CDP It also reports PoE power usage to the switch The default value is Enabled VLAN ID Null 0 through 4094 Phone s 802 1Q VLAN identifier The default value is Null Note Null no VLAN tagging VLAN Filtering Enabled Disabled Filter received Ethernet packets so that the TCP IP stack does not process bad data or too much data Enable disable the VLAN filtering state The default value is Enabled Storm Filtering Enabled Disabled Filter received Ethernet packets so that the TCP IP stack does not process bad data or too much data Enable disable the DoS storm prevention state The default value is Enabled LAN Port Mode 0 Auto The network speed over the Ethernet 1 10HD The default value is Auto 2 10FD 3 100HD 4 100FD 5 1000FD PC Port Mode Au
128. ample of Application sip Id is not compatible with this phone boot failure messages hwSigParseRemove could not find key Bad image signature Error updating application 0223220834 sig 0223220834 ctg 0223220834 cfg 0223220834 appl1 0223220904 appi1 0223220904 appi1 Application sip ld is not compatible Loaded application sip ld successful loading b T time is THU FEB 022322082S5 cfg 3 00 New load header information 022322082S5 cfg 3 00 Code length 0x0013F22D 022322082S cfg 3 00 Header check Sum 0x20147429 022322082S cfg 3 00 Code check Sum OxOA8COD28 0223220825 cfg 3 00 Options 0x00000003 0223220828 cfg 3 00 New load header information 0223220828 cfg 3 00 Code length Ox0013F1i6D 0223220828 cfg 3 00 Header check Sum 020147369 0223220828 cfg 3 00 Code check Sum OxO0A8SDE14D 0223220828 cfg 3 00 Options 0x00000003 0223220828 cfg 3 00 Using compatible image 1 0223220834 sig 4 og m 3 09 sS oq s og 4 og 4 og 6 og Administrator s Guide SoundPoint IP SoundStation IP Reading an Application Log The following figure shows a portion of an application log file 0224000058 so 00 Initial log entry 0224000058 so 00 Platform Model SoundPoint IP 500 Assembly 2345 11500 020 Rev A 0224000058 so 00 Platform MAC 0004 2015a51 IP 172 23 2 172 Subnet Mask 255 255 0 0 0224000058 so 00 Platform BootBlock 2 5 0 11500_020 20 Aug 04 16 05 0224000058 so 00
129. analog chassis IP_330 36 voice gain tx analog chassis IP_430 36 voice gain tx analog chassis IP_601 0 voice gain tx analog chassis IP_650 36 voice gain tx digital handset 0 voice gain tx digital handset IP_330 10 voice gain tx digital handset IP_430 10 voice gain tx digital handset IP_650 6 voice gain tx digital headset 0 Configuration Files Attribute Default voice gain tx digital headset IP_330 10 voice gain tx digital headset IP_430 10 voice gain tx digital headset IP_650 6 voice gain tx digital chassis 3 voice gain tx digital chassis IP_330 12 voice gain tx digital chassis IP_430 12 voice gain tx digital chassis IP_4000 0 voice gain tx digital chassis IP_601 6 voice gain tx digital chassis IP_650 12 voice gain tx analog preamp handset 14 voice gain tx analog preamp headset 23 voice gain tx analog preamp chassis 32 voice gain tx analog preamp chassis IP_601 32 voice handset rxag adjust IP_330 1 voice handset rxag adjust IP_430 1 voice handset rxag adjust IP_650 1 voice handset txag adjust IP_330 18 voice handset txag adjust IP_430 18 voice handset txag adjust IP_650 18 voice handset sidetone adjust IP_330 3 voice handset sidetone adjust IP_430 3 voice handset sidetone adjust IP_650 0 voice headset rxag adjust IP_330 4 voice headset rxag adjust IP_430 1 voice headset rxag adjust IP_650 1 voice headset txag adjust IP_330 21 voice h
130. anual Ww data entry POLYCOM The following table shows the manually entered networking parameters that may be overridden by parameters obtained from a DHCP server an alternate DHCP server or configuration file Alternate Configuration File Local Parameter DHCP Option DHCP DHCP application only FLASH gt priority when more than one source exists gt 1 2 3 4 IP address 1 d subnet mask 1 IP gateway 3 d Setting up Your System Parameter DHCP Option DHCP Alternate DHCP Configuration File application only Local FLASH boot server address Refer to DHCP Menu on page 3 7 SIP server address 151 Note This value is configurable SNTP server 42 then 4 address SNTP GMT offset 2 DNS server IP 6 address alternate DNS 6 server IP address DNS domain 15 Refer to DHCP Warning Cisco Discovery Protocol CDP overrides Local FLASH Menu on page that overrides DHCP VLAN Discovery VLAN ID 3 7 For more information on DHCP options go to http www ietf org rfc rfc2131 txt number 2131 or http www ietf org rfc rfc2132 txt number 2132 Note The configuration file value for SNTP server address and SNTP GMT offset can be configured to override the DHCP value Refer to tcpIpApp sntp address overrideDHCP in Time Synchronization lt sntp gt on page A 53 The CDP value can
131. ation call hold localReminder enabled 0 1 0 If set to 1 periodically notify the local user that calls have been on hold for an extended period of time call hold localReminder period non negative 60 Time in seconds between subsequent integer reminders call hold localReminder startDelay non negative 90 Time in seconds to wait before the integer initial reminder Directory lt dir gt This attribute includes e Local Directory lt local gt e Corporate Directory lt corp gt Administrator s Guide SoundPoint IP SoundStation IP Local Directory lt local gt The local directory is stored in either flash memory or RAM on the phone The local directory size is limited based on the amount of flash memory in the phone Different phone models have variable flash memory When the volatile storage option is enabled ensure that a properly configured boot server that allows uploads is available to store a back up copy of the directory or its contents will be lost when the phone reboots or loses power Permitted Attribute Values Default Interpretation dir local volatile 2meg 0 1 0 Attribute applies to platforms with 2 Mbytes of flash memory If set to 1 use volatile storage for phone resident copy of the directory to allow for larger size dir local nonVolatile maxSize 2meg 1 to 20 20 Attribute applies to platforms with 2 Mbytes of flash memory Maximum size in Kbytes of non volatile storage that the di
132. ation IP 7000 conference phone is provisioned over the Ethernet by the boot server However when two SoundStation IP 7000 phones are daisy chained together the one that is not directly connected to the Ethernet can still be provisioned known as the secondary The provisioning over CLink feature is automatically enabled when a SoundStation IP 7000 phone is not connected to the Ethernet Both SoundStation IP 7000 phones must be running the same version of the SIP application The steps for provisioning the secondary SoundStation IP 7000 phone are the same as for the primary SoundStation IP 7000 phone You can reboot the primary without rebooting the secondary However the primary and secondary should be rebooted together for the primary secondary relationship to be recognized If you power up both SoundStation IP 7000 phones the primary will power up first Currently provisioning over CLink is supported for the following configurations of SoundStation IP 7000 conference phones e Two SoundStation IP 7000 conference phone daisy chained together e Two SoundStation IP 7000 conference phone daisy chained together with one external microphone specifically designed for the SoundStation IP 7000 conference phone Refer to Daisy Chaining Phones on page 1 4 for an illustration of two SoundStation IP 7000 conference phone daisy chained together The provisioning boot server or proxy for the secondary is determined by the following criteria e
133. ault Interpretation up toneControl bass 4 to 4 Null 0 Bass equalization control Each step is an increment of 1 dB at 225 kHz and 2 dB lt 225 Hz up toneControl treble 4 to 4 Null 0 Treble equalization control Each step is an increment of 1 dB at 3 7 kHz and 2 dB gt 10 kHz Administrator s Guide Addendum for the SoundStation IP 7000 Permitted Attribute Values Default Interpretation up audioSetup auxInput 0 Other Null Auxiliary audio input Input If set to Null default value is 2 1 Polycom Wireless Mic 2 off up audioSetup auxOutput 0 Other Null Auxiliary audio output Input If set to Null default value is 2 1 Polycom Wireless Mic 2 off Sampled Audio for Sound Effects lt saf gt The following new sampled audio WAVE file wav formats are supported e 1L16 32000 16 bit 32 kHz sampling rate mono e 1L16 48000 16 bit 48 kHz sampling rate mono Voice Coding Algorithms lt codecs gt These new codecs include e Codec Preferences lt codecPref gt e Codec Profiles lt audioProfile gt Administrator s Guide Addendum for the SoundStation IP 7000 Codec Preferences lt codecPref gt Permitted Attribute Values Default Interpretation voice codecPref IP_7000 G711Mu Null 1 16 6 Specifies the codec preferences for the SoundStation I
134. be obtained from a connected Ethernet switch if the switch supports CDP In the case where you do not have control of your DHCP server or do not have the ability to set the DHCP options an alternate method of automatically discovering the provisioning server address is required Connecting to a secondary DHCP server that responds to DHCP INFORM queries with a requested boot server value is one possibility For more information refer to http www ietf org rfc rfc3361 txt number 3361 and http www ietf org rfc rfc3925 txt number 3925 Administrator s Guide SoundPoint IP SoundStation IP Supported Provisioning Protocols Note Note The bootROM performs the provisioning functions of downloading configuration files uploading and downloading the configuration override file and user directory and downloading the dictionary and uploading log files The protocol that will be used to transfer files from the boot server depends on several factors including the phone model and whether the bootROM or SIP application stage of provisioning is in progress By default the phones are shipped with FTP enabled as the provisioning protocol If an unsupported protocol is specified this may result in a defined behavior see the table below for details of which protocol the phone will use The Specified Protocol listed in the table can be selected in the Server Type field or the Server Address can include a transfer protocol for example http u
135. ber of buttons to 12 registrations on the IP 601 and 34 registrations on the IP 650 Each registration can optionally be associated with a private array of servers for completely segregated signaling The SoundStation IP 4000 supports a single registration Administrator s Guide SoundPoint IP SoundStation IP In the following table x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Permitted Attribute Values Default Interpretation reg x csta 0 1 Null If set to 1 uaCSTA is enabled If reg x csta is not Null this attribute overrides the global CSTA flag in the sip cfg configuration file reg x displayName UTF 8 encoded Null Display name used for local user interface as string well as SIP signaling reg x address string in the format Null The user part or the user and the host part of userPart from the phone s SIP URI userPart domain The user part of the phone s SIP URI For example reg x address 1002 from 1002 polycom com or reg x address 1002 polycom com reg x label UTF 8 encoded Null Text label to appear on the display adjacent string to the associated line key If omitted the label will be derived from the user part of reg x address reg x Ics 0 1 0 If set to 1 the Microsoft Live Communications Server is supported for registration x reg x type private OR shared private
136. ber of conference parties for the platform can exist and there is a manage conference page Note The manage conference feature is always disabled on the SoundPoint IP 301 320 330 430 501 600 601 phone Note feature 16 name nway conference feature 17 name call recording and feature 19 name corporate directory are charged for separately To activate these features you must go to the Polycom Resource Center http extranet polycom com csnprod signon html to retrieve the activation code Resource lt res gt A 84 This attribute s settings control the maximum size or an external resource retrieved at run time This attribute also includes e Finder lt finder gt e Quotas lt quotas gt Finder lt finder gt Configuration Files This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation res finder sizeLimit positive If a resource that is being downloaded to the phone integer is larger than this value 1024 bytes the maximum size the resource will be automatically truncated to the maximum size defined Note For the SoundPoint IP 650 platform this value is internally replaced by 2X the value res finder minfree 1 to 2048 A resource will not be downloaded to the phone if the amount of free memory is less than this value 1024 bytes the minimum size This parameter is used for 16MB SDRAM platforms and scaled
137. between old and new versions of configuration files are explained in the Release Notes that accompany the software Both mandatory and optional changes may present Changes to site wide configuration files such as sip cfg can be done manually but a scripting tool is useful to change per phone configuration files The configuration files listed in CONFIG_FILES attribute of the master configuration file must be updated when the software is updated Any new configuration files must be added to the CONFIG_FILES attribute in the appropriate order Mandatory changes must be made or the software may not behave as expected For more information refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice 3 Save the new configuration files and images such as sip ld on the boot server 4 Reboot the phones by pressing the reboot multiple key combination For more information refer to Multiple Key Combinations on page C 9 Since the APPLICATION APP_FILE_PATH attribute of the lt Ethernet address gt cfg files references the individual sip Id files it is possible to verify that an update is applied to phones of a particular model For example the reference to sip ld is changed to 2345 11605 001 sip Id to boot the SoundPoint IP 601 image The phones can be rebooted remotely through the SIP signaling protocol Refer to Special Events lt specialEvent gt on page A 15 Setting up Your
138. bute Default voice txEq hs IP_330 preFilter enable voice txEq hs IP_430 preFilter enable voice txEq hs IP_500 preFilter enable voice txEq hs IP_600 preFilter enable voice txEq hs IP_601 preFilter enable oO o oO Oo voice txEq hs IP_650 preFilter enable voice txEq hs IP_330 postFilter enable voice txEq hs IP_430 postFilter enable voice txEq hs IP_500 postFilter enable voice txEq hs IP_600 postFilter enable voice txEq hs IP_601 postFilter enable voice txEq hs IP_650 postFilter enable voice txEq hd IP_330 preFilter enable voice txEq hd IP_430 preFilter enable voice txEq hd IP_500 preFilter enable voice txEq hd IP_600 preFilter enable voice txEq hd IP_601 preFilter enable oOo o o o o voice txEq hd IP_650 preFilter enable voice txEq hd IP_330 postFilter enable voice txEq hd IP_430 postFilter enable voice txEq hd IP_500 postFilter enable voice txEq hd IP_600 postFilter enable voice txEq hd IP_601 postFilter enable voice txEq hd IP_650 postFilter enable oO o o o oao o 45 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice txEq hf IP_330 preFilter enable 0 voice txEq hf IP_430 preFilter enable 0 voice txEq hf IP_500 preFilter enable 0 voice txEq hf IP_600 preFilter enable 0 voice txEq hf IP_601 preFilter enable 0 voi
139. c Profiles lt audioProfile gt on page A 36 Local Web Server Specify codec priority preferred payload sizes and jitter buffer tuning if enabled parameters Navigate to http lt phonelPAddress gt coreConf htm au Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server 50 Configuring Your System Background Noise Suppression Comfort Noise Fill Background noise suppression BNS is designed primarily for hands free operation and reduces background noise to enhance communication in noisy environments There are no related configuration changes Comfort noise fill is designed to help provide a consistent noise level to the remote user of a hands free call Fluctuations in perceived background noise levels are an undesirable side effect of the non linear component of most AEC systems This feature uses noise synthesis techniques to smooth out the noise level in the direction toward the remote user providing a more natural call experience There are no related configuration changes Automatic Gain Control Automatic Gain Control AGC is applicable to hands free operation and is used to boost the transmit gain of the local talker in certain circumstances This increases the ef
140. can occur on SounaPoint IP 430 501 550 560 600 601 and 650 and SoundStation IP 4000 phones Configuration Files Attribute Permitted Values Default Interpretation mb idleDisplay refresh 0 or an integer gt 5 0 The period in seconds between refreshes of the idle display Microbrowser s content If set to 0 the idle display Microbrowser is not refreshed The minimum refresh period is 5 seconds values from 1 to 4 are ignored and 5 is used Note If an HTTP Refresh header is detected it will be respected even if this parameter is set to 0 The refresh parameter will be respected only in the event that a refresh fails Once a refresh is successful the value in the HTTP refresh header if available will be used Main Browser lt main gt This setting controls the home page used by the Microbrowser when that function is selected Attribute Permitted Values Default Interpretation mb main home Any fully formed valid Null URL used for Microbrowser home page If blank HTTP URL Length the browser will notify the user that a blank up to 255 characters home page was used For example http www example com xhtml frontpage cgi pa ge home mb main statusbar 0 1 Null Flag to determine whether or not to turn off display of status messages If set to 1 the display of the status bar is enabled If set to 0 or Null the display of the status bar is disabled mb mai
141. cation of the IP 330 0 101 upper left corner of the indictor measured in pixels from left to IP 400 0 122 none IP 500 0 159 IP 600 0 319 IP 4000 0 247 A 74 Configuration Files Attribute Permitted Values Interpretation ind gi x y physY IP 300 0 3 IP 330 0 19 IP 400 0 45 IP 500 0 79 IP 600 0 159 IP 4000 0 67 For Graphic Icon type indicators this is the y axis location of the upper left corner of the indicator measured in pixels from top to bottom ind gi x y physW IP 300 n a IP 330 1 87 IP 400 1 102 IP 500 1 160 IP 600 1 320 IP 4000 1 248 For Graphic Icon type indicators this is the width of the indicator measured in pixels ind gi x y physH IP 300 n a IP 330 1 20 IP 400 1 23 IP 500 1 80 IP 600 1 160 IP 4000 1 68 For Graphic Icon type indicators this is the height of the indicator measured in pixels Event Logging lt log gt Logging parameter changes can impair system operation Do not change any logging parameters without prior consultation with Polycom Technical Support The event logging system supports the following classes of events Level 0 O a A WwW N Interpretation Debug only High detail event class Moderate detail event class Low detail event class Minor error graceful recovery Major error will eventually incapacitate the system Fatal error Administrator s Guide SoundPoint IP
142. ce qualityMonitoring collector alert dela Null 10 to Null Threshold value of one way delay y threshold critical 2000 in ms that causes phone to senda critical alert quality report If set to Null critical alerts are not generated due to one way delay One way delay includes both network delay and end system delay Server lt server gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation voice qualityMonitoring collector server x a Dotted decima Null IP address or host name and port of ddress IP address or a SIP server report collector that host name accepts voice quality reports contained in SIP PUBLISH messages Set x to 1as only one report collector is supported at this time voice qualityMonitoring collector server x p 0 Null 1 to 5060 If port is O or Null port 5060 will be ort 65535 used Set x to 1as only one report collector is supported at this time RTCP XR lt rtcpxr gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation voice qualityMonitoring enable rtcpxr 0 1 0 Enables generation of RTCP XR packets Administrator s Guide SoundPoint IP SoundStation IP Quality of Service lt QOS gt These settings control the Quality of Service QOS options This attribute includes e Ethernet IEEE 802 1p Q lt ethernet gt e IP TOS lt
143. ce txEq hf IP_650 preFilter enable 1 voice txEq hf IP_4000 preFilter enable 0 voice txEq hf IP_330 postFilter enable 1 voice txEq hf IP_430 postFilter enable 1 voice txEq hf IP_500 postFilter enable 1 voice txEq hf IP_600 postFilter enable 1 voice txEq hf IP_601 postFilter enable 1 voice txEq hf IP_650 postFilter enable 1 voice txEq hf IP_4000 postFilter enable 0 Voice Activity Detection lt vad gt These settings control the performance of the voice activity detection silence suppression feature Permitted Attribute Values Default Interpretation voice vadEnable 0 1 0 If set to 1 enable VAD voice vadThresh integer from 0 15 The threshold for determining what is active voice and to 30 what is background noise in dB This does not apply to G 729AB codec operation which has its own built in VAD function Configuration Files Permitted Attribute Values Default Interpretation voice vad signalAnnex 0 1 Null If set to 1 and voice vadEnable is set to 1 Annex B is B used A new line can be added to SDP depending on the setting of this parameter and the voice vadEnable parameter If voice vadEnable is set to 1 add attribute line a fmtp 18 annexb yes below a rtpmap attribute line where 18 could be replaced by another payload e Ifvoice vadEnable is set to 0 add attribute line a fmtp 18 annexb no below a rtpmap attribute line where 18 could be replaced by anothe
144. ces lt up gt e Tones lt tones gt e Sampled Audio for Sound Effects lt saf gt e Sound Effects lt se gt e Voice Settings lt voice gt e Quality of Service lt QOS gt e Basic TCP IP lt TCP_IP gt e Web Server lt httpd gt e Call Handling Configuration lt call gt e Directory lt dir gt e Presence lt pres gt e Fonts lt font gt e Keys lt key gt e Backgrounds lt bg gt e Bitmaps lt bitmap gt e Indicators lt ind gt e Event Logging lt log gt e Security lt sec gt e License lt license gt e Provisioning lt prov gt e RAM Disk lt ramdisk gt e Request lt request gt e Feature lt feature gt Administrator s Guide SoundPoint IP SoundStation IP e Resource lt res gt e Microbrowser lt mb gt Polycom recommends that you create another file with your organization s wy modifications If you must change any Polycom templates back them up first POLYCOM For more information refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice Protocol lt volpProt gt This attribute includes e Local lt local gt e Server lt server gt e SDP lt SDP gt e SIP lt SIP gt Local lt local gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt local port 0 to 65535 5060 Local port for sending and receiving SIP signaling packe
145. cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User Specify per registration the number of calls that can be active or on hold per line key assigned to that registration using the SIP Configuration menu Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations not a mixture of these options When the SIP Configuration menu is used it is assumed that all registrations use the same server Shared Call Appearances Calls and lines on multiple phones can be logically related to each other A call that is active on one phone will be presented visually to phones that share that call appearance Mutual exclusion features emulate traditional PBX or key system privacy for shared calls Incoming calls can be presented to multiple phones simultaneously Users at the different locations have the ability to interrupt remote active calls This feature is dependent on support from a SIP server that binds the appearances together logically and looks after the necessary state notifications and performs an access control function For more information refer to Shared Call Appearance Signaling on page B 10 Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can pe
146. cy lt DTMF gt on page A 25 DTMF Event RTP Payload The phone is compatible with RFC 2833 RTP Payload for DTMF Digits Telephony Tones and Telephony Signals RFC 2833 describes a standard RTP compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream The phone generates RFC 2833 DTMF only events but does not regenerate nor otherwise use DTMF events received from the remote end of the call Configuration changes can performed centrally at the boot server Central boot server Configuration file Enable or disable RFC 2833 support in SDP offers and specify the payload value to use in SDP offers For more information refer to Dual Tone Multi Frequency lt DTMF gt on page A 25 Acoustic Echo Cancellation The phone employs advanced acoustic echo cancellation AEC for hands free operation Both linear and non linear techniques are employed to aggressively reduce echo yet provide for natural full duplex communication patterns When using the handset on any SoundPoint IP phones AEC is not normally required In certain situations where echo is experienced by the far end party when the user is on the handset AEC may be enabled to reduce avoid this echo To achieve this make the following changes in the sip cfg configuration file default settings for these parameters are disabled voice aec hs enable 1 voice aes hs enable 1 voice ns hs enable 1 v
147. d This is the same as the previous behavior If set to 0 the phone will send a reinvite with a stream mode attribute of inactive when a call is put on hold NOTE The phone will ignore the value of this parameter if set to 1 when the parameter volpProt SIP useRFC2543hold s also set to 1 default is 0 volpProt SIP Ics 0 1 0 If set to 1 the proprietary epid parameter is added to the From field of all requests to support Microsoft Live Communications Server volpProt SIP ms forking 0 1 0 If set to 0 support for MS forking is disabled If set to 1 support for MS forking is enabled and the phone will reject all Instant Message INVITEs This parameter is relevant for Microsoft Live Communications Server server installations Note that if any end point registered to the same account has MS forking disabled all other end points default back to non forking mode Windows Messenger does not use MS forking so be aware of this behavior if one of the end points is Windows Messenger volpProt SIP dialog usePvalue 0 1 0 If set to 0 phone uses pval field name in Dialog This obeys the draft ietf sipping dialog package 06 txt draft If set to 1 phone uses a field name of pvalue Configuration Files Attribute Permitted Values Default Interpretation volpProt SIP connectionReuse useAli as 0 1 0 If set to 0 this is the old behavior If set to 1 phone uses the co
148. d Indicators The phone s user interface can be customized by changing the fonts and graphic icons used on the display and the LED indicator patterns Pre existing fonts embedded in the software can be overwritten or new fonts can be downloaded The bitmaps and bitmap animations used for graphic icons on the display can be changed and repositioned LED flashing sequences and colors can be changed Configuration changes can performed centrally at the boot server Central boot Configuration File Specify fonts to overwrite existing ones or specify new fonts server sip cfg For more information refer to Fonts lt font gt on page A 66 Specify which bitmaps to use For more information refer to Bitmaps lt bitmap gt on page A 71 Specify how to create animations and LED indicator patterns For more information refer to Indicators lt ind gt on page A 71 Instant Messaging 4 28 The phone supports sending and receiving instant text messages The user is alerted to incoming messages visually and audibly The user can view the messages immediately or when it is convenient For sending messages the user can either select a message from a preset list of short messages or an alphanumeric text entry mode allows the typing of custom messages using the dial pad Message sending can be initiated by replying to an incoming Configuring Your System message or by initiating a new dialog The destination for
149. d from the boot server Local Phone User Interface The user can set the call forward all setting from the idle display enable disable and specify the forward to contact as well as divert callers while the call is alerting Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Directed Call Pick Up Calls to another phone can be picked up by dialing the extension of the other phone This feature depends on support from a SIP server Configuration changes can performed centrally at the boot server Central boot server Configuration file sip cfg Turn this feature on or off For more information refer to Feature lt feature gt on page A 83 Group Call Pick Up Calls to another phone within a pre defined group can be picked up without dialing the extension of the other phone This feature depends on support from a SIP server Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server Central Configuration file Turn this feature on or off boot server sip cfg e For more information refer to Feature lt feature gt on page A 83 Call Park Retrieve An active cal
150. d on comma separated list hardware platform y where y can be IP_500 or IP_600 To add new languages to those included with the distribution 1 2 6 Create a new dictionary file based on an existing one Change the strings making sure to encode the XML file in UTF 8 but also ensuring the UTF 8 characters chosen are within the Unicode character ranges indicated in the tables below Place the file in an appropriately named folder according to the format language_region parallel to the other dictionary files under the SoundPointIPLocalization folder on the boot server Add a 1cl ml lang clock menu x attribute to the configuration file Add lcl ml lang clock x 24HourClock lcol ml lang clock x format lcl ml lang clock x longFormat and 1lcl ml lang clock x dateTop attributes and set them according to the regional preferences Optional Set 1c1 ml lang to be the new language_region string Basic character support includes the following Unicode character ranges Name Range CO Controls and Basic Latin U 0000 U 007F C1 Controls and Latin 1 Supplement U 0080 U 00FF Cyrillic partial U 0400 U 045F Administrator s Guide SoundPoint IP SoundStation IP Extended character support available on SoundPoint IP 600 and SoundStation IP 4000 platforms includes the following Unicode character ranges Name Range CJK Symbols and Punctuation U 3000
151. dation gt e Special Events lt specialEvent gt e Conference Setup lt conference gt Configuration Files Outbound Proxy lt outboundProxy gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP outboundProxy address dotted deci Null IP address or host name and port of a SIP mal IP server to which the phone shall send all address or requests host name volpProt SIP outboundProxy port 1 to 65535 5060 volpProt SIP outboundProxy transpor DNSnaptror DNSnapt If set to Null or DNSnaptr t TCPpreferre r If volpProt SIP outboundProxy address is a dor hostname and UDPOnly or volIpProt SIP outboundProxy port is 0 or TLS or Null do NAPTR then SRV look ups to try to TCPOnly discover the transport ports and servers as per RFC 3263 If volIpProt SIP outboundProxy address is an IP address or a port is given then UDP is used If set to TCPpreferred TCP is the preferred transport UDP is used if TCP fails If set to UDPOnly Only UDP will be used If set to TLS If TLS fails transport fails Leave port field empty will default to 5061 or set to 5061 If set to TCPOnly Only TCP will be used NOTE TLS is not supported on SoundPoint IP 300 and 500 phones Administrator s Guide SoundPoint IP SoundStation IP Alert Information lt alertInfo gt This configuration attribute is defined as follows
152. defined as follows Permitted Attribute Values Default Interpretation attendant uri string Null For attendant console busy lamp field BLF feature This specifies the list SIP URI on the server If this is just a user part the URI is constructed with the server host name IP attendant reg positive 1 For attendant console BLF feature This is the index of integer the registration which will be used to send a SUBSCRIBE to the list SIP URI specified in attendant uri For example attendant reg 2 means the second registration will be used Roaming Buddies lt roaming_buddies gt Note This attribute is used in conjunction with Microsoft Live Communications Server 2005 only This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation roaming_buddies reg positive Null Specifies the line registration number which has roaming integer buddies support enabled If Null roaming buddies is disabled If value lt 1 then value is replaced with 1 Warning This parameter must be enabled value gt 0 if the call server is Microsoft Live Communications Server 2005 Roaming Privacy lt roaming_privacy gt Note This attribute is used in conjunction with Microsoft Live Communications Server 2005 only This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation roaming_privacy reg positive Nul
153. dicate that one or more messages voice mail are waiting at the message center se appLocalEnabled 0 1 1 If set to 1 local user interface sound effects such as confirmation error tones will be enabled This attribute also includes e Patterns lt pat gt e Ring type lt rt gt Patterns lt pat gt Patterns use a simple script language that allows different chord sets or wave files to be strung together with periods of silence The script language uses the following instructions Instruction Meaning Example sampled n Play sampled audio file se pat callProg x inst y type sampled sampled audio n file instruction type se pat callProg x inst y value 3 specifies sampled audio file 3 chord n d Play chord set n d is se pat callProg x inst y type chord chord set optional and allows the instruction type chord set ON duration to se pat callProg x inst y value 3 specifies call sera tod progress chord set 3 plllisegends se pat callProg x inst y param 2000 override ON duration of chord set to 2000 milliseconds silence d Play silence for d se pat callProg x inst y type silence silence milliseconds Rx audio instruction type is not muted se pat callProg x inst y value 300 specifies silence is to last 300 milliseconds branch n Advance n instructions se pat callProg x inst y type branch branch and execute that instructi
154. dies reg element to the number corresponding to the LCS registration For example roaming_buddies reg 2 Refer to Roaming Buddies lt roaming_buddies gt on page A 104 Configuring Your System k Locate the roaming_privacy attribute l Set the roaming_privacy reg element to the number corresponding to the LCS registration For example roaming_privacy reg 2 Refer to Roaming Privacy lt roaming_privacy gt on page A 104 m Save the modified phonel cfg configuration file Setting Up Audio Features Proprietary state of the art digital signal processing DSP technology is used to provide an excellent audio experience This section provides information for making configuration changes for the following audio related features Low Delay Audio Packet Transmission Jitter Buffer and Packet Error Concealment Voice Activity Detection DTMF Tone Generation DTMF Event RTP Payload Acoustic Echo Cancellation Audio Codecs Background Noise Suppression Comfort Noise Fill Automatic Gain Control IP Type of Service IEEE 802 1p Q Voice Quality Monitoring Low Delay Audio Packet Transmission The phone is designed to minimize latency for audio packet transmission There are no related configuration changes Administrator s Guide SoundPoint IP SoundStation IP Jitter Buffer and Packet Error Concealment The phone employs a high performance jitter buffer and packet error concealment system designed to mitigate packe
155. ding is active e If server based call forwarding is enabled but inactive and the user selects the call forward soft key the moving arrow icon does not appear on the user s phone incoming calls are not forwarded Server based call forwarding is disabled if Shared Call Appearance or Bridged Line Appearance is enabled Configuring Your System Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Enable or disable server based call forwarding For more information refer to SIP lt SIP gt on page A 10 Configuration file phonet cfg Enable or disable server based call forwarding as a per registration feature For more information refer to Registration lt reg gt on page A 89 Set all call diversion settings including a global forward to contact and individual settings for call forward all call forward busy call forward no answer and call forward do not disturb For more information refer to Diversion lt divert gt on page A 96 Local Web Server if enabled Set all call diversion settings Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is remove
156. e SoundStation IP 4000 supports a single registration Network Address Translation The phones can work with certain types of network address translation NAT Presence Allows the phone to monitor the status of other users devices and allows other users to monitor it Requires call server support Real Time Transport Protocol Ports The phone treats all real time transport protocol RTP streams as bi directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports Recording and Playback of Audio Calls Recording and playback allows the user to record any active conversation using the phone on a USB device The files are date and time stamped for easy archiving and can be played back on the phone or on any computer with a media playback program what supports the wav format Server Redundancy Server redundancy is often required in VoIP deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance the server fails or the connection from the phone to the server fails Shared Call Appearances Calls and lines on multiple phones can be logically related to each other Requires call server support Synthesized Call Progress Tones In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment call progress tones ar
157. e and messaging features on or off For more information refer to Feature lt feature gt on page A 83 Configuration file phonet cfg Specify the number of line keys to assign per registration For more information refer to Registration lt reg gt on page A 89 Specify the line registration number which has roaming buddies support enabled For more information refer to Roaming Buddies lt roaming_buddies gt on page A 104 Specify the line registration number which has roaming privacy support enabled For more information refer to Roaming Privacy lt roaming_privacy gt on page A 104 Configuration File Examples SoundPoint IP phones can be deployed in two basic methods In the first method Microsoft Live Communications Server 2005 serves as the call server and the phones have a single registration In the second method the phone has a primary registration to call server that is not Microsoft Live Communications Server LCS and a secondary registration to LCS for presence purposes To set up a single registration with Microsoft Live Communications Server 2005 as the call server 1 Modify the sip cfg configuration file as follows a Open sip cfg in an XML editor Locate the feature parameter For the feature 1 name presence attribute set feature 1 enabled to 1 Configuring Your System d For the feature 2 name messaging attribute set feature 2 enabled to
158. e and the resulting log file fmt SEO orn scheduled log sched 1 name log sched 1 level log sched 1 period log sched 1 startMode log sched 1 startTime log sched 1 startDay log sched 2 name log sched 2 level log sched 2 period log sched 2 startMode log sched 2 startTime log sched 2 startDay a ae wwe Se ee ee ee ee ee ee 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog 0223152507 slog santanna aale zr D E E cl cl a a E E E A A oo oo a oo ol oo ool oo a a ood RR Se RE PAE A i AS oooocococococo0cococo0c0c0c0co0c0c0c0cC0cC0c0cC00o o0o0o0o000000000000000000000 amp g Manual Log Upload Lig OO a Ac change SN gt o ete showCpuLoad Pe oe ERRE E E E EEEE E EE EEEE EEE EE E EE EEE EE EEE EE E EE EE E E E d Running memShow OT a EE EE E E E E E E E EEE EE E EE E EEE EE E EEEE EEE ERE EEE EEE EEE ELE E ELE E E id ind the average is 90 8 EEEE EEEE EEEE EE EE EEEE E E EEE E d 49593 A Zagen A e status bytes current free 6631528 alloc 5498224 cumulative 10509324 EEEE
159. e as the working server As soon as the primary server registration succeeds it will return to being the working server If reg x server y register is set to 0 then phone will not register to that server However the INVITE will fail over to that server if all higher priority servers are down Recommended Practices for Fallback Deployments In situations where server redundancy for fall back purpose is used the following measures should be taken to optimize the effectiveness of the solution 1 Deploy an on site DNS server to avoid long call initiation delays that can result if the DNS server records expire 2 Do not use OutBoundProxy configurations on the phone if the OutBoundProxy could be unreachable when the fallback occurs Sound Point IP phones can only be configured with one OutBoundProxy per registration and all traffic for that registration will be routed through this proxy for all servers attached to that registration If Server 2 is not accessible through the configured proxy call signaling with Server 2 will fail 3 Avoid using too many servers as part of the redundancy configuration as each registration will generate more traffic 4 Educate users as to the features that will not be available when in fallback operating mode Presence Note Configuring Your System The Presence feature allows the phone to monitor the status of other users devices and allows other users to monitor it The status of monito
160. e boot server or locally Central boot server Configuration file Specify the boot up language and the selection of language choices sip cfg to be made available to the user For more information refer to Multilingual lt ml gt on page A 20 For instructions on adding new languages refer to To add new languages to those included with the distribution on page A 21 Local Local Phone User The user can select the preferred language under the Settings menu Interface Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Administrator s Guide SoundPoint IP SoundStation IP Downloadable Fonts New fonts can be loaded onto the phone For guidelines on downloading fonts refer to Fonts lt font gt on page A 66 Synthesized Call Progress Tones In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment call progress tones are synthesized during the life cycle of a call These call progress tones are easily configurable for compatibility with worldwide telephony standards or local preferences Configuration changes can performed centrally at the boot server Central boot server C
161. e configured to the address of the router gateway that provides the fallback telephony support and is on site For example reg 1 server 2 address 172 23 0 1 Administrator s Guide SoundPoint IP SoundStation IP Note Note It is possible to configure the phone for more than two servers per registration but you need to exercise caution when doing this to ensure that the phone and network load generated by registration refresh of multiple registrations do not become excessive This would be of particularly concern if a phone had multiple registrations with multiple servers per registration and it is expected that some of these servers will be unavailable Phone Operation for Registration After the phone has booted up it will register to all the servers that are configured Server 1 is the primary server and supports greater SIP functionality than any of servers For example SUBSCRIBE NOTIFY services used for features such as shared lines presence and BLF will only be established with Server 1 Upon registration timer expiry of each server registration the phone will attempt to re register If this is unsuccessful normal SIP re registration behavior typically at intervals of 30 to 60 seconds will proceed and continue until the registration is successful for example when the Internet link is once again operational While the primary server registration is unavailable the next highest priority server in the list will serv
162. e delay on known good networks Use larger values to minimize packet loss on networks with large jitter 3000 ms voice audioProfile x jitterBufferMax gt The largest jitter buffer depth to be supported jitterBufferMin in milliseconds Jitter above this size will multiple of 10 always cause lost packets This parameter lt 300 for IP should be set to the smallest possible value 320 330 430 that will support the expected network jitter 501 550 600 601 and 650 lt 200 for IP 301 Volume Persistence lt volume gt Configuration Files The user s selection of the receive volume during a call can be remembered between calls This can be configured per termination handset headset and hands free chassis In some countries regulations exist which dictate that receive volume should be reset to nominal at the start of each call on handset and headset Permitted Attribute Values Default Interpretation voice volume persist handset 0 1 0 If set to 1 the receive volume will be i ERER F remembered between calls E Salsa ae If set to 0 the receive volume will be reset voice volume persist handsfree 0 1 1 to nominal at the start of each call Gains lt gain gt The default gain settings have been carefully adjusted to comply with the TIA 810 A digital telephony standard we POLYCOM Polycom recommends that you do not change these values
163. e ee eee eee eee 4 55 Configuring SoundPoint IP SoundStation IP Phones Locally 4 55 5 Troubleshooting Your SoundPoint IP SoundStation IP PRUNE EEEE EE E E e E TEE ET TE T 9 Error Messages pii ii ine a na aea iaa D eai E 5 2 BootROM Error Messag Susscersuperorverocs ers n eee a Pisy 5 2 Application Error Messages 00 000 eee eee eee 5 3 Administrator s Guide SoundPoint IP SoundStation IP viii Status Ment iis siiceiciisncs se tedan evens imels bee wSd isan 5 4 Log Files os ine knee eors eiue tedre RE tae En enn a ieaie 5 4 Reading a Boot Log 0 000 cece eee eee eee 5 7 Reading an Application Log 2 ccc eee eens 5 8 Power and Startup eersten ii nioso ie ha B6ee ee hee Pee eda wee ess 5 9 COmtols i2 28i4 Bediv nel aweeavor e4 adi sim ys ides Shee ae 5 10 Access to Screens and Systems 0 e eee eee eee eee 5 11 Calne iene tie Oak kik ESE a hee KERRE KSR pIE E EERE gas 5 12 Displays 2 52 serat tota Seu tee ods howe SS ee ee 5 13 AMIGO 6 ois ciccr edews ata ien atin HAE OERE EERE Abe Has ema 5 14 UP STAGING asics wraths rekoninte i e e rer eina EEEE ln ee bee 5 14 A Configuration Files 0 cece cece ee eee es An Master Configuration Files 0 0 00 e ee eee eee eee eee A 2 Application Configuration 00 00 A 4 Protocol lt volpProt gt sacs secacnt ehh pork ok bee hee beh eee Bea A 6 Dial Plan lt dialplan gt 0 0 eee eee eee A
164. e is chord this optional parameter specifies the on m integer duration to be used overriding the on duration specified in the chord set definition Call Progress Patterns The following table maps call progress patterns to their usage within the phone Call progress pattern number Use within phone 1 dial tone busy tone ring back tone reorder tone stuttered dial tone 2 3 4 5 6 call waiting tone Configuration Files Call progress pattern number Use within phone 7 alternate call waiting tone distinctive 8 confirmation tone 9 howler tone off hook warning 10 record warning 11 message waiting tone 12 alerting 13 intercom announcement tone 14 barge in tone 15 secondary dial tone Ringer Patterns The following table maps ringer pattern numbers to their default descriptions Ringer pattern number Default description 1 Silent Ring 2 Low Trill 3 Low Double Trill 4 Medium Trill 5 Medium Double Trill 6 High Trill 7 High Double Trill 8 Highest Trill Highest Double Trill 10 Beeble 11 Triplet 12 Ringback style 13 Sampled audio file 2 14 Sampled audio file 3 15 Sampled audio file 4 16 Sampled audio file 5 17 Sampled audio file 6 Administrator s Guide SoundPoint IP SoundStation IP Note Ringer pattern number D
165. e is defined as follows Attribute Permitted Values Default Interpretation font IP_600 x name fontName_height_Uxx00 _UyyFF fon OR fontName_height_Uxx00 _UxxFF fnt Null Defines the font file that will be loaded from boot server during boot up Note When several font P_600 x name are defined the index x must follow consecutive increasing order A 68 Configuration Files Keys lt key gt These settings control the scrolling behavior of keys and can be used to change key functions Permitted Attribute Values Default Interpretation key scrolling timeout positive 1 The time out after which a key that is enabled for integer scrolling will go into scrolling mode until the key is released Keys enabled for scrolling are menu navigation keys left right up down arrows volume keys and some context specific soft keys The value is an integer multiple of 500 milliseconds 1 500ms POLYCOM Sound Point IP 301 320 330 430 501 550 560 600 601 and 650 and SoundStation IP 4000 key functions can be changed from the factory defaults although this is typically not necessary For each key whose function you wish to change add an XML attribute in the format described in the following table to the lt keys gt element of the configuration file These will override the built in assignments Polycom does not recommend the remapping for keys In the followin
166. e other AF11 AF12 gos ip rtp parameters Default AF13 AF21 of Null which means the other AF22 AF23 gos ip rtp parameters will be AF31 AF32 used AF33 AF41 AF42 AF43 qos ip rtp min_delay 0 1 1 If set to 1 set min delay bit in the IP TOS field of the IP header or else don t set it qos ip rtp max_throughput 0 1 1 If set to 1 set max throughput bit in the IP TOS field of the IP header or else don t set it qos ip rtp max_reliability 0 1 0 If set to 1 set max reliability bit in the IP TOS field of the IP header or else don t set it qos ip rtp min_cost 0 1 0 If set to 1 set min cost bit in the IP TOS field of the IP header or else don t set it qos ip rtp precedence 0 7 5 If set to 1 set precedence bits in the IP TOS field of the IP header or else don t set them Administrator s Guide SoundPoint IP SoundStation IP Call Control lt callControl gt These parameters apply to call control packets such as the network protocol signaling Permitted Attribute Values Default Interpretation qos ip callControl dscp 0 to 63 or Null This parameter allows the DSCP of EF or packets to be specified If set to a any of value this will override the other AF11 AF12 gos ip callControl AF13 AF21 parameters Default of Null which AF22 AF23 means the other AF31 AF32 gos ip callControl AF33 AF41 parameters will be used AF42 AF43 qos ip callControl min_delay 0 1 1 If set to 1 set m
167. e synthesized during the life cycle of a call Customizable for certain regions for example Europe has different tones from North America Voice Mail Integration Compatible with voice mail servers Audio Features Acoustic Echo Cancellation Employs advanced acoustic echo cancellation for hands free operation Audio Codecs Supports the standard audio codecs Automatic Gain Control Designed for hands free operation boosts the transmit gain of the local user in certain circumstances Administrator s Guide SoundPoint IP SoundStation IP Background Noise Suppression Designed primarily for hands free operation reduces background noise to enhance communication in noisy environments Comfort Noise Fill Designed to help provide a consistent noise level to the remote user of a hands free call DTMF Event RTP Payload Conforms to RFC 2833 which describes a standard RTP compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream DTMF Tone Generation Generates dual tone multi frequency DTMF tones in response to user dialing on the dial pad IEEE 802 1p Q The phone will tag all Ethernet packets it transmits with an 802 10 VLAN header IP Type of Service Allows for the setting of TOS settings Jitter Buffer and Packet Error Concealment Employs a high performance jitter buffer and packet error concealment system designed to mitigate packet inter arrival jitt
168. e type are allowed The configuration order dictates how the attributes are displayed and sorted The first attribute is the primary soft index and the second attribute is the secondary sort index The other attributes are not used in sorting To limit the amount of data displayed in the corporate directory filtering of the entries can be configured for all attribute types Filtering can be configured to be retained if the phone reboots For more information on LDAP attributes refer to RFC 4510 Lightweight Directory Access Protocol LDAP Techncial Specification Road Map Browsing the Corporate Directory The SoundPoint IP or SoundStation IP phone will establish a session with the corporate directory and download enough entries to fill its cache e when the corporate directory is first accessed e when the phone boots up if the background synchronization parameter is enabled The requested entries are based on the configured attributes see previous section If the background synchronization parameter is enabled a timer is initiated to permit a periodic download from the corporate directory Entries are sorted according to the order in which the first two attributes are configured for example last name then first name The browse position within the corporate directory as well as the attribute filters are maintained for subsequent corporate directory access can be saved if so configured Configuration File Example T
169. eadset txag adjust IP_430 39 voice headset txag adjust IP_650 21 39 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice headset sidetone adjust IP_330 3 voice headset sidetone adjust IP_430 3 voice headset sidetone adjust IP_650 3 Acoustic Echo Cancellation lt aec gt These settings control the performance of the speakerphone acoustic echo canceller Polycom recommends that you do not change these values POLYCOM Attribute Default voice aec hs enable 0 voice aec hs lowFreqCutOff 100 voice aec hs highFreqCutOff 7000 voice aec hs erlTab_0_300 24 voice aec hs erlTab_300_600 24 voice aec hs erlTab_600_ 1500 24 voice aec hs erlTab_1500_ 3500 24 voice aec hs erlTab_3500_ 7000 24 voice aec hd enable 0 voice aec hd lowFreqCutOff 100 voice aec hd highFreqCutOff 7000 voice aec hd erlTab_O_300 24 voice aec hd erlTab_ 300 600 24 voice aec hd erlTab_600_ 1500 24 voice aec hd erlTab_1500_3500 24 voice aec hd erlTab_3500_7000 24 voice aec hf enable 1 voice aec hf lowFreqCutOff 100 voice aec hf highFreqCutOff 7000 A 40 Configuration Files Attribute Default voice aec hf erlTab_O 300 6 voice aec hf erlTab_300_600 6 voice aec hf erlTab_600 1500 6 voice aec hf erlTab_ 1500 3500 6 voice aec hf erlTab_3500_7000 6 Acoustic Echo Suppression lt aes gt The
170. econd Sunday in March at 2 am e Stop DST on the first Sunday in November at 2 am Permitted Attribute Values Default Interpretation tcplpApp sntp resyncPeriod positive 86400 24 Time in seconds between integer hours Simple Network Time Protocol SNTP re syncs tcplpApp sntp address valid host clock Address of the SNTP name or IP server address Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Default Interpretation tcplpApp sntp address overrideDHCP 0 1 0 These parameters determine whether configuration file parameters override DHCP parameters for the SNTP server address and Greenwich Mean Time GMT offset If set to 0 DHCP values will override configuration file parameters If set to 1 the configuration file parameters will override DHCP values tcplpApp sntp gmtOffset positive or negative integer 28800 Pacific time Offset in seconds of the local time zone from GMT 3600 seconds 1 hour tcplpApp sntp gmtOffset overrideDHCP 0 1 0 These parameters determine whether configuration file parameters override DHCP parameters for the SNTP server address and GMT offset If set to 0 DHCP values will override configuration file parameters If set to 1 the configuration file parameters will override DHCP values tcplpApp sntp daylightSavings enable 0 1 If set to 1 apply daylight savings r
171. ecs gt e Gains lt gain gt e Receive Equalization lt rxEq gt Transmit Equalization lt txEq gt e Feature lt feature gt Sampled Audio for Sound Effects lt saf gt The following new sampled audio WAVE file wav formats are supported e 1L16 32000 16 bit 32 kHz sampling rate mono e 1L16 48000 16 bit 48 kHz sampling rate mono Administrator s Guide Addendum for the SoundStation IP 6000 Voice Coding Algorithms lt codecs gt The codecs include e Codec Preferences lt codecPref gt Codec Preferences lt codecPref gt S Permitted Attribute Values Default Interpretation voice codecPref IP_6000 G711Mu Null 1 13 5 Specifies the codec preferences for the SoundStation IP 6000 platform voice codecPref IP_6000 G711A 6 Interpretation as above voice codecPref IP_6000 G722 3 voice codecPref IP_6000 G7221 16kbps Null voice codecPref IP_6000 G7221 24kbps Null voice codecPref IP_6000 G7221 32kbps 4 voice codecPref IP_6000 G729AB 7 voice codecPref IP_6000 G7221C 24kbp Null s voice codecPref IP_6000 G7221C 32kbp Null S voice codecPref IP_6000 G7221C 48kbp 1 s voice codecPref IP_6000 Siren14 24kbp Null S voice codecPref IP_6000 Siren14 32kbp Null s voice codecPref IP_6000 Siren14 48kbp 2 Note Codecs with a default of Null are available for test purposes only and are not expected to be used in your deployment
172. ed for all phones in a deployment If this approach is not used then changes will need to be made to all the lt Ethernet address gt cfg files for SoundPoint IP 300 and 500 phones or all of the lt Ethernet address gt cfg files if it is not explicitly known which phones are SoundPoint IP 300 and 500 phones For more information refer to Technical Bulletin 35311 Supporting SoundPoint IP 300 and IP 500 Phones with SIP 2 2 and Later Releases at http www polycom com support voice Configuring Your System After you set up your SoundPoint IP SoundStation IP phones on the network you can allow users to place and answer calls using the default configuration however you may be require some basic changes to optimize your system for best results This chapter provides information for making configuration changes for e Setting Up Basic Features e Setting Up Advanced Features e Setting Up Audio Features Setting Up Security Features This chapter also provides instructions on e Configuring SoundPoint IP SoundStation IP Phones Locally To troubleshoot any problems with your SoundPoint IP SoundStation IP phones on the network refer to Troubleshooting Your SoundPoint IP SoundStation IP Phones on page 5 1 For more information on the configuration files refer to Configuration Files on page A 1 Setting Up Basic Features This section provides information for making configuration changes for the following basic featu
173. edit contents then remove from file name Telephones without a local directory such as new units from the factory will download the 00000000000 directory xml directory and base their initial directory on it These files should be edited with an XML editor These files can be downloaded once per reflash For information on file format refer to Local Contact Directory File Format the following section XML file lt Ethernet This file can be created manually using an XML editor address gt directory For information on file format refer to Local Contact Directory File xml Format the following section Local Local Phone User The user can edit the directory contents if configured in that way Interface Changes will be stored in the phone s flash file system and backed up to the boot server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the boot server copy of the directory if present will overwrite the local copy Local Contact Directory File Format An example of a local contact directory is shown below The subsequent table provides an explanation of each element lt xml version 1 0 encoding UTF 8 standalone yes gt lt directory gt lt item_list gt 4 10 Configuring Your System lt item gt lt ln gt Doe lt 1n gt lt fn gt John lt fn gt lt ct gt 1001 lt ct gt lt sd gt 1 lt sd gt lt rt gt 1 lt rt gt lt dc gt
174. efault description 18 Sampled audio file 7 19 Sampled audio file 8 20 Sampled audio file 9 21 Sampled audio file 10 22 Sampled audio file 11 Silent Ring will only provide a visual indication of an incoming call but no audio indication Sampled audio files 1 21 all use the same built in file unless that file has been replaced with a downloaded file For more information refer to Sampled Audio for Sound Effects lt saf gt on page A 27 Miscellaneous Patterns The following table maps miscellaneous patterns to their usage within the phone Miscellaneous pattern number Use within phone 1 new message waiting indication 2 new instant message 3 Not used 4 local hold notification 5 positive confirmation 6 negative confirmation 7 welcome boot up Configuration Files Ring type lt rt gt Ring type is used to define a simple class of ring to be applied based on some credentials that are usually carried within the network protocol The ring class includes attributes such as call waiting and ringer index if appropriate The ring class can use one of four types of ring that are defined as follows ring Play a specified ring pattern or call waiting indication visual Provide only a visual indication no audio indication of incoming call no ringer needs to be specified answer Provide auto answer on incoming call ring answer Provide auto answer on incoming call after a rin
175. ement A 94 busy A 97 calls A 93 dial plan emergency A 101 digit map A 99 do not disturb A 94 A 98 forward all A 96 message waiting indicator A 102 messaging A 101 missed call configuration A 94 Network Address Translation A 102 no answer A 97 quotas A 85 registration A 89 roaming buddies A 104 roaming privacy A 104 routing A 100 routing server A 100 per phone configuration file A 89 phonel cfg A 89 port lt port gt A 56 presence 4 43 presence lt pres gt A 65 protocol lt volpProt gt A 6 protocol server lt server gt A 7 protocol special events lt specialEvent gt A 15 provisioning lt prov gt A 81 provisioning protocols 3 4 provisioning protocols supported 3 4 Q QOS See also Quality of Service Quality of Service lt QOS gt A 50 quotas lt quotas gt A 85 R RAM disk lt ramdisk gt A 81 receive equalization lt rxEq gt A 43 registration lt reg gt A 89 reliability of provisional responses B 9 request lt request gt A 82 request delay lt delay gt A 82 request validation lt requestValidation gt A 14 resetting to factory defaults 3 5 resource lt res gt A 84 resource files overview 2 7 restarting phones 3 17 3 18 RFC support B 2 ring type lt rt gt A 33 ringer patterns A 31 roaming buddies lt roaming_buddies gt A 104 roaming privacy lt roaming_provacy gt A 104 routing lt routing gt A 100 routing server lt server gt A 19 A 100 RTP lt RTP gt A 50 A 51 A 56 S sampled audio
176. emove the step of selecting between instant messages and voice mail after pressing the Messages key on the SoundPoint IP 430 500 501 550 560 600 601 and 650 instant messages are still accessible from the Main Menu Navigate to http lt phonelPAddress gt coreConf htm us On a per registration basis specify a subscribe contact for solicited NOTIFY applications a callback mode self call back or another contact to call when the user accesses voice mail Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Multiple Registrations The SoundPoint IP 301 320 330 and 430 support a maximum of two registrations the SoundPoint IP 501 supports three the SoundPoint IP 550 and 560 supports four and the SoundPoint IP 600 601 and 650 support 6 Up to three SoundPoint IP Expansion Modules can be added to a single host SoundPoint IP 601 and 650 phone increasing the total number of buttons to 12 registrations on the IP 601 and 34 registrations on the IP 650 The SoundStation IP 4000 supports a single registration Each registration can be mapped to one or more line keys a line key can be used for only one registration The user can select w
177. en and none is found through DNS 5060 will be used Refer to http www ietf org rfc rfc3263 txt for an example Failure to resolve a DNS name is treated as signalling failure that will cause a failover Behavior When the Primary Server Connection Fails For Outgoing Calls INVITE Fallback When the user initiates a call the phone will go through the following steps to connect the call 1 Try to make the call using the working server 2 If the working server does not respond correctly to the INVITE then try and make a call using the next server in the list even if there is no current registration with these servers This could be the case if the Internet connection has gone down but the registration to the working server has not yet expired 3 If the second server is also unavailable the phone will try all possible servers even those not currently registered until it either succeeds in making a call or exhausts the list at which point the call will fail At the start of a call server availability is determined by SIP signaling failure SIP signaling failure depends on the SIP protocol being used as described below e If TCP is used then the signaling fails if the connection fails or the Send fails e If UDP is used then the signaling fails if ICMP is detected or if the signal times out If the signaling has been attempted through all servers in the list and this is the last server then the signaling fails after the
178. ence Setup lt conference gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP conference address ASCII string Null If Null conferences are set up on the phone up to 128 locally characters If set to some value conferences are set up long by the server using the conferencing agent specified by this address The acceptable values depend on the conferencing server implementation policy A 15 Administrator s Guide SoundPoint IP SoundStation IP Dial Plan lt dialplan gt Note The dial plan is not applied against Placed Call List VoiceMail last call return and remote control dialed numbers This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan applyToCallListDial 0 1 0 This attribute covers dialing from Received Call List and Missed Call List including dialing from Edit or Info sub menus If set to 0 the dial plan is not applied against the dialed number if set to 1 the dial plan is applied against the dialed number dialplan applyToDirectoryDial 0 1 This attribute covers dialing from Directory as well as Speed Dial List Value interpretation is the same as for dialplan applyToCallListDial Note An Auto Call Contact number is considered a dial from directory dialplan applyToUserDial 0 1 This attribute covers the case w
179. er Supported Notes Accept Yes Accept Encoding No Accept Language No Alert Info Yes Allow Yes Allow Events Yes Authentication Info No Authorization Yes Call ID Yes Call Info Yes Contact Yes Content Disposition No Content Encoding No Content Language No Content Length Yes Content Type Yes CSeq Yes Date No Session Initiation Protocol SIP Header Supported Notes Diversion Yes Error Info No Event Yes Expires Yes From Yes In Reply To No Max Forwards Yes Min Expires No Min SE Yes MIME Version No Organization No P Asserted Identity Yes P Preferred Identity Yes Priority No Proxy Authenticate Yes Proxy Authorization Yes Proxy Require No RAck Yes Record Route Yes Refer To Yes Referred By Yes Remote Party ID Yes Replaces Yes Reply To No Require Yes Retry After No Route Yes RSeq Yes Server No Session Expires Yes Administrator s Guide SoundPoint IP SoundStation IP Header Supported Notes Subject No Subscription State Yes Supported Yes Timestamp No To Yes Unsupported No User Agent Yes Via Yes Warning No WWW Authenticate Yes Response Support The following SIP responses are supported Note In the following table a Yes in the Supported column means the header is sent and properly parsed The phone may not actually generate the response 1x
180. er and out of order or lost lost or excessively delayed by the network packets Low Delay Audio Packet Transmission Designed to minimize latency for audio packet transmission Voice Activity Detection Conserves network bandwidth by detecting periods of relative silence in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring Voice Quality Monitoring Generates various quality metrics including MOS and R factor for listening and conversational quality e Security Features Local User and Administrator Privilege Levels Several local settings menus are protected with two privilege levels user and administrator each with its own password Configuration File Encryption Confidential information stored in configuration files must be protected encrypted The phone can recognize encrypted files which it downloads from the boot server and it can encrypt files before uploading them to the boot server Custom Certificates When trying to establish a connection to a boot server for application provisioning the phone trusts certificates issued by widely recognized certificate authorities CAs Incoming Signaling Validation Levels of security are provided for validating incoming network signaling For more information on each feature and its associated configuration parameters see the appropriate section in Configuring Your System on page 4 1
181. ernate DHCP server responds this is functionally equivalent to the scenario where the primary DHCP server responds with a valid boot server value If no alternate DHCP server responds the INFORM query process will retry and eventually time out 2 Static The phone will use the boot server configured through the Server Menu For more information refer to the following section Server Menu 3 Custom Option 66 The phone will first use the custom option if present or use Option 66 if the custom option is not present If the DHCP server sends nothing the following scenarios are possible Ifa boot server value is stored in flash memory and the value is not 0 0 0 0 then the value stored in flash is used e Otherwise the phone sends out a DHCP INFORM query Ifa single alternate DHCP server responds this is functionally equivalent to the scenario where the primary DHCP server responds with a valid boot server value The phone prefers the custom option value over the Option 66 value but if no custom option is given the phone will use the Option 66 value If no alternate DHCP server responds the INFORM query process will retry and eventually time out Boot Server Option 128 through 254 Cannot be the same as VLAN ID Option When the boot server parameter is set to Custom this parameter specifies the DHCP option number in which the phone will look for its boot server Boot Server Option Type
182. ers For information on supported RFC s and Internet drafts refer to the following section RFC and Internet Draft Support This chapter also describes e Request Support e Header Support e Response Support e Hold Implementation e Reliability of Provisional Responses e Transfer e Third Party Call Control e SIP for Instant Messaging and Presence Leveraging Extensions e Shared Call Appearance Signaling e Bridged Line Appearance Signaling Administrator s Guide SoundPoint IP SoundStation IP RFC and Internet Draft Support The following RFC s and Internet drafts are supported RFC 1321 The MD5 Message Digest Algorithm RFC 2327 SDP Session Description Protocol RFC 2387 The MIME Multipart Related Content type RFC 2976 The SIP INFO Method RFC 3261 SIP Session Initiation Protocol replacement for RFC 2543 RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol SIP RFC 3263 Session Initiation Protocol SIP Locating SIP Servers RFC 3264 An Offer Answer Model with the Session Description Protocol SDP RFC 3265 Session Initiation Protocol SIP Specific Event Notification RFC 3311 The Session Initiation Protocol SIP UPDATE Method RFC 3325 SIP Asserted Identity RFC 3515 The Session Initiation Protocol SIP Refer Method RFC 3555 MIME Type of RTP Payload Formats RFC 3611 RTP Control Protocol Extended reports RTCP XR RFC 3665
183. es 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 e Language dictionaries e Custom fonts Administrator s Guide SoundPoint IP SoundStation IP Ring tones Synthesized tones Contact directories Available Features Note The SoundPoint IP 300 and 500 phones will be supported on the latest maintenance patch release of the SIP 2 1 software stream currently SIP 2 1 2 Any new features introduced after SIP 2 1 2 are not supported This section provides information the features available on the SoundPoint IP SoundStation IP phones Basic Features Automatic Off Hook Call Placement Supports an optional automatic off hook call placement feature for each Call Forward Provides a flexible call forwarding feature to forward calls to another destination Call Hold Pauses activity on one call so that the user may use the phone for another task such as making or receiving another call Call Log Contains call information such as remote party identification time and date and call duration in three separate lists missed calls received calls and placed calls on most platforms Call Park Retrieve An active call can be parked A parked call can be retrieved by any phone Call Timer A separate call timer in hours minutes and seconds is
184. es on page C 3 1 Optional Create per phone configuration files by performing the following steps Note This step may be omitted if per phone configuration is not needed a Obtain a list of phone Ethernet addresses barcoded label on underside of phone and on the outside of the box b Create per phone phone MACaddress cfg file by using the phonel cfg file from the distribution as templates For more information on the phonel cfg file refer to Per Phone Configuration on page A 89 Note Throughout this guide the terms Ethernet address and MAC address are used interchangeable c Edit contents of phone MACaddress cfg if desired For example edit the parameters Administrator s Guide SoundPoint IP SoundStation IP Note 2 Optional Create new configuration file s in the style of sip cfg by performing the following steps For more information on why to create another configuration file refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice For more information especially on the SIP server address refer to SIP lt SIP gt on page A 10 For more information on the sip cfg file refer to Application Configuration on page A 4 Most of the default settings are typically adequate however if SNTP settings are not available through DHCP the SNTP GMT offset and possibly the SNTP server address will need to be edited for the correct local cond
185. et to a path name the phone will attempt to download this file at boot time from the boot server If set to a URL the phone will attempt to download this file at boot time from the Internet Note A TFTP URL is expected to be in the format tftp lt host gt pathname lt filename gt for example tftp somehost example com sounds example wav The following table defines the default usage of the sampled audio files with the phone Sampled Audio File 1 oO AON DODO F amp F WwW DN Oo 12 24 Default use within phone pattern reference Welcome Sound Effect se pat misc 7 Ringer 13 se pat ringer 13 Ringer 14 se pat ringer 14 Ringer 15 se pat ringer 15 Ringer 16 se pat ringer 16 Ringer 17 se pat ringer 17 Ringer 18 se pat ringer 18 Ringer 19 se pat ringer 19 Ringer 20 se pat ringer 20 Ringer 21 se pat ringer 21 Ringer 22 se pat ringer 22 Not used Sound Effects lt se gt A 28 The phone uses both synthesized based on the chord sets refer to Chord Sets lt chord gt on page A 26 and sampled audio sound effects Sound effects are defined by patterns rudimentary sequences of chord sets silence periods and wave files Configuration Files This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation se stutterOnVoiceMail 0 1 1 If set to 1 stuttered dial tone is used in place of normal dial tone to in
186. ey1 39 n a 7 Redial 18 Dialpad6 29 ArrowDown 40 n a 8 VolUp 19 Dialpad3 30 Select 41 n a 9 VolDown 20 Dialpad2 31 ArrowLeft 42 n a 10 DialpadStar 21 Dialpad1 32 Menu 11 DialpadO 22 ArrowRight 33 MicMute C 12 SoundPoint IP 501 T pog rr EROR o o i a u 3 Ced Eus d Ee da Transfer A Redial a D Miscellaneous Administrative Tasks FA POLYCOM Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 ArrowUp 12 VolDown 23 Dialpad2 34 Line2 2 ArrowLeft 13 DialpadPound 24 Dialpad1 35 Line1 3 Select 14 DialpadO 25 SoftKey4 36 Redial 4 ArrowRight 15 DialpadStar 26 SoftKey3 37 Transfer 5 ArrowDown 16 Dialpad9 27 SoftKey2 38 Headset 6 Delete 17 Dialpad8 28 SoftKey1 39 MicMute 7 Menu 18 Dialpad7 29 Conference 40 Handsfree 8 Messages 19 Dialpad4 30 CallHistory 41 n a 9 DoNotDisturb 20 Dialpad5 31 Services 42 n a 10 Hold 21 Dialpad6 32 Directories 11 VolUp 22 Dialpad3 33 Line3 Administrator s Guide SoundPoint IP SoundStation IP SoundPoint IP 550 560 600 601 650 Key ID Note The SoundPoint IP 550 and 560 has have only the top four lines keys Key IDs 31 and 42 are not used on SoundPoint IP 550 and 560 phones Key ID Function Key ID Function Key ID Function Key ID Function 1 ArrowUp 12 VolDown 23 Dialp
187. face makes extensive use of intuitive context sensitive soft key menus The soft key function is shown above the key on the graphic display There are no related configuration changes Entries in the local directory can be linked to the speed dial system The speed dial system allows calls to be placed quickly from dedicated keys as well as from a speed dial menu If Presence watching is enabled for speed dial entries their status will be shown on the idle display if the SIP server supports this feature For more information refer to Presence on page 4 43 Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server or locally Central XML file The lt sd gt x lt sd gt element in the lt Ethernet address gt directory xml boot server lt Ethernet file links a directory entry to a speed dial resource within the phone address gt directory Speed dial entries are mapped automatically to unused line keys line xml keys are not available on the SoundStation IP 4000 and are available for selection within the speed dial menu Press the up arrow key from the idle display to jump to SpeedDial For more information refer to Local Contact Directory on page 4 9 Local Local Phone User The next available Speed Dial Index is assigned to new directory Interface entries Key pad short cuts are available to facilitate assigning and modifying the Speed Dial Index va
188. fective user phone radius and helps with the intelligibility of soft talkers There are no related configuration changes IP Type of Service The type of service field in an IP packet header consists of four type of service TOS bits and a 3 bit precedence field Each TOS bit can be set to either 0 or 1 The precedence field can be set to a value from 0 through 7 The type of service can be configured specifically for RTP packets and call control packets such as SIP signaling packets Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify protocol specific IP TOS settings boot server sip ctg For more information refer to IP TOS lt IP gt on page A 51 Local Web Server Specify IP TOS settings if enabled Navigate to http lt phonelPAddress gt netConf htm qo Administrator s Guide SoundPoint IP SoundStation IP IEEE 802 1p Q The phone will tag all Ethernet packets it transmits with an 802 10 VLAN header for one of the following reasons e When it has a valid VLAN ID set in its network configuration e When it is instructed to tag packets through Cisco Discovery Protocol CDP running on a connected Ethernet switch e When a VLAN ID is obtained from DHCP refer to DHCP Menu on page 3 7 The 802 1p Q user_priority field can be set to a value from 0 to 7 The user_priority can be configured specifically for RTP packets and call control
189. files A 28 sampled audio for sound effects lt saf gt A 27 SCA See also shared call appearances scheduled logging parameters A 78 SDP lt SDP gt A 9 security lt sec gt A 79 server menu 3 9 server redundancy 4 39 server based call forwarding See also call forwarding server based DND See also do not disturb Services key See also Applications key Session Initiation Protocol setting up advanced features 4 22 audio features 4 47 basic features 4 1 boot server 3 12 network 3 2 security features 4 53 shared call appearance signaling B 10 shared call appearances shared calls lt shared gt A 60 shared lines barge in 4 26 A 92 Index SIP 1xx Responses Provisional B 6 2xx Responses Success B 6 3xx Responses Redirection B 7 4xx Responses Request Failure B 7 5xx Responses Server Failure B 8 6xx Responses Global Failure B 8 application architecture 2 3 basic protocols hold implementation B 9 basic protocols request support B 3 basic protocols response support B 6 basic protocols RFC and Internet draft support B 2 basic protocols transfer B 9 instant messaging and presence leveraging extensions B 10 RFC 2 1 SIP application description 2 4 installing 3 15 upgrading 3 17 SIP basic protocols header support B 4 SIP See also Session Initiation Protocol sip cfg A 4 SIP lt SIP gt A 10 sound effects lt se gt A 28 SoundPoint IP SoundStation IP phones features overview 2 8 introduction 1 1 network 2 2
190. g bla 02 txt Implementing Bridged Line Appearances BLA Using Session Initiation Protocol SIP e draft ietf sip privacy 04 txt SIP Extensions for Network Asserted Caller Identity and Privacy within Trusted Networks e draft levy sip diversion 06 txt Diversion Indication in SIP e draft ietf sipping cc conferencing 03 txt SIP Call Control Conferencing for User Agents e draft ietf sipping rtcp summary 02 txt Session Initiation Protocol Package for Voice Quality Reporting Event e draft ietf sip connect reuse 04 txt Connection Reuse in the Session Initiation Protocol SIP The following SIP request messages are supported Method Supported Notes REGISTER Yes INVITE Yes ACK Yes CANCEL Yes BYE Yes OPTIONS Yes SUBSCRIBE Yes NOTIFY Yes REFER Yes PRACK Yes Administrator s Guide SoundPoint IP SoundStation IP Method Supported Notes INFO Yes RFC 2976 the phone does not generate INFO requests but will issue a final response upon receipt No INFO message bodies are parsed MESSAGE Yes Final response is sent upon receipt Message bodies of type text plain are sent and received UPDATE Yes Header Support The following SIP request headers are supported Note In the following table a Yes in the Supported column means the header is sent and properly parsed Head
191. g period Note The auto answer on incoming call is currently only applied if there is no other call in progress on the phone at the time In the following table x is the ring class number The x index needs to be sequential Attribute Permitted Values Interpretation se rt enabled 0 1 Set to 1 to enable the ring type feature within the phone 0 otherwise se rt modification enabled 0 1 Set to 1 to allow user modification through local user interface of the pre defined ring type enabled for modification se rt x name UTF 8 encoded string Used for identification purposes in the user interface se rt x type ring OR visual OR answer As defined in table above OR ring answer se rt x ringer integer only relevant if the The ringer index to be used for this class of ring type is set to ring or The ringer index should match one of Ringer ring answer Patterns on page A 31 se rt x callWait integer only relevant ifthe The call waiting index to be used for this class of type is set to ring or ring The call waiting index should match one ring answer defined in Call Progress Patterns on page A 30 se rt x timeout positive integer only The duration of the ring in milliseconds before the relevant if the type is setto call is auto answered If this field is omitted or is left ring answer Default blank a value of 2000 is used value is 2000 se rt x mod 0 1 Set t
192. g table x IP_300 IP_330 IP 430 IP_500 IP_550 IP_600 IP_650 and IP_4000 and y is the key number Note that IP_300 parameters affect SoundPoint IP 301 phones IP_330 parameters affect SoundPoint IP 320 and 330 phones IP_430 parameters affect SoundPoint IP 430 phones IP_500 parameters affect SoundPoint IP 501 phones IP_550 parameters affect SoundPoint IP 550 and 560 phones IP_600 parameters affect SoundPoint IP 600 and 601 phones IP_650 parameters affect SoundPoint IP 650 phones IP_4000 parameters affect the SoundStation IP 4000 phones IP 300 y 1 35 IP 330 y 1 34 IP 430 y 1 35 IP 500 y 1 40 IP_550 y 1 40 IP 600 y 1 42 IP_650 y 1 42 IP_4000 y 1 29 Attribute Permitted Values Interpretation key x y function prim Functions listed below Sets the function for key y on platform x key x y subPoint prim positive integer Sets the sub identifier for key functions with a secondary array identifier such as SpeedDial Administrator s Guide SoundPoint IP SoundStation IP The following table lists the functions that are available Functions ArrowDown ArrowLeft ArrowRight ArrowUp BuddyStatus CallList Conference Delete DialpadO Dialpad1 Dialpad2 Dialpad3 Dialpad4 Dialpad5 Dialpad6 Dialpad7 Dialpad8 Dialpad9 DialpadStar DialpadPound Directories DoNotDisturb Handsfree Headset Hold Line1 Line2 Select Line3 Setup Line4 SoftKey1 Lined SoftKey2 Line6 S
193. ge requests with digest authentication using the local credentials for the associated registration line both or alt apply both of the above methods Configuration Files Permitted Attribute Values Default Interpretation volpProt SIP requestValidation x req A valid string Null Determines which events specified with the uest y event Event header should be validated only applicable when volpProt SIP requestValidation x re quest is set to SUBSCRIBE or NOTIFY If set to Null all events will be validated volpProt SIP requestValidation dige A valid string Polycom Determines string used for Realm st realm SPIP Special Events lt specialEvent gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP specialEvent lineSeize n O 1 1 If set to 1 process a 200 OK response for a onStandard line seize event SUBSCRIBE as though a line seize NOTIFY with Subscription State active header had been received this speeds up processing volpProt SIP specialEvent checkSync 0 1 0 If set to 1 always reboot when a NOTIFY alwaysReboot message is received from the server with event equal to check sync If set to 0 only reboot if any of the files listed in lt MAC address gt cfg have changed on the FTP server when a NOTIFY message is received from the server with event equal to check sync Confer
194. he configuration files bridged lines are configured by shared line parameters Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Specify whether diversion should be disabled on shared lines For more information refer to Call Handling Configuration lt call gt on page A 58 Configuration file phonet cfg Specify per registration line type private or shared and the shared line third party name A shared line will subscribe to a server providing call state information For more information refer to Registration lt reg gt on page A 89 Specify per registration whether diversion should be disabled on shared lines For more information refer to Diversion lt divert gt on page A 96 Local Web Server if enabled Specify per registration line type private or shared and third party name and whether diversion should be disabled on shared lines Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User Interface Specify per registration line type private or shared and the shared line third party name
195. he following excerpt from the sip cfg configuration file shows an example where downloaded entries are limited to any where the phone number is in the 604 area code dir corp address dir corp port dir corp transport TCP dir corp baseDN cn Users dc yourcompany dc local dir corp user ldapadmin dir corp password 12345678 Configuring Your System dir corp filterPrefix objectclass person dir corp attribute 1 name sn dir corp attribute 1 label Last Name dir corp attribute 1 type last_name dir corp attribute 1 filter wu dir corp attribute 1 sticky 1 1 1 1 dir corp attribute 2 name givenName dir corp attribute 2 label First Name dir corp attribute 2 type first_name 2 2 3 3 3 3 filter wu dir corp attribute dir corp attribute 2 sticky dir corp attribute 3 name telephoneNumber label Phone Number type phone_number filter 604 dir corp attribute 3 sticky 1 dir corp attribute dir corp attribute dir corp attribute dir corp backGroundSync 1 dir corp backGroundSync period 60 dir corp viewPersistence 1 Recording and Playback of Audio Calls Note This feature requires a license key for activation Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller SoundPoint IP phones that have a USB port can be configured to
196. he phone is compatible with the comprehensive codec independent comfort noise transmission algorithm specified in RFC 3389 This algorithm is derived from G 711 Appendix II which defines a comfort noise CN payload format or bit stream for G 711 use in packet based multimedia communication systems The phone generates CN packets also known as Silence Insertion Descriptor SID frames and also decodes CN packets efficiently regenerating a facsimile of the background noise at the remote end Configuration changes can performed centrally at the boot server Central boot server Configuration file Enable or disable VAD and set the detection threshold For more information refer to Voice Activity Detection lt vad gt on page A 46 Configuring Your System DTMF Tone Generation The phone generates dual tone multi frequency DTMF tones in response to user dialing on the dial pad These tones are transmitted in the real time transport protocol RTP streams of connected calls The phone can encode the DTMF tones using the active voice codec or using RFC 2833 compatible encoding The coding format decision is based on the capabilities of the remote end point Configuration changes can performed centrally at the boot server Central boot server Configuration file Set the DTMF tone levels autodialing on and off times and other parameters For more information refer to Dual Tone Multi Frequen
197. hen the user presses the Dial soft key to send dialed number when in idle state display Value interpretation is the same as for dialplan applyToCallListDial dialplan applyToUserSend 0 1 This attribute covers the case when the user presses the Send soft key to send the dialed number Value interpretation is the same as for dialplan applyToCallListDial dialplan impossibleMatchHandling 0 1o0r2 If set to 0 the digits entered up to and including the point where an impossible match occurred are sent to the server immediately If set to 1 give reorder tone If set to 2 allow user to accumulate digits and dispatch call manually with the Send soft key dialplan removeEndOfDial 0 1 If set to 1 strip trailing digit from digits sent out Configuration Files This attributes also includes e Digit Map lt digitmap gt e Routing lt routing gt Digit Map lt digitmap gt A digit map is defined either by a string or by a list of strings Each string in the list is an alternative numbering scheme specified either as a set of digits or timers or as an expression over which the gateway will attempt to find a shortest possible match Digit map extension letter R indicates that certain matched strings are replaced The following examples shows the semantics of the syntax RORRXXxxxxx Remove 9 at the beginning of the dialed number For example if a customer di
198. hich registration to use for outgoing calls or which to use when initiating new instant message dialogs Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Specify the local SIP signaling port and an array of SIP servers to register to For each server specify the registration period and the signaling failure behavior For more information refer to Local lt local gt on page A 6 and Server lt server gt on page A 7 Configuration file For up to maximum number of registrations specify a display name phonet cfg a SIP address an optional display label an authentication user ID and password the number of line keys to use and an optional array of registration servers The authentication user ID and password are optional and for security reasons can be omitted from the configuration files The local flash parameters will be used instead The optional array of servers and their associated parameters will override the servers specified in sip cfg if non Null e For more information refer to Registration lt reg gt on page A 89 Administrator s Guide SoundPoint IP SoundStation IP Local Web Server Specify the local SIP signaling port and an array of SIP servers to if enabled register to Navigate to http lt phonelPAddress gt appConf htm se For up to six registrations depending on the phone model in this case the max
199. his section provides instructions on how to add a background logo to all SoundPoint IP phones in your organization You must be running at least BootROM 2 x x and SIP 1 x x One bitmap file is required for each model but SoundPoint IP 301 phones do not support bitmap logos Administrator s Guide SoundPoint IP SoundStation IP Model Width Height Color Depth IP 301 n a n a n a IP 320 330 102 23 monochrome IP 430 94 23 monochrome IP 501 114 51 2 bit grayscale or monochrome IP 600 601 209 109 2 bit grayscale or monochrome IP 550 560 650 209 109 4 bit grayscale or monochrome IP 4000 150 33 monochrome Logos smaller than described in the table above are acceptable but larger logos may be truncated or interfere with other areas of the user interface The SoundPoint IP 501 600 601 phones only support the four colors black dark gray light gray and white Any other colors will be approximated RGB Values Color RGB Values Decimal Hexadecimal Black 0 0 0 00 00 00 Dark Gray 96 96 96 60 60 60 Light Gray 160 160 160 A0 A0 A0 White 255 255 255 FF FF FF The SoundPoint IP 550 560 650 phones support a 4 bit grayscale which is a smooth gradient from black 0 0 0 to white FF FF FF The SoundStation IP 4000 phone only supports black and white Any other colors will be rendered as either black or white Configuration File Changes In the l
200. hree separate lists Missed Calls Received Calls and Placed Calls The call lists can be cleared manually by the user and will be erased when the phone is restarted On some SoundPoint IP platforms missed calls and received calls appear in one list Missed calls appear as gj and received calls appear as f The call list feature can be disabled on all SoundPoint IP and SoundStation IP platforms except the SoundPoint IP 330 320 Configuration changes can performed centrally at the boot server Central boot server sip cfg Configuration File Enable or disable all call lists or individual call lists For more information refer to Feature lt feature gt on page A 83 Call Timer Call Waiting A call timer is provided on the display A separate call timer is maintained for each distinct call in progress The call duration appears in hours minutes and seconds There are no related configuration changes When an incoming call arrives while the user is active on another call the incoming call is presented to the user visually on the LCD display A configurable sound effect such as the familiar call waiting beep will be mixed with the active call audio as well Configuration changes can performed centrally at the boot server Central boot server Configuration File Specify the ring tone heard on an incoming call when another call is phonet cfg active For more information refer
201. ial log entry weeves 4 00 Ini iia aon eusle koe cDE is DISABLED cdp 0223214053 s0 BEIE 802 10 VLAN tagging is DISABLED 0223214053 m 3 00 Platforn Board 2345 11500 020 A 0223214053 3 00 Platform MAC 0004f2015a51 IP Unknown Subnet Mask Unknown 0223214053 so j 00 Platform BootBlock 2 5 0 11500_020 20 Aug 04 16 05 0223214053 so 3 O0 Application main Label BOOT Version 3 0 1 0026 29 Mar 05 10 29 0223214053 so 3 00 Application main P N 3150 11069 301 0223214053 app1l 4 Initial log entry 0223214054 so 13 0 ink status is Net up Speed 10 half Duplex PC down 022321405S app1 3 Ising resolver server 172 23 a 200 alternate server 172 23 0 239 and do pezaztdi23lapp 3 00 T DHCP returned result Ox3E from server 172 23 0 232 0223214123 app1 E oo Phone IP address is 172 23 2 172 0223214123 app1 3 00 Subnet mask is 255 255 0 0 e E Gateway address is 172 23 2 240 0223214123 app1 E ooi Time server is 172 23 0 235 0223214123 app1 3 00 GMT offset is 28800 seconds ee eE ce DNS server is 172 23 0 200 0223214123 app1 3 00 Bo DNS alternate server is 172 23 0 239 0223214123 app1 3 00 Bo y 0223214259 cfg 3 00 Imass DNS domain is Ebert er polycom com 0223214300 cfg 3 00 Image sip ld has not changed 0223214326 app1 4 00 Loaded application sip ld successfully errors 0x0 0223214326 appl 6 00 Uploading boot log time is THU FEB 23 21 43 26 2006 Boot Failure Messages The following figure shows an ex
202. icense and or sell copies of the Software and to permit persons to whom the Software is furnished to do so subject to the following conditions The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software THE SOFTWARE IS PROVIDED AS IS WITHOUT WARRANTY OF ANY KIND EXPRESS OR IMPLIED INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM DAMAGES OR OTHER LIABILITY WHETHER IN AN ACTION OF CONTRACT TORT OR OTHERWISE ARISING FROM OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE curl COPYRIGHT AND PERMISSION NOTICE Copyright c 1996 2004 Daniel Stenberg lt daniel haxx se gt All rights reserved Permission to use copy modify and distribute this software for any purpose with or without fee is hereby granted provided that the above copyright notice and this permission notice appear in all copies THE SOFTWARE IS PROVIDED AS IS WITHOUT WARRANTY OF ANY KIND EXPRESS OR IMPLIED INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OF THIRD PARTY RIGHTS IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM DAMAGES OR OTHER LIABILITY WHETHER IN AN ACTION OF CONTRACT TORT OR OTHERWISE ARISING FROM
203. ile gt lt edit and rename this file to lt Ethernet address gt cfg for each phone gt lt SRCSfile 000000000000 cfg v Revision gt lt APPLICATION APP_FILE_PATH sip MACADDRESS 1d Administrator s Guide SoundPoint IP SoundStation IP CONFIG_FILES phonel MACADDRESS cfg sip cfg MISC_FILES LOG FILE DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY gt If you have a requirement for separate application loads on different phones on the same boot server you can modify the application that is loaded when each phone reboots An example is below lt xml version 1 0 standalone yes gt lt Default Master SIP Configuration File gt lt edit and rename this file to lt Ethernet address gt cfg for each phone gt lt S RCSfile 000000000000 cfg v SRevision gt lt APPLICATION APP_FILE_PATH sip PHONE_PART_NUMBER 1d CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG FILE DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY gt You can also use the substitution strings PHONE_MODEL PHONE_PART_NUMBER and PHONE_MAC_ADDRESS in the master configuration file You can also direct phone upgrades to a software image and configuration files based on the phone model number and part number All XML attributes can be modified in this manner An example is below lt xml version 1 0 standalone
204. ile Enable or disable server based DND as a per registration feature phonet cfg e For more information refer to Registration lt reg gt on page A 89 Specify whether DND is treated as a per registration feature or a global feature on the phone For more information refer to Do Not Disturb lt dnd gt on page A 98 Local Local Phone User Enable or disable DND using the Do Not Disturb key on the Interface SoundPoint IP 301 501 550 560 600 601 and 650 or the Do Not Disturb option on the Features menu on the SoundPoint IP 320 330 and 430 and SoundStation IP 4000 Configuring Your System Handset Headset and Speakerphone SoundPoint IP phones come standard with a handset and a dedicated connector is provided for a headset not supplied The SoundPoint IP 320 330 430 500 501 550 560 600 601 and 650 desktop phones and SoundStation IP 4000 conference phone are full duplex speakerphones The SoundPoint IP 301 phones is a listen only speakerphone The SoundPoint IP phones provide dedicated keys for convenient selection of either the speakerphone or headset Only the SoundPoint IP 320 330 430 550 560 600 601 and 650 desktop phones can be configured to use the electronic hookswitch For more information refer to Technical Bulletin 35150 Using an Electronic Hookswitch with SountPoint IP Phones at http www polycom com support voip Configuration changes can performed ce
205. imum is six even for the IP 601 and 650 specify a display name a SIP address an optional display label an authentication user ID and password the number of line keys to use and an optional array of registration servers The authentication user ID and password are optional and for security reasons can be omitted from the configuration files The local flash parameters will be used instead The optional array of servers will override the servers specified in sip cfg in non Null This will also override the servers on the appConf htm web page Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Local Phone User Use the SIP Configuration menu to specify the local SIP signaling continued Interface port a default SIP server to register to and registration information for up to twelve registrations depending on the phone model The SIP Configuration menu contains a sub set of all the parameters available in the configuration files Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations not a mixture of these options When the SIP Configuration menu is used it is as
206. in delay bit in the IP TOS field of the IP header or else don t set it qos ip callControl max_throughput 0 1 0 If set to 1 set max throughput bit in the IP TOS field of the IP header or else don t set it qos ip callControl max_reliability 0 1 0 If set to 1 set max reliability bit in the IP TOS field of the IP header or else don t set it qos ip callControl min_cost 0 1 0 If set to 1 set min cost bit in the IP TOS field of the IP header or else don t set it qos ip callControl precedence 0 7 5 If set to 1 set precedence bits in the IP TOS field of the IP header or else don t set them Basic TCP IP lt TCP_IP gt This attribute includes A 52 Network Monitoring lt netMon gt Time Synchronization lt sntp gt Port lt port gt Keep Alive lt keepalive gt Configuration Files Network Monitoring lt netMon gt Polycom recommends that you do not change these values POLYCOM This configuration attribute is defined as follows Permitted Attribute Values Default tcplpApp netMon enabled 0 1 1 tcplpApp netMon period 1 to 86400 30 Time Synchronization lt sntp gt The following table describes the parameters used to set up time synchronization and daylight savings time The defaults shown will enable daylight savings time DST for North America Daylight savings defaults Do not use fixed day use first or last day of week in the month e Start DST on the s
207. ing normal 2 medium 3 use of the phone high The default value is medium up backlight idlelntensity 0 off Null This parameter controls the intensity of the 1 low LCD backlight when the phone is idle m i 3 The default value is low nigh Note If idleintensity is set higher than onintensity it will be replaced with the onintensity value up idleTimeout positive Null Timeout for the idle display or default call integer handling display seconds If set to 0 there is no timeout If set to Null the default timeout of 20 seconds is used If set to value greater than 0 the timeout is for that number of seconds maximum 65536 up mwiVisible 0 Disabled 0 If set to O or Null there is no MWI for 1 Enabled registration x SIP 2 1 0 and 2 1 1 behavior If set to 1 msg mwi x callBackMode is set to disabled MWI notification will be displayed for registration x Pre SIP 2 1 0 behavior up handsfreeMode 0 Disabled 1 If set to 1 or Null hands free speakerphone is 1 Enabled enabled If set to 0 hands free speakerphone is disabled Tones lt tones gt This attribute describes configuration items for the tone resources available in the phone This attribute includes e Dual Tone Multi Frequency lt DTMF gt e Chord Sets lt chord gt Dual Tone Multi Frequency lt DTMF gt Configuration Files This configuration attribute is defined as follows Attribute Permitted Values Default Interpre
208. ion Vendor Class Identifier section of the packet and includes the phone s part number and the bootROM version The format of this option s data is not specified in RFC 2132 but is left to each vendor to define its own format To be useful every vendor s format must be distinguishable from every other vendor s format To make our format uniquely identifiable the format follows RFC 3925 which uses the C217 Administrator s Guide SoundPoint IP SoundStation IP C 18 IANA Private Enterprise number to determine which vendor s format should be used to decode the remaining data The private enterprise number assigned to Polycom is 13885 0x0000363D This vendor ID information is not a character string but an array of binary data The steps for parsing are as follows 1 Check for the Polycom signature at the start of the option 4 octet 00 00 36 3d 2 Get the length of the entire list of sub options 1 octet Read the field code and length of the first sub option 1 1 octets If this is a field you want to parse save the data Skip to the start of the next sub option on kk w Repeat steps 3 to 5 until you have all the data or you encounter the End of Suboptions code OxFF For example the following is a sample decode of a packet from an IP601 3c 74 Option 60 length of Option data part of the DHCP spec 00 00 36 3d Polycom signature always 4 octects 6f Length of Polycom data 01 07 50 6 6c 79 63 6f
209. ion attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan x routing server y add dotted decimallP address Null IP address or host name and ress or host name port of a SIP server that will be used for routing calls dialplan x routing server y port 1 to 65535 5060 Multiple servers can be listed starting with y 1 2 for fault tolerance A 100 Emergency lt emergency gt Configuration Files In the following attributes y is the index of the emergency entry description and z is the index of the server associated with the emergency entry y For each emergency entry index y one or more server entry indexes y z can be configured y and z must both follow single step increasing numbering starting at 1 This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan x routing emergency Comma separated list of Null This represents the URLs y value entries or single entry Example that should be watched for representing a or a 15 17 18 911 emergency routing combination of SIP URL sog When one of these defined URL is detected as being dialed by the user the call will be automatically directed to the defined emergency server dialplan x routing emergency y server z positive integer Null Index representing the server defined in Server lt server gt
210. is required the file should be named lt Ethernet address gt cfg where Ethernet address is the MAC address of the phone in question For A F hexadecimal digits use upper or lower case for example 0004 200106c cfg The Ethernet address can be viewed using the About soft key during the auto restart countdown of the bootROM or through the Menu gt Status gt Platform gt Phone menu in the application It is also printed on a label on the back of the phone If this file cannot be downloaded the phone will search for the default master configuration file described next Default master configuration file For systems in which the configuration is identical for all phones no per phone lt Ethernet address gt cfg files the default master configuration file may be used to set the configuration for all phones The file named 000000000000 cfg lt 12 zeros gt cfg is the default master configuration file and it is recommended that one be present on the boot server If a phone does not find its own lt Ethernet address gt cfg file it will use this one and establish a baseline configuration This file is part of the standard Polycom distribution of configuration files It should be used as the template for the lt Ethernet address gt cfg files The default master configuration file 000000000000 cfg is shown below lt xml version 1 0 standalone yes gt lt Default Master SIP Configuration File gt lt edit and
211. it is assumed that all registrations use the same server 4 24 Configuring Your System Multiple Call Appearances The phone supports multiple concurrent calls The hold feature can be used to pause activity on one call and switch to another call The number of concurrent calls per line key is configurable Each registration can have more than one line key assigned to it refer to the previous section Multiple Line Keys per Registration Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Specify the default number of calls that can be active or on hold per line key For more information refer to Call Handling Configuration lt call gt on page A 58 Configuration file Specify per registration the number of calls that can be active or on phonet cfg hold per line key assigned to that registration This will override the default value specified in sip cfg For more information refer to Registration lt reg gt on page A 89 Local Web Server Specify the default number of calls that can be active or on hold per if enabled line key and the number of calls per registration that can be active or on hold per line key assigned to that registration Navigate to http lt phonelPAddress gt appConf htm ls and http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone
212. it will display the function that has been assigned to each of the keys in the configuration This is also where you can test the LCD for faulty pixels In addition to the hardware tests the Diagnostics menu has a series of real time graphs for CPU network and memory utilization that can be helpful in diagnosing performance issues SoundPoint IP and SoundStation IP phones will log various events to files stored in the flash file system and will periodically upload these log files to the boot server The files are stored in the phone s home directory or a user configurable directory There is one log file for the bootROM and one for the application When a phone uploads its log files they are saved on the boot server with the MAC address of the phone prepended to the file name For example 00 4 200360b boot log and 00f4 200360b app log are the files associated with Troubleshooting Your SoundPoint IP SoundStation IP Phones MAC address 00f4f200360b The bootROM log file is uploaded to the boot server after every reboot The application log file is uploaded periodically or when the local copy reaches a predetermined size Both log files can be uploaded on demand using a multiple key combination described in Multiple Key Combinations on page C 9 The phone uploads four files namely mac boot log app boot log mac now boot log and mac now app log The now_ logs are uploaded manually The amount of logging that the phone perform
213. itions Changing the default daylight savings parameters will likely be necessary outside of North American locations a b Optional Disable the local web HTTP server or change its signalling port if local security policy dictates Change the default location settings for user interface language and time and date format Optional Create a master configuration file by performing the following steps a Create per phone or per platform lt Ethernet address gt cfg files by using the 00000000000 cfg and files from the distribution as templates For more information refer to Master Configuration Files on page A 2 Edit the CONFIG_FILES attribute of the lt Ethernet address gt cfg files so that it references the appropriate phone MACaddress cfg file For example replace the reference to phonel cfg with phone MACaddress cfg Edit the CONFIG_FILES attribute of the lt Ethernet address gt cfg files so that it references the appropriate sipXXXX cfg file For example replace the reference to sip cfg with sip650 cfg Edit the LOG_FILE_DIRECTORY attribute of the lt Ethernet address gt cfg files so that it points to the log file directory Edit the CONTACT_DIRECTORY attribute of the lt Ethernet address gt cfg files so that it points to the organization s contact directory Setting up Your System 4 Reboot the phones by pressing the reboot multiple key combination For more information refer to Multiple Key Co
214. keep alive A 57 keys A 69 local protocol A 6 localization A 19 main browser A 87 Microbrowser A 86 multilingual A 20 network monitoring A 53 outbound proxy A 13 password lengths A 80 platform A 71 port A 56 presence A 65 protocol A 6 protocol server A 7 protocol special events A 15 provisioning A 81 Quality of Service A 50 RAM disk A 81 receive equalization A 43 request A 82 request delay A 82 request validation A 14 resource A 84 ring type A 33 routing server A 19 RTP A 50 A 51 A 56 sampled audio for sound effects A 27 SDP A 9 security A 79 shared calls A 60 SIP A 10 sound effect patterns A 29 sound effects A 28 tones A 24 transmit equalization A 45 user preferences A 23 Index 1 Administrator s Guide SoundPoint IP SoundStation IP voice activity detection A 46 voice coding algorithms voice coding algorithms lt codecs gt A 35 voice settings A 34 volume persistence A 37 web server A 57 application configuration file A 4 application error messages 5 3 application files overview 2 6 Applications key 4 30 attendant lt attendant gt A 103 attended transfers 4 18 audio codecs 4 50 audio playback 4 35 A 83 audio recording 4 35 A 83 automatic call distribution 4 38 automatic gain control 4 51 automatic gain control lt agc gt A 43 automatic off hook call placement 4 17 automatic off hook call placement lt autoOffHook gt A 94 B background logo adding C 5 configura
215. l If non Null this port will be used by the phone for SIP signaling overriding the value set for volpProt local signalPort in sip cfg nat mediaPortStart 1024 to 65535 Null If non Null this attribute will be used to set the initially allocated RTP port overriding the value set for tcpIpApp port rtp mediaPortRangeStart in sip cfg Refer to RTP lt rtp gt on page A 56 nat keepalive interval 0 to 3600 Null If non Null or 0 the keepalive interval in seconds This parameter is used to set the interval at which phones will send a keep alive packet to the gateway NAT device to keep the communication port open so that NAT can continue to function as setup initially The Microsoft Live Communications Server 2005 keepalive feature will override this interval If you want to deploy phones behind a NAT and connect them to Live Communications Server the keepalive interval received from the Live Communications Server must be short enough to keep the NAT port open Once the TCP connection is closed the phones stop sending keep alive packets Attendant lt attendant gt Note These attributes are available on SoundPoint IP 600 601 and 650 phones with an attached Expansion Module only The Busy Lamp Field BLF attendant console feature enhances support for a phone based attendant console A 103 Administrator s Guide SoundPoint IP SoundStation IP This configuration attribute is
216. l Specifies the line registration number which has roaming integer privacy support enabled If Null roaming privacy is disabled If value lt 1 then value is replaced with 1 A 104 Configuration Files User Preferences lt user_preferences gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation up analogHeadsetOpti 0 1 on 0 Selects optional external hardware for use with a headset attached to the phone s analog headset jack If set to 0 no compatible headset is attached If set to 1 a DHSG compatible headset is attached and can be used as an electonic hookswitch Flash Parameter Configuration Caution Any field in the bootROM setup menu and the application SIP Configuration menu can be set through a configuration file A DHCP server can be configured to point the phones to a boot server that has the required configuration files The new settings will be downloaded by the phones and used to configure them This removes the need for manual interaction with phones to configure basic settings This is especially useful for initial installation of multiple phones These device settings are detected when the application starts If the new settings would normally cause a reboot if they were changed in the application Network Configuration menu then they will cause a reboot when the application starts The parameters for thi
217. l can be parked and the parked call can be retrieved by another phone This feature depends on support from a SIP server Configuration changes can performed centrally at the boot server Central Configuration file Turn this feature on or off boot server sip ctg For more information refer to Feature lt feature gt on page A 83 Last Call Return The phone allows server based last call return This feature depends on support from a SIP server 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 lt feature gt on page A 83 Specify the string sent to the server for last call return For more information refer to Call Handling Configuration lt call gt on page A 58 Setting Up Advanced Features This section provides information for making configuration changes for the following advanced features e Configurable Feature Keys e Multiple Line Keys per Registration e Multiple Call Appearances e Shared Call Appearances e Bridged Line Appearance e Busy Lamp Field e Customizable Fonts and Indicators 4 22 Configuring Your System e Instant Messaging e Multilingual User Interface e Downloadable Fonts e Synthesized Call Progress Tones e Microbrowser e Real Time Transport Protocol Ports e Network Address Translation e Corporate Direc
218. lable from a shared line call park is the call park and park retrieve features group call pickup is the group call pickup feature directed call pickup is the directed call pickup feature last call return is the last call return feature acd login logout is the ACD login logout feature acd agent available is the ACD agent available unavailable feature nway conference is the conference managing feature call recording is the call recording and playback feature corporate directory is the corporate directory feature Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Interpretation feature x enabled 0 or 1 default except for x 9 If set to 0 the feature will be disabled If set to 1 the feature will be enabled and usable by the local user Note The url dialing feature must be disabled by setting feature 9 enabled to 0 in order to prevent unknown callers from being identified on the display by an IP address Note The call list feature can be disabled on all SoundPoint IP and SoundStation IP platforms except the SoundPoint IP 330 320 Note For feature 16 name nway conference e If set to 0 the n way conferencing feature is disabled meaning that three way conferencing can exist but there is no manage conference page e If setto the n way conferencing feature is enabled the maximum num
219. latform refer to Supported Provisioning Protocols on page 3 4 All of the boot servers must be reachable by the same protocol and the content available on them must be identical The parameters described in section Server Menu on page 3 9 can be used to configure the number of times each server will be tried for a file transfer and also how long to wait between each attempt The maximum number of servers to be tried is configurable For more information contact your Certified Polycom Reseller Be aware of how logs overrides and directories are uploaded to servers that maps to multiple IP addresses The server that these files are uploaded to may change over time If you want to use redundancy for uploads synchronize the files between servers in the background However you may want to disable the redundancy for uploads by specifying specific IP addresses instead of URLs for logs overrides and directory in the lt MAC address gt cfg To set up the boot server Use this procedure as a recommendation if this is your first boot server setup 1 Install boot server application or locate suitable existing server s Polycom recommends that you use RFC compliant servers 2 Create account and home directory If the provisioning protocol requires an account name and password the server account name and password must match those configured in the phones Defaults are provisioning protocol FTP name PlemSplp password PlemSplp
220. le but is not intended to be used for long term operation of the phones Administrator s Guide SoundPoint IP SoundStation IP Setting Up the Network Regardless of whether or not you will be installing a centrally provisioned system you must perform basic TCP IP network setup such as IP address and subnet mask configuration to get your organization s phones up and running The bootROM application uses the network to query the boot server for upgrades which is an optional process that will happen automatically when properly deployed For more information on the basic network settings refer to DHCP or Manual TCP IP Setup on page 3 2 The bootROM on the phone performs the provisioning functions of downloading the bootROM the lt Ethernet address gt cfg file and the SIP application and uploading log files For more information refer to Supported Provisioning Protocols on page 3 4 Basic network settings can be changed during bootROM download using the bootROM s setup menu A similar menu system is present in the application for changing the same network parameters For more information refer to Modifying the Network Configuration on page 3 5 DHCP or Manual TCP IP Setup Basic network settings can be derived from DHCP or entered manually using the phone s LCD based user interface or downloaded from configuration files Polycom recommends using DHCP where possible to eliminate repetitive m
221. led A 96 Busy lt busy gt Configuration Files Calls can be automatically diverted when the phone is busy Attribute Permitted Values Default Interpretation divert busy x enabled 0 1 1 If set to 1 calls will be forwarded on busy to the contact specified below Note If server based call forwarding is enabled this parameter is disabled divert busy x timeout positive integer 60 Time in seconds to allow altering before initiating the diversion divert busy x contact ASCII encoded string containing digits the user part of a SIP URL or a string that constitutes a valid SIP URL 6416 or 6416 polycom com Null Forward to contact for calls forwarded due to busy status if Null divert x contact will be used No Answer lt noanswer gt The phone can automatically divert calls after a period of ringing Attribute Permitted Values Default Interpretation divert noanswer x enabled 0 1 1 If set to 1 calls will be forwarded on no answer to the contact specified Note If server based call forwarding is enabled this parameter is disabled divert noanswer x timeout positive integer 60 Time in seconds to allow altering before initiating the diversion divert noanswer x contact ASCII encoded string containing digits the user part of a SIP URL or a string that constitutes a valid SIP URL 6416
222. ll Network signaling is employed to request that the remote party stop sending media and to inform them that they are being held A configurable local hold reminder feature can be used to remind the user that they have placed calls on hold The call hold reminder is always played through the speakerphone Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Specify whether RFC 2543 c 0 0 0 0 or RFC 3264 a sendonly or sip cfg a inactive outgoing hold signaling is used For more information refer to SIP lt SIP gt on page A 10 Specify local hold reminder options For more information refer to Hold Local Reminder lt hold gt lt localReminder gt on page A 61 Local Web Server Specify whether or not to use RFC 2543 c 0 0 0 0 outgoing hold if enabled signaling The alternative is RFC 3264 a sendonly or a inactive Navigate to http lt phonelPAddress gt appConf htm ls Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User Use the SIP Configuration menu to specify whether or not to use RFC Interface 2543 c 0 0
223. locally if configured in that way Contact information from previous calls may be easily added to the directory for convenient future access Administrator s Guide SoundPoint IP SoundStation IP The directory is the central database for several other features including speed dial distinctive incoming call treatment presence and instant messaging Note If a user makes a change to the local contact directory there is a five second timeout before it is uploaded to the boot server as lt mac address gt directory cfg Configuration changes can performed centrally at the boot server or locally Central Configuration file Set whether the directory uses volatile storage on the phone boot server sip cfg required on the SoundPoint IP 500 platform for directories greater than 25 entries For more information refer to Local Directory lt local gt on page A 62 Specify whether or not the local contact directory is read only For more information refer to Local Directory lt local gt on page A 62 XML file A sample file named 000000000000 directory xml Note the extra 000000000000 direct in the filename is included with the application file distribution ory xml This file can be used as a template for the per phone lt Ethernet address gt directory xml directories edit contents then rename to lt Ethernet address gt directory xml It also can be used to seed new phones with an initial directory
224. log files to be uploaded and changes made by the phone user to the configuration through the web server and local user interface and changes made to the directory to be backed up For organizational purposes configuring a separate log file directory is recommended but not required For more information on LOG_FILE_DIRECTORY refer to Master Configuration Files on page A 2 File permissions should give the minimum access required and the account used should have no other rights on the server The phone s server account needs to be able to add files to which it can write in the log file directory and the root directory It must also be able to list files in all directories mentioned in the lt MAC address gt cfg file All other files that Setting up Your System the phone needs to read such as the application executable and the standard configuration files should be made read only through file server file permissions Deploying Phones From the Boot Server You can successfully deploy SoundPoint IP and SoundStation IP phones from one or more boot servers Multiple boot servers can be configured by having the boot server DNS name map to multiple IP addresses The default number of boot servers is one and the maximum number is eight HTTPS HTTP and FTP are supported for redundant boot servers To deploy phones from the boot server Note For more information on encrypting configuration files refer to Encrypting Configuration Fil
225. lpProt server x expires lineSeize positive 30 Requested line seize subscription period integer minimum 10 volpProt server x Ics 0 1 0 This attribute overrides the volpProt SIP lcs If set to 1 the proprietary epid parameter is added to the From field of all requests to support Microsoft Live Communications Server SDP lt SDP gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SDP answer useLocalPrefer Oor1 0 If set to 1 the phones uses its own ences preference list when deciding which codec to use rather than the preference list in the offer If set to 0 it is disabled Administrator s Guide SoundPoint IP SoundStation IP SIP lt SIP gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP useContactInReferTo 0 1 0 If set to 0 the To URI is used in the REFER If set to 1 the Contact URI is used in the REFER volpProt SIP useRFC2543hold 0 1 0 If set to 1 use the obsolete c 0 0 0 0 RFC2543 technique otherwise use SDP media direction attributes such as a sendonly per RFC 3264 when initiating hold In either case the phone processes incoming hold signaling in either format volpProt SIP useSendonlyHold 0 1 1 If set to 1 the phone will send a reinvite with a stream mode attribute of sendonly when a call is put on hol
226. lt routing gt Digit Map lt digitmap gt For more information on digit map syntax refer to Digit Map lt digitmap gt on page A 17 Administrator s Guide SoundPoint IP SoundStation IP This configuration attribute is defined as follows separated by Permitted Attribute Values Default Interpretation dialplan x digitmap A string compatible with the Null When present this attribute digit map feature of MGCP overrides the global dial plan described in 2 1 5 of RFC defined in the sip cfg 3435 string is limited to 768 configuration file bytes and 30 segments a For more information refer to comma is also allowed a Digit Map lt digitmap gt on page comma is also allowed A 17 when reached in the digit map a comma will turn dial tone back on is allowed as a valid digit extension letter R is used as defined above dialplan x digitmap timeOut string of positive integers Null When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For more information refer to Digit Map lt digitmap gt on page A 17 Routing lt routing gt This attribute allows specific routing paths for outgoing SIP calls to be configured independent of other default configuration This attribute includes Server lt server gt Emergency lt emergency gt Server lt server gt This configurat
227. lue for entries in the directory The Speed Dial Index field is used to link directory entries to speed dial operations Changes will be stored in the phone s flash file system and backed up to the boot server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the boot server copy of the directory if present will overwrite the local copy Time and Date Display The phone maintains a local clock and calendar Time and date can be displayed in certain operating modes such as when the phone is idle and during a call The clock and calendar must be synchronized to a remote Simple Network Time Protocol SNTP timeserver The time and date displayed on the phone will flash continuously until a successful SNTP response is received to indicate that they are not accurate The time and date display can use one of several different formats and can be turned off Configuration changes can performed centrally at the boot server or locally Central Configuration file Turn time and date display on or off boot server sip cfg For more information refer to User Preferences lt up gt on page A 23 Set the time and date display formats For more information refer to Date and Time lt datetime gt on page A 23 Set the basic SNTP settings and daylight savings parameters For more information refer to Time Synchronization lt sntp gt on page A 53 Configuri
228. mation refer to Encryption lt encryption gt on page A 80 Configuration file Change the encryption key lt device gt cfg For more information refer to Flash Parameter Configuration on page A 105 Changing the Key on the Phone For security purposes it may be desirable to change the key on the phones and the server from time to time To change a key 1 Put the new key into a configuration file that is in the list of files downloaded by the phone specified in 000000000000 cfg or lt Ethernet address gt cfg Use the device sec configEncryption key parameter to specify the new key Manually reboot the phone so that it will download the new key The phone will automatically reboot a second time to use the new key At this point the phone expects all encrypted configuration files on the boot server to use the new key and it will continue to reboot until this is the case The files on the server must be updated to the new key or they must be made available in unencrypted format Updating to the new key requires decrypting the file with the old key then encrypting it with the new key Note that configuration files contact directory files and configuration override files may all need to be updated if they were already encrypted In the case of configuration override files they can be deleted from the boot server so that the phone will replace them when it successfully boots Adding a Background Logo T
229. mbinations on page C 9 The bootROM and SIP application modify the APPLICATION APP_FILE_PATH attribute of the lt Ethernet address gt cfg files so that it references the appropriate sip ld files For example the reference to sip ld is changed to 2345 11605 001 sip Id to boot the SoundPoint IP 601 image Note At this point the phone sends a DHCP Discover packet to the DHCP server This is found in the Bootstrap Protocol option Vendor Class Identifier section of the packet and includes the phone s part number and the bootROM version For example a SoundPoint IP 650 might send the following information 5EL DC 5cSc52 46 9N7 lt u6 pPolycomSoundPointIP SPIP_6502345 12600 001 1B R 4 0 0 0155 23 May 07 13 35BR 4 0 0 0155 23 May 07 13 35 For more information refer to Parsing Vendor ID Information on page C 17 5 Monitor the boot server event log and the uploaded event log files if permitted Ensure that the configuration process completed correctly All configuration files used by the boot server are logged You can now instruct your users to start making calls Upgrading SIP Application You can upgrade the SIP application that is running on the SoundPoint IP and SoundStation IP phones in your organization The exact steps that you perform are dependent on the version of the SIP application that is currently running on the phones and the version that want to upgrade to The bootROM application executable and configuratio
230. mends that you create another file with your organization s modifications If you must change any Polycom templates back them up first By default sip cfg is included e Per phone It contains parameters unique to a particular phone user Typical parameters include display name unique addresses Each phone in an installation usually has its own customized version of user files derived from Polycom templates By default phonel cfg is included Central Provisioning The phones can be centrally provisioned from a boot server through a system of global and per phone configuration files The boot server also facilitates automated application upgrades logging and a measure of fault tolerance Multiple redundant boot servers can be configured to improve reliability In the central provisioning method there are two major classifications of configuration files e System configuration files e Per phone configuration files Parameters can be stored in the files in any order and can be placed in any number of files The default is to have 2 files one for per phone setting and one for system settings The per phone file is typically loaded first and could contain system level parameters letting you override that parameter for a given user For example it might be desirable to set the default CODEC for a remote user differently than for all the users who reside in the head office By adding the CODEC settings to a particular use
231. n files can be updated automatically through the centralized provisioning model These files are read only by default Most organization can use the instructions shown in the next section Supporting SoundPoint IP and SoundStation IP Phones However if your organization has a mixture of SoundPoint IP 300 and or 500 phones deployed along with other models you will need to change the phone configuration files to continue to support the SoundPoint IP 300 and IP 500 phones when software releases SIP 2 2 0 or later are deployed These models were discontinued as of May 2006 In this case refer to Supporting SoundPoint IP 300 and 500 Phones on page 3 19 Administrator s Guide SoundPoint IP SoundStation IP Warning The SoundPoint IP 300 and 500 phones will be supported on the latest maintenance patch release of the SIP 2 1 software stream currently SIP 2 1 2 Any critical issues that affect SoundPoint IP 300 and 500 phones will be addressed by a maintenance patch on this stream until the End of Life date for these products Phones should be upgraded to BootROM 4 0 0 for these changes to be effective Supporting SoundPoint IP and SoundStation IP Phones Warning To automatically update 1 Back up old application and configuration files The old configuration can be easily restored by reverting to the backup files 2 Customize new configuration files or apply new or changed parameters to the old configuration files Differences
232. n idle Timeout 0 600 seconds Null Timeout for the interactive browser If the interactive browser remains idle for a defined period of time the phone should return to the idle browser If set to 0 there is no timeout If set to Null the value from up idleTimeout is used Refer to User Preferences lt up gt on page A 23 If mb main idleTimeout and up idleTimeout are Null the timeout is 20 seconds If set to value greater than 0 and less than 600 the timeout is for that number of seconds Administrator s Guide SoundPoint IP SoundStation IP Browser Limits lt limits gt These settings limit the size of object which the Microbrowser will display by limiting the amount of memory available for the Microbrowser Attribute Permitted Values Default Interpretation mb limits nodes positive integer 256 Limits the number of tags that the XML parser will handle This limits the amount of memory used by complicated pages A maximum total of 500 256 each is recommended This value is used as referent values for 16MB of SDRAM Note Increasing this value may have a detrimental effect on performance of the phone mb limits cache positive integer 200 Limits the total size of objects downloaded for each page both XHTML and images Once this limit is reached no more images are downloaded until the next page is requested Units kBytes This value is used as referent values for 16MB of SDRAM Note
233. n the Polycom Support Knowledgebase Upgrading Symptom Problem Corrective Action SoundPoint IP 300 and or 500 behave incorrectly or do not display new features New features are not supported on the SoundPoint IP 300 and 500 and the configuration files have not been correctly modified The SoundPoint IP 300 and 500 will not understand the new configuration parameters and will attempt to load the new application The attempt to load the new application will fail since there is no 300 500 image contained within the sip ld file so the phone will continue on and run the current version of application that it has in memory It will however use the new configuration files Refer to Supporting SoundPoint IP 300 and 500 Phones on page 3 19 5 14 Configuration Files Caution Note Note This appendix provides detailed descriptions of certain configuration files used by the Session Initiation Protocol SIP application It is a reference for all parameters that are configurable when using the centralized provisioning installation model This appendix contains information on e Master Configuration Files MAC address cfg or 000000000000 cfg e Application Configuration sip cfg e Per Phone Configuration phonel cfg e Flash Parameter Configuration The application configuration files dictate the behavior of the phone once it is running the executable specified in the master co
234. n the bootROM until the password prompt appears e IP 301 501 550 600 601 and 650 4 6 8 and dial pad keys e IP 320 330 and 430 560 1 3 5 and 7 dial pad keys e IP 4000 6 8 and dial pad keys Enter the administrator password to initiate the reset Resetting to factory defaults will also reset the administrator password factory default password is 456 Uploading Log Files For the key combination press and hold certain key combinations depending on the phone model simultaneously until a confirmation tone is heard or for about three seconds e IP 301 The two Line keys and the Up and Down arrow keys e IP 320 and 330 Menu Dial and the two Line keys e IP 430 501 550 560 600 601 and 650 Up Down Left and Right arrow keys Administrator s Guide SoundPoint IP SoundStation IP e JP 4000 Menu Exit Off hook Hands free Redial Default Feature Key Layouts The following figures and tables show the default SIP key layouts for the SoundPoint IP 301 320 330 430 501 550 560 600 601 and 650 and SoundStation IP 4000 models SoundPoint IP 301 ri 2 Cae CY east 07 E p Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 Line1 12 DialpadPound 23 DoNotDisturb 34 n a 2 Line2 13 Dialpad9 24 n a 35 Headset 3 n a 14 Dialpad8 25 SoftKey3 36 n a 4 n a 15 Dialpad7 26 MicMu
235. nd configured to receive and send SIP messages For more information on IP PBX and softswitch vendors go to http www polycom com techpartners1 This chapter contains information on e Where SoundPoint IP SoundStation IP Phones Fit e Session Initiation Protocol Application Architecture e Available Features To install your SoundPoint IP SoundStation IP phones on the network refer to Setting up Your System on page 3 1 To configure your SoundPoint IP SoundStation IP phones with the desired features refer to Configuring Your Administrator s Guide SoundPoint IP SoundStation IP System on page 4 1 To troubleshoot any problems with your SoundPoint IP SoundStation IP phones on the network refer to Troubleshooting Your Sound Point IP SoundStation IP Phones on page 5 1 Where SoundPoint IP SoundStation IP Phones Fit The phones connect physically to a standard office twisted pair IEEE 802 3 10 100 megabytes per second Ethernet LAN and send and receive all data using the same packet based technology Since the phone is a data terminal digitized audio being just another type of data from its perspective the phone is capable of vastly more than traditional business phones AsSoundPoint IP SoundStation IP phones run the same protocols as your office personal computer many innovative applications can be developed without resorting to specialized technology Remote Boot Server Remote Application Serer
236. ndPoint IP SoundStation IP family are available e Quick Start Guides which describe how to assemble the phones e Quick User Guides which describe the most basic features available on the phones e User Guides which describe the basic and advanced features available on the phones e Developer s Guide which assists in the development of applications that run on the SoundPoint IP SoundStation IP phone s Microbrowser e Technical Bulletins which describe workarounds to existing issues e Release Notes which describe the new and changed features and fixed problems in the latest version of the software For support or service please contact your Polycom reseller or go to Polycom Technical Support at http www polycom com support voice Polycom recommends that you record the phone model numbers software both the bootROM and SIP and partner platform for future reference SoundPoint IP SoundStation IP models BootROM version SIP Application version Partner Platform Administrator s Guide SoundPoint IP SoundStation IP Contents About This Guide ccc cece cece eee e eee ee H 1 Introducing the SoundPoint IP SoundStation IP Family 1 1 SoundPoint IP Desktop Phones 0c eee cece 1 2 SoundStation IP Conference Phone 2000000 1 4 Key Features of Your SoundPoint IP SoundStation IP Phones 1 5 2 Overview oi 6h oS Ok KES ROR RO a eee
237. ned IP address and subnet mask will never be used in this configuration Resetting to Factory Defaults The basic network configuration referred to in the following sections can be reset to factory defaults using a multiple key combination described in Multiple Key Combinations on page C 9 Administrator s Guide SoundPoint IP SoundStation IP Main Menu The following configuration parameters can be modified on the main setup menu Name Possible Values Description DHCP Client Enabled Disabled If enabled DHCP will be used to obtain the parameters discussed in DHCP or Manual TCP IP Setup on page 3 2 DHCP Menu Refer to DHCP Menu on page 3 7 Note Disabled when DHCP client is disabled Phone IP Address dotted decimal IP address Phone s IP address Note Disabled when DHCP client is enabled Subnet Mask dotted decimal subnet mask Phone s subnet mask Note Disabled when DHCP client is enabled IP Gateway dotted decimal IP address Phone s default router Server Menu Refer to Server Menu on page 3 9 SNTP Address dotted decimal IP address Simple Network Time Protocol SNTP server from OR which the phone will obtain the current time domain name string GMT Offset 13 through 12 Offset of the local time zone from Greenwich Mean Time GMT in half hour increments DNS Server dotted decimal IP address Primary server to which the phone directs Domain Name
238. new dialog messages can be entered manually or selected from the contact directory the preferred method There are no related configuration changes Multilingual User Interface Note Note Note This feature is not available on SoundPoint IP 301 phones The system administrator or the user can select the language Support for major western European languages is included and additional languages can be easily added Support for Asian languages Chinese Japanese and Korean is also included but will display only on the SoundPoint IP 650 and SoundStation IP 4000 s higher resolution displays For basic character support and extended character support available on Sound Point IP 550 560 600 601 and 650 and SoundStation IP platforms refer to Multilingual lt ml gt on page A 20 Note that within a Unicode range some characters may not be supported due to their infrequent usage The multilingual feature relies on dictionary files resident on the boot server The dictionary files are downloaded from the boot server whenever the language is changed or at boot time when a language other than the internal US English language has been configured If the dictionary files are inaccessible the language will revert to the internal language Currently the multilingual feature is only available in the application At this time the bootROM application is available in English only Configuration changes can performed centrally at th
239. nfiguration changes can performed centrally at the boot server or locally Central Configuration File Specify patterns used for sound effects and the individual tones or boot server sip cfg sampled audio files used within them For more information refer to Sampled Audio for Sound Effects lt saf gt on page A 27 or Sound Effects lt se gt on page A 28 Local Web Server Specify sampled audio wave files to replace the built in defaults if enabled Navigate to http lt phonelPAddress gt coreConf htm sa Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Message Waiting Indication The phone will flash a message waiting indicator MWI LED when instant messages and voice messages are waiting Configuration changes can performed centrally at the boot server Central Configuration file Specify per registration whether the MWI LED is enabled or disabled boot server phonet cfg e For more information refer to Message Waiting Indicator lt mwi gt on page A 102 Specify whether MWI notification is displayed for registration x pre SIP 2 1 behavior is enabled For more information refer to User Preferences lt up gt on page A 23 Distinctive
240. nfiguration file Configuration files should only be modified by a knowledgeable system administrator Applying incorrect parameters may render the phone unusable The configuration files which accompany a specific release of the SIP software must be used together with that software Failure to do this may render the phone unusable In the tables in the following sections Null should be interpreted as the empty string that is attributeName when the file is viewed in an XML editor To enter special characters in a configuration file enter the appropriate sequence using an XML editor amp as amp amp e as amp quot e as amp apos lt as amp lt e gt as amp gt The various hd parameters in sip cfg such as voice aec hd enable voice ns hd enable and voice agc hd enable are headset parameters They are not connected to high definition or HD voice Administrator s Guide SoundPoint IP SoundStation IP Master Configuration Files The master configuration files can be one of Specified master configuration file The master configuration file can be explicitly specified in the boot server address for example http usr pwd server dir examplel cfg The filename must end with cfg and be at least five characters long If this file cannot be downloaded the phone will search for the per phone master configuration file described next Per phone master configuration file If per phone customization
241. ng Your System Local Web Server Set the basic SNTP and daylight savings settings if enabled Navigate to http lt phonelPAddress gt coreConf htm ti Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User The basic SNTP settings can be made in the Network Configuration Interface menu For more information refer to DHCP or Manual TCP IP Setup on page 3 2 The user can edit the time and date format and enable or disable the time and date display under the Settings menu Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server They will permanently override global settings unless deleted through the Reset Local Config menu selection Idle Display Animation All phones except the SoundPoint IP 301 can display a customized animation on the idle display in addition to the time and date For example a company logo could be displayed refer to Adding a Background Logo on page C 5 Configuration changes can performed centrally at the boot server Central Configuration file To turn idle display animation on or off boot server sip cfg For more information refer to Indicators lt ind gt on page A 71
242. ng configuration changes for the following security related features e Local User and Administrator Privilege Levels e Custom Certificates e Incoming Signaling Validation e Configuration File Encryption Local User and Administrator Privilege Levels Several local settings menus are protected with two privilege levels user and administrator each with its own password The phone will prompt for either the user or administrator password before granting access to the various menu options When the user password is requested the administrator password will also work The web server is protected by the administrator password refer to Configuring SoundPoint IP SoundStation IP Phones Locally on page 4 55 Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify the minimum lengths for the user and administrator boot server sip cfg passwords For more information refer to Password Lengths lt pwd gt lt length gt on page A 80 Local Web Server None if enabled Local Phone User The user and administrator passwords can be changed under the Interface Settings menu or through configuration parameters refer to Flash Parameter Configuration on page A 105 Passwords can consist of ASCII characters 32 127 0x20 0x7F only Changes are saved to local flash but are not backed up to lt Ethernet address gt
243. ng table x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Attribute Permitted Values Default Interpretation call autoOffHook x enabled 0 1 0 If set to 1 a call will be automatically placed to call autoOffHook x contact ASCII encoded string containing digits Null the contact specified the user part of a SIP URL or a string upon going off hook on that constitutes a valid SIP URL 6416 this registration or 6416 polycom com Missed Call Configuration lt serverMissedCall gt The phone supports a per registration configuration of which events will cause the locally displayed missed calls counter to be incremented In the following table x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Permitted Attribute Values Default Interpretation call serverMissedCall x enabled 0 1 0 If set to 0 all missed call events will increment the counter If set to 1 only missed call events sent by the server will increment the counter A 94 Configuration Files Missed Call Tracking lt missedCallTracking gt You can enable disable missed call tracking on a per line basis In the following table x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550
244. nge is 1 to 24 and the default is 24 For all other phones the permitted range is 1 to 8 and the default is 8 This is the number of calls that may be active or on hold per line key on the phone Note that this may be overridden by the per registration attribute of reg x callsPerLineKey Refer to Registration lt reg gt on page A 89 call stickyAutoLineSeize Null O 1 If set to 1 makes the phone use sticky line seize behavior This will help with features that need a second call object to work with The phone will attempt to initiate a new outgoing call on the same SIP line that is currently in focus on the LCD this was the behavior in SIP 1 6 5 If set to O or Null the feature is disabled this was the behavior in SIP 1 6 6 Note This may fail due to glare issues in which case the phone may select a different available line for the call call stickyAutoLineSeize onHook Dialing Null O 1 Null If call stickyAutoLineSeize is set to 1 this parameter has no effect The regular stickyAutoLineSeize behavior is followed If call stickyAutoLineSeize is set to 0 or Null and this parameter is set to 1 this overrides the stickyAutoLineSeize behavior for hot dial only Any new call scenario seizes the next available line If call stickyAutoLineSeize is set to 0 or Null and this parameter is set to 0 or Null there is no difference between hot dial and new call scenarios Note A hot dial
245. nnection reuse draft which introduces alias volpProt SIP sendCompactHdrs 0 1 If set to 0 SIP header names generated by the phone use the long form for example From If set to 1 SIP header names generated by the phone use the short form for example f volpProt SIP keepalive sessionTimer s 0 1 If set to 1 the session timer will be enabled If set to 0 the session timer will be disabled and the phone will not declare timer in Support header in INVITE The phone will still respond to a re INVITE or UPDATE The phone will not try to re INVITE or do UPDATE even if remote end point asks for it volpProt SIP requestURI E164 addGl obalPrefix 0 1 If set to 1 global prefix is added to E 164 user parts in sip URIs volpProt SIP allowTransferOnProcee ding 0 1 If set to 1 a transfer can be completed during the proceeding state of a consultation call If set to 0 a transfer is not allowed during the proceeding state of a consultation call If set to Null the default value is used volpProt SIP dialog useSDP 0 1 If set to 0 new dialog event package draft is used no SDP in dialog body If set to 1 for backwards compatibility use this setting to send SDP in dialog body volpProt SIP pingInterval 0 to 3600 The number in seconds to send PING message This feature is disabled by default volpProt SIP useContactInReferTo 0 1 If set
246. notification for the line disabled If set to contact a call will be placed to the contact specified in the callback attribute when the user invokes message retrieval If set to registration a call will be placed using this registration to the contact registered the phone will call itself If set to disabled message retrieval and message notification are disabled msg mwi x callBack ASCII encoded string containing Null Contact to call when retrieving digits the user part of a SIP messages for this registration URL or a string that constitutes a valid SIP URL 6416 or 6416 polycom com Network Address Translation lt nat gt A 102 These parameters define port and IP address changes used in NAT traversal The port changes will change the port used by the phone while the IP entry simply changes the IP advertised in the SIP signaling This allows the use of simple NAT devices that can redirect traffic but do not allow for port mapping For example port 5432 on the NAT device can be sent to port 5432 on an internal device but not port 1234 Configuration Files This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation nat ip dotted decima IP address Null IP address to advertise within SIP signaling should match the external IP address used by the NAT device nat signalPort 1024 to 65535 Nul
247. nt IP 330 320 430 501 550 560 600 601 and 650 and the SoundStation IP 4000 Configuring Your System The RTCP XR packets are complaint with RFC 3611 RTP Control Extended Reports RTCP XR The packets are sent to a report collector as specified in draft RFC draft ietf_sipping_rtcp summary 02 Three types of quality reports can be enabled e Alert Generated when the call quality degrades below a configurable threshold e Periodic Generated during a call at a configurable period e Session Generated at the end of a call A wide range of performance metrics are generated Some are based on current values such as jitter buffer nominal delay and round trip delay while others cover the time period from the beginning of the call until the report is sent such as network packet loss Some metrics are computed using other metrics as input such as listening Mean Opinion Score MOS conversational MOS listening R factor and conversational R factor Configuration changes can performed centrally at the boot server Central boot server Configuration file Specify the location of the central report collector how often the reports are generated and the warning and critical threshold values that will cause generation of alert reports For more information refer to Quality Monitoring lt quality monitoring gt on page A 47 Setting Up Security Features This section provides information for maki
248. ntrally at the boot server or locally Central boot server Configuration file sip cfg Enable or disable persistent headset mode For more information refer to User Preferences lt up gt on page A 23 Enable or disable hands free speakerphone mode For more information refer to User Preferences lt up gt on page A 23 Configuration file Specify whether or not the electronic hookswitch is enabled and what phonet cfg type of headset is attached For more information refer to User Preferences lt user_preferences gt on page A 89 Local Web Server Enable or disable persistent headset mode if enabled Navigate to http lt phonelPAddress gt coreConf htm us Local Phone User Interface Enable or disable persistent headset mode through the Settings menu Settings gt Basic gt Preferences gt Headset gt Headet Memory Mode Enable or disable hands free speakerphone mode through the Settings menu Settings gt Advanced gt Admin Settings gt Phone Settings Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Contact Directory The phone maintains a local contact directory The directory can be downloaded from the boot server and edited
249. o 1 if the user interface should allow for modification by the user of the ringer index used for this ring class Administrator s Guide SoundPoint IP SoundStation IP Note Modification of se rt modification enabled and se rt x name parameters through the user interface will be implemented in a future release Voice Settings lt voice gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation voice txPacketFilter 0 1 Null Flag to determine whether or not narrowband Tx high pass filtering should be enabled If set to 1 narrowband Tx high pass filter is enabled If set 0 or Null no Tx filtering is performed This attribute includes e Voice Coding Algorithms lt codecs gt e Volume Persistence lt volume gt e Gains lt gain gt e Acoustic Echo Cancellation lt aec gt e Acoustic Echo Suppression lt aes gt e Background Noise Suppression lt ns gt e Automatic Gain Control lt agc gt e Receive Equalization lt rxEq gt Transmit Equalization lt txEq gt e Voice Activity Detection lt vad gt e Quality Monitoring lt quality monitoring gt Voice Coding Algorithms lt codecs gt The following voice codecs are supported Configuration Files Sample Effective Audio Algorithm MIME Type Label Bit Rate Rate Frame
250. odec preferences for the SoundStation IP 4000 platform voice codecPref IP_4000 G711A 2 Interpretation as above voice codecPref IP_4000 G729AB Null Not supported by default so that G 711Mu and G 711A local conferences can be supported Codec Profiles lt audioProfile gt The following profile attributes can be adjusted for each of the five supported codecs In the table x G711Mu G711A G722 G7221 and G729AB Permitted Attribute Values Interpretation voice audioProfile x payloadSize 10 20 30 80 Preferred Tx payload size in milliseconds to be provided in SDP offers and used in the absence of ptime negotiations This is also the range of supported Rx payload sizes voice audioProfile x jitterBufferMin 20 40 50 60 The smallest jitter buffer depth in milliseconds multiple of that must be achieved before play out begins 10 for the first time Once this depth has been achieved initially the depth may fall below this point and play out will still continue This parameter should be set to the smallest possible value which is at least two packet payloads and larger than the expected short term average jitter The IP4000 values are the same as the IP30x values voice audioProfile x jitterBufferShrink 10 20 30 The absolute minimum duration time in multiple of 10 milliseconds of RTP packet Rx with no packet loss between jitter buffer size shrinks Use smaller values 1000 ms to minimize th
251. oftKey3 Messages SoftKey4 Menu SpeedDial MicMute SpeedDialMenu MyStatus Transfer Null VolDown Offline VolUp Redial Release Backgrounds lt bg gt The backgrounds used by the SoundPoint IP 550 560 and 650 phones are defined in this section In the following table x 1 to 4 This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation bg hiRes gray pat solid x name any string White Solid pattern name For x 1 White x 2 Light Gray x 3 4 Null bg hiRes gray pat solid x red 0 to 255 The screen background layouts bg hiRes gray pat solid x green For x 1 red 255 green 255 blue 255 For x 2 red 160 green 160 blue 160 bg hiRes gray pat solid x blue For x 3 and 4 all values are Null bg hiRes gray bm x name any string User defined graphic files For x 1 string is Leaf bmp For x 2 string is World bmp A 70 Bitmaps lt bitmap gt Configuration Files The bitmaps used by each phone model are defined in this section Platform lt IP_300 gt lt IP 330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt In the following table x IP_300 IP_330 IP_400 IP_500 IP_600 IP_4000 and y is the bitmap number Note that IP_300 parameters affect SoundPoint IP 301 phones IP_330 parameters affect SoundPoint IP 320 and 330 phones IP_400 parameters affect SoundPoint IP 430 phones IP_500 parame
252. oice ns hs signalAttn 6 voice ns hs silenceAttn 9 Administrator s Guide SoundPoint IP SoundStation IP For more information refer to Acoustic Echo Cancellation lt aec gt on page A 34 Acoustic Echo Suppression lt aes gt on page A 41 and Background Noise Suppression lt ns gt on page A 42 For the SoundPoint IP 501 and 601 utilizing acoustic echo cancellation will introduce a small delay increase into the audio path which might cause a lower voice quality Note Audio Codecs AEC on the SoundPoint IP 301 handset is not supported The following table summarizes the phone s audio codec support Effective Sample audio Algorithm MIME Type Ref Bit Rate Rate Frame Size bandwidth G 711p law PMCU RFC 1890 64 Kbps 8 Ksps 10ms 80ms 3 5KHz G 711a law PCMA RFC 1890 64 Kbps 8 Ksps 10ms 80ms 3 5KHz G 722 G722 8000 RFC 1890 64 Kbps 16 Ksps 10ms 80ms 7 KHz G 722 1 G722 16000 RFC 3047 16 Kbps 16 Ksps 20ms 80ms 7 KHz 24 Kbps 32 Kbps G 729AB G729 RFC 1890 8 Kbps 8 Ksps 10ms 80ms 3 5KHz SID CN RFC 3389 N A N A N A N A RFC 2833 phone event RFC 2833 N A N A N A N A Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify codec priority preferred payload sizes and jitter buffer tuning boot server sip cfg parameters For more information refer to Codec Preferences lt codecPref gt on page A 35 and Code
253. on phonet cfg e For more information refer to Registration lt reg gt on page A 89 Server Redundancy Note Warning Server redundancy is often required in VoIP deployments to ensure continuity of phone service for events where the call server needs to be taken offline for maintenance the server fails or the connection from the phone to the server fails Two types of redundancy are possible Fail over In this mode the full phone system functionality is preserved by having a second equivalent capability call server take over from the one that has gone down off line This mode of operation should be done using DNS mechanisms or IP Address Moving from the primary to the back up server Fallback In this mode a second less featured call server router or gateway device with SIP capability takes over call control to provide basic calling capability but without some of the richer features offered by the primary call server for example shared lines presence and Message Waiting Indicator Polycom phones support configuration of multiple servers per SIP registration for this purpose In some cases a combination of the two may be deployed Your SIP server provider should be consulted for recommended methods of configuring phones and servers for fail over configuration Prior to SIP 2 1 the reg x server y parameters refer to Registration lt reg gt on page A 89 could be used for fail over configuration The olde
254. on type instruction n must be se pat callProg x inst y value 5 step back 5 negative and must not instructions and execute that instruction branch beyond the first instruction Administrator s Guide SoundPoint IP SoundStation IP Note Currently patterns that use the sampled instruction are limited to the following format sampled followed by optional silence and optional branch back to the beginning In the following table x is the pattern number y is the instruction number Both x and y need to be sequential There are three categories of sound effect patterns callProg Call Progress Patterns ringer Ringer Patterns and misc Miscellaneous Patterns Permitted Attribute Values Interpretation se pat callProg x name UTF 8 Used for identification purposes in the user interface currently encoded used for ringer patterns only for patterns that use a sampled string audio file which has been overridden by a downloaded replacement the se pat ringer x name parameter will be overridden in the user interface by the file names of the wave file se pat callProg x inst y type sampled OR As above chord OR silence OR branch se pat callProg x inst y valu integer Instruction type Interpretation e sampled sampled audio file number chord chord set number silence silence duration in ms branch number of instructions to advance se pat callProg x inst y para positive If instruction typ
255. onfiguration file Specify the basic tone frequencies levels and basic repetitive sip cfg cadences For more information refer to Chord Sets lt chord gt on page A 26 Specify downloaded sampled audio files for advanced call progress tones For more information refer to Sampled Audio for Sound Effects lt saf gt on page A 27 Specify patterns For more information refer to Patterns lt pat gt on page A 29 and Call Progress Patterns on page A 30 Microbrowser Note The SoundPoint IP 430 501 550 560 600 601 and 650 phones and the SoundStation IP 4000 phone supports an XHTML Microbrowser This can be launched by pressing the Applications key or if there isn t one on the phone it can be accessed through the Menu key by selecting Features and then Applications As of SIP 2 2 0 the Services key and menu entry are renamed Applications however the functionality remains the same Two instances of the Microbrowser may run concurrently e An instance with standard interactive user interface e Aninstance that does not support user input but appears in a window on the idle display For more information refer to the Web Application Developers s Guide Configuring Your System Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Specify the Application browser home page a proxy to use and size limits
256. ons of 1 3 and 1 4 support this feature Refer to the latest Release Notes to verify if the image is signed There is a new image file in each release of software The application performs the following tasks in order Downloads system and per phone configuration files and resource files These files are called sip cfg and phonel cfg by default You can customized the filenames If the Application is any SIP version prior to 1 5 the bootROM would have downloaded all the configuration files that are listed in the master configuration file Configuration Warning Overview 2 Controls all aspects of the phone after it has restarted 3 Uploads log files BootROM and Application Wrapper Both the bootROM and the application run on multiple platforms meaning all previously released versions of hardware that are still supported The file stored on the boot server is a wrapper with multiple hardware specific images contained within When a new bootROM or application is being saved the file is read until a header matching the hardware model and revision are found and then only this image is saved to flash memory The SoundPoint IP SoundStation IP phones can be configured automatically through files stored on a central boot server manually through the phone s local UI or web interface or a combination of the automatic and manual methods The recommended method for configuring phones is automatically through a central b
257. oot server but if one is not available the manual method will allow changes to most of the key settings The phone configuration files consist of e Master Configuration Files e Application Configuration Files Configuration files should only be modified by a knowledgeable system administrator Applying incorrect parameters may render the phone unusable The configuration files which accompany a specific release of the SIP software must be used together with that software Failure to do this may render the phone unusable Master Configuration Files The master configuration files can be one of e Specified master configuration file e Per phone master configuration file e Default master configuration file For more information refer to Master Configuration Files on page A 2 Administrator s Guide SoundPoint IP SoundStation IP Note Application Configuration Files Typically the files are arranged in the following manner although parameters may be moved around within the files and the filenames themselves can be changed as needed These files dictate the behavior of the phone once it is running the executable specified in the master configuration file The application files are e Application It contains parameters that affect the basic operation of the phone such as voice codecs gains and tones and the IP address of an application server All phones in an installation usually share this category of files Polycom recom
258. or s Guide SoundPoint IP SoundStation IP Save the modified phonel cfg configuration file To set up a dual registration with Microsoft Live Communications Server 2005 as the presence server 1 Optional Modify the sip cfg configuration file as follows a b c g Open sip cfg in an XML editor Locate the feature parameter For the feature 1 name presence attribute set feature 1 enabledto 1 For the feature 2 name messaging attribute set feature 2 enabled to 1 Locate the volpProt parameter If SIP forking is desired set voIpProt SIP ms forking attribute to 1 Refer to SIP lt SIP gt on page A 10 Save the modified sip cfg configuration file Modify the phonel cfg configuration file as follows a Open phonel cfg in an XML editor Locate the registration parameter Select a registration to be used for the Microsoft Live Communications Server 2005 Typically this would be 2 Set the reg x address to the LCS address For example reg 2 address 7778 Set the reg x server y address to the LCS server name Optional Set the reg 2 server y transport attribute to TCPpreferred or TLS Your selection depends on the LCS configuration Set reg x auth userId to the phone s LCS username For example reg 2 auth userId jbloggs Set reg x auth password to the LCS password For example reg 2 auth password Password2 Locate the roaming_buddies attribute Set the roaming_bud
259. or1 For description refer to DHCP or Manual TCP IP Setup on page 3 2 A 106 Configuration Files Name Possible Values Description device dhcp offerTimeout 1 to 600 Number of seconds the phone waits for secondary DHCP Offer messages before selecting an offer device dhcp bootSrvUseOp 0 to 3 For descriptions refer to DHCP Menu on page 3 7 t device dhcp bootSrvOpt 128 to 254 Cannot be the same as VLAN ID Option device dhcp bootSrvOptTy Oor1 pe device dhcp dhcpVlanDisc 0 to 2 UseOpt device dhcp dhcpVlanDisc Opt 128 to 254 Cannot be the same as Boot Server Option device prov serverName any string For descriptions refer to Server Menu on page 3 9 device prov serverType 0to4 device prov user any string device prov password any string device prov appProvType Oor1 device prov appProvString any string device prov redunAttemptLi 10 Null mit device prov reduninterAtte 300 Null mptDelay device sntp serverName any string Can be dotted decimal IP address or domain name string SNTP server from which the phone will obtain the current time device sntp gmtOffset 43200 to 46800 GMT offset in seconds corresponding to 12 to 13 hours device dns serverAddress dotted decimal IP address Primary server to which the phone directs Domain Name System queries device dns altSrvAddress dotted decimal IP address Secondary server
260. ot server Missed Call Notification The phone can display the number of calls missed since the user last looked at the Missed Calls list The types of calls that are counted as missed can be configured per registration Remote missed call notification can be used to notify the phone when a call originally destined for it is diverted by another entity such as a Session Initiation Protocol SIP server Note On some SoundPoint IP platforms missed calls and received calls appear in one list Configuring Your System Configuration changes can performed centrally at the boot server Central boot server Configuration file Turn this feature on or off sip cfg For more information refer to Feature lt feature gt on page A 83 Configuration file Specify per registration whether all missed call events or only phonet cfg remote server generated missed call events will be displayed For more information refer to Missed Call Configuration lt serverMissedCall gt on page A 94 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 Vol
261. ot simply be copied and put under another distribution licence including the GNU Public Licence zlib C 1995 2002 Jean loup Gailly and Mark Adler This software is provided as is without any express or implied warranty In no event will the authors be held liable for any damages arising from the use of this software Permission is granted to anyone to use this software for any purpose including commercial applications and to alter it and redistribute it freely subject to the following restrictions 1 The origin of this software must not be misrepresented you must not claim that you wrote the original software If you use this software in a product an acknowledgment in the product documentation would be appreciated but is not required 2 Altered source versions must be plainly marked as such and must not be misrepresented as being the original software 3 This notice may not be removed or altered from any source distribution Jean loup Gailly Mark Adler jloup gzip org madler alumni caltech edu Administrator s Guide SoundPoint IP SoundStation IP Expat Copyright c 1998 1999 2000 Thai Open Source Software Center Ltd and Clark Cooper Permission is hereby granted free of charge to any person obtaining a copy of this software and associated documentation files the Software to deal in the Software without restriction including without limitation the rights to use copy modify merge publish distribute subl
262. p to the boot server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the boot server copy of the directory if present will overwrite the local copy Microsoft Live Communications Server 2005 Integration Note SoundPoint IP phones can used with Microsoft Live Communications Server 2005 and Microsoft Office Communicator to help improve business efficiencies and increase productivity and to share ideas and information immediately with business contacts For instructions on changing the configuration files refer to Configuration File Examples on page 4 44 Any contacts added through the SoundPoint IP phone s buddy list will appear in as a contact in Microsoft Office Communicator and Windows Messenger Administrator s Guide SoundPoint IP SoundStation IP POLYCOM Polycom recommends that the BLF not be used in conjunction with the Microsoft Wye Live Communications Server 2005 feature For more information refer to Busy Lamp Field on page 4 27 Configuration changes can performed centrally at the boot server Central boot server Configuration file sip cfg Specify that support for Microsoft Live Communications Server 2005 is enabled For more information refer to SIP lt SIP gt on page A 10 Specify the line registration number used to send SUBSCRIBE for presence For more information refer to Presence lt pres gt on page A 65 Turn the presenc
263. plyToDirectoryDial 0 1 0 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 Configuration Files Permitted Attribute Values Default Interpretation dialplan x applyToUserDial 0 1 1 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 dialplan x applyToUserSend 0 1 1 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 dialplan x impossibleMatchHandling 0 1 0r2 0 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 dialplan x removeEndOfDial 0 1 1 When present and if dialplan x digitmap s not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 16 This attribute also includes e Digit Map lt digitmap gt e Routing
264. r Boot Server 0 Option 66 The phone will look for option number 66 string type in the response received from the DHCP server The DHCP server should send address information in option 66 that matches one of the formats described for Server Address in the following section Server Menu If the DHCP server sends nothing the following scenarios are possible Ifa boot server value is stored in flash memory and the value is not 0 0 0 0 then the value stored in flash is used e Otherwise the phone sends out a DHCP INFORM query If a single alternate DHCP server responds this is functionally equivalent to the scenario where the primary DHCP server responds with a valid boot server value If no alternate DHCP server responds the INFORM query process will retry and eventually time out Administrator s Guide SoundPoint IP SoundStation IP Name Possible Values Description Boot Server continued 1 Custom The phone will look for the option number specified by the Boot Server Option parameter below and the type specified by the Boot Server Option Type parameter below in the response received from the DHCP server If the DHCP server sends nothing the following scenarios are possible Ifa boot server value is stored in flash memory and the value is not 0 0 0 0 then the value stored in flash is used e Otherwise the phone sends out a DHCP INFORM query If a single alt
265. r s per phone file the values in the system file are ignored Verify the order of the configuration files Parameters in the configuration file loaded first will overwrite those in later configuration files Resource Files Overview The following figure shows one possible layout of the central provisioning method Boot Server lil contig overrides directory 000422002999 ety 000422002999 phose cty noost2002999 noct 1og event log files aip id 000422002999 dicectocy ctg 000422002995 app 10y aip ety phom2995 ctg SousdPotatIe dickianaey ml master config file application binary config files dictionary files user interface resource files SoundPoint IP SIP Local User Interface MAC 00 04 12 00 29 99 Local Web Server 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 In addition to the application and the configuration files the phon
266. r chord set components can be specified y 1 2 3 4 tone chord cat x level y 57 to 3 Level of this component in dBm0 tone chord cat x onDur positive On duration in milliseconds O infinite integer tone chord cat x offDur positive Off duration in milliseconds O infinite integer tone chord cat x repeat positive Specifies how many times the ON OFF cadence is integer repeated O infinite Sampled Audio for Sound Effects lt saf gt Note The following sampled audio WAVE file wav formats are supported e mono 8 kHz G 711 Law e G 711 A Law e L16 16000 16 bit 16 kHz sampling rate mono L16 16000 is not supported on SoundPoint IP 301phones and SoundStation IP 4000 phones The phone uses built in wave files for some sound effects The built in wave files can be replaced with files downloaded from the boot server or from the Internet however these are stored in volatile memory so the files will need to remain accessible should the phone need to be rebooted Files will be truncated to a maximum size of 300 kilobytes Administrator s Guide SoundPoint IP SoundStation IP In the following table x is the sampled audio file number Attribute Permitted Values Interpretation saf x Null OR valid path name OR an RFC 1738 compliant URL to a HTTP FTP or TFTP wave file resource Note Refer to the above wave file format restrictions If Null the phone will use a built in file If s
267. r payload If set to 0 or Null there is no change to SDP Quality Monitoring lt quality monitoring gt This attribute includes e Central Report Collector lt collector gt e Alert Reports lt alert gt e Server lt server gt e RTCP XR lt rtcpxr gt Central Report Collector lt collector gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation voice qualityMonitoring collector enable pe 0 1 0 Enables generation of periodic riodic quality reports throughout a call voice qualityMonitoring collector enable se 0 1 0 Enables generation of a quality ssion report at the end of each call Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Default Interpretation voice qualityMonitoring collector enable tri ggeredPeriodic 0 1 2 0 Controls the generation of periodic quality reports triggered by alert states If set to 0 alert states do not cause periodic reports to be generated If set to 1 periodic reports will be generated when an alert state is critical If set to 2 periodic reports will be generated when an alert state is either warning or critical Note This parameter is ignored when qualityMonitoring collector e nable periodic is set 1 since periodic reports are sent throughout the duration of a call voice qualityMonitoring collector period
268. r activation Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller we POLYCOM Configuring Your System The SoundPoint IP phones can be configured to interface with a corporate directory server that supports the Lightweight Directory Access Protocol LDAP version 3 Microsoft s Active Directory is included Both corporate directories that support server side sorting and those that do not not are supported In the latter case the sorting is performed on the phone Polycom recommends using corporate directories that have server side sorting Polycom recommends that you consult your LDAP Administrator when making any configuration changes for this feature The corporate directory can be browsed or searched Entries retrieved from the LDAP server can be saved to the local contact directory on the phone Phone calls can be placed based on the phone number contained in the LDAP entry The corporate directory interface shall be read only so that editing or deleting existing directory entries as well as adding new directory entries from the phone shall not be possible All attributes are considered to be Unicode text Validity checking will be performed when a call isplaced or the entry is saved to the local contact directory The corporate directory LDAP server status can be reviewed through the Status menu Status gt CD Server Stat
269. r behavior is no longer supported Customers that are using the reg x server y configuration parameters where y gt 2 should take care to ensure that their current deployments are not adversely affected For example the phone will only support advanced SIP features such as shared lines missed calls presence with the primary server y 1 For more information refer to Technical Bulletin 5844 SIP Server Fallback Enhancements on SoundPoint IP Phones at http www polycom com support voice Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server Central boot server Configuration file Specify global primary and fallback server configuration parameters sip cfg e For more information refer to Protocol lt volpProt gt on page A 6 Configuration file Specify per registration primary and fallback server configuration phonet cfg parameters values that override those in sip cfg For more information refer to Registration lt reg gt on page A 89 4 40 Note DNS SIP Server Name Resolution If a DNS name is given for a proxy registrar address the IP address es associated with that name will be discovered as specified in RFC 3263 If a port is given the only lookup will be an A record If no port is given NAPTR and SRV records will be tried before falling back on A records if NAPTR and SRV records return no results If no port is giv
270. r elects to retrieve messages Voice mail access can be configured to be through a single key press for example the Messages key on the SoundPoint IP 430 500 501 550 560 600 601 and 650 A message waiting signal from a voice mail server triggers the message waiting indicator to flash and the call waiting audio tone is played through the active audio path Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file For one touch voice mail access enable the one touch voice mail sip cfg user preference e For more information refer to User Preferences lt up gt on page A 23 Configuration file For one touch voice mail access bypass instant messages to remove phonet cfg the step of selecting between instant messages and voice mail after pressing the Messages key on the SoundPoint IP 430 500 501 550 560 600 601 and 650 instant messages are still accessible from the Main Menu On a per registration basis specify a subscribe contact for solicited NOTIFY applications a callback mode self call back or another contact and the contact to call when the user accesses voice mail For more information refer to Messaging lt msg gt on page A 101 Configuring Your System Local Web Server if enabled For one touch voice mail access enable the one touch voice mail user preference and bypass instant messages to r
271. r log files if required A URL can also be specified This is blank by default e CONTACTS_DIRECTORY An alternative directory to use for user directory files if required A URL can also be specified This is blank by default e OVERRIDES_DIRECTORY An alternative directory to use for configuration overrides files if required A URL can also be specified This is blank by default e LICENSE_DIRECTORY An alternative directory to use for license files if required A URL can also be specified This is blank by default The order of the configuration files listed in CONFIG_FILES is significant e The files are processed in the order listed left to right e The same parameters may be included in more than one file e The parameter found first in the list of files will be the one that is effective This provides a convenient means of overriding the behavior of one or more phones without changing the baseline configuration files for an entire system For more information refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice If you have a requirement for different application loads on different phones on the same boot server you can create a variable in the master configuration file that is replaced by the MAC address of each phone when it reboots An example is shown below lt xml version 1 0 standalone yes gt lt Default Master SIP Configuration F
272. ration Files These settings control the activation or deactivation of a feature at run time In the table below x is the feature number Attribute Permitted Values Interpretation feature x name presence messaging directory calllist ring download calllist received calllist placed calllist missed url dialing call park group call pickup directed call pickup last call return acd login logout acd agent available nway conference call recording corporate directory These are features offered on the phone presence is the presence feature including management of buddies and own status messaging is the instant messaging feature directory is the local directory feature calllist is the locally controlled call lists ring download is run time downloading of ringers calllist received is the received calls list feature the calllist feature must be enabled for this feature to be available calllist placed is the placed calls list feature the calllist feature must be enabled for this feature to be available calllist missed is the missed calls list feature the calllist feature must be enabled for this feature to be available url dialing controls whether URL name dialing is available from a private line it is never avai
273. ration attribute is defined as follows Permitted Attribute Values Default Interpretation call shared disableDivert 0 1 1 If set to 1 disable diversion feature for shared lines Note This feature is disabled on most call servers call shared seizeFailReorder 0 1 1 If set to 1 play re order tone locally on shared line seize failure Configuration Files Permitted Attribute Values Default Interpretation call shared oneTouchResume 0 1 0 If set to 1 when a shared line has a call on hold the remote user can press that line and resume the call If more than one call is on hold on the line then the first one will be selected and resumed automatically If set to 0 pressing the shared line will bring up a list of the calls on that line and the user can select which call the next action should be applied to Note This parameter affects the SoundStation IP 4000 phone For other phones a quick press and release of the line key will resume a call whereas pressing and holding down the line key will show a list of calls on that line call shared exposeAutoHolds 0 1 0 If set to 1 on a shared line when setting up a conference a re INVITE will be sent to the server If set to 0 no re INVITE will be sent to the server Hold Local Reminder lt hold gt lt localReminder gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpret
274. rectory will be permitted to consume dir local volatile 4meg 0 1 0 Applies to platforms with 4 Mbytes of flash memory If set to 1 use volatile storage for phone resident copy of the directory to allow for larger size dir local nonVolatile maxSize 4meg 1 to 50 50 Applies to platforms with 4 Mbytes of flash memory Maximum size in Kbytes of non volatile storage that the directory will be permitted to consume dir local volatile maxSize 1 to 100 100 Maximum size in Kbytes of volatile storage that the directory will be permitted to consume Note For the SoundPoint IP 650 platform this value is internally replaced by 2X the value dir local volatile 8meg 0 1 0 Attribute applies only to platforms with 8 Mbytes of flash memory If set to 1 use volatile storage for phone resident copy of the directory to allow for larger size Configuration Files Attribute Permitted Values Default Interpretation dir local nonVolatile maxSize 8meg 1 to 100 100 Attribute applies only to platforms with 8 Mbytes of flash memory This is the maximum size of non volatile storage that the directory will be permitted to consume dir local readonly 0 1 Specifies whether or not local contact directory is read only If set to O or Null the local contact directory is editable If set to 1 the local contact directory is read only dir search field 0 1 Null
275. red users is displayed visually and is updated in real time in the Buddies display screen or for speed dial entries on the phone s idle display Users can block others from monitoring their phones and are notified when a change in monitored status occurs Phone status changes are broadcast automatically to monitoring phones when the user engages in calls or invokes do not disturb The user can also manually specify a state to convey overriding and perhaps masking the automatic behavior Notification when a change in monitored status occurs will be available in a subsequent release The presence feature works differently when Microsoft Live Communications Server 2005 is used as the call server For more information refer to the following section Microsoft Live Communications Server 2005 Integration Configuration changes can performed centrally at the boot server Central boot server XML file lt Ethernet The lt bw gt 0 lt bw gt buddy watching and lt bb gt 0 lt bb gt buddy address gt directory blocking elements in the lt Ethernet address gt directory xml file xml dictate the Presence aspects of directory entries For more information refer to Local Contact Directory on page 4 9 Local Local Phone User The user can edit the directory contents The Watch Buddy and Interface Block Buddy fields control the buddy behavior of contacts Changes will be stored in the phone s flash file system and backed u
276. rename this file to lt Ethernet address gt cfg for each phone gt lt Revision 1 14 Date 2005 07 27 18 43 30 gt lt APPLICATION APP_FILE_PATH sip 1d CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG FILE DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY gt Master configuration files contain six XML attributes APP_FILE_PATH The path name of the application executable It can have a maximum length of 255 characters This can be a URL with its own protocol user name and password for example http usr pwd server dir sip d Note Warning Configuration Files e CONFIG_FILES A comma separated list of configuration files Each file name has a maximum length of 255 characters and the list of file names has a maximum length of 2047 characters including commas and white space Each configuration file can be specified as a URL with its own protocol user name and password for example ftp usr pwd server dir phone2034 cfg e MISC_FILES A comma separated list of other required files Dictionary resource files listed here will be stored in the phone s flash file system So if the phone reboots at a time when the boot server is unavailable it will still be able to load the preferred language On the SoundPoint IP 500 there is insufficient room for a language file Specifying one will cause a reboot loop e LOG_FILE_DIRECTORY An alternative directory to use fo
277. res e Call Log e Call Timer e Call Waiting e Called Party Identification e Calling Party Identification e Missed Call Notification Administrator s Guide SoundPoint IP SoundStation IP Connected Party Identification e Context Sensitive Volume Control e Customizable Audio Sound Effects e Message Waiting Indication e Distinctive Incoming Call Treatment e Distinctive Ringing e Distinctive Call Waiting e Do Not Disturb e Handset Headset and Speakerphone e Local Contact Directory e Local Digit Map e Microphone Mute Soft Key Activated User Interface e Speed Dial Time and Date Display e Idle Display Animation e Ethernet Switch e Graphic Display Backgrounds This section also provides information for making configuration changes for the following basic call management features e Automatic Off Hook Call Placement e Call Hold e Call Transfer e Local Centralized Conferencing e Call Forward e Directed Call Pick Up e Group Call Pick Up e Call Park Retrieve e Last Call Return Call Log Note Configuring Your System The phone maintains a call log The log contains call information such as remote party identification time and date and call duration It can be used to redial previous outgoing calls return incoming calls and save contact information from call log entries to the contact directory The call log is stored in volatile memory and is maintained automatically by the phone in t
278. rformed centrally at the boot server or locally Central Configuration file Specify whether diversion should be disabled on shared lines boot server sip cfg e For more information refer to Shared Calls lt shared gt on page A 60 Specify line seize subscription period For more information refer to Server lt server gt on page A 7 Specify standard or non standard behavior for processing line seize subscription for mutual exclusion feature For more information refer to Special Events lt specialEvent gt on page A 15 Configuration file Specify per registration line type private or shared barge in phonet cfg capabilities and line seize subscription period if using per registration servers A shared line will subscribe to a server providing call state information For more information refer to Registration lt reg gt on page A 89 Specify per registration whether diversion should be disabled on shared lines For more information refer to Diversion lt divert gt on page A 96 Local Web Server Specify line seize subscription period if enabled Navigate to http lt phonelPAddress gt appConf htm se Specify standard or non standard behavior for processing line seize subscription for mutual exclusion feature Navigate to http lt phonelPAddress gt appConf htm ls Specify per registration line type private or shared and line seize subscription period if using per registration servers
279. rtising or publicity pertaining to distribution of the software without specific written prior permission M I T makes no representations about the suitability of this software for any purpose It is provided as is without express or implied warranty OpenSSL The OpenSSL toolkit stays under a dual license i e both the conditions of the OpenSSL License and the original SSLeay license apply to the toolkit See below for the actual license texts Actually both licenses are BSD style Open Source licenses In case of any license issues related to OpenSSL please contact openssl core opensslL org OpenSSL License Copyright c 1998 2003 The OpenSSL Project All rights reserved Redistribution and use in source and binary forms with or without modification are permitted provided that the following conditions are met 1 Redistributions of source code must retain the above copyright notice this list of conditions and the following disclaimer 2 Redistributions in binary form must reproduce the above copyright notice this list of conditions and the following disclaimer in the documentation and or other materials provided with the distribution 3 All advertising materials mentioning features or use of this software must display the following acknowledgment This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit http www openssl org Administrator s Guide SoundPoint IP SoundStation IP 4
280. rty Software and or other materials provided with the distribution 3 All advertising materials mentioning features or use of this software must display the following acknowledgement This product includes cryptographic software written by Eric Young eay cryptsoft com The word cryptographic can be left out if the routines from the library being used are not cryptographic related 4 If you include any Windows specific code or a derivative thereof from the apps directory application code you must include an acknowledgement This product includes software written by Tim Hudson tjh cryptsoft com THIS SOFTWARE IS PROVIDED BY ERIC YOUNG AS IS AND ANY EXPRESS OR IMPLIED WARRANTIES INCLUDING BUT NOT LIMITED TO THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT INDIRECT INCIDENTAL SPECIAL EXEMPLARY OR CONSEQUENTIAL DAMAGES INCLUDING BUT NOT LIMITED TO PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES LOSS OF USE DATA OR PROFITS OR BUSINESS INTERRUPTION HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY WHETHER IN CONTRACT STRICT LIABILITY OR TORT INCLUDING NEGLIGENCE OR OTHERWISE ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE The licence and distribution terms for any publicly available version or derivative of this code cannot be changed i e this code cann
281. rwrite the font used for SoundPoint IP 500 soft keys for ASCII the name should be fontPropSoftkey_10_U0000_U00FF fnt e toadd support for a new font that will be used everywhere and that is not currently supported For example for the Eastern Central European Czech language this is Unicode range 100 17F the name could be fontCzechIP500_10_U0100_U01FF fnt and fontCzechIP600_19_U0100_U01FF fnt When defining a single fon file there is a need for a font delimiter currently Copyright Polycom Canada Ltd is used as an embedded delimiter but this can be configured using font delimiter The font delimiter is important to retrieve the different scrambled fnt blocks This font delimiter must be placed in the copyright attribute of the fnt header fon files are useful if you want to include support for a large number of font ranges at once otherwise if simply adding or changing a few fonts currently in use multiple fnt files are recommended since they are easier to work with individually This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation font delimiter string up to 256 ASCII characters Null Delimiter required to retrieve different grouped fnt blocks This attribute also includes e IP _330 font lt IP_330 gt e IP_400 font lt IP_400 gt e IP_500 font lt IP_500 gt e IP 600 font lt IP_600 gt IP_330 font lt IP
282. s Yok eae dame See 4 18 Local Centralized Conferencing 0000 00005 4 19 Call Forward sis prany ien iio ed ae h pees aoe wk bs BAS 4 20 Directed Call Pick Up 0 cece eee cee eee eee 4 21 Group Call Pick Up cn Gviean a E eed Chew eV esa Chee ess 4 21 Call Park Retrieve 0 0 0 0 ccc cece tenes 4 22 Last Call Retty is sso tse tesde N pas acai ees eas ete WI wholes 4 22 Setting Up Advanced Features 0 0 000 e eee eee eee 4 22 Configurable Feature Keys 0 cece eee 4 23 Multiple Line Keys per Registration 000 4 24 Multiple Call Appearances 0 cece e eee eee 4 25 Shared Call Appearances cece eee eee eee 4 25 Bridged Line Appearance 00000 e eee eee eee 4 26 Busy Lamp Field sie ia a lain A E E 4 27 Customizable Fonts and Indicators 0 000000 4 28 Instant Messaging 6 eects 4 28 Contents Multilingual User Interface 0 00 e eee eee 4 29 Downloadable Fonts 0 00 4 30 Synthesized Call Progress Tones 0 00 c cece eee eee 4 30 Microbr wser seriis eiieeii ere se EE e e le ea Hite sled octet 4 30 Real Time Transport Protocol Ports 0 0000 4 31 Network Address Translation 00 e eee eee 4 32 Corporate Directory 0 cece eee 4 32 Recording and Playback of Audio Calls 0 0055 4 35 Voice Mail Integration
283. s can be tuned for the application to provide more or less detail on specific components of the phone s software For example if you are troubleshooting a SIP signaling issue you are not likely interested in DSP events Logging levels are adjusted in the configuration files or via the web interface You should not modify the default logging levels unless directed to by Polycom Technical Support Inappropriate logging levels can cause performance issues on the phone In addition to logging events the phone can be configured to automatically execute command line instructions at specified intervals that output run time information such as memory utilization task status or network buffer contents to the log file These techniques should only be used in consultation with Polycom Technical Support Application Logging Options Each of the components of the application software is capable of logging events of different severity This allows you to capture lower severity events in one part of the application while still only getting high severity event for other components The parameters for log level settings are found in the sip cfg configuration file They are log level change module_name Log levels range from 1 to 6 1 for the most detailed logging 6 for critical errors only There are currently 27 different log levels that can be adjusted to assist with the investigation of different problems When testing is complete remember to ret
284. s feature should be put in separate configuration files to simplify maintenance Do not add them to existing configuration files such as sip cfg One new configuration file will be required for parameters that should apply to all phones and individual configuration files will be required for phone specific parameters such as SIP registration information The global device set parameter must be enabled when the initial installation is done and then it should be disabled This prevents subsequent reboots by individual phones triggering a reset of parameters on the phone that may have been tweaked since the initial installation A 105 Administrator s Guide SoundPoint IP SoundStation IP Caution This feature is very powerful and should be used with caution For example an incorrect setting could set the IP Address of multiple phones to the same value Note that some parameters may be ignored for example if DHCP is enabled it will still override the value set with device net ipAddress Individual parameters are checked to see whether they are in range however the interaction between parameters is not checked If a parameter is out of range an error message will appear in the log file and parameter will not be used Incorrect configuration could cause phones to get into a reboot loop For example server A has a configuration file that specifies that server B should be used which has a configuration file that specifies that server
285. s support of the leading protocols and industry partners makes it a future proof choice Field upgradeable Upgrade SoundPoint IP SoundStation IP as standards develop and protocols evolve Extends the life of the phone to protect your investment Administrator s Guide SoundPoint IP SoundStation IP Application flexibility for call management and new telephony applications e Large LCD Easy to use easily readable and intuitive interface Support of rich application content including multiple call appearances presence and instant messaging and XML services 4 line x 20 character monochrome LCD for the SoundPoint IP 301 102 x 23 pixel graphical LCD for the SoundPoint IP 320 330 160 x 80 pixel graphical grayscale LCD for the SoundPoint IP 501 320 x 160 pixel graphical grayscale LCD for the SoundPoint IP 550 560 600 601 650 supports Asian characters 248 x 68 pixel graphical LCD for the SoundStation IP 4000 e Dual auto sensing 10 100baseT Ethernet ports Leverages existing infrastructure investment No re wiring with existing CAT 5 cabling Simplifies installation e Power over Ethernet PoE port Unused pairs on Ethernet port pairs are used to deliver power to the phone via a wall adapter allowing fewer wires to desktop Optional accessory cable for CiscoR Inline Powering and IEEE 802 3af on the SoundPoint IP 301 and SoundPoint IP 501 Built in PoE on the SoundPoint IP 550 560 600 601 and 650 auto sensing
286. s to the phone the secure HTTPS protocol is not available To guarantee software integrity the bootROM will only download cryptographically signed bootROM or application images For HTTPS widely recognized certificate authorities are trusted by the phone and custom certificates can be added refer to Trusted Certificate Authority List on page C 1 Modifying the Network Configuration You can access the network configuration menu During bootROM Phase The network configuration menu is accessible during the auto boot countdown of the bootROM phase of operation Press the Setup soft key to launch the main menu During Application Phase The network configuration menu is accessible from the phone s main menu Select Menu gt Settings gt Advanced gt Admin Settings gt Network Configuration Advanced Settings are locked by default Enter the administrator password to unlock The factory default password is 456 Phone network configuration parameters may be modified by means of e Main Menu DHCP Menu e Server Menu Ethernet Menu e Syslog Menu Use the soft keys the arrow keys the Select and Delete keys to make changes Certain parameters are read only due to the value of other parameters For example if the DHCP Client parameter is enabled the Phone IP Addr and Subnet Mask parameters are dimmed or not visible since these are guaranteed to be supplied by the DHCP server mandatory DHCP parameters and the statically assig
287. s will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Network Address Translation The phone can work with certain types of network address translation NAT The phone s signaling and RTP traffic use symmetric ports the source port in transmitted packets is the same as the associated listening port used to receive packets and the external IP address and ports used by the NAT on the phone s behalf can be configured on a per phone basis Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify the external NAT IP address and the ports to be used for boot server sip cfg signaling and RTP traffic For more information refer to Network Address Translation lt nat gt on page A 102 Local Web Server Specify the external NAT IP address and the ports to be used for if enabled signaling and the RTP traffic Navigate to http lt phonelPAddress gt netConf htm na Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Corporate Directory Note This feature requires a license key fo
288. se settings control the performance of the speakerphone acoustic echo suppressor Polycom recommends that you do not change these values POLYCOM Attribute Default voice aes hs enable 0 voice aes hs duplexBalance voice aes hd enable oO voice aes hd duplexBalance oO voice aes hf enable voice aes hf duplexBalance 0 voice aes hf duplexBalance 1 voice aes hf duplexBalance 2 voice aes hf duplexBalance 3 voice aes hf duplexBalance 4 voice aes hf duplexBalance 5 voice aes hf duplexBalance 6 voice aes hf duplexBalance 7 voice aes hf duplexBalance 8 Nl ola Ri a a oal n wn voice aes hf duplexBalance IP_4000 0 oO voice aes hf duplexBalance IP_4000 1 voice aes hf duplexBalance IP_4000 2 Al Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice aes hf duplexBalance IP_4000 3 voice aes hf duplexBalance IP_4000 4 voice aes hf duplexBalance IP_4000 5 voice aes hf duplexBalance IP_4000 7 7 6 5 voice aes hf duplexBalance IP_4000 6 4 3 2 voice aes hf duplexBalance IP_4000 8 Background Noise Suppression lt ns gt These settings control the performance of the transmit background noise suppression feature Polycom recommends that you do not change these values we POLYCOM Attribute Defa
289. servers TCPpreferred or tr Neat specified in sip cfg in Server lt server gt on UDPOnly or AT TLS or paga TCPOnly Note If the reg x server y address parameter is non Null all of the reg x server y xxx reg x server y expires positive integer Null parameters will override the parameters specified in sip cfg in Server lt server gt on reg x server y register 0 1 Null page A 7 reg x server y expires overlap positive integer 60 Note If the reg x server y address parameter minimum 5 is non Null it takes precedence even if the maximum 65535 DHCP server is available s Note TLS is not supported on SoundPoint IP reg x server y retry TimeOut Null or Null 300 and 500 phones non negative integer reg x server y retryMaxCount Null or Null non negative integer reg x server y expires lineSeize positive integer Null reg x server y Ics 0 1 0 This attribute overrides the reg x 1cs If set to 1 the Microsoft Live Communications Server is supported for registration x reg x acd login logout 0 1 0 If both parameters are set to 1 fora i registration the ACD feature will be enabled reg x acd agent available 0 1 0 for that registration reg x ringlype 1 to 22 2 The ringer to be used for calls received by this registration Default is the first non silent ringer reg x lineKeys 1 to max 1 max the number of line keys on the phone max 1 on SoundStation IP 4000 max 2 on IP 301 320 330 430 max 3 on IP 501 max 4 on IP
290. size limits to be used by the Microbrowser when it is selected to provide services The Microbrowser is supported on the SoundPoint IP 430 501 550 560 601 and 650 and the SoundStation IP 4000 phones This configuration attribute is defined as follows formed valid HTTP URL Length up to 255 characters Attribute Permitted Values Default Interpretation mb proxy Null or Null Address of the desired HTTP proxy to be used domain name or Default by the Microbrowser If blank normal unproxied IP address in the port HTTP is used by the Microbrowser format 8080 lt address gt lt port gt This attribute also includes e Idle Display lt idleDisplay gt e Main Browser lt main gt e Browser Limits lt limits gt Idle Display lt idleDisplay gt The Microbrowser can be used to create a display that will be part of the phone s idle display These settings control the home page and the refresh rate Attribute Permitted Values Default Interpretation mb idleDisplay home Null or any fully Null URL used for Microbrowser idle display home page For example http www example com xhtml frontpage cgi pa ge home If empty there will be no Microbrowser idle display feature Note that the Microbrowser idle display will displace the idle display indicator refer to ind idleDisplay enabled in Indicators lt ind gt on page A 71 Note If ind idleDisplay enabled is enabled miscellaneous XML errors
291. sport Web Server lt httpd gt The phone contains a local web server for user and administrator features This can be disabled for applications where it is not needed or where it poses a security threat The web server supports both basic and digest authentication The authentication user name and password are not configurable for this release This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation httpd enabled 0 1 1 If set to 1 the HTTP server will be enabled Administrator s Guide SoundPoint IP SoundStation IP This attribute a lso includes Configuration lt cfg gt Configuration lt cfg gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation httpd cfg enabled 0 1 1 If set to 1 the HTTP server configuration interface will be enabled httpd cfg port 1 65535 80 Port is 80 for HTTP servers Care should be taken when choosing an alternate port Call Handling Configuration lt call gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation call rejectBusyOnDnd 0 1 1 If set to 1 reject all incoming calls with the reason busy if do not disturb is enabled Note This attribute is ignored when the line is configured as shared The reason being that even though one party has
292. sr pwd server refer to Server Menu on page 3 9 The boot server address can be an IP address domain string name or URL The boot server address can also be obtained through DHCP Configuration file names in the lt Ethernet address gt cfg file can include a transfer protocol for example https usr pwd server dir file cfg If a user name and password are specified as part of the server address or file name they will be used only if the server supports them A URL should contain forward slashes instead of back slashes and should not contain spaces Escape characters are not supported If a user name and password are not specified the Server User and Server Password will be used refer to Server Menu on page 3 9 Protocol used by Protocol used by bootROM SIP Application 301 320 330 430 301 320 330 430 Specified 501 550 560 600 501 550 560 600 Protocol 601 650 4000 601 650 4000 FTP FTP FTP TFTP TFTP TFTP HTTP HTTP HTTP HTTPS HTTP HTTPS There are two types of FTP methods active and passive As of SIP 1 5 and bootROM 3 0 the SIP application is no longer compatible with active FTP At that time secure provisioning was implemented Setting up Your System Note Setting Option 66 to tftp 192 168 9 10 has the effect of forcing a TFTP download Using a TFTP URL for example tftp provserver polycom com has the same effect For downloading the bootROM and application image
293. sumed that all registrations use the same server Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server For more information refer to Local lt local gt on page A 6 Server lt server gt on page A 7 and Registration lt reg gt on page A 89 Automatic Call Distribution The phone allows automatic call distribution ACD login and logout This feature depends on support from a SIP server Configuration changes can performed centrally at the boot server Central Configuration file Turn this feature on or off boot server sip cfg e For more information refer to Feature lt feature gt on page A 83 Configuration file Enable this feature per registration phonet cfg e For more information refer to Registration lt reg gt on page A 89 Configuring Your System The phone also supports ACD agent available and unavailable This feature depends on support from a SIP server Configuration changes can performed centrally at the boot server Central boot server Configuration file Turn this feature on or off sip cfg e For more information refer to Feature lt feature gt on page A 83 Configuration file Enable this feature per registrati
294. t 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 divertcontact 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
295. t bitmaps gt section of sip cfg find the end of each model s bitmap list and add your bitmap to the end do not include the bmp extension lt bitmaps gt lt IP_300 gt bitmap IP_330 66 name 1ogo 330 gt bitmap IP_500 66 name 1ogo 500 gt bitmap IP_600 70 name logo 600 gt lt IP_330 lt IP_500 a lt IP_600 a lt IP_4000 bitmap IP_4000 70 name logo 4000 gt Miscellaneous Administrative Tasks lt bitmaps gt Next enable the idle display feature and modify the idle display animation for each model to point to your bitmap again without the bmp extension lt indicators ind idleDisplay enabled 1 gt lt Animations gt lt IP_300 gt lt IP_300 gt lt IP_330 gt lt IDLE_DISPLAY ind anim IP_3300 38 frame 1 bitmap logo 330 ind anim IP_330 38 frame 1 duration 0 gt lt IP_330 gt lt IP_500 gt lt IDLE_DISPLAY ind anim IP_500 38 frame 1 bitmap logo 500 ind anim IP_500 38 frame 1 duration 0 gt lt IP_500 gt lt IP_600 gt lt IDLE_DISPLAY ind anim IP_600 38 frame 1 bitmap logo 600 ind anim IP_600 38 frame 1 duration 0 gt lt IP_600 gt lt IP_4000 gt lt IDLE_DISPLAY ind anim IP_4000 38 frame 1 bitmap logo 4000 ind anim IP_4000 38 frame 1 duration 0 gt lt IP_4000 gt lt Animations gt lt indicators gt BootROM SIP Application Dependencies Not withstanding the hardware backward compatibility mandate there have been times
296. t call waiting types Distinctive Incoming Call Treatment The phone can automatically apply distinctive treatment to calls containing specific attributes Distinctive Ringing The user can select the ring type for each line and the ring type for specific callers can be assigned in the contact directory Do Not Disturb A do not disturb feature is available to temporarily stop all incoming call alerting Graphic Display Backgrounds A picture or design displayed on the background of the graphic display Handset Headset and Speakerphone SoundPoint IP phones come standard with a handset and a dedicated headset connection not supplied The SoundPoint IP 320 330 430 500 501 550 560 600 601 and 650 phones and SoundStation IP 4000 phone are full duplex speakerphones The SoundPoint IP 301 phone is a listen only speakerphone Idle Display Animation All phones except the SoundPoint IP 301 can display a customized animation on the idle display in addition to the time and date Last Call Return The phone allows call server based last call return Local Centralized Conferencing The phone can conference together the local user with the remote parties of two independent calls and can support centralized conferences for which external resources are used such as a conference bridge Local Contact Directory The phone maintains a local contact directory that can be downloaded from the boot server and edited
297. t inter arrival jitter and out of order or lost lost or excessively delayed by the network packets The jitter buffer is adaptive and configurable for different network environments When packets are lost a concealment algorithm minimizes the resulting negative audio consequences Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression For more information refer to Codec Profiles lt audioProfile gt on page A 36 Local Web Server if enabled Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression Navigate to http lt phonelPAddress gt coreConf htm au Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Voice Activity Detection The purpose of voice activity detection VAD is to conserve network bandwidth by detecting periods of relative silence in the transmit data path and replacing that silence efficiently with special packets that indicate silence is occurring For those compression algorithms without an inherent VAD function such as G 711 t
298. t necessary For more information refer to Keys lt key gt on page A 69 For more information on the default feature key layouts refer to Default Feature Key Layouts on page C 10 Multiple Line Keys per Registration More than one Line Key can be allocated to a single registration phone number or line on SoundPoint IP phones The number of Line Keys allocated per registration is configurable Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify the number of line keys to assign per registration boot server phonet cfg e For more information refer to Registration lt reg gt on page A 89 Local Web Server Specify the number of line keys to assign per registration if enabled Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Local Phone User Interface Specify the number of line keys to assign per registration using the SIP Configuration menu Either the Web Server or the boot server configuration files or the local phone user interface should be used to configure registrations not a mixture of these options When the SIP Configuration menu is used
299. t to TCPOnly Only TCP will be used NOTE TLS is not supported on SoundPoint IP 300 and 500 phones volpProt server x expires positive 3600 The phone s requested registration period in integer seconds minimum Note The period negotiated with the server 300 may be different The phone will attempt to re register at the beginning of the overlap period For example if expires 3600 and overlap 60 the phone will re register after 3540 seconds 3600 60 volpProt server x expires overlap positive 60 The number of seconds before the expiration integer time returned by server x at which the phone minimum 5 should try to re register The phone will try to maximum re register at half the expiration time returned 65535 by the server if that value is less than the configured overlap value volpProt server x register 0 1 1 If set to 0 calls can be routed to an outbound proxy without registration volpProt server x retryTimeOut Null or 0 If set to 0 or Null use standard RFC 3261 non negativ signaling retry behavior Otherwise e integer retryTimeOut determines how often retries will be sent Units milliSeconds Finest resolution 100ms Configuration Files Permitted Attribute Values Default Interpretation volpProt server x retryMaxCount Null or 3 If set to O or Null 3 is used retryMaxCount non negativ retries will be attempted before moving on to e integer the next available server vo
300. tact has been configured In the following tables x is the registration number IP 301 320 330 430 x 1 2 IP 501 x 1 3 IP 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 x 1 34 IP 4000 x 1 Attribute Permitted Values Default Interpretation divert x contact ASCII encoded string Null The forward to contact used for containing digits the user all automatic call diversion part of a SIP URL or a string features unless overridden by a that constitutes a valid SIP specific contact of a per call URL 6416 or diversion feature refer to 6416 polycom com below divert x autoOnSpecificCaller 0 1 1 If set to 1 calls may be diverted using the Auto Divert feature of the directory This is a global flag Note If server based call forwarding is enabled this parameter is disabled divert x sharedDisabled 0 1 1 If set to 1 all diversion features on that line will be disabled if the line is configured as shared This attribute also includes e Forward All lt fwd gt e Busy lt busy gt e No Answer lt noanswer gt e Do Not Disturb lt dnd gt Forward All lt fwd gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation divert fwd x enabled 0 1 1 If set to 1 the user will be able to enable universal call forwarding through the soft key menu Note If server based call forwarding is enabled this parameter is enab
301. tation prov fileSystem rfs0 minFreeSpace prov fileSystem ffs0 4meg minFreeSpace prov fileSystem ffs0 2meg minFreeSpace prov fileSystem ffs0 8meg minFreeSpace 5 512 5 420 48 512 Minimum free space in Kbytes to reserve in the file system when downloading files from the boot server Note Polycom recommends that you do not change these parameters Note For the SoundPoint IP 650 platform prov fileSystem ffs0 8meg m inFreeSpace s internally replaced by 2X the value prov polling enabled 0 1 If set to 1 automatic periodic boot server polling for upgrades is enabled prov polling mode abs rel abs Polling mode is absolute or relative prov polling period integer greater than 3600 86400 Polling period in seconds Rounded up to the nearest number of days in abs mode Measured relative to boot time in rel mode prov polling time Format is hh mm 03 00 Only used in abs mode Polling time RAM Disk lt ramdisk gt This attribute s settings control the phone s internal RAM disk feature Polycom recommends that you do not change these values POLYCOM Administrator s Guide SoundPoint IP SoundStation IP This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation ramdisk enable 0 1 1 If set to 1 RAM disk will be available The RAM disk is
302. tation tone dtmf level 33 to 3 15 Level of the high frequency component of the DTMF digit measured in dBm0 the low frequency tone will be two dB lower tone dtmf onTime positive integer 50 When a sequence of DTMF tones is played out automatically this is the length of time in milliseconds the tones will be generated for this is also the minimum time the tone will be played for when dialing manually even if key press is shorter tone dtmf offTime positive integer 50 When a sequence of DTMF tones is played out automatically this is the length of time in milliseconds the phone will pause between digits this is also the minimum inter digit time when dialing manually tone dtmf chassis masking 0 1 If set to 1 DTMF tones will be substituted with a non DTMF pacifier tone when dialing in hands free mode This prevents DTMF digits being broadcast to other surrounding telephony devices or being inadvertently transmitted in band due to local acoustic echo Note tone dtmf chassis masking should only be enabled when tone dtmf viaRtp is disabled tone dtmf stim pac offHookOnly 0 1 Not currently used Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Default Interpretation tone dtmf viaRtp 0 1 1 If set to 1 encode DTMF in the active RTP stream otherwise DTMF may be encoded within the signaling protocol only
303. te 37 n a 5 Hold 16 Dialpad4 27 SoftKey2 38 n a 6 n a 17 Dialpad5 28 SoftKey1 39 n a 7 Redial 18 Dialpad6 29 ArrowDown 40 n a 8 VolUp 19 Dialpad3 30 n a 41 n a 9 VolDown 20 Dialpad2 31 ArrowUp 42 n a 10 DialpadStar 21 Dialpad1 32 Menu 11 DialpadO 22 n a 33 n a C 10 SoundPoint IP 320 330 Miscellaneous Administrative Tasks ee Sy eB e L Pr line gl va Ro a 0 24 Oo 16 23 Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 Dialpad2 12 n a 23 VolUp 34 Menu 2 Dialpad5 13 SoftKey2 24 VolDown 35 n a 3 Dialpad8 14 ArrowUp 25 Dialpad3 36 n a 4 Dialpad7 15 Select 26 Dialpad6 37 n a 5 Dialpad4 16 ArrowDown 27 Dialpad9 38 n a 6 Dialpad1 17 n a 28 DialpadO 39 n a 7 SoftKey3 18 n a 29 DialpadStar 40 n a 8 Line1 19 Hold 30 MicMute 41 n a 9 ArrowRight 20 Headset 31 SoftKey1 42 n a 10 Line2 21 Handsfree 32 Dial 11 n a 22 DialpadPound 33 ArrowLeft 11 Administrator s Guide SoundPoint IP SoundStation IP SoundPoint IP 430 Key ID Function Key ID Function Key ID Function Key ID Function 1 Line1 12 DialpadPound 23 Messages 34 Softkey3 2 Line2 13 Dialpad9 24 n a 35 Handsfree 3 n a 14 Dialpad8 25 SoftKey4 36 n a 4 ArrowUp 15 Dialpad7 26 Headset 37 n a 5 Hold 16 Dialpad4 27 SoftKey2 38 n a 6 n a 17 Dialpad5 28 SoftK
304. te above time else display time above date User Preferences lt up gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation up headsetMode 0 1 0 If set to 1 the headset will be selected as the preferred transducer after its first use until the headset key is pressed again otherwise hands free will be selected preferentially over the headset up useDirectoryNames If set to 1 the name fields of directory entries which match incoming calls will be used for caller identification display and in the call lists instead of the name provided through network signaling up one TouchVoiceMail 0 1 0 If set to 1 the voice mail summary display is bypassed and voice mail is dialed directly if configured up welcomeSoundEnabled 0 1 1 If set to 1 play welcome sound effect after a reboot Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation up welcomeSoundOnWarmBootE 0 1 0 If set to 1 play welcome sound effect on warm nabled as well as cold boots otherwise only a cold boot will trigger the welcome sound effect up localClockEnabled 0 1 1 If set to 1 display the date and time on the idle display up backlight onIntensity 0 off Null This parameter controls the intensity of the 1 low LCD backlight when it turns on dur
305. te tee S npara tek enews B 9 Third Party Call Control 0 0 2 B 9 SIP for Instant Messaging and Presence Leveraging Extensions B 10 Shared Call Appearance Signaling 00 B 10 Bridged Line Appearance Signaling 0 B 10 C Miscellaneous Administrative Tasks C l Trusted Certificate Authority List 0 0 eee eee eee C 1 Encrypting Configuration Files 0 6 c cece eee eee eee C 3 Changing the Key on the Phone 0 00 cece eee eee C 5 Adding a Background Logo 0 cee eee eee ee C 5 BootROM SIP Application Dependencies 0008 C 7 Migration Dependencies 0 00 c eee nren C 8 Multiple Key Combinations 6 c cece eee ees C 9 Default Feature Key Layouts 0 000 e eee eee eee ee C 10 Administrator s Guide SoundPoint IP SoundStation IP Assigning a VLAN ID Using DHOP icicevisaieaieersbeeretnies ed C 16 Parsing Vendor ID Information 0 0 0002 eee eee C 17 D Third Party Software 0 cece cece cece eee DI Index 666 02 0dbeeSdcanesee es eundhterssne secon Introducing the SoundPoint IP SoundStation IP Family This chapter introduces the SoundPoint IP SoundStation IP family which is supported by the software described in this guide The SoundPoint IP SoundStation IP family provides a powerful yet flexible IP communications solution
306. ted to comply with the TIA 810 A digital telephony standard Polycom recommends that you do not change these values we POLYCOM Attribute Default voice gain rx analog chassis IP_7000 0 voice gain rx analog ringer IP_7000 0 voice gain rx digital chassis IP_7000 0 voice gain rx digital ringer IP_7000 21 voice gain tx analog chassis IP_7000 0 voice gain tx digital chassis IP_7000 6 Receive Equalization lt rxEq gt These settings control the performance of the receive equalization feature Polycom recommends that you do not change these values we POLYCOM Attribute Default voice rxEq hf IP_7000 preFilter enable 0 voice rxEq hf IP_7000 postFilter enable 0 Administrator s Guide Addendum for the SoundStation IP 7000 Transmit Equalization lt txEq gt These settings control the performance of the hands free transmit equalization feature Polycom recommends that you do not change these values we POLYCOM Attribute Default voice txEq hf IP_7000 preFilter enable 0 voice txEq hf IP_7000 postFilter enable 0 Call Handling Configuration lt call gt These new configuration attributes are defined as follows Permitted Attribute Values Default Interpretation call singleKeyPressConference 0 1 0 If set to 1 the conference will be setup after a user presses the Conference soft key or Conference key the first time Also all sound effects
307. ters affect SoundPoint IP 501 phones IP_600 parameters affect SoundPoint IP 550 560 600 601 and 650 phones IP_4000 parameters affect SoundStation IP 4000 phones Attribute Permitted Values Interpretation bitmap x y name The name of a bitmap This is the name of a bitmap to be used for creating an to be used animation If the bitmap is to be downloaded from the boot server its name must e Be different from any name already in use in sip cfg Match the name of the corresponding lt fileName gt bmp to be retrieved from the boot server Indicators lt ind gt The following indicators are used by the phone e Animations lt anim gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt e Patterns lt pattern gt e Classes lt class gt e Assignments Administrator s Guide SoundPoint IP SoundStation IP This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation ind idleDisplay mode 1 default 2 Null The idle display animation screen layouts 3 For example for the SoundPoint IP 330 320 e Ifsetto 1 or Null the idle display animation size is 87 x 11 pixels e If set to 2 the idle display animation size is 87 x 22 pixels e If set to 3 the idle display animation size is 102 x 22 pixels ind idleDisplay enabled 0 1 0 If set to 1 the idle display may support presentation of a c
308. tion refer to Registration lt reg gt on page A 89 XML File lt Ethernet address gt directory xml This file can be created manually using an XML editor For more information refer to Local Contact Directory on page 4 9 Local Local Phone User Interface The user can edit the ring types selected for each line under the Settings menu The user can also edit the directory contents Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server Distinctive Call Waiting The volpProt SIP alertInfo x value and volpProt SIP alertInfo x class fields can be used to map calls to distinct call waiting types currently limited to two styles Configuration changes can performed centrally at the boot server Central boot server Configuration file sip cfg Specify the mapping of Alert Info strings to call waiting types For more information refer to Alert Information lt alertInfo gt on page A 14 Administrator s Guide SoundPoint IP SoundStation IP Do Not Disturb Note A Do Not Disturb DND feature is available to temporarily stop all incoming call alerting Calls can optionally be treated as though the phone is busy while DND is enabled DN
309. tion IP 6000 conference phone are the same as the SoundStation IP 4000 conference phone The SoundStation IP 6000 conference phone default key layout is as follows 2 Gy 57 6 40 CoP Cr Ga We CTD Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 Dialpad1 12 n a 23 Select 34 n a 2 Dialpad2 13 Dialpad7 24 n a 35 n a 3 Dialpad3 14 Dialpad8 25 SoftKey3 36 n a 4 VolUp 15 Dialpad9 26 Exit 37 n a Administrator s Guide Addendum for the SoundStation IP 6000 Key ID Function Key ID Function Key ID Function Key ID Function 5 Handsfree 16 MicMute 27 Menu 38 n a 6 n a 17 ArrowUp 28 SoftKey1 39 n a 7 Dialpad4 18 n a 29 SoftKey2 40 n a 8 Dialpad5 19 DialpadStar 30 n a 41 n a 9 Dialpad6 20 Dialpad0 31 n a 42 n a 10 VolDown 21 DialpadPound 32 n a 11 ArrowDown 22 Redial 33 n a Administrator s Guide Addendum for the SoundStation IP 6000
310. tion file changes C 6 background noise suppression 4 51 background noise suppression lt ns gt A 42 basic logging A 77 basic protocols header support B 4 hold implementation B 9 request support B 3 response support B 6 RFC and Internet draft support B 2 transfer B 9 basic TCP IP A 52 blind transfers 4 18 BNS See also background noise suppression boot failure messages 5 7 boot server security policy 3 14 boot servers deploying phones 3 15 redundant 3 13 security policy 3 14 setting up 3 13 bootROM 2 3 bootROM and application wrapper 2 5 Index 2 bootROM error messages 5 2 bootROM tasks 2 3 bootROM SIP application dependencies C 7 bridged line appearance signaling B 10 bridged line appearances 4 26 browser limits A 88 busy lt busy gt A 97 busy lamp field 4 27 C call control lt callControl gt A 50 call control third party B 9 call forwarding 4 20 A 96 call handling configuration lt call gt A 58 call hold 4 17 call log 4 3 call park retrieve 4 22 call progress patterns A 30 call progress tones synthesized 4 30 call timer 4 3 call transfer 4 18 call waiting 4 3 called party identification 4 4 calling party identification 4 4 calls lt calls gt A 93 central provisioning overview 2 6 changing the key on the phone C 5 charge for software feature 4 19 4 32 4 35 4 52 A 84 chord sets lt chord gt A 26 codec preferences lt codecPref gt A 35 codec profiles lt audioProfile
311. to 10HD 10FD 100HD The network speed over the Ethernet 100FD 1000FD The default value is Auto Note The LAN Port Mode and PC Port Mode parameters are only available on SoundPoint IP 330 430 550 560 601 and 650 phones HD means half duplex and FD means full duplex It is recommended that you leave the LAN and PC parameters set to Auto Only the SoundPoint IP 560 supports the LAN Port Mode and PC Port Mode setting of 1000FD Administrator s Guide SoundPoint IP SoundStation IP Syslog Menu Syslog is a standard for forwarding log messages in an IP network The term syslog is often used for both the actual syslog protocol as well as the application or library sending syslog messages The syslog protocol is a very simplistic protocol the syslog sender sends a small textual message less than 1024 bytes to the syslog receiver The receiver is commonly called syslogd syslog daemon or syslog server Syslog messages can be sent through UDP TCP or TLS The data is sent in cleartext Syslog is supported by a wide variety of devices and receivers Because of this syslog can be used to integrate log data from many different types of systems into a central repository The syslog protocol is defined in RFC 3164 For more information on syslog go to http www ietf org rfc rfc3164 txt number 3164 The following syslog configuration parameters can be modified on the Syslog menu
312. to 1 the Contact URI is used If set to 0 the TO URI is used previous behavior volpProt SIP serverFeatureControl cf 0 1 If set to 1 server based call forwarding is enabled The call server has control of call forwarding If set to 0 server based call forwarding is not enabled This is the old behavior Administrator s Guide SoundPoint IP SoundStation IP Attribute Permitted Values Default Interpretation volpProt SIP serverFeatureControl dn d 0 1 0 If set to 1 server based DND is enabled The call server has control of DND If set to 0 server based DND is not enabled This is the old behavior volpProt SIP authOptimizedInFailover 0 1 If set to 1 when failover occurs the first new SIP request is sent to the server that sent the proxy authentication request If set to 0 when failover occurs the first new SIP request is sent to the server with the highest priority in the server list If reg x auth optimizedInFailover set to Null this attribute is checked If volpProt SIP authOptimizedInFailover is Null then this feature is disabled If both attributes are set the value of reg x auth optimizedInFailover takes precedence volpProt SIP csta 0 1 If set to 1 uaCSTA is enabled This attribute also includes e Outbound Proxy lt outboundProxy gt e Alert Information lt alertInfo gt e Request Validation lt requestVali
313. to Call Waiting lt callWaiting gt on page A 95 For related configuration changes refer to Customizable Audio Sound Effects on page 4 5 Administrator s Guide SoundPoint IP SoundStation IP Called Party Identification The phone displays and logs the identity of the remote party specified for outgoing calls This is the party that the user intends to connect with There are no related configuration changes Calling Party Identification The phone displays the caller identity derived from the network signalling when an incoming call is presented if the information is provided by the call server For calls from parties for which a directory entry exists the local name assigned to the directory entry may optionally be substituted Configuration changes can performed centrally at the boot server or locally Central Configuration File Specify whether or not to use directory name substitution boot server sip cfg e For more information refer to User Preferences lt up gt on page A 23 Local Web Server Specify whether or not to use directory name substitution if enabled Navigate to http lt phonelPAddress gt coreConf htm us Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the bo
314. tory e Recording and Playback of Audio Calls This section also provides information for making configuration changes for the following advanced call server features e Voice Mail Integration e Multiple Registrations e Automatic Call Distribution e Server Redundancy e Presence e Microsoft Live Communications Server 2005 Integration Configurable Feature Keys Note All key functions can be changed from the factory defaults The scrolling timeout for specific keys can be configured No feature keys on the SoundStation IP 4000 can be remapped Since there is no Redial key on the SoundPoint IP 330 320 phone the redial function cannot be remapped The rules for remapping of key functions are e The phone keys that have removable key caps can be mapped to the following Any function that is implemented as a removable key cap on any of the phones Directories Applications Conference Transfer Redial Menu Messages Do Not Disturb Call Lists Aspeed dial Null Administrator s Guide SoundPoint IP SoundStation IP e The phone keys without removable key caps cannot be remapped These include Any keys on the dial pad Volume control Handsfree Mute Headset Hold Navigation Cluster Configuration changes can performed centrally at the boot server Central boot server Configuration File sip cfg Set the key scrolling timeout key functions and sub pointers for each key usually no
315. tribution Zip File e Configurable Feature Keys e Multilingual User Interface e Single Registration e Audio Codecs e Voice Quality Monitoring Configuration file changes are described in Configuration File Changes on page 1 3 The default SIP key layout for the SoundPoint IP 6000 conference phone is shown in Multiple Key Combinations and Default Key Layout on page 1 6 For more information refer to the Release Notes for the SIP Application Version 3 0 2 For more information on the SoundPoint IP 6000 conference phone refer to the User Guide at http www polycom com support voip Administrator s Guide Addendum for the SoundStation IP 6000 New or Changed Features Distribution Zip File As well as the sip Id file in the archive there is a separate file for the SoundPoint IP 6000 conference phone called 3111 15600 001 sip Id Configurable Feature Keys No feature keys on the SoundStation IP 6000 can be remapped Multilingual User Interface Support for major western European languages is included and additional languages can be easily added Support for Asian languages Chinese Japanese and Korean is also included and will display on the SoundStation IP 6000 s higher resolution display Downloadable fonts are not supported on the SoundStation IP 6000 Single Registration The SoundStation IP 6000 phone supports a single registration When the phone is unable to register with the call control server the icon
316. ts If set to 0 or Null 5060 is used for the local port but it is not advertised in the SIP signaling If set to some other value that value is used for the local port and it is advertised in the SIP signaling Configuration Files Server lt server gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt server dhcp available 0 1 0 If set to 1 check with the DHCP server for SIP server IP address If set to 0 do not check with DHCP server volpProt server dhcp option 128 to 255 Null Option to request from the DHCP server if volpProt server dhcp available 1 There is no default value for this parameter it must be filled in with a valid value Note If the reg x server y address parameter in Registration lt reg gt on page A 89 is non Null it takes precedence even if the DHCP server is available volpProt server dhcp type 0 1 Null If set to 0 IP request address If set to 1 request string Type to request from the DHCP server if volpProt server dhcp available 1 There is no default value for this parameter it must be filled in with a valid value volpProt server x address dotted deci Null IP address or host name and port of a SIP mal IP server that accepts registrations Multiple address or servers can be listed starting with x 1 2 host name for fault tolerance volpProt server x port 0 Null 1 to Null Nore if a reg x
317. ttrs srtp usb efk clink Idap and peer log render level 0 6 1 Specifies the lowest class of event that will be rendered to the log files This is the output filter from the internal memory based log system The log render level maps to syslog severity as follows 0 gt SeverityDebug 7 1 gt SeverityDebug 7 2 gt Severitylnformational 6 3 gt Severitylnformational 6 4 gt SeverityError 3 5 gt SeverityCritical 2 6 gt SeverityEmergency 0 7 gt SeverityNotice 5 For more information refer to Syslog Menu on page 3 12 log render type 0 2 2 Refer to above table for timestamp type log render realtime 0 1 1 Set to 1 Note Polycom recommends that you do not change this value log render stdout 0 1 1 Set to 1 Note Polycom recommends that you do not change this value log render file 0 1 1 Set to 1 Note Polycom recommends that you do not change this value Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation log render file size positive 16 Maximum local application log file integer 1 to size in Kbytes When this size is 179 5 exceeded the file is uploaded to the boot server and the local copy is erased log render file upload period positive 172800 Time in seconds between log file integer uploads to the boot server Note The log file will not be uploaded if no ne
318. turned on DND the other person people sharing that line do not necessarily want all calls to that number diverted away Note If server based DND is enabled this parameter is disabled call enableOnNotRegistered 0 1 If set to 1 calls will be allowed when the phone is not successfully registered otherwise calls will not be permitted without a valid registration call offeringTimeOut positive integer 60 Time in seconds to allow an incoming call to ring before dropping the call O infinite Note The call diversion no answer feature will take precedence over this feature if enabled For more information refer to No Answer lt noanswer gt on page A 97 call ringBackTimeOut positive integer 60 Time in seconds to allow an outgoing call to remain in the ringback state before dropping the call O infinite Configuration Files Attribute Permitted Values Default Interpretation call dialtoneTimeOut Null positive integer 60 Time in seconds to allow the dialtone to be played before dropping the call If set to 0 the call is not dropped If set to Null call dropped after 60 seconds call lastCallReturnString string of maximum length 32 69 The string sent to the server when the user selects the last call return action call callsPerLineKey 1 to 24 OR 1to8 24 OR For the SoundPoint IP 600 601 and 650 the permitted ra
319. ty detection VLAN ID using DHCP C 16 voice activity detection 4 48 voice activity detection lt vad gt A 46 voice mail integration 4 36 voice quality monitoring 4 52 A 47 voice setting lt voice gt A 34 volume persistence lt volume gt A 37 W web server lt httpd gt A 57 Index 6 Addendum to SIP 3 0 Administrator s Guide This addendum addresses changes to the SoundPoint IP SoundStation IP SIP 3 0 Administrator s Guide specific to the release of the SoundStation IP 7000 conference phone The SoundStation IP 7000 conference phone is a next generation phone with a modern industrial design and the latest advancements in Polycom voice technology Some of the features include Extraordinary audio performance Strong IP telephony feature set Built in voice bridge for multi party conferences Ability to connect multiple consoles and microphones together for convenience performance and flexibility A port built into the console to allow calls from a mobile phone Daisy chaining Provisioning over CLink The new or changed features include Distribution Zip File Managing Conferences Configurable Feature Keys Multilingual User Interface Single Registration Audio Codecs Treble Bass Controls Voice Quality Monitoring Administrator s Guide Addendum for the SoundStation IP 7000 Note e Daisy Chaining Phones e Provisioning Phones Over CLink Configuration file changes are described in Configuration File
320. ublic Primary Certification Authority e Verisign Class 1 Public Primary Certification Authority G2 e Verisign Class 1 Public Primary Certification Authority G3 e Verisign Class 2 Public Primary Certification Authority e Verisign Class 2 Public Primary Certification Authority G2 e Verisign Class 2 Public Primary Certification Authority G3 e Verisign Class 3 Public Primary Certification Authority e Verisign Class 3 Public Primary Certification Authority G2 e Verisign Class 3 Public Primary Certification Authority G3 e Verisign Class 4 Public Primary Certification Authority G2 e Verisign Class 4 Public Primary Certification Authority G3 e Verisign RSA Commercial CA e Verisign RSA Secure Server CA Encrypting Configuration Files The phone can recognize encrypted files which it downloads from the boot server and it can encrypt files before uploading them to the boot server There must be an encryption key on the phone to perform these operations Configuration files excluding the master configuration file contact directories and configuration override files can be encrypted A separate SDK with a readme file is provided to facilitate key generation and configuration file encryption and decrypt on a UNIX or Linux server The utility is distributed as source code that runs under the UNIX operating system For more information contact Polycom Technical Support Administrator s Guide SoundPoint IP SoundStation IP Note
321. ules to displayed time tcplpApp sntp daylightSavings fixedDayEnable 0 1 If set to 0 month date and dayOfWeek are used in DST date calculation If set to 1 then only month and date are used tcplpApp sntp daylightSavings start month 3 March Month to start DST Mapping 1 Jan 2 Feb 12 Dec A 54 Configuration Files Attribute Permitted Values Default Interpretation tcplpApp sntp daylightSavings start date 1 31 8 If fixedDayEnable is set to 1 use as day of the month to start DST If fixedDayEnable is set to 0 us the mapping 1 the first occurrence of a given day of the week in a month 8 the second occurrence of a given day of the week in a month 15 the third occurrence of a given day of the week in a month 22 the fourth occurrence of a given day of the week in a month tcplpApp sntp daylightSavings start time 0 23 Time of day to start DST in 24 hour clock Mapping 2 2 am 14 2 pm tcplpApp sntp daylightSavings start dayOfWeek Day of week to apply DST Mapping 1 Sun 2 Mon Sat tcplpApp sntp daylightSavings start dayOfWeek la stInMonth 0 1 If set to 1 and fixedDayEnable is set to 0 DST starts on the last day specified by start dayOfWeek of the week in the month The start date is ignored tcplpApp sntp daylightSavings stop month 1 12 11 Month to stop DST tcplpApp sntp da
322. ult voice ns hs enable 0 voice ns hs signalAttn 6 voice ns hs silenceAttn 9 voice ns hd enable 0 voice ns hd signalAttn 0 voice ns hd silenceAttn 0 voice ns hf enable 1 voice ns hf signalAttn 6 voice ns hf silenceAttn 9 voice ns hf IP_4000 enable 1 voice ns hf IP_4000 signalAttn 6 voice ns hf IP_4000 silenceAttn 9 Configuration Files Automatic Gain Control lt agc gt These settings control the performance of the transmit automatic gain control feature Note Automatic Gain Control will be implemented in a future release Polycom recommends that you do not change these values POLYCOM Attribute Default voice agc hs enable 0 voice agc hd enable 0 voice agc hf enable 0 Receive Equalization lt rxEq gt These settings control the performance of the receive equalization feature Polycom recommends that you do not change these values POLYCOM Attribute Default voice rxEq hs IP_330 preFilter enable 1 voice rxEq hs IP_430 preFilter enable 1 voice rxEq hs IP_500 preFilter enable 1 voice rxEq hs IP_600 preFilter enable 1 voice rxEq hs IP_601 preFilter enable 1 voice rxEq hs IP_650 preFilter enable 1 voice rxEq hs IP_330 postFilter enable voice rxEq hs IP_430 postFilter enable voice rxEq hs IP_500 postFilter enable voice rxEq hs IP_600 postFilter enable E C S voice rxEq hs IP_601 postFilter enable Administrator s Guide SoundPoint
323. ume 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 Adjust receive and handset headset volume sip cfg For more information refer to Volume Persistence lt volume gt on page A 37 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 e mono G 711 13 bit dynamic range 8 khz sample rate e mono L16 16000 16 bit dynamic range 16 kHz sample rate Administrator s Guide SoundPoint IP SoundStation IP Note L16 16000 is not supported on SoundPoint IP 301 and SoundStation IP 4000 phones Note The alternate sampled audio sound effect files must be present on the boot server or the Internet for downloading at boot time Co
324. uration password required SIP Configuration SSL Security settings Reset to Default local configuration device settings and file system format User password Restart Phone required Warning Changes made through the web server or local user interface are stored internally as overrides These overrides take precedence over settings contained in the configuration obtained from the boot server If the boot server permits uploads these override setting will be saved in a file called lt Ethernet address gt phone cfg on the boot server as well in flash memory Local configuration changes will continue to override the boot server derived configuration until deleted through the Reset Local Config menu selection Troubleshooting Your SoundPoint IP SoundStation IP Phones This chapter provides you with some tools and techniques for troubleshooting SoundPoint IP SoundStation IP phones and installations The phone can provide feedback in the form of on screen error messages status indicators and log files for troubleshooting issues This chapter includes information on BootROM Error Messages e Application Error Messages e Status Menu e Log Files This chapter also presents phone issues likely causes and corrective actions Issues are grouped as follows e Power and Startup e Controls e Access to Screens and Systems e Calling e Displays e Audio e Upgrading Review the latest Release Notes
325. urn all logging levels to the default value of 4 There are other logging parameters that you may wish to modify Changing these parameters does not have the same impact as changing the logging levels but you should still understand how your changes will affect the phone and the network e log render level Sets the lowest level that can be logged default 1 e log render file size Maximum size before log file is uploaded default 16 kb e log render file upload period Frequency of log uploads default is 172800 seconds 48 hours e log render file upload append Controls if log files on the boot server are overwritten or appended not supported by all servers Administrator s Guide SoundPoint IP SoundStation IP e log render file upload append sizeLimit Controls the maximum size of log files on the boot server default 512 kb e log render file upload append 1limitMode Controls action to take when server log reaches max size actions are stop and delete Scheduled Logging Scheduled logging is a powerful tool for anyone who is trying to troubleshoot an issue with the phone that only occurs after some time in operation The output of these instructions is written to the application log and can be examined later for trend data The parameters for scheduled logging are found in the sip cfg configuration file They are log sched module_name The following figure shows an example of a configuration fil
326. us Configuration changes can performed centrally at the boot server Central boot server sip cfg Configuration file Specify the location of the corporate directory s LDAP server the LDAP attributes how often to refresh the local cache from the LDAP server and other miscellaneous parameters For more information refer to Corporate Directory lt corp gt on page A 63 Local Local Phone User Enable or disable persistent viewing through the Settings menu Interface Settings gt Basic gt Preferences gt Corporate Directory gt View Persistency Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the boot server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the boot server This section contains the following information e Corporate Directory LDAP Attributes e Browsing the Corporate Directory e Configuration File Example Administrator s Guide SoundPoint IP SoundStation IP Corporate Directory LDAP Attributes The entry attributes in the corporate directory are mapped through sip cfg configuration file attributes to the LDAP attributes first_name last_name phone_number and others so the SIP application knows how to use them for searching dialing or saving to the local contact directory Multiple attributes of the sam
327. used to cache downloaded wave files and other resources for the user interface ramdisk bytesPerBlock 0 32 33 0 These three parameters use internal defaults 1024 when value is set to 0 ramdisk blocksPerTrack 0 1 2 0 Note For the SoundPoint IP 650 platform 65536 ramdisk bytesPerBlock is internally replaced by 2X the value ramdisk nBlocks 0 1 2 4096 65536 ramdisk minsize 50 to 16384 50 Smallest size in Kbytes of RAM disk to create before returning an error RAM disk size is variable depending on the amount of device memory ramdisk minfree 512 to 16384 3072 Minimum amount of free space that must be left after the RAM disk has been created The RAM disk s size will be reduced as necessary in order to leave this amount of free RAM Request lt request gt This attribute includes e Delay lt delay gt Delay lt delay gt These settings control the phone s behavior when a request for restart or reconfiguration is received Permitted Attribute Values Default Interpretation request delay type Null audio or call Defines the strategy to adopt before a request gets call executed If set to audio a request can be executed as soon as there is no active audio on the phone independently of any call state If set to call a request can be executed as soon as there are no calls in any state on the phone Feature lt feature gt Configu
328. ustom animation if configured in the animation section of sip cfg Animations lt anim gt lt IP_300 gt lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt This section defines bitmap animations composed of bitmap duration couples In the following table x IP_300 IP_330 IP_400 IP_500 IP_600 IP_4000 y is the animation number z is the step in the animation Note that IP_300 parameters affect SoundPoint IP 301 phones IP_330 parameters affect SoundPoint IP 320 and 330 phones IP_400 parameters affect SoundPoint IP 430 phones IP_500 parameters affect SoundPoint IP 501 phones and IP_600 parameters affect SoundPoint IP 550 560 600 601 and 650 phones IP_4000 parameters affect SoundStation IP 4000 phones Note As of SIP 2 2 0 a maximum of 24 frames per animation is supported Attribute Permitted Values Interpretation ind anim x y frame z bitmap A bitmap name defined Bitmap to use previously Note that it must be defined already refer to Platform lt IP_300 gt lt IP 330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt on page A 71 ind anim x y frame z duration positive integer Duration in milliseconds for this step O infinite A 72 Patterns lt pattern gt Configuration Files This section defines patterns for the LED indicators In the following table x is the pattern number y is the step in the pattern
329. w events have been logged since the last upload log render file upload append 0 1 1 If set to 1 use append mode when uploading log files to server Note HTTP and TFTP don t support append mode unless the server is set up for this log render file upload append sizeLimit positive 512 Maximum log file size on boot integer server in Kbytes log render file upload append limitMode delete stop delete Behavior when server log file has reached its limit delete delete file and start over stop stop appending to file Scheduled Logging Parameters lt sched gt The phone can be configured to schedule certain advanced logging tasks ona periodic basis These attributes should be set in consultation with Polycom Technical Support Each scheduled log task is controlled by a unique attribute set starting with log sched x where x identifies the task Permitted Attribute Values Interpretation log sched x name alphanumeric Name of an internal system command to be periodically executed string To be supplied by Polycom log sched x level 0 5 Event class to assign to the log events generated by this command This needs to be the same or higher than log level change slog for these events to appear in the log log sched x period positive Seconds between each command execution 0 run once integer Configuration Files Permitted Attribute Values Interpretation log sched x startMode abs rel Start
330. warranty gives you specific legal rights which may vary depending on local law Copyright Notice Portions of the software contained in this product are Copyright 1998 1999 2000 Thai Open Source Software Center Ltd and Clark Cooper Copyright 1998 by the Massachusetts Institute of Technology Copyright 1998 2003 The OpenSSL Project Copyright 1995 1998 Eric Young eay cryptsoft com All rights reserved Copyright 1995 2002 Jean Loup Gailly and Mark Adler Copyright 1996 2004 Daniel Stenberg lt daniel haxx se gt Permission is hereby granted free of charge to any person obtaining a copy of this software and associated documentation files the Software to deal in the Software without restriction including without limitation the rights to use copy modify merge publish distribute sublicense and or sell copies of the Software and to permit persons to whom the Software is furnished to do so subject to the following conditions The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software THE SOFTWARE IS PROVIDED AS IS WITHOUT WARRANTY OF ANY KIND EXPRESS OR IMPLIED INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM DAMAGES OR OTHER LIABILITY WHETHER IN AN ACTION OF CONTRACT TORT OR OTHERWISE
331. x Responses Provisional Response Supported Notes 100 Trying Yes 180 Ringing Yes 181 Call Is Being Forwarded No 182 Queued No 183 Session Progress Yes 2xx Responses Success Response Supported Notes 200 OK Yes 202 Accepted Yes In REFER transfer Note 3xx Responses Redirection Session Initiation Protocol SIP Response Supported Notes 300 Multiple Choices Yes 301 Moved Permanently Yes 302 Moved Temporarily Yes 305 Use Proxy No 380 Alternative Service No Axx Responses Request Failure All 4xx responses for which the phone does not provide specific support will be treated the same as 400 Bad Request Response Supported Notes 400 Bad Request Yes 401 Unauthorized Yes 402 Payment Required No 403 Forbidden No 404 Not Found Yes 405 Method Not Allowed Yes 406 Not Acceptable No 407 Proxy Authentication Required Yes 408 Request Timeout No 410 Gone No 413 Request Entity Too Large No 414 Request URI Too Long No 415 Unsupported Media Type Yes 416 Unsupported URI Scheme No 420 Bad Extension No 421 Extension Required No 423 Interval Too Brief No 480 Temporarily Unavailable Yes Administrator s Guide SoundPoint IP SoundStation IP Response Supported Notes 481 Call Transaction Does Not Exist Yes 482 Loop Detected
332. ylightSavings stop date 1 31 Day of the month to stop DST tcplpApp sntp daylightSavings stop time 0 23 Time of day to stop DST in 24 hour clock tcplpApp sntp daylightSavings stop dayOfWeek 1 7 Day of week to stop DST tcplpApp sntp daylightSavings stop dayOfWeek la stInMonth 0 1 If set to 1 and fixedDayEnable set to 0 DST stops on the last day specified by stop dayOfWeek of the week in the month The stop date is ignored Administrator s Guide SoundPoint IP SoundStation IP Port lt port gt This attribute includes e RTP lt rtp gt RTP lt rtp gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation tcplpApp port rtp filterBylp 0 1 1 If set to 1 reject RTP packets arriving from sent from a non negotiated through SDP IP address tcplpApp port rtp filterByPort 0 1 0 If set to 1 reject RTP packets arriving from sent from a non negotiated through SDP port tcplpApp port rtp forceSend Null Null When non Null send all RTP 1024 65534 packets to and expect all RTP packets to arrive on the specified port Note both tcplpApp port rtp filterBylp and tcplpApp port rtp filterByPort must be enabled for this to work tcplpApp port rtp mediaPortRangeStart Null even Null If set to Null the value 2222 will integer from be used for the first allocated 1024 65534 RTP port otherwise the
333. ze for G7221 G7221C Siren14 and Siren22 are further subdivided voice audioProfile x jitterBufferMin 20 40 50 60 The smallest jitter buffer depth in milliseconds multiple of that must be achieved before play out begins 10 for the first time Once this depth has been achieved initially the depth may fall below this point and play out will still continue This parameter should be set to the smallest possible value which is at least two packet payloads and larger than the expected short term average jitter The IP4000 values are the same as the IP30x values voice audioProfile x jitterBufferShrink 10 20 30 The absolute minimum duration time in multiple of 10 milliseconds of RTP packet Rx with no packet loss between jitter buffer size shrinks Use smaller values 1000 ms to minimize the delay on known good networks Use larger values to minimize packet loss on networks with large jitter 3000 ms voice audioProfile x jitterBufferMax gt The largest jitter buffer depth to be supported jitterBufferMin in milliseconds Jitter above this size will multiple of 10 always cause lost packets This parameter lt 300 for IP should be set to the smallest possible value 320 330 430 that will support the expected network jitter 501 550 600 601 and 650 lt 200 for IP 301 Administrator s Guide Addendum for the SoundStation IP 7000 Gains lt gain gt The default gain settings have been carefully adjus

Polycom SIP 3.0.2 User's Manual

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