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Polycom SIP 3.1 Telephone User Manual

1. 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 550 560 600 601 650 670 A vV Le LDE gt E 2 H 5 g o o N gv 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 Dialpad2 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 Dialpad9 27 SoftKey2 38 Headset 6 Delete 17 Dialpad8 28 SoftKey1 39 Handsfree 7 Menu 18 Dialpad7 29 Applications 40 Hold 8 Messages 19 Dialpad4 30 Directories 41 Line4 9 DoNotDisturb 20 Dial
2. Permitted Attribute Values Default Interpretation 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 dir corp password UTF 8 encoded Null The password used to authenticate string to the LDAP server dir corp filterPrefix UTF 8 encoded objectclas Predefined filter string string S person If set to Null or invalid objectclass person is used dir corp scope one sub sub Type of search base If set to one a search of the level one below the baseDN is performed If set to sub or Null a recursive search of all levels below the baseDN is performed If set to base a search at the baseDN level is performed 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 last_name This parameter defines how the last_name attribute is interpreted by the phone phone_number Entries can have multiple attributes SIP_address of the same
3. ls 00 Initial log entry Current logging level 4 0224000058 log O0 Initial log entry Current logging level 4 O0 Initial log entry Current logging level 4 00 Resolver initialized added 2 nameservers and set vancouver polycom c O0 Initial log entry Current logging level 4 O0 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 0224000058 curl 3 3 00 UtilCopyC curl error Curl Error strings have been compiled out 3 3 3 3 0224000058 copy 0224000058 sec 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 copy 0224000058 so 0224000058 dns s 3 0223160058 log 00 Log render Ievel set to 1 0223160058 Initial log entry Current logging level 0223160058 rdisk 00 Initial log entry Current logging level 0223160058 res 00 Initial log entry Current logging level 0223160058 httpa 00 Initial log entry Current logging level 0223160058 cdp 00 Initial log entry Current logging level 0223160058 cdp 00 CDP is DISABLED 0223160058 sys 00 Initial log entry Current logging level 0223160058 wdog 00 Initial log entry Current logging level 0223160118 sec 4 00 utilCryptoConfigFileEncrypted Could not read
4. 0 0 4 78 IP Type of Service oi o55 ceied tant ne Seek kee be ane e Lae uke 4 78 TERE 802 1p7 0 scutes iad N a peek tae hand bailed 4 78 Voice Quality Monitoring 0 cee eee eee ee 4 79 Dynamic Noise Reduction 00 000 e eee eee 4 80 Treble Bass Controls 00 cee cee ce cece tenet eens 4 80 Setting Up Security Features 0 eeeeee es 4 80 Local User and Administrator Privilege Levels 4 81 Administrator s Guide SoundPoint IP SoundStation IP viii Custom Certificates caressant erone ena ELTE teed ees baaeee wees 4 81 Incoming Signaling Validation 0000000 4 82 Secure Real Time Transport Protocol 004 4 82 Configuration File Encryption 0 00 cece eee eee 4 82 Configuring SoundPoint IP SoundStation IP Phones Locally 4 83 5 Troubleshooting Your SoundPoint IP SoundStation IP Phones Error Messages iy aria ates ick iiaa aea ivack dw E AAEE E REE 5 2 BootROM Error Messages 0 0 cee eee eee 5 2 Application Error Messages 00 000 eee eee eee 5 3 Status Menu is ociiv edad atetan iran EEIT E EAN e nee ade 5 4 Eog Files serar tA Boas Serene a a A ods ts eh acted anc teed 5 5 Reading a Boot Log 1 eect eee nes 5 8 Reading an Application Log 000 0 00008 5 9 Testing Phone Hardware cn s60s cosa eee ek bebe OEM ekis wea 5 9 P werand Startup s
5. 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 enabled is 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 contact 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 650 670 x 1 34 IP 4000 x 1 IP 600
6. volpProt SIP serverFeatureControl dnd 0 1 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 Administrator s Guide SoundPoint IP SoundStation IP Attribute volpProt SIP authOptimizedInFailover volpProt SIP csta Permitted Values 0 1 0 1 Default 0 Interpretation 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 volIpProt SIP authOptimizedInFailover is Null then this feature is disabled If both attributes are set the value of reg x auth optimizedInFailover takes precedence If set to 1 uaCSTA is enabled volpProt SIP strictLineSeize 0 1 Null If set to 1 forces the phone to wait for 200 OK response when receiving a TRYING notify If set to 0 or Null this is old behavior volpProt SIP strictUserValidation 0 1 Null If set to 1 forces the phone to match user portion of signaling exactly If set to 0 or Null phone will use first registration if the user part does not match any registration volpProt SIP lineSeize retries 3 to 10 10 Controls the number
7. There are audio and echo issues on the headset Possible issues include Echo on external calls through a gateway Internal calls no gateway handsfree echo e Internal calls no gateway handset to handset echo Refer to Technical Bulletin 16249 Troubleshooting Audio and Echo Issues on SoundPoint IP Phones at http www polycom com usa en support v oice soundpoint_ip VoIP_Technical_Bullet ins_pub html 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 Id 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 20 Administrator s Guide SoundPoint IP SoundStation IP 5 16 Configuration Files Caution Note Note This appendix provides detailed descriptions of certain configuration files used by the Session Initiation Protocol SIP app
8. 606 Not Acceptable No Session Initiation Protocol SIP Hold Implementation 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 Note Even if the phone is set to use c 0 0 0 0 it will not do so if it gets any sendrecv sendonlly or inactive from the server These flags will cause it to revert to the other hold method Reliability of Provisional Responses The phone fully supports RFC 3262 Reliability of Provisional Responses Transfer 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 fro
9. For related configuration changes refer to Local Contact Directory on page 4 9 Configuring Your System Distinctive Ringing There are three options for distinctive ringing 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 Specify the mapping of Alert Info strings to ring types sip cfg For more information refer to Alert Information lt alertInfo gt on page A 15 Configuration file Specify the ring type to be used for each line phonet cfg e For more information refer to Registration lt reg gt on page A 107 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
10. high Note If idlelntensity is set higher than onintensity it will be replaced with the onintensity value 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 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 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 Configuration Files Permitted Attribute Values Default Interpretation 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 speakerp
11. Download the file to the boot server 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 background will be displayed e For more information refer to Backgrounds lt bg gt on page A 77 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 112 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 call 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 Configurati
12. 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 addGlobalPrefix 0 1 If set to 1 global prefix is added to E 164 user parts in sip URIs volpProt SIP allowTransferOnProceeding 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 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
13. 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 For more information refer to Product Model and Part Number Mapping on page C 26 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 yes gt lt Default Master SI lt edit and rename t each phone gt lt SRCSfile 000000000000 cfg v SRevision gt lt APPLICATION APP_FILE_ sip cfg MISC_F LES OVERRIDES_DIRECTORY CONTACTS_ DIRECTO APP FILE PATH SP CONFIG_FILES_SP APP FILE PATH SP CONFIG_FILES_SP Ry IP300 IP500 P Configuration File gt his file to lt Ethernet address gt cfg for PATH sip ld CONFIG_FILES phonel cfg LOG_FILE_DIRECTORY LICENSE_DIRECTORY SPIP300 sip 1d P300 phonel_SPIP300 cfg sip_SPIP300 cfg SPIP500 sip 1d P500 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 usa en support voice soundpoint_ip VoIP_T ech
14. The syslog server IP address or host name The default value is NULL device syslog transport None 0 UDP 1 TCP 2 TLS 3 The protocol that the phone will use to write to the syslog server If set to None transmission is turned off but the server address is preserved 126 Configuration Files Name Possible Values Description 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 0to6 Specifies the lowest class of event that will be rendered to syslog It is based on log render level and can be a lower value Refer to Basic Logging lt level gt lt change gt and lt render gt on page A 86 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 etherVlanFilter 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 serial enable 0 1 Enables the debug serial port The default value is 1 device sec SSL certList all custom
15. Administrator s Guide for the Polycom SoundPoint IP SoundStation IP Family SIP 3 1 August 2008 Edition 1725 11530 31 pra A ha PO LYCO M 3 1 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 w
16. Configuration file sip cfg 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 e For more information refer to Corporate Directory lt corp gt on page A 69 Local Local Phone User Interface Enable or disable persistent viewing through the Settings menu 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 Administrator s Guide SoundPoint IP SoundStation IP Note This section contains the following information e Corporate Directory LDAP Attributes e Browsing the Corporate Directory e Configuration File Example 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 same type are allowed The maximum of eight attributes can be configured in sip cfg The con
17. Customers replacing legacy telephony PBX or key system would like to get equivalent functionality from their new VoIP telephony system With SIP 3 0 this feature allowed system administrators to program the speed dials on their phones to interact with the phone user to implement commonly used functions such as Call Park in an intuitive fashion This capability applies to the SoundPoint IP 301 320 330 430 501 550 560 601 650 and 670phones The enhanced feature key functionality is implemented using Star Code sequences and SIP messaging The enhanced feature key definition language was defined to follow current configuration file standards and to be extensible The particular Star Code sequence and the associated prompts displayed on the SoundPoint IP phone for the enhanced feature are defined by macros These macros are case sensitive An enhanced feature key can be accessed from all instances where the speed dial is accessible for example unused line keys speed dial lists or programmed to hard function keys This section provides detailed information on e Enhanced Feature Key Definition Language e Macro Definition e Configuration File Changes e Useful Tips e Examples Enhanced Feature Key Definition Language This section defines the additional fields to be entered into a configuration file for controlling the enhanced feature key behavior The definition language follows the XML style notation The follow
18. Null If set to 0 the New Call soft key is not displayed when there is another way to place a call If set to 1 or Null the New Call soft key is displayed softkey feature endcall 0 1 default Null If set to 0 the End Call soft key is not displayed If set to 1 or Null the EndCall soft key is displayed softkey feature split 0 1 default Null If set to 0 the Split soft key is not displayed If set to 1 or Null the Split soft key is displayed softkey feature join 0 1 default Null If set to 0 the Join soft key is not displayed If set to 1 or Null the Join soft key is displayed softkey feature forward 0 1 default Null If set to 0 the Forward soft key is not displayed If set to 1 or Null the Forward soft key is displayed A 104 Configuration Files Attribute softkey feature directories Permitted Values 0 1 Null default Default Null Interpretation If set to Null the Dir soft key is displayed on the SoundPoint IP 320 330 phone but not on any other phone If set to 0 the Dir soft key is not displayed on any phone If set to 1 the Dir soft key is displayed on all phones as follows e Inthe idle state it is displayed after the New Call and Callers soft keys e Inthe dialtone state itis displayed after the End Call and Callers soft keys During a conference or transfer it is displayed
19. 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 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 Log Files Troubleshooting Your SoundPoint IP SoundStation IP Phones SoundPoint IP and SoundStation IP phones will log various events to files stored in the fla
20. dir local readonly 0 1 Specifies whether or not local contact directory is read only If set to 0 or Null the local contact directory is editable If set to 1 the local contact directory is read only Note If the local contact directory is read only speed dial entry on the SoundPoint IP 320 330 is disabled enter the speed dial index followed by dir search field 0 1 Null 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 Administrator s Guide SoundPoint IP SoundStation IP
21. 6000 or 7000 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 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 A speed dial Null 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 not necessary For more information refer to Keys lt key gt on page A 75 For more information on the default feature key layouts refer to Default Feature Key Layouts on page C 12 Configuring Your System 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 assig
22. 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 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
23. 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 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 SoundPoint IP 330 320 430 501 550 560 600 601 650 and 670 phones and the SoundStation IP 4000 phone Voice Quality Monitoring is not supported on the SoundStation IP 6000 and 7000 conference phones at this time 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 4 79 Administrator s Guide SoundPoint IP SoundStation IP
24. 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 Administrator s Guide SoundPoint IP SoundStation IP 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 ring 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
25. add attribute line a fmtp 18 annexb no below a rtpmap attribute line where 18 could be replaced by another payload If set to 0 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 A 52 e Server lt server gt e RTCP XR lt rtepxr gt Central Report Collector lt collector gt Configuration Files This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation voice qualityMonitoring collector enable periodic 0 1 0 Enables generation of periodic quality reports throughout a call voice qualityMonitoring collector enable session 0 1 Enables generation of a quality report at the end of each call voice qualityMonitoring collector enable triggeredPeriodic 0 1 2 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 qualityMon
26. default The type of certificate list device sec SSL customCert X 509 certificate The certificate value A 127 Administrator s Guide SoundPoint IP SoundStation IP A 128 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 timers 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 dr
27. 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 A 20 Routing lt routing gt 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 Configuration Files Server lt server gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan routing server x dotted decimal IP address Null IP address or host name and port of address or host name a SIP server that will be used for routing calls Multiple servers can be listed starting with x 1 2 for fault tolerance dialplan routing server x port 1 to 65535 5060 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
28. e 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 three 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 and SoundStation IP 7000 Configuration changes can performed centrally at the boot server Central boot server sip cfg Configuration File
29. or proxy for the secondary is determined by the following criteria e If the 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 Setting up Your System 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 configuration 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 20 Warnin
30. 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 ora comma separated list A list of the languages supported on hardware platform y where y can be IP_500 IP_600 or IP4000 The IP_500 platform does not support any Asian languages The IP_4000 platform does not support Slovenian Icl ml lang tags x string in the format language_region language preference level The format is e The first two letters are the ISO 639 language abbreviation e The next two letters are the ISO 3166 country code e The next two letters are the ISO 639 language abbreviation e The remainder of the string is the preference level for the display of the language or English if the language is not available For example lcl ml lang tags 1 zh cn zh q 0 9 en q 0 8 For more information refer to the Accept Language header definition in the HTTP RFC 2616 at http www w3 org Protocols rfc261 6 rfc2616 sec14 html sec14 4 To add new languages to those included with the distribution 1 2 Create a new dictionary file based on an existing one Change the strings making sure to encode the XML fil
31. page A 16 Configuration file phonet cfg Specify per registration line type private or shared barge in 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 107 Specify per registration whether diversion should be disabled on shared lines e For more information refer to Diversion lt divert gt on page A 114 Local Web Server if enabled Specify line seize subscription period 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 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 using the SIP Configuration me
32. this will override the 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 Sou ndStation 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
33. 6 Aa E ah Meh Rane Sol NOR Sah a 4 3 Gall Waitin es e e ta ae Varded bond 4 tb eae eee echoes ema e 4 3 Called Party Identification 0 0 eee eee 4 4 Calling Party Identification 0 eee eee 4 4 Missed Call Notification 0 00 4 4 Connected Party Identification 0 eee eee eee 4 5 Context Sensitive Volume Control snssnsnnsann nrn 4 5 Customizable Audio Sound Effects 0 0 00 cece ee eee eee 4 5 Message Waiting Indication 000 e eee ee eee 4 6 Distinctive Incoming Call Treatment 06 0 060 e eee eee 4 6 Distinctive Ringing episte tersi eee ees 4 7 Distinctive Call Waiting 0 0 000s 4 7 Do NOE Disturb waco ek ees Sala in te inte oa ae 88 Meee nee 4 8 Handset Headset and Speakerphone 04 4 8 Local Contact Directory s 0 0 0 4 9 Local Digit Map o oscicsss ios a aa a eh we iad oh 4 12 Microphone Mute 0 00 cee eee eee ee 4 13 Soft Key Activated User Interface 0000000 4 13 Speed Dial ies sealers eich etna ah ake t avid sh sge eet Eat kG 4 13 Time and Date Display 00 e eee eee 4 14 Idle Display Animation 00 e eee eee eee eee 4 15 Ethernet Switeh sezere nu oea a e i e E E eats 4 15 Graphic Display Backgrounds 0000000 4 16 Automatic Off Hook Call Placement 0 0000004 4 17 Call Hold sheds Gs dik eG SG Ses ES SOS FSS GEN a
34. 97 The star code to initiate a directed call pickup Note The default value supports the BroadWorks calls server only You must change the value if your organization uses a different call server 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 configuration attribute is defined as follows Attribute Permitted Values Default Configuration Files 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 If set to 1 play re order tone locally on shared line seize failure call shared oneTouchResume 0 1 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 6000 and 7000 phones 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 ex
35. 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 10 Since the APPLICATION APP_FILE_PATH attribute of the lt Ethernet address gt cfg files references the individual sip ld 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 ld 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 16 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 90 Supporting SoundPoint IP 300 and 500 Phones 3 20 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 n
36. 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 and lt IP_7000 gt tag above A 82 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 next 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_4000 gt and lt IP_7000 gt on page A 83 LEDs lt led gt Configuration Files 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 82 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 and lt IP_7000 gt In the following table x IP_300 IP_330 IP_400 IP_500 IP_600 IP_4000 or IP_7000 y is the graphic icon number N
37. Guide SoundPoint IP SoundStation IP e Deutsche Telekom AG Root CA e Digital Signature Trust Co Global CA 1 e Digital Signature Trust Co Global CA 2 e Digital Signature Trust Co Global CA 3 e Digital Signature Trust Co Global CA 4 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 GeoTrust Primary Certification Authority e GeoTrust Global CA e GeoTrust Global CA 2 e GeoTrust Universal CA e GeoTrust Universal CA 2 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 Miscellaneous Administrative Tasks 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 Thawte Personal Freemail CA Thawte Personal Premium CA Thawte Premium Server CA Thawte Server CA Thawte Universal CA Root UPS Document Exchange by DST ValiCert Class 1 VA ValiCert Class 2 VA ValiCert Class 3 VA VeriSign Class 4 Primary CA Verisign Class
38. However the primary and secondary should be rebooted together for the primary secondary relationship to be recognized If you power up both SoundStation IP family phones the primary will power up first Currently provisioning over CLink is supported for the following configurations of SoundStation IP family conference phones e Two SoundStation IP family conference phone daisy chained together e Two SoundStation IP family conference phone daisy chained together with one external microphone specifically designed for the SoundStation IP family conference phone Refer to Daisy Chaining Phones on page 4 38 for an illustration of two SoundStation IP family conference phone daisy chained together The provisioning boot server or proxy for the secondary is determined by the following criteria e 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 Administrator s Guide SoundPoint IP SoundStation IP Enhanced Feature Keys 4 40 Note The Enhanced Feature Key feature from SIP 3 0 is compatible with Enhanced Feature Key feature from SIP 3 1 However improvements have been made and Polycom recommends that existing configuration files be reviewed and updated
39. IP phones features overview 2 8 introduction 1 1 network 2 2 new features overview 2 13 SoundPoint IP 330 switching text entry mode 3 7 SoundPoint IP 650 playback 4 37 A 92 recording 4 37 A 92 SoundPoint IP 670 playback 4 37 A 92 recording 4 37 A 92 SoundStation IP applications 4 33 configuring phones locally 4 83 features list of 1 6 supported languages 4 31 SoundStation IP 7000 daisy chaining 4 38 provisioning over CLink 4 39 treble bass controls 4 80 speed dial 4 13 SRTP See also secure real time transport protocol static DNS cache 4 68 status menu 5 4 T text entry mode switching 3 7 time and date display 4 14 time synchronization A 59 transmit equalization lt txEq gt A 50 Index 6 troubleshooting Application is not compatible 5 2 application error messages 5 3 application logging options 5 5 audio issues 5 15 blinking time 5 4 boot failure messages 5 8 bootROM error messages 5 2 calling issues 5 13 Config file error Error is 5 3 controls issues 5 11 Could not contact boot server 5 2 displays issues 5 14 Error loading 5 3 Error application is not present 5 3 Failed to get boot parameters via DHCP 5 2 log files 5 5 manual log upload 5 7 Network link is down 5 3 Not all configuration files were present 5 3 power and startup issues 5 10 reading a boot log 5 8 reading an application log 5 9 registration status 5 4 scheduled logging 5 6 screens and systems access issues 5 12 trusted certificate
40. IP_400 font lt IP_400 gt Null Defines the font file that will be loaded from boot server during boot up Note When several font P_330 x name are defined the index x must follow consecutive increasing order 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_400 x name are defined the index x must follow consecutive increasing order A 74 Attribute IP_500 font lt IP_500 gt Configuration Files This configuration attribute is defined as follows 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 P_500 x name are defined the index x must follow consecutive increasing order IP_600 font lt IP_600 gt This configuration attribute is defined as follows Attribute font IP_600 x name Permitted Values fontName_height_Uxx00 _UyyFF fon OR fontName_height_Uxx00 _UxxFF fnt Default Null Interpretation Defines the font file that will be loaded from boot serve
41. If set to 1 or Null all of these soft keys are displayed Per Phone Configuration wy POLYCOM A 106 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 14 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 Registration lt reg gt Configuration Files e User Preferences lt user_preferences gt Sound Point 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 600 601 650 and 670 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 number of butto
42. 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 b c 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 enabledto 1 Configuring Your System 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 voIpProt 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
43. L16 32000 51 2 Kops 32 Ksps 14 KHz L16 48000 76 8 Kops 48 Ksps 22 KHz Siren14 SIREN14 SIREN14 24 Kbps 32 Ksps 20ms 80ms 14 KHz 16000 32 Kbps 48 Kbps Siren22 SIREN22 SIREN22 32 Kbps 48 Ksps 20ms 80ms 22 KHz 48000 48 Kbps 64 Kbps 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 Configuration Files Permitted Attribute Values Default Interpretation voice codecPref IP_300 G711Mu Null 1 3 1 Specifies the codec preferences for 3 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 550 560 650 and 670 voice codecPref IP_650 G711A 3 platform Interpretation as above voice codecPref IP_650 G729AB 4 voice codecPref IP_650 G722 1 voice codecPref IP_4000 G711Mu Null 1 3 1 Specifies the codec preferences for the Sou
44. Note Note Setting up Your System 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 Each phone may open multip
45. Note The SoundPoint IP 430 501 550 560 600 601 650 and 670 phones and the SoundStation IP 4000 6000 and 7000 phones supports an XHTML Microbrowser This can be launched by pressing the Applications key or 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 Developer s Guide Configuring Your System Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Specify the Application browser home page a proxy to use and size limits e For more information refer to Microbrowser lt mb gt on page A 95 Specify the telephone notification and state polling events to be recorded and location of the push server For more information refer to Applications lt apps gt on page A 98 Local Web Server Specify the Applications browser home page and proxy to use if enabled 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
46. OR OTHERWISE 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 301 320 330 430 501 550 560 600 601 650 and 670 desktop phones and the SoundStation IP 4000 6000 and 7000 conference phones
47. 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 ie tod progress chord set 3 illi n miSecONgS 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 instruction 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 32 Note Configuration Files 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
48. VLAN Discovery to Fixed or Custom When set to Fixed the phone will examine DHCP options 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 V V V LAN A 10 LAN A 0x0a LAN A 012 Note If a VLAN tag is assigned by CDP DHCP VLAN tags will be ignored Administrator s Guide SoundPoint IP SoundStation IP 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 siring max 10 esponse to DHCP Discover on VLAN X received More VLANs in DVD string Yes No Boot process continues with VLAN tag assigned Phone Reboots Boot process continues without any VLAN assigne
49. after the Callers and Cancel soft keys softkey feature callers 0 1 Null default Null If set to Null the Callers soft key is displayed on the SoundPoint IP 320 330 phone but not on any other phone If set to 0 the Callers soft key is not displayed on any phone If set to 1 the Callers soft key is displayed on all phones as follows e Inthe idle state it is displayed after the New Call soft key and before the Dir soft key e Inthe dialtone state itis displayed after the End Call soft key and before the Dir soft key During a conference or transfer it is displayed before the Cancel soft key softkey feature mystatus Oor1 If set to 0 the MyStatus soft key is not displayed If set to 1 or Null the MyStatus soft key is displayed Note pres idleSoftKeys must be set to 1 for this soft key to be displayed A 105 Administrator s Guide SoundPoint IP SoundStation IP Attribute softkey feature buddies Permitted Values Default Interpretation Oor1 1 If set to 0 the Buddies soft key is not displayed If set to 1 or Null the Buddies soft key is displayed Note pres idleSoftKeys must be set to 1 for this soft key to be displayed softkey feature Oor1 1 If set to 0 and the phone has hard keys basicCallManagement redundant mapped for Hold Transfer and Conference functions all must be mapped all of these soft keys are not displayed
50. 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 15 Administrator s Guide SoundPoint IP SoundStation IP Do Not Disturb 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 DND 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 registration
51. authority list C 1 type of service bits 4 78 U uaCSTA A 12 A 107 B 9 upgrading SIP application 3 19 USB device 4 37 USB devices supported 4 38 user interface soft key activated 4 13 user preferences lt up gt A 25 V VAD See also voice activity detection VLAN ID using DHCP C 23 voice activity detection 4 74 voice activity detection lt vad gt A 52 voice mail integration 4 54 voice quality monitoring 4 79 A 52 voice setting lt voice gt A 37 volume persistence lt volume gt A 42 WwW web server lt httpd gt A 63 welcome sound reboot A 31 POLYCOM INC APPLICATION PROGRAMMING INTERFACE LICENSE API FOR SOUNDPOINT IP AND SOUNDSTATION IP PRODUCTS Product or Products 1 Agreement You understand and agree that by using the API you will be bound by the terms of the End User License and Warranty Terms included with the Product s and this document together the Agreement In the event of any conflicts between the End User License and Warranty Terms and this document this document shall govern with respect to the API 2 Parties For purposes of this Agreement you or your shall mean the individual or entity accepting this Agreement or using the API The relationship between you and Polycom is that of licensee licensor No legal partnership or agency relationship is created between you and Polycom Neither you nor Polycom is a partner an agent or has any authority to bind the oth
52. 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 digits the user part of a SIP URL or a string that constitutes a valid SIP URL 6416 or 6416 polycom com Null Contact to call when retrieving messages for this registration Network Address Translation lt nat gt A 120 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 Null If non Null this port will be used by the phone for SIP signali
53. 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 Downloadable Fonts New fonts can be loaded onto the phone For guidelines on downloading fonts refer to Fonts lt font gt on page A 72 Note Downloadable fonts are not supported on the SoundStation IP 6000 and 7000 Administrator s Guide SoundPoint IP SoundStation IP 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 Configuration 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 29 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 30 Specify patterns For more information refer to Patterns lt pat gt on page A 32 and Call Progress Patterns on page A 33 Microbrowser
54. 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 Ib UTF 8 encoded string label of up to 40 bytes Note In some cases this will be less than 40 characters due to UTF 8 s variable length encoding Note The label of a contact directory item is by default the label attribute of the item If the label attribute does not exist or is Null then the concatenation of first name and last name will be used as label A space is added between first and last names 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 Note In some cases this will be less than 40 characters due to UTF 8 s variable length encoding ct UTF 8 encoded string contact containing digits the user part of a SIP URL or a string that constitutes a valid SIP URL Used by the phone to address a remote party in the same way that a string of digits or a SIP URL are dialed manually by the user This element is also used to associate incoming callers with a particular directory entry Note This field cannot be null or duplicated Administrator s Guide SoundPoint IP SoundStation IP Element Permitted Values Interpretation sd Null 1 to 9999 speed dial index Asso
55. call is parked at that location by the call server The active call is put on hold during this operation 14941 I To 684000 i Call Park ie Hold Ross Dutkiewicz Call Pickup 1442 2 01 Mon Dec 17 9 52 AM Hold End Call Trnsfer More Configurable Soft Keys This feature enables phone system administrators to program certain frequently used functions onto the soft keys at the bottom of the phone display This programming can be controlled based on call state For example a Call Park function can be presented to the user when in an active call state If certain hard keys are missing you may want to create a soft key For example if there is no Do Not Disturb key on a phone you could create a Do Not Disturb soft key New soft keys can be mapped into e An Enhanced Feature Key sequence e A speed dial contact directory entry Note Configuring Your System e Directly into an Enhanced Feature Key macro e Directly into a URL e A chained list of actions It is possible to disable the display of specific standard keys the soft keys that are displayed on SoundStation IP and SoundStation IP phones to make room for other soft keys that your organization wants displayed To ensure that the usability of features is not compromised the disabling of certain soft keys in certain circumstances may be restricted When a standard soft key is disabled the space where it was remains empty The standard keys that can be
56. 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 Attribute Permitted Values Interpretation log sched x name alphanumeric string Name of an internal system command to be periodically executed 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 Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Interpretation log sched x startMode abs rel Start 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
57. can be modified 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 Note Only implicit FTPS is supported Administrator s Guide SoundPoint IP SoundStation IP Name Possible Values Description Server Address Server User dotted decimal IP address OR domain name string OR URL All addresses can be followed by an optional directory and optional file name any string 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 The user name used when the phone logs into the server if required for the
58. chain two SoundStation IP 7000 conference phones together read the information in Provisioning SoundStation IP 7000 Phones Using CLink on page 3 18 to understand the different provisioning options available Provisioning Phones To deploy phones from the boot server For more information on encrypting configuration files refer to Encrypting Configuration Files on page C 4 1 Optional Create per phone configuration files by performing the following steps 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 106 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 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 S
59. 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 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 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 NOTE This feature is supported with the Sylantro call server only A 112 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 550 560 x 1 4 IP 600 x 1 6 IP 601 x 1 12 IP 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 x 1 Permitted Attribute Values Default Interpretation call missedCallTracking x enabled 0 1 1 If set to 1 or Null missed call tracking is
60. 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 attributes must be sequential lcl 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 SoundPointIPLocalization folder on the boot server for the phone to be able to locate the dictionary file For example lcl ml lang menu 1 Chinese_China 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
61. dialplan routing emergency x Single entry representing for x 1 This determines the URLs value a SIP URL value 911 Null that should be watched for for all others When one of these defined URLs is detected as having been dialed by the user the call will automatically be directed to the defined emergency server dialplan routing emergency x positive integer for x 1 y 1 Null Index representing the server y for all others server defined in Server lt server gt on page A 21 that will be used for emergency routing Localization lt Icl 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 29 and Call Progress Patterns on page A 33 This attribute includes e Multilingual lt ml gt e Date and Time lt datetime gt 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
62. directed pickup and group pickup Some functions depend on the feature being enabled For example BuddyStatus and MyStatus require the presence feature to be enabled Hard key remappings do not require the Enhanced Feature key feature to be enabled This include the SpeedDial function The table below shows only Line1 to Line6 functions For the SoundPoint IP 602 650 and 670 phones with attached Expansion Modules Line7 to Line48 functions are also supported Function Label ACDAvailable ACDAvailableFromldle ACDLogin ACDLoginLogout ACDLogout ACDLoginLogout ACDUnavailable ACDAvailableFromldle Answer Answer Call screen only Applications Main Browser ArrowDown ArrowDown ArrowLeft ArrowLeft ArrowRight ArrowRight ArrowUp ArrowUp Bargeln BarglnShowAppearances Bargeln Call screen only BuddyStatus Buddy Status Callers Callers CallList Call Lists CallPark ParkEntry Call screen only CallPickup CallPickupEntry Call screen only Conference ConferenceCall Call screen only Delete Delete DialpadO DialpadO Dialpad1 Dialpad1 Dialpad2 Dialpad2 C 20 Miscellaneous Administrative Tasks Label Function Dialpad3 Dialpad3 Dialpad4 Dialpad4 Dialpad5 Dialpad5 Dialpad6 Dialpad6 Dialpad7 Dialpad7 Dialpad8 Dialpad8 Dialpadg Dialpadg DialpadPound Dial
63. disabled include e New Call e End Call e Split e Join e Forward e Directories or Dir as it is called on the SoundPoint IP 320 330 e Callers appears on the SoundPoint IP 320 330 e MyStatus and Buddies e Hold Transfer and Conference The Hold Transfer and Conference are grouped together to avoid usability issues Custom soft keys can be added in the following call states e Idle There are no active calls e Active This state starts when a call is connected It stops when the call stops or changes to another state like hold or dial tone e Alerting or ringing or incoming proceeding The phone is ringing e Dial tone You can hear the dial tone e Proceeding or outgoing proceeding This state starts when the phone sends a request to the network It stops when the call is connected e Setup This state starts when the user starts keying in a phone number This state ends when the Proceeding state starts e Hold The call is put on hold locally Administrator s Guide SoundPoint IP SoundStation IP Custom soft keys can be configured to precede the standard soft keys that are still displayed The order of the custom soft keys follows the configuration order The standard soft keys are shifted to the right and any empty spaces are removed If the custom soft keys are configured to not precede the standard soft keys then the standard soft keys do not move The order of the custom soft keys st
64. displayed web content with automatic phone behaviour which are valid actions from within the Microbrowser context Spontaneous web content will only be retrieved and displayed for a call if that call occupies or will occupy the UI focus at the time of the event e Passive Mode Web content can also be retrieved when the user chooses to do so The fact that web content is available for viewing is shown through the call appearance based web content icon descibed in Web Content Status Indication on page 4 66 The Select key can be used to fetch the associated web content for the call that is in focus If the web content has expired the icon will be removed and the Select key will perform no function Passive mode is recommended for applications where the Microbrowser is used for other applications In the SIP 2 2 feature interactive microbrowser sessions will be interrupted by the arrival of active mode web content URLs which may cause annoyance although the Back navigation function will work in this context Settings Menu If enabled a new SIP web content entry is added to the Setting gt Basic gt Preferences menu to allow the user to change the current content retrieval mode Two options are provided passive mode and active mode Signaling Changes A new SIP header must be used to report web content associated with SIP phone calls the SSAWC header follow the BNF for the standard SIP header Alert Info Alert Info Alert In
65. erussspenkepunisp unesene i sag Rlaren tube heap Reb 5 10 Controls rsa eck Ridge a haga ale Soe EE Rober oe oe ee we thee 5 11 Access to Screens and Systems 0000 e eee eee eee ee 5 12 Gallino seag ed nn eae Cees BEA ae ated Oe ee aes Baie 5 13 Displays yea ii n KE E E E E a T A 5 14 Audi 5p chia toate Sante eee p EERE ae y Whee EDE 5 15 Uperading iui sacrar i an ENE ETA el mab ENET 5 15 A Configuration Files ccc cece eee eee es A Master Configuration Files 0 0 e eee eee eee eee A 2 Application Configuration 0000 e eee eee A 4 Protocol lt VolpProt gt wie ics asics cade se aed baer se alee sae ese A 6 Dial Plan lt dialplan gt 0 0000 A 17 Localization lt lel gt 0 5d00 sais ae ead eee s Ode oedema see A 21 User Preferences lt up gt 0 0 0 0 0c cece eee eee eee A 25 Ton s lt Stones gt eredetien ete oes MRE Raita Bd gett RA aaa td A 27 Sampled Audio for Sound Effects lt saf gt 06 A 30 Sound Effects lt se gt 0 6 ccc cece eens A 31 Voice Settings lt voice gt 6 eee A 37 Quality of Service lt QOS gt 0 A 55 Basic TCP IP TCR IPA gt iare AE ERRER A 58 Web Server lt htipd gt cccc cts4scuaterens eieae a ESE A 63 Call Handling Configuration lt call gt 00 e eee A 64 Contents Directory lt dit Purere orian esar ea TETE TENNET VEENA A 68 Presence lt pr
66. family 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 family phone that has the Ethernet connection is referred to as the primary The SoundStation IP family 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 25 foot lt Network Cable O i 4 38 Configuring Your System Instructions for daisy chaining SoundStation IP family conference phones are available in the SoundStation IP 7000 User Guide Provisioning Phones Over CLink Normally the SoundStation IP family conference phone is provisioned over the Ethernet by the boot server However when two SoundStation IP family 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 family phone is not connected to the Ethernet Both SoundStation IP family phones must be running the same version of the SIP application The steps for provisioning the secondary SoundStation IP family phone are the same as for the primary SoundStation IP family phone You can reboot the primary without rebooting the secondary
67. flash memory Warning 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 e BootROM Error Messages e Application Error Messages e Status Menu Log Files e Testing Phone Hardware 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 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 Administrator s Guide SoundPoint IP SoundStation IP Error Messages If your problems is not listed in this chapter nor described in the latest Release Notes contact your Certified Polycom Reseller for support There are several different error messages that can be displayed on the phone when it is bo
68. 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 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 Administrator s Guide SoundPoint IP SoundStation IP Displays 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 The contrast needs adjustment Do one of the following e Check that the display is illuminated
69. 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 If set to 0 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 during normal 2 medium 3 use of the phone high The default value is medium up backlight idleIntensity 0 off Null This parameter controls the intensity of the 1 low LCD backlight when the phone is idle a 3 The default value is low
70. 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 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 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 SRCSfile 000000000000 cfg v S Revision gt lt APPLICATION APP_FILE_PATH PHONE_PART_NUMBER sip 1d CONFIG_FILES phonel cfg sip cfg
71. interface is available in the following languages by default Chinese if displayable Danish Dutch English French German Italian Japanese if displayable Korean if displayable Norwegian Polish Portuguese Russian Slovenian Spanish and Swedish Note Slovenian is not supported on the SoundStation IP 4000 Note 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 Note 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 the boot server or locally Central Configuration file Specify the boot up language and the selection of language choices boot server sip cfg to be made available to the user For more information refer to Multilingual lt ml gt on page A 22 For instructions on adding new languages refer to To add new languages to those included with the distribution on page A 23 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
72. local contact directory that can be downloaded from the boot server and edited locally Local Digit Map The phone has a local digit map to automate the setup phase of number only calls 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 Administrator s Guide SoundPoint IP SoundStation IP 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 c
73. logo 600 ind anim IP_600 46 frame 1 duration 0 gt lt IP_600 gt lt IP_4000 gt lt IDLE_DISPLAY ind anim IP_4000 45 frame 1 bitmap logo 4000 ind anim IP_4000 45 frame 1 duration 0 gt lt IP_4000 gt lt IP_7000 gt lt IDLE_DISPLAY ind anim IP_7000 46 frame 1 bitmap logo 4000 ind anim IP_7000 46 frame 1 duration 0 gt lt IP_7000 gt Miscellaneous Administrative Tasks lt Animations gt lt indicators gt BootROM SIP Application Dependencies Not withstanding the hardware backward compatibility mandate there have been times throughout the life of the SoundPoint IP SoundStation IP phones where certain dependencies on specific bootROM and application versions have been necessitated 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 670 CEM 4 1 1 or later 3 0 3 or later IP 4000 3 1 2 or later 1 4 or later IP 6000 4 1 1 or later 3 0 2 or later IP 7000 4 1 1 or later 3 0 2 or later Migration Dependencies In addition
74. lt IP 330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt In the following table x IP_300 IP_330 IP_400 IP_500 IP_600 IP_4000 or IP_7000 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 parameters affect SoundPoint IP 501 phones IP_600 parameters affect SoundPoint IP 550 560 600 601 and 650 and 670 phones IP_4000 parameters affect SoundStation IP 4000 and 6000 phones and IP_7000 parameters affect SoundStation IP 7000 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 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 A 80 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 and lt IP_7000 gt e Patterns lt pattern gt e Classes lt class gt e Assignments Configuration Files This configuration attribute is defined as follows Attribute Permi
75. 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 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 58 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 e Do not use fixed day use first or last day of
76. more information refer to Registration lt reg gt on page A 107 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 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 92 Configuration file Enable this feature per registration phonet cfg For more information refer to Registration lt reg gt on page A 107 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 Note Warning Configuring Your System Two types of redundancy are possible e 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 e 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 cal
77. 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 4 77 Administrator s Guide SoundPoint IP SoundStation IP 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 effective 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 cfg e For more information refer to IP TOS lt I
78. not comply it is not included in the digit plan as a valid one That is no matching is done against it 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 Using T in the left part of RRR syntax is not recommended For example ROTR322R should be avoided Administrator s Guide SoundPoint IP SoundStation IP 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 0 2 9poorxxx during the setup phase of 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 commais 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
79. of a SIP server to which the phone shall send all requests A 109 Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation reg x outboundProxy transport DNSnaptr or DNSnap If set to Null or DNSnaptr TCPpreferred or tr If reg x outboundProxy address is a UDPOnly or hostname and reg x outboundProxy port is 0 TLS or or Null do NAPTR then SRV look ups to try TCPOnly 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 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 server
80. 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 RORRxXXxxxxx Remove 9 at the beginning of the dialed number For example if a customer dials 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 to 6044539400 is placed e R9R604Rxxxxxxx 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 Configuration Files For example if a customer dials 16092345678 a call is placed to 16002345678 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 You must use only or 0 9 between second and third R If a digit map does
81. 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 recommends 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 i
82. om x em name any string bg hiRes gray bm x adj integer Default Interpretation Graphic files for display on the phone and Expansion Module and also the brightness adjustment to the graphic For x 1 name is Leaf jpg name is LeafEM jpg adjustment is 0 For x 2 name is Sailboat jpg name is SailboatEM jpg adjustment is 3 For x 3 name is Beach jpg name is BeachEM jpg adjustment is 0 For x 4 name is Palm jpg name is PalmEM jpg adjustment is 3 For x 5 e name is Jellyfish jpg name is JellyfishEM jpg adjustment is 2 For x 6 name is Mountain jpg name is MountainEM jpg adjustment is 0 Note If the file is missing or unavailable the built in default solid pattern is displayed Note The adjustment value is changed on each individual phone when the user lightens or darkens the graphic during preview button gray selection x y modify any string The label color for soft keys and line key labels associated with the defined backgrounds These values can be modified locally by the user The format is rgbHILo lt parameter list gt By default all defaults are set to none 79 Administrator s Guide SoundPoint IP SoundStation IP Bitmaps lt bitmap gt The bitmaps used by each phone model are defined in this section Platform lt IP_300 gt
83. on a supported USB device Only the SoundPoint IP 650 and 670 have a functioning 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 Administrator s Guide SoundPoint IP SoundStation IP Note Notify your users that they may be required by federal state and or local laws to notify some or all called parties when they are recording 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 a list of supported USB devices refer to Technical Bulletin 38084 SoundPoint IP 650 and 670 Supported USB Devices for Recording at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuration changes can be performed centrally at the boot server Central Configuration file Turn this feature on or off boot server sip cig For more information refer to Feature lt feature gt on page A 92 Daisy Chaining Phones You can join two SoundStation IP
84. prefix You can also add labels in the first name fn fields For information on file format refer to Local Contact Directory File Format on page 4 10 Useful Tips The following information should be noted e Activation of the enhanced feature key will fail if configured values are invalid except where noted in previous sections e All failures are logged at level 4 minor Configuring Your System e Iftwo macros have the same name the first one will be used and the subsequent ones will be ignored V and macro prefixes cannot be mixed in the same macro line ayy e A sequence of characters prefixed with The exception is the speed dial reference which starts with contains digits only are parsed as a macro name lh 1 and e A sequence of characters prefixed with is the action string e The sequence of characters accessed from speed dial keys must be prefixed by either or so it will be processed as an enhanced feature key All macro references and action strings added to the local directory contact field must be prefixed by either or e Action strings used in soft key definitions do not need to be prefixed by However the prefix must be used if macros or speed dials are referenced For more information refer to Configurable Soft Keys on page 4 50 e Asequence of macro names in the same macro is supported for example tm
85. 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 thead 04 F remembered between calls ell aie el i 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 Polycom recommends that you do not change these values POLYCOM 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 chassis IP_6000 0 voice gain rx analog chassis IP_7000 0 voice gain rx analog ringer 0 voice gain rx analog ringer IP_300 6 voice gain rx analog ringer IP_330 0 Configuration Files Attribute Default voice gain rx analog ringer IP_430 0 voice gain rx analog ringer IP_601 6 voice gain rx analog ringer IP_650 0 voice gain rx analog ringer IP_6000 0 voice gain rx ana
86. 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 complete UDP 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 107 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 Note Configuring Your System Host
87. 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 assumed 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 107 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 92 Configuration file Enable this feature per registration phonet cfg For
88. the absence of a separate written agreement to the contrary Polycom shall be free to use any information suggestions or recommendations you provide to Polycom for any purpose subject to any applicable patents or copyrights 6 Harmful Code You agree not to include any Harmful Code in any products you develop by use of the API including but not limited to any code that i contains hidden files time bombs or viruses or ii can alter damage disclose or erase any data or other computer programs without control of a person operating the computing equipment on which it resides or iii retrieves or collects information without the consent of the user or for any illegal or unauthorized purpose or iv contains a key node lock time out or other function whether implemented by electronic mechanical or other means which restricts or may restrict use or access to programs or data on the Products frequency or duration of use or other limiting criteria or v any code which may restrict inhibit disrupt or interfere with the functionality of the Products as provided by Polycom You agree not to use the API for any illegal or unauthorized purpose 7 Marketing Trademarks You are free to market any products you develop using the API provided you agree not use the Polycom logo the marks Polycom SoundPoint SoundStation any other marks belonging or licensed to Polycom or any marks that are confusingly similar to marks belonging or
89. this prompt is invalid and all parameters depending on this prompt are invalid Note A mix of numeric and text is not supported lt version gt This element contains the version of the enhanced feature key elements The version element has the following format lt version efk version 2 gt If this parameter is omitted or has an invalid value including Null the enhanced feature key is disabled In SIP 3 0 1 is the only supported version In SIP 3 1 2 is the only supported version Special Characters The following special characters are used to implement the enhanced feature key functionality e The characters following it are a macro name e This character delimits the parts of the macro string This character must exist in pairs where the delimits the characters to be expanded e This character indicates that the following characters represent the expanded macro as in the action string Administrator s Guide SoundPoint IP SoundStation IP Macro names and action s unpredictable results may Macro Definition trings cannot contain these characters If they do occur The action string in the efklist element can be defined by either e Macro Action e Prompt Macro Substitution e Expanded Macros Macro Action The action string is executed in the order it appears User input is collected before any action is taken The action string contains the follo
90. 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 registration to the call server refer to Rebooting the Phone on page C 10 The display shows Network Link is Down The LAN 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 registration to the call server refer to Rebooting the Phone on page C 10 Calling Troubleshooting Your SoundPoint IP SoundStation IP Phones Symptom Problem Corrective Action There is no dial tone Power is not correctly applied to the SoundPoint IP family SIP phone Dial tone is not present on one of audio paths 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 Do one of the
91. 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 Administrator s Guide SoundPoint IP SoundStation IP 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 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 Multiple Key Combinations C 10 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 pol
92. 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 call to the phone_number and SIP_address from the corporate directory Configuration Files Attribute dir corp attribute x sticky Permitted Values 0 1 Default Null Interpretation If set to 0 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 dir corp backGroundSync UTF 8 encoded string 0 1 Null The filter string for this attribute which is edited when searching If set to 0 or Null there will be no background downloading from the LDAP server If set to 1 there will be background downloading of data from the LDAP server dir corp backGroundSync period 3600 to 604800 seconds 86400 The corporate directory cache is refreshed after the corporate directory feature has not been used for this period of time The default period is 24 hours The minimum is 1 hour and the maximum is 7 days dir corp viewPersistence 0 1 If set to 0 the browse position in the data on the LDAP server and the attribute filters are rese
93. voice soundpoint_ip VoIP_Technical_Bull etins_pub html 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 manual we data entry POLYCOM The followi
94. 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 36 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 One of Null Sets the name of the method for which request 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 kai some cases therefore use it wisely MESSAGE SUBSCRIBE NOTIFY REFER PRACK or UPDATE volpProt SIP requestValidation x Null or Null If Null no validation is done Otherwise this method 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 challenge requests with digest authentication using the local credentials for the associated registration line both or all apply both of the above
95. 0 330 bitmap IP_330 61 name IP 430 bitmap IP_400 61 name IP 501 bitmap IP_500 61 name IP 550 560 600 601 650 670 bitmap IP_600 65 name IP 4000 6000 bitmap IP_4000 65 name IP 7000 bitmap IP_7000 65 name For example lt bitmaps gt lt IP_300 n gt lt IP_330 bitmap IP_330 61 name logo 330 gt Administrator s Guide SoundPoint IP SoundStation IP lt IP_400 bitmap IP_400 61 name logo 430 gt lt IP_500 bitmap IP_500 61 name logo 500 gt lt IP_600 bitmap IP_600 65 name logo 600 gt lt IP_4000 bitmap IP_4000 65 name logo 4000 gt lt IP_7000 bitmap IP_7000 65 name logo 7000 gt 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_330 30 frame 1 bitmap Logo 330 ind anim IP_330 30 frame 1 duration 0 gt lt IP_330 gt lt IP_400 gt lt IDLE_DISPLAY ind anim IP_400 30 frame 1 bitmap logo 400 ind anim IP_400 30 frame 1 duration 0 gt lt IP_500 gt lt IP_500 gt lt IDLE_DISPLAY ind anim IP_500 42 frame 1 bitmap logo 500 ind anim IP_500 42 frame 1 duration 0 gt lt IP_500 gt lt IP_600 gt lt IDLE_DISPLAY ind anim IP_600 46 frame 1 bitmap
96. 0 bridged line appearances 4 27 browser limits A 98 busy lt busy gt A 115 busy lamp field 4 28 C call control lt callControl gt A 56 call control third party B 9 call forwarding 4 20 A 114 call handling configuration lt call gt A 64 call hold 4 17 call log 4 3 call park retrieve 4 22 call progress patterns A 33 call progress tones synthesized 4 32 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 111 central provisioning overview 2 6 changing the key on the phone C 5 chord sets lt chord gt A 29 codec preferences lt codecPref gt A 38 codec profiles lt audioProfile gt A 41 comfort noise fill 4 77 conference setup lt conference gt A 16 configurable feature keys 4 24 configurable soft keys 4 50 configuration file encryption 4 82 configuration file example 4 62 connected party identification 4 5 consultative transfers 4 18 context sensitive volume control 4 5 corporate directory 4 35 A 69 A 92 custom certificates 4 81 customizable audio sound effects 4 5 customizable fonts and indicators 4 29 D daisy chaining phones 4 38 date and time lt datetime gt A 25 default feature key layouts C 12 default password 3 5 4 83 C 11 C 27 deploying phones from the boot server 3 14 device lt device gt A 124 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 di
97. 0 messaging lt msg gt A 119 Index 4 Microbrowser 4 32 4 65 microphone mute 4 13 Microsoft Live Communications Server 2005 Integration 4 61 migration dependencies C 9 miscellaneous patterns A 35 missed call configuration lt serverMissedCall gt A 112 missed call notification 4 4 model number substitution A 4 modifying network configuration 3 5 multilingual lt ml gt A 22 multilingual user interface 4 30 multiple call appearances 4 25 multiple line keys per registration 4 25 multiple registrations 4 55 music on hold 4 18 music on hold lt musicOnHold gt A 17 N Network Address Translation lt nat gt A 120 network configuration modifying 3 5 network monitoring lt netMon gt A 59 new features 2 13 no answer lt noanswer gt A 115 O Option 66 3 7 outbound proxy lt outboundProxy gt A 14 P packet error concealment 4 74 password lt pwd gt A 89 patterns lt pat gt A 32 patterns lt pattern gt A 82 peer networking lt pnet gt application configuration peer networking A 100 per phone configuration attendant A 121 automatic off hook call placement A 112 busy A 115 calls A 111 dial plan emergency A 119 digit map A 117 do not disturb A 112 A 116 forward all A 114 message waiting indicator A 120 messaging A 119 missed call configuration A 112 Network Address Translation A 120 no answer A 115 quotas A 94 registration A 107 roaming buddies A 122 roaming privacy A 123 routing A 118 routi
98. 0 x 1 IP 7000 x 1 URL 6416 or 6416 polycom com 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 diversion feature refer to 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 enabled A 114 Attribute Busy lt busy gt Configuration Files Calls can be automatically diverted when the phone is busy Permitted Values De
99. 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 relevantifthe 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 34 se rt x callWait integer only relevantifthe 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 33 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 to 1 if the user interface should allow for modification by the user of the ringer index used for this ring class Note Configuration Files 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 Inte
100. 000 and 7000 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 Configuration Files 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 set 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 Note The following table defines the default usage of the sampled audio files with the phone Sampled Audio File A o ON DOO F ODN Ul o 12 24 Default use within phone pattern reference Ringer 12 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 Ring
101. 1 Public Primary Certification Authority Verisign Class 1 Public Primary Certification Authority G2 Verisign Class 1 Public Primary Certification Authority G3 Verisign Class 2 Public Primary Certification Authority Verisign Class 2 Public Primary Certification Authority G2 Verisign Class 2 Public Primary Certification Authority G3 Verisign Class 3 Public Primary Certification Authority Verisign Class 3 Public Primary Certification Authority G2 Verisign Class 3 Public Primary Certification Authority G3 Verisign Class 4 Public Primary Certification Authority G2 Verisign Class 4 Public Primary Certification Authority G3 Verisign RSA Commercial CA Verisign RSA Secure Server CA Administrator s Guide SoundPoint IP SoundStation IP POLYCOM Polycom endeavors to maintain a built in list of the most commonly used CA Certificates Due to memory contraints we cannot keep as thorough a list as some other applications for example browsers If you are using a certificate from a commercial Certificate Authority not in the list above you may submit a Feature Request for Polycom to add your CA to the trusted list by visiting https jira polycom com 8443 secure Createlssue default jspa 0s_usernamesjirag uest amp 0s_password polycom At this point you can use the Custom Certificate method to load your particular CA certificate into the phone refer to Technical Bulletin 17877 using Custom Certificates on SoundPoint IP Phones at h
102. 1 m2 e A sequence of speed dial references is supported for example 1 2 e A sequence of macro names and speed dial references is supported for example m1 2 m2 e Macro names that appear in the local contact directory must follow the format lt macro name gt where lt macro name gt must match an lt elklist gt mname entry The maximum macro length is 100 characters e Asequence of macros is supported but cannot be mixed with other action types e Action strings that appear in the local contact directory must follow the format lt action string gt Action strings can reference other macros or speed dial indexes Protection against recursive macro calls exists the enhanced feature keys fails once 50 macro substitutions is reached Examples For BroadWorks specific examples refer to Technical Bulletin 42250 Using Enhanced Feature Keys and Configurable Soft Keys on SoundPoint IP Phones with BroadWorks at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html For specific examples for other call servers go to http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Administrator s Guide SoundPoint IP SoundStation IP Configuration File Changes You must make the the following changes to the lt feature gt parameter that is defined in the sip cfg configuration file lt feature feature 18 n
103. 160 IP 600 1 320 IP 4000 1 248 IP 7000 1 256 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 IP 7000 1 128 For Graphic Icon type indicators this is the height of the indicator measured in pixels Event Logging lt log gt 84 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 O a A O PD 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 Configuration Files 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 event class cumulative log events missed due to excessive CPU load 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 sec
104. 2 Overview 65665664 6 Sk 4 oO eh eee Ne eee 22l Where SoundPoint IP SoundStation IP Phones Fit 2 2 Session Initiation Protocol Application Architecture 2 3 BootROM 3 i ocacvchadnee enpre hee tea tee btees e aaa ad autehaws 2 3 Application ssh Seale bog hh a Oech le pos aE E ESES 2 4 Configuration ne me re a 2 5 Resource BIOS errire tina Ace Sorte Nea eg ERAAN ERREEN endo been ages Sen re 2 7 Available Features 00 cece cence teen nee ens 2 8 New Features in SIP 3 1 2 0 2 ee eee cece eens 2 13 3 Setting up Your System cece eee eee eee Ord Setting Up the Network 0 00 00 2 3 2 DHCP or Manual TCP IP Setup 0 eee eee ee eee 3 2 Supported Provisioning Protocols 0 00008 3 4 Modifying the Network Configuration 000 3 5 Setting Up the Boot Server 0 eee eee eee 3 12 Deploying Phones From the Boot Server 0 000 3 14 Upgrading SIP Application 0 002 eee eee 3 19 Supporting SoundPoint IP and SoundStation IP Phones 3 19 Supporting SoundPoint IP 300 and 500 Phones 3 20 4 Configuring Your System ce cece cece ee Al Setting Up Basic Features 0 vise tiina iia 4 1 Call LOS Snccandeekenweceklos bebe E EEE EEEE EE nin EE E 4 3 Administrator s Guide SoundPoint IP SoundStation IP vi Cali IEn 275
105. 2 weight 1 cache SRV 2 port 5075 cache SRV 2 target secondary sipserver example com The reg 1 server 1 port and reg 1 server 2 port values in this example are set to null to force SRV lookups Example 3 This example shows how to configure static DNS cache where your DNS provides NAPTR and SRV records for server X address When the static DNS cache is not used the sip cfg configuration would look as follows reg 1 address 1002 sipserver example com reg 1 server 1 address 172 23 0 140 reg l server 1 port 5075 reg 1 server 1 transport UDPOnly reg 1 server 2 address 172 23 0 150 reg 1 server 2 port 5075 reg 1l server 2 transport UDPOnly When the static DNS cache is used the sip cfg configuration would look as follows reg reg reg reg reg reg reg dns dns dns address 1002 server 1 address sipserver example com server 1 port server 1 transport server 2 port Pe PPP Pe 1 1 server 2 address 2 2 server 2 transport cache NAPTR 1 name Sipserver example com cache NAPTR 1 ttl 3600 cache NAPTR 1 order 1 Administrator s Guide SoundPoint IP SoundStation IP dns cache NAPTR 1 preference 1 dns cache NAPTR 1 flag s dns cache NAPTR 1 service SIP D2U dns cache NAPTR 1 regexp dns cache NAPTR 1 replacement _sip _udp sipserver example com dns cache SRV 1 name _sip _udp sipserver example com dns cache SRV 1 ttl 3600 dns cac
106. 20 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 and 670 phones IP_4000 parameters affect the SoundStation IP 4000 and 6000 phones and IP_7000 parameters affect the SoundStation IP 7000 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 IP_7000 y 1 30 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 The following table lists the functions that are available Functions ArrowDown Dialpad5 Line2 Select ArrowLeft Dialpad6 Line3 Setup ArrowRight Dialpad7 Line4 SoftKey1 ArrowUp Dialpad8 Line5 SoftKey2 BuddyStatus Dialpad9 Line6 SoftKey3 CallList DialpadStar Messages SoftKey4 Conference DialpadPound Menu SpeedDial Delete Directories MicMute SpeedDialMenu Dialpado DoNotDisturb MyStatus Transfer Dialpad1 Handsfree Null VolDown Dialpad2 Headset Offline VolUp Dialpad3 Hold Redial Dialpad4 Line1 Release A 76 Backgrounds lt bg gt Configuration Files The ba
107. 26 9000 4750 Willow Road Fax 408 526 9100 Pleasanton CA 94588 U S A By downloading the following Sample Applications you agree to the below end user license agreement LICENSE AGREEMENT FOR DEVELOPMENT PURPOSES This License Agreement for Development Purposes the Agreement is a legal agreement between you and Polycom Inc a Delaware corporation Polycom The software you are about to download the Software comprises sample code that may be useful in the development of applications designed to operate on or in conjunction with Polycom Products Polycom is willing to license the Software to you only upon the condition that you accept all of the terms contained in this agreement Select the Accept button at the bottom of the page to confirm your acceptance If you are not willing to be bound by these terms select the Do Not Accept button and the downloading process will not continue PLEASE NOTE POLYCOM OFFERS NO SUPPORT FOR THIS SOFTWARE AND THE SOFTWARE IS BEING LICENSED WITHOUT DOCUMENTATION WITHOUT WARRANTY AS IS AND WITH ALL FAULTS THE SOFTWARE HAS NOT BEEN TESTED BY POLYCOM AND SHOULD NOT BE LOADED ON PRODUCTION SYSTEMS 1 GRANT OF LICENSE 1 1 License Subject to the terms of this Agreement Polycom grants to you a nonexclusive nontransferable license to copy install use and modify the Software including the Software in source code format and to produce your own commercial or othe
108. 3152407 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 anpa noe Ma Da SPO POSCOPGOGOS a ll cl ae a cl ooo oe ol oo oo ol oo oo od oll ol ood ooooc0cococo0c0ccoc0c0c0c0c0c0cC0cCc0cC0cC0cC000o amp a so se Manual Log Upload oopoooooooooooooooo0oo0oo0o0o0000O0 1 pA ee ey eee showCpuLoad od tarm 47293 er AA EEEE EEE EEEE EEEE EEEE EEEE EEE EEEE EE EEE EE EE EEE EE EEE EE EE E id and the average is 90 8 EEEE EEEE EEEE EEE EE EEE EEEE HHS EEEEEEEEEE EEE EEEE EE EEEE EEEE EEEE EEEE EEEE E E E E E Running memShow verer RRE status bytes blocks avg block max block current free 6631528 45 147367 6430376 alloc 5498224 23179 237 cumulative alloc 10503508 47342 221 HHAAHHHHHHHHKHHHHHHHHHHHHHH HAAS HHH HHH HHH HHH EREE E E E EEEE E EEE EEEE EEEE E E EE EEE EE EEE E E EEE E E E a Running showCpuLoad Cpu load is 0 0 and the average is 88 5 EEEE EEE EEEEEEE EEEE EEEE EEEE EEEE EEEE EEEE EEEE E Ed ERRE E E E E EE EE EEEE EE EE EEE E E E E E EE E EE E EE E E E E E E E E a Running memShow status bytes blocks avg block max block current free 6631528 45 147367 6
109. 3FF 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 FFOO U FFFF Note Within a Unicode range some characters may not be supported due to their infrequent usage A 24 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 set to 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 set to 1 display the day and month in long format Friday November otherwise use abbreviations Fri Nov Icl datetime date dateTop 0 1 If set to 1 display date 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
110. 4000 150 33 monochrome IP 6000 150 33 32 bit grayscale or monochrome IP 7000 255 128 32 bit grayscale or 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 Miscellaneous Administrative Tasks 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 SoundPoint IP 670 phones support a 12 bit color scale from black 0 0 0 to white FFFF FFFF FFFF The SoundStation IP 4000 phone only supports black and white Any other colors will be rendered as either black or white The SoundStation IP 6000 phone is the same as the IP 7000 The SoundStation IP 7000 phone supports a 32 bit grayscale which is a smooth gradient from black 0 0 0 to white FF FF FF Configuration File Changes In the lt 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 Model Associate Parameter IP 301 n a IP 32
111. 430376 alloc 5498224 23179 237 cumulative alloc 10509324 47391 221 HHAAHAHHHHHHHHHHHHHHHHHHHHAHAHAHHHH HHH HHH HHS EEEE EEE EEEE EE EEE EE EEE EEE EE EEE EEEE E HHH HHH HHS Running showGperl gad Bs k a LP a sate o_o a Ns te 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 For more information refer to Multiple Key Combinations on page C 10 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 Administrator s Guide SoundPoint IP SoundStation IP Reading a Boot Log The following figure shows a portion of a boot log file EE a Moles Initial log entry 0223214053 so 00 m 0229214053 vdog 400 1 Pht Note that bootrom log tines are in GMT 0223214053 Hae Int isl log entry 00 0225214053 odp L00 JCD SCDE is DISABLED cdp 0223214053 so 3 00 P 1802 10 VLAN tagging is DISABLED 0223214053 a 3100 Eee Board 2345 11500 020 A 0223214053 3 00 Platform MAC 0004 2015a51 IP Unknown Subnet Mask Unknown 0223214053 so 00 Platform BootBlock 2 5 0 11500_020 20 Aug 04 16 05 0223214053 so Q0 Application main Label BOOT Version 3 0 1 0026 29 Mar 05 10 29 0223214053 so O0 Application main P N 3150 11069 301 0223214053lapp1 4 Tnitial log
112. 5534 RTP port otherwise the 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 idleTransmitinterval tcplpApp keepalive tcp noResponseTrasmitinterval 10 to 7200 5 to 120 Null 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 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 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
113. 7000 Siren22 32kbps voice codecPref IP_7000 Siren22 48kbps voice codecPref IP_7000 Siren22 64kbps voice codecPref IP_7000 Siren14 24kbps voice codecPref IP_7000 Siren14 32kbps voice codecPref IP_7000 Siren14 48kbps Permitted Values Null 1 16 Default Null Null Null Null Null Null Null Null Null Null Null Interpretation Specifies the codec preferences for the SoundStation IP 7000 platform Interpretation as above Note Codecs with a default of Null are available for test purposes only and are not expected to be used in your deployment Codec Profiles lt audioProfile gt Configuration Files The following profile attributes can be adjusted for each of the five supported codecs In the table x G711Mu G711A G722 G7221 G7221C and G729AB Lin16 Siren14 and Siren22 Attribute Permitted 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 size for G7221 G7221C Siren14 and Siren22 are further subdivided voice audioProfile x jitterBufferMin 20 40 50 60 multiple of 10 The smallest jitter buffer depth in milliseconds that must be achieved bef
114. 959 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 08 txt Diversion Indication in SIP e draft anil sipping 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 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 sup
115. 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 Configuration file Specify the location of the central report collector how often the boot server sip cfg reports are generated and the warning and critical threshold values that will cause generation of alert reports e For more information refer to Quality Monitoring lt quality monitoring gt on page A 52 Dynamic Noise Reduction Dynamic noise reduction DNR provides maximum microphone sensitivity while automatically reducing background noise from fans projectors heating and air conditioning for clearer sound and more efficient conferencing There are no related configuration changes 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 Configuration changes can performed cent
116. ANTIES 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 cannot 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 r
117. 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 Administrator s Guide SoundPoint IP SoundStation IP Status Menu Status When the phone is unable to register with the call control server the icon 3 isshown outline Once the phone is registered the icon T is shown solid On the SoundStation IP 7000 the icons are gt and 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
118. E 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 names of the authors and copyright holders must not be used in advertising or otherwise to promote the sale use or other dealing in this Software without specific written prior permission Title to copyright in this Software shall at all times remain with copyright holders OpenLDAP is a registered trademark of the OpenLDAP Foundation Copyright 1999 2003 The OpenLDAP Foundation Redwood City California USA All Rights Reserved Permission to copy and distribute verbatim copies of this document is granted 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 openss l core openssl org OpenSSL License Copyright c 1998 2003 The OpenSSL Project All rights reserved Redistribution and use in source and binary forms with or wi
119. EE 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 jitter 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 This feature is part of the Productivity Suite 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
120. Enable or disable all call lists or individual call lists For more information refer to Feature lt feature gt on page A 92 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 phone1 cfg active For more information refer to Call Waiting lt callWaiting gt on page A 113 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 pre
121. FeatureControl 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 is not Null this attribute overrides the global server based call forwarding flag in the sip cfg configuration file Configuration Files Attribute reg x auth optimized InFailover reg x strictLineSeize Permitted Values 0 1 0 1 Default 0 Null Interpretation 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 If set to 1 forces phone to wait for 200 OK on registration x when receiving a TRYING notify If set to O or Null this is old behavior If this parameter is Null volpProt SIP strictLineSeize Is checked If both parameters are set this parameter takes precedence reg x musicOnHold uri string Null A URI that provides the media stream to play for the remote party on hold When present and if reg x musicOnHoldis not Null this attribute overrides the global Music on Hold defined in the sip cfg configu
122. HCP 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 107 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 hid if me ee aoaress p arameter 65535 in Registration lt reg gt on page A 107 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 voIpProt server x addressisa 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 i
123. I ie Menu ah d L 5 Line 1 g E 34 S oz 10 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 13 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 SoftKey1 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 14 SoundPoint IP 501 D EROR o o moppanoyg FA POLYCOM Miscellaneous Administrative Tasks
124. IFY 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 B 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 Internal Key Functions Assigning a VLAN ID Using DHCP Parsing Vendor ID Information Product Model and Part Number Mapping Disabling PC Ethernet Port 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 CA American Express Global CA BelSign Object Publishing CA BelSign Secure Server CA Administrator s
125. IP 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 conditions Changing the default daylight savings parameters will likely be necessary outside of North American locations a Optional Disable the local web HTTP server or change its signalling port if local security policy dictates b 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 b 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 Setting up Your System c 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 d Edit the LOG_FILE_DIRECTORY attribute of the
126. IP_6000 preFilter enable voice txEq hf IP_7000 preFilter enable voice txEq hf IP_330 postFilter enable oOo o o 51 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default 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 txEq hf IP_6000 postFilter enable 0 voice txEq hf IP_7000 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 voice vad 0 1 Null If set to 1 or Null and voice vadEnable is set to 1 signalAnnexB Annex B is used A new line can be added to SDP depending on the setting of this parameter and the voice vadEnable parameter e 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 If voice vadEnable is set to 0
127. 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 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 ld 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
128. If set to O or Null the soft key is not displayed in the idle state If set to 1 the soft key is displayed in the idle state softkey x use active 0 default 1 Null If set to 0 or Null the soft key is not displayed in the active call state If set to 1 the soft key is displayed in the active call state A 103 Administrator s Guide SoundPoint IP SoundStation IP Attribute softkey x use alerting Permitted Values 0 default 1 Default Null Interpretation If set to 0 or Null the soft key is not displayed in the alerting state If set to 1 the soft key is displayed in the alerting state softkey x use dialtone 0 default 1 Null If set to 0 or Null the soft key is not displayed in the dialtone state If set to 1 the soft key is displayed in the dialtone state softkey x use proceeding 0 default 1 Null If set to 0 or Null the soft key is not displayed in the proceeding state If set to 1 the soft key is displayed in the proceeding state softkey x use setup 0 default Null If set to 0 or Null the soft key is not displayed in the setup state If set to 1 the soft key is displayed in the setup state softkey x use hold 0 default 1 Null If set to 0 or Null the soft key is not displayed in the hold state If set to 1 the soft key is displayed in the hold state softkey feature newcall 0 1 default
129. 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 Do one of the following e Refer to the appropriate SoundPoint IP SoundStation IP SIP phone User Guide Reboot the phone to obtain a default level of contrast refer to Rebooting the Phone on page C 10 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 Audio Troubleshooting Your SoundPoint IP SoundStation IP Phones 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 Ensure the headset amplifier if present is turned on and or the volume is correctly adjusted
130. NFIG_FILES_2345 11605 001 phone1_2345 11605 001 cfg sip_2345 11605 001 cfg will override CONFIG_FILES_SPIP601 phone1_SPIP601 cfg sip_SPIP601 cfg which will override CONFIG_FILES phonel cfg sip cfg for an SoundPoint IP 601 You can also add variables to the master configuration file that are replaced when the phone reboots The variables include PHONE_MODEL PHONE_PART_NUMBER and PHONE_MAC_ADDRESS The following table shows the product name model name and part number mapping for SoundPoint IP and SoundStation IP phones Product Name Model Name Product Part Number SoundPoint IP 301 SPIP301 2345 11300 001 SoundPoint IP 320 SPIP320 2345 12200 002 2345 12200 005 SoundPoint IP 330 SPIP330 2345 12200 001 2345 12200 004 SoundPoint IP 430 SPIP430 2345 11402 001 SoundPoint IP 501 SPIP501 2345 11500 030 2345 11500 040 SoundPoint IP 550 SPIP550 2345 12500 001 SoundPoint IP 560 SPIP560 2345 12560 001 SoundPoint IP 600 SPIP600 2345 11600 001 SoundPoint IP 601 SPIP601 2345 11605 001 SoundPoint IP 650 SPIP650 2345 12600 001 SoundPoint IP 670 SPIP670 2345 12670 001 SoundStation IP 4000 SSIP4000 2101 06642 001 SoundStation IP 6000 SSIP6000 3111 15600 001 SoundStation IP 7000 SSIP7000 3111 40000 001 Miscellaneous Administrative Tasks Disabling PC Ethernet Port Certain SoundPoint IP phones have a PC Ethernet port If it is unused it can be disabled The PC Eth
131. OM e Microbrowser lt mb gt e Applications lt apps gt e Peer Networking lt pnet gt e DNS Cache lt dns gt e Soft Keys lt softkey gt 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 This attribute includes e Local lt local gt e Server lt server gt SDP lt SDP gt e SIP lt SIP gt e Music on Hold lt musicOnHold 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 packets If set to O 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 D
132. P gt on page A 56 Local Web Server Specify IP TOS settings if enabled Navigate to http lt phonelPAddress gt netConf htm qo IEEE 802 1p Q 4 78 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 packets such as SIP signaling packets with default settings configurable for all other packets Configuring Your System Configuration changes can performed centrally at the boot server or locally Central boot server Local Configuration file Specify default and protocol specific 802 1p Q settings sip cfg e For more information refer to Ethernet IEEE 802 1p Q lt ethernet gt on page A 55 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
133. P 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 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 107 call stickyAutoLineSeize Null 0 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 onHookDialing 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 n
134. 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 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 remov
135. P_650 0 1 2 2048 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 Administrator s Guide SoundPoint IP SoundStation IP Feature lt feature gt 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 calll
136. Point IP SoundStation IP Provisioning SoundStation IP 7000 Phones Using CLink Normally the SoundStation IP family conference phone is provisioned over the Ethernet by the boot server However when two SoundStation IP family phones are daisy chained together the one that is not directly connected to the Ethernet can still be provisioned known as the secondary 12 foot Ethernet Cable 25 foot lt Network Cable Gy E The provisioning over CLink feature is automatically enabled when a SoundStation IP family phone is not connected to the Ethernet Both SoundStation IP family phones must be running the same version of the SIP application The steps for provisioning the secondary SoundStation IP family phone are the same as for the primary SoundStation IP family 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 family phones the primary will power up first Currently provisioning over CLink is supported for the following configurations of SoundStation IP family conference phones e Two SoundStation IP family conference phone daisy chained together e Two SoundStation IP family conference phone daisy chained together with one external microphone specifically designed for the SoundStation IP family conference phone The provisioning boot server
137. SIP signalling connections that use TLS transport 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 Permitted Attribute Values Default Interpretation call rejectBusyOnDnd 0 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 be
138. ST tcplpApp sntp daylightSavings 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 lastInMonth 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 6
139. Server Option Type 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 next section Server Menu Setting up Your System Name VLAN Discovery Possible Values Description 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 23 Note If multiple alternate DHCP servers respond e The phone should gather the responses from alternate DHCP servers e If configured for Customt Option66 the phone will select the first response that contains a valid custom option value e If 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
140. 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 122 Locate the roaming_privacy attribute Set the roaming_privacy reg element to 1 Refer to Roaming Privacy lt roaming_privacy gt on page A 123 Save the modified phonel cfg configuration file Administrator s Guide SoundPoint IP SoundStation IP 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 voIpProt 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 phone1 cfg configuration file as follows a Open phonel cfg in an XML editor Locate the registration parameter Select a reg
141. Syslog Menu on page 3 11 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 Configuration Files 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 new 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
142. The following related documents for SoundPoint 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 eee e ccc c cece cece H 1 Introducing the SoundPoint IP SoundStation IP Family 1 1 SoundPoint IP Desktop Phones 0 0 eee eee eee 1 1 SoundStation IP Conference Phones 0000000 00008 1 4 Key Features of Your SoundPoint IP SoundStation IP Phones 1 6
143. 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 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
144. 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 4 74 Configuring Your System 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 e For more information refer to Voice Activity Detection lt vad gt on page A 52 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 Frequency lt DTMF gt on page A 28 DTMF Event RTP Payload The phone is compatible with RFC 2833 RTP P
145. 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 124 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 phone cfg on the boot server for security reasons Custom Certificates Note 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 For more information on using custom certificates refer to Technical Bulletin 17877 Using Custom Certificates With SoundPoint IP Phones at http www polycom com usa en support voice soundpoint_ip VoIP_Technical_Bulle tins_pub html Administrator s Guide SoundPoint IP SoundStation IP 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 i
146. 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 Configuration Files Music on Hold lt musicOnHold gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP musicOnHold uri string Null A URI that provides the media stream to play for the remote party on hold If reg x musicOnHold is set to Null this attribute is checked Note The SIP URI parameter transport is supported when configured with the values of UDP TCP or TLS 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 Permitted Attribute Values Default Interpretation dialplan apply ToCallListDial 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 0 This attribute covers
147. WARRANTIES TERMS OR CONDITIONS EXPRESS OR IMPLIED EITHER IN FACT OR BY OPERATION OF LAW STATUTORY OR OTHERWISE INCLUDING WARRANTIES TERMS OR CONDITIONS OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE SATISFACTORY QUALITY CORRESPONDENCE WITH DESCRIPTION AND NON INFRINGEMENT ARE EXPRESSLY DISCLAIMED POLYCOM NEITHER ASSUMES NOR AUTHORIZES ANY OTHER PERSON TO ASSUME FOR IT ANY OTHER LIABILITY IN CONNECTION WITH THE SALE INSTALLATION MAINTENANCE OR USE OF THIS SOFTWARE 6 LIMITATION OF LIABILITY 6 1 Limitations TO THE MAXIMUM EXTENT PERMITTED BY APPLICABLE LAW IN NO EVENT SHALL POLYCOM OR ITS SUPPLIERS BE LIABLE FOR ANY SPECIAL INCIDENTAL INDIRECT OR CONSEQUENTIAL DAMAGES WHATSOEVER INCLUDING WITHOUT LIMITATION DAMAGES FOR LOSS OF BUSINESS PROFITS BUSINESS INTERRUPTION LOSS OF BUSINESS INFORMATION OR ANY OTHER PECUNIARY LOSS ARISING OUT OF THE USE OR INABILITY TO USE THE SOFTWARE EVEN IF POLYCOM HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES IN ANY CASE POLYCOM S ENTIRE LIABILITY SHALL BE LIMITED TO THE GREATER OF THE AMOUNT ACTUALLY PAID BY YOU FOR THE SOFTWARE OR USS 5 00 7 DISCLAIMER 7 1 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 limi
148. 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 For more information refer to Message Waiting Indicator lt mwi gt on page A 120 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 25 Distinctive 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
149. When another party calls the ToVMail soft key is displayed When the user presses ToV Mail soft key the other party is transferred to voice mail Administrator s Guide SoundPoint IP SoundStation IP 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 user 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 430 500 501 550 560 600 601 650 and 670 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 For more information refer to User Preferences lt up gt on page A 25 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 430 500 501 550 560 600 601 650 and 670 Instant messages are still accessible from the Main Menu On a p
150. 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 Yes Only warning codes 300 to 399 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 1xx Responses Provisional Response Supported Notes 100 Trying Yes 180 Ringing Yes 181 Call ls 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
151. _650 preFilter enable e voice rxEq hd IP_330 postFilter enable voice rxEq hd IP_430 postFilter enable voice rxEq hd IP_500 postFilter enable voice rxEq hd IP_600 postFilter enable voice rxEq hd IP_601 postFilter enable voice rxEq hd IP_650 postFilter enable Gl O OT S a G 49 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default 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_6000 preFilter enable 0 voice rxEq hf IP_7000 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 voice rxEq hf IP_4000 postFilter enable voice rxEq hf IP_6000 postFilter enable a OoN G O voice rxEq hf IP_7000 postFilter enable 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 hs IP_330 preFilte
152. _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 Remove any lt Ethernet address gt cfg files that may have been used with earlier releases from the boot server 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 used 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 Administrator s Guide SoundPoint IP SoundStation IP 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 usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuring Your System After you set up your SoundPoint IP SoundStation IP phones on the netw
153. 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 54 Instant Messages will still be accessible from the Main Menu This attribute also includes e Message Waiting Indicator lt mwi gt 119 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 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 x 1 This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation msg mwi x subscribe 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 If non Null the phone will send a SUBSCRIBE request to this contact after boot up msg mwi x callBackMode contact or registration or disabled registration Configures message retrieval and notification for the line If set to contact a call will
154. aec gt These settings control the performance of the speakerphone acoustic echo canceller Polycom recommends that you do not change these values we POLYCOM Attribute Default voice aec hs enable 1 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 45 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice aec hd lowFreqCutOff 100 voice aec hd highFreqCutOff 7000 voice aec hd erlTab_0 300 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 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 These 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 1 voice aes hs duplexBalance 7 voice aes hd enable 0 voice aes hd duplexBalance 0 voic
155. afts 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 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 3
156. agnostics phone 5 9 dial plan lt dialplan gt A 17 digit map default A 20 examples A 18 match and replace A 18 digit map lt digitmap gt A 117 directed call pick up 4 21 directory lt dir gt A 68 distinctive call waiting 4 7 distinctive incoming call treatment 4 6 distinctive ringing 4 7 diversion A 114 DND See also do not disturb DNS cache lt dns gt A 100 DNS SIP server name resolution 4 57 do not disturb 4 8 do not disturb lt dnd gt A 112 A 116 downloadable fonts 4 31 DTMF event RTP payload 4 75 DTMF tone generation 4 75 DTMF See also dual tone multi frequency dual tone multi frequency lt DMTF gt A 28 dynamic noise reduction 4 80 Index E electronic hookswitch supported 4 9 A 123 emergency lt emergency gt A 21 A 119 emergency routing A 21 A 119 encryption lt encryption gt A 89 enhanced feature keys 4 40 A 92 definition language 4 40 examples 4 47 macro definitions 4 44 useful tips 4 46 Ethernet IEEE 802 1p Q A 55 Ethernet menu 3 11 F feature lt feature gt A 92 feature licensing 4 19 4 34 4 37 4 79 A 93 features list of 1 6 finder lt finder gt A 94 flash parameter configuration A 124 flash parameter See also device fonts lt font gt A 72 forward all lt fwd gt A 114 G gains lt gain gt A 42 graphic display backgrounds 4 16 A 77 graphic icons lt gi gt A 83 group call pick up 4 22 H handset headset and speakerphone 4 8 hands free disabled A 27 ho
157. 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 57 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 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 Network Address Translation The phone can work with certain types of network address translation NAT The phone s signaling and RTP traffic use symmet
158. ame enhanced feature keys feature 18 enabled 1 gt Action String Example The action string SChangup 444 SP1IN4SS Tinvites Cwaitconnect P2N3 S Cpause2 Tdt mf Changup is executed as follows 1 The user is prompted for 4 digits For example 1234 The user is prompted for 3 digits For example 567 The user s active call is disconnected The string 444 1234 is sent using the INVITE method a Fw ND Once connected there is a 2 second pause and then the string 567 is sent using DTMF dialing on the active call 6 The active call is disconnected Speed Dial Example Your organization voice mail system is accessible through 7700 and your voice mail password is 2154 You could use a speed dial key to access your voice mail if you entered 7700 Cpause3 2154 as the contact number Enhanced Feature Key XML Files You must ensure that the following XML code exists for the definition of Call Park lt efklist efk efklist 2 mname callpark efk efklist 2 status 1 efk efklist 2 label Call Park efk efklist 2 use idle 1 efk efklist 2 use active 1 efk efklist 2 use alerting 1 efk efklist 2 use dialtone 1 efk efklist 2 use proceeding 1 efk efklist 2 use setup 1 efk efklist 2 type invite efk efklist 2 action string 68 SP1N105S gt lt efkprompt efk efkprompt 1 status 1 efk efkprompt 1 label Enter Number Note Configuring Your Syst
159. and FD means full duplex 4 100FD Note Polycom recommends that you do not change this 5 1000FD setting unless you want to disable the PC port 1 Disabled Note The LAN Port Mode and PC Port Mode parameters are only available on SoundPoint IP 330 430 550 560 601 650 and 670 phones Only the SoundPoint IP 560 and 670 supports the LAN Port Mode and PC Port Mode setting of 1000FD 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 Administrator s Guide SoundPoint IP SoundStation IP Name 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 Possible Values Description Server Address dott
160. ange 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 Administrator s Guide SoundPoint IP SoundStation IP 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 JP_500 font lt IP_500 gt e IP _600 font lt IP_600 gt IP_330 font lt IP_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
161. appear Fonts lt font gt 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 Configuration Files 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 lt fontHeightInPixels gt describes the font height in number of screen pixels 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 fnt file or Uxx00_UyyFF fon file For more information refer to Multilingual User Interface on page 4 30 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 a
162. 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 Overview Secure Real Time Transport Protocol Encrypting audio streams to avoid interception and eavesdropping For more information on each feature and its associated configuration parameters see the appropriate section in Configuring Your System on page 4 1 New Features in SIP 3 1 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 3 Any new features introduced after SIP 2 1 3 are not supported The following new features were introduced in SIP 3 1 e Access URL in SIP Message Ability for the SoundPoint IP phones to be able to receive a URL inside a SIP message for example as a SIP header extension in a SIP INVITE and subsequently access that given URL in the Microbrowser e Configurable Soft Keys Allows customers to create their own soft keys and have them displayed with or without the standard SoundPoint IP and SoundStation IP soft keys e Enhanced Feature Keys Allows customers to redefine soft keys to suit their needs In SIP 3 0 this feature required a license key e Dynamic Noise Reduction Provides maximum microphone sensitivity while automatically reducing background noise on Sou
163. apps telNotification Oor1 0 If set to 0 incoming call notification is disabled incomingEvent If set to 1 incoming call notification is enabled apps telNotification Oori 0 If set to 0 outgoing call notification is disabled outgoingEvent If set to 1 outgoing call notification is enabled apps telNotification Oor1 0 If set to 0 offhook notification is disabled offhookEvent If set to 1 offhook notification is enabled apps telNotification Oor1 0 If set to 0 onhook notification is disabled onhookEvent If set to 1 onhook notification is enabled Configuration Files Aitribute Permitted Values Default Interpretation apps statePolling URL URL Null The URL to which the phone sends call processing state device network information The protocol used can be either HTTP or HTTPS Note To enable state polling the attributes apps statePolling URL apps statePolling username and apps statePolling password must be set to non Null values apps statePolling username string Null The user name to access the state polling URL apps statePolling password string Null The password to access the state polling URL apps push messageType 0 to 3 Select the allowable push priority messages on phone The values are 0 None Discard push messages e 1 Critical Allows only critical push messages e 2 Normal Allows only normal push messages 3 Both Allows bo
164. aptive 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 41 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 the phone is compatible with the comprehensive codec independent comfort noise transmission algorithm specified in RFC 3389
165. arranty This 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
166. arting from the leftmost empty space follows the empty spaces Any extra custom soft keys that are left after all empty spaces are used are appended at the end Up to 10 soft keys can be configured Any additional soft keys are ignored If more soft keys are defined than fit on the graphic display at one time a More soft key is displayed followed by the remainder of the soft keys that you have defined This capability applies to the SoundPoint IP 301 320 330 430 501 550 560 601 650 and 670phones This capability is linked to the Enhanced Feature Key feature refer to Enhanced Feature Keys on page 4 40 Configuration changes can be performed centrally at the boot server Central boot server Configuration file Specify the soft key label in what states it should be displayed and sip cfg prompt for input if required For more information refer to Soft Keys lt softkey gt on page A 103 Configuration File Examples For BroadWorks specific examples refer to Technical Bulletin 42250 Using Enhanced Feature Keys and Configurable Soft Keys on SoundPoint IP Phones with BroadWorks at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html For specific examples on other call servers go to http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html To disable the New Call soft key 1 Update the sip cfg configurat
167. ation A 45 acoustic echo suppression A 46 animations A 81 automatic gain control A 48 background noise suppression A 47 backgrounds A 77 bitmaps A 80 call handling configuration A 64 call progress patterns A 33 chord sets A 29 codec preferences A 38 codec profiles A 41 conference setup A 16 date and time A 25 dial plan A 17 dial plan emergency A 21 directory A 68 DNS cache A 100 dual tone multi frequency A 28 encryption A 89 Ethernet call control A 56 event logging A 84 feature A 92 finder A 94 fonts A 72 gains A 42 graphic icons A 83 hold local reminder A 67 idle display A 96 indicator classes A 82 indicator patterns A 82 indicators assignments A 82 IP TOS call control A 58 keep alive A 63 keys A 75 local protocol A 6 localization A 21 main browser A 97 multilingual A 22 music on hold A 17 network monitoring A 59 outbound proxy A 14 password lengths A 89 platform A 80 port A 62 presence A 72 protocol A 6 protocol server A 7 protocol special events A 16 provisioning A 90 Quality of Service A 55 RAM disk A 90 receive equalization A 49 request A 91 request delay A 91 request validation A 15 resource A 93 ring type A 36 routing server A 21 RTP A 56 A 57 A 62 Index 1 Administrator s Guide SoundPoint IP SoundStation IP sampled audio for sound effects A 30 SDP A 9 security A 88 shared calls A 67 SIP A 10 soft keys A 103 sound effect patterns A 32 sound effects A 31 ton
168. ation syntax is based on recommendations in 2 1 5 of RFC 3435 The phone behavior when the user dials digits that do not match the digit map is configurable It is also possible to strip a trailing from the digits sent or to replace certain matched digits with the introduction of R to the digit map 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 17 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 116 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 Wh
169. ault 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 attendant ringType 1 to 22 Null The ring tone to play when a BLF dialog is in the offering state 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 A 122 Roaming Privacy lt roaming_privacy gt Note Configuration Files This attribute is used in conjunction with Microsoft Live Communications Server 2005 only This configuration attribute is defined as follows Permitted Attrib
170. auto sensing e Multiple language support Set on screen language to your preference Select from Chinese Danish Dutch English French German Italian Japanese Korean Norwegian Polish Portuguese Russian Slovenian Spanish and Swedish Note In SIP 3 0 default support for Chinese Japanese and Korean was removed from the SoundPoint IP 600 and 601 e Microbrowser Supports a subset of XHTML constructs otherwise runs like any other Web browser Administrator s Guide SoundPoint IP SoundStation IP 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 Router
171. ayload 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 28 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 4 75 Administrator s Guide SoundPoint IP SoundStation IP 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 voice ns hs signalAttn 6 voice
172. bps Siren22 SIREN22 SIREN22 32 Kbps 32 Ksps 20ms 80ms 14 KHz 48000 48 Kbps 64 Kbps RFC 2833 phone event RFC 2833 N A N A N A N A 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 G 722 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 e For more information refer to Codec Preferences lt codecPref gt on page A 38 and Codec Profiles lt audioProfile gt on page A 41 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 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
173. c written prior permission M LT makes no representations about the suitability of this software for any purpose It is provided as is without express or implied warranty OpenLDAP The OpenLDAP Public License Version 2 8 17 August 2003 Redistribution and use of this software and associated documentation Software with or without modification are permitted provided that the following conditions are met 1 Redistributions in source form must retain copyright statements and notices 2 Redistributions in binary form must reproduce applicable copyright statements and notices this list of conditions and the following disclaimer in the documentation and or other materials provided with the distribution and 3 Redistributions must contain a verbatim copy of this document The OpenLDAP Foundation may revise this license from time to time Administrator s Guide SoundPoint IP SoundStation IP Each revision is distinguished by a version number You may use this Software under terms of this license revision or under the terms of any subsequent revision of the license THIS SOFTWARE IS PROVIDED BY THE OPENLDAP FOUNDATION AND ITS CONTRIBUTORS 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 OPENLDAP FOUNDATION ITS CONTRIBUTORS OR THE AUTHOR S OR OWNER S OF THE SOFTWARE BE LIABL
174. 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 e SoundStation IP Conference Phones For a list of key features available on the SoundPoint IP SoundStation IP phones running the latest software refer to Key Features of Your SoundPoint IP SoundStation IP Phones on page 1 6 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 Administrator s Guide SoundPoint IP SoundStation IP The currently supported desktop phones are e SoundPoint IP 301 CE E ECEE e SoundPoint IP 320 330 e SoundPoint IP 430 Introducing the SoundPoint IP SoundStation IP Family e SoundPoint IP 501 e SoundPoint IP 550 560 e SoundPoint IP 600 601 Administrator s Guide SoundPoint IP SoundStation IP e SoundPoint IP 650 e SoundPoint IP 670 SoundStation IP Conference Phones This secti
175. ces e Shared Call Appearances e For more information refer to Feature lt feature gt on page A 92 e For more information refer to Feature lt feature gt on page A 92 For more information refer to Feature lt feature gt on page A 92 For more information refer to Call Handling Configuration lt call gt Configuring Your System e Bridged Line Appearance e Busy Lamp Field e Customizable Fonts and Indicators 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 Directory e Recording and Playback of Audio Calls e Daisy Chaining Phones e Provisioning Phones Over CLink e Enhanced Feature Keys e Configurable Soft Keys 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 e Access URL in SIP Message e Static DNS Cache e Display of Warnings from SIP Headers Administrator s Guide SoundPoint IP SoundStation IP 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
176. ch 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 Administrator s Guide SoundPoint IP SoundStation IP Passwords Administrator Network Configuration password SIP Configuration required SSL Security settings Reset to Default local configuration device settings and file system format User password Reboot Phone required 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
177. ch the phone directs Domain Name 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 11 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 Offe
178. ciates a particular entry with a speed dial bin for one touch dialing or dialing from the speed dial menu Note On the SoundPoint IP 330 320 and the SoundStation IP 6000 and 7000 the maximum speed dial index is 99 rt Null 1 to 21 ring type When incoming calls can be associated with a directory entry by matching the address fields this field is used to specify ring type to be used dc UTF 8 encoded string divert contact containing digits the The forward to address for the autodivert feature user part of a SIP URL or a string that constitutes a valid SIP URL ad 0 1 auto divert If set to 1 automatically diverts callers that match the directory entry to the address specified in divert contact Note If auto divert is enabled it has precedence over auto reject ar 0 1 auto reject If set to 1 automatically rejects callers that match the directory entry Note If auto divert is also enabled it has precedence over auto reject bw 0 1 buddy watching If set to 1 add this contact to the list of watched phones bb 0 1 buddy block If set to 1 block this contact from watching this phone Local Digit Map The phone has a local digit map feature to automate the setup phase of number only calls When properly configured this feature eliminates the need for using the Dial or Send soft key when making outgoing calls As soon as a digit pattern matching the digit map is found the call setup process will complete automatically The configur
179. ckgrounds used by the SoundPoint IP 550 560 650 and 670 phones are defined in this section In the following table w 1 to 3 x 1 to 6 This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation bg hiRes color selection W X 1 1 Specify which type of background w and index for that type x is selected on reboot where w 1 to 3 x 1 to 6 bg hiRes color pat solid x any string Solid pattern name name For x 1 Light Blue x 2 Teal x 3 Tan x 4 Null bg hiRes color pat solid x red 0 to 255 The screen background layouts ba hiR l tsolid 0 to 255 For x 1 red 151 green 207 blue 249 hiRes color pat solid x o ie P For x 2 red 73 green 148 blue 148 For x 3 red 245 green 157 blue 69 bg hiRes color pat solid x blue 0 to 255 For x 4 red Null green Null blue Null bg hiRes color om x name any string built in Graphic files for display on the phone and value of Expansion Module bg hiRes color om x em name any string Thistle For x 1 name is Leaf jpg name is LeafEM jpg For x 2 name is Sailboat jpg name is SailboatEM jpg For x 3 name is Beach jpg name is BeachEM jpg For x 4 name is Palm jpg name is PalmEM jpg For x 5 name is Jellyfish jpg name is JellyfishEM jpg For x 6 name is Mountain jpg name is MountainEM j
180. cros Expanded macros are prefixed with the character and are inserted directly into the local directory contact field For more information refer to Local Contact Directory File Format on page 4 10 Configuration File Changes The configuration file changes and the enhanced feature key definitions can be included together in one configuration file A sample configuration for this feature including the enhanced feature keys definitions shown in the following section Examples may be included with the SIP 3 1 release Create a new configuration file in the style of sip cfg in order to make configuration changes 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 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 92 Configuration file Specify two calls per line key phonet cfg For more information refer to Registration lt reg gt on page A 107 XML file lt Ethernet This file holds the macro names which correspond to the mname fields address gt directory in the configuration file where the enhanced feature keys are defined xml Macro names must be embedded into the contact cn fields with the I
181. d 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 option 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 Miscellaneous Administrative Tasks 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 kw 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
182. d 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 Configuration Files 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 main idleTimeout 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 25 lf mo 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 Admini
183. d 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 The Presence feature allows the phone to monitor the status of other users devices and allows other users to monitor it The status of monitored 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 Configuring Your System The presence feature works differently when Microsoft Live Communications Server 2005 is used as the call server For more information refer to the next section Microsoft Live Communications Server 2005 Integration Configuration changes can performed centra
184. d to use a DNS server For more information refer to DNS Cache lt dns gt on page A 100 Configuration File Examples 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 Example 1 This example shows how to configure static DNS cache using A records IP addresses in SIP server address fields When the static DNS cache is not used the sip cfg configuration would look as follows reg 1 address 1001 reg 1 server 1 address 172 23 0 140 reg 1 server 1 port 5075 Administrator s Guide SoundPoint IP SoundStation IP Note reg 1 server 1 transport UDPOnly reg 1 server 2 address 172 23 0 150 reg 1l server 2 port 5075 reg 1l server 2 transport UDPOnly When the static DNS cache is used the sip cfg configuration would look as follows reg 1 address 1001 reg 1 server 1 address sipserver example com reg 1 server 1 port 5075 reg l server 1 transport UDPOnly reg l server 2 address reg l server 2 port reg l server 2 transport dns cache A 1 name sipserver example com dns cache A 1 ttl 3600 dns cache A 1 address 172 23 0 140 dns cache A 2 name sipserver example com dns cache A 2 ttl 3600 dns cache A 2 address 172 23 0 150 Above addresses are pr
185. d troubleshooting information you can access certain menus on the SoundPoint IP and SoundStation IP phone that test the phone hardware From the diagnostics menu you can test The phone s microphones speaker handset and any third party handset if present e Keypad mapping You can verify the function assign to each key e Graphic display You can test the LCD for faulty pixels To test the phone hardware gt gt Press Menu and then select Status gt Diagnostics gt Test Hardware gt Audio Diagnostics Keypad Diagnostics or Display Diagnostics Administrator s Guide SoundPoint IP SoundStation IP Power and Startup Symptom Problem Corrective Action There are power issues The SoundPoint IP SoundStation IP family SIP phone has no power Do one of the following Verify that no lights appear on the unit when it is powered up Check if the phone is properly plugged into a functional AC outlet Make sure that the phone isn t plugged into a plug controlled by a light switch that is off If plugged into a power strip try plugging directly into a wall outlet instead Try the phone in another room where the electricity is known to be working on a particular outlet If using PoE the power supply voltage may be too high or too low Troubleshooting Your SoundPoint IP SoundStation IP Phones Controls Symptom Problem Corrective Action The dial pad does n
186. 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 301 320 330 430 501 550 560 600 601 650 and 670 and the SoundStation IP 4000 6000 and 7000 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 2 5 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 information refer to Encryption lt encryption gt on page A 89 Configuration file Change the encryption key lt device gt cfg For more information refer to Flash Parameter Configuration on page A 124 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 Administrator s Guide SoundPo
187. ded 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 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 Troubleshooting Your SoundPoint IP SoundStation IP Phones The following figure shows an example of a configuration file and the resulting log file Change 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 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 0223152407 slog 022
188. 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 1 This attribute covers the case when 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 Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation dialplan apply ToUserSend 0 1 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 0 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 1 If set to 1 strip trailing digit from digits sent out 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
189. ding the bootROM and application images 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 e 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
190. e 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 Administrator s Guide SoundPoint IP SoundStation IP 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 four 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 The following sampled audio WAVE file wav formats are supported Note mono 8 kHz G 711 u Law G 711 A Law L16 16000 16 bit 16 kHz sampling rate mono L16 32000 16 bit 32 kHz sampling rate mono L16 48000 16 bit 48 kHz sampling rate mono L16 16000 is not supported on SoundPoint IP 301phones and SoundStation IP 4000 phones L16 32000 and L16 48000 are supported on SoundStation IP 6
191. e Loaded application sip ld successful time is THU FEB 022322082S5 ctg 3 00 New load header information 0223220825 cfg 3 00 Code length 0x0013F22D 022322082S5 cfg 3 00 Header check Sum 0x20147429 022322082S cfg 3 00 Code check Sum Ox0A8COD28 0223220825 cfg 3 00 Options 0 00000003 0223220828 cfg 3 00 New load header information 0223220828 cfg 3 00 Code length 0x0013F16D 0223220828 cfg 3 00 Header check Sum 0x20147369 0223220828 cfg 3 00 Code check Sum 0x0ASDE1 4D 0223220828 cfg 3 00 Options 0x00000003 0223220828 cfg 3 00 Using compatible image 1 0223220834 sig 4 0g 3 0 s aq 5 og 4 og 4 og 6 ag Troubleshooting Your SoundPoint IP SoundStation IP Phones 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 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 0224000058 dns 0224000058 cfg 0224000058 dns
192. e Communications Server 2005 feature For more information refer to Busy POLYCOM Lamp Field on page 4 28 Administrator s Guide SoundPoint IP SoundStation IP 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 e 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 72 Turn the presence and messaging features on or off For more information refer to Feature lt feature gt on page A 92 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 107 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 122 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 123 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
193. e aes hf enable 1 voice aes hf duplexBalance 0 7 voice aes hf duplexBalance 1 7 Configuration Files Attribute Default 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 mp o A AJ oa oj o voice aes hf duplexBalance IP_4000 0 oO voice aes hf duplexBalance IP_4000 1 voice aes hf duplexBalance IP_4000 2 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 6 voice aes hf duplexBalance IP_4000 7 voice aes hf duplexBalance IP_4000 8 pO wo AJ aJ ol NI oO Background Noise Suppression lt ns gt These settings control the performance of the transmit background noise suppression feature y Polycom recommends that you do not change these values POLYCOM Attribute Default voice ns hs enable 1 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 47 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default voice ns hf enable 1 voice ns hf signalAttn 6 voice ns hf silenceAttn 9 voice ns hf IP_4000 enable 1 voice
194. e 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 Troubleshooting Your SoundPoint IP SoundStation IP Phones 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 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 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
195. e 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 Sound PointIPLocalization folder on the boot server Administrator s Guide SoundPoint IP SoundStation IP 4 Adda 1cl ml lang clock menu x attribute to the configuration file Add 1cl ml lang clock x 24HourClock lcl ml lang clock x format lcl ml lang clock x longFormat and 1cl ml lang clock x dateTop attributes and set them according to the regional preferences 6 Optional Set 1cl m1 1lang 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 Extended character support available on SoundPoint IP 600 and SoundStation IP 4000 and 7000 platforms includes the following Unicode character ranges Name Range CJK Symbols and Punctuation U 3000 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 Enclosed CJK Letters and Months U 31A0 U 31BF U 3200 U 327F CJK Compatibility U 3300 U 3
196. e phone max 1 on SoundStation IP 4000 6000 7000 max 2 on IP 301 320 330 430 max 3 on IP 501 max 4 on IP 550 560 max 6 on IP 600 max 48 on IP 601 650 670 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 reg x callsPerLineKey 1 to 34 OR 1 to 24 OR 1to8 24 OR For the SoundPoint IP 650 and 670 the permitted range is 1 to 34 and the default is 34 For the SoundPoint IP 550 560 600 and 601 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 64 Note A call active on another phone ona shared line counts as a call for every phone sharing that registration 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 xX 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
197. e qualityMonitoring collector alert delay threshold critical Null 10 to 2000 Null Threshold value of one way delay 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 Configuration Files This configuration attribute is defined as follows RTCP XR lt rtcpxr gt Permitted Attribute Values Default Interpretation voice qualityMonitoring collector server x Dotted decima Null IP address or host name and port of address 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 0 Null 1 to 5060 If port is O or Null port 5060 will be port 65535 used Set x to 1as only one report collector is supported at this time 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 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 IPTOS41IP gt Ethernet IEEE 802 1p Q lt e
198. e 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 e For more information refer to Local Directory lt local gt on page A 68 Specify whether or not the local contact directory is read only For more information refer to Local Directory lt local gt on page A 68 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 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 the next section Local Contact Dir
199. ectory File Format XML file lt Ethernet This file can be created manually using an XML editor address gt directory For information on file format refer to the next section Local Contact xml Directory File Format 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 Elements can appear in any order lt xml version 1 0 encoding UTF 8 standalone yes gt lt directory gt lt item_list gt lt item gt lt lb gt Mr lt lb gt lt 1n gt Doe lt 1n gt lt fn gt John lt fn gt 4 10 lt ct gt 1001 lt ct gt lt sd gt 1 lt sd gt lt rt gt 1l 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 gt lt lb gt Dr lt lb gt lt ln gt Smith lt 1n gt lt fn gt Bill lt fn gt lt ct gt 1003 lt ct gt Configuring Your System 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
200. ed 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 and or other materials provided with the distribution Administrator s Guide SoundPoint IP SoundStation IP 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 WARR
201. ed 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 and lt IP_7000 gt on page A 80 ind anim x y frame z duration positive integer Duration in milliseconds for this step O infinite Administrator s Guide SoundPoint IP SoundStation IP Patterns lt pattern gt This section defines patterns for the LED indicators In the following table x is the pattern number y is the step in the pattern 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
202. ed VoIP Service Provider g aa Ball Server 18 Call Server 1A U D VoIP SMB Customer Premise SIP Capable Router PSTN Gateway Phone Configuration The phones at the customer site are configured as follows e 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 e Server 2 the fallback server will be 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 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 ser
203. ed decimal IP address OR domain name string The syslog server IP address or host name The default value is NULL 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 0to6 Specifies the lowest class of event that will be rendered to syslog It is based on log render level and can be a lower value Refer to Basic Logging lt level gt lt change gt and lt render gt on page A 86 Note Use left and right arrow keys to change values Prepend MAC Address Enabled Disabled If enabled the phone s MAC address is prepended to the 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 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 platform refer to Supported Provisioning Protocols on page 3 4 Note Note we POLYCOM
204. ed 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 usr pwd
205. ed in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 17 A 116 Configuration Files Permitted Attribute Values Default Interpretation dialplan x apply ToUserDial 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 17 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 17 dialplan x impossibleMatchHandling 0 1or2 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 17 dialplan x removeEndOfDial 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 17 This attribute also includes e Digit Map lt digitmap gt e Routing lt routing gt Digit Map lt digitmap gt For more information on digit map syntax refer to Digit Map lt digitmap gt on
206. ed line will subscribe to a server providing call state information For more information refer to Registration lt reg gt on page A 107 Specify per registration whether diversion should be disabled on shared lines For more information refer to Diversion lt divert gt on page A 114 Local Web Server Specify per registration line type private or shared and third party if enabled 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 Specify per registration line type private or shared and the shared Interface line third party name 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 320 330 430 550 560 600 601 650 and 670 phones However on the SoundPoint IP 320 330 the LED is not lit Depending on your call ser
207. egory the resource will not be downloaded and a predefined default will be used instead For res quotas x value the default is 300 KB for tones 10 KB for bitmaps and fonts and 600KB for backgrounds Note For the SoundPoint IP 550 560 650 and 670 phones this value is internally replaced by 2X the value For the SoundStation IP 6000 and 7000 phones this value is internally replaced by 4X the value Microbrowser lt mb gt This attribute s settings control the home page proxy and size limits to be used by the Microbrowser when it is selected to provide services The Microbrowser is supported on the SoundPoint IP 430 501 550 560 601 650 and 670 and the SoundStation IP 4000 6000 and 7000 phones This configuration attribute is defined as follows 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 mb ssawc enabled 0 1 Null If set to 0 or Null spontaneous display of web content is disabled If set to 1 spontaneous display of web content is enabled mb ssawc call mode Active Passive Null Control the spontaneous display of web content If set to passive or Null the web content is displayed only when requested by the user If set to active the web c
208. em efk efkprompt 1 userfeedback visible efk efkprompt 1 type numeric efk efkprompt 1 digitmatching stylel gt Contact Directory Changes You must make the following contact directory changes for the definition of Call Park lt directory gt lt item_list gt lt item gt lt fn gt Call Park lt fn gt lt ct gt callpark lt ct gt lt sd gt 2 lt sd gt lt rt gt 4 lt rt 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 To avoid users accidently deleting the definitions in the contact directory make the contact directory read only For more information refer to Local Directory lt local gt on page A 68 Using Call Park Key The following figure shows the second speed dial key mapped to Call Park as well as others mapped to Park Return and Call Pickup 41441 fTo Ross Dutkiewicz CallPark 181442 Mon Dec 17 951 AM Hold End Call Trnsfer Confrne Administrator s Guide SoundPoint IP SoundStation IP To use the Call Park key during an active call 1 When there is an active call on line 2233 a Select the Call Park soft key The Call Park screen appears Call Park i 7 Enter Number Next Cancel b Enter the number where you want to park the active call then select the Next soft key The Call Park code 68 is prepended to the number you entered and the
209. en 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 interface 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 For SoundPoint IP 320 330 desktop phones and SoundStation IP 6000 and 7000 conference phones the speed dial index range is 1 to 99 For all other SoundPoint IP and SoundStation IP phones the range is 1 to 9999 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 60 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
210. en 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 Administrator s Guide SoundPoint IP SoundStation IP Provisioning lt prov gt This attribute s settings control aspects of the phone s boot server provisioning system Permitted Attribute Values Default Interpretation prov fileSystem rfsO minFreeSpace 5 512 5 Minimum free space in Kbytes to reserve in the file system when prov fileSystem ffs0 4meg minFreeSpace 420 downloading files from the boot rver 48 serve prov fileSystem ffsO 2meg minFreeSpace Note Polycom recommends that prov fileSystem ffsO 8meg minFreeSpace 512 you do not change these parameters Note For the SoundPoint IP 650 platform prov fileSystem ffs0 8meg m inFreeSpace Is internally replaced by 2X the value Note For the SoundPoint IP 7000 pla
211. ence feature is always enabled on the SoundStation IP 7000 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 This attribute s settings control the maximum size or an external resource retrieved at run time For more information refer to Technical Bulletin 35704 Allocating Adequate Memory for resources on SoundPoint IP and SoundStation IP Phones at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html This attribute also includes e Finder lt finder gt e Quotas lt quotas gt Administrator s Guide SoundPoint IP SoundStation IP Finder lt finder gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation res finder sizeLimit positive 300 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 550 560 650 and 670 phones this value is internally replaced by 2X the value For the SoundStation IP 6000 and 7000 phones this value is internally r
212. entry 0223214054 so OO tink status is Net up Speed 10 half Duplex PC down 0223214055 app1 ye Sang resolver server 172 23 0 200 alternate server 172 23 0 239 and do 0223214123 app1 3 00 OOlTEDHCP returned result 0 3E7 from server 172 23 0 232 00 Phone IP address is 172 23 2 172 00 Subnet mask is 255 255 0 0 00 Gateway address is 172 23 2 240 00 Time server is 172 23 0 235 00 GMT offset is 28800 seconds 00 DNS server is 172 23 0 200 00 Bo DNS alternate server is 172 23 0 239 00 Bo DNS domain is vancouver polycom com 00 Imoye wooeeum nos uov cuouyou 00 Image sip ld has not changed 00 Loaded application sip ld successfully errors Ox0 00 Uploading boot log time is THU FEB 23 21 43 26 2006 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 appl 0223214123 app1 0223214123 app1 0223214123 app1 0223214259 cfg 0223214300 cfg 0223214326 app1 0223214326 app1 O a LI L O9 L9 LA L OI O9 L9 LA LA OA O9 OA EA Eaa ama O Ga ALO O O9 ON OE ee a Boot Failure Messages The following figure shows an example of Application sip ld is not compatible with this phone boot failure messages hwSigParseRemove could not find key Bad image signature Error updating application 0223220834 sig 0223220834 cftg 0223220834 cftg 0223220834 app1 0223220904 appi1 0223220904 appi Application sip ld is not compatibl
213. eplaced by 4X the value res finder minfree 1 to 2048 600 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 up for platforms with more SDRAM If set to 0 or Null the default value of 600 is used Note For the SoundPoint IP 550 560 650 and 670 phones this value is internally replaced by 2X the value For the SoundStation IP 6000 and 7000 phones this value is internally replaced by 4X the value Quotas lt quotas gt This configuration attribute is defined as follows 5 background 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 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 Configuration Files Attribute res quotas x value Permitted Values positive integer Interpretation 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 cat
214. ept 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 6 Configuration changes can performed centrally at the boot server Central boot server Configuration file sip cfg To turn idle display animation on or off e For more information refer to Indicators lt ind gt on page A 80 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 and lt IP_7000 gt on page A 81 To change the position of the idle display animation e For more information refer to Graphic Icons lt gi gt lt IP_300 s lt IP_330 gt lt IP_400 gt lt IP_500 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt on page A 83 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 contain 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
215. equired 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 Third Party Software 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 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 ARISING FROM OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE curl COPYRIGHT AND PERMISSION NOTICE Copyrigh
216. er You agree not to represent otherwise 3 License Ownership Subject to your compliance with this Agreement Polycom hereby grants you a limited license to use the API solely for the purposes of developing and testing your own proprietary software to be used in conjunction with the Product s The foregoing license does not grant you any distribution rights or other rights to use the API for any other purpose and you agree that you shall not rent lease loan sell sublicense assign or otherwise transfer any rights in the API Polycom retains ownership of the API and except as expressly set forth herein no other rights or licenses are granted Polycom may change suspend or discontinue providing the API at any time 4 Term Survival Without prejudice to any other rights Polycom may terminate this Agreement if you fail to comply with any of the terms and conditions of this Agreement In such an event you must destroy all copies of the API You may terminate this Agreement at any time by destroying the API In the event of any termination of this Agreement Sections 1 2 5 and 7 11 shall survive termination 5 Development Nothing in this Agreement shall impair Polycom s right to develop acquire license market promote or distribute products software or technologies that perform the same or similar functions as or otherwise compete with any other products software or technologies that you may develop produce market or distribute In
217. er 18 se pat ringer 18 Ringer 19 se pat ringer 19 Ringer 20 se pat ringer 20 Ringer 17 se pat ringer 17 Ringer 21 se pat ringer 21 Ringer 22 se pat ringer 22 Not used In SIP 3 1 the SoundPoint IP welcome sound was removed from saf 1 If you want the welcome sound to be played when a phone reboots or restarts set saf 1 to SoundPointIPWelcome wav Sound Effects lt se gt The phone uses both synthesized based on the chord sets refer to Chord Sets lt chord gt on page A 29 and sampled audio sound effects Sound effects are defined by patterns rudimentary sequences of chord sets silence periods and wave files Administrator s Guide SoundPoint IP SoundStation IP 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 indicate 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
218. er 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 119 Local Web Server For one touch voice mail access enable the one touch voice mail if enabled user preference and bypass instant messages to remove the step of selecting between instant messages and voice mail after pressing the Messages key on the SoundPoint IP 430 500 501 550 560 600 601 650 and 670 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 Configuring Your System 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 suppor
219. ernet port can be disabled on the SoundPoint IP 330 430 450 550 560 601 650 and 670 and 850 The port cannot be disabled on the SoundPoint IP 301 501 and 600 due to a lack of hardware support for the feature To disable the Ethernet port on a supported SoundPoint IP phone 1 2 So E Press Select Settings gt Advanced gt Network Configuration gt Ethernet Menu You must enter the administrator password to access the network configuration The factory default password is 456 Scroll down to PC Port Mode and select Edit Select Disabled and then press the OK soft key Press the Exit soft key Select Save Config The SoundPoint IP phone reboots When the reboot is complete the PC Ethernet port is disabled Administrator s Guide SoundPoint IP SoundStation IP 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 advertising or publicity pertaining to distribution of the software without specifi
220. erred value is auto meaning that the configuration of the peer role is automatic slaveOnly The other value definitions are slavePreferred e standAlone IP 7000 is always only standalone masterOnly IP 7000 is always the master e masterPreferred The configuration is automatic but if the call capability of the daisy chained IP 7000 is the same as this one this one is the master slaveOnly IP 7000 is always the slave e slavePreferred The configuration is automatic but if the call capability of the daisy chained IP 7000 is the same as this one this one is the slave DNS Cache lt dns gt In the tables below a maximum of 12 entries of NAPTR SRV and A record can be added This attribute includes e NAPTR lt NAPTR gt attribute SRV lt SRV gt A lt A gt A 100 NAPTR lt NAPTR gt Configuration Files This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dns cache NAPTR x domain name string Null The domain name to which this resource record name refers dns cache NAPTR x ttl 300 to 65535 300 Specifies the time interval in seconds that the resource record may be cached before the source of the information should again be consulted dns cache NAPTR x 0 to 65535 0 A 16 bit unsigned integer specifying the order in order which the NAPTR records mu
221. es gt rire eese tepsi ene IE VEE E ta nee A 72 Fonts lt font gt 2 0 0 ccc cnet a E pE nine ni A 72 Keys Shey gt crre irena E aA E E EER EE EERE EERS A 75 Backgrounds lt bg gt res bs ccdeie sek eee bandh RAEES A 77 Bitmaps lt bitmap gt 0 ec eee eens A 80 Indicators sind gt 6 amp 0 ccc Maidan ee tard be tabs Wad de ed A 80 Event Logging lt log gt 2 0 2 0 0 eee eee eee eee eee A 84 Security lt SeC gt tc purih nnd Deas as eed et A 88 License lt license gt 2sccscii hckates a AnA ee eee hehe ee heals A 89 Provisioning lt prov gt 2 0 0 6 cece cece eee eens A 90 RAM Disk lt ramdisk gt 0c cee eee eee eee A 90 Request lt request gt sos tsere tasie t eae aE ee tenes bol betes A 91 Feature lt feature gt u un nunun nnne rere A 92 Resource PES gt sarr e eee OAE E EEEE Re A 93 Microbrowser lt mb gt 0 cee cece cette eee e eee A 95 Applications lt apps gt 00 ce cece eee A 98 Peer Networking lt pnet gt 0 00 eee eee A 100 DNS Cache lt dns gt 0 cece cee nce e eens A 100 Soft Keys lt softkey gt 0 0 00 A 103 Per Phone Configuration 0 00 eee eee eee ee A 106 Registration lt f66 gt ii sccsacewiaie basing tiasseetussiacas A 107 Calls lt call gt oc tec nk a a eek peda A 111 Diversion lt divert gt 0 cc cece cece eens A 114 Dial Plan lt dialplan gt 0 0 eee A 116 Me
222. es 4 17 Call Transfer oye nE E EA bes E E ee eee ere ee dates Oe 4 18 Local Centralized Conferencing 0 00000008 4 19 Call Forward ets etia th cotta ieii a hier a Shope a Hades ne Glo 4 20 Directed Call Pick Up 0 ccc cece cece ce eee eee eens 4 21 Group Call Pick2U pts wess eed i Bed Me RS Me 4 22 Call Park Retrieve 2 0 0 0 E e A ccc etn eas 4 22 Last Call Return sencan ani ie te ea et eee a als dae Maes 4 22 Setting Up Advanced Features 0 0000 e eee eee 4 22 Configurable Feature Keys 0 00 c cece eee eee 4 24 Multiple Line Keys per Registration 00 4 25 Multiple Call Appearances 0 cece eee 4 25 Shared Call Appearances cece eee 4 26 Bridged Line Appearance 0 0000 c eee eee eee eee 4 27 Busy Lamp Field oseni a cok gus ste eet ecb wesc vaca bowen 4 28 Customizable Fonts and Indicators 0000005 4 29 Contents Instant Messaging susastra Aa Beet Dhyana ede 4 30 Multilingual User Interface 0 0 e eee eee 4 30 Downloadable Fonts 0 0 0 e eee 4 31 Synthesized Call Progress Tones 0 00000 e eee ee eee 4 32 MicrobroWSer Sis oriens iana Stk o Sioa 4s ie een a a ee 4 32 Real Time Transport Protocol Ports 000 4 33 Network Address Translation 0 0 00 000s eee 4 34 Corporate Directory 2 334 ls BE i hg ein de kee 4 34 Recordin
223. es A 27 transmit equalization A 50 user preferences A 25 voice activity detection A 52 voice coding algorithms voice coding algorithms lt codecs gt A 38 voice settings A 37 volume persistence A 42 web server A 63 application configuration file A 4 application error messages 5 3 application files overview 2 6 applications 4 33 Applications key 4 32 attendant lt attendant gt A 121 attended transfers 4 18 audio codecs 4 76 audio playback 4 37 A 92 audio recording 4 37 A 92 automatic call distribution 4 56 automatic gain control 4 78 automatic gain control lt agc gt A 48 automatic off hook call placement 4 17 automatic off hook call placement lt autoOffHook gt A 112 B background logo adding C 6 configuration file changes C 7 background noise suppression 4 77 background noise suppression lt ns gt A 47 backgrounds lt bg gt A 77 basic logging A 86 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 Index 2 basic TCP IP A 58 blind transfers 4 18 BNS See also background noise suppression boot failure messages 5 8 boot server security policy 3 14 boot servers deploying phones 3 15 redundant 3 12 security policy 3 14 setting up 3 13 bootROM 2 3 bootROM and application wrapper 2 5 bootROM error messages 5 2 bootROM tasks 2 3 bootROM SIP application dependencies C 9 bridged line appearance signaling B 1
224. esented to SIP application in order for example dns cache A 1 dns cache A 2 and so on Example 2 This example shows how to configure static DNS cache where your DNS provides A records for server X address but not SRV In this case the static DNS cache on the phone provides SRV records For more information go to http tools ietf org html rfc3263 When the static DNS cache is not used the sip cfg configuration would look as follows reg 1 address 1002 sipserver example com reg 1 server 1 address primary sipserver example com reg 1 server 1 port 5075 reg l server 1 transport UDPOnly reg 1 server 2 address secondary sipserver example com reg 1 server 2 port 5075 reg l server 2 transport UDPOnly When the static DNS cache is used the sip cfg configuration would look as follows reg 1 address 1002 reg 1 server 1 address sipserver example com reg l server 1 port reg 1 server 1 transport UDPOnly Note reg reg reg dns dns dns dns dns dns dns dns dns dns dns dns Configuring Your System l server 2 address l server 2 port l server 2 transport cache SRV 1 name _sip _udp sipserver example com cache SRV 1 ttl 3600 cache SRV 1 priority 1 cache SRV 1 weight 1 cache SRV 1 port 5075 cache SRV 1 target primary sipserver example com cache SRV 2 name _sip _udp sipserver example com cache SRV 2 ttl 3600 cache SRV 2 priority 2 cache SRV
225. ew 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 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 loadin
226. ew call scenarios Note A hot dial 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 call singleKeyPressConference 0 1 Default 0 Interpretation 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 O or Null sound effects are only heard by conference initiator old behavior Only supported for SoundPoint IP 550 560 650 and 670 and SoundStation IP 7000 For all others set to 0 call localConferenceCallHold 0 1 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 and 670 and SoundStation IP 7000 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 0 or Null the Normal soft key appears Note This parameter is supported on the SoundPoint IP 330 320 only call directedCallPickupString star code
227. ew release You must rename the sip ld sip cfg and phonel1 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 Note Setting up Your System To upgrade your SIP application 1 Do one of the following steps a Place the bootrom d 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 Copy sip ld sip cfg 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 Rename sip ld sip cfg and phonel cfg from the previous distribution to sip_212 1d 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 Modify the 000000000000 cf 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 CONFIG_FILES PHONE
228. f servers and their associated parameters will override the servers specified in sip cfg if non Null For more information refer to Registration lt reg gt on page A 107 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 maximum is six even for the IP 601 650 and 670 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 Administrator s Guide SoundPoint IP SoundStation IP Local Local Phone User Use the SIP Configuration menu to
229. f the codec negotiation answer If set to 0 or Null RFC 3264 is followed for transmit and receive RTP payload type values volpProt SDP answer useLocalPreferences Oor1 If set to 1 the phones uses its own 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 hold 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 vol
230. fault 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 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 or 6416 polycom com Null Forward to contact used for calls forwarded due to no answer if Null divert x contact will be used A 115 Administra
231. figuration order dictates how the attributes are displayed and sorted The first attribute is the primary sort 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 Technical 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 Configuring Your System Configuration File Example The
232. file ffs0 local local dir 0223160118 ssps 00 Initial log entry Current logging level 4 0223160118 net 00 Initial log entry Current logging level 4 0223160118 httpd 00 Initial log entry Current logging level 4 0223160118 key OO0 Initial log entry Current logging level 4 0223160118 ssps 00 Application comp 1 Label PolyDSP Orion Mem2 FS1 Version 1 3 3 0010 0 0223160118 ssps O00 Application comp 1 P N 3150 11580 133 0223160118 pps 00 Initial log entry Current logging level 4 Do231e0118 ceo ResFinderC Failed to download file SoundPointJP elcone vav errno Oxdl SoToneC Failed to find tone SoundPoint PWelcone vay using default 0223160121 so VY LASSE LHUGLUJ POLLGU VU UUWHLUOU Lado JUUHULULHVILWOLUUMG WOV Siu VAU 0223160121 slog 0223160121 res 4100 SoToneC Failed to find tone SoundPointIPWelcome wav using default 4 00lRecRegistration failed User 2125551212 Error Code 404 Not Fo 00 UtilCopyC curl error buffer Failed writing body 00 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 oo Uti Caner Tia f ferenaoaa feundl Wedsatinew file 76 fel lein nfa con siecrConfiguration phonel cfg sip cfg Res W 4 fi a E o S maom ee a 0223160121 so 0223160153 sip Testing Phone Hardware To obtain more detaile
233. fo HCOLON alert param COMMA alert param alert param LAQUOT absoluteURI RAQUOT SEMI generic param The web content must be located with an absolute URI which begins with the scheme identifier Currently only the HTTP scheme is supported So an example header might look like Access URL lt http server polycom com content23456 xhtml gt This header may be placed in SIP requests and responses as appropriate so long as the messages are part of an INVITE initiated dialog and the phone can associate them with an existing phone call This feature also requires the definition of two optional parameters Administrator s Guide SoundPoint IP SoundStation IP Static DNS Cache 4 68 e An expires parameter is defined to indicate the lifespan of the URL itself or assuming that the URL is permanent the time span for which the content is expected to have relevance to the call with which it is associated If the parameter is absent or invalid this will be interpreted to mean that the content or the URL itself will be persistent in nature A value if it is present will indicate the lifespan of the content in seconds zero has special significance see example below When the lifespan expires the phone will remove both the indication of the URL and the ability of the user to retrieve it For example Access URL lt http server polycom com content23456 xhtml1 gt expires 60 If the server wishes to invalidate a previ
234. 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 389 dir corp transport TCP dir corp baseDN cn Users dc yourcompany dc local dir corp user ldapadmin dir corp password 12345678 dir corp filterPrefix objectclass person dir corp scope sub 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 dir corp attribute 1 sticky 0 dir corp attribute 2 name givenName dir corp attribute 2 label First Name dir corp attribute 2 type first_name dir corp attribute 2 filter dir corp attribute 2 sticky 0 dir corp attribute 3 name telephoneNumber 3 label Phone Number 3 type phone_number 3 filter 604 dir corp attribute 3 sticky 0 dir corp attribute dir corp attribute dir corp attribute dir corp backGroundSync 0 dir corp backGroundSync period 86400 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 allow recording of audio calls
235. g 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 3 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 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 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 Administrator s Guide SoundPoint IP SoundStation IP Warning 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
236. g and Playback of Audio Calls 00005 4 37 Daisy Chaining Phones 0000 0 e eee eee 4 38 Provisioning Phones Over CLink 06600 c cee eee eens 4 39 Enhanced Feature Keys 0 00 e eee eee eee eee eee 4 40 Configurable Soft Keys eee eee ee 4 50 Voice Mail Integration 0 0 00 eee eee eee eee 4 54 Multiple Registrations 00 02 4 55 Automatic Call Distribution 0 0 c cece eee eee 4 56 Server Redundancy 2 0 disci ieee ena as 4 56 PReSENCEs 23 01 nee abate tnd eee dite Peas Ta eee tie gee 4 60 Microsoft Live Communications Server 2005 Integration 4 61 Access URL in SIP Message 0 0 nren rrna 4 65 Static DNS Cache ae ve saes owns ieee eRe we EE 4 68 Display of Warnings from SIP Headers 00 4 72 Setting Up Audio Features 0000s 4 73 Low Delay Audio Packet Transmission 000 4 74 Jitter Buffer and Packet Error Concealment 005 4 74 Voice Activity Detection oei yos gA e ana ccc eee 4 74 DTMF Tone Generation 0 0 cee eens 4 75 DTMF Event RTP Payload nneur rrer nrn 4 75 Acoustic Echo Cancellation 0 0 0 0 e eee eee eee 4 75 A di Codecs iii x dene sede s eee ideas onde aden Bou tees 4 76 Background Noise Suppression 0 00 000000 4 77 Comfort Noise Full sss ceases bab eyes ae as ig oes 4 77 Automatic Gain Control
237. g 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 versions 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 t
238. gain tx digital handset IP_650 6 voice gain tx digital headset 0 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 digital chassis IP_6000 6 voice gain tx digital chassis IP_7000 6 voice gain tx analog preamp handset 23 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 Configuration Files Attribute Default 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 headset txag adjust IP_430 39 voice headset txag adjust IP_650 21 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
239. gging 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 return all logging levels to the default value of 4 Administrator s Guide SoundPoint IP SoundStation IP 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 uploa
240. gister 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 retry TimeOut 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 0 or Null 3 is used retryMaxCount non negativ retries will be attempted before moving on to e integer the next available server volpProt 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 Attribute Permitted Values Default Interpretation volpProt SDP useLegacyPayloadNegotiation 0 1 Null If set to 1 the phone transmits and receives RTP using the payload type identified by the first codec listed in the SDP o
241. guration Files 6 6 c cece eee eee C 4 Changing the Key on the Phone ranns eee cece C 5 Adding a Background Logo 6 c cece eens C 6 BootROM SIP Application Dependencies 000 C 9 Migration Dependencies rnnr rnrn eee eee ee C 9 Multiple Key Combinations 0 000 e eee eee eee eee C 10 Default Feature Key Layouts 0 00 e eee eee eee ee C 12 Internal Key Functions 0 00 0 e eee eee eee C 19 Assigning a VLAN ID Using DACP i445 0 0i spn rnnr rnn C 23 Parsing Vendor ID Information 000 e eee eee C 24 Product Model and Part Number Mapping C 26 Disabling PC Ethernet Port 0 0c eee C 27 D Third Party Software ce cece cece cece ee DI Index 6c ck sc deb eeieveseadeebvesccvess seas ss Index 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 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
242. guration file Adjust receive and handset headset volume sip cfg For more information refer to Volume Persistence lt volume gt on page A 42 Customizable Audio Sound Effects Audio sound effects used for incoming call alerting and other indications are customizable Sound effects can be composed of patterns of synthesized tones or sample audio files The default sample audio files may be replaced with alternates in wav file format Supported wav formats include e mono G 711 13 bit dynamic range 8 khz sample rate e mono L16 16000 16 bit dynamic range 16 kHz sample rate e mono L16 32000 16 bit dynamic range 32 kHz sample rate e mono L16 48000 16 bit dynamic range 48 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 L16 32000 and L16 48000 are only supported on SoundPoint IP 7000 phones Note The alternate sampled audio sound effect files must be present on the boot server or the Internet for downloading at boot time Configuration 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 30 or Sound Effects lt se gt on page A 31 Local
243. he 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 enhanced feature keys is the enhanced feature keys feature corporate directory is the corporate directory feature 92 Configuration Files 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 and SoundStation IP 7000 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 number 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 The manage confer
244. he SRV 1 priority 1 dns cache SRV 1 weight 1 dns cache SRV 1 port 5075 dns cache SRV 1 target primary sipserver example com dns cache SRV 2 name _sip _udp sipserver example com dns cache SRV 2 ttl 3600 dns cache SRV 2 priority 2 dns cache SRV 2 weight 1 dns cache SRV 2 port 5075 dns cache SRV 2 target secondary sipserver example com dns cache A 1 name primary sipserver example com dns cache A 1 ttl 3600 dns cache A 1 address 172 23 0 140 dns cache A 2 name secondary sipserver example com dns cache A 2 ttl 3600 dns cache A 2 address 172 23 0 150 Note The reg 1 server 1 port reg 1 server 2 port reg 1 server 1 transport and reg 1 server 2 transport values in this example are set to null to force NAPTR lookups Display of Warnings from SIP Headers The Warning Field from a SIP header may be used to cause the phone to display a three second pop up to the user For example this feature can be used to inform the user of information such as the reason that a call transfer action failed bad extension entered for example For more information refer to Header Support on page B 4 These messages are displayed in any language supported by the phone for three seconds unless overidden by another message or action Configuring Your System For example if a user parks a call the following message appears on their phone Park success 801 ny ee voces Th New Call Forward x Sa Conf
245. he 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 107 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 locally if configured in that way Contact information from previous calls may be easily added to the directory for convenient future access The directory is the central database for several other features including speed dial distinctive incoming call treatment presence and instant messaging Administrator s Guide SoundPoint IP SoundStation IP Note If a user makes a chang
246. he 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 boot 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
247. hey are on a call for example a SIP re INVITE The information accessible at the URL can be anything that you want to have displayed 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 Microbrowser lt mb gt on page A 95 This section provides detailed information on e Web Content Examples e User Interface e Signaling Changes Web Content Examples This feature can be used in the following circumstances e Call Center Customer information The URL provided allows the phone to access information about a customer and display it before the agent takes the call e Call Center Scripts for different call center groups The phone can access a script of questions for an agent to ask a caller when a call comes in The script can be different for each agent group e Restaurant menu on a hotel phone A guest dials a number for the restaurant and a voice indicates that the menu is now available for viewing on the phone Administrator s Guide SoundPoint IP SoundStation IP User Interface There are three user interface aspects to this feature e Web content status indication e Web content retrieval spontaneous and on demand e Settings menu item to control active versus passive behaviour Web Content Status Indication When valid web content validity is determined thro
248. hone is disabled up numberFirst CID 0 1 0 If set to O or Null caller ID display will show caller s name first If set to 1 caller ID display will show caller s number first 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 Administrator s Guide SoundPoint IP SoundStation IP Dual Tone Multi Frequency lt DTMF gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation 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 tone dtmf off Time positive integer positive integer 50 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 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 dial
249. hone 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 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 Note Note 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 HTTPS HTTP and FTP are supported for redundant boot servers For all SoundPoint IP and SoundStation IP phones follow the normal provisioning process in the next section Provisioning Phones However if you have decided to daisy
250. ide the servers TCPpreferred or tr E specified in sip cfg in Server lt server gt on UDPOnly or A 7 TLS or page 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 Note TLS is not supported on SoundPoint IP reg x server y retry TimeOut Null or l 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 z registration the ACD feature will be enabled reg x acd agent available 0 1 0 for that registration reg x ring Type 1 to 22 2 The ringer to be used for calls received by this registration Default is the first non silent ringer A 108 Configuration Files Attribute reg x lineKeys Permitted Values 1 to max Default 1 Interpretation max the number of line keys on th
251. ies 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 boot server Configuration file Specify the conference hold behavior all parties on hold or only host sip cfg is on hold e For more information refer to Call Handling Configuration lt call gt on page A 64 Specify whether or not all parties hear sound effects while setting up a conference e For more information refer to Call Handling Configuration lt call gt on page A 64 Specify which type of conference to establish and the address of the centralized conference resource e For more information refer to Conference Setup lt conference gt on page A 16 Manage Conferences Note This feature is supported on the SoundPoint IP 550 560 650 and 670 desktop phones and the SoundStation IP 7000 conference phone This feature requires a license key for activation on all phones except the SoundStation IP 7000 Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller Administrator s Guide SoundPoint IP SoundStation IP The individual parties within a conference can be managed New parties can be added and information about the conference participants can be viewed for example names phone numbe
252. iguration changes can be performed centrally at the boot server Central boot server Configuration file Turn this feature on or off and specify which warnings are sip cfg displayable e For more information refer to SIP lt SIP gt on page A 10 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 e Low Delay Audio Packet Transmission e Jitter Buffer and Packet Error Concealment e Voice Activity Detection DTMF Tone Generation e DTMF Event RTP Payload e Acoustic Echo Cancellation e Audio Codecs e Background Noise Suppression e Comfort Noise Fill e Automatic Gain Control IP Type of Service e TIEFE 802 1p Q Voice Quality Monitoring 4 73 Administrator s Guide SoundPoint IP SoundStation IP e Dynamic Noise Reduction e Treble Bass Controls Low Delay Audio Packet Transmission The phone is designed to minimize latency for audio packet transmission There are no related configuration changes Jitter Buffer and Packet Error Concealment The phone employs a high performance jitter buffer and packet error concealment system designed to mitigate packet inter arrival jitter and out of order or lost lost or excessively delayed by the network packets The jitter buffer is ad
253. ine Keys per Registration More than one line key can be allocated to a single Multiple Registrations SoundPoint IP desktop phones support multiple registrations per phone However SoundStation IP conference phones support 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 This feature is part of the Productivity Suite 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 logica
254. ing elements are part of the definition language e lt efk gt e lt efklist gt e lt efkprompt gt e lt version gt e Special Characters Note Configuring Your System lt efk gt This element indicates the start of enhanced feature key definition section The efk element has the following format lt efk gt lt efk gt lt efklist gt This element describes behavior of enhanced feature key The different blocks of the enhanced feature key definitions are uniquely identified by number following efk efklist prefix for example efk efklist 1 lt suffix gt In SIP 3 1 a maximum of 50 element groups is supported however the exact number is dependent on available RAM and processing speed The disabled elements are included in the total count This element contains the following parameters Name Interpretation mname This is the unique identifier that is used for the speed dial configuration to reference the enhanced feature key entry It cannot start with a digit This parameter must have a value and it cannot be Null status This parameter has the following values e If set to 1 this key is enabled e If set to 0 or Null this key is disabled If this parameter is omitted the value 0 is used label This field defines the text string that will be used as a label on any user text entry screens during enhanced feature key operation The value can be any string including the null st
255. ing 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 viaRip is disabled tone dtmf stim pac offHookOnly 0 1 Not currently used Configuration Files Attribute tone dtmf viaRtp Permitted Values Default Interpretation 0 1 1 If set to 1 encode DTMF in the active RTP stream otherwise DTMF may be encoded within the signaling protocol only when the protocol offers the option Note tone dtmf chassis masking should be enabled when tone adtmf viaRtp is disabled tone dtmf rfe2833Control 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 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 cadenc
256. ing that even though one party has 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 offering TimeOut 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 115 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 64 Configuration Files Attribute call dialtone TimeOut Permitted Values Null positive integer Default 60 Interpretation 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 I
257. int IP SoundStation IP 2 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 This 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 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 670 209 109 12 bit color IP
258. int IP SoundStation IP 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 DNSnaptr or DNSnapt If setto Null or DNSnaptr transport 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 volpProt SIP outboundProxy address S 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 Alert Information lt alertInfo gt Configuration Files This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP alertInfo x value string to Null Alert Info fields from INVITE requests
259. ion as follows softkey feature newcall 0 2 Reboot the phone The New Call soft key is not displayed and the space where it usually appears is empty Note Configuring Your System To map a chained list of actions to a soft key 1 Configure speed dial index 2 in contact directory with a regular phone number For example enter 2900 in the contact field 2 Configure speed dial index 1 in contact directory with 2 in contact field 3 Update the sip cfg configuration as follows softkey 1 label ChainAct softkey 1l action S1 Tinvite softkey 1l use idle 1 4 Reboot the phone soft key chainact if you press phone dials number 2900 To map the Do Not Disturb Enhanced Feature Key sequence to a soft key 1 Update sip cfg as follows softkey 1 label DND softkey 1l action FDoNotDisturb softkey 1l use idle 1 2 Reboot the phone A DND soft key is displayed on the phone when it is in the idle state When the DND soft key is pressed the Do Not Disturb icon is displayed When another party calls the phone does not ring and the other party hears a busy dial tone To map a Send to Voice Mail Enhanced Feature Key sequence to a soft key The exact star code to transfer the active call to Voice Mail will differ depending on your call server 1 Update sip cfg as follows softkey 2 label ToVMail softkey 2 action 55 P1N10 Tinvite softkey 2 use alerting 1 2 Reboot the phone
260. is configuration attribute is defined as follows where x 1 to maximum number of defined soft keys Attribute Permitted Values Default Interpretation softkey x label string Null This is the text displayed with the soft key If set to Null the label to display is determined as follows e Ifthe soft key is mapped to a enhanced feature key macro the label of the enhanced feature key macro will be used Ifthe soft key is mapped to a speed dial the label of the corresponding directory entry will be used If this label does not exist as well and the directory entry is a enhanced feature key macro then the label of the enhanced feature key macro will be used e If the soft key is mapped to chained actions only the first one is considered for label using the rules above e f no labels are found after the above steps the soft key label will be blank softkey x action string Null The same syntax as the enhanced feature key action For more information refer to Macro Definition on page 4 44 softkey x enable 0 default 1 Null If set to 0 or Null the soft key is disabled If set to 1 the soft key is enabled softkey x precede 0 default 1 Null If set to 0 or Null the soft key replaces any empty space from the leftmost position If set to 1 the soft key is displayed before the first standard soft key softkey x use idle 0 default 1 Null
261. ist 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 enhanced feature keys 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 available 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 t
262. istinctive 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 headset not supplied The SoundPoint IP 320 330 430 500 501 550 560 600 601 and 650 and 670 phones and SoundStation IP 4000 6000 and 7000 phones 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 The advanced aspects of conferencing are part of the Productivity Suite Local Contact Directory The phone maintains a
263. istration 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_buddies 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 122 Locate the roaming_privacy attribute Configuring Your System 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 123 m Save the modified phonel cfg configuration file Access URL in SIP Message Introduced in SIP 2 2 this feature that allows information contained in incoming SIP signaling to refer to XHTML web content that can be rendered by the SoundPoint IP phone s Microbrowser Supporting this feature allows use of the SoundPoint IP phone s display to provide information before someone takes a call and while t
264. itoring collector period 5 to 20 20 The time interval between successive periodic quality reports Administrator s Guide SoundPoint IP SoundStation IP 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 voice qualityMonitoring collector alert delay threshold warning Null 10 to 2000 Null Threshold value of one way delay 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 voic
265. iversion 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 e For more information refer to Diversion lt divert gt on page A 114 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 removed 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 si
266. l 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 107 could be used for fail over configuration The older 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 usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuration changes can performed centrally at the boot server Central boot server sip cfg Configuration file Specify global primary and fallback server configuration parameters For more information refer to Protocol lt volpProt gt on page A 6 Configuration file Specify per registration primary and fallback server configuration phone1 cfg parameters
267. ld lt hold gt A 67 I idle display lt idleDisplay gt A 96 idle display animation 4 15 incoming signaling validation 4 82 indicator classes lt class gt A 82 indicators A 80 assignments A 82 installing SIP application 3 14 instant messaging 4 30 IP TOS A 56 IP TOS call control lt callControl gt A 58 IP_400 font A 74 Index 3 Administrator s Guide SoundPoint IP SoundStation IP IP_500 font A 75 IP_600 font A 75 J jitter buffer 4 74 K keep alive lt keepalive gt A 63 key features 1 6 keys lt key gt A 75 L language support 1 7 languages adding new A 23 languages supported 4 31 last call return 4 22 LEDs A 83 length lt length gt A 89 local centralized conferencing 4 19 local lt local gt A 6 local contact directory 4 9 local contact directory file format 4 10 local digit map 4 12 local reminder lt localReminder gt A 67 local user and administrator privilege levels 4 81 localization lt Icl gt A 21 log files 5 5 logging lt log gt A 84 low delay audio packet transmission 4 74 M MAC address definition A 2 substitution 3 15 3 21 A 3 main browser lt main gt A 97 main menu 3 6 manage conferences 4 19 manual configuration overview 2 7 manual log upload 5 7 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 12
268. ld 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 430 501 550 560 600 601 650 670 and 7000 Up Down Left and Right arrow keys e IP 4000 and 6000 Menu Exit Off hook Hands free Redial Administrator s Guide SoundPoint IP SoundStation IP Default Feature Key Layouts The following figures and tables show the default SIP key layouts for the Sound Point IP 301 320 330 430 501 550 560 600 601 650 and 670 and SoundStation IP 4000 6000 and 7000 models SoundPoint IP 301 nm 2 CA 3 CY Menu 32 De Mrt Dstt Redial 7 Hold 5 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 MicMute 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 12 SoundPoint IP 320 330 Miscellaneous Administrative Tasks gt D D
269. le 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 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 Administrator s Guide SoundPoint IP SoundStation IP POLYCOM 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 log files to be uploaded and changes made by the p
270. lication 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 configuration 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 subsequent 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 e 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
271. licensed to Polycom in any way except as otherwise expressly authorized by Polycom in each instance In no event shall you i expressly state or imply that any products developed by you were created by or on behalf of Polycom or are being marketed by or on behalf of Polycom or ii expressly state or imply that Polycom has reviewed sanctioned or endorsed your product in any way 8 No Warranty You understand the API provided to you is supplied AS IS AND WITH ALL FAULTS WITHOUT ANY WARRANTY OF ANY KIND WHETHER EXPRESS OR IMPLIED INCLUDING WITHOUT LIMITATION THE IMPLIED WARRANTIES OF MERCHANTIBILITY NON INFRINGEMENT ACCURACY COMPLETENESS PERFORMANCE AND FITNESS FOR A PARTICULAR PURPOSE AND POLYCOM PROVIDES NO SUPPORT FOR THIS API You understand that Polycom is under no obligation to provide updates enhancements or corrections or to notify you of any API changes that Polycom may make In the event you market a product you develop using the API any obligations representations or warranties provided by you to an end user shall be solely your obligations and in no event shall Polycom be responsible to fulfill any such obligations 9 Indemnity You shall indemnify and hold Polycom harmless from and against any and all costs damages losses liability or expenses including reasonable attorneys fees arising from your use of the API including without limitation any actions arising from acts or omissions of your employees or agents
272. ling 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 A 107 Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation 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 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 line seize 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 Velo fault tolerance If specified these servers may overr
273. lly at the boot server Central boot server XML file lt Ethernet address gt directory xml The lt bw gt 0 lt bw gt buddy watching and lt bb gt 0 lt bb gt buddy blocking elements in the lt Ethernet address gt directory xml file dictate the Presence aspects of directory entries e For more information refer to Local Contact Directory on page 4 9 Local Local Phone User Interface The user can edit the directory contents The Watch Buddy and Block Buddy fields control the buddy behavior of contacts 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 Microsoft Live Communications Server 2005 Integration 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 62 Note Any contacts added through the SoundPoint IP phone s buddy list will appear in as a contact in Microsoft Office Communicator and Windows Messenger Polycom recommends that the BLF not be used in conjunction with the Microsoft wy Liv
274. lly related to each other Requires call server support Static DNS Cache Set up a static DNS cache and provide for negative caching 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 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 Administrator s Guide SoundPoint IP SoundStation IP Automatic Gain Control Designed for hands free operation boosts the transmit gain of the local user in certain circumstances 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 IE
275. log ringer IP_7000 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_6000 6 voice gain rx digital chassis IP_7000 6 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 digital ringer IP_6000 21 voice gain rx digital ringer IP_7000 21 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 analog chassis IP_330 36 43 Administrator s Guide SoundPoint IP SoundStation IP Attribute Default 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 analog chassis IP_6000 0 voice gain tx analog chassis IP_7000 0 voice gain tx digital handset 0 voice gain tx digital handset IP_330 10 voice gain tx digital handset IP_430 10 voice
276. lt Ethernet address gt cfg files so that it points to the log file directory e Edit the CONTACT_DIRECTORY attribute of the lt Ethernet address gt cfg files so that it points to the organization s contact directory 4 Reboot the phones by pressing the reboot multiple key combination For more information refer to Multiple Key Combinations on page C 10 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 24 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 Administrator s Guide Sound
277. m 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 107 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 The phone is compatible with the Presence and Instant Messaging features of Microsoft Windows Messenger 5 1 In a future release support for the Presence and Instant Message recommendations in the SIP Instant Messaging and Presence Leveraging Extensions SIMPLE proposals will be provided by the following Internet drafts or their successors e draft ietf simple cpim mapping 01 e draft ietf simple presence 07 e draft ietf simple presencelist package 00 e draft ietf simple winfo format 02 e draft ietf simple winfo package 02 Shared Call Appearance Signaling A shared line is an address 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 NOT
278. methods Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation volpProt SIP requestValidation x A valid string Null Determines which events specified with the request 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 A valid string Polycom Determines string used for Realm digest realm SPIP Special Events lt specialEvent gt This configuration attribute is defined as follows checkSync alwaysReboot Permitted Attribute Values Default Interpretation volpProt SIP specialEvent lineSeize 0 1 1 If set to 1 process a 200 OK response for a nonStandard 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 0 1 0 If set to 1 always reboot when a NOTIFY 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 Conference Setup lt conference gt This configuration attribute is defined as follows Permitted Attribute
279. n 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 200 200 When the volatile storage option is set refer to see dir local volatile 4meg and dir local volatile 8meg this attribute is the maximum size of contact directory file that the phone supports Note that phones with 16 MB RAM support up to 50 Kbytes of directory file and phones with more than 16 MB RAM support up to 200 Kbytes of directory file When the value specified for this attribute exceeds the limit the limit will be used as the max directory size dir local volatile 8meg 0 1 0 Attribute applies only to platforms with 8 Mbytes or more of flash memory If set to 1 use volatile storage for phone resident copy of the directory to allow for larger size A 68 Configuration Files Attribute dir local nonVolatile maxSize 8meg Permitted Values 1 to 100 Default 100 Interpretation Attribute applies only to platforms with 8 Mbytes or more of flash memory This is the maximum size of non volatile storage that the directory will be permitted to consume
280. n existing font use these lt fontName gt _ lt fontHeightInPixels gt SoundPoint IP 320 330 430 500 and 501 fontProp_10 fontPropSoftkey_10 This is the font used widely in the current implementation This is the soft key specific font SoundPoint IP 550 560 600 601 650 and 670 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 time on the SoundPoint IP 650 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 to overwrite the font used for SoundPoint IP 500 soft keys for ASCII the name should be fontPropSoftkey_10_U0000_U00FF fnt to add 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 r
281. n on e Setting Up the Network Setting Up the Boot Server e Deploying Phones From the Boot Server e 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 SoundStation 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 available but is not intended to be used for long term operation of the phones However if you want to register a single SoundPoint IP SoundStation IP phone refer to Quick Tip 44011 Registering Standalone SoundPoint IP and SoundStation IP Phones at http www polycom com usa en support
282. n per registration boot server phonet cfg e For more information refer to Registration lt reg gt on page A 107 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 it is assumed that all registrations use the same server 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 sip cfg Specify
283. ncoming network signaling are e Source IP address validation e Digest authentication e Source IP address validation and digest authentication Configuration changes can performed centrally at the boot server Central boot server Configuration File Specify the type of validation to perform on a request by request sip cfg basis appropriate to specific event types in some cases For more information refer to Request Validation lt requestValidation gt on page A 15 Secure Real Time Transport Protocol Secure Real Time Transport Protocol SRTP provides means of encrypting the audio stream s of VoIP phone calls to avoid interception and eavesdropping on phone calls For detailed configuration instructions refer to Technical Bulletin 25751 Secure Real Time Transport Protocol on SoundPoint IP Phones at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuration File Encryption 4 82 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 301 320 330 430 501 550 560 600 601 650 and 670 and the So
284. nct 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 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 Overview 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 distinct call waiting types D
285. nd 670 or the Do Not Disturb option on the Features menu on the SoundPoint IP 320 330 and 430 and SoundStation IP 4000 6000 and 7000 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 320 330 430 500 501 550 560 600 601 650 and 670 desktop phones and SoundStation Central boot server Configuring Your System IP 4000 6000 and 7000 conference phones 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 320 330 430 550 560 650 and 670 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 usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuration changes can performed centrally at the boot server or locally Configuration file sip cfg Enable or disable persistent headset mode For more information refer to User Preferences lt up gt on page A 25 Enable or disable hands free speakerphone mode For more information refer to User Preferences lt up gt on page A 25 Configuration file Specify whether or not t
286. nd another user party C selected by party A The phone offers three types of transfers e 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 Configuring Your System 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 e For more information refer to Call Handling Configuration lt call gt on page A 64 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 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 rel
287. ndStation IP Flash Parameter Configuration A 124 Warning Warning 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 this 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 This feature is very p
288. ndStation 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 voice codecPref IP_6000 G711Mu Null 1 13 5 Specifies the codec preferences for the f 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 Null 24kbps voice codecPref IP_6000 G7221C Null 32kbps voice codecPref IP_6000 G7221C 1 48kbps voice codecPref IP_6000 Siren14 Null 24kbps voice codecPref IP_6000 Siren14 Null 32kbps voice codecPref IP_6000 Siren14 2 48kbps Administrator s Guide SoundPoint IP SoundStation IP Attribute voice codecPref IP_7000 G711Mu voice codecPref IP_7000 G711A voice codecPref IP_7000 G722 voice codecPref IP_7000 G7221 16kbps voice codecPref IP_7000 G7221 24kbps voice codecPref IP_7000 G7221 32kbps voice codecPref IP_7000 G7221C 24kbps voice codecPref IP_7000 G7221C 32kbps voice codecPref IP_7000 G7221C 48kbps voice codecPref IP_7000 G729AB voice codecPref IP_7000 Lin16 16ksps voice codecPref IP_7000 Lin16 32ksps voice codecPref IP_7000 Lin16 48ksps voice codecPref IP_
289. ndStation IP 7000 conference phones e Treble Bass Controls Equalizes the tone of the high and low frequency sound from the speakers on SoundStation IP 7000 conference phones e Display of Warnings from SIP Headers Displays a pop up to user that is found in the Warning Field from a SIP header The following existing features were changed in SIP 3 1 e Call Forward The Diversion field can be used by the call server to inform the phone of a call s history e Call Hold If supported by the call server you can supply a Music on Hold URI e Local Contact Directory A new Label field has been added to each contact directory entry e Busy Lamp Field The attendant can now monitor all calls states and pickup remote calls Administrator s Guide SoundPoint IP SoundStation IP e Microbrowser An XML API allows for the creation of more advanced applications e Multilingual User Interface Polish and Slovenian are now available as languages choices Documentation of the newly released SoundPoint IP 560 and 670 desktop phones and SoundStation IP 6000 and 7000 conference phones has also been added Setting up Your System POLYCOM 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 informatio
290. ng 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 62 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 320 330 430 550 560 600 601 650 and 670 phones only The Busy Lamp Field BLF attendant console feature enhances support for a phone based attendant console A 121 Administrator s Guide SoundPoint IP SoundStation IP This configuration attribute is defined as follows Permitted Attribute Values Def
291. ng server A 118 per phone configuration file A 106 phone diagnostics 5 9 phonel cfg A 106 port lt port gt A 62 presence 4 60 presence lt pres gt A 72 product model part number mapping C 26 protocol lt volpProt gt A 6 protocol server lt server gt A 7 protocol special events lt specialEvent gt A 16 provisioning lt prov gt A 90 provisioning over CLink 4 39 provisioning protocols 3 4 provisioning protocols supported 3 4 Q QOS See also Quality of Service Quality of Service lt QOS gt A 55 quotas lt quotas gt A 94 R RAM disk lt ramdisk gt A 90 rebooting phones 3 17 3 20 receive equalization lt rxEq gt A 49 registration lt reg gt A 107 reliability of provisional responses B 9 request lt request gt A 91 request delay lt delay gt A 91 request validation lt requestValidation gt A 15 resetting to factory defaults 3 5 resource lt res gt A 93 resource files overview 2 7 RFC support B 2 ring type lt rt gt A 36 ringer patterns A 34 roaming buddies lt roaming_buddies gt A 122 roaming privacy lt roaming_provacy gt A 123 routing lt routing gt A 118 routing server lt server gt A 21 A 118 Index RTP lt RTP gt A 56 A 57 A 62 S sampled audio files A 31 sampled audio for sound effects lt saf gt A 30 SCA See also shared call appearances scheduled logging parameters A 87 SDP lt SDP gt A 9 secure real time transport protocol 4 82 security lt sec gt A 88 server menu 3 9 server
292. ng 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 subnet mask 1 IP gateway 3 Setting up Your System Alternate Configuration File Local Parameter DHCP Option DHCP DHCP application only FLASH Refer to DHCP boot server Menu on page address 3 7 151 Note This value SIP server address 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 59 The CDP Compatibility value can be obtain
293. nical_Bulletins_pub html 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 Preferences 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 e Resource lt res gt Administrator s Guide SoundPoint IP SoundStation IP Protocol lt volpProt gt POLYC
294. nistrative Tasks SoundStation IP 7000 EL a oe A 3 T 4 Le a5 9 ibid Led D eE EP 4 E gt EP 9 28 roq E E 5 EA CE 4 EZ U 4 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 Internal Key Functions A complete list of internal key functions for enhanced feature keys and hard key mappings is shown in the following table The following guidelines should be noted e Some functions are dependent on call state Generally if the soft key appears on a call screen the soft key function is executable There are some exceptions on the SoundPoint IP 320 330 phone because it does not display as many soft keys C 19 Administrator s Guide SoundPoint IP SoundStation IP On the Sound Point IP 320 330 phone CallPickup and ParkedPickup refer to the same function On other phones CallPickup refers to the soft key function that provides the menu with separate soft keys for parked pickup
295. ns 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 Configuring Your System 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 Shared Calls lt shared gt on page A 67 Specify line seize subscription period e 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 e For more information refer to Special Events lt specialEvent gt on
296. ns hf IP_4000 signalAttn 6 voice ns hf IP_4000 silenceAttn 9 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 Configuration Files Attribute Default voice rxEq hs IP_330 preFilter enable 1 voice rxEq hs IP_430 preFilter enable voice rxEq hs IP_500 preFilter enable voice rxEq hs IP_600 preFilter enable voice rxEq hs IP_601 preFilter enable voice rxEq hs IP_650 preFilter enable 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 voice rxEq hs IP_601 postFilter enable voice rxEq hs IP_650 postFilter enable voice rxEq hd IP_330 preFilter enable voice rxEq hd IP_430 preFilter enable voice rxEq hd IP_500 preFilter enable voice rxEq hd IP_600 preFilter enable voice rxEq hd IP_601 preFilter enable o OoO o o o o o yoyo aoyo voice rxEq hd IP
297. ns hs silenceAttn 9 For more information refer to Acoustic Echo Cancellation lt aec gt on page A 37 Acoustic Echo Suppression lt aes gt on page A 46 and Background Noise Suppression lt ns gt on page A 47 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 AEC on the SoundPoint IP 301 handset is not supported Audio Codecs The following table summarizes the phone s audio codec support Effective Sample audio Algorithm MIME Type Ref Bit Rate Rate Frame Size bandwidth G 711u 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 G7221 16000 RFC 3047 16 Kbps 16 Ksps 20ms 80ms 7 KHz 24 Kbps 32 Kbps G 722 1C G7221 G7221C 24 Kbps 32 Ksps 20ms 80ms 14 KHz 32000 32 Kbps 48 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 Lini6 L16 16000 RFC 1890 25 6 Kops 16 Ksps 10ms 7 KHz L16 32000 51 2 Kops 32 Ksps 14 KHz L16 48000 76 8 Kops 48 Ksps 22 KHz 4 76 Configuring Your System Effective Sample audio Algorithm MIME Type Ref Bit Rate Rate Frame Size bandwidth Siren14 SIREN14 SIREN14 24 Kbps 32 Ksps 20ms 80ms 14 KHz 16000 32 Kbps 48 K
298. ns 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 6000 and 7000 supports a single registration 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 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 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 If set to private use standard call signa
299. nta Clara County California Each of the parties agrees to the exercise over them of the personal jurisdiction of such courts for such purpose 9 3 U S Government Restricted Rights The Software and documentation are provided with Restricted Rights The Software programs and documentation are deemed to be commercial computer software and commercial computer software documentation respectively pursuant to DFAR Section 227 7202 and FAR 12 212 b as applicable Any use modification reproduction release performance display or disclosure of the Software programs and or documentation by the U S Government or any of its agencies shall be governed solely by the terms of this Agreement and shall be prohibited except to the extent expressly permitted by the terms of this Agreement Any technical data provided that is not covered by the above provisions is deemed to be technical data commercial items pursuant to DFAR Section 227 7015 a Any use modification reproduction release performance display or disclosure of such technical data shall be governed by the terms of DFAR Section 227 7015 b 9 4 Relationship Between the Parties The relationship between you and Polycom is that of licensee licensor Neither party will represent that it has any authority to assume or create any obligation express or implied on behalf of the other party nor to represent the other party as agent employee franchisee or in any other capacity Nothing in
300. nu 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 Administrator s Guide SoundPoint IP SoundStation IP 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 the configuration files bridged lines are configured by shared line parameters Configuration changes can performed centrally at the boot server or locally Central Configuration file Specify whether diversion should be disabled on shared lines boot server sip ctg e For more information refer to Call Handling Configuration lt call gt on page A 64 Configuration file Specify per registration line type private or shared and the shared phonet cfg line third party name A shar
301. 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 integer Instruction type Interpretation value 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 positive If instruction type is chord this optional parameter specifies the on param 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 Administrator s G
302. o 1 apply daylight savings rules 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 60 Configuration Files Attribute tcplpApp snip daylightSavings start date Permitted Values 1 31 Default 8 Interpretation If fixedDayEnable is set to 1 use as day of the month to start DST If ixedDayEnable 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 ina 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 7 Sat tcplpApp sntp daylightSavings start dayOfWeek lastInMonth 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 D
303. o select the label color for soft key and line key labels Users can select which background and label color appears on their phone You can modify the supported solid color and pictures backgrounds For example you can add a grey solid color background or modify a picture 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 e Configuring Your System For the solid backgrounds set the name and RGB values For example bg hiRes gray pat solid 3 name Gray bg hiRes gray pat solid 3 red 128 bg hiRes gray pat solid 3 green 128 bg hiRes gray pat solid 3 blue 128 For images select a filename For example bg hiRes gray bm 3 name polycom jpg bg hiRes gray bm 3 em name polycomEM jpg bg hiRes gray bm 3 adj 0 The default size for images on a phone is 320 x 160 The default size for images on an Expansion Module is 160 x 320 Use a photo editor on a computer to adjust the image you want to display Edit the image so the main subject is centered in the upper right corner of the display
304. o statically configure a set of DNS NAPTR SRV and or A records into the phone Configuring Your System When a phone is configured with a DNS server it will behave as follows by default e An initial attempt to resolve a hostname that is within the static DNS cache for example to register with its SIP registrar results in a query to the DNS e Ifthe initial DNS query returns no results for the hostname or cannot be contacted then the values in the static cache are used for their configured time interval e After the configured time interval has elapsed a resolution attempt of the hostname will again result in a query to the DNS e Ifa DNS query for a hostname that is in the static cache returns a result the values from the DNS are used and the statically cached values are ignored When a phone is not configured with a DNS server it will behave as follows e An attempt to resolve a hostname that is within the static DNS cache will always return the results from the static cache Support for negative DNS caching as described in RFC 2308 is also provided to allow faster failover when prior DNS queries have returned no results from the DNS server For more information go to http tools ietf org html rfc2308 Configuration changes can be performed centrally at the boot server Central Configuration file Specify DNS NAPTR SRV and A records for use when the phone is boot server sip cfg WY POLYCOM not configure
305. of times the phone will retry a notify when attempting to seize a line BLA volpProt SIP header diversion enable 0 1 If set to 1 the diversion header is displayed if received If set to 0 or Null the diversion header is not displayed Configuration Files Attribute volpProt SIP header list useFirst Permitted Values 0 1 Default p Interpretation If set to 1 or Null the first diversion header is displayed If set to 0 the last diversion header is displayed volpProt SIP header warning codes comma accept separated list Null A list of accepted warning codes If set to Null all codes are accepted Only codes between 300 and 399 are supported For example if you want to accept only codes 325 to 330 volpProt SIP header warning codes acc ept 325 326 327 328 329 330 Text will be shown in the appropriate language For more information refer to lcl ml lang tags x in Multilingual lt ml gt on page A 22 volpProt SIP header warning enable 0 1 If set to 1 the warning header is displayed if received If set to 0 or Null the warning header is not displayed This attribute also includes Outbound Proxy lt outboundProxy gt Alert Information lt alertInfo gt Request Validation lt requestValidation gt Special Events lt specialEvent gt Conference Setup lt conference gt A 13 Administrator s Guide SoundPo
306. on describes the current SoundPoint IP conference 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 Introducing the SoundPoint IP SoundStation IP Family The currently supported conference phones are e SoundStation IP 4000 e SoundStation IP 6000 e SoundStation IP 7000 Administrator s Guide SoundPoint IP SoundStation IP Key Features of Your SoundPoint IP SoundStation IP Phones The key features of the SoundPoint IP SoundStation IP phones are e 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 e 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 e Platform independent Supports multiple protocols and platforms enabling standardization on one phone for multiple locations systems and vendors Polycom s support of the leading protocols and industry partners makes it a future proof choice e Field upgradeable Upgrade SoundPoint IP SoundStation IP as s
307. on file sip cfg Specify whether RFC 2543 c 0 0 0 0 or RFC 3264 a sendonly or 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 67 Specify the Music on Hold URI For more information refer to Music on Hold lt musicOnHold gt on page A 17 Configuration file Specify the Music on Hold URI phone1 cfg e For more information refer to Music on Hold lt musicOnHold gt on page A 17 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 Interface Use the SIP Configuration menu to specify whether or not to use RFC 2543 c 0 0 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 a
308. onds 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 Administrator s Guide SoundPoint IP SoundStation IP 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 ttrs srtp usb efk clink Idap and peer and pnetm 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
309. onsole phone Requires call server support Configurable Feature Keys Certain 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 This feature is part of the Productivity Suite 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 430 501 550 560 600 601 650 and 670 desktop phones and the SoundStation IP 4000 6000 and 7000 conference phones 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 Multilingual User Interface All phones except SoundPoint IP 301 have multilingual user interfaces Overview 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 L
310. ontent is displayed immediately This attribute also includes e Idle Display lt idleDisplay gt e Main Browser lt main gt e Browser Limits lt limits gt Administrator s Guide SoundPoint IP SoundStation IP 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 formed valid HTTP page For example URL Length up to http www example com xhtml frontpage cgi pa 255 characters 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 80 Note If ind idleDisplay enabled is enabled miscellaneous XML errors can occur on SoundPoint IP 430 501 550 560 600 601 650 and 670 and SoundStation IP 4000 6000 and 7000 phones 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 respecte
311. or any failure by you to comply with the terms of this Agreement 10 Disclaimer of Liability UNDER NO CIRCUMSTANCES SHALL POLYCOM BE LIABLE FOR SPECIAL INDIRECT INCIDENTAL OR CONSEQUENTIAL DAMAGES INCLUDING WITHOUT LIMITATION DAMAGES RESULTING FROM DELAY OF DELIVERY OR FROM LOSS OF PROFITS DATA BUSINESS OR GOODWILL ON ANY THEORY OF LIABILITY WHETHER ARISING UNDER TORT INCLUDING NEGLIGENCE CONTRACT OR OTHERWISE WHETHER OR NOT POLYCOM HAS BEEN ADVISED OR IS AWARE OF THE POSSIBILITY OF SUCH DAMAGES POLYCOM S ENTIRE LIABILITY FOR DIRECT DAMAGES UNDER THIS AGREEMENT IS LIMITED TO FIVE DOLLARS 5 00 11 Miscellaneous f any provision is found to be unenforceable or invalid that provision shall be limited or eliminated to the minimum extent necessary so that this Agreement shall otherwise remain in full force and effect and enforceable This Agreement constitutes the entire agreement between the parties with respect to its subject matter and supersedes all prior or contemporaneous understandings regarding such subject matter No addition to or removal or modification of any of the provisions of this Agreement will be binding upon Polycom unless made in writing and signed by an authorized representative of Polycom YOUR USE OF THIS API ACKNOWLEDGES THAT YOU HAVE READ UNDERSTAND AND AGREE TO BE BOUND BY THE TERMS AND CONDITIONS INDICATED ABOVE Polycom Inc 2008 ALL RIGHTS RESERVED www polycom com Corporate Headquarters Phone 408 5
312. ore play out begins 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 multiple of 10 The absolute minimum duration time in 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 jitterBufferMin multiple of 10 lt 300 for IP 320 330 430 501 550 600 601 and 650 lt 200 for IP 301 The largest jitter buffer depth to be supported in milliseconds Jitter above this size will always cause lost packets This parameter should be set to the smallest possible value that will support the expected network jitter Administrator s Guide SoundPoint IP SoundStation IP Volume Persistence lt volume gt 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
313. ork 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 Setting Up Advanced Features Setting Up Audio Features e 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 features 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 e Soft Key Activated User Interface e Speed Dial
314. ot work The dial pad on the SoundPoint Do one of the following IP SoundStation IP family SIP Check for a response from other phone does not respond feature keys or from the dial pad Place acall to the phone from a known working telephone Check for display updates e Press the Menu key followed by System Status and Server Status to check if the telephone is correctly registered to the server e Press the Menu key followed by System Status and Network Statistics Scroll down to see if LAN port shows active or Inactive e 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 Administrator s Guide SoundPoint IP SoundStation IP 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
315. ote 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 550 560 600 601 650 and 670 phones IP_4000 parameters affect SoundStation IP 4000 and 6000 phones and IP_7000 parameters affect SoundStation IP 7000 phones Attribute Permitted 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 82 for this indicator ind gi x y physX IP 300 0 19 IP 330 0 101 IP 400 0 122 IP 500 0 159 IP 600 0 319 IP 4000 0 247 IP 7000 0 255 For Graphic Icon type indicators this is the x axis location of the upper left corner of the indictor measured in pixels from left to right Administrator s Guide SoundPoint IP SoundStation IP Attribute ind gi x y physY Permitted Values IP 300 0 3 IP 330 0 19 IP 400 0 45 IP 500 0 79 IP 600 0 159 IP 4000 0 67 IP 7000 0 127 Interpretation 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
316. oting 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 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 th
317. ous URL it can send a new header through UPDATE with expires 0 The expires parameter is ignored when determining whether to spontaneously retrieve the web content unless expires 0 e A mode parameter is defined to indicate whether the web content should be displayed spontaneously or retrieved on demand Two values are allowed active and passive If the parameter is absent or invalid this will be interpreted the same as passive meaning that the web content will be retrievable on demand but will not be spontaneously displayed If the value is set to active the web content will be spontaneously displayed subject to the rules discussed under Active Mode in Web Content Retrieval on page 4 66 For example Access URL lt http server polycom com content23456 xhtml gt expires 60 mode passive In this case the phone will indicate in the call appearance user interface that web content is available for a period of 60 seconds and will retrieve the web content at the request of the user for a period of up to 60 seconds but the phone will not spontaneously switch to the microbrowser application and download the content Starting with SIP 2 1 0 failover redundancy can only be utilized when the configured IP server hostname resolves through SRV or A record to multiple IP addresses Unfortunately some customer s are unable to configure the DNS to take advantage of failover redundancy The solution in SIP 3 1 is to provide the ability t
318. owerful 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 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 Configuration Files 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 0 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 Oor 1 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 ne
319. p cfg Turn this feature on or off For more information refer to Feature lt feature gt on page A 92 Administrator s Guide SoundPoint IP SoundStation IP 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 Configuration changes can performed centrally at the boot server boot server sip cfg Central Configuration file Turn this feature on or off Call Park Retrieve An active call 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 boot server sip cfg Central Configuration file Turn this feature on or off 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 boot server sip cfg Central Configuration file Turn this feature on or off Specify the string sent to the server for last call return on page A 64 Setting Up Advanced Features 4 22 This section provides information for making configuration changes for the following advanced features Configurable Feature Keys e Multiple Line Keys per Registration e Multiple Call Appearan
320. pProt 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 volpProt SIP connectionReuse useAlias Permitted Values 0 1 Default 0 Interpretation If set to 0 this is the old behavior If set to 1 phone uses the connection 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 sessionTimers 0 1
321. pad5 31 Line6 42 Line5 10 MicMute 21 Dialpad6 32 Conference 11 VolUp 22 Dialpad3 33 Line2 C 16 Miscellaneous Administrative Tasks SoundStation IP 4000 s A AZ oo Ss 5 E C C 40 ZALAN OREO 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 SoundStation IP 6000 j Gaye 10 CQ 743 PONO CCDC 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 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 DialpadO 31 n a 42 n a 10 VolDown 21 DialpadPound 32 n a 11 ArrowDown 22 Redial 33 n a C 18 Miscellaneous Admi
322. padPound DialpadStar DialpadStar DialpadURL Dialname Call screen only DirectedPiclup DirectedPickup Call screen only Directories Directories Divert Forward DoNotDisturb Do Not Disturb menu Exit Exist existing menu Menu only GroupPickup GroupPickup Handsfree Handsfree Hold Toggle Hold Join Join Call screen only LCR LastCallReturn Line1 Line Key 1 Line2 Line Key 2 Line3 Line Key 3 Line4 Line Key 4 Lined Line Key 5 Line6 Line Key 6 ListenMode Turn on speaker to listen only SoundPoint IP 301 only Menu Menu Messages Messages menu MicMute MicMute 21 Administrator s Guide SoundPoint IP SoundStation IP Label Function MyStatus MyStatus NewCall NewCall Call screen only Null Do nothing Offline Offline for presence EnterRecord enterCallRecord Call screen only Redial Redial Call screen only Release EndCall or Cancel hot dial SoundStation IP 7000 only ParkedPickup ParkedPickup Call screen only Select Select Setup Settings menu Silence RingerSilence Call screen only SoftKey1 SoftKey1 SoftKey2 SoftKey2 SoftKey3 SoftKey3 SoftKey4 SoftKey4 SpeedDial SpeedDial Split Split Call screen only Transfer Transfer Call screen only VolDown VolDown VolUp VolUp Assigning a VLAN Miscellaneous Administrative Tasks D Using DHCP To assign a VLAN ID to a phone using DHCP gt gt Inthe DHCP menu of the Main setup menu set
323. page A 18 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 18 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 18 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 configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan x routing server
324. parameters and the statically assigned IP address and subnet mask will never be used in this configuration Resetting to Factory Defaults The basic network configuration referred to in the subsequent sections can be reset to factory defaults using a multiple key combination described in Multiple Key Combinations on page C 10 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 Subnet Mask dotted decimal IP address dotted decimal subnet mask Phone s IP address Note Disabled when DHCP client is enabled 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 whi
325. pg Note If the file is missing or unavailable the built in default solid pattern is displayed Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values button color selection x y modify any string Default Interpretation The label color for soft keys and line key labels associated with the defined backgrounds These values can be modified locally by the user The format is rgoHILo lt parameter list gt For example rbgHiLo 51 255 68 255 0 119 is the default button color associated with the built in background bg gray selection W X 2 1 Specify which type of background w and index x for that type is selected on reboot bg hiRes gray pr x adj bg hiRes gray pat solid x name any string White Specify the brightness adjustment to the graphic 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 bg hiRes gray pat solid x green 0 to 255 bg hiRes gray pat solid x blue 0 to 255 The screen background layouts For x 1 red 255 green 255 blue 255 For x 2 red 160 green 160 blue 160 For x 3 and 4 all values are Null Note The values for red green and blue must be the same to display correctly on grayscale A 78 Configuration Files Attribute bg hiRes gray om x name Permitted Values any string bg hiRes gray
326. 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 To disabled the PC Ethernet port refer to Disabling PC Ethernet Port on page C 27 SoundPoint IP Switch Port Priorities To help ensure good voice quality the Ethernet switch 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 55 Graphic Display Backgrounds Note 4 16 You can set up a picture or design to be displayed on the background of the graphic display of all SoundPoint IP 550 560 650 and 670 phones There are a number of default backgrounds both solid color and pictures Both BMP and JPEG files are supported You can als
327. ported 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 Header Supported Notes Accept Yes Accept Encoding No Accept Language Yes 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 ldentity
328. poseAutoHolds 0 1 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 Interpretation 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 Administrator s Guide SoundPoint IP SoundStation IP Directory lt dir gt This attribute includes e Local Directory lt local gt e Corporate Directory lt corp gt 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 Interpretatio
329. pted 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 Note Note Miscellaneous Administrative Tasks You can check whether an encrypted file is the same as an unencrypted file by 1 Run the configFileEncrypt utility 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
330. 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 for 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
331. r messages before selecting an offer 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 next section Server Menu If the DHCP server sends nothing the following scenarios are possible e 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 If no alternate DHCP server responds the INFORM query process will retry and eventually time out Administrator s Guide SoundPoint IP SoundStation IP Name Boot Server continued Possible Values 1 Custom Description 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 que
332. r 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 Setting up Your System The following Ethernet configuration parameters can be modified on the Ethernet menu Name Possible Values Description CDP Compatibility Enabled Disabled If enabled the phone will use a CDP compatibility method 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 Disabled 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 F Cp HD means half duplex and FD means full duplex 4 100FD Note Polycom recommends that you do not change this 5 1000FD setting PC Port Mode 0 Auto The network speed over the Ethernet 1 10HD The default value is Auto r an HD means half duplex
333. r during boot up Note When several font IP_600 x name are defined the index x must follow consecutive increasing order 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 SoundPoint IP 301 320 330 430 501 550 560 600 601 and 650 and 670 and SoundStation IP 4000 6000 and 7000 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 Administrator s Guide SoundPoint IP SoundStation IP POLYCOM Polycom does not recommend the remapping for keys In the following table x IP_300 IP_330 IP 430 IP_500 IP_550 IP_600 IP_650 and IP_4000 and IP_7000 and y is the key number Note that IP_300 parameters affect SoundPoint IP 301 phones IP_330 parameters affect SoundPoint IP 3
334. r enable 0 voice txEq hs IP_430 preFilter enable 0 voice txEq hs IP_500 preFilter enable 0 voice txEq hs IP_600 preFilter enable 0 A 50 Configuration Files Attribute Default voice txEq hs IP_601 preFilter enable 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 o 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 voice txEq hf IP_330 preFilter enable voice txEq hf IP_430 preFilter enable voice txEq hf IP_500 preFilter enable voice txEq hf IP_600 preFilter enable voice txEq hf IP_601 preFilter enable o o o o o o o yo ao oyo voice txEq hf IP_650 preFilter enable e voice txEq hf IP_4000 preFilter enable voice txEq hf
335. r purposes derivative works thereof Except as provided below this License Agreement does not grant you any rights to patents copyrights trade secrets trademarks or any other rights related to the Software 2 DESCRIPTION OF OTHER RIGHTS AND LIMITATIONS 2 1 Copyright All title and copyrights in and to the Software and any copies of the Software are owned by Polycom or its suppliers The Software is protected by copyright laws and international treaty provisions Title ownership rights and intellectual property rights in the Software shall remain in Polycom or its suppliers 2 2 Ownership of Derivative Works As between you and Polycom you will own copyright and other intellectual property rights in derivative works of the Software that you develop 2 3 Reservation Polycom reserves all rights in the Software not expressly granted to you in this Agreement 3 SUPPORT SERVICES 3 1 No Support Services Polycom provides no support services for the Software 4 TERMINATION 4 1 Termination Without prejudice to any other rights Polycom may terminate this Agreement if you fail to comply with any of the terms and conditions of this Agreement In such event you must destroy all copies of the Software and all of its component parts You may terminate this Agreement at any time by destroying the Software and all of its component parts 5 NO WARRANTY THE SOFTWARE IS LICENSED WITHOUT WARRANTY AS IS AND WITH ALL FAULTS ALL
336. rally at the boot server Central Configuration file Specify the user s preferences for treble and bass boot server sip cfg e For more information refer to User Preferences lt up gt on page A 25 Setting Up Security Features This section provides information for making configuration changes for the following security related features e Local User and Administrator Privilege Levels e Custom Certificates e Incoming Signaling Validation 4 80 Configuring Your System e Secure Real Time Transport Protocol 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 83 Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file Specify the minimum lengths for the user and administrator sip cfg passwords For more information refer to Password Lengths lt pwd gt lt length gt on page A 89 Local Web Server None if enabled Local Phone
337. ration file 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 A 111 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 NOTE If volpProt SIP serverFeatureControl dnd is set to 1 enabled this parameter is ignored For more information refer to SIP lt SIP gt on page A 10 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 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 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 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
338. ration file Turn time and date display on or off boot server sip cfg e For more information refer to User Preferences lt up gt on page A 25 Set the time and date display formats For more information refer to Date and Time lt datetime gt on page A 25 Set the basic SNTP settings and daylight savings parameters e For more information refer to Time Synchronization lt sntp gt on page A 59 Configuring Your System Local Web Server if enabled Set the basic SNTP and daylight savings settings 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 Interface The basic SNTP settings can be made in the Network Configuration 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 exc
339. redundancy 4 56 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 73 basic features 4 1 boot server 3 12 network 3 2 security features 4 80 shared call appearance signaling B 10 shared call appearances shared calls lt shared gt A 67 shared lines barge in 4 27 A 109 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 14 upgrading 3 19 Index 5 Administrator s Guide SoundPoint IP SoundStation IP SIP basic protocols header support B 4 SIP header diversion A 12 warning A 13 SIP headers warnings 4 72 SIP See also Session Initiation Protocol sip cfg A 4 SIP lt SIP gt A 10 soft keys lt softkey gt A 103 sound effects lt se gt A 31 SoundPoint IP applications 4 33 configuring phones locally 4 83 features list of 1 6 supported languages 4 31 SoundPoint IP SoundStation
340. rformed 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 92 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 112 Connected Party Identification The identity of the remote party to which the user has connected is displayed and logged if the name and ID is provided by the call server The connected party identity is derived from the network signaling In some cases the remote party will be different from the called party identity due to network call diversion There are no related configuration changes Context Sensitive Volume Control The volume of user interface sound effects such as the ringer and the receive volume of call audio is adjustable While transmit levels are fixed according to the TIA EIA 810 A standard receive volume is adjustable For SoundPoint IP and phones if using the default configuration parameters the receive handset headset volume resets to nominal after each call to comply with regulatory requirements Handsfree volume persists with subsequent calls Configuration changes can performed centrally at the boot server Central boot server Confi
341. ric 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 120 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 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 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 ver
342. ring in this case no label appears If this parameter is omitted the Null string is used Note If you exceed the phone physical layout text limits the text will be shortened and will be appended Administrator s Guide SoundPoint IP SoundStation IP Note Name Interpretation type The SIP method to be performed once the macro starts executing This parameter has the following values e If set to invite the action required is performed using the SIP INVITE method Note This parameter is included for backwards compatability only Do not use if at all possible If the action string contains types this parameter is ignored If this parameter is omitted the default is INVITE action string The action string contains a macro definition of the action to be performed For more information refer to Macro Definition on page 4 44 This parameter must have a value and it cannot be Null lt efkprompt gt This element describes the behavior of the user prompts The different blocks are uniquely identified by number following efk efkprompt prefix for example efk efkprompt 1 lt suffix gt In SIP 3 0 a maximum of four user prompts were supported In SIP 3 1 a maximum of ten user prompts are supported This element contains the following parameters Name Interpretation status This parameter has the following values e If set to 1 this key is enabled e If se
343. ring For descriptions refer to Server Menu on page 3 9 device prov serverlype 0to4 device prov user any string device prov password any string device prov appProvType Oor1 device prov appProvString any string device prov 10 Null redunAttemptLimit device prov 300 Null redunIinterAttemptDelay 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 to which the phone directs Domain Name System queries device dns domain any string The phone s DNS domain device auth any string The phone s local administrator password localAdminPassword device auth any string The phone user s local password localUserPassword 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 any string Configuration encryption key that is used for configEncryption key encryption of configuration files device syslog serverName dotted decimal IP address OR domain name string
344. rpretation 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 Administrator s Guide SoundPoint IP SoundStation IP Voice Coding Algorithms lt codecs gt The following voice codecs are supported Sample Effective Audio Algorithm MIME Type Label Bit Rate Rate Frame Size Bandwidth G 711u 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 G7221 16000 G7221 16 Kbps 16 Ksps 20ms 80ms 7 KHz 24 Kbps 32 Kbps G 722 1C G7221 G7221C 24 Kbps 32 Ksps 20ms 80ms 14 KHz 32000 32 Kbps 48 Kbps G 729AB G729 G729AB 8 Kbps 8 Ksps 10ms 80ms 3 5 KHz Lin16 L16 16000 L16 25 6 Kbps 16 Ksps 10ms 7 KHz
345. rs send receive status or media flow receive and transmit codecs and hold status Configuration changes can be performed centrally at the boot server Central boot server sip cfg Configuration file Turn this feature on or off For more information refer to Feature lt feature gt on page A 92 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 SIP 2 1 feature with the following exception
346. ry 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 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 next 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
347. s Attribute Permitted Values Default Interpretation dns cache SRV x name domain name string Null The domain name string with SRV prefix dns cache SRV x ttl 0 to 65535 seconds 300 Specifies the time interval that the resource record may be cached before the source of the information should again be consulted dns cache SRV x priority 0 to 65535 0 The priority of this target host For more information go to http tools ietf org html rfc2782 dns cache SRV x weight 0 to 65535 0 A server selection mechanism For more information go to http tools ietf org html rfc2782 dns cache SRV x port 0 to 65535 0 The port on this target host of this service For more information go to http tools ietf org html rfc2782 dns cache SRV x target domain name string Null The domain name of the target host For more information go to http tools ietf org html rfc2782 A lt A gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dns cache A x name valid hostname Null Hostname dns cache A x ttl 0 to 65535 300 Specifies the time interval that the resource record may be cached before the source of the information should again be consulted dns cache A x address dotted decimal IP Null IP address that hostname dns cache A x name version 4 address maps to A 102 Soft Keys lt softkey gt Configuration Files Th
348. s There is no indication on the phone s user interface whether or not server based call forwarding 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 The Diversion field with a SIP header is often used by the call server to inform the phone of a call s history For example when a phone has been set to enable call forwarding the Diversion header allows the receiving phone to indicate who the call was from and from which phone number it was forwarded For more information refer to Header Support on page B 4 Configuration changes can performed centrally at the boot server or locally Central boot server Configuration file sip cfg Configuration file phonet cfg Enable or disable server based call forwarding e For more information refer to SIP lt SIP gt on page A 10 Enable or disable display of Diversion header and the order in which to display the caller ID and number e For more information refer to SIP lt SIP gt on page A 10 Enable or disable server based call forwarding as a per registration feature For more information refer to Registration lt reg gt on page A 107 Set all call d
349. s 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 phones may require resource files that are used by some of the advanced features These files are optional but if the particular feature is being employed these files are required Some examples of resource files include e Language dictionaries e Custom fonts Administrator s Guide SoundPoint IP SoundStation IP Available Features Ring tones Synthesized tones Contact directories This section provides information the features available on the SoundPoint IP SoundStation IP phones Basic User Features Automatic Off Hook Call Placement Supports an optional automatic off hook call placement feature for each registration 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 maintained for each disti
350. s 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 A call server is active and 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 e New Features in SIP 3 1 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 SoundS
351. s 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 SIP 2 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 Configuration changes can be performed centrally at the boot server or locally Central Configuration file Enable or disable server based DND boot server sip cfg e 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 64 Configuration file Enable or disable server based DND as a per registration feature phonet cfg For more information refer to Registration lt reg gt on page A 107 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 116 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 a
352. s 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 set 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 re
353. selected Server Type Note If the Server Address is a URL 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 1to 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 14 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 value is Disabled Note 3 10 The Server User and Serve
354. sending can be initiated by replying to an incoming message or by initiating a new dialog The destination for new dialog messages can be entered manually or selected from the contact directory the preferred method Configuration changes can be 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 92 Multilingual User Interface 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 550 560 650 and 670 and SoundStation IP 4000 6000 and 7000 s higher resolution displays A WGL4 character set is displayed the SoundStation IP 7000 For more information refer to http www microsoft com OpenType otspec WGL4E HTM For basic character support and extended character support available on Sound Point IP 550 560 600 601 and 650 and 670 and SoundStation IP platforms refer to Multilingual lt ml gt on page A 22 Note that within a Unicode range some characters may not be supported due to their infrequent usage Configuring Your System The SoundPoint IP and SoundStation IP user
355. sented 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 25 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 boot 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 pe
356. 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 301 320 330 430 301 320 330 430 501 550 560 600 501 550 560 600 Specified 601 650 670 4000 601 650 670 4000 Protocol 6000 7000 6000 7000 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 downloa
357. sh 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 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 10 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 performs 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 lo
358. sion 3 Microsoft s Active Directory is included Both corporate directories that support server side sorting and those that do 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 POLYCOM 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 is placed 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 Status For more information refer to Technical Bulletin 41137 Best Practices When Using Corporate Directory on SoundPoint IP and SoundStation IP Phones at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Configuration changes can performed centrally at the boot server Central boot server
359. ssaging lt mso gt sig oct ok one cals tAn eee gurney ewe eee nd A 119 Network Address Translation lt nat gt 0 00 eee eee A 120 Attendant lt attendant gt 0 c cece cece eens A 121 Roaming Buddies lt roaming_buddies gt A 122 Roaming Privacy lt roaming_privacy gt 00 A 123 User Preferences lt user_preferences gt 0000005 A 123 Flash Parameter Configuration 0 00 e eee ee eee A 124 B Session Initiation Protocol SIP Brl RFC and Internet Draft Support 0 00 eee eee eee B 2 REQUESE Suppor x4 ecu raura no LEV Ae ee ok ee weed ees B 3 Header Support 22 cuts reee eiei FAS eR ASRS bh aah onto sites B 4 Response SUPPOLE isc eee aod ee wha ketene eke ee Rae Ra ee B 6 Hold Implementation 0 0 eee eee eens B 9 Reliability of Provisional Responses 00 0000 eee ee ene B 9 Administrator s Guide SoundPoint IP SoundStation IP Wransem 2 i234 speran as ran eTR ness incre ies oe teeta ee Seed B 9 Third Party Call Control 0 0 B 9 SIP for Instant Messaging and Presence Leveraging Extensions B 10 Shared Call Appearance Signaling 00 8 B 10 Bridged Line Appearance Signaling 0000 B 10 C Miscellaneous Administrative Tasks 22 C l Trusted Certificate Authority List 0 0 eee eee eee eee C 1 Encrypting Confi
360. st be processed to ensure the correct ordering of rules dns cache NAPTR x 0 to 65535 0 A 16 bit unsigned integer that specifies the order preference in which NAPTR records with equal order values should be processed low numbers being processed before high numbers dns cache NAPTR x string Flags to control aspects of the rewriting and flags interpretation of the fields in the record Flags are single characters from the set A Z 0 9 The alphabetic characters are case insensitive At this time only four flag S A U and P are defined For more information go to http tools ietf org html rfc2915 dns cache NAPTR x string Specifies the service s available down this service rewrite path For more information go to http tools ietf org html rfc2915 dns cache NAPTR x string Null A string containing a substitution expression that regexp is applied to the original string held by the client in order to construct the next domain name to lookup The grammar of the substitution expression is given in RFC 2915 Note This attribute is currently not used dns cache NAPTR x domain name string Null The next name to query for NAPTR SRV or replacement with SRV prefix address records depending on the value of the flags field It must be a fully qualified domain name A 101 Administrator s Guide SoundPoint IP SoundStation IP SRV lt SRV gt This configuration attribute is defined as follow
361. strator 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 Increasing this value may have a detrimental effect on performance of the phone Applications lt apps gt This attribute s settings control the telephone notification events state polling events and the push server controls This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation apps telNotification URL URL Null The URL to which the phone sends notifications of specified events The protocol used can be either HTTP or HTTPS
362. t 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 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 Administrator s Guide SoundPoint IP SoundStation IP Index Numerics 802 10 VLAN header 4 78 A access URL 4 65 ACD See also automatic call distribution acoustic echo cancellation 4 75 acoustic echo cancellation lt aec gt A 45 acoustic echo suppression lt aes gt A 46 AEC See also acoustic echo cancellation AGC See also automatic gain control alert information A 15 animations lt anim gt A 81 application configuration acoustic echo cancell
363. t for subsequent usage of the corporate directory If set to 1or Null 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 behavior after exiting from the corporate directory Administrator s Guide SoundPoint IP SoundStation IP Permitted Attribute Values Default Interpretation dir corp cacheSize 8 to 256 64 The maximum number of entries that can be cached locally on the phone dir corp pageSize 8 to 64 16 The maximum number of entries requested from the corporate directory server with each query 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 is not a valid line registration number it is ignored pres idleSoftkeys 0 1 Null If set to Null or 0 the presence idle soft keys MyStat and Buddies do not appear If set to Null or 1 the presence idle soft keys
364. t 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 vianid 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 Oor 1 For description refer to DHCP or Manual TCP IP Setup on page 3 2 device dhcp offerTimeout 1 to 600 Number of seconds the phone waits for secondary DHCP Offer messages before selecting an offer device dhcp 0to3 For descriptions refer to DHCP Menu on page 3 7 bootSrvUseOpt device dhcp bootSrvOpt 128 to 254 Cannot be the same as VLAN ID Option device dhcp Oor1 bootSrvOptType device dhcp 0to2 dhcpVlanDiscUseOpt device dhcp 128 to 254 Cannot be the dhcpVlanDiscOpt same as Boot Server Option A 125 Administrator s Guide SoundPoint IP SoundStation IP Name Possible Values Description device prov serverName any st
365. t 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 user 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 lag 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 parameter
366. t to 0 this key is disabled This parameter must have a value and it cannot be Null Note If a macro attempts to use a prompt that is disabled or invalid the macro execution fails label This parameter sets the prompt text that will be presented to the user on the user prompt screen The value can be any string including the null string in this case no label appears If this parameter is omitted the Null string is used Note If you exceed the phone physical layout text limits the text will be shortened and will be appended Note Configuring Your System Name Interpretation userfeedback This parameter specifies the user input feedback method It has the following values e If set to visible the text appears as clear text If set to masked the text appears as characters For example if a password is entered If this parameter is omitted the value visible is used If this parameter has an invalid value including Null this prompt is invalid and all parameters depending on this prompt are invalid type The type of characters entered by the user This parameter has the following values e Ifsetto numeric the characters are interpreted as numbers If set to text the characters are interpreted as letters If this parameter is omitted the value numeric is used If this parameter has an invalid value including Null
367. 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 A 88 This attribute also includes e Encryption lt encryption gt e Password Lengths lt pwd gt lt length gt Attribute Encryption lt encryption gt Configuration Files This configuration attribute is defined as follows 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 This will replace whatever phone specific contact directory is on the server even if it is unencrypted sec encryption upload overrides 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 ev
368. tandards develop and protocols evolve Extends the life of the phone to protect your investment 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 Aline x 20 character monochrome LCD for the SoundPoint IP 301 102 x 23 pixel graphical LCD for the SoundPoint IP 320 330 160x 80 pixel graphical grayscale LCD for the SoundPoint IP 501 320x160 pixel graphical grayscale LCD for the SoundPoint IP 550 560 600 601 650 670 supports Asian characters 248 x 68 pixel graphical LCD for the SoundStation IP 4000 6000 256 x 128 pixel graphical grayscale LCD for the SoundStation IP 7000 Introducing the SoundPoint IP SoundStation IP Family e Dual auto sensing 10 100 1000baseT Ethernet ports Leverages existing infrastructure investment Nore 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 550 560 600 601 650 and 670 and the SoundStation IP 6000 and 7000
369. tation 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 Server PSTN Gateway Voice Bridge 40 100 Application Boot Server Ethernet Server Hub 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 n
370. ted in their application to you 8 EXPORT CONTROLS 8 1 Export Controls The Software may not be downloaded or otherwise exported or re exported 1 into or to a national or resident of Cuba Iraq Libya North Korea Yugoslavia Iran Syria Republic of Serbia or any other country to which the U S has embargoed goods or ii to anyone on the U S Treasury Department s List of Specially Designated Nationals or the U S Commerce Department s Table of Denial Orders By downloading or using this Software you are agreeing to the foregoing and you are representing and warranting that you are not located in under the control of or a national or resident of any such country or on any such list If you obtained this Software outside of the United States you are also agreeing that you will not export or re export it in violation of the laws of the country in which it was obtained 9 MISCELLANEOUS 9 1 Governing Law This Agreement shall be governed by the laws of the State of California as such laws are applied to agreements entered into and to be performed entirely within California between California residents and by the laws of the United States The United Nations Convention on Contracts for the International Sale of Goods 1980 is hereby excluded in its entirety from application to this Agreement 9 2 Venue for Resolving Disputes Any disputes relating to this Agreement will be resolved only in the state or federal courts located in Sa
371. tform prov fileSystem rfs0 minFre eSpace is internally replaced by 4X the value prov polling enabled 0 1 0 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 86400 Polling period in seconds greater than Rounded up to the nearest 3600 number of days in abs mode Measured relative to boot time in rel mode prov polling time Format is 03 00 Only used in abs mode Polling hh mm 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 we POLYCOM Configuration Files 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 used to cache downloaded wave files and other resources for the user interface ramdisk bytesPerBlock 0 32 33 0 These four parameters use internal defaults when 1024 value is set to 0 ramdisk blocksPerTrack 0 1 2 a 0 Note Forte Sound gat I B5 PATON ramdisk bytesPerBlock is internally replaced by 2X 65536 the value ramdisk nBlocks 0 1 2 4096 Note For the SoundPoint IP 7000 platform 65536 ramdisk bytesPerBlock is internally replaced by 4X the value ramdisk nBlocks I
372. th critical and normal push messages apps push serverRootURL URL Null The relative URL received from HTTP URL Push message is appended to the application server root URL and the resultant URL is sent to the Microbrowser For example if the application server root URL is http 172 24 128 85 8080 sampleapps and the relative URL is examples sample html the URL that is sent to the Microbrowser is http 172 24 128 85 8080 sampleapps example s sample html The protocol used can be either HTTP or HTTPS apps push username string Null The user name to access the push server URL Note To enable the push functionality the attributes apps push username and apps push password must be set to non Null values apps push password string Null The password to access the push server URL Administrator s Guide SoundPoint IP SoundStation IP Peer Networking lt pnet gt Peer networking manages communications between Polycom devices For the SoundStation IP 7000 conference phone it manages daisy chaining and video integation with the HDX video systems This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation pnet role auto Null The role of the SoundStation IP 7000 when standAlone communicating with other Polycom devices masterOnly If the attribute is not defined or is null the default masterPref
373. 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 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 1d 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 Administrator s Guide SoundPoint IP SoundStation IP Product Model and Part Number Mapping In SIP 2 1 2 enhancements to the master configuration file were made to allow you to direct phone upgrades to a software image and configuration files based on phone model number firmware part number or MAC address The part number specific version has precedence over the model number version which has precedence over the original version For example CO
374. the SoundStation IP 4000 6000 and 7000 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 value 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 Configu
375. the default number of calls that can be active or on hold per line key e For more information refer to Call Handling Configuration lt call gt on page A 64 Configuration file phonet cfg Specify per registration the number of calls that can be active or on 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 107 Administrator s Guide SoundPoint IP SoundStation IP 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 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 Interface 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 registratio
376. thernet gt The following settings control the 802 1p Q user_priority field e RIP lt RTP gt e Call Control lt callControl gt e Other lt other gt Administrator s Guide SoundPoint IP SoundStation IP 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 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 A 56 RTP lt rtp gt These parameters apply to RTP packets Configuration Files 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
377. this agreement shall be construed to limit either party s right to independently develop or distribute software that is functionally similar to the other party s products so long as proprietary information of the other party is not included in such software 9 5 Entire Agreement This Agreement represents the complete agreement concerning this license and may be amended only by a writing executed by both parties If any provision of this Agreement is held to be unenforceable such provision shall be reformed only to the extent necessary to make it enforceable www polycom com Corporate Headquarters 4750 Willow Road Pleasanton CA 94588 USA Phone 408 526 9000 Fax 408 526 9100
378. thout 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 4 The names OpenSSL Toolkit and OpenSSL Project must not be used to Third Party Software 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
379. tion refer to Attendant lt attendant gt on page A 121 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 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 Administrator s Guide SoundPoint IP SoundStation IP Configuration changes can performed centrally at the boot server Central boot server Configuration File Specify fonts to overwrite existing ones or specify new fonts sip cfg For more information refer to Fonts lt font gt on page A 72 Specify which bitmaps to use For more information refer to Bitmaps lt bitmap gt on page A 80 Specify how to create animations and LED indicator patterns For more information refer to Indicators lt ind gt on page A 80 Instant Messaging 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
380. tor 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 650 670 x 1 34 IP 4000 x 1 IP 6000 x 1 IP 7000 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 17 dialplan x apply ToDirectoryDial 0 1 0 When present and if dialplan x digitmap is not Null this attribute overrides the global dial plan defin
381. ts 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 SoundPoint IP 601 and 34 registrations on the SoundPoint IP 650 and 670 The SoundStation IP 4000 6000 and 7000 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 which 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 o
382. tted Values Default Interpretation ind idleDisplay mode ind idleDisplay enabled 1 default 2 Null The idle display animation screen layouts 3 For example for the SoundPoint IP 330 320 Ifsetto 1 or Null the idle display animation size is 87 x 11 pixels Ifsetto 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 0 1 0 If set to 1 the idle display may support presentation of a custom animation if configured in the animation section of sip cfg Note 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 and lt IP_7000 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 or IP_7000 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 550 560 600 601 650 and 670 phones IP_4000 parameters affect SoundStation IP 4000 and 6000 phones and IP_7000 parameters affect SoundStation IP 7000 phones As of SIP 2 2 0 a maximum of 24 frames per animation is supported Attribute Permitt
383. ttp www polycom com usa en support voice soundpoint_ip VoIP_Technical_Bulle tins_pub html 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 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 encry
384. ugh a SIP header parameter is available for a SIP call it is indicated by an icon that appears after the call appearance status text regardless of the call state In the examples shown below a lightning bolt symbol is used to indicate that web content is available for the displayed call appearance and the user is encouraged to press the Select key to retrieve and display the content through the Microbrowser SoundPoint IP 330 Graphic Display From James Deon 7 End Call SoundPoint IP 550 Graphic Display 6721 From Greg Slowski 6721 6722 01 16 m 6723a m 67234 FromSandra Lee Web Content Retrieval Web content is retrieved either spontaneously active mode or at the request of the user passive mode e Active Mode Two methods can be used to achieve spontaneous web content retrieval static configuration parameters or parameters received as part of the SIP signaling If parameters received in the SIP signaling conflict with the static configuration the parameters in the SIP signaling will take precedence If the phone is configured to spontaneously retrieve web content the phone will launch the interactive Microbrowser and have it fetch the Configuring Your System appropriate URL upon arrival of the appropriate SIP signaling subject to some conditions described below Since new web content URLs can be received at any time as the first URL for a call or a replacement URL rules are needed to match
385. uide SoundPoint IP SoundStation IP Call progress pattern number 7 Use within phone 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 9 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 Note Configuration Files Ringer pattern number Default 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 30
386. undStation IP 4000 6000 and 7000 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 Configuring Your System 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 4 Configuration changes can performed centrally at the boot server Central boot server Configuration File Specify the phone specific contact directory and the sip cfg phone specific configuration override file For more information refer to Encryption lt encryption gt on page A 89 Configuration file Change the encryption key lt device gt cfg e For more information refer to refer to Flash Parameter Configuration on page A 124 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 63 The phone s local user interface also permits many application settings to be modified su
387. ute Values Default Interpretation roaming_privacy reg positive Null 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 User Preferences lt user_preferences gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation up analogHeadsetOption 0 1 2 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 electronic hookswitch If set to 2 a Plantronics compatible headset is attached and can be used an electronic hookswitch up offHookAction none 0 1 Null If set to 0 or Null the behavior introduced in SIP 2 1 2 occurs When users go off hook the phone tries to seize a line Which line is seized depends on volpProt SIP strictLineSeize voIPProt SIP lineSeize retries and reg x strictLineSeize If set to 1 the behavior from SIP 1 6 7 occurs When users go off hook the phone does not seize a line or answer a ringing call The user must use the line keys to either make a new call or answer a ringing call This will apply under all ringer settings not just SilentRing A 123 Administrator s Guide SoundPoint IP Sou
388. ution of the action string fails P lt prompt num gt N lt num digits gt The user input prompt string Refer to Prompt Macro Substitution on this page S lt speed dial index gt The speed dial index Only digits are valid The action is found in the contact field of the local directory entry pointed to by the index F lt internal function gt An internal function For more information refer to Internal Key Functions on page C 19 URL A URL Only one per action string is supported Prompt Macro Substitution The action string in the efklist element can be defined by a macro substitution string PnNn where e Pnis the prompt x as defined in the efk efkprompt x e Nnis the number of digits or letters that the user can enter The maximum number is 32 The user needs to press the Enter soft key to complete data entry Note If the maximum number of characters is greater than 32 or less than one macro execution fails Administrator s Guide SoundPoint IP SoundStation IP Note The macros provide a generic and easy to manage way to define the prompt to be displayed to the user the maximum number of characters that the user can input and action that the phone performs once all user input has been collected The macros are case sensitive If a macro attempts to use a prompt that is disabled the macro execution fails A prompt is not required for every macro Expanded Ma
389. values that override those in sip cfg e For more information refer to Registration lt reg gt on page A 107 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 Administrator s Guide SoundPoint IP SoundStation IP Note Warning SRV records will be tried before falling back on A records if NAPTR and SRV records return no results If no port is given 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
390. ver certain aspects of this feature work may not work as described below we POLYCOM Note Configuring Your System 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 In the SIP 3 1 release the BLF feature is updated for the following e Visual indication when a remote line is in an alerting state e Display of the caller ID of calls on remotely monitored lines e Single button Directed Call Pickup on a remote line For more information refer to Quick Tip 37381 Enhanced BLF at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Polycom recommends that the BLF not be used in conjunction with the Microsoft Live Communications Server 2005 feature For more information refer to Microsoft Live Communications Server 2005 Integration on page 4 61 Use this feature with TCPpreferred transport refer to Server lt server gt on page A 7 You can also use UDP transport on SoundPoint IP 650 and 670 phones Configuration changes can performed centrally at the boot server Central boot server Configuration file Specify the list SIP URI and index of the registration which will be phonet cfg used to send a SUBSCRIBE to the list SIP URI specified in attendant uri For more informa
391. vices 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 Administrator s Guide SoundPoint IP SoundStation IP Presence Note Note 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 serve 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 an
392. week in the month e Start DST on the second 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 tcplpApp sntp address overrideDHCP tcplpApp sntp gmtOffset Permitted Values 0 1 positive or negative integer Default 0 28800 Pacific time Interpretation 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 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 t
393. 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 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
394. wing fields Name Interpretation L lt label gt This is the label for the entire operation The value can be any string including the null string in this case no label appears This label will be used if no other operation label collection method worked up to the point where this field is introduced Make this the first entry in action string to be sure this label is used otherwise another label may be used and this one ignored digits The digits to be sent The appearance of this this parameter depends on the action string C lt command gt This is the command It can appear anywhere in the action string Supported commands or shortcuts include hangup hu hold h e waitconnect wc pause lt number of seconds gt p lt num sec gt where the maximum value is 10 Configuring Your System Name Interpretation T lt type gt The embedded action type Multiple actions can be defined Supported action types include invite dtmf refer Note Polycom recommends that you always define this field If it is not defined the supplied digits will be dialed using INVITE if no active call or DTMF if an active call The use of refer method is call server dependentand may require the addition of star codes M lt macro gt The embedded macro The lt macro gt string must begin with a letter If the macro name is not defined the exec
395. y dotted decimal IP address Null IP address or host name and address 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 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 sos 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 on page A 118 that will be used for emergency routing Messaging lt msg gt Message waiting indication is supported on
396. ycom 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 and 670 Volume Volumet Mute and Messages e IP 4000 and 6000 Volume and Select e IP 7000 Volume and Volume Note Miscellaneous Administrative Tasks As of SIP 3 1 users can restart their phones by pressing the Menu key and then selecting Settings gt Basic gt Restart Phone Any new bootROM and SIP applications will be downloaded to the phone as a result of this restart Restoring Factory Defaults For the key combination press and hold certain key combinations depending on the phone model simultaneously during the countdown process in the bootROM until the password prompt appears e IP 301 501 550 600 601 and 650 and 670 4 6 8 and dial pad keys e IP 320 330 and 430 560 7000 1 3 5 and 7 dial pad keys IP 4000 and 6000 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 ho

Polycom SIP 3.1 Telephone User Manual

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