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SIP 3.2 Administrator`s Guide

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1. Permitted Attribute Values Default Interpretation httpd enabled Oor1 1 If set to 1 the HTTP server will be enabled httpd lp port 1 65535 80 Port is 80 for HTTP servers Care should be taken when choosing an alternate port Note This feature is supported on the Polycom VVX 1500 only This attribute also includes Configuration lt cfg gt Configuration lt cfg gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation httpd cfg enabled Oori 1 If set to 1 the HTTP server configuration interface will be enabled httpd cfg port 1 65535 80 Port is 80 for HTTP servers Care should be taken when choosing an alternate port Call Handling Configuration lt call gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation call rejectBusyOnDnd Oori 1 If set to 1 reject all incoming calls with the reason busy if do not disturb is enabled Note This attribute is ignored when the line is configured as shared The reason being that even though one party has 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 Oor1 If set to 1 calls will be allowed when the phone is not successfully registered oth
2. 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 Line3 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 T 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 550 560 650 670 Miscellaneous Administrative Tasks 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 Dialpad5 31 Line6 42 Line5 10 MicMute 21 Dialpad6 32 Conference 11 VolUp 22 Dialpad3 33 Line2 15 Administrator s Guide SoundPoint IP SoundStation IP VVX SoundStation
3. Attribute Permitted Values Default Interpretation attendant resourceList x type normal or normal Type of resource being monitored automata If set to normal the default action when pressing the line key adjacent to this monitored user is to initiate a call if the user is idle or busy and to perform a directed call pickup if the user is ringing Any active calls are first placed on hold If set to automata the default action when pressing the line key adjacent to this monitored user is to perform a park blind transfer of any currently active call If there is no active call and the monitored user is ringing busy an attempt to perform a directed call pickup park retrieval is made Behaviors lt behaviors gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation attendant behaviors display Oor 1 1 A flag to determine whether or spontaneousCallAppearances normal not a call appearance is spontaneously presented to the attendant behaviors display Oor1 0 attendant when calls are alerting spontaneousCallAppearances automata on a monitored resource The information displayed after a press and hold of a resource s line key is unchanged by this parameter If set to 1 the display is enabled attendant behaviors display Oor 1 1 A flag to determine whether or remoteCallerlID normal not remote party caller ID information is presente
4. 0 cece eee eee C 22 Parsing Vendor ID Information 6 000 cee eee eee C 23 Product Model and Part Number Mapping C 25 Disabling PC Ethernet Port 0 0 0 eee eee eee C 26 Modifying Phone s Configuration Using the Web Interface C 26 Capturing Phone s Current Screen 00 eee renren eee C 29 Contents LLDP and Supported TLVs 0 eee cece eee eee C 29 Supported TLVS cs tcsisteesade des needs teh PEREOS EAD EEEE C 31 D Third Party Software 2 2 ce cece cece cece ee D 1 Index 666 02 0d bse Sacanssee as eund hterssne secon xi Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family xii Introducing the SoundPoint IP SoundStation IP VVX Family This chapter introduces the SoundPoint IP SoundStation IP VVX family which is supported by the software described in this guide The SoundPoint IP SoundStation IP VVX 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 VVX family supports advanced functionality including multiple call and flexible line appearances HTTPS secure provisioning presence custom ring tones and local conferencing The SoundPo
5. Permitted Attribute Values Default Interpretation acd reg 1 to 34 1 The registration index used to support BroadSoft server based ACD If set to Null line 1 is used acd stateAtSignIn Oori 1 The state of the user when signing in If set to 1 or Null the user is available If set to 0 the user is unavailable A 150 Configuration Files Flash Parameter Configuration 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 provisioning 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
6. If set to1 and the phone is indicating a ringing inbound call appearance phone will transmit a 486 response to the received INVITE when the Reject soft key is pressed If set to 0 no 486 response is transmitted This attribute also includes SDP lt SDP gt 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 Dialog lt dialog gt Connection Reuse lt dialog gt Music on Hold lt musicOnHold gt Compliance lt compliance gt A 15 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones SDP lt SDP gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation volpProt SDP useLegacyPayloadTypeNegotiation Oori Null If set to 1 the phone transmits and receives RTP using the payload type identified by the first codec listed in the SDP of the codec negotiation answer If set to 0 or Null RFC 3264 is followed for transmit and receive RTP payload type values 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 volpProt SDP iLBC 13_33kbps includeMode Oori Null If set to 1 or Null the
7. Defines the font file that will be loaded from provisioning server during boot up Note When several font IP_400 x name are defined the index x must follow consecutive increasing order A 88 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 XxxFF fnt Null Defines the font file that will be loaded from provisioning 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 _UxxFF fnt Default Null Interpretation Defines the font file that will be loaded from provisioning server during boot up Note When several font P_600 x name are defined the index x must follow consecutive increasing order Keys lt key gt Note Use of this parameter is not supported for the Polycom VVX 1500 phone 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 u
8. 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 Administrator s Guide SoundPoint IP SoundStation IP VVX Product Name Model Name Product Part Number SoundPoint IP 670 SPIP670 2345 12670 001 SoundStation IP 4000 SSIP4000 2201 06642 001 SoundStation IP 6000 SSIP6000 3111 15600 001 SoundStation IP 7000 SSIP7000 3111 40000 001 Polycom VVX 1500 VVX1500 2345 17960 001 Disabling PC Ethernet Port Certain SoundPoint IP phones have a PC Ethernet port If it is unused it can be disabled The PC Ethernet port can be disabled on the SoundPoint IP 33x 430 450 550 560 601 650 and 670 and Polycom VVX 1500 through the menu shown below The Ethernet port can also be disabled through the configuration files To disable the Ethernet port on a supported SoundPoint IP phone 1 2 Se Modifying Phone Press A 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
9. Valid DVD string present in DHCP option Yes Release DHCP address For each VLAN listed in DVD string max 10 esponse to DHCP Discover on VLAN X received Yes Boot process continues with VLAN tag assigned Boot process continues without any VLAN assigned Parsing Vendor ID Information More VLANs in DVD string No Phone Reboots After the phone boots 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 Administrator s Guide SoundPoint IP SoundStation IP VVX C 24 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 fie
10. 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 platform prov fileSystem rfs0 minFre eSpace is internally replaced by 4X the value prov polling enabled Oor 1 0 If set to 1 automatic periodic provisioning 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 prov quickSetup enabled Oor1 Null If set to 1 the quick setup feature is enabled If set to 0 or Null the quick setup feature is disabled A 108 Configuration Files RAM Disk lt ramdisk gt This attribute s settings control the phone s internal RAM disk feature Polycom recommends that you do not change these values POLYCOM This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation ramdisk enable Oor1 1 If set to 1 RAM disk will be available The RAM disk is used to cach
11. SoundPoint IP 32x 33x 430 450 fontProp_10 This is the font used widely in the current implementation fontPropSoftkey_10 This is the soft key specific font SoundPoint IP 550 560 650 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 range 100 17F the name could be fontCzechIP500_10_U0100_U01FF fnt and fontCzechIP600_1
12. 0O1 Prm Configuration file O1 Prm Configuration file O1 Prm Configuration file a E TEL file EE A 19A Success provisioning aa baam oa Aa aan F f Nea gg AN Tathil Easy error tren S043 db tenting stat ot 2307 local O60 Adu iy Q1 Initial log entry Current logging level 4 4 01 ldap Not Enabled 4 01 cDynamicData cDynamicData cDynamicData Failed O1 Initial log entry Current logging level 4 s 01 So0oNcasC App Ctx 6045551234 0 6045551234 O01 phonel cfg is from template phonel c O01 phonel cfg SHA1 digest B712DCCA395E 001 sip cfg is from template sip cfg r O01 sip cfg SHA1 digest B4E4534529797EC ee ee psizisc is etg 0522120608 ldap 0522120608 ldap 0522120608 ldap 0522120608 efk 0522120608 so 0522120608 sip 0522120608 appi 0522120608 sip 0522120608 cfg a ee slog 5 4 Q1 NAPTR query for host as test returned no results O1 InitializeBacklightIntensity m_nDefaultMin 0 m_nDefaultLow 4 01 Registration failed User 6045551234 Error Code 404 Not Found 4 O01 Edit Error Ox380003 attempting stat of ffs0 local O0004 21db094 p OY ae A male aa Ni OY a Ae Administrator s Guide SoundPoint IP SoundStation IP VVX Reading a Syslog The following shows a portion of a syslog log file the messages look identical to the normal log with the addition of a timestamp and IP address Jan 00 00 00 172 23 7 249 0100000000 so
13. up pictureFrame 3 to 300 Null The time to display the image timePerlmage seconds If set to Null the default time is 5 seconds A 149 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default up screenSaver enabled Oor1 0 Interpretation If set to 1 a USB flash drive is attached to the phone and the idle browser is not configured a slide show will cycle through the images from the USB flash drive The images must stored in the appropriate directory of the USB flash drive up pictureFrame folder in phonet cfg The slide show does not appear when the phone is in the active state If set to 1 but there is no USB flash drive attached to the phone there is not change on the screen However the screen saver will start working once a USB flash drive is attached If set to 0 the feature is disabled Note If the idle browser is also enabled the idle browser is displayed until the screen saver times out then the screen saver appears When the screen saver exits the idle browser is displayed again and is up to date it is refreshed in the background up screenSaver waitTime 1 to 9999 Null The time to wait In minutes in the idle state until the screen saver starts If set to Null the default time is 15 minutes Automatic Call Distribution lt acd gt This configuration attribute is defined as follows
14. Ringer pattern number Default description 18 Sampled audio file 6 19 Sampled audio file 7 20 Sampled audio file 8 21 Sampled audio file 9 22 Sampled audio file 10 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 34 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 for the SoundPoint IP SoundStation IP VVX Phones 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 aut
15. Permitted Attribute Values Default Interpretation volpProt SIP dialog useSDP Oor1 0 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 dialog usePvalue Oor1 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 Connection Reuse lt dialog gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP connectionReuse Oori 0 If set to 0 this is the old behavior useAlias If set to 1 phone uses the connection reuse draft which introduces alias 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 A 20 Compliance lt compliance gt Configuration Files This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP compli
16. 4 O00fServer 172 23 2 92 said bootrom ld is not present 4 O00fCould not get all 512 bytes of the header 3 Q0fbootROM file not prenent on boot server 13 00 tp plemspip PITE er ae 4 00 Download of O004f21db094 cfg FAILED on attempt 1 4 00 Server 172 23 2 92 said OO04f21db094 cfg is not present 3 00 Update of ffs0 init mac failed leaving local copy intact 3 O00 ftp plemspip 172 23 2 92 000000000000 cfg from 172 23 2 3 00 Download of OO0000000000 cfg succeeded on attempt 1 addr 1 of addr 1 of 1 Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones Reading an Application Log The following figure shows portions of an application log file 522184554 log 0522184554 so 0522184554 so 0522184554 so 0522184554 so 0522184554 so 0522184554 so 0522184554 so 0522184554 so O1 Initial log entry Current logging level 4 O1 Initial log entry Current logging level 3 Initial log entry O1 Platform Model SoundPoint IP 450 Assembly 2345 12450 001 Rev 3 01 Platform Mac 0004f21db094 IP 172 23 61 141 Subnet Mask 255 255 01 Platform BootBlock 2 8 1 12450 001 04 Jun 08 17 04 O01 Platform Bootrom 4 1 2 0009 20 Jul 08 21 57 01 Application main Label SIP Version 3 1 3 0439 26 Apr 09 23 52 O1 Application main P N 3150 11530 313 0522164554 wdog 01 Initial log entry Current logging level 4 05221684554 ethf 01 Initial log entry Curr
17. 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 provisioning server Use this procedure as a recommendation if this is your first provisioning server setup 1 Install a provisioning server application or locate suitable existing server s Polycom recommends that you use RFC compliant servers 2 Create an 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 multiple connections to the server The phone will attempt to upload log files a configuration override file and a directory file to the server This requires that the phone s account has 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 for the SoundPoint IP Sou
18. LLDP MED Extended Power Via MDI TLV Power Type Power Source Power Priority Power Value e Optional Port Description System Name Admininstrator assigned name System Description Includes device type phone number hardware version and software version System Capabilities Set as Telephone capability MAC PHY config status Detects duplex mismatch Management Address Used for network discovery LLDP MED Location Identification Location data formats Co ordinate Civic Address ECS ELIN LLDP MED Inventory Management Hardware Revision Firmware Revision Software Revision Serial Number Manufacturer s Name Model Name Asset ID An LLDP frame shall contain all mandatory TLVs The frame will be recognized as LLDP only if it contains mandatory TLVs SoundPoint IP SoundStation IP VVX phones will support LLDP frames with both mandatory and optional TLVs The basic structure of an LLDP frame and a table containing all TLVs along with each field is explained in Supported TLVs on page C 31 Note Supported TLVs Miscellaneous Administrative Tasks As per section 10 2 4 4 of the LLDP MED standard LLDP MED endpoint devices need to transmit Location Identification TLVs if they are capable of either automatically determining their physical location by use of GPS or radio beacon or capable of being statically configured with this information At present the SoundPoint IP SoundStation IP VVX phone
19. The absolute minimum duration time in milliseconds of RTP packet Rx with no packet loss between jitter buffer size shrinks Use smaller values 33 ms to minimize the delay on known good networks Use larger values 1000ms to minimize packet loss on networks with large jitter 3000 ms video profile H264 payloadType 96 to 127 RTP payload format type for H264 90000 default 109 MIME type video profile H264 profileLevel 1 1b 1 1 1 2 This value is H 264 s level used in the phone 1 3 default The Level is a constraint set to selected key algorithm parameters codec in different level has different ability at this time Polycom VVX 1500 support these level 1 1b 1 1 1 2 1 3 as to detailed level definition For more information refer to ITU T H 264 video profile H263 jitterBufferMax video profil e H263 jitter BufferMin 500ms to 2500ms default 2000ms 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 for the SoundPoint IP SoundStation IP VVX Phones Attribute video profile H263 jitterBufferMin Permitted Values 33ms to 1000ms default 150ms Interpretation The smallest jitter buffer depth in milliseconds that must be achieved before play out begins for the first tim
20. The phone configuration files consist of e Master Configuration Files e Application Configuration Files e Override Files This section also contains information on e Central Provisioning e Manual Configuration 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 for the SoundPoint IP SoundStation IP VVX Family Application Configuration Files Typically the files are arranged in the following manner although parameters may be moved around within the files and the filenames themselves can be changed as needed These files dictate the behavior of the phone once it is running the executable specified in the master configuration file The application files are e Application It contains parameters that affect the basic operation of the phone such as voice codecs gains and tones and the IP address of an application server All phones in an installation usually share this category of files Polycom 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 un
21. 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 Oor1 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 29 lf mb main idleTimeout and up idleTimeout are Null the timeout is 40 seconds If set to value greater than 0 and less than 600 the timeout is for that number of seconds A 115 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Permitted Values Default Interpretation mb main autoBackKey Oor 1 1 If set to 1 the phone will automatically supply a Back soft key in all main browser screens which if pressed
22. 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 Configuring Your System c 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 d 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 Download the file to the provisioning server e Save the modified sip cfg configuration file Automatic Off Hook Call Placement The phone supports an optional automatic off hook call placement feature for each registration This feature is sometimes referred to as hot dialing Configuration changes can be performed centrally at the provisioning server Central Configuration file Specify which registrations have the feature and what contact to call provisioning phonet cfg when going off hook server For more inform
23. APP_FILE_PATH_SPIP600 sip_313 1d APP_FILE_PATH_SPIP501 sip_313 1d APP_FILE_PATH_SPIP301 sip_313 1d APP_FILE_PATH_SSIP4000 sip_313 1d CONFIG_FILES PHONE_MAC_ADDRESS user cfg phonel cfg sip cfg CONFIG_FILES_SPIP500 PHONE_MAC_ADDRESS user cfg _S _S 5 BEN _S ES CONFIG_FILES_SPIP300 PHONE_MAC_ADDRESS user cfg CONFIG_FILES_SPIP601 PHONE_MAC_ADDRESS user cfg CONFIG_FILES_SPIP600 PHONE_MAC_ADDRESS user cfg CONFIG_FILES_SPIP501 PHONE_MAC_ADDRESS user cfg CONFIG_FILES_SPIP301 PHONE_MAC_ADDRESS user cfg phonel_313 cfg sip_313 cfg CONFIG_FILES_SSIP4000 PHONE_MAC_ADDRESS user cfg phonel_313 cfg sip_313 cfg MISC_FILES LOG_FILE_DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY gt Note Setting up Your System 6 Remove any lt MACaddress gt cfg files that may have been used with earlier releases from the provisioning 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 MACaddress 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 MACaddress gt cfg files for Sound
24. End Call Trnsfer Confrne 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 CalPak e Park Enter Number Enter Cancel Cancel b Enter the number where you want to park the active call then select the Enter soft key The Call Park code 68 is prepended to the number you entered and the call is parked at that location by the call server The active call is put on hold during this operation 4 48 Configuring Your System To 6849000 684000 eae Hold Ross Dutkiewicz Call Pickup BPP 2 01 Mon Dec 17 9 52 AM End Call Trnsfer More Hold 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 e Directly into the Enhanced Feature Key macro e Directly into a URL e A chained list of actions It is possible to disab
25. Match the name of the corresponding lt fileName gt bmp to be retrieved from the provisioning server Indicators lt ind gt The following indicators are used by the phone e Animations lt anim gt lt IP_330 gt lt IP_400 gt lt IP_450 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation ind idleDisplay mode 1 default 2 Null The idle display animation screen layouts 3 For example for the SoundPoint IP 330 320 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 ind idleDisplay enabled Oor1 0 If set to 1 the idle display may support presentation of a custom animation if configured in the animation section of sip cfg Animations lt anim gt lt IP_330 gt lt IP_400 gt lt IP_450 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_330 IP_400 IP_450 IP_600 IP_4000 or IP_7000 y is the animation number z is the step in the animation Note that
26. 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family e Missed Call Notification 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 e Time and Date Display e Idle Display Animation 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 infor
27. SoundStation IP VVX 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 provisioning server This error is typically a result of one of the above errors This error is fatal but recoverable Contact your system administrator 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 provisioning 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 The probable cause of this error is a misconfiguration of the lt MACaddress gt cfg file Note This error does not occur with BootROM 3 2 2 B or later Error loading lt file name gt When the required file does not exist in flash memory and cannot be found on the provisioning 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 BootROM 3 2 2 B or later Application Error Messages Config file error Error is lt Hex gt If there is an error in the configuration file you will not be ab
28. 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 Ringer 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 33 and sampled audio sound effects Sound effects are defined by patterns rudimentary sequences of chord sets silence periods and wave files Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation se stutterOnVoiceMail Oori 1 If set to 1 stuttered dial tone is used i
29. 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Name The following syslog configuration parameters can be modified on the Syslog menu Possible Values Description Server Address dotted 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 Oto6 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 101 Note Use left and right arrow keys to change values Prepend
30. tcplpApp sntp gmtOffset Permitted Values Oori 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 Oor1 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 Oor1 If set to 1 apply daylight savings rules to displayed time tcplpApp sntp daylightSavings fixedDayEnable Oor1 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 72 Configuration Files Attribute tcplpApp snip daylightSavings start date Permitted Values 1 31 Default 8 Interpretation If
31. voice txEq hf IP_650 postFilter enable Configuration Files Attribute Default voice txEq hf IP_6000 postFilter enable 0 voice txEq hf IP_7000 postFilter enable 0 voice txEq hf VVX_1500 postFilter enable 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 Oor1 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 Oor1 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 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Quality Monitoring lt quality monitoring gt This attribute include
32. volpProt SIP keepalive sessionTimers Permitted Values Oor1 Default 0 Interpretation 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 Oor1 If set to 1 global prefix is added to E 164 user parts in sip URIs volpProt SIP allowTransferOnProceeding Oor1 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 pingInterval 0 to 3600 The number in seconds to send PING message This feature is disabled by default volpProt SIP useContactInReferTo Oor1 If set to 1 the Contact URI is used If set to 0 the TO URI is used previous behavior volpProt SIP serverFeatureControl cf Oor1 Null If set to 1 server based call forwarding is enabled The call server has control of call forwarding If set to 0 or Null server based call forwarding is not enabled This is the old behavior volpProt SIP serverFeatureControl loc alProcessing cf Oor1 Null If set to
33. 421 Extension Required No 423 Interval Too Brief No 480 Temporarily Unavailable Yes 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 Session Initiation Protocol SIP 6xx Responses Global Failure Response Supported Notes 600 Busy Everywhere No 603 Decline Yes 604 Does Not Exist Anywhere No 606 Not Acceptable No 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 11 but both methods are supported when signaled by the remote end p
34. 650 and 670 and SoundStation IP 6000 and 7000 e Video Codecs Support the standard video codecs on the Polycom VVX 1500 phones Note Overview e Mutual TLS Authentication Support for phone authentication of the server and server authentication of the phone e Digital Certificates Support for digital certificates and associated private keys on certain models of SoundPoint IP phones e Capturing Phone s Current Screen Allows the phone s current display to be displayed in a web browser The following existing features were changed in SIP 3 2 e Busy Lamp Field The BLF feature has been enhanced as follows To provide individual subscription based BLF monitoring without requiring a centralized resource list to be maintained by the call server Toallow the single button remote pick up feature to be implemented using Directed Call Pick Up using SIP signalling as well as the star code method supported in SIP 3 1 e Secure Real Time Transport Protocol Information has been transferred from the Technical Bulletin 25751 Secure Real Time Transport Protocol on SoundPoint IP Phones to this guide Documentation of the newly released SoundPoint IP 321 331 and 450 desktop phones and Polycom VVX 1500 business media phone has also been added When SoundPoint IP 32x 33x is used in this guide it includes the SoundPoint IP 320 321 330 and 331 phones Administrator s Guide for the SoundPoint
35. Expansion Module s are detached from the phone 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 specific 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
36. For example for LOG CONTACTS and OVERRIDES allow full access read and write and for all others read only access For more information on LOG_FILE_DIRECTORY OVERRIDES CONTACTS and LICENSE 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 Setting up Your System Deploying Phones From the Provisioning Server Note You can successfully deploy SoundPoint IP SoundStation IP VVX phones from one or more provisioning servers For all SoundPoint IP SoundStation IP VVX phones follow the normal provisioning process in the next section Provisioning Phones However if you have decided to daisy chain two SoundStation IP 7000 conference phones together read the information in Provisioning SoundStation IP 7000 Phones Using C Link on page 3 20 to understand the different provisioning options available Provisioning Phones The default configuration files will work without any changes however if you change
37. In the configuration files bridged lines are configured by shared line parameters Central provisioning server Local Configuring Your System Configuration changes can be performed centrally at the provisioning server or locally Configuration file sip cfg Specify whether diversion should be disabled on shared lines For more information refer to Call Handling Configuration lt call gt on page A 76 Configuration file phonet cfg Web Server if enabled Specify per registration line type private or shared and the shared line third party name A shared line will subscribe to a server providing call state information For more information refer to Registration lt reg gt on page A 128 Specify per registration whether diversion should be disabled on shared lines For more information refer to Diversion lt divert gt on page A 136 Specify per registration line type private or shared and third party name and whether diversion should be disabled on shared lines Navigate to http lt phonelPAddress gt reg htm Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the provisioning 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 provisioning server Local Phone User Interface Specify per regist
38. 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 limited 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 Departme
39. SoundStation IP VVX Family H handset headset and speakerphone 4 9 hands free disabled A 31 hold lt hold gt A 80 I idle display lt idleDisplay gt A 114 idle display animation 4 16 iLBC 4 81 incoming signaling validation 4 89 indicator classes lt class gt A 97 indicators A 95 assignments A 97 installing SIP application 3 17 instant messaging 4 29 IP TOS A 68 IP TOS call control lt callControl gt A 70 IP_400 font A 88 IP_500 font A 89 IP_600 font A 89 J jitter buffer 4 79 K keep alive lt keepalive gt A 75 key features 1 6 keys lt key gt A 89 L languages adding new A 27 languages supported 4 29 last call return 4 25 LDAP directory virtual list view support A 85 LEDs A 98 length lt length gt A 104 link layer discovery protocol C 29 LLDP See also link layer discovery protocol local centralized conferencing 4 21 local lt local gt A 7 local camera view lt localCameraView gt A 67 local contact directory 4 10 local contact directory file format 4 11 local digit map 4 13 local reminder lt localReminder gt A 80 local user and administrator privilege levels 4 88 localization lt Icl gt A 25 Index 4 log files 5 5 logging lt log gt A 99 low delay audio packet transmission 4 78 M MAC address definition A 2 substitution 3 17 3 18 3 25 A 4 main browser lt main gt A 115 main menu 3 7 manage conferences 4 22 manual configuration overview 2 7 manual lo
40. call hold 4 19 call log 4 3 call park retrieve 4 24 call progress patterns A 37 call progress tones synthesized 4 30 call timer 4 3 call transfer 4 20 call waiting 4 3 called party identification 4 4 calling party identification 4 4 calls lt calls gt A 133 camera controls lt camera gt A 66 central provisioning overview 2 6 changing the key on the phone C 5 chord sets lt chord gt A 33 comfort noise fill 4 83 compliance lt compliance gt A 21 conference setup lt conference gt A 19 configurable feature keys 4 26 configurable soft keys 4 49 configuration file encryption 4 90 configuration file example 4 66 connected party identification 4 5 connection reuse lt connectionReuse gt A 20 consultative transfers 4 20 context sensitive volume control 4 5 corporate directory 4 35 corporate directory feature A 83 A 111 custom certificates 4 89 customizable audio sound effects 4 6 customizable fonts and indicators 4 28 D date and time lt datetime gt A 29 default feature key layouts C 12 default password 3 6 4 95 C 11 C 26 deploying phones from the boot server 3 17 device lt device gt A 151 DHCP secondary server 3 3 DHCP INFORM 3 3 3 8 3 9 DHCP menu 3 8 DHCP or manual TCP IP setup 3 2 diagnostics phone 5 10 dial plan lt dialplan gt A 21 dialog lt dialog gt A 20 digit map default A 24 examples A 23 match and replace A 23 protocol A 23 timer A 23 digit map lt digitmap gt A 141 d
41. ce cece cece ee ARI Setting Up Basic Features everr iirin tia aana E eee 4 1 Call Log a Ree ae Ree ee 4 3 Call MMe s4ctjacabasnetsim Gu seis etait steesa Tee ey 4 3 Call Waiting oeenn tone baa cee ails gntniwon EE ok ahem ale abe 4 3 Called Party Identification 0 0 eee ee eee 4 4 Calling Party Identification 00 0 4 4 Missed Call Notification 0000 000 00088 4 5 Connected Party Identification 0 0 eee eee eee 4 5 Context Sensitive Volume Control 0 0 00 00 eee eee 4 5 Customizable Audio Sound Effects 0 0 000 e eee 4 6 Message Waiting Indication 000 0000 4 7 Distinctive Incoming Call Treatment 0000008 4 7 Distinctive Ringing i oc serpe ekant keia Te pee a bie eta obey 4 7 Distinctive Call Waiting n 0 c cece 4 8 Do NOt Distut cic bin cpa cnrgs cee g sw ninan ENEE EEE AES 4 8 Handset Headset and Speakerphone 000 4 9 Local Contact Directory 00000 4 10 Local Digit Map eri c os iseg tae hte e es thiet n ese nena aA 4 13 Microphone Mute 0 0 eee 4 14 Soft Key Activated User Interface 000 4 14 Speed Dial pornn a sie op aie eG had agee en Sh bet bie Ewa iies 4 15 Time and Date Display 6 0 c cece eee eens 4 15 Idle Display Animation 0 0000 e eee eee eee 4 16 Ethernet Switeh oseere tetanic
42. digit map A 141 do not disturb A 134 A 138 forward all A 137 message waiting indicator A 143 messaging A 143 missed call configuration A 134 Network Address Translation A 144 no answer A 138 quotas A 113 registration A 128 resource list A 146 roaming buddies A 148 roaming privacy A 148 routing A 141 routing server A 142 per phone configuration file A 127 phone diagnostics 5 10 phone quick setup 4 77 phonel cfg A 127 Polycom VVX 1500 application launch pad 4 32 power saving feature 4 52 port lt port gt A 74 power saving lt powerSaving gt A 126 presence 4 64 presence lt pres gt A 86 product model part number mapping C 25 protocol lt volpProt gt A 7 protocol server lt server gt A 8 protocol special events lt specialEvent gt A 19 Index provisioning lt prov gt A 108 provisioning protocols 3 4 provisioning protocols supported 3 4 Q QOS See also Quality of Service Quality of Service lt QOS gt A 67 quick setup feature 4 77 quotas lt quotas gt A 113 R RAM disk lt ramdisk gt A 109 rebooting phones 3 19 3 23 receive equalization lt rxEq gt A 54 registration lt reg gt A 128 reliability of provisional responses B 9 request lt request gt A 109 request delay lt delay gt A 110 request validation lt requestValidation gt A 18 resetting to factory defaults 3 6 resource lt res gt A 112 resource files overview 2 8 resource list lt resourceList gt A 146 RFC support B 2 ring type lt r
43. lt up gt e Automatic Call Distribution lt acd gt SoundPoint IP 32x 33x and 430 support a maximum of two unique registrations SoundPoint IP 450 supports three the SoundPoint IP 550 and 560 supports four and SoundPoint IP 650 and 670 and the Polycom VVX 1500 support six Up to three SoundPoint IP Expansion Modules can be added toa single host SoundPoint IP 650 and 670 phone increasing the total number of buttons to 34 registrations on the IP 650 and 670 Each registration can optionally be associated with a private array of servers for completely segregated signaling The SoundStation IP 6000 and 7000 supports a single registration In the following table x is the registration number IP 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Permitted Attribute Values Default Interpretation reg x csta Oor1 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 in SIP signaling as the string default caller ID 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 A 128
44. reflash For information on file format refer to the next section Local Contact Directory 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 provisioning server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the provisioning 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 1n gt Doe lt 1n gt lt fn gt John lt fn gt lt ct gt 1001 lt ct gt lt sd gt 1 lt sd gt lt lb gt Mr lt 1b gt lt rt gt 1 lt rt gt lt dc gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family lt ad gt 0 lt ad gt lt ar gt 0 lt ar gt lt bw gt 0 lt bw gt lt bb gt 0 lt bb gt lt item gt lt item gt lt ln gt Smith lt 1n gt lt fn gt Bill lt fn gt lt ct gt 1003 lt ct gt lt sd gt
45. 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 provisioning 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 provisioning server Local Phone User Use the SIP Configuration menu to specify whether or not to use RFC Interface 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 and 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 be performed centrally at the provisioning server Centr
46. 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 330 1 87 IP 400 1 102 IP 450 1 170 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 330 1 20 IP 400 1 23 IP 450 1 73 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 Warning 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 fF O PN Interpretation Debug only High detail event class Moderate detail event class Low detail event class Minor error graceful recovery Major error will eventually incapacitate the system Fatal error Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Each event in the log contains the following fields separated by the character e time or time date stamp e 1 5 character component identifier such as so e event class e cumulative log events missed due to excessive CPU load e free form text the event des
47. 0522183251 co 3 O00 tp plems Lp 8172 23 2 9 45 12450 001 sip 1d CRN ot 17 pus we a Seann DF ob 4 00 Initial log entry 4 00 Note that bootrom log times are in GMT 4 00 Initial log entry 3 00 Initial log entry 4 00 Initial log entry 14 00 Initial log entry 4 00 Initial log entry 3 O00 CDP is DISABLED 3 00 Platform Model SoundPoint IP 450 Assembly 2345 12450 001 Rev 3 3 00 Platform Board 2345 12450 001 2 3 00 Platform Mac 0004f21db094 IP Resolving Subnet Mask Resolving 3 00 Platform BootBlock 2 6 1 12450 001 04 Jun 08 17 04 3 00 application main Label BOOT Version 4 1 2 0009 20 Jul 06 21 5 3 O00 application main P N 3150 11069 412 14 00 Initial log entry 3 00 Link status is Net up Speed 100 full Duplex PC down 13 00 CDP received a response from a switch CDP enabled 3 00 Native VLAN Id is 1 3 00 No Auxiliary VLAN found ao RY ee meer te ee _ Boot Failure Messages A Mase The following figure shows a number of boot failure messages eo rd teens 3 00 Beginning to provision phone A 3 O00 ftp plemspip 172 23 2 92 2345 12450 001 bootrom 1d trom amp 4 00 Download of 2345 12450 001 bootrom ld FAILED on attempt 1 addr 2 14 00 Server 172 23 2 92 said 2345 12450 001 bootrom ld is not presen a cae ae se not ger all 512 bytes of the header 131 00 i ee ra ee 4 00 Dosnload of bootrom ld FAILED on atene 1 addr 1 of 1
48. 1500 FTP FTP FTP TFTP TFTP TFTP HTTP HTTP HTTP HTTPS HTTP HTTPS There are two types of FTP methods active and passive The SIP application is not compatible with active FTP Secure provisioning was implemented in a previous release Note Note Note Setting up Your System 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 Both digest and basic authentication are supported when using HTTP S for the SIP application Only digest authentication is supported when using HTTP by the BootROM If the Server Type is configured as HTTPS the BootROM will contact the same address and apply the same username and password to authentication challenges only the protocol used will be HTTP No SSL negotiation will take place so servers that do not allow unsecured HTTP connections will not be able to provision files For downloading 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 refer to Trusted Certificate Authority List on page C 1 and custom certificates can be added to the phone refer to Technical Bulletin 17877 Using Custom Certificates With SoundPoint IP SoundStation IP and E Phon
49. 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 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 Agr
50. 3 lt sd gt lt lb gt Dr lt 1b gt lt rt gt 3 lt rt gt lt dc gt lt ad gt 0 lt ad gt lt ar gt 0 lt ar gt lt bw gt 0 lt bw gt lt bb gt 0 lt bb gt lt item gt lt item_list gt lt directory gt Element Permitted Values Interpretation fn UTF 8 encoded string first name of up to 40 bytes Note In some cases this will be less than 40 characters due to UTF 8 s variable length encoding In UTF 8 encoded string last name of up to 40 bytes 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 Used by the phone to address a remote party in the same way that a user part of a SIP string of digits or a SIP URL are dialed manually by the user This URL or a string that element is also used to associate incoming callers with a particular constitutes a valid SIP directory entry URL Note This field cannot be null or duplicated sd Null 1 to 9999 speed dial index Associates a particular entry with a speed dial bin for one touch dialing or dialing from the speed dial menu Note On the SoundPoint IP 32x 33x and the SoundStation IP 6000 and 7000 the maximum speed dial index is 99 4 12 Configuring Your System 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 charact
51. 4 O1 HRRRHHRHRHRAEREREARRERRRRRRR RR RR RRR 4 01 Running memShow y 1401 status bytes blocks avg block max block A A OL eeseer nee a a 90522163048 slog 4101 cPrent z a P52 eee hog F eet ee a tenn alent mata Manual Log Upload 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 VVX Reading a Boot Log The following figure shows a portion of a boot log file 100000000 s0 0100000000 so 100000000 cfg 0100000000 copy o0100000000 hw 0100000000 etht 0522182911 wdog 0522182911 cap 0522182911 s0 0522182911 s0 0522182911 s0 0522182911 s0 0522182911 s0 0522182911 s0 0522182911 appl 0522182912 s0 0522182916 cdp 0522182916 cap 0522182916 cdp aaz cdi oy 0522183251 cfg 0522183251 copy 0522183251 copy 0522183251 copy 0522183251 cfg 0522183251 copy 0522183251 copy 0522183251 copy 0522183251 cfg 0522183251 cfg 0522183251 copy 0522183251 copy 0522183251 copy 0522183251 copy 0522183251 copy 0522183251 copy Sovzlbs oD appiy O0f WS Tomara 4 vandduver ply
52. 5 If set to 1 set precedence bits in the IP TOS field of the IP header or else don t set them Configuration Files Basic TCP IP lt TCP_IP gt This attribute includes e Network Monitoring lt netMon gt e Time Synchronization lt sntp gt e Port lt port gt e Keep Alive lt keepalive gt Network Monitoring lt netMon gt y Polycom recommends that you do not change these values POLYCOM This configuration attribute is defined as follows Permitted Attribute Values Default tcplpApp netMon enabled Oor1 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 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 for the SoundPoint IP SoundStation IP VVX Phones Attribute tcplpApp sntp address overrideDHCP
53. Disturb option on the Features menu on the SoundPoint IP 32x 33x 430 and 450 and SoundStation IP 5000 6000 and 7000 Note The LED on the Do Not Disturb key on the Polycom VVX 1500 is red when pressed or when server based DND is enabled Handset Headset and Speakerphone SoundPoint IP phones come standard with a handset and a dedicated connector is provided for a headset not supplied All Polycom phones are full duplex speakerphones The SoundPoint IP phones provide dedicated keys for convenient selection of either the speakerphone or headset All Polycom desktop phones can be configured to use the electronic hookswitch For more information refer to Technical Bulletin 35150 Using an Electronic Hookswitch with SoundPoint IP and Polycom VVX 1500 Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Enable or disable persistent headset mode provisioning sip cfg For more information refer to User Preferences lt up gt on page A 29 server Enable or disable hands free speakerphone mode For more information refer to User Preferences lt up gt on page A 29 Configuration file Specify whether or not the electronic hookswitch is enabled and what phonet cfg type of hea
54. Do one of the following is Down connected Check termination at the switch or hub furthest end of the cable from the phone e Check that the switch or hub is operational flashing link status lights or contact your system administrator e Press Menu followed by Status gt Network Scroll down to verify that the LAN is active Ping phone from another machine e Reboot the phone to attempt re registration to the call server refer to Rebooting the Phone on page C 10 Administrator s Guide SoundPoint IP SoundStation IP VVX Calling 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 e Check that the display is illuminated Make sure the LAN cable is inserted properly at the rear of the phone try unplugging and re inserting the cable e If using in line powering have your system administrator check that the switch is supplying power to the phone Do one of the following e 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 e Check configuration for gain levels The phone is not registered Contact your system ad
55. Do one of the following steps a Place all bootrom d files corresponding to BootROM release zip file onto the provisioning server b Ensure that all phones are running BootROM 4 0 0 or later code Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 2 phonel_214 cfg sip_214 cfg phonel_214 cfg sip_214 cfg phonel_313 cfg sip_313 cfg phonel_313 cfg sip_313 cfg phonel_313 cfg sip_313 cfg Copy sip Id sip cfg and phonel cfg from the SIP 3 2 0 or later release distribution onto the provisioning server These are the relevant files for all phones except the SoundPoint IP 300 301 500 501 600 601 and SoundStation IP 4000 phones Rename sip Id sip cfg and phonel cfg from the previous distribution to sip_21x ld sip_21x cfg and phonel1_21x cfg respectively on the provisioning server These are the relevant files for supporting the SoundPoint IP 300 and 500 phones Rename sip ld sip cfg and phone1 cfg from the previous distribution to sip_31y ld sip_31ly cfg and phone1_31y cfg respectively on the provisioning server These are the relevant files for supporting the SoundPoint IP 301 501 600 601 and SoundStation IP 4000 phones Modify the 000000000000 cfg file if required to match your configuration file structure For example lt APPLICATION APP_FILE_PATH sip 1d APP_FILE_PATH_SPIP500 sip_214 1d APP_FILE_PATH_SPIP300 sip_214 1d APP_FILE_PATH_SPIP601 sip_313 1d
56. 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 408 526 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
57. IP SoundStation IP VVX Phones Attribute No Answer lt noanswer gt The phone can automatically divert calls after a period of ringing Permitted Values Default Interpretation divert noanswer x enabled Oor1 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 55 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 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 Oori 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
58. IP 6000 GA Ge Gr OD 40 CX 733 PRENO CICA 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 16 Miscellaneous Administrative Tasks SoundStation IP 7000 TOQ cg A BOO A ai n l a Lig ay Luuga 1 Sal e E zs z 4 E gt EP 9 28 fol 23 E 5 DA E x a 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 17 Administrator s Guide SoundPoint IP SoundStation IP VVX Po
59. IP_330 parameters affect SoundPoint IP 32x 33x phones IP_400 parameters affect SoundPoint IP 430 phones IP_450 parameters affect SoundPoint IP 450 phones IP_600 parameters affect SoundPoint IP 550 560 600 601 650 and 670 phones IP_4000 parameters affect SoundStation IP 6000 phones and IP_7000 parameters affect SoundStation IP 7000 phones Note As of SIP 2 2 0 a maximum of 24 frames per animation is supported Attribute Permitted Values Interpretation ind anim x y frame z bitmap A bitmap name defined Bitmap to use previously Note that it must be defined already refer to Platform lt IP 330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt on page A 95 ind anim x y frame z duration positive integer Duration in milliseconds for this step O infinite A 96 Patterns lt pattern gt Configuration Files This section defines patterns for the LED indicators In the following table x is the pattern number y is the step in the pattern 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 cl
60. Label Function Notes Dialpado Dialpado Dialpad1 Dialpad1 Dialpad2 Dialpad2 Dialpad3 Dialpad3 Dialpad4 Dialpad4 Dialpad5 Dialpad5 Dialpad6 Dialpad6 Dialpad7 Dialpad7 Dialpad8 Dialpad8 Dialpadg Dialpadg DialpadPound DialpadPound 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 Headset Headset Desktop phones only 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 Miscellaneous Administrative Tasks Label Function Notes ListenMode Turn on speaker to listen only Menu Menu Messages Messages menu MicMute MicMute MyStatus MyStatus NewCall NewCall Call screen only Null Do nothing Offline Offline for presence QuickSetup Quick Setup feature Call screen only 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 SpeedDia
61. 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 Provisioning Server The provisioning server can be on the local LAN or anywhere on the Internet Multiple provisioning servers can be configured by having the provisioning server DNS name map to multiple IP addresses The default number of provisioning servers is one and the maximum number is eight The following protocols are supported for redundant provisioning servers HTTPS HTTP and FTP For more information on the protocol used on each platform refer to Supported Provisioning Protocols on page 3 4 All of the provisioning 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 10 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 Note Note wy POLYCOM Note Note Setting up Your System Be aware of how logs overrides and directories are uploaded to servers that map 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
62. 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 CONFIG_FILES_2345 11560 001 phone1_2345 11560 001 cfg sip_2345 11560 001 cfg will override CONFIG_FILES_SPIP560 phone1_SPIP560 cfg sip_SPIP560 cfg which will override CONFIG_FILES phonel cfg sip cfg for an SoundPoint IP 560 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 SoundStation IP and Polycom VVX 1500 phones Product Name Model Name Product Part Number SoundPoint IP 300 SPIP300 2345 11300 001 SoundPoint IP 301 SPIP301 2345 11300 010 SoundPoint IP 320 SPIP320 2345 12200 002 2345 12200 005 SoundPoint IP 321 SPIP321 2345 13600 001 SoundPoint IP 330 SPIP330 2345 12200 001 2345 12200 004 SoundPoint IP 331 SPIP331 2345 1365 001 SoundPoint IP 430 SPIP430 2345 11402 001 SoundPoint IP 450 SPIP450 2345 12450 001 SoundPoint IP 500 SPIP500 2345 11500 001 2345 11500 010 2345 11500 020
63. Ox0012bb 9 Polycom inventory manufactur er name 21 LLDP MED 127 min len gt 0x0012bb 10 Refer to Model Names on inventory 0 max len page C 36 model lt 32 name 22 LLDP MED 127 4 Oxfe08 0x0012bb 11 Empty Zero length string inventory asset ID 23 End of 0 0 0x0000 2 LLDP DU Administrator s Guide SoundPoint IP SoundStation IP VVX Note 1 For other subtypes refer to IEEE 802 1AB March 2005 at http Awww ieee802 o0rg 1 pages 802 1ab html 2 For other application types refer to TIA Standards 1057 April 2006 at http tia nufu eu std ANSI TIA 1057 3 At this time this TLV is not sent by the phone System Names Model System Name IP 320 Polycom SoundPoint IP 320 IP 321 Polycom SoundPoint IP 321 IP 330 Polycom SoundPoint IP 330 IP 331 Polycom SoundPoint IP 331 IP 430 Polycom SoundPoint IP 430 IP 450 Polycom SoundPoint IP 450 IP 550 Polycom SoundPoint IP 550 IP 560 Polycom SoundPoint IP 560 IP 650 Polycom SoundPoint IP 650 IP 670 Polycom SoundPoint IP 670 IP 6000 Polycom SoundStation IP 6000 IP 7000 Polycom SoundStation IP 7000 VVX 1500 Polycom VVX 1500 Model Names Model Model Name IP 320 SoundPointIP SPIP_320 IP 321 SoundPointIP SPIP_321 IP 330 SoundPointIP SPIP_330 IP 331 SoundPointIP SPIP_331 IP 430 SoundPointIP SPIP_430 IP 450 SoundPointIP SPIP_450 IP 550 SoundPointIP SPIP_550 IP 560 SoundP
64. Profiles lt audioProfile gt on page A 46 Web Server Set the jitter buffer tuning parameters including minimum and if enabled 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection 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 This algorithm is derived from G 711 Appendix II which defines a comfort noise CN payload format or bit stream for G 711 use in packet based multimedia communication systems The phone generates CN packets also known as Silence Insertion Descriptor SID frames and also decodes CN packets efficiently regenerating a facsimile of the background noise at the remote end Configuration changes can be performed centrally at the provisioning server Central provisioning server Configuration file Ena
65. 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 23 5 Ensure that the configuration process completed correctly For example on the phone press the Menu key and then select Status gt Platform gt Application to see the SIP application version and Status gt Platform gt Configuration to see the configuration files downloaded to the phone Monitor the provisioning server event log and the uploaded event log files if permitted All configuration files used by the provisioning server are logged You can now instruct your users to start making calls Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Provisioning SoundStation IP 7000 Phones Using C Link Normally the SoundStation IP 7000 conference phone is provisioned over the Ethernet by the provisioning server However when two SoundStation IP 7000 phones are daisy chained together the one that is not directly connected to the Ethernet can still be provisioned known as the secondary 12 foot Ethernet Cable 25 foot we Network Cable Gay The provisioning over C Link feature is automatically enabled when a SoundStation IP 7000 phone is not connected to the Ethernet Both SoundStation IP 7000 phones must be running the same version of the SIP application The steps for provisioning the secondary SoundStation IP 7000 phone are the same as for th
66. Registration Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify the default number of calls that can be active or on hold per provisioning sip cfg line key server e For more information refer to Call Handling Configuration lt call gt on page A 76 Configuration file Specify per registration the number of calls that can be active or on phonet cfg hold per line key assigned to that registration This will override the default value specified in sip cfg For more information refer to Registration lt reg gt on page A 128 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 provisioning 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 provisioning server Local Phone User Interface Specify per registration the number of calls that can be active or on hold per line key assigned to that registration using the SIP Configuration menu Either the Web Server
67. Technical Bulletin 17877 Using Custom Certificates With SoundPoint IP Phones at http www polycom com support voice soundpoint_ip VoIP_Technical_Bulletins_pu b html Configuration changes can be performed locally Local Local Phone User The custom certificate can be specified and the type of certificate to Interface trust can be set under the Settings menu Incoming Signaling Validation The three optional levels of security for validating incoming network signaling are e Source IP address validation e Digest authentication e Source IP address validation and digest authentication Configuration changes can be performed centrally at the provisioning server Central provisioning 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 18 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 Both RTP and RTCP signaling may be encrypted using an AES algorithm as described in RFC3711 When this feature is enabled phones will negotiate with the other end point whether and what type of encryption or Administrator s Guide for the SoundPoint IP SoundStation
68. Up to eight attributes can be configured x 1 to 8 dir corp attribute x label UTF 8 encoded string Null A UTF 8 encoded string that is used as the label when data is displayed dir corp attribute x type first_name last_name phone_number SIP_address URL other last_name This parameter defines how the attribute is interpreted by the phone Entries can have multiple attributes of the same type Type other is used for display purposes only 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 dir corp attribute x sticky Oori Null If set to O or Null the filter criteria for this attribute is reset after a reboot If set to 1 the filter criteria for this attribute is retained through a reboot Such attributes are denoted with a before the label when displayed on the phone dir corp attribute x filter UTF 8 encoded string Null The filter string for this attribute which is edited when searching dir corp attribute x searchable Oor1 A flag to determine if the attribute is searchable through quick search This flag applies for x 2 or greater If set to O or Null quick search on this attribute is disabled If set to 1 quick search on this attribute is enabled dir corp backGro
69. 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 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 configurat
70. 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 Note 2xx Responses Success Session Initiation Protocol SIP Response Supported Notes 200 OK Yes 202 Accepted Yes In REFER transfer 3xx Responses Redirection Response Supported Notes 300 Multiple Choices Yes 301 Moved Permanently Yes 302 Moved Temporarily Yes 305 Use Proxy No 380 Alternative Service No Axx Responses Request Failure All 4xx responses for which the phone does not provide specific support will be treated the same as 400 Bad Request Response Supported Notes 400 Bad Request Yes 401 Unauthorized Yes 402 Payment Required No 403 Forbidden No 404 Not Found Yes 405 Method Not Allowed Yes 406 Not Acceptable No 407 Proxy Authentication Required Yes 408 Request Timeout No 410 Gone No 413 Request Entity Too Large No 414 Request URI Too Long No Administrator s Guide SoundPoint IP SoundStation IP VVX Response Supported Notes 415 Unsupported Media Type Yes 416 Unsupported URI Scheme No 420 Bad Extension No
71. 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 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 supplie
72. any configuration file then the others will have to adjusted accordingly For more information on why to create another configuration file refer to the Configuration File Management on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones white paper at http www polycom com global documents support technical products voice white_paper_configuration_file_management_on_soundpoint_ip_ph ones pdf For more information on phone configuration and provisioning refer to the appropriate Technical Bulletins and Quick Tips at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html For more information on encrypting configuration files refer to Encrypting Configuration Files on page C 4 To deploy phones from the provisioning server 1 Create per phone configuration files by performing the following steps 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 127 Throughout this guide the terms Ethernet address and MAC address are used interchangeable Do not use MACaddress phone cfg as the per phone filename This filename is used by the phone itself to store user preferences overrides Administrator s Guide for the SoundP
73. any other video device If set to 1 enabled or Null video is sent in outgoing calls and received in incoming calls If set to 0 video is not sent in outgoing calls and not received in incoming calls All calls are audio only video autoStartVideoTx Oor1 Null Flag to determine whether or not video transmission occurs when a call starts If set to 0 video transmission does not start If set to 1or Null video transmission from the near end starts when a call starts video screenMode normal full crop Null Applies to the video window shown in the normal mode If set to normal or Null all pixels are displayed black bars appear on the top bottom or sides of the window if necessary to maintain the correct aspect ratio If set to full all pixels are displayed and the image is stretched linearly and independently to fill the video frame If set to crop the black bars do not appear the image size is re sized to maintain the correct aspect ratio and any parts of the image that do not fit in the display are cropped video screenModeFS normal full crop Null Applies to the video window in Full Screen mode The image is re sized to maintain the correct aspect ratio and any parts of the image that do not fit in the display are cropped video quality motion sharpness Null Determine the quality of vi
74. are built in w 1 solids w 2 and bitmaps w 3 w 2 is used when selecting any image as a background w is used when selecting any image from the Digital Picture Frame as a background This image is stored under Local File Only one local file at a time is supported bg VVX_1500 color om x name any string Null Graphic files for display on the phone For example if you set bg VVX_1500 color bm 1 name to Polycom bmp the user will be able to select Polycom bmp as a background on the phone bg hiRes color selection 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 name any string bg hiRes color pat solid x red 0 to 255 bg hiRes color pat solid x green 0 to 255 bg hiRes color pat solid x blue 0 to 255 Solid pattern name For x 1 Light Blue x 2 Teal x 3 Tan x 4 Null The screen background layouts For x 1 red 151 green 207 blue 249 For x 2 red 73 green 148 blue 148 For x 3 red 245 green 157 blue 69 For x 4 red Null green Null blue Null Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute bg hiRes color om x name Permitted Values any string bg hiRes color om x em name any string Default built in value of Thistle Interpretation Graphic
75. auth regPasswordx any string The SIP registration password for registration x where x 1 to 48 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 The syslog server IP address or host name The default value is NULL A 153 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Name Possible Values Description device syslog transport None 0 The protocol that the phone will use to write to the UDP 1 syslog server TCP 2 If set to None transmission is turned off but the TLS 3 server address is preserved device syslog facility 0 to 23 A description of what generated the log message For more information refer to section 4 1 1 of RFC 3165 The default value is 16 which maps to local 0 device syslog renderLevel 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 101 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 7 device net etherVlanFilter E
76. 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 Lined 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 90 Backgrounds lt bg gt Configuration Files The backgrounds used by the Sound Point IP 450 550 560 650 and 670 and the Polycom VVX 1500 phones are defined in this section In the following table w 1 to 3 x 1 to 6 hiRes parameters are used by SoundPoint IP 550 560 650 and 670 phones medRes parameters are used by SoundPoint IP 450 phones and VVX_1500 parameters are used by Polycom VVX 1500 phones This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation bg VVX_1500 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 The type of backgrounds
77. change these values POLYCOM Attribute Default voice gain rx analog handset 0 voice gain rx analog handset VVX_1500 2 voice gain rx analog headset 0 voice gain rx analog headset VVX_1500 2 voice gain rx analog chassis 0 voice gain rx analog chassis IP_330 0 voice gain rx analog chassis IP_430 0 voice gain rx analog chassis IP_450 0 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 chassis VVX_1500 3 voice gain rx analog ringer 0 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Default voice gain rx analog ringer IP_330 0 voice gain rx analog ringer IP_430 0 voice gain rx analog ringer IP_450 0 voice gain rx analog ringer IP_650 0 voice gain rx analog ringer IP_6000 0 voice gain rx analog ringer IP_7000 0 voice gain rx analog ringer VVX_1500 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_450 5 voice gain rx digital chassis IP_6000 5 voice gain rx digital chassis IP_7000 5 voice gain rx digital chassis VVX_1500 0 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_450 12 voice gain
78. classes A 97 indicator patterns A 97 indicators assignments A 97 IP TOS call control A 70 keep alive A 75 keys A 89 local camera view A 67 local protocol A 7 localization A 25 main browser A 115 multilingual A 26 music on hold A 20 network monitoring A 71 outbound proxy A 17 password lengths A 104 platform A 95 port A 74 power saving A 126 presence A 86 protocol A 7 protocol server A 8 protocol special events A 19 provisioning A 108 Quality of Service A 67 RAM disk A 109 receive equalization A 54 request A 109 request delay A 110 request validation A 18 resource A 112 ring type A 40 routing server A 25 Index 1 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family RTP A 68 A 69 A 74 sampled audio for sound effects A 34 SDP A 16 security A 103 shared calls A 80 SIP A 11 soft keys A 123 sound effect patterns A 36 sound effects A 35 tones A 31 transmit equalization A 55 user preferences A 29 video A 61 video codec preferences A 62 video codec profiles A 63 voice activity detection A 57 voice coding algorithms voice coding algorithms lt codecs gt A 41 voice settings A 41 volume persistence A 47 web server A 75 application configuration file A 5 application error messages 5 3 application files overview 2 6 application launch pad 4 32 applications 4 32 Applications key 4 31 attendant lt attendant gt A 145 attended transfers 4 20 audio codec i
79. contained in this product are Copyright 1998 1999 2000 Thai Open Source Software Center Ltd and Clark Cooper Copyright 1998 by the Massachusetts Institute of Technology Copyright 1998 2003 The OpenSSL Project Copyright 1995 1998 Eric Young eay cryptsoft com All rights reserved Copyright 1995 2002 Jean Loup Gailly and Mark Adler Copyright 1996 2004 Daniel Stenberg lt daniel haxx se gt Permission is hereby granted free of charge to any person obtaining a copy of this software and associated documentation files the Software to deal in the Software without restriction including without limitation the rights to use copy modify merge publish distribute sublicense and or sell copies of the Software and to permit persons to whom the Software is furnished to do so subject to the following conditions The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software THE SOFTWARE IS PROVIDED AS IS WITHOUT WARRANTY OF ANY KIND EXPRESS OR IMPLIED INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM DAMAGES OR OTHER LIABILITY WHETHER IN AN ACTION OF CONTRACT TORT OR OTHERWISE ARISING FROM OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE 2009 Polycom Inc Al
80. defined as follows Permitted Attribute Values Default Interpretation volpProt local port 0 to 65535 Null 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Server lt server gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt server dhcp available Oor1 0 If set to 1 check with the DHCP server for SIP server IP address If set to 0 do not check with DHCP server volpProt server dhcp option 128 to 255 Null Option to request from the DHCP server if volpProt server dhcp available 1 There is no default value for this parameter it must be filled in with a valid value Note If the reg x server y address parameter in Registration lt reg gt on page A 128 is non Null it takes precedence even if the DHCP server is available volpProt server dhcp type Oor1 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 hos
81. delay and end system delay Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Server lt server gt 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 rtcpxr enable 0 1 0 Enables generation of RTCP XR packets Video Settings lt video gt Note Configuration Files This configuration attribute is only supported for use on the Polycom VVX 1500 These configuration attributes are defined as follows Attribute Permitted Values Default Interpretation video enable 0 Disable 1 Enable Null Flag to determine whether or not video calls are established This applies to all calls between two Polycom VVX 1500s and between Polycom VVX 1500 and
82. eens 4 80 DTMF Event RTP Payload 0 0000 4 80 Acoustic Echo Cancellation 5 000000000 4 80 Audio Codes sse nieces ihe Race lateral ye a Beaute ai tee le Gye EEA le toe 4 81 Background Noise Suppression 000 00000 4 82 Comfort Noise Fill isteach itive aE aaa ek 4 83 Automatic Gain Control 002 eee ee 4 83 IP Type OF Service secs tise taba ee ea ee bes BSE Sis 4 83 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family viii IERE 302 1p O 2s2cn cee es iaewan need ia creimee eae bees hobs 4 83 Voice Quality Monitoring 0 eee eee eee eee eee 4 84 Dynamic Noise Reduction 0066 e cece eee eee 4 85 Treble Bass Controls riire riiin tiir EEr EEN 4 85 Setting Up Video Features 0 000 02 4 86 Video Transmission 000000 e eee eee eee eee 4 86 VideO COMES oi 4 55 64 teenecd A aewieckct enact Slane agen Mies dae a Peden eR 4 87 Setting Up Security Features 0 cece eee 4 88 Local User and Administrator Privilege Levels 4 88 Custom Certificates serer sion nune i EERE EEEE ey E REER 4 89 Incoming Signaling Validation 6 66 cece eee eens 4 89 Secure Real Time Transport Protocol 0004 4 89 Configuration File Encryption 0 0 0 e cece eee ee eee 4 90 Digital Certificates 2 2344 5 49 6 sess ied ets i ve REA
83. emergency gt 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 dialplan x routing emergency y server z 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 aora 15 17 18 911 emergency routing combination of SIP URL sog When one of these defined URL is detected as being dialed by the user the call will be automatically directed to the defined emergency server positive integer Null Index representing the server defined in Server lt server gt on page A 142 that will be used for emergency routing A 142 Configuration Files Messaging lt msg gt Message waiting indication is supported on a per registration basis This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation msg bypassinstantMessage Oor1 0 If set to 1 the display offering a choice of Message Center and Instant Messages will be bypassed when pressing the Messages
84. ends when the Proceeding state starts e Hold The call is put on hold locally 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 starting 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 32x 33x 430 450 550 560 650 and 670 and Polycom VVX 1500 phones This capability is linked to the Enhanced Feature Key feature refer to Enhanced Feature Keys on page 4 38 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 123 Configuring You
85. ewan 4 91 Mutual TLS Authentication 0000 2 eee eee 4 93 Configuring SoundPoint IP SoundStation IP VVX Phones Locally 4 94 5 Troubleshooting Your SoundPoint IP SoundStation IP VVX PHONES 36 sa x5 54 20d E E se oeW hs SES ae aeenen een Error Messages feg 42 ots caer at cares haves dee oa hae wh ek wes 5 2 BootROM Error Messages 0000s 5 2 Application Error Messages 000 0000 eee eee eee 5 3 Status Menu ois csvivadeis ecan al Coe Oise AES ee HS 5 4 LOS PUES osu viwenereevheaes bene eRe E CEEI ee Ded newt we ie 5 5 Reading a Boot Log 6 cette teens 5 8 Reading an Application Log s s cece cece 5 9 R ading a Syslog osere sacs tne pe haiiiupes EERE E EREEREER 5 10 Testing Phone Hardware 0 e eee eee eee 5 10 Power and Startup 0 cece cette nee ee 5 11 COMOL ar a a E E E EE ERDE iiri 5 12 Access to Screens and Systems 0 000 e eee eee eee eee 5 13 CAMA 5 ie avebGle E sdb ine alee we ale wed hee Biss wae quem 5 14 Displays cerrierp repki p niki pi nei n Reaves biases kra EAA 5 15 7251 6 0 nn ea a E 5 16 Productivity SUE seepia ices eee ce eek ene cee eee DEES 5 16 UO perading ace os oes sadn Sad iat RGSS Seen shed es abet he 5 17 Contents A Configuration Files 0 cece cece ee eee ee AH Master Configuration Files 00 00 0 e eee eee eee ee A 2 Application Configuration 00000 A 5 Proto
86. feature depends on support from a SIP server With many SIP servers this feature is implemented using a particular star code sequence With some SIP servers specific network signaling is used to implement this feature Configuration changes can be performed centrally at the provisioning server Central Configuration file Turn this feature on or off provisioning sip ctg server For more information refer to Feature lt feature gt on page A 110 Determine the type of call park and retrieval string e For more information refer to Call Handling Configuration lt call gt on page A 76 4 24 Configuring Your System Last Call Return The phone allows server based last call return This feature depends on support from a SIP server With many SIP servers this feature is implemented using a particular star code sequence With some SIP servers specific network signaling is used to implement this feature Configuration changes can be performed centrally at the provisioning server Central Configuration file Turn this feature on or off provisioning sip cfg e For more information refer to Feature lt feature gt on page A 110 server Specify the string sent to the server for last call return For more information refer to Call Handling Configuration lt call gt on page A 76 Setting Up Advanced Features This section provides information for making configuration changes f
87. files for display on the phone and Expansion Module 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 e name is Jellyfish jpg name is JellyfishEM jpg For x 6 name is Mountain jpg name is MountainEM jpg Note If the file is missing or unavailable the built in default solid pattern is displayed bg hiRes gray selection 2 1 Specify which type of background w and index x for that type is selected on reboot bg hiRes gray pr x adj Specify the brightness adjustment to the graphic bg hiRes gray pat solid x name any string White Solid pattern name For x 1 White x 2 Light Gray x 3 4 Null bg hiRes gray pat solid x red 0 to 255 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 Configuration Files Attribute bg hiRes gray om x name Permitted Values any string bg hiRes gray om x em name any string bg hiR
88. 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 Oor1 If set to 1 and fixedDayEnable is set to 0 DST starts on the last day specified by start dayOfWeek of the week in the month The start date is ignored tcplpApp sntp daylightSavings stop month 1 12 11 Month to stop DST tcplpApp sntp 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 Oor1 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 Administrato
89. for the SoundPoint IP SoundStation IP VVX Family 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 presented 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
90. gradient from black 0 0 0 to white FF FF FF The SoundPoint IP 670 phone support a 12 bit color scale from black 0 0 0 to white FFFF FFFF FFFF The SoundStation IP 6000 phone is the same as the IP 7000 The SoundStation IP 7000 phone supports a 32 bit grayscale whichis 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 32x 33x bitmap IP_330 68 name IP 430 bitmap IP_400 61 name IP 450 bitmap IP_450 82 name IP 550 560 650 670 bitmap IP_600 83 name IP 6000 bitmap IP_4000 83 name IP 7000 bitmap IP_7000 84 name Administrator s Guide SoundPoint IP SoundStation IP VVX For example lt bitmaps gt lt IP_330 bitmap IP_330 68 name logo 330 gt lt IP_400 bitmap IP_400 61 name logo 430 gt lt IP_450 bitmap IP_450 82 name logo 450 gt lt IP_600 bitmap IP_600 83 name logo 650 gt lt IP_4000 bitmap IP_4000 83 name logo 6000 gt lt IP_7000 bitmap IP_7000 84 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 idleDispl
91. gt Permitted Attribute Values Default Interpretation video codecPref H264 1 to 4 1 Specifies the video codec preferences for the Polycom VVX 1500 phone video codecPref H2631998 1 to 4 2 video codecPref H263 1 to 4 3 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 profile gt Configuration Files The profile attributes can be adjusted for each of the new supported video codecs Attribute Permitted Values Interpretation video profile H264 jitterBufferMax video profil e H264 jitter BufferMin The largest jitter buffer depth to be supported in milliseconds Jitter above this size will always cause lost packets This parameter 500ms to should be set to the smallest possible value 2500ms that will support the expected network jitter default 2000ms video profile H264 jitterBufferMin 33ms to The smallest jitter buffer depth in milliseconds 1000ms that must be achieved before play out begins default 150ms 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 video profile H264 jitterBufferShrink 33ms to 1000ms default 70ms
92. gt on page A 139 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 provisioning 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 provisioning server Microphone Mute A microphone mute feature is provided When activated visual feedback is provided This is a local function and cannot be overridden by the network There are no related configuration changes Soft Key Activated User Interface 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 Using the Configurable Soft Key configuration parameters an administrator can modify the default soft keys by removing them at different call stages and or adding specific single or multiple functions Refer to Enhanced Feature Keys on page 4 38 and Configurable Soft Keys on page 4 49 Speed Dial Configuring Your System 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 S
93. gt on page A 128 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 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection Configuring Your System Local continued Local Phone User Interface Use the SIP Configuration menu to specify the local SIP signaling 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 param
94. 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 121 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones SRV lt SRV gt This configuration attribute is defined as follows 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 rfc278
95. 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 Configuration Files This attribute s settings control the telephone notification events state polling events and the push server controls For more information refer to the Web Application Developer s Guide which can be found at http www polycom com voicedocumentation This attribute also includes e Telephone Notification lt telNotification gt State Polling lt statePolling gt e Push lt push gt Telephone Notification lt telNotification gt This configuration attribute is defined as follows onhookEvent If set to 1 onhook notification is enabled 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 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 Oor1 0 If set to 0 outgoing call notification is disabled outgoingEvent If
96. max len lt 255 Manufacturer s name Polycom Refer to Model Names on page C 36 Hardware version Application version BootROM version Capabilities 0x0e04 System Capabilities Telephone and Bridge if the phone has PC port support and it is not disabled Enabled Capabilities Telephone and Bridge if phone has PC port support it is not disabled and PC port is connected to PC Note PC port supported Phones IP 330 IP 331 IP 430 IP 450 IP 550 IP 560 IP 650 and IP 670 PC port not supported phones IP6000 IP7000 IP320 and IP321 Manageme nt Address 0x100c Address String Len 5 IPV4 subtype IP address Interface subtype Unknown Interface number 0 ODI string Len 0 IEEE 802 3 MAC PHY config statu s 127 Oxfe09 0x00120f e Auto Negotiation Supported 1 enabled disabled Refer to PMD Advertise and Operational MAU on page C 37 Miscellaneous Administrative Tasks No Name Type 7 bits 0 6 Length 9 bits 7 15 Type Length Org Unique Code 3 bytes Version Sub Type Information 10 LLDP MED capabilities 127 7 Oxfe07 0x0012bb 1 Capabilities 0x33 LLDP Med capabilities Network policy Extended Power Via MDI PD Inventory Class Type IIl Note Once support for configuring location Identification information is locally av
97. ml lang clock x longFormat and lcl 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 33FF CJK Unified Ideographs U 4E00 U 9FFF Hangul Syllables U AC00 U D7A3 CJK Compatibility Ideographs U F900 U FAFF CJK Half width forms U FF00 U FFFF Note Within a Unicode range some characters may not be supported due to their infrequent usage A 28 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
98. 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 provisioning server or boot server to look for any updates to the bootROM application If updates are found they are downloaded and saved to flash memory eventually overwriting itself after verifying the integrity of the download If a new bootROM is downloaded formats the file system clearing out any application software and configuration files that may have been present Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Application 5 Downloads the master configuration file This file is either called lt MAC address gt cfg or 000000000000 cfg This file is used by the bootROM and the application for a list of other file
99. of the Software and to permit persons to whom the Software is furnished to do so subject to the following conditions The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software THE SOFTWARE IS PROVIDED AS IS WITHOUT WARRANTY OF ANY KIND EXPRESS OR IMPLIED INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM DAMAGES OR OTHER LIABILITY WHETHER IN AN ACTION OF CONTRACT TORT OR OTHERWISE ARISING FROM OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE curl COPYRIGHT AND PERMISSION NOTICE Copyright c 1996 2008 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 W
100. or later 2 1 1 or later IP 321 331 4 1 3 or later 3 1 3C or later IP 430 3 1 3 C or later 1 6 6 or later IP 450 4 1 2 or later 3 1 0 C or later IP 5501 3 2 2 B or later 2 1 or later IP 5601 4 0 1 or later 2 2 2 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 6000 4 1 1 or later 3 0 2 or later IP 70002 4 1 1 or later 3 0 2 or later VVX 1500 4 1 2 or later 3 1 2B or later Note 1 SoundPoint IP 550 560 and 650 phones manufactured as of February 2009 have additional bootROM SIP application dependencies For more information refer to Technical Bulletin TB 46440 Notice of Product Shipping Configuration Change at http www polycom com support voice soundpoint_ip VoIP_Technical_Bulletin s_pub html 2 Ifthe SoundStation IP 7000 is connected to a Polycom HDX system the bootROM must be 4 1 2 or later Administrator s Guide SoundPoint IP SoundStation IP VVX Migration Dependencies In addition to the bootROM and application dependencies there are certain restrictions with regard to upgrading or downgrading from one bootROM release to another bootROM release These restrictions are typically caused by the addition of features that change the way bootROM provisioning is done so the older version become incompatible There is always a way to move forward with bootROM releases although it may be a two or three step procedure someti
101. or the provisioning 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 Customizable Fonts and Indicators Note 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 Customizable fonts and indicators are not supported on the Polycom VVX 1500 Configuring Your System Configuration changes can be performed centrally at the provisioning server server Central Configuration File Specify fonts to overwrite existing ones or specify new fonts provisioning sip cfg For more information refer to Fonts lt font gt on page A 86 Specify which bitmaps to use For more information refer to Bitmaps lt bitmap gt on page A 95 Specify how to create animations and LED indicator patterns For more information refer to Indicators lt ind gt on page A 95 Instant Messaging The phone supports sending and receiving instant text messages The user is alerted to incoming messages visually and audi
102. phones will be supported on the latest maintenance patch release of the SIP 3 1 software stream currently SIP 3 1 3 Any new features introduced after 3 1 3 are not supported Configuration parameters related to these phones will be removed from the sip cfg and phone1 cfg files in the next major release To administer these phones refer to the S P 3 1 Administrator s Guide which is available at http www polycom com voicedocumentation The following new features were introduced in SIP 3 1 2 e Feature Synchronized Automatic Call Distribution Supports ACD agent available and unavailable and allows ACD sign in and sign out Requires call server support e Quick Setup of SoundPoint IP SoundStation IP VVX Phones Simplifies the process of entering provisioning server parameters The following new feature enhancement was introduced in SIP 3 1 3 e Corporate Directory The phone s user interface to access your corporate directory has changed Also Microsoft ADAM and SunLDAP are also supported in addition to Active Directory and OpenLDAP The following new features were introduced in SIP 3 2 e LLDP and Supported TLVs Support for Link Layer Discovery Protocol LLDP and media extensions LLDP MED such as VLAN configuration For provisioning information refer to Ethernet Menu on page 3 12 e iLBC added to Audio Codecs Support for Internet Low Bitrate Codec iLBC added for the SoundPoint IP 32x 33x 450 550 560
103. polycom com support voicedocumentation 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 you 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 SoundStation IP and Polycom VVX Phones However if your organization has a mixture of SoundPoint IP 300 301 500 501 600 601 and or SoundStation IP 4000 phones deployed along with other models you will need to change the phone configuration files to continue to support the SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 phones when software releases SIP 3 2 0 or later are deployed These models were discontinued as follows e The SoundPoint IP 300 and 500 phones as of May 2006 e The SoundPoint IP 301 600 and 601 phones as March 2008 The SoundPoint IP 501 phone as of August 2009 The SoundStation IP 4000 phone as of May 2009 In all cases refer to Supporting SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones on page 3 23 Administrator s Guide for the
104. refer to Multilingual lt ml gt on page A 26 Note that within a Unicode range some characters may not be supported due to their infrequent usage The SoundPoint IP and SoundStation IP user interface is available in the following languages by default Simplified Chinese if displayable Danish Dutch English French German Italian Japanese if displayable Korean if displayable Norwegian Polish Brazilian Portuguese Russian Slovenian International Spanish and Swedish Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note The multilingual feature relies on dictionary files resident on the provisioning server The dictionary files are downloaded from the provisioning 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 be performed centrally at the provisioning server or locally Central Configuration file Specify the boot up language and the selection of language choices provisioning sip cfg to be made available to the user server For more information refer to Multilingual lt ml gt on page A 26 For instructions on adding new languag
105. 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 digital ringer VVX_1500 21 voice gain rx analog handset sidetone 20 voice gain rx analog handset sidetone VVX_1500 15 voice gain rx analog headset sidetone 24 voice gain rx analog headset sidetone VVX_1500 31 voice gain tx analog handset 6 voice gain tx analog handset VVX_1500 48 voice gain tx analog headset 3 voice gain tx analog headset VVX_1500 47 Configuration Files Attribute Default voice gain tx analog chassis 3 voice gain tx analog chassis IP_330 36 voice gain tx analog chassis IP_430 36 voice gain tx analog chassis IP_450 36 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 analog chassis VVX_1500 25 voice gain tx digital handset 0 voice gain tx digital handset IP_330 10 voice gain tx digital handset IP_430 6 voice gain tx digital handset IP_450 6 voice gain tx digital handset IP_650 6 voice gain tx digital handset VVX_1500 12 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_450 6 voice gain tx digital headset IP_650 6 voice gain tx digital headset VVX_1500 12 voice gain tx digital chassis 3 voice gain tx digital chassis
106. 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 117 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones State Polling lt statePolling gt This configuration attribute is defined as follows Attribute 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 string Null The user name to access the state polling URL username apps statePolling string Null The password to access the state polling URL password Attribute Push lt push gt Configuration Files This configuration attribute is defined as follows Permitted Values Default Interpretation apps push messageType apps push serverRootURL 0to 3 URL 0 Null 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 n
107. the lt MACaddress 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 Note Setting up Your System c Edit the CONFIG_FILES attribute of the lt MACaddress 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 lt MACaddress gt cfg files so that it points to the log file directory e Edit the CONTACT_DIRECTORY attribute of the lt MACaddress 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 MACaddress gt cfg files so that it references the appropriate sip ld files For example the reference to sip ld is changed to 2345 11670 001 sip Id to boot the SoundPoint IP 670 image 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
108. 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 previous 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 70 For example Access URL lt http server poly
109. 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 13 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note 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 650 and 670 phones To switch the text entry mode on the SoundPoint IP 32x 33x 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 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 Ifa boot server value is s
110. the provisioning 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 provisioning server If a phone does not find its own lt Ethernet address gt cfg file it will use this one and establish a baseline configuration T
111. 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 Oori If set to 1 DTMF tones will be substituted with a non DTMF pacifier tone when dialing in hands free mode This prevents DTMF digits being broadcast to other surrounding telephony devices or being inadvertently transmitted in band due to local acoustic echo Note tone dtmf chassis masking should only be enabled when tone dtmf viaRtp is disabled tone dtmf stim pac offHookOnly Oor1 Not currently used Configuration Files Attribute tone dtmf viaRtp Permitted Values Default Interpretation Oor1 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 atmf viaRtp is disabled tone dtmf rfe2833Control Oor1 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 thi
112. 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 net ipAddress set 1 then use the contents of the device net ipAddress field device net ipAddress dotted decimal IP address Phone s IP address Note This field is not used when DHCP client is enabled device net subnetMask dotted decimal IP address Phone s subnet mask Note This field is not used when DHCP client is enabled device net IPgateway dotted decimal IP address Phone s default router IP gateway Note This field is not used when DHCP client is enabled device 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 provisioning server Option 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 and negotiate power through CDP device dhcp enabled Oor 1 For description r
113. to Null the value 0 is used video camera sharpness 0to6 Null Set sharpness level The value range is from 0 Lowest to 6 Highest If set to Null the value 3 is used Local Camera View lt localCameraView gt These settings control how the local camera is viewed on the screen These configuration attributes are defined as follows Permitted Attribute Values Default Interpretation video localCameraView 0 Disable Null Determines whether the local camera view is fullscreen enabled 1 Enable shown in the full screen layout If set to 0 the local camera view is not shown If set to 1 or Null the local camera view is shown video localCameraView pip or Null How the local camera view is shown fullscreen mode Null If set to pip the local camera view appears as a picture in picture with the far end window If set to Null the local camera view appears side by side with the far end window Quality of Service lt QOS gt These settings control the Quality of Service QOS options This attribute includes Ethernet IEEE 802 1p Q lt ethernet gt IPTOS lt IP gt Ethernet IEEE 802 1p Q lt ethernet gt The following settings control the 802 1p Q user_priority field e RIP lt RTP gt e Call Control lt callControl gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones e Other lt other gt RTP lt RTP gt These parameters apply
114. to RTP packets Permitted Attribute Values Default Interpretation qos ethernet rtp user_priority 0 7 5 User priority used for Voice RTP packets qos ethernet rtp video user_priority 0 7 5 User priority used for Video 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 68 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 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 AF3
115. 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 4 70 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 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 70 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 Microbr
116. you are having trouble connecting to the provisioning 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 provisioning 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 provisioning 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 provisioning server but the causes may be numerous It may be cabling issue it may be related to DHCP configuration or it could be a problem with the provisioning 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
117. 0 and volpProt SIP serverFeatureControl cf 1 the phone will not perform local Call Forward behavior If set to 1 or Null the phone will perform local Call Forward behavior on all calls received volpProt SIP serverFeatureControl dnd Oor1 Null If set to 1 server based DND is enabled The call server has control of DND If set to 0 or Null server based DND is not enabled This is the old behavior volpProt SIP serverFeatureControl localProcessing dnd If set to 0 and volpProt SIP serverFeatureControl dnd 1 the phone will not perform local DND call behavior If set to 1 or Null the phone will perform local DND call behavior on all calls received Configuration Files Attribute volpProt SIP authOptimizedInFailover volpProt SIP csta Permitted Values 0 1 Oori 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 volpProt 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 Oor1 Nul
118. 0 for that registration reg x ringType 1 to 22 2 The ringer to be used for calls received by this registration Default is the first non silent ringer reg x lineKeys 1 to max 1 max the number of line keys on the phone max 1 on SoundStation IP 6000 7000 max 2 on IP 32x 33x 430 max 3 on IP 450 max 4 on IP 550 560 max 6 on VVX 1500 max 34 on IP 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 A 129 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute reg x callsPerLineKey Permitted Values 1 to 34 OR 1 to 24 OR 1 to 8 OR 1to4 Default 34 OR 24 OR 8 OR 4 Interpretation For the SoundPoint IP 650 and 670 the permitted range is 1 to 34 and the default is 34 For the SoundPoint IP 550 and 560 and the VVX 1500 the permitted range is 1 to 24 and the default is 24 For the SoundPoint IP 430 the permitted range is 1 to 4 and the default is 4 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 76 If reg 1 callsPerLineKey is set to 1 call wait
119. 00 in the middle of the dialed number that matches For example if a customer dials 16092345678 a call is placed to 16002345678 e 911xxx T A period which matches an arbitrary number including zero of occurrences of the preceding construct For example 911123 with waiting time to comply with T is a match 9111234 with waiting time to comply with T is a match 91112345 with waiting time to comply with T is a match and the number can grow indefinitely given that pressing the next digit takes less than T The following guidelines should be noted e The letters x T R are case sensitive e You must use only or 0 9 between second and third R e Ifadigit map does not comply it is not included in the digit plan as a valid one That is no matching is done against it e There is no limitation on the number of R triplet sets in a digit map However a digit map that contains less than full number of triplet sets for example a total of 2Rs or 5Rs is considered an invalid digit map Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones e Using T in the left part of RRR syntax is not recommended For example ROTR322R should be avoided This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan digitmap string compatible with the 2 9 11 OT When this attribute is digit map feature of 011xxx T present number only diali
120. 000 RFC 1890 128 Kbps 8 Ksps 10ms 3 5 KHz L16 16000 256 Kbps 16 Ksps 7 KHz L16 32000 512 Kbps 32 Ksps 14 KHz L16 44100 705 6 Kbps 44 1 Ksps 20 KHz L16 48000 768 Kbps 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 RFC 2833 phone event RFC 2833 N A N A N A N A iLBC iLBC RFC 3951 13 33Kbps 8 Ksps 30ms 60ms 3 5KHz 15 2Kbps 20ms 80ms 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 about 100kbps of network bandwidth two way audio Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify codec priority preferred payload sizes and jitter buffer tuning provisioning sip cfg parameters server e For more information refer to Codec Preferences lt codecPref gt on page A 42 and Codec Profiles lt audioProfile gt on page A 46 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 provisioning server Changes will permanently override global settings unless deleted through the Reset
121. 0000 e eee eee eee A 151 B Session Initiation Protocol SIP 22 Bol RFC and Internet Draft Support 00 0 eee eee eee B 2 Request Support ak tobe E Reo Row Sakae B 3 Header Support erdee teer i nioe RENER eae eae oe ate i dats B 4 Response SUPPOLt arest tr ERINA N eee EAEEREN EERIE ENS B 6 Hold Implementation 00 eee eens B 9 Reliability of Provisional Responses 60 00 0c eee eens B 9 Transfer Geshe eke Lala eg aod a hae bea se nda bea he tipe aaa B 9 Third Party Call Control 0 0 000000 B 9 SIP for Instant Messaging and Presence Leveraging Extensions B 10 Shared Call Appearance Signaling 00 0 B 10 Bridged Line Appearance Signaling 00 eee rnrn B 10 C Miscellaneous Administrative Tasks C l Trusted Certificate Authority List 0 0 eee eee eee C 1 Encrypting Configuration Files 0 6 6 c cece cence eee C 4 Changing the Key on the Phone 00 cece cece C 5 Adding a Background Logo 6 cee nee ees C 6 BootROM SIP Application Dependencies 0005 C 9 Migration Dependencies 0000 e eee eee eee eee C 10 Multiple Key Combinations 00 02 eee eee ee eee C 10 Default Feature Key Layouts 0 000 e eee eee eee eee C 12 Internal Key Functions 00 00 e eee eee ee eee C 18 Assigning a VLAN ID Using DHCP
122. 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Permitted Attribute Values Default Interpretation call serverMissedCall x enabled Oor1 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 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 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Permitted Attribute Values Default Interpretation call missedCallTracking x enabled Oor 1 1 If set to 1 or Null missed call tracking is enabled If call missedCallTracking x enabled is set to 0 then missedCall counter is not updated regardless of what call serverMissedCalls x enabled is set to and regardless of how the server is configured There is no Missed Call List provided under Menu gt Features of the phone If call missedCallTracking x enabled is set to 1 and call serverMissedCalls x enabled is set to 0 then the number of missedCall counter is incremented regardless of how the server is configured If call missedCallTracking x enabled is set to 1 and call serverMissedCalls x enabled is set to 1 then the ha
123. 1995 1998 Eric Young eay cry ptsoft com All rights reserved This package is an SSL implementation written by Eric Young eay cryptsoft com The implementation was written so as to conform with Netscape s SSL This library is free for commercial and non commercial use as long as the following conditions are adhered to The following conditions apply to all code found in this distribution be it the RC4 RSA lhash DES etc code not just the SSL code The SSL documentation included with this distribution is covered by the same copyright terms except that the holder is Tim Hudson tjh cryptsoft com Copyright remains Eric Young s and as such any Copyright notices in the code are not to be removed If this package is used in a product Eric Young should be given attribution as the author of the parts of the library used This can be in the form of a textual message at program startup or in documentation online or textual provided with the package Redistribution and use in source and binary forms with or without modification are permitted provided that the following conditions are met 1 Redistributions of source code must retain the copyright notice this list of conditions and the following disclaimer 2 Redistributions in binary form must reproduce the above copyright notice this list of conditions and the following disclaimer in the documentation and or other materials provided with the distribution Administrator s Guide S
124. 2 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 122 Soft Keys lt softkey gt Configuration Files This 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 chain
125. 3 AF41 AF42 AF43 qos ip rtp min_delay Oor1 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 Oor1 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 Oori 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 Oor1 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 qos ip rtp video 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 AF 12 gos ip rtp video parameters AF13 AF21 Default of Null which means the AF22 AF23 other gos ip rtp video AF31 AF282 parameters will be used AF33 AF41 AF42 AF43 qos ip rtp video min_delay Oor1 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 video max_throughput Oor1 1 If set to 1 set max throughput bit in the IP TOS field of the IP header or else don t set it Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation qos ip rtp video max_reliability Oor1 0 If set to 1 set
126. 4 34 Corporate Directory cies eee eta eee 4 34 Recording and Playback of Audio Calls 00000 4 37 Digital Picture Frame 0 00 00 4 38 Enhanced Feature Keys 0 0 e eee eee eee eee eee 4 38 Configurable Soft Keys eee eee eee 4 49 ECD PowerSaving ieri fog 8 508 eee Ga Sade Ghee sy 4 52 Shared Call Appearances cece eee 4 52 Bridged Line Appearance 0 0000 e eee eee eee eee 4 54 Busy Lamp Riel dss oui scsi eee oie 4 55 Voice Mail Integration 0 0 4 56 Multiple Registrations 00 e eee eee eee 4 57 SIP B Automatic Call Distribution 000008 4 59 Feature Synchronized Automatic Call Distribution 4 60 Server Redundancy 0 eee eee eee eee ee 4 60 Presence soyiga ans aces deed a gu aie e EE AAE oe 4 64 Microsoft Live Communications Server 2005 Integration 4 65 Access URL in SIP Message 0 0 cece cece ee eee 4 69 Static DNS Cache ssi oie se Ona seine Heese wwe Meike Pewee 4 72 Display of Warnings from SIP Headers 0000005 4 76 Quick Setup of SoundPoint IP SoundStation IP VVX Phones 4 77 Setting Up Audio Features 0002 4 78 Low Delay Audio Packet Transmission 0000055 4 78 Jitter Buffer and Packet Error Concealment 54 4 79 Voice Activity Detection 0 cece eee 4 79 DTMF Tone Generations etess inaia cee
127. 7 POLYCOM gt Administrator s Guide for the Polycom SoundPoint IP SoundStation IP VVX Family SIP 3 2 August 2009 1725 11530 320 Rev A Trademark Information POLYCOM the Polycom Triangles logo and the names and marks associated with Polycom s products are trademarks and or service marks of Polycom Inc and are registered and or common law marks in the United States and various other countries All other trademarks are property of their respective owners No portion hereof may be reproduced or transmitted in any form or by any means for any purpose other than the recipient s personal use without the express written permission of Polycom Patent Information The accompanying product is protected by one or more U S and foreign patents and or pending patent applications held by Polycom Inc Disclaimer Some countries states or provinces do not allow the exclusion or limitation of implied warranties or the limitation of incidental or consequential damages for certain products supplied to consumers or the limitation of liability for personal injury so the above limitations and exclusions may be limited in their application to you When the implied warranties are not allowed to be excluded in their entirety they will be limited to the duration of the applicable written warranty This warranty gives you specific legal rights which may vary depending on local law Copyright Notice Portions of the software
128. 9_U0100_U01FF fnt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 If you are 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 Null characters Delimiter required to retrieve different grouped fnt blocks This attribute also includes e JP_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 Null XxFF fnt Defines the font file that will be loaded from provisioning server during boot up Note When several font P_330 x name are defined the index x must follow consecutive increasing order IP_400 font lt IP_400 gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation font IP_400 x name fontName_height_Uxx00_U Null xxFF fnt
129. A 8 Specify standard or non standard behavior for processing line seize subscription for mutual exclusion feature For more information refer to Special Events lt specialEvent gt on page A 19 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 e For more information refer to Registration lt reg gt on page A 128 Specify per registration whether diversion should be disabled on shared lines For more information refer to Diversion lt divert gt on page A 136 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Local Web Server Specify line seize subscription period if enabled Navigate to http lt phonelPAddress gt appConf htm se Specify standard or non standard behavior for processing line seize subscription for mutual exclusion feature Navigate to http lt phonelPAddress gt appConf htm ls Specify per registration line type private or shared and line seize subscription period if using per registration servers 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 provisioning server Changes will permanently
130. CSfile 000000000000 cfg v Revision 1 21 gt Configuration Files lt APPLICATION APP_FILE_PATH PHONE_PART_NUMBER sip 1d CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG FILE DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY gt You can also use the substitution strings PHONE_MODEL PHONE_PART_NUMBER MACADRESS and PHONE_MAC_ADDRESS in the master configuration file For more information refer to Product Model and Part Number Mapping on page C 25 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 SIP Configuration File gt lt For information on configuring Polycom VoIP phones please refer to the gt lt Configuration File Management white paper available from gt lt http www polycom com common documents whitepapers configuration_file _management_on_soundpoint_ip_phones pdf gt lt SRCSfile 000000000000 cfg v Revision 1 21 gt lt APPLICATION APP_FILE_PATH sip ld CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG_FILE_DIRECTORY OVERRIDES_DIRECTORY CONTACTS_DIRECTORY LICENSE_DIRECTORY gt lt APPLICATION APP_FILE_PATH_SPIP300 SPIP300 sip 1ld CONFIG_FILES_SPIP300 phonel_SPIP300 cfg sip_
131. 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 66 Note Any contacts added through the SoundPoint IP phone s buddy list will appear as a contact in Microsoft Office Communicator and Windows Messenger Polycom recommends that the BLF not be used in conjunction with the Microsoft wy Live Communications Server 2005 feature For more information refer to Busy POLYCOM Lamp Field on page 4 55 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can performed centrally at the provisioning server Central Configuration file Specify that support for Microsoft Live Communications Server 2005 provisioning sip cfg is enabled server For more information refer to SIP lt SIP gt on page A 11 Specify the line registration number used to send SUBSCRIBE for presence For more information refer to Presence lt pres gt on page A 86 Turn the presence and messaging features on or off e For more information refer to Feature lt feature gt on page A 110 Configuration file Specify the number of line keys to assign per registration phonet cfg e
132. Configuration Files Permitted Attribute Values Default Interpretation 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 Oor1 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 signaling If set to shared augment call signaling with call state subscriptions and notifications and use access control for outgoing calls reg x thirdPartyName string in the same Null This field must match the reg x address format as value of the other registration which makes reg x address up the bridged line appearance BLA It must be Null in all other cases reg x auth userld string Null User ID to be used for authentication challenges for this registration If non Null will override the Reg User x parameter entered into the Authentication submenu off of the Settings menu on the phone reg x auth password string Null Password to be used for authentication challenges for this registration If non Null will override the Reg Password x parameter entered into the Authentication submenu off of the Settings menu on the phone reg x acd login logout Oor1 0 If both parameters are set to 1 fora r registration the ACD feature will be enabled reg x acd agent available Oor 1
133. Configuration file Specify the conference hold behavior all parties on hold or only host provisioning sip cfg is on hold server e For more information refer to Call Handling Configuration lt call gt on page A 76 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 76 Specify which type of conference to establish and the address of the centralized conference resource For more information refer to Conference Setup lt conference gt on page A 19 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note Manage Conferences This feature is supported on the SoundPoint IP 450 550 560 650 and 670 desktop phones the SoundStation IP 7000 conference phone and the Polycom VVX business media phone This feature requires a license key for activation on all phones except the SoundStation IP 7000 and the Polycom VVX 1500 Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller The individual parties within a conference can be managed New parties can be added and information about the conference participants can be viewed for example names phone numbers send receive status or media flow receive and transmit codecs and hold status Configuration changes can be perfo
134. DP overrides Local FLASH which VLAN ID 3 8 overrides DHCP VLAN Discovery 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 71 The CDP Compatibility value can be obtained from a connected Ethernet switch if the switch supports CDP In the case where you do not have control of your DHCP server or do not have the ability to set the DHCP options an alternate method of automatically discovering the provisioning server address is required Connecting to a secondary DHCP server that responds to DHCP INFORM queries with a requested provisioning 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 for the SoundPoint IP SoundStation IP VVX Family 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 provisioning server depends on se
135. Extended Reports RTCP XR The packets are sent to a report collector as specified in draft RFC draft ietf_sipping_rtcp summary 02 Three types of quality reports can be enabled e Alert Generated when the call quality degrades below a configurable threshold e Periodic Generated during a call at a configurable period e Session Generated at the end of a call A wide range of performance metrics are generated Some are based on current values such as jitter buffer nominal delay and round trip delay while others cover the time period from the beginning of the call until the report is sent such as network packet loss Some metrics are computed using other metrics as input such as listening Mean Opinion Score MOS conversational MOS listening R factor and conversational R factor Configuration changes can be performed centrally at the provisioning server Central Configuration file Specify the location of the central report collector how often the provisioning sip cfg reports are generated and the warning and critical threshold values server that will cause generation of alert reports e For more information refer to Quality Monitoring lt quality monitoring gt on page A 58 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 mo
136. Feb 13 01 12 45 172 23 7 249 0213011245 appl 3 00 DNS resolver servers are 172 23 0 200 172 23 0 23 Feb 13 01 12 45 172 23 7 249 0213011245 appl 3 00 DNS resolver search domain is vancouver polycom com Feb 13 01 12 45 172 23 7 249 0213011245 appl 3 00 Bootline esw 3 0 bootHost flash e 172 23 7 249 f fff Apr 15 22 32 22 172 23 7 249 041522322Z appl 3 00 Time has been set from 172 23 0 200 172 23 0 200 Apr 15 22 32 22 172 23 7 249 041522322Z2 appl 3 00 DHCP returned result Ox3E7 from server 172 23 0 232 Apr 15 22 32 22 172 23 7 249 041522322Z appl 13100 Phone IP address is 172 23 7 249 Apr 15 22 32 22 172 23 7 249 041522322Z appl 3100 Subnet mask is 255 255 0 0 Apr 15 22 32 22 172 23 7 249 041522322Z appl 131001 Gateway address is 172 23 2 240 Apr 15 22 32 22 172 23 7 249 041522322Z appl 131001 Time server is 172 23 0 200 Apr 15 22 32 22 172 23 7 249 04152232ZZ2 appl 131001 GMT offset is 28800 seconds RT MRS ST eee at sean Ue Rear Rear ae Fae ee A Testing Phone Hardware To obtain more detailed 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 e 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 hard
137. For more information refer to Registration lt reg gt on page A 128 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 148 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 148 Configuration File Examples SoundPoint IP phones can be deployed in two basic methods In the first method Microsoft Live Communications Server 2005 serves as the call server and the phones have a single registration In the second method the phone has a primary registration to call server that is not Microsoft Live Communications Server LCS and a secondary registration to LCS for presence purposes To set up a single registration with Microsoft Live Communications Server 2005 as the call server 1 Modify the sip cfg configuration file as follows a Open sip cfg in an XML editor b Locate the feature parameter c For the feature 1 name presence attribute set feature 1 enabled to 1 d For the feature 2 name messaging attribute set feature 2 enabled to 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 SoundPo
138. IF Refer to Bit H263 1998 90000 768 kbps 30 fps QCIF Rate column Rx Frame Size CIF QCIF SQCIF QVGA SIF H 264 H264 90000 64 kbpsto 5 fps to Tx Frame size CIF Refer to Bit 768 kbps 30 fps QCIF Rate column Rx Frame Size CIF QCIF SQCIF QVGA SIF Configuration changes can be performed centrally at the provisioning server Central Configuration file Specify codec priority payload type and jitter buffer tuning provisioning sip cfg parameters server For more information refer to Codec Preferences lt codecPref gt on page A 62 and Codec Profiles lt profile gt on page A 63 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 Secure Real Time Transport Protocol e Configuration File Encryption e Digital Certificates e Mutual TLS Authentication 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
139. IP SoundStation IP VVX Family 2 16 Setting up Your System POLYCOM Your SoundPoint IP SoundStation IP VVX 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 VVX phones This chapter contains information on Setting Up the Network e Setting Up the Provisioning Server e Deploying Phones From the Provisioning 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 SoundPoint IP SoundStation IP VVX phones the use of a provisioning 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 provisioning server when it boots up it will ope
140. IP VVX Family authentication to utilize for the session This negotiation process is compliant with RFC4568 Session Description Protocol SDP Security Descriptions for Media Streams Authentication proves to the phone receiving the RTP RTCP stream that the packets are from the expected source and have not been tampered with Encryption modifies the data in the RTP RTCP streams so that if the data is captured or intercepted it cannot be understood it sounds like noise Only the receiver knows the key to restore the data A number of configuration parameters have been added to allow you to turn off authentication and encryption for RTP and RTCP streams This is done mainly to allow the system administrator to reduce the CPU usage on legacy Polycom phones in certain deployment scenarios for example if three way local conferencing is required Note When using SRTP with Polycom VVX 1500 phone limit the video bandwidth on the Polycom VVX 1500 to 384kbps otherwise you will experience performance issues If the call is completely secure RTP authentication and encryption and RTCP authentication and RTCP encryption are enabled then the user sees a padlock symbol appearing in the last frame of the connected context animation two arrow moving towards each other In SIP 2 2 the SRTP feature has been implemented in a very configurable manner The reason for this is to allow deployment in a mixed environment where some elemen
141. IP_330 12 voice gain tx digital chassis IP_430 12 voice gain tx digital chassis IP_450 12 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 digital chassis VVX_1500 6 voice gain tx analog preamp handset 23 voice gain tx analog preamp headset 23 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Default 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_450 1 voice handset rxag adjust IP_650 1 voice handset txag adjust IP_330 18 voice handset txag adjust IP_430 18 voice handset txag adjust IP_450 18 voice handset txag adjust IP_650 18 voice handset sidetone adjust IP_330 3 voice handset sidetone adjust IP_430 3 voice handset sidetone adjust IP_450 0 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_450 1 voice headset rxag adjust IP_650 1 voice headset txag adjust IP_330 21 voice headset txag adjust IP_430 21 voice headset txag adjust IP_450 21 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_450 3 voice head
142. ITH 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 83 A access URL 4 69 ACD See also automatic call distribution acoustic echo cancellation 4 80 acoustic echo cancellation lt aec gt A 51 acoustic echo suppression lt aes gt A 52 AEC See also acoustic echo cancellation AGC See also automatic gain control alert information A 18 animations lt anim gt A 96 application configuration acoustic echo cancellation A 51 acoustic echo suppression A 52 animations A 96 audio codec preferences A 42 audio codec profiles A 46 automatic gain control A 53 background noise suppression A 53 backgrounds A 91 bitmaps A 95 call handling configuration A 76 call progress patterns A 37 camera controls A 66 chord sets A 33 compliance A 21 conference setup A 19 connection reuse A 20 date and time A 29 dial plan A 21 dial plan emergency A 25 dialog A 20 directory A 81 DNS cache A 120 dual tone multi frequency A 32 encryption A 104 Ethernet call control A 68 event logging A 99 feature A 110 finder A 112 fonts A 86 gains A 47 graphic icons A 98 hold local reminder A 80 idle display A 114 indicator
143. LBC 4 81 audio codec preferences lt codecPref gt A 42 audio codec profiles lt audioProfile gt A 46 audio codecs 4 81 audio playback feature 4 37 A 111 audio recording feature 4 37 A 111 automatic call distribution lt acd gt A 150 automatic gain control 4 83 automatic gain control lt agc gt A 53 automatic off hook call placement 4 19 automatic off hook call placement lt autoOffHook gt A 134 B background logo adding C 6 configuration file changes C 7 background noise suppression 4 82 background noise suppression lt ns gt A 53 backgrounds lt bg gt A 91 Index 2 basic logging A 101 basic protocols header support B 4 hold implementation B 9 request support B 3 response support B 6 RFC and Internet draft support B 2 transfer B 9 basic TCP IP A 71 behaviors lt behaviors gt A 147 blind transfers 4 20 BNS See also background noise suppression boot failure messages 5 8 boot server security policy 3 16 boot servers deploying phones 3 17 redundant 3 14 security policy 3 16 setting up 3 15 bootROM 2 3 bootROM and application wrapper 2 4 bootROM error messages 5 2 bootROM tasks 2 3 bootROM SIP application dependencies C 9 bridged line appearance signaling B 10 bridged line appearances 4 54 browser limits A 116 busy lt busy gt A 137 busy lamp field 4 55 C call control lt callControl gt A 68 call control third party B 9 call forwarding 4 22 A 136 call handling configuration lt call gt A 76
144. Local Config menu selection Background Noise Suppression Background noise suppression BNS is designed primarily for hands free operation and reduces background noise to enhance communication in noisy environments Comfort Noise Fill Configuring Your System There are no related configuration changes Comfort noise fill is designed to help provide a consistent noise level to the remote user of a hands free call Fluctuations in perceived background noise levels are an undesirable side effect of the non linear component of most AEC systems This feature uses noise synthesis techniques to smooth out the noise level in the direction toward the remote user providing a more natural call experience There are no related configuration changes Automatic Gain Control Automatic Gain Control AGC is applicable to hands free operation and is used to boost the transmit gain of the local talker in certain circumstances This increases the 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 signali
145. Management on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones white paper at http www polycom com global documents support technical products voice white_ paper_configuration_file_management_on_soundpoint_ip_phones pdf 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 Video Settings lt video 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 Protocol lt volpProt gt Configuration Files 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 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 e LCD Power Saving lt powerSaving gt This attribute includes e Local lt local gt e Server lt server gt e SIP lt SIP gt Local lt local gt This configuration attribute is
146. More than one line key can be allocated to a single registration Multiple Registrations SoundPoint IP desktop phones and Polycom VVX 1500 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 serv
147. Null 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 Configuration Files Permitted Attribute Values Default Interpretation voice codecPref VVX_1500 G711Mu Null 1 16 4 Specifies the audio codec preferences for the Polycom VVX 1500 phone voice codecPref VVX_1500 G711A 5 Interpretation as above voice codecPref VVX_1500 G722 3 voice codecPref VVX_1500 Null G7221 16kbps voice codecPref VVX_1500 Null G7221 24kbps voice codecPref VVX_1500 2 G7221 32kbps voice codecPref VVX_1500 Null G7221C 24kbps voice codecPref VVX_1500 Null G7221C 32kbps voice codecPref VVX_1500 1 G7221C 48kbps voice codecPref VVX_1500 G729AB 6 voice codecPref VVX_1500 Null Lin16 16ksps voice codecPref VVX_1500 Null Lin16 32ksps voice codecPref VVX_1500 Null Lin16 44_1ksps voice codecPref VVX_1500 Null Lin16 48ksps voice codecPref VVX_1500 Lin16 8ksps Null voice codecPref VVX_1500 Null Siren14 24kbps voice codecPref VVX_1500 Null Siren14 32kbps voice codecPref VVX_1500 Null Siren14 48kbps Note Codecs with a default of Null are available for test purposes only and are not expected to be used in your deployment Administrator
148. Order list of attributes Null The list of attributes in the exact order to be used by the LDAP server when indexing For example sn givenName telephoneNumber Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation dir corp autoQuerySubmitTimeout 0 to 60 seconds 0 To control if there is a timeout after the user stops entering characters in the quick search and if there is how long the timeout is If set to 0 there is not disabled dir corp sortCtrl Oor1 Null Controls how client makes queries and does it sort entries locally It should not be used by customers If set to O or Null leave sorting as negotiated between client and server If set to 1 force sorting of queries Note Polycom does not recommend setting dir corp sortCtrl to 1 as it causes excessive LDAP queries It should be used to diagnose LDAP servers with sorting problems only 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
149. PELOD ooo Initial log entry 0 4 Jan 0 00 00 00 172 23 7 249 0100000000 so 14100 Note that bootrom log times are in GMT a Jan 0 00 00 00 172 23 7 249 0100000000 cfqg 4 00 Initial log entry e Jan 0 00 00 00 172 23 7 249 0100000000 copy 1 3100 Initial log entry 3 Jan 0 00 00 00 172 23 7 249 0100000000 hw 14100 Initial log entry Jan 0 00 00 00 172 23 7 249 0100000000 ethf 4 00 Initial log entry Feb 13 01 12 39 172 23 7 249 0213011239 wdog 4 00 Initial log entry Feb 13 01 12 39 172 23 7 249 0213011239 cdp 3 00 CDP is DISABLED Feb 13 01 12 39 172 23 7 249 0213011239 s0 13 00 Platform Model SoundPoint IP 650 Assembly Z2345 1260 Feb 13 01 12 39 172 23 7 249 0213011239 s0 13 00 Platform Board 2345 12600 001 1 Feb 13 01 12 39 172 23 7 249 0213011239 s0 13100 Platform MAC 0004f2111511 IP Resolving Subnet Mask Feb 13 01 12 39 172 23 7 249 0213011239 s0 13 00 Platform BootBlock 2 7 0 12600_001 30 May 06 15 58 Feb 13 01 12 39 172 23 7 249 0213011239 s0 13100 Application main Label BOOT Version 4 1 0 0219 1o28 Feb 13 01 12 39 172 23 7 249 0213011239 s0 13 00 Application main P N 3150 11063 410 3 Feb 13 01 12 39 172 23 7 249 0213011239 appl 4100 Initial log entry Feb 13 01 12 40 172 23 7 249 0213011240 s0o 13100 Link status is Net down PC down Feb 13 01 12 41 172 23 7 2493 0213011241 so0 13 00 Link status is Net up Speed 100 half Duplex PC down Feb 13 01 12 41 172 23 7 249 0213011Z24lilcdp 3 00 CDP is disabled
150. PIP Special Events lt specialEvent gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP specialEvent lineSeize Oori 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 Oor1 0 If set to 1 always reboot when a NOTIFY checkSync alwaysReboot message is received from the server with event equal to check sync If set to 0 only reboot if any of the files listed in lt MAC address gt cfg have changed on the FTP server when a NOTIFY message is received from the server with event equal to check sync Conference Setup lt conference gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP conference address ASCII string Null If Null conferences are set up on the phone up to 128 locally characters If set to some value conferences are set up long by the server using the conferencing agent specified by this address The acceptable values depend on the conferencing server implementation policy A 19 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Dialog lt dialog gt This configuration attribute is defined as follows
151. Phone s Current Screen on page C 29 Refer to the appropriate SoundPoint IP SoundStation IP VVX SIP phone User Guide Reboot the phone to obtain a default level of contrast refer to Rebooting the Phone on page C 10 Use the screen capture feature Refer to Capturing Phone s Current Screen on page C 29 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 Use the screen capture feature Refer to Capturing Phone s Current Screen on page C 29 The display is flickering Certain type of older fluorescent lighting causes the display to appear to flicker Do one of the following Move the SoundPoint IP SoundStation IP VVX SIP phone away from the lights Replace the lights Use the screen capture feature Refer to Capturing Phone s Current Screen on page C 29 Administrator s Guide SoundPoint IP SoundStation IP VVX Audio Symptom Problem Corrective Action There is no audio on the headset The connections are not correct Do one of the following Ensure the headset is plugged into the jack marked Headset at the rear of the phone Ensure the headset amplifier if present is t
152. Point IP 300 301 500 501 600 and 601 and SoundStation IP 4000 phones or all of the lt MACaddress gt cfg files if it is not explicitly known which phones are SoundPoint IP 300 and 500 phones For more information refer to Technical Bulletin 35311 Supporting SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones with SIP 2 2 0 or SIP 3 2 0 and Later Releases at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuring Your System After you set up your SoundPoint IP SoundStation IP VVX phones on the network you can allow users to place and answer calls using the default configuration however you may 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 Setting Up Video Features Setting Up Security Features This chapter also provides instructions on e Configuring SoundPoint IP SoundStation IP VVX Phones Locally To troubleshoot any problems with your SoundPoint IP SoundStation IP VVX phones on the network refer to Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones on page 5 1 For more information on the configuration files refer to Configuration Files on page A 1
153. SIP Server Fallback Enhancements on SoundPoint IP Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuration changes can be performed centrally at the provisioning server Ceniral provisioning 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 7 Configuration file Specify per registration primary and fallback server configuration phonet cfg parameters values that override those in sip cfg e For more information refer to Registration lt reg gt on page A 128 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 for the SoundPoint IP SoundStation IP VVX Family 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 Fallbac
154. SPIP300 cfg gt lt APPLICATION APP_FILE_PATH_SPIP500 SPIP500 sip 1ld CONFIG_FILES_SPIP500 phonel_SPIP500 cfg sip_SPIP500 cfg gt For more information e Refer to Technical Bulletin 35311 Supporting SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones with SIP 2 2 0 or SIP 3 2 0 and Later Releases at http www polycom com usa en support voice soundpoint_ip Vol P_Technical_Bulletins_pub html e 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 Vol P_Technical_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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 Polycom recommends that you create another file with your organization s modifications If you must change any Polycom templates back them up first POLYCOM For more information refer to the Configuration File
155. Setting up Your System on page 3 1 To configure your SoundPoint IP SoundStation IP VVX phones with the desired features Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family refer to Configuring Your System on page 4 1 To troubleshoot any problems with your SoundPoint IP SoundStation IP VVX phones on the network refer to Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones on page 5 1 Where SoundPoint IP SoundStation IP VVX 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 As SoundPoint IP SoundStation IP VVX phones run the same protocols as your office personal computer many innovative applications can be developed without resorting to specialized technology Remote a i Boot Server Internet PSTN og N i emote A J il a Y Server Qa N Router SY Ethernet Switches Ss rs ill 10 100 Ethernet Switch Voice Bridge i Local Application Server Or Local 10 100 Boot Server Ethernet Hub Overview Session Initiation Protocol Application Architecture BootROM The software architecture of SIP application is made
156. SoundPoint IP SoundStation IP VVX Family Warning The SoundPoint IP 300 and 500 phones will be supported on the latest maintenance patch release of the SIP 2 1 software stream currently SIP 2 1 4 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 The SoundPoint IP 301 501 600 and 601 and the SoundStation IP 4000 phones will be supported on the latest maintenance patch release of the SIP 3 1 software stream currently SIP 3 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 or later for these changes to be effective Supporting SoundPoint IP SoundStation IP and Polycom VVX Phones Warning To automatically update 1 Back up old application and configuration files The old configuration can be easily restored by reverting to the backup files 2 Customize new configuration files or apply new or changed parameters to the old configuration files Differences 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 c
157. Station IP VVX Family e SoundPoint IP 650 e SoundPoint IP 670 SoundStation IP Conference Phones This section describes the current SoundPoint IP conference phones For individual guides refer to the product literature available at http www polycom com support voicedocumentation Additional options are also available For more information contact your Polycom distributor Introducing the SoundPoint IP SoundStation IP VYX Family The currently supported conference phones are e SoundStation IP 6000 e SoundStation IP 7000 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Polycom VVX 1500 Business Media Phone This section describes the current Polycom VVX 1500 business media phone For the individual guide refer to the product literature available at http www polycom com support voicedocumentation Additional options are also available For more information contact your Polycom distributor Key Features of Your SoundPoint IP SoundStation IP VVX Phones The key features of the SoundPoint IP SoundStation IP VVX phones are e Award winning sound quality and full duplex speakerphone or conference phone Permits natural high quality two way conversations Uses Polycom s industry leading Acoustic Clarity Technology e Easy to use An easy transition from traditional PBX systems into the world of IP Upto 18 dedicated hard keys for access to commonl
158. This character delimits the commands within the macro 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 Macro names and action strings cannot contain these characters If they do unpredictable results may occur Macro Definition The action string in the efklist element can be defined by either Configuring Your System 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 following 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 inc
159. _313 1d FIG_FILES_SPIP600 phonel_313 cfg sip_313 cfg gt lt APPLICATION_SPIP601 APP FILE PATH SPIP601 sip_313 1d FIG_FILES_SPIP601 phonel_313 cfg sip_313 cfg gt lt APPLICATION_SSIP4000 APP_FILE_PATH_SSIP4000 sip_313 1d FIG_FILES_SSIP4000 phonel_313 cfg sip_313 cfg gt lt APPLICATION_VVX1500 APP_FILE_PATH_VVX1500 sip 1ld FIG_FILES_VVX1500 phonel_313 cfg sip_313 cfg gt lt APPLICATION gt Master configuration files contain the following 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 1d CONFIG_FILES A comma separated list of configuration files Each file name has a maximum length of 255 characters and the list of file names has a maximum length of 2047 characters including commas and white space Each configuration file can be specified as a URL with its own protocol user name and password for example ftp usr pwd server dir phone2034 cfg 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 provisioning server is unavailable it will still be able to load the preferred language LOG_FILE_DIRECTORY An alternativ
160. a lot of confusion about where parameters are being set and so it is best to avoid using the manual method unless you have good reason to do so Resource Files 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 e Ring tones e Synthesized tones e Contact directories Note If you need to remove the resource files from a phone at some later date for example you are giving the phone to a new user instructions on how to put the phone into the factory default state can be found in Quick Tip 18298 Resetting and Rebooting SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at http Awww polycom com support voice soundpoint_ip VoIP_Technical_Bulletins_p ub html Available Features This section provides information about the features available on the SoundPoint IP SoundStation IP VVX phones e 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 s
161. aa E teens wee aera 4 17 Graphic Display Backgrounds 0 000000 4 17 Automatic Off Hook Call Placement 00000 4 19 Call HONG noite be 35 ot nated sad eee AS ye T RR ewes 4 19 Call Transter sicevts tie wha ede ev e et eats Baa nine ad ace STs 4 20 Local Centralized Conferencing 0000000 4 21 Call FORwatd serores tunadi ia Wego Mea seen ea aie 4 22 Directed Call Pick Up 0 cece eee erences 4 24 Group Call Pick Up crci riras ia E e e e eee 4 24 Call Park Retrieve doa ead eee daa E EA E EREE 4 24 Last Call Refur sse resereerei ewig ac seeiprg ween EEE E EE 4 25 Setting Up Advanced Features 00 0 00008 4 25 Configurable Feature Keys 0 000 eee eee eee eee eee 4 26 Multiple Line Keys per Registration 0 4 27 Multiple Call Appearances 0 cece ee 4 28 Customizable Fonts and Indicators 0000 4 28 vi Contents Instant Messaging rania arada nA Ret kane kde ah es 4 29 Multilingual User Interface 0 0 0000000 4 29 Downloadable Fonts 0 0 2 e eee 4 30 Synthesized Call Progress Tones 0 00 e ee eee 4 30 MicrobrowSser meiss ynn soe autos ok o Gea p Cena a oases 4 31 Application Launch Pad 0 cece eee 4 32 Real Time Transport Protocol Ports 0 000 4 33 Network Address Translation 6 60 06 c cece ee eens
162. administrator password refer to Configuring SoundPoint IP SoundStation IP VVX Phones Locally on page 4 94 Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify the minimum lengths for the user and administrator provisioning sip cfg passwords server For more information refer to Password Lengths lt pwd gt lt length gt on page A 104 Local Web Server None if enabled Local Phone User The user and administrator passwords can be changed under the Interface Settings menu or through configuration parameters refer to Flash Parameter Configuration on page A 151 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 provisioning server for security reasons 4 88 Custom Certificates Configuring Your System The phone trusts certificates issued by widely recognized certificate authorities when trying to establish a connection to a provisioning server for application provisioning Refer to Trusted Certificate Authority List on page C 1 In addition custom certificates can be added to the phone This is done by using the SSL Security menu on the phone to provide the URL of the custom certificate then select an option to use this custom certificate Note For more information on using custom certificates refer to
163. ailable Capabilities 0x37 LLDP Med capabilities Network policy Location Identification Extended Power Via MDI PD Inventory Class Type III 11 LLDP MED network policy 127 Oxfe08 0x0012bb ApplicationType Voice 1 Policy Unknown 1 Defined 0 Unknown if phone is in booting stage or if switch doesn t support network policy TLV Defined if phone is operational stage and Netwokpolicy TLV is received from the switch Tagged Untagged Vlanld L2 priority and DSCP Administrator s Guide SoundPoint IP SoundStation IP VVX No Name Type 7 bits 0 6 Length 9 bits 7 15 Type Length Org Unique Code 3 bytes Version Sub Type Information 12 LLDP MED network policy 127 8 Oxfe08 0x0012bb 2 ApplicationType Voice Signaling 2 Policy Unknown 1 Defined 0 Unkknown if phone is in booting stage or if switch doesn t support network policy TLV Defined if phone is operational stage and Netwokpolicy TLV is received from the switch Tagged Untagged Vianld L2 priority and DSCP Note Voice signaling TLV is sent only if it contains configuration parameters that are different from voice parameters 13 LLDP MED network policy 127 Oxfe08 0x0012bb ApplicationType Video Conferencing 6 Policy Unknown 1 Defined 0 Unkknown if phone is in booting stage or if sw
164. akes provisioning server management easier After encrypting a configuration file it is useful to rename the file to avoid confusing it with the original version for example rename sip cfg to sip enc However the directory and override filenames cannot be changed in this manner You can check whether an encrypted file is the same as an unencrypted file by 1 Run the configFileEncrypt utility 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 provisioning server provides an encrypted file that can be read an unencrypted file or the file is removed from the master configuration file list Encrypted configuration files can only be decrypted on the SoundPoint IP 32x 33x 430 450 550 560 650 and 670 the SoundStation IP 6000 and 7000 phones and the Polycom VVX 1500 phones The master configuration file cannot be encrypted on the provisioning 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 re
165. al Configuration file Specify whether to allow a transfer during the proceeding state of a provisioning sip cfg consultation call server e For more information refer to SIP lt SIP gt on page A 11 Specify whether a transfer is blind or not e For more information refer to Call Handling Configuration lt call gt on page A 76 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 All phones support three party local conferencing The SoundPoint IP 450 550 560 650 and 670 phones may support four way local conferencing Note Four party conferencing 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 If the initiator of a three party local conference ends the call the other members of the call may still communicate If the initiator of a four party local conference ends the call the conference ends The phone also supports centralized conferences for which external resources are used such as a conference bridge This relies on network signaling Configuration changes can be performed centrally at the provisioning server Central
166. alues 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 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 25 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 to
167. am is included in SDP of a SIP invite sec srtp require Oori Null If set to 1 the phone is only allowed to use secure media streams Any offered SIP INVITEs must include a secure media description in the SDP or the call will be rejected For outgoing calls only a secure media stream description is include in the SDP of the SIP INVITE meaning that the non secure media description is not included If sec srtp require is set to 1 sec srtp offer is logically set to 1 no matter what the value in the configuration file If set to O or Null secure media streams are not required sec srtp offer HMAC_SHA1_80 Oor1 Null If set to 1 or Null a crypto line with the AES_CM_128 HMAC_SHA 1_80 crypto suite will be included in offered SDP If set to 0 the crypto line is not included Note This attribute was added in SIP 2 2 1 sec srtp offer HMAC_SHA1_ 32 Oor1 Null If set to 1 a crypto line with the AES_CM_128_HMAC_SHA1_32 crypto suite will be included in offered SDP If set to O or Null the crypto line is not included Note This attribute was added in SIP 2 2 1 A 105 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute sec srtp key lifetime Permitted Values 0 positive integer minimum 1024 Default Null Interpretation The master key lifetime used for the cryptographic attribute in the SDP The value specified is
168. ame as per SIP 2 1 per registration feature with the following exceptions e Server based DND cannot be used if the phone is configured as a shared line 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 provisioning server or locally Central provisioning server Configuration file Enable or disable server based DND For more information refer to SIP lt SIP gt on page A 11 Enable or disable local DND behavior when server based enabled e For more information refer to SIP lt SIP gt on page A 11 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 76 Configuration file Enable or disable server based DND as a per registration feature phonet cfg e For more information refer to Registration lt reg gt on page A 128 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 138 Local Local Phone User Enable or disable DND using the Do Not Disturb key on the SoundPoint IP 550 560 650 and 670 and the Polycom VVX 1500 or the Do Not
169. ameters 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 for the SoundPoint IP SoundStation IP VVX Phones Note You can make changes to the configuration files through the web interface to the phone Using your chosen browser enter the phone s IP address as the browser address For more information refer to Modifying Phone s Configuration Using the Web Interface on page C 26 Changes made through the web interface are written to the override file highest priority These changes remain active until Reset Local Config is performed Master Configuration Files The master configuration files can be one of Specified master configuration file The master configuration file can be explicitly specified in
170. ance RFC3261 Oori Null If set to 1 validation of the SIP header validate contentLanguage content language is enabled If set to 0 or Null validation is disabled volpProt SIP compliance RFC3261 Oor1 Null If set to 1 or Null validation of the SIP header validate contentLength content length is enabled If set to 0 validation is disabled volpProt SIP compliance RFC3261 Oor1 Null If set to 1 or Null validation of the SIP header validate uriScheme URI scheme is enabled If set to 0 validation is disabled Dial Plan lt dialplan gt Note The dial plan is not applied against Placed Call List VoiceMail last call return remote control dialed numbers and on hook dialing This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan applyToCallListDial Oori 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 digit map replacement operations are not applied against the dialed number if set to 1 the digit map replacement operations are applied against the dialed number dialplan applyToDirectoryDial Oor1 This attribute covers dialing from Directory as well as Speed Dial List Value interpretation is the same as for dialplan applyToCallListDial Note An Auto Call Contact number is considered a dial from directory Administrator s Guid
171. anch 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 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 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
172. and SoundStation IP 4000 and the configuration files have not been correctly modified These phones 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 300 301 500 501 600 601 4000 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 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones on page 3 23 Administrator s Guide SoundPoint IP SoundStation IP VVX 5 18 Configuration Files Warning Note Note This appendix provides detailed descriptions of certain configuration files used by the Session Initiation Protocol SIP application It is a reference for all parameters that are configurable when using the centralized provisioning installation model This appendix contains information on e Master Configuration Files MAC address cfg or 000000000000 cfg e Application Configuration sip cfg e Per Phone Configuration phonel cfg e Flash Parameter Configuration The application configuration files dictate the behavior of the phone once it is running the executable specified in the master configuration file Configuration files should only be modified by a knowledgeable system administrator Applying incorrect par
173. asses for the LED and graphical icon indicator types In the following table x is the class number y is the identifier of the state number for that class Attribute Permitted Values Interpretation ind class x state y index positive integer For LED type indicators index refers to the pattern index such as index x in the Patterns lt pattern gt tag above For Graphic Icon type indicators index refers to the animation index such as index y in the Animations lt anim gt lt IP_330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt tag above Assignments This attribute assigns a type and a class to an indicator In the case of the Graphic Icon type it also assigns a physical location and size in pixels on the LCD display refer to the next section In the case of the LED type it assigns a physical LED number refer to Graphic Icons lt gi gt lt IP_330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt on page A 98 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute LEDs lt led gt In the following table x is the LED number Permitted 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 clas
174. ation refer to Automatic Off Hook Call Placement lt autoOffHook gt on page A 134 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 4 19 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify whether RFC 2543 c 0 0 0 0 or RFC 3264 a sendonly or provisioning sip cfg a inactive outgoing hold signaling is used server For more information refer to SIP lt SIP gt on page A 11 Specify local hold reminder options For more information refer to Hold Local Reminder lt hold gt lt localReminder gt on page A 80 Specify the Music on Hold URI For more information refer to Music on Hold lt musicOnHold gt on page A 20 Configuration file Specify the Music on Hold URI phonet cfg For more information refer to Music on Hold lt musicOnHold gt on page A 20 Local Web Server Specify whether or not to use RFC 2543 c 0 0 0 0 outgoing hold if enabled
175. ation File Changes e Useful Tips e Examples For more examples including sample configuration files refer to Technical Bulletin 42250 Using Enhanced Feature Keys and Configurable Soft Keys on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html 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 following 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 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 Note In SIP 3 2 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 co
176. 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 30 If set to 0 there is no timeout If set to Null the default timeout of 40 seconds is used If set to value greater than 0 the timeout is for that number of seconds maximum 65535 Configuration Files Permitted Attribute Values Default Interpretation up mwiVisible Oor1 0 If set is O or Null the incoming MWI notifications for lines where the MWI callback mode is disabled msg mwi x callBackMode is set to 0 are ignored and do not appear in the message retrieval menus If set to 1 the MWI for lines whose MWI is disabled is displayed pre SIP 2 1 behavior even though MWI notifications have been received for those lines up handsfreeMode Oor 1 1 If set to 1 or Null hands free speakerphone is enabled If set to 0 hands free speakerphone is disabled up numberFirst CID Oor 1 0 If set to 0 or Null caller ID display will show caller s name first If set to 1 caller ID display will show caller s number first up idleBrowser enabled Oor1 0 A flag to determine whether or not the back
177. ault Interpretation 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 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 Oor1 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 SIP lt SIP gt This configuration attribute is defined as follows Attribute Permitted Values Default Configuration Files Interpretation volpProt SIP useContactInReferTo Oori 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 Oori If set to 1 use SDP media direction attributes such as a sendonly per RFC 3264 when initiating hold otherwise use the obsolete c 0 0 0 0 RFC2543 technique In either case the phone processes incoming hold signaling in either format volpProt SIP useSendonlyHold Oori If set to 1 the phone will
178. aviors attributes are ignored attendant reg attendant ringType positive integer 1 to 22 For attendant console BLF feature This is the index of 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 The ring tone to play when a BLF dialog is in the offering state This attribute also includes e Resource List lt resourceList gt e Behaviors lt behaviors gt Resource List lt resourceList gt In the following table x is the monitored user number For IP 450 x 1 2 IP 550 IP 560 X 1 3 IP 650 IP 670 x 1 47 This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation attendant resourceList x address string that Null The user referenced by constitutes a valid attendant reg will subscribe to this SIP URI URI for dialog If a user part is present the sip 6416 polyco phone will subscribe to a sip URI m com or contains constructed from user part and the domain the user part of a of the user referenced by attendant reg SIP URI 6416 attendant resourceList x label UTF 8 encoded Null Text label to appear on the display string adjacent to the associated line key If set to Null the label will be derived from the user part of attendant resourceList x address A 146 Configuration Files
179. ay enabled 1 gt lt Animations 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_400 gt lt IP_450 gt lt IDLE_DISPLAY ind anim IP_450 45 frame 1 bitmap logo 450 ind anim IP_450 45 frame 1 duration 0 gt lt IP_450 gt lt IP_600 gt lt IDLE_DISPLAY ind anim IP_600 46 frame 1 bitmap logo 650 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 6000 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 7000 ind anim IP_7000 46 frame 1 duration 0 gt Miscellaneous Administrative Tasks lt IP_7000 gt 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 VVX 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 320 330 3 2 3 B
180. be used A 138 Configuration Files Dial Plan lt dialplan gt Per registration dial plan configuration is supported In the following table x is the registration number IP 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Permitted Attribute Values Default Interpretation dialplan x applyToCallListDial Oori 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 21 dialplan x apply ToDirectoryDial Oor1 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 21 dialplan x apply ToUserDial Oor1 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 21 dialplan x applyToUserSend Oor1 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 21 dialplan x impossibleMatchHandling 0 1or2 0 Wh
181. bility 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 SIP B Automatic Call Distribution Supports ACD agent available and unavailable and allows ACD login and logout Requires call server support Bridged Line Appearance Calls and lines on multiple phones can be logically related to each other Requires call server support Busy Lamp Field Allows monitoring the hook status and remote party information of users through the busy lamp field BLF LEDs and displays on an attendant console phone This feature may require call server support Configurable Feature Keys Certain key functions can be changed from the factory defaults Overview 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 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 Display of Warnings from SIP Headers Displays a pop up to user that is found in the Warning Field from a SIP header D
182. ble or disable VAD and set the detection threshold For more information refer to Voice Activity Detection lt vad gt on page A 57 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 be performed centrally at the provisioning server Central Configuration file Set the DTMF tone levels autodialing on and off times and other provisioning sip cfg parameters server For more information refer to Dual Tone Multi Frequency lt DTMF gt on page A 32 DTMF Event RTP Payload The phone is compatible with RFC 2833 RTP Payload for DTMF Digits Telephony Tones and Telephony Signals RFC 2833 describes a standard RTP compatible technique for conveying DTMF dialing and other telephony events over an RTP media stream The phone generates RFC 2833 DTMF only events but does not regenerate nor otherwise use DTMF events received from the remote end of the call Configuration changes can be performed centrally at the provisioning server Central Configuration fil
183. bly The user can view the messages immediately or when it is convenient For sending messages the user can either select a message from a preset list of short messages or an alphanumeric text entry mode allows the typing of custom messages using the dial pad Message sending can be initiated by replying to an incoming 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 provisioning server server Central Configuration file Turn this feature on or off provisioning sip cfg For more information refer to Feature lt feature gt on page A 110 Multilingual User Interface 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 higher resolution displays of the SoundPoint IP 450 550 560 650 and 670 SoundStation IP 6000 and 7000 and Polycom VVX 1500 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 450 550 560 650 and 670 and SoundStation IP platforms
184. cal 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 provisioning 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 provisioning 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 provisioning server They will permanently override global settings unless deleted through the Reset Local Config menu selection Idle Display Animation Note All phones 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 Currently customized animations are not supported on the Polycom VVX 1500 Configuring Your System Configuration changes can be performed centrally at the provisioning server Central Configuration file To turn idle display ani
185. 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 Guide for the SoundPoint IP SoundStation IP VVX Phones A 38 Call progress pattern number Use within phone 7 alternate call waiting tone distinctive 8 confirmation tone 9 howler tone off hook warning 10 record warning 11 message waiting tone 12 alerting 13 intercom announcement tone 14 barge in tone 15 secondary dial tone Ringer Patterns The following table maps ringer pattern numbers to their default descriptions Ringer pattern number Default description 1 Silent Ring 2 Low Trill 3 Low Double Trill 4 Medium Trill 5 Medium Double Trill 6 High Trill 7 High Double Trill 8 Highest Trill 9 Highest Double Trill 10 Beeble 11 Triplet 12 Ringback style 13 Sampled audio file 1 14 Sampled audio file 2 15 Sampled audio file 3 16 Sampled audio file 4 17 Sampled audio file 5 Note Configuration Files
186. col lt volpProt gt 2 0 0 ccc cece A 7 Dial Plan lt dialplan gt acesscneitadesrteeestpiwetanas odes bos A 21 Localization lt Icl gt 2 0 0 cece eee e eens A 25 User Preferences lt up gt 0 c cece cece eee eee A 29 Tones lt tOnes Pe ca c destin hee LER G Ee Beek TEAR REAA A 31 Sampled Audio for Sound Effects lt saf gt 06 A 34 Sound Effects lt s gt l kere nteer hnr nren ei ee ae do cedar A 35 Voice Settings lt voice gt 0 0 eee eee A 41 Video Settings lt video gt 0 0 00 A 61 Quality of Service lt QOS gt 00 0 A 67 Basic TCP IP lt TCP IP gt terpeene orket epai piei nabbed A 71 Web Server lt httpd gt 0 0 cee eee eee A 75 Call Handling Configuration lt call gt 0 e eee A 76 Directory lt dit gt r 0 054 bots seek hate aah ie ta E A 81 PLreSENCE SPTES gt roen atiet aiio AE K RAEE EEEE ite A 86 Fonts Tona pierrier ienie Era AERA E RAE aeds A 86 Keys Skey gt eenei noen nPE Coe EEIN E T OEA EAE A 89 Backgrounds lt bg gt e rges espireiscn casas eee eee eee A 91 Bitmaps lt bitmap gt 0 A 95 Indicators Sind gt airean ERER tne AEEA E EEr ERARAS A 95 Event Logging lt l6G gt serises veisi egos EERE EETA AERES A 99 Security SSOC gt 3 uieii ie a E E EEE E D EEEE A 103 License lt license gt cis eck Ciir ki ia he eek eaten eta eke A 107 Provisioning lt prov gt 66 A 108 RAM Disk l
187. com 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 SIP 3 2 is to provide the ability to 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 a
188. cord can be added This attribute includes e NAPTR lt NAPTR gt attribute e SRV lt SRV gt A lt A gt 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 must 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
189. correctly registered to the server Press the Menu key followed by System Status and Network Statistics Scroll down to see if LAN port shows active or Inactive Check the termination at the switch or hub end of the network LAN cable Ensure that the switch hub port connected to the telephone is operational if not accessible contact your system administrator Before restarting your phone contact your system administrator since this may allow more detailed troubleshooting to occur before losing any current status information 5 12 Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones Access to Screens and Systems Symptom Problem Corrective Action There is no response from The SoundPoint IP Do one of the following feature key presses SoundStation IP VVX family e Press the keys more slowly SIP phone ie not in active state e Check to see whether or not the key has been mapped to a different function or disabled e Make a call to the phone to check for inbound call display and ringing as normal If successful try to press feature keys within the call to access Directory or Buddy Status for example e 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 The LAN cable is not properly
190. creasing the total number of registrations to 34 The SoundStation IP 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 4 57 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify the local SIP signaling port and an array of SIP servers to provisioning sip cfg register to For each server specify the registration period and the server signaling failure behavior For more information refer to Local lt local gt on page A 7 and Server lt server gt on page A 8 Configuration file For up to maximum number of registrations specify a display name phonet cfg a SIP address an optional display label an authentication user ID and password the number of line keys to use and an optional array of registration servers The authentication user ID and password are optional and for security reasons can be omitted from the configuration files The local flash parameters will be used instead The optional array of servers and their associated parameters will override the servers specified in sip cfg if non Null For more information refer to Registration lt reg
191. cription Example 011511 006 so 2 00 soCoreAudioTermChg chassis gt idle time stamp f ID event class missed events text Three formats are available for the event timestamp Type Example 0 seconds milliseconds 011511 006 1 hour 15 minutes 11 006 seconds since booting 1 absolute time with minute resolution 0210281716 2002 October 28 17 16 2 absolute time with seconds resolution 1028171642 October 28 17 16 42 Two types of logging are supported e Basic Logging lt level gt lt change gt and lt render gt e Scheduled Logging Parameters lt sched gt A 100 Configuration Files Basic Logging lt level gt lt change gt and lt render gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation log level change xxx 0 5 4 Control the logging detail level for individual components These are the input filters into the internal memory based log system Possible values for xxx are so app1 sip sspsc ssps pps net cfg cdp pmt ftp ares dns cxss httpd rdisk copy slog res key log curl rtos mb ib sotet ttrs srtp usb efk clink Idap and peer pnetm cmp cmr usbio pres pwrsv and Ildp 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 ma
192. cryption of RTP is required A call placed to a phone configured with noAuth require must offer the UNENCRYPTED_SRTP session parameter in its SDP If sec srtp sessionParams noEncryptRIP requir e is set to 1 sec srtp sessionParams noEncryptRIP offer is logically set to 1 no matter what the value in the configuration file If set to 0 or Null encryption of RTP is required sec srtp requireMatching Tag Oor1 Null A flag to determine whether or not to check the tag value in the crypto attribute in an SDP answer If set to 1 or Null the tag values must match If set to 0 the tag value is ignored License lt license gt This attribute s settings control aspects of the feature licensing system This configuration attribute is defined as follows A 107 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation license polling time 00 00 23 59 2 00am The time to check whether or not the license has expired Provisioning lt prov gt This attribute s settings control aspects of the phone s provisioning server provisioning system Permitted Attribute Values Default Interpretation prov fileSystem rfs0 minFreeSpace 5 512 5 Minimum free space in Kbytes to reserve in the file system when downloading files from the 48 provisioning server prov fileSystem ffs0 4meg minFreeSpace 420
193. curity troubleshooting Application is not compatible 5 2 application error messages 5 3 application logging options 5 5 audio issues 5 16 blinking time 5 4 boot failure messages 5 8 bootROM error messages 5 2 calling issues 5 14 Config file error Error is 5 3 controls issues 5 12 Could not contact boot server 5 2 displays issues 5 15 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 11 productivity suite 5 16 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 13 trusted certificate authority list C 1 type length values type of service bits 4 83 U uaCSTA A 13 A 128 B 9 upgrading SIP application 3 21 USB device 4 37 USB devices supported 4 37 user interface soft key activated 4 14 user preferences lt up gt A 29 v VAD See also voice activity detection video lt video gt A 61 video codec preferences lt codecPref gt A 62 video codec profiles lt profile gt A 63 VLAN ID using DHCP C 22 voice activity detection 4 79 voice activity detection lt vad gt A 57 voice mail integration 4 56 voice quality monitoring 4 84 A 58 voice setting lt voice gt A 41 volume persistence lt volume gt A 47 WwW web server lt httpd gt A 75 welco
194. d 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 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
195. d 7000 and the Polycom VVX 1500 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 Oor1 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Permitted Values Default Interpretation 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 content is displayed immediately mb LaunchPad enabled Oori 1 If set to 1 the application launch pad is enabled If set to O or Null the application launch pad is disabled Note This feature is supported on the Polycom VVX 1500 only This attribute also includes e Idle Display lt idleDisplay gt e Main Browser lt main gt e Browser Limits lt limits gt Idle Display lt idleDisplay gt The Microbrowser can be used to create a display that will be part of the phone s idle display These s
196. d certain key combinations depending on the phone model simultaneously during the countdown process in the bootROM until the password prompt appears e IP 450 550 600 601 and 650 and 670 and VVX 1500 4 6 8 and dial pad keys e IP 32x 33x 430 560 7000 1 3 5 and 7 dial pad keys e IP 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 hold certain key combinations depending on the phone model simultaneously until a confirmation tone is heard or for about three seconds e IP 32x 33x Menu Dial and the two Line keys e IP 430 450 550 560 600 601 650 670 and 7000 and VVX 1500 Up Down Left and Right arrow keys e IP 6000 Menu Exit Off hook Hands free Redial Administrator s Guide SoundPoint IP SoundStation IP VVX Default Feature Key Layouts The following figures and tables show the default SIP key layouts for the SoundPoint IP 32x 33x 430 450 550 560 650 and 670 SoundStation IP 6000 and 7000 and Polycom VVX 1500 models SoundPoint IP 320 321 330 331 la cp lt a L4 Menu Litine 1 a gt 32 15 33 Py wa 2 Ne w W716 lt gt 24 23 z o 9 O N Key ID Key ID Fu
197. d to the attendant behaviors display Oor1 1 attendant If set to 0 disabled remoteCallerlID automata the string unknown would be substituted for both name and number information A 147 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Roaming Buddies lt roaming_buddies gt Note This attribute is used in conjunction with Microsoft Live Communications Server 2005 only This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation roaming_buddies reg positive Null Specifies the line registration number which has roaming integer buddies support enabled If Null roaming buddies is disabled If value lt 1 then value is replaced with 1 Warning This parameter must be enabled value gt 0 if the call server is Microsoft Live Communications Server 2005 Roaming Privacy lt roaming_privacy gt Note This attribute is used in conjunction with Microsoft Live Communications Server 2005 only This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation roaming_privacy reg positive 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 A 148 Configuration Files User Preferences lt up gt Note The Digital Picture Frame
198. dPoint IP Configuration Home General Network SIP Lines Configuration Updated Please wait a few seconds for your phone to reconfigure and restart TET TTT Your phone will reboot g Select General from the menu tab A web page similar to the one shown below appears General Configuration Parameters User Preferences Time Audio Processing Video Processing Background Sampled Audio Microbrowser Logging Applications Headset Memory Mode Use Directory Names One Touch Voice Mail Bypass Instant Message Welcome Sound All Boot Welcome Sound arm Boot h If you make any changes scroll down to the bottom of the section i Select the Submit button Your phone will reboot Miscellaneous Administrative Tasks Capturing Phone s Current Screen You can capture the current screen on a SoundPoint IP SoundStation IP or Polycom VVX phone through the web interface to the phone To capture the phone s current screen 1 Modify the sip cfg configuration file as follows a Open sip cfg in an XML editor b Locate the user preference parameter c Change up screenCapture enabledto 1 d Save the modified sip cfg configuration file 2 On the phone do the following a Press the Menu key and then select Settings gt Basic gt Preferences gt Screen Capture b Using the arrow keys select Enabled and then press the Select soft key Note You wi
199. ddress gt directory cfg If so configured the first and last name fields of the local contact 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 Configuring Your System Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Set whether the directory uses volatile storage on the phone provisioning sip cfg e For more information refer to Local Directory lt local gt on page server A 81 Specify whether or not the local contact directory is read only For more information refer to Local Directory lt local gt on page A 81 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
200. deo shown in a call or conference Use motion for people or other video with motion Use sharpness or Null for video with little or no movement Moderate to heavy motion can cause some frames to be dropped Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation video maxCallRate 128 1024 Null Limits the maximum network bandwidth used in a kbps call It is used in the SDP bandwidth signaling If honored by the far end both Rx and Tx network bandwidth used in a call will not exceed this value in kbps If set to Null the value 448 is used video autoFullScreen Oor1 Null Flag to determine whether or not video calls use the full screen layout If set to 1 video calls will use the full screen layout by default When a video call is first created upon discovery that far end is video capable or when an audio call transitions to a video call through far end transfer the full screen layout will be used If set to 0 or Null video calls only use the full screen layout if it is selected by the user These attributes also include e Video Coding Algorithms lt codecs gt e Camera Controls lt camera gt e Local Camera View lt localCameraView gt Video Coding Algorithms lt codecs gt These codecs include e Codec Preferences lt codecPref gt e Codec Profiles lt profile gt Codec Preferences lt codecPref
201. 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 provisioning server so that the phone will replace them when it successfully boots Adding a Background Logo Note Background logos are not supported on the Polycom VVX 1500 phone 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 Model Width Height Color Depth IP 32x 33x 102 23 monochrome IP 430 94 23 monochrome IP 450 256 116 4 bit grayscale or monochrome IP 550 560 650 209 109 4 bit grayscale or monochrome Miscellaneous Administrative Tasks Model Width Height Color Depth IP 670 209 109 12 bit color 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 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 450 550 560 650 phone support a 4 bit grayscale which is a smooth
202. dset is attached For more information refer to User Preferences lt up gt on page A 128 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection Local Contact Directory Note The phone maintains a local contact directory The directory can be downloaded from the provisioning 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 The maximum number of entries in the local contact directory is phone dependent If a user makes a change to the local contact directory there is a five second timeout before it is uploaded to the provisioning server as lt mac a
203. e 4 42 This parameter must have a value and it cannot be Null Note Configuring Your System 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 2 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 set 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 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 o
204. e Enable or disable RFC 2833 support in SDP offers and specify the provisioning sip cfg payload value to use in SDP offers server For more information refer to Dual Tone Multi Frequency lt DTMF gt on page A 32 Acoustic Echo Cancellation The phone employs advanced acoustic echo cancellation AEC for hands free operation Both linear and non linear techniques are employed to aggressively reduce echo yet provide for natural full duplex communication patterns When using the handset on any SoundPoint IP phones AEC is not normally required In certain situations where echo is experienced by the far end party when the user is on the handset AEC may be enabled to reduce avoid this echo To achieve this make the following changes in the sip cfg configuration file default settings for these parameters are disabled voice aec hs enable 1 voice aes hs enable 1 voice ns hs enable 1 voice ns hs signalAttn 6 voice ns hs silenceAttn 9 Configuring Your System For more information refer to Acoustic Echo Cancellation lt aec gt on page A 41 Acoustic Echo Suppression lt aes gt on page A 52 and Background Noise Suppression lt ns gt on page A 53 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 Audio Codecs The following table shows which audio codecs are support by each of the SoundPo
205. e 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 video profile H263 jitterBufferShrink video profile H263 CifMpi 33ms to 1000ms default 70ms 1 default to 32 The absolute minimum duration time in milliseconds of RTP packet Rx with no packet loss between jitter buffer size shrinks Use smaller values 33 ms to minimize the delay on known good networks Use larger values 1000ms to minimize packet loss on networks with large jitter 3000 ms This value is H263 90000 format parameter CIF used to signal Polycom VVX 1500 receiving capability in SDP This value also controls the TX frame size If set to 1 CIF is used provided the far end supports CIF 1 otherwise QCIF is used video profile H263 QcifMpi 1 default to 32 This value is H263 90000 format parameter QCIF used to signal Polycom VVX 1500 receiving capability in the SDP video profile H263 SqcifMpi 1 default to 32 This value is H263 90000 format parameter SQCIF used to signal Polycom VVX 1500 receiving capability in the SDP video profile H263 1998 jitterBufferMax video profil e H2631998 ji The largest jitter buffer depth to be supported in milliseconds Jitter above this size will defa
206. e Specify whether to filter incoming RTP packets by IP address provisioning sip cfg whether to require symmetric port usage or whether to jam the server destination port and specify the local RTP port range start For more information refer to RTP lt rtp gt on page A 69 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection Network Address Translation The phone can work with certain types of network address translation NAT The phone s signaling and RTP traffic use symmetric ports the source port in transmitted packets is the same as the associated listening port used to receive packets and the external IP address and ports used by the NAT on the phone s behalf can be configured on a per phone basis Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify the external NAT IP address and the ports to be used for provisioning sip cfg signaling and RTP traffic server e For more informa
207. e directory to use for log files if required A URL can also be specified This is blank by default CONTACTS_DIRECTORY An alternative directory to use for user directory files if required A URL can also be specified This is blank by default Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Warning e OVERRIDES_DIRECTORY An alternative directory to use for configuration overrides files if required A URL can also be specified This is blank by default e LICENSE_DIRECTORY An alternative directory to use for license files if required A URL can also be specified This is blank by default The order of the configuration files listed in CONFIG_FILES is significant e The files are processed in the order listed left to right e The same parameters may be included in more than one file e The parameter found first in the list of files will be the one that is effective This provides a convenient means of overriding the behavior of one or more phones without changing the baseline configuration files for an entire system For more information refer to the Configuration File Management on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones white paper at http www polycom com global documents support technical products voice white_ paper_configuration_file_management_on_soundpoint_ip_phones pdf If you have a requirement for different application loads on different phones on the same p
208. e 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 4s 0 Note For te SoundPoint iF 630 plattorm 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 IP_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 3150 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 A 109 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Delay lt delay gt These settings control the phone s behavior when a request for restart or reconfiguration is received call Permitted Attribute Values Default Interpretation request delay type Null audio or call Defines the strategy to adopt before a request gets executed If set to audio a request can be executed as so
209. e for the SoundPoint IP SoundStation IP VVX Phones Attribute dialplan apply ToUserDial Permitted Values Oori Default 1 Interpretation 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 dialplan apply ToUserSend dialplan impossibleMatchHandling Oori 0 10or2 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 Affects digits entered while in dial mode For example the digits are affected after a user has picked up the handset headset or pressed the dial key and not when hot dialing contact dialing or call list dialing 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 Oori If set to 1 strip trailing digit from digits sent out dialplan applyToTelUriDial Oor1 A flag to determine if the dial plan applies to uses of the tel URI If set to 1 or Null the dial plan applies If set to 0 the dial plan does not apply dialplan applyToRemoteDialing Oor1 A f
210. e investment No re wiring with existing CAT 5 cabling Simplifies installation 1000baseT is supported by the SoundPoint IP 560 and 670 and Polycom VVX 1500 only Power over Ethernet PoE port or Power pack option Built in IEEE 802 3af PoE port on the SoundPoint IP 320 321 330 331 450 550 560 650 and 670 the SoundStation IP 6000 and 7000 and Polycom VVX 1500 auto sensing Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Unused pairs on Ethernet port are used to deliver power to the phone via a wall adapter allowing fewer wires to desktop for the SoundStation IP 6000 and 7000 conference phones e Multiple language support on most phones Set on screen language to your preference Select from Chinese Simplified Danish Dutch English Canada United Kingdom and United States French German Italian Japanese Korean Norwegian Polish Portuguese Brazilian Russian Slovenian Spanish International and Swedish Chinese Simplified Japanese and Korean are not supported on the SoundPoint IP 32x 33x phones e Microbrowser Supports a subset of XHTML constructs otherwise runs like any other Web browser e XML status control API Ability to poll phones for call status and device information Ability to receive telephony notification events Overview This chapter provides an overview of the Session Initiation Protocol SIP application and how the phones fit i
211. e on or off sip cfg For more information refer to Feature lt feature gt on page A 110 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Digital Picture Frame Central provisioning server Note Note This feature is only supported on the Polycom VVX 1500 A slide show of multiple personal images stored on a USB flash drive can be displayed on the Polycom VVX 1500 phone during the idle mode The supported formats include JPEG BMP and PNG The maximum image size is 9999x9999 A maximum of 1000 images can be displayed and these must be stored in a directory of the USB flash drive that you create Although 9999x9999 images and progressive multiscan JPEG images are supported the maximum image size that can be downloaded is restricted by the available memory in the phone Configuration changes can be performed centrally at the provisioning server Configuration file Turn this feature on or off and configure how it appears sip cfg For more information refer to Feature lt feature gt on page A 110 For more information refer to User Preferences lt up gt on page A 149 Configuration file Configure how the feature appears phonet cfg For more information refer to User Preferences lt up gt on page A 149 Enhanced Feature Keys Note The Enhanced Feature Key feature from SIP 3 0 is compatible with Enhanced Feature Key feature from SIP 3 2 Howeve
212. e primary SoundStation IP 7000 phone You can reboot the primary without rebooting the secondary However the primary and secondary should be rebooted together for the primary secondary relationship to be recognized If you power up both SoundStation IP 7000 phones the primary will power up first Currently provisioning over C Link is supported for the following configurations of SoundStation IP 7000 conference phones e Two SoundStation IP 7000 conference phone daisy chained together e Two SoundStation IP 7000 conference phone daisy chained together with one external microphone specifically designed for the SoundStation IP 7000 conference phone The provisioning server or proxy for the secondary is determined by the following criteria e The primary phone must be powered up using Multi Interface Module PoE will not provide enough power for both phones e Ifthe secondary is configured for DHCP use the primary s provisioning server if the primary is configured for DHCP Setting up Your System e Ifthe secondary is not configured for DHCP use the secondary s static provisioning server if it exists e Ifthe secondary s static provisioning server does not exists use the primary s provisioning server ignoring the source For more information on daisy chaining and setting up the SoundStation IP 7000 conference phone refer to the Setup Guide for the Polycom SoundStation IP 7000 Phone which is available at http www
213. e provisioning server Central provisioning server Configuration File Specify the ring tone heard on an incoming call when another call is phonet cfg active e For more information refer to Call Waiting lt callWaiting gt on page A 136 Disable call waiting e For more information refer to Registration lt reg gt on page A 128 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family For related configuration changes refer to Customizable Audio Sound Effects on page 4 6 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 The identity displayed is based on the number of the placed call and information obtained from the network signaling Note The phone does not match the number of the placed call to any entries in the Local Contact Directory or Corporate Directory There are no related configuration changes Calling Party Identification The phone displays the caller identity derived from the network signalling when an incoming call is presented if the information is provided by the call server For calls from parties for which a directory entry exists the local name assigned to the Contact Directory entry may optionally be substituted Note The phone does not match the received number to any entries in the Corporate Directory Configuration chan
214. e 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 148 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 they 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 For more information refer to Microbrowser lt mb gt on page A 113 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
215. e rxEq hs VVX_1500 postFilter enable 0 voice rxEq hd IP_330 preFilter enable 0 voice rxEq hd IP_430 preFilter enable 0 voice rxEq hd IP_450 preFilter enable 0 voice rxEq hd IP_650 preFilter enable 1 voice rxEq hd VVX_1500 preFilter enable 0 A 54 Configuration Files Attribute Default voice rxEq hd IP_330 postFilter enable voice rxEq hd IP_430 postFilter enable voice rxEq hd IP_450 postFilter enable voice rxEq hd IP_650 postFilter enable oloj ojl oj o voice rxEq hd VVX_1500 postFilter enable voice rxEq hf IP_330 preFilter enable 1 voice rxEq hf IP_430 preFilter enable 1 voice rxEq hf IP_450 preFilter enable 1 voice rxEq hf IP_650 preFilter enable 1 voice rxEq hf IP_6000 preFilter enable oO oO voice rxEq hf IP_7000 preFilter enable e voice rxEq hf VVX_1500 preFilter enable voice rxEq hf IP_330 postFilter enable voice rxEq hf IP_430 postFilter enable voice rxEq hf IP_450 postFilter enable voice rxEq hf IP_650 postFilter enable voice rxEq hf IP_6000 postFilter enable voice rxEq hf IP_7000 postFilter enable o o o o o o o voice rxEq hf VVX_1500 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 p
216. e 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 11 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 registration to be used for the Microsoft Live Communications Server 2005 Typically this would be 2 Set the reg x address to the LCS address For example reg 2 address 7778 Set the reg x server y address to the LCS server name Optional Set the reg 2 server y transport attribute to TCPpreferred or TLS Your selection depends on the LCS configuration Set reg x auth userId to the phone s LCS username For example reg 2 auth userId jbloggs Set reg x auth password to the LCS password For example reg 2 auth password Password2 Locate the roaming_buddies attribute Set the roaming_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 148 Locate th
217. e 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 Hosted VoIP Service Provider g aa Ball Server 18 all Server 1A DNS Server 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 e
218. e volatile storage for phone resident copy of the directory to allow for larger size dir local nonVolatile maxSize 2meg 1 to 20 20 Applies to platforms with 2 Mbytes of flash memory Maximum size in Kbytes of non volatile storage that the directory will be permitted to consume dir local volatile 4meg Oor1 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default dir local volatile maxSize 1 to 200 200 dir local volatile 8meg Oor1 0 Interpretation 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 Attribute applies only to platforms with 8 Mbytes or more of flash memory If set to 1 use volatile sto
219. ecify 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 83 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Local Local Phone User Enable or disable persistent viewing through the Settings menu Interface Settings gt Basic gt Preferences gt Corporate Directory gt View Persistency Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection This section contains the following information e Corporate Directory LDAP Attributes e Browsing the Corporate Directory 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 Note The maximum of eight attributes can be configured in sip cfg The configuration order dictates how the attributes are displayed and sorted The first attribute is the primary sort index and the seco
220. ed 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 42 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 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 123 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 s
221. eement 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
222. ef IP_6000 Siren14 48kbps voice codecPref iLBC IP_6000 13 33kbps voice codecPref iLBC IP_6000 15 2kbps Permitted Values Null 1 13 Default 5 6 3 Null Null Null Null Null Null Null Null Interpretation Specifies the codec preferences for the SoundStation IP 6000 platform Interpretation as above Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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_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 voice codecPref iLBC IP_7000 13_33kbps voice codecPref iLBC IP_7000 15_2kbps Permitted Values Null 1 16 Default Null
223. efer to DHCP or Manual TCP IP Setup on page 3 2 device dhcp 0to3 For descriptions refer to DHCP Menu on page 3 8 bootSrvUseOpt A 152 Configuration Files Name Possible Values Description device prov serverName any string For descriptions refer to Server Menu on page 3 10 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 prov 1 to 8 maxRedunServers 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 48 device
224. en 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 21 A 139 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation dialplan x removeEndOfDial Oori 1 When present and if dialplan x digitmap s not Null this attribute overrides the global dial plan defined in the sip cfg configuration file For interpretation refer to Dial Plan lt dialplan gt on page A 21 dialplan x applyToTelUriDial Oor1 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 21 This attribute also includes e Digit Map lt digitmap gt e Routing lt routing gt A 140 Digit Map lt digitmap gt Configuration Files For more information on digit map syntax refer to Digit Map lt digitmap gt on page A 23 This configuration attribute is defined as follows 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 s
225. endorse or promote products derived from this software without prior written permission For written permission please contact openssl core openssl org 5 Products derived from this software may not be called OpenSSL nor may OpenSSL appear in their names without prior written permission of the OpenSSL Project 6 Redistributions of any form whatsoever must retain the following acknowledgment This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit http www openssl org THIS SOFTWARE IS PROVIDED BY THE OpenSSL PROJECT AS IS AND ANY EXPRESSED OR IMPLIED WARRANTIES INCLUDING BUT NOT LIMITED TO THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED IN NO EVENT SHALL THE OpenSSL PROJECT OR ITS CONTRIBUTORS BE LIABLE FOR ANY DIRECT INDIRECT INCIDENTAL SPECIAL EXEMPLARY OR CONSEQUENTIAL DAMAGES INCLUDING BUT NOT LIMITED TO PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES LOSS OF USE DATA OR PROFITS OR BUSINESS INTERRUPTION HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY WHETHER IN CONTRACT STRICT LIABILITY OR TORT INCLUDING NEGLIGENCE OR OTHERWISE ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE This product includes cryptographic software written by Eric Young eay cryptsoft com This product includes software written by Tim Hudson tjh cryptsoft com Original SSLeay License Copyright C
226. ensions 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 VVX RFC and Internet Draft Support The following RFC s and Internet drafts are supported RFC 1321 The MD5 Message Digest Algorithm RFC 2327 SDP Session Description Protocol RFC 2387 The MIME Multipart Related Content type RFC 2976 The SIP INFO Method RFC 3261 SIP Session Initiation Protocol replacement for RFC 2543 RFC 3262 Reliability of Provisional Responses in the Session Initiation Protocol SIP RFC 3263 Session Initiation Protocol SIP Locating SIP Servers RFC 3264 An Offer Answer Model with the Session Description Protocol SDP RFC 3265 Session Initiation Protocol SIP Specific Event Notification RFC 3311 The Session Initiation Protocol SIP UPDATE Method RFC 3325 SIP Asserted Identity RFC 3515 The Session Initiation Protocol SIP Ref
227. ent logging level 4 0522 184554 s0 5 O1 utilCertificateInit failed 0522184554 hw QO1 Initial log entry Current logging level 4 0522164554 ares 01 Initial log entry Current logging level 4 0522184554 dns 01 Initial log entry Current logging level 3 0522184554 cfg 01 Initial log entry Current logging level 3 ae al AM po Fis P rA t Osud 129 6 Oey dw jafda Inverell log Aut cent logging level g Seed eee er i 0522114602 50 O1 System Info Reports ry 052218 Piik o IP parameters GES GT w penama ND rev 2 running at 150MHz with memory at 12 0522114602 s0 O1 CPU is TNETV1055 C55x 0522114602 s0 O01 Board is identified as PolycomSoundPointIP SPIP 450 0522114602 so0 101 DRAM LO Ox94000000 DRAM SIZE 32 MB 0522114602 so0 O01 Clocks are VBEUSP 125882 VBUS SNHz USB 25MHz LCD 20HHz 0522114602 key 01 Initial log entry Current logging level 4 0522114602 ht 01 Initial log entry Current logging level 4 0522114602 httpd 01 Initial log entry Current logging level 4 0522114602 ssps 01 Application comp 1 Label PolyDSP Titan Memi FSS G 729 Versi E S ee amen ae a ae ome pl ett petite pL n x re S a a sul tabi feta Si Pie dated wocw etp cig af PE MB Se See ee i i 0522185324 cfgq ee iisi of configuration files suceeded 0522165324 cfg 3 01 Prm Phone successfully provisioned a 0522185324 cfg 0522185324 cfg 0522185324 ctg 0522185324 cfg
228. er 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 3959 The Early Session Disposition Type for the Session Initiation Protocol SIP RFC 3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol SIP RFC 3968 The Internet Assigned Number Authority IANA Header Field Parameter Registry for the Session Initiation Protocol SIP Request Support Session Initiation Protocol SIP e RFC 3969 The Internet Assigned Number Authority IANA Uniform Resource Identifier URI Parameter Registry for the Session Initiation Protocol SIP e RFC 4028 Session Timers in the Session Initiation Protocol SIP e RFC 4235 An INVITE Initiated Dialog Event Package for the Session Initiation Protocol SIP e draft levy sip diversion 08 txt Diversion Indicat
229. er fails Shared Call Appearances Calls and lines on multiple phones can be logically 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 a wide range of industry standard audio codecs 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 Overview DTMF Tone Generation Generates dual tone multi frequenc
230. erience 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 e 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 e IP Type of Service e IEEE 802 1p Q e Voice Quality Monitoring 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 4 78 Configuring Your System 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 adaptive and configurable for different network environments When packets are lost a concealment algorithm minimizes the resulting negative audio consequences Configuration changes can be performed centrally at the provisioning server or locally Central provisioning server Local Configuration file Set the jitter buffer tuning parameters including minimum and maximum size and shrink aggression For more information refer to Codec
231. ern matching the digit map is found the call setup process will complete automatically The configuration syntax is based on recommendations in 2 1 5 of RFC 3435 The phone behavior when the user dials digits that do not match the digit map is configurable It is possible to strip a trailing from the digits sent or to replace certain matched digits with the introduction of R to the digit map Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family For more information digit maps refer to Technical Bulletin 11572 Changes to Local Digit Maps on SoundPoint IP SoundStation IP VVX Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Note Digit maps do not apply to on hook dialing The parameter settings described in Dial Plan lt dialplan gt on page A 21 are ignored during on hook dialing Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify impossible match behavior trailing behavior digit map provisioning sip cfg matching strings and time out value server For more information refer to Dial Plan lt dialplan gt on page A 21 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
232. ers 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 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 patt
233. erwise calls will not be permitted without a valid registration Configuration Files Permitted Attribute Values Default Interpretation call offeringTimeOut positive 60 Time in seconds to allow an incoming call to ring integer 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 138 call ringBackTimeOut positive 60 Time in seconds to allow an outgoing call to integer remain in the ringback state before dropping the call O infinite call dialtoneTimeOut Null positive 60 Time in seconds to allow the dial tone to be integer 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 69 The string sent to the server when the user maximum selects the last call return action length 32 call callsPerLineKey 1 to 24 OR 34 24 8 For the SoundPoint IP 650 and 670 the 1108 OR 4 permitted range is 1 to 34 and the default is 34 For the SoundPoint IP 550 and 560 the permitted range is 1 to 24 and the default is 24 For the SoundPoint IP 32x 33x and 430 the permitted range is 1 to 8 and the default is 4 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
234. ery 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 22 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 can be modified on the Server menu Name Possible Values Description Server Type O FTP 1 TFTP 2 HTTP 3 HTTPS 4 FTPS 5 Invalid The protocol that the phone will use to obtain configuration and phone application files from the provisioning server Refer to Supported Provisioning Protocols on page 3 4 Note Active FTP is not supported for bootROM version 3 0 or later Passi
235. es at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html For more information refer to Technical Bulletin 46792 Best Practices When Using HTTP and HTTPS Provisioning on SoundPoint IP SoundStation IP and Polycom VVX Phones at http www polycom com usa en support voice soundpoint_ip VoIP_T echnical_Bulletins_pub html As of SIP 3 2 Mutual Transport Layer Security TLS authentication is available For more information refer to Mutual TLS Authentication on page 4 93 If you want to use digest authentication against the Microsoft Internet Information Services server e Use Microsoft Internet Information Server 6 0 or later e Digest authentication needs the user name and password to be saved in reversible encryption e The user account on the server must have administrative privileges e The wildcard must be set as MIME type otherwise the phone will not download cfg ld and other required files This is due to the fact that the Microsoft Internet Information Server cannot recognize these extensions and will return a File not found error To configure wildcard for MIME type refer to http support microsoft com kb 326965 For more information refer to http www microsoft com technet prodtechnol WindowsServer2003 Library IIS 809 552a3 3473 48a7 9683 c6df0cdfda21 mspx mfr true Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Mod
236. es Accept Encoding No Accept Language Yes Access Network Info No Alert Info Yes Allow Yes Allow Events Yes Authentication Info No Authorization Yes Call ID Yes Call Info Yes Contact Yes Content Disposition No Content Encoding No Content Language No Content Length Yes Content Type Yes CSeq Yes Session Initiation Protocol SIP Header Supported Notes Date No 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 Yes P Preferred Identity Yes Priority No Privacy No Proxy Authenticate Yes Proxy Authorization Yes Proxy Require Yes RAck Yes Record Route Yes Refer To Yes Referred By Yes Referred To Yes Remote Party ID Yes Replaces Yes Reply To No Requested By No Require Yes Response Key No Administrator s Guide SoundPoint IP SoundStation IP VVX Header Supported Notes Retry After Yes Route Yes RSeq Yes Server Yes Session Expires Yes Subject Yes Subscription State Yes Supported Yes Timestamp Yes To Yes Unsupported Yes 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
237. es refer to To add new languages to those included with the distribution on page A 27 Local Local Phone User The user can select the preferred language under the Settings menu Interface The languages appears in the list in the language itself For example German appears in the list as Deutsch and Swedish appears as Svenska For administrator convenience the ISO representation of each language is also included in the language selection menu Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the provisioning 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 provisioning 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 86 Note Downloadable fonts are not supported on the SoundStation IP 6000 and 7000 and the Polycom VVX 1500 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 4 30 Configuring Your System Configuration changes can be performed cen
238. es 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 bg medRes gray selection 2 1 Specify which type of background w and index x for that type is selected on reboot bg medRes gray pr x adj Specify the brightness adjustment to the graphic bg medRes gray pat solid x name any string White Solid pattern name For x 1 White x 2 Light Gray x 3 4 Null bg medRes gray pat solid x red 0 to 255 bg medRes gray pat solid x green 0 to 255 bg medRes gray pat solid x blue 0 to 255 The screen background layouts For x 1
239. et 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 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 I
240. eter is disabled divert x sharedDisabled Oori 1 If set to 1 all diversion features on that line will be disabled if the line is configured as shared A 136 This attribute also includes Forward All lt fwd gt Busy lt busy gt No Answer lt noanswer gt Do Not Disturb lt dnd gt Forward All lt fwd gt This configuration attribute is defined as follows Configuration Files Permitted Attribute Values Default Interpretation divert fwd x enabled Oor1 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 Busy lt busy gt Calls can be automatically diverted when the phone is busy Attribute Permitted Values Default Interpretation divert busy x enabled Oor1 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 Null Forward to contact for calls containing digits the user part forwarded due to busy status if of a SIP URL or a string that Null divert x contact will be constitutes a valid SIP URL used 6416 or 6416 polycom com A 137 Administrator s Guide for the SoundPoint
241. eters available in the configuration files Either the Web Server or the provisioning 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection For more information refer to Local lt local gt on page A 7 Server lt server gt on page A 8 and Registration lt reg gt on page A 128 SIP B Automatic Call Distribution Note For more information on SIP B and supported features on SoundPoint IP SoundStation IP and Polycom VVX phones contact Polycom Product Management The phone allows Automatic Call Distribution ACD login and logout This feature depends on support from a SIP server Configuration changes can be performed centrally at the provisioning server Central Configuration file Turn this feature on or off provisioning sip cfg e For more information refer to Feature lt feature gt on page A 110 server Configuration file Enable this feature per registration phone1 cfg For more information refer to Registration lt reg gt on page A 128 The phone also supports ACD agent availability This fea
242. ettings 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 95 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 A 114 Configuration Files Attribute Permitted Values Default Interpretation mb idleDisplay refresh 0 or an integer gt 5 0 The period in seconds between refreshes of the idle display Microbrowser s content If set to 0 the idle display Microbrowser is not refreshed The minimum refresh period is 5 seconds values from 1 to 4 are ignored and 5 is used Note If an HTTP Refresh header is detected it will be respected even if this parameter is set to 0 The refresh parameter will be respected only in the event that a refresh fails Once a refresh is successful the value in the HTTP refresh header if available will be used Main Browser lt main gt
243. eturn all logging levels to the default value of 4 There are other logging parameters that you may wish to modify Changing these parameters does not have the same impact as changing the logging levels but you should still understand how your changes will affect the phone and the network e log render level Sets the lowest level that can be logged default 1 e log render file size Maximum size before log file is uploaded default 16 kb e log render file upload period Frequency of log uploads default is 172800 seconds 48 hours e log render file upload append Controls if log files on the provisioning 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 provisioning server default 512 kb e log render file upload append 1imitMode 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 VVX Phones The following figure shows an exam
244. ey You can control of the following features of the Polycom VVX 1500 phone s camera e Flicker avoidance e Frame rate e Brightness level e Saturation level e Contrast level e Sharpness level Central provisioning server Configuring Your System Configuration changes can be performed centrally at the provisioning server or locally Configuration file sip cfg Turn video transmission off at the near end when calls start and transmit still image if video not available For more information refer to Video Settings lt video gt on page A 61 Specify camera parameters For more information refer to Camera Controls lt camera gt on page A 66 Determine how the local camera is displayed For more information refer to Local Camera View lt localGameraView gt on page A 67 Local Local Phone User Interface The user can set the individual video settings from the menu through Settings gt Basic gt Video gt Video Call Settings Video Screen Mode and Local Camera View The user can set the individual camera settings from the menu through Settings gt Basic gt Video gt Camera Settings Video Codecs The following table summarizes the Polycom VVX 1500 phone s video codec support Effective Frame video Algorithm MIME Type Bit Rate Rate Frame Size bandwidth H 263 H263 90000 64 kbps to 5 fps to Tx Frame size C
245. f offHours hours If set to 0 this feature is disabled If set to Null the default value is 2 This value was chosen for good performance in a typically office environment and is biased for difficult detection during off hours Per Phone Configuration we POLYCOM 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 Provisioning Server on page 3 17 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 SoundStation IP and Polycom VVX 1500 Phones white paper at http Awww polycom com global documents support technical products voice white_ paper_configuration_file_management_on_soundpoint_ip_phones pdf The parameters include e Registration lt reg gt e Calls lt call gt e Diversion lt divert gt A 127 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Registration lt reg gt e Dial Plan lt dialplan gt e Messaging lt msg gt e Network Address Translation lt nat gt e Attendant lt attendant gt e Roaming Buddies lt roaming_buddies gt e Roaming Privacy lt roaming_privacy gt e User Preferences
246. f enabled Navigate to http lt phonelPAddress gt netConf htm qo Local Phone User Specify whether CDP is to be used or manually set the VLAN ID or Interface configure DHCP VLAN Discovery Phase 1 bootRom Navigate to SETUP menu during auto boot countdown Phase 2 Application Navigate to Menu gt Settings gt Advanced gt Admin Settings gt Network Configuration For more information refer to Setting Up the Network on page 3 2 Voice Quality Monitoring Note Note This feature requires a license key for activation except for the Polycom VVX 1500 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 sent as SIP PUBLISH messages to a central voice quality report collector The collection of these metrics is supported on the SoundPoint IP 32x 33x 430 450 550 560 650 and 670 phones and the Polycom VVX 1500 phone Voice Quality Monitoring is not supported on the SoundStation IP 6000 and 7000 conference phones at this time Only voice Quality Monitoring of the audio portion is supported on the Polycom VVX 1500 at this time Configuring Your System The RTCP XR packets are compliant with RFC 3611 RTP Control
247. f set to 0 the Forward soft key is not displayed If set to 1 or Null the Forward soft key is displayed A 124 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 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 so
248. f so that the flashing display is not a distraction Debugging of single phone may be possible through an examination of the phone s status menu Press Menu select Status and then press the Select soft key Under the Platform selection you can get details on the phone s serial number or MAC address the current IP address the bootROM version the application version the name of the configuration files in use and the address of the provisioning 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 has 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 Log Files Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones In addition to the hardware tests the Diagnostic
249. feature and screen saver are supported on the Polycom VVX 1500 only This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation up analogHeadsetOption 0 1 0r 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 Oor1 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 up pictureFrame folder string Null The path name for images The maximum length is 40 characters If set to Null images stored in the root folder on the USB flash drive are displayed For example if the images are stored in the images phone folder on the USB flash drive set up pictureFrame folder to images phone
250. fects 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 Distinctive Incoming Call Treatment The phone can automatically apply distinctive treatment to calls containing specific attributes Distinctive Ringing The user can select the ring type for each line and the ring type for specific callers can be assigned in the contact directory Do Not Disturb A do not disturb feature is available to temporarily stop all incoming call alerting Graphic Display Backgrounds A picture or design displayed on the background of the graphic display Handset Headset and Speakerphone SoundPoint IP phones come standard with a handset and a dedicated headset connection headset not supplied All SoundPoint IP SoundStation IP and Polycom VVX phones have full duplex speakerphones Idle Display Animation All phones can display a customized animation on the idle display in addition to the time and date Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Last Call Return The phone allows call server based last call return Local Cen
251. fg can be done manually but a scripting tool is useful to change per phone configuration files The configuration files listed in CONFIG_FILES attribute of the master configuration file must be updated when the software is updated Any new configuration files must be added to the CONFIG_FILES attribute in the appropriate order Mandatory changes must be made or the software may not behave as expected For more information refer to the Configuration File Management on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones white paper at http Awww polycom com global documents support technical products voice white_ paper_configuration_file_management_on_soundpoint_ip_phones pdf 3 Save the new configuration files and images such as sip ld on the provisioning server Setting up Your System 4 Reboot the phones using automatic methods such as polling or check sync Using the reboot multiple key combination should be used as a backup option only For more information refer to Multiple Key Combinations on page C 10 Since the APPLICATION APP_FILE_PATH attribute of the lt MACaddress 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 11670 001 sip Id to boot the SoundPoint IP 670 image The phones can be rebooted remotely through the SIP signaling protocol Refer to Special Events l
252. fication 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 Miscellaneous Administrative Tasks GlobalSign Partners CA GlobalSign Primary Class 1 CA GlobalSign Primary Class 2 CA GlobalSign Primary Class 3 CA GlobalSign Root CA National Retail Federation by DST TC TrustCenter Germany Class 1 CA TC TrustCenter Germany Class 2 CA TC TrustCenter Germany Class 3 CA TC TrustCenter Germany Class 4 CA Thawte Personal Basic CA 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 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 Verisi
253. fk efklist 2 gt lt efkprompt efk efkprompt efk efkprompt efk efkprompt efk efkprompt efk efkprompt gt use use use USE USe use PPP RPP mname callpark status 1 label Call Park idle 1 active 1 alerting 1 dialtone 1 proceeding 1 setup 1 type invite action string 68 SPIN10 status 1 label Enter Number userfeedback visible type numeric digitmatching stylel 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family lt directory gt Note 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 81 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 41941 H Call Park To Ross Dutkiewicz 1442 0 16 i Park Rtr i Call Pickup Mon Dec 17 951 AM Hold
254. ft 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 125 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones LCD Power Saving lt powerSaving gt Note basicCallManagement redundant Permitted Attribute Values Default Interpretation softkey feature buddies 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 mapped for Hold Transfer and Conference functions all must be mapped all of these soft keys are not displayed If set to 1 or Null all of these soft keys are displayed This attribute is supported for use on the Polycom VVX 1500 only This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation powerSaving enabled Oor1 1 If set to 1 or Null the LCD power saving feature is enabled If set to 0 the LCD power saving feature is disabled startHour xxx powerSaving officeHours 0 to 23 The starting hour for the day s office hours where xxx is one of monday tuesday wednesday thursday friday saturday and sunday If set to Null t
255. g upload 5 7 master configuration file model number version A 5 part number substitution A 4 master configuration files details A 2 overview 2 5 message waiting indication 4 7 message waiting indicator lt mwi gt A 143 messaging lt msg gt A 143 Microbrowser 4 31 4 69 microphone mute 4 14 Microsoft Live Communications Server 2005 Integration 4 65 migration dependencies C 10 miscellaneous patterns A 39 missed call configuration lt serverMissedCall gt A 134 missed call notification 4 5 model number substitution A 5 modifying network configuration 3 6 multilingual lt ml gt A 26 multilingual user interface 4 29 multiple call appearances 4 28 multiple line keys per registration 4 27 multiple registrations 4 57 music on hold 4 20 music on hold lt musicOnHold gt A 20 mutual TLS support for 4 93 N Network Address Translation lt nat gt A 144 network configuration modifying 3 6 network monitoring lt netMon gt A 71 new features 2 14 no answer lt noanswer gt A 138 O Option 66 3 8 outbound proxy lt outboundProxy gt A 17 P packet error concealment 4 79 password lt pwd gt A 104 patterns lt pat gt A 36 patterns lt pattern gt A 97 peer networking lt pnet gt application configuration peer networking A 120 per phone configuration attendant A 145 automatic call distribution A 150 automatic off hook call placement A 134 behaviors A 147 busy A 137 calls A 133 dial plan emergency A 142
256. ges 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 be performed centrally at the provisioning server Central XML file lt Ethernet The lt bw gt 0 lt bw gt buddy watching and lt bb gt 0 lt bb gt buddy provisioning address gt directory blocking elements in the lt Ethernet address gt directory xml file server xml dictate the Presence aspects of directory entries e For more information refer to Local Contact Directory on page 4 10 Local Local Phone User The user can edit the directory contents The Watch Buddy and Interface Block Buddy fields control the buddy behavior of contacts Changes will be stored in the phone s flash file system and backed up to the provisioning server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the provisioning server copy of the directory if present will overwrite the local copy Microsoft Live
257. ges can performed centrally at the provisioning server or locally Central Configuration File Specify whether or not to use directory name substitution provisioning sip ctg e For more information refer to User Preferences lt up gt on page server A 29 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection Configuring Your System Missed Call Notification The phone can display the number of calls missed since the user last looked at the Missed Calls list The phone can be configured to use a built in missed call counter or to display information provided by a Session Initiation Protocol SIP server Note On some SoundPoint IP platforms missed calls and received calls appear in one list Configuration changes can performed centrally at the provisioning server Central Configuration file Turn this feature on or off provisioning sip cfg For more information refer to Feature lt feature gt on page A 110 server Configuration file Specify per registration whether all missed call events or only phone1 cfg remote server generated missed call events will be displayed For more inf
258. gn Class 3 Public Primary Certification Authority G3 Administrator s Guide SoundPoint IP SoundStation IP VVX we POLYCOM e Verisign Class 4 Public Primary Certification Authority G2 e Verisign Class 4 Public Primary Certification Authority G3 e Verisign RSA Commercial CA e Verisign RSA Secure Server CA 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 http 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 provisioning server and it can encrypt files before uploading them to the provisioning server There must be an encryption key on the phone to perform these operations Configuration files excluding the master configuration file contact directo
259. gration 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 430 450 550 560 650 and 670 and the MSG key on the Configuring Your System Polycom VVX 1500 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 be performed centrally at the provisioning server or locally Central provisioning server Configuration file For one touch voice mail access enable the one touch voice mail user preference For more information refer to User Preferences lt up gt on page A 29 Configuration file For one touch voice mail access bypass instant messages to phonet cfg remove the step of selecting between instant messages and voice mail after pressing the Messages key on the SoundPoint IP 430 450 550 560 650 and 670 and the MSG key on the Polycom VVX 1500 Instant messages are still accessible from the Main Menu On a per registration basis specify a subscribe contact for solicited NOTIFY applications a callback mode self call back or a
260. ground takes priority over the idle browser Used in conjunction with up prioritizeBackground enable up prioritizeBackgroundMenultem 0 or 1 1 If set to 1 the Prioritize Background menu is enable available to the user The user can then decide whether or not the background takes priority over the idle browser Used in conjunction with up idleBrowser enabled up screenCapture enabled Oor1 0 A flag to determine whether or not the user can get a screen capture of the current screen shown on a phone The flag is cleared when the phone reboots If set to 1 the Screen Capture menu is available to the user Refer to Capturing Phone s Current Screen on page C 29 Tones lt tones gt This attribute describes configuration items for the tone resources available in the phone This attribute includes Chord Sets lt chord gt Dual Tone Multi Frequency lt DTMF gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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
261. he third 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 is second in priority 3 The volpProt SIP alertInfo x value and volpProt SIP alertInfo x class fields can be used to map calls to specific ring types This option requires server support and is first highest in priority Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify the mapping of Alert Info strings to ring types provisioning sip cfg e For more information refer to Alert Information lt alertInfo gt on server page A 18 Configuration file Specify the ring type to be used for each line phonet cfg For more information refer to Registration lt reg gt on page A 128 XML File lt Ethernet This file can be created manually using an XML editor address gt directory For more information refer to Local Contact Directory on page xml 4 10 Local Local Phone User The user can edit the ring types selected for each line under the Interface Settings menu The user can also edit the directory contents Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the provisioning server Changes will permanently override global settings unle
262. he application for changing the same network parameters For more information refer to Modifying the Network Configuration on page 3 6 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 following table shows the manually entered networking parameters that may be overridden by parameters obtained from a DHCP server an alternate DHCP server or configuration file Alternate Configuration File Local Parameter DHCP Option DHCP DHCP application only FLASH gt priority when more than one source exists gt 1 2 3 4 IP address 1 E i subnet mask 1 E IP gateway 3 e 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 8 151 Note This value SIP server address is configurable SNTP server 42 then 4 address SNTP GMT offset 2 DNS server IP 6 S address alternate DNS 6 e server IP address DNS domain 15 Refer to DHCP Warning Link Layer Discovery Protocol LLDP overrides Cisco Menu on page Discovery Protocol CDP C
263. he default value is 8 duration xxx powerSaving officeHours Oto 12 10 or O The duration of the day s office hours where xxx is one of monday tuesday wednesday thursday friday saturday and sunday If set to Null the default value for the week days is 10 hours and the default value for Saturday and Sunday is 0 hours officeHours powerSaving idleTimeout 1 to 600 10 The office hours mode idle timeout in minutes If set to Null the default value is 10 A 126 Configuration Files Permitted Attribute Values Default Interpretation powerSaving idleTimeout 1to10 1 The off hours mode idle timeout in minutes offHours If set to Null the default value is 1 powerSaving idleTimeout 1 to 20 10 The minimum idle timeout after user input userlnputExtension events in minutes If set to Null the default value is 10 powerSaving 0 to 10 7 The sensitivity of the algorithm used to detect userDetectionSensitivity the presence of the phone s user during office office Hours hours If set to 0 this feature is disabled If set to Null the default value is 7 This value was chosen for good performance in a typically office environment and is biased for easy detection during office hours powerSaving 0 to 10 2 The sensitivity of the algorithm used to detect userDetectionSensitivity the presence of the phone s user during of
264. he following e Configurable list of remote parties to a maximum of 47 with configurable line key labels e The introduction of configurable default key press actions e The ability to remove spontaneous call appearances from incoming calls on monitored lines The SIP 3 2 update to the BLF feature is not supported on the SoundPoint IP 430 For more information refer to Quick Tip 37381 Enhanced BLF at http www polycom com support voice soundpoint_ip VoIP_Technical _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 65 Use this feature with TCPpreferred transport refer to Server lt server gt on page A 8 You can also use UDP transport on SoundPoint IP 650 and 670 phones Configuration changes can be performed centrally at the provisioning server Central provisioning 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 information refer to Attendant lt attendant gt on page A 145 Specify the list of monitored resources For more information refer to Resource List lt resourceList gt on page A 146 and Behaviors lt behaviors gt on page A 147 Voice Mail Inte
265. he user presses the QSetup soft key anew menu will immediately appear that allows them to configure the necessary parameters for the phone to access the provisioning server for configuration The QSetup soft key may be disabled using a configuration file setting such that it does not appear after it has been successfully configured The Quick Setup feature is supported on all SoundPoint IP 32x 33x 430 450 550 560 650 and 670 desktop phones SoundStation IP 6000 and 7000 conference phones and Polycom VVX 1500 phones System administrators can enable the Quick Setup feature through the use of a new parameter in sip cfg configuration file or through the phone s menu For details on how to configure SoundPoint IP SoundStation IP and VVX phones for quick setup refer to Technical Bulletin 45460 Using Quick Setup with SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html 4 77 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be 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 Provisioning lt prov gt on page A 108 Setting Up Audio Features Proprietary state of the art digital signal processing DSP technology is used to provide an excellent audio exp
266. he v1 3 on Microsoft Windows XP For more information on using Mutual TLS with Microsoft Internet Information Services IIS 6 0 refer to Technical Bulletin 52609 Mutual Transport Layer Security Provisioning Using Microsoft Internet Information Services 6 0 at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuring SoundPoint IP SoundStation IP VVX Phones Locally 4 94 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 75 The phone s local user interface also permits many application settings to be modified such as SIP server address ring type or regional settings such as time date format and language Configuring Your System 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 men
267. hen startMode is abs hh mm log sched x startDay 1 7 When startMode is abs specifies the day of the week to start command execution 1 Sun 2 Mon 7 Sat Security lt sec gt This attribute s settings affect security aspects of the phone This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation sec tagSerialNo Oor1 Null If set to 1 the phone may advertise its serial number Ethernet address through protocol signaling If set to 0 or Null the phones does advertise its serial number This attribute also includes Encryption lt encryption gt e Password Lengths lt pwd gt lt length gt e SRTP lt srtp gt A 103 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Encryption lt encryption gt This configuration attribute is defined as follows Permitted Values Default Interpretation sec encryption upload dir Oori 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 encryptio
268. her distribution licence including the GNU Public Licence zlib version 1 2 3 July 18th 2005 Copyright C 1995 2005 Jean loup Gailly and Mark Adler This software is provided as is without any express or implied warranty In no event will the authors be held liable for any damages arising from the use of this software Permission is granted to anyone to use this software for any purpose including commercial applications and to alter it and redistribute it freely subject to the following restrictions 1 The origin of this software must not be misrepresented you must not claim that you wrote the original software If you use this software in a product an acknowledgment in the product documentation would be appreciated but is not required 2 Altered source versions must be plainly marked as such and must not be misrepresented as being the original software 3 This notice may not be removed or altered from any source distribution Jean loup Gailly Mark Adler jloup gzip org madler alumni caltech edu 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
269. his 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 for SIP 3 2 is shown below lt xml version 1 0 standalone yes gt lt Default Master SIP Configuration File gt lt For information on configuring Polycom VoIP phones please refer to the gt lt Configuration File Management white paper available from gt Configuration Files lt http www polycom com common documents whitepapers configuration_file _management_on_soundpoint_ip_phones pdf gt lt SRCSfile 000000000000 cfg v Revision 1 21 gt lt APPLICATION APP_FILE_PATH sip 1d CONFIG_FILES phonel cfg sip cfg MISC_FILES LOG_FILE_DIRECTORY OVERRIDES_DIRECTORY CO CO CO CO CO CO CO CO CO TACTS_DIRECTORY LICENSE_DIRECTORY gt lt APPLICATION_SPIP300 APP FILE PATH SPIP300 sip_212 1d FIG_FILES_SPIP300 phonel_212 cfg sip_212 cfg gt lt APPLICATION_SPIP500 APP FILE PATH SPIP500 sip_212 1d FIG_FILES_SPIP500 phonel_212 cfg sip_212 cfg gt lt APPLICATION_SPIP301 APP FILE PATH SPIP301 sip_313 1d FIG_FILES_SPIP301 phonel_313 cfg sip_313 cfg gt lt APPLICATION_SPIP501 APP FILE PATH SPIP501 sip_313 1d FIG_FILES_SPIP501 phonel_313 cfg sip_313 cfg gt lt APPLICATION_SPIP600 APP FILE PATH SPIP600 sip
270. ictures Both BMP and JPEG files are supported You can also 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 For Polycom VVX 1500 phones You can select the pictures or designs displayed on the background The supported formats include JPEG BMP and PNG and the maximum size is 800x480 A default picture is displayed when the phone starts up the first time Users can select which background appears on their individual phones Users can also select a background from an image displayed by the digital picture frame feature refer to Digital Picture Frame on page 4 38 Support for resolutions greater than 800x480 is inconsistent Content may be truncated or nor displayed Progressive multiscan JPEG images are not supported at this time Configuration changes can be performed centrally at the provisioning server or locally Central provisioning server Configuration file Specify which background will be displayed phonet cfg 7 For more information refer to Backgrounds lt bg gt on page A 91 Local Local Phone User On the Polycom VVX 1500 the user can save one of the images as Interface the background by selecting Save as Background on the touch screen 4 18
271. idleSoftkeys Oor1 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 appear Fonts lt font gt 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 Configuration Files font support for Unicode ranges not already embedded The font file must be a Microsoft fnt file format The font file name must follow a specific pattern as described Font filename lt fontName gt _ lt fontHeight InPixels gt _ lt fontRange gt lt fontExtension gt 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 are 256 characters based blocks the lt fontRange gt is Uxx00_UxxFF fint file For more information refer to Multilingual User Interface on page 4 29 lt fontExtension gt describes the file type Either fnt for single 256 characters font If it is necessary to overwrite an existing font use these lt fontName gt _ lt fontHeightInPixels gt
272. 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 volpProt server x expires positive 3600 The phone s requested registration period in integer seconds minimum 10 Note The period negotiated with the server may be different The phone will attempt to re register at the beginning of the overlap period For example if expires 300 and overlap 5 the phone will re register after 295 seconds 300 5 volpProt server x expires overlap positive 60 The number of seconds before the expiration integer time returned by server x at which the phone minimum 5 should try to re register The phone will try to maximum re register at half the expiration time returned 65535 by the server if that value is less than the configured overlap value volpProt server x register Oor1 1 If set to 0 calls can be routed to an outbound proxy without registration Refer to reg x server y register in Registration lt reg gt on page A 128 For more information refer to Technical Bulletin 5844 SIP Server Fallback Enhancements on SoundPoint IP Phones at http www polycom com usa en support voic e soundpoint_ip VoIP_Technical_Bulletins_p ub html Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Def
273. ificates are set to expire on March 9 2044 An X 509 digital certificate is a digitally signed statement The X 509 standard defines what information can go into a certificate All X 509 certificates have the following fields in addition to the signature e Version This identifies which version of the X 509 standard applies to this certificate which affects what information can be specified in it Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Serial Number The entity that created the certificate is responsible for assigning it a serial number to distinguish it from other certificates it issues e Signature Algorithm Identifier This identifies the algorithm used by the Certificate Authority CA to sign the certificate e Issuer Name The X 500 name of the entity that signed the certificate This is normally a CA Using this certificate implies trusting the entity that signed this certificate e Validity Period Each certificate is valid only for a limited amount of time This period is described by a start date and time and an end date and time and can be as short as a few seconds or almost as long as a century Subject Name The name of the entity whose public key the certificate identifies This name uses the X 500 standard so it is intended to be unique across the Internet e Subject Public Key Information This is the public key of the entity being named together with an algorithm ident
274. ifier which specifies which public key cryptographic system this key belongs to and any associated key parameters The following is an example of a Polycom device certificate if opened with the Microsoft Internet Explorer 7 or Firefox 3 5 browser on a computer running Microsoft XP Service Pack 3 Certificate pong General Details Certification Path Show lt All gt Field Value fF Version v3 E Serial number 5f 17 61 f1 00 00 00 01 a6 2f E Signature algorithm sha256RSA FE issuer Polycom Equipment Issuing CA 1 E valid from Monday March 23 2009 2 33 Elvaiid to Saturday March 23 2024 2 4 E Subject 0004F222335F Polycom Inc EfPubickey RSA 2048 Bits _ The device certificate and associated private key are stored on the phone in its non volatile memory as part of the manufacturing process For more information on digital certificates refer to http www ietf org html charters pkix charter html and http www ietf org rfc rfc2459 txt Configuring Your System To determine if there is a digital certificate on a SoundPoint IP SoundStation IP or Polycom VVX phone 1 Press the Menu key and then select Status gt Platform gt Phone 2 Scroll down to the bottom of screen One of three messages will be displayed Device Certificate Installed is displayed if the certificate is available in flash memory all the certificate fields are valid listed above and certificate has not ex
275. ifying the Network Configuration You can access the network configuration menu e 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 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 menu selection from the Advanced Settings menu or using a multiple key combination described in Multiple Key Combinations on page C 10 Mai
276. igital certificates support for 4 91 digital picture frame feature A 111 directed call pick up 4 24 directory lt dir gt A 81 distinctive call waiting 4 8 distinctive incoming call treatment 4 7 distinctive ringing 4 7 Index diversion A 136 DND See also do not disturb DNS cache lt dns gt A 120 DNS SIP server name resolution 4 61 do not disturb 4 8 do not disturb lt dnd gt A 134 A 138 downloadable fonts 4 30 DTMF event RTP payload 4 80 DTMF tone generation 4 80 DTMF See also dual tone multi frequency dual tone multi frequency lt DMTF gt A 32 dynamic noise reduction 4 85 E electronic hookswitch supported 4 9 A 149 emergency lt emergency gt A 25 A 142 emergency routing A 25 A 142 encryption lt encryption gt A 104 enhanced feature keys definition language 4 39 examples 4 46 macro definitions 4 42 useful tips 4 45 enhanced feature keys feature 4 38 A 111 Ethernet IEEE 802 1p Q A 67 Ethernet menu 3 12 F feature lt feature gt A 110 feature licensing 4 21 4 22 4 34 4 37 4 84 A 111 feature synchronized ACD feature 4 60 A 150 features list of 1 6 finder lt finder gt A 112 flash parameter configuration A 151 flash parameter See also device fonts lt font gt A 86 forward all lt fwd gt A 137 G gains lt gain gt A 47 graphic display backgrounds 4 17 A 91 graphic icons lt gi gt A 98 group call pick up 4 24 Index 3 Administrator s Guide for the SoundPoint IP
277. ill fail over to that server if all higher priority servers are down Recommended Practices for Fallback Deployments In situations where server redundancy for fall back purpose is used the following measures should be taken to optimize the effectiveness of the solution 1 Deploy an on site DNS server to avoid long call initiation delays that can result if the DNS server records expire 2 Do not use OutBoundProxy configurations on the phone if the OutBoundProxy could be unreachable when the fallback occurs Sound Point IP phones can only be configured with one OutBoundProxy per registration and all traffic for that registration will be routed through this proxy for all servers attached to that registration If Server 2 is not accessible through the configured proxy call signaling with Server 2 will fail 3 Avoid using too many servers as part of the redundancy configuration as each registration will generate more traffic 4 Educate users as to the features that will not be available when in fallback operating mode 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 chan
278. ill 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 fix the destination transport port to a specified value regardless of the negotiated port This can be useful for communicating through firewalls When this is enabled all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well Incoming packets are sorted by the source IP address and port allowing multiple RTP streams to be multiplexed The RTP port range used by the phone can be specified Since conferencing and multiple RTP streams are supported several ports can be used concurrently Consistent with RFC 1889 the next higher odd port is used to send and receive RTCP Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Configuration fil
279. ing server Configuration file Specify the number of line keys to assign per registration phonet cfg For more information refer to Registration lt reg gt on page A 128 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 provisioning 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 provisioning server Local Phone User Specify the number of line keys to assign per registration using the Interface SIP Configuration menu Either the Web Server or the provisioning 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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
280. ing can be disabled Note A call active on another phone ona shared line counts as a call for every phone sharing that registration reg x bargelnEnabled Oor1 Null Allow remote user of SCA to interrupt call Works in a similar way to resume If set to 1 barge in is enabled for line x If set to 0 or Null barge in is disabled for line x reg x outboundProxy address dotted decimal IP address or host name Null reg x outboundProxy port 1 to 65535 5060 IP address or host name and port of a SIP server to which the phone shall send all requests A 130 Configuration Files Attribute reg x outboundProxy transport Permitted Values DNSnaptr or TCPpreferred or UDPOnly or TLS or TCPOnly Default DNSnap tr Interpretation If set to Null or DNSnaptr If reg x outboundProxy address is a hostname and reg x outboundProxy port is 0 or Null do NAPTR then SRV look ups to try to discover the transport ports and servers as per RFC 3263 If reg x outboundProxy address is an IP address or a port is given then UDP is used If set to TCPpreferred TCP is the preferred transport UDP is used if TCP fails If set to UDPOnly Only UDP will be used If set to TLS If TLS fails transport fails Leave port field empty will default to 5061 or set to 5061 If set to TCPOnly Only TCP will be used reg x proxyRequire string Null The string that need
281. ing server copy of lt Ethernet address gt directory xml if this is configured When the phone boots the provisioning 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 to indicate that they are not accurate until a successful SNTP response is received The time and date display can use one of several different formats and can be turned off The SoundPoint IP 32x 33x and IP 4xx phones have a limited selection of date formats due to a smaller display size Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Central provisioning server Configuration changes can be performed centrally at the provisioning server or locally Configuration file sip cfg Turn time and date display on or off For more information refer to User Preferences lt up gt on page A 29 Set the time and date display formats For more information refer to Date and Time lt datetime gt on page A 29 Set the basic SNTP settings and daylight savings parameters e For more information refer to Time Synchronization lt sntp gt on page A 71 Lo
282. ins digits only are parsed as a macro name V and A sequence of characters prefixed with is the action string 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 49 A sequence of macro names in the same macro is supported for example tm1 m2 A sequence of speed dial references is supported for example 1 2 A sequence of macro names and speed dial references is supported for example m1 2 m2 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 A sequence of macros is supported but cannot be mixed with other action types 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 indexe
283. int IP SoundStation IP VVX 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 e Polycom VVX 1500 Business Media Phone For a list of key features available on the SoundPoint IP SoundStation IP VVX phones running the latest software refer to Key Features of Your SoundPoint IP SoundStation IP VVX 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 voicedocumentation Additional options are also available For more information contact your Polycom distributor Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note Documentation for the SoundPoint IP 300 301 500 501 600 and 601 desktop phones and the SoundStation IP 4000 conference phone is available at http Awww polycom com voicedocumentation The currently supported desktop phones are e SoundPoint IP 320 321 330 331 e SoundPoint IP 430 Introducing the SoundPoint IP SoundStation IP VVX Family e SoundPoint IP 450 e SoundPoint IP 550 560 Administrator s Guide for the SoundPoint IP Sound
284. int IP SoundStation IP and Polycom VVX phones Phone Supported Audio Codecs SoundPoint IP 430 G 711p law G 711a law G 729AB SoundPoint IP 320 321 330 331 G 711p law G 711a law G 729AB iLBC SoundPoint IP 450 550 560 650 and 670 G 711p law G 711a law G 722 G 729AB iLBC SoundStation IP 6000 G 711p law G 711a law G 722 G 722 1 G722 1C G 729AB Siren14 iLBC SoundStation IP 7000 G 711p law G 711a law G 722 G 722 1 G 722 1C G 729AB Lin16 Siren14 Siren22 iLBC Polycom VVX 1500 G 711p law G 711a law G 722 G 722 1 G722 1C G 729AB Lin16 Siren14 The following table summarizes the supported audio codecs 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Effective Sample audio Algorithm MIME Type Ref Bit Rate Rate Frame Size bandwidth Lin16 L16 8
285. int 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 11 Save the modified sip cfg configuration file 2 Modify the phonel cfg configuration file as follows a Open phonel cfg in an XML editor Locate the registration parameter Set the reg 1 address to the LCS address For example reg 1 address 7778 Set the reg 1 server y address to the LCS server name Optional Set the reg 1 server y transport attribute to TCPpreferred or TLS Your selection depends on the LCS configuration Set reg 1 auth userId to the phone s LCS username For example reg 1 auth userId jbloggs Set reg 1 auth password to the LCS password For example reg 1 auth password Password2 Locate the roaming_buddies attribute Set the roaming_buddies reg element to 1 Refer to Roaming Buddies lt roaming_buddies gt on page A 148 Locate the roaming_privacy attribute Set the roaming_privacy reg element to 1 Refer to Roaming Privacy lt roaming_privacy gt on page A 148 Save the modified phonel cfg configuration file Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family To set up a dual registration with Microsoft Live Communications Server 2005 as the presenc
286. ion lt reg gt on page A 128 Set all call diversion settings including a global forward to contact and individual settings for call forward all call forward busy call forward no answer and call forward do not disturb For more information refer to Diversion lt divert gt on page A 136 Local Web Server Set all call diversion settings 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 provisioning 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 provisioning server Local Phone User The user can set the call forward all setting from the idle display Interface 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 provisioning 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 provisioning server Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Directed Call Pick Up Calls to another phone can be picked up by dialing the extension of the other phone This feature depends on
287. ion 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 supported Method Supported Notes REGISTER Yes INVITE Yes ACK Yes CANCEL Yes BYE Yes OPTIONS Yes SUBSCRIBE Yes NOTIFY Yes REFER Yes PRACK Yes Administrator s Guide SoundPoint IP SoundStation IP VVX Header Support Note 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 The following SIP request headers are supported In the following table a Yes in the Supported column means the header is sent and properly parsed Header Supported Notes Accept Y
288. ion 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 for the SoundPoint IP SoundStation IP VVX Family 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 cache 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 exa
289. ique 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 Override Files This file contains all changes that are made by a user through the their phone for example time date formats ring types and backlight intensity The file allows the phone to keep user preferences through reboots and upgrades There is an option to clear the override file available to the system administrator press the Menu key and then select Settings gt Advanced gt Admin Settings gt Reset to Default gt Reset Local Config You will be prompted to enter the administrative password Central Provisioning The phones can be centrally provisioned from a provisioning server througha system of global and per phone configuration files The provisioning server also facilitates automated application upgrades logging and a measure of fault tolerance Multiple redundant provisioning 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 Note Overview 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 sy
290. itch doesn t support network policy TLV Defined if phone is operational stage and Networkpolicy TLV is received from the switch Tagged Untagged Vianld L2 priority and DSCP Note Video Conferencing TLV is sent only from Video capable phones currently Polycom VVX 1500 only 14 LLDP MED location identificatio ns 127 min len gt 0 max len lt 511 0x0012bb ELIN data format 10 digit emergency number configured on the switch Civic Address physical address data such as city street number and building information Miscellaneous Administrative Tasks Org Version Type Length Unique 7 bits 9 bits Type Code Sub No Name 0 6 7 15 Length 3 bytes Type Information 15 Extended 127 7 Oxfe07 0x0012bb 4 PowerType PD device power via PowerSource PSE amp local a Power Priority Unknown PowerValue Refer to Power Values on page C 38 16 LLDP MED 127 min len gt 0x0012bb 5 Hardware part number and inventory 0 max len revision hardware lt 32 revision 17 LLDP MED 127 min len gt 0x0012bb 6 BootROM revision inventory 0 max len firmware lt 32 revision 18 LLDP MED 127 min len gt 0x0012bb 7 Application SIP revision inventory 0 max len software lt 32 revision 19 LLDP MED 127 min len gt 0x0012bb 8 MAC Address ASCII inventory 0 max len string serial lt 32 number 20 LLDP MED 127 11 OxfeOb
291. itive 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 32000 and L16 48000 are supported on SoundStation IP 6000 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 provisioning 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 provisioning 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
292. k When the user initiates a call the phone will go through the following steps to connect the call 1 Try to make the call using the working server 2 If the working server does not respond correctly to the INVITE then try and make a call using the next server in the list even if there is no current registration with these servers This could be the case if the Internet connection has gone down but the registration to the working server has not yet expired 3 If the second server is also unavailable the phone will try all possible servers even those not currently registered until it either succeeds in making a call or exhausts the list at which point the call will fail At the start of a call server availability is determined by SIP signaling failure SIP signaling failure depends on the SIP protocol being used as described below e If TCP is used then the signaling fails if the connection fails or the Send fails e If UDP is used then the signaling fails if ICMP is detected or if the signal times out If the signaling has been attempted through all servers in the list and this is the last server then the signaling fails after the 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 8 and Registration lt reg gt on page A 128 If DNS is used to resolv
293. key The phone will act as if Message Center was chosen Refer to Voice Mail Integration on page 4 56 Instant Messages will still be accessible from the Main Menu This attribute also includes e Message Waiting Indicator lt mwi gt Message Waiting Indicator lt mwi gt In the following table x is the registration number IP 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 A 143 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation msg mwi x subscribe ASCII encoded string containing Null If non Null the phone will send digits the user part of a SIP a SUBSCRIBE request to this URL or a string that constitutes contact after boot up a valid SIP URL 6416 or 6416 polycom com msg mwi x contact or registration Configures message retrieval callBackMode registration or and notification for the line disabled If set to contact a call will be placed to the contact specified in the callback attribute when the user invokes message retrieval If set to registration a call will be placed using this registration to the contact registered the phone will call itself If set to disabled message retrieval and message notification are disabled msg mwi x callBack ASCII encoded
294. kyAutoLineSeize is set to 0 or Null and this parameter is set to 1 this overrides the stickyAutoLineSeize behavior for hot dial only Any new call scenario seizes the next available line If call stickyAutoLineSeize is set to 0 or Null and this parameter is set to 0 or Null there is no difference between hot dial and new call scenarios Note A hot dial occurs on the line which is currently in the call appearance Any new call scenario seizes the next available line call singlekeyPressConference 0 1 0 If set to 1 the conference will be setup after a user presses the Conference soft key or Conference key the first time Also all sound effects dial tone DTMF tone while dialing and ringing back are heard by all existing participants in the conference If set to 0 or Null sound effects are only heard by conference initiator old behavior Only supported for SoundPoint IP 550 560 650 and 670 and SoundStation IP 7000 For all others set to 0 Configuration Files Permitted Attribute Values Default call localConferenceCallHold Oor1 0 Interpretation 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
295. l 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 Oor1 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 of times the phone will retry a notify when attempting to seize a line BLA volpProt SIP header diversion enable Oor1 If set to 1 the diversion header is displayed if received If set to 0 or Null the diversion header is not displayed volpProt SIP header diversion list useFirst Oor1 If set to 1 or Null the first diversion header is displayed If set to 0 the last diversion header is displayed A 13 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute volpProt SIP header warning codes accept Permitted Values comma separated list Default Null Interpretation 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
296. l SpeedDial Split Split Call screen only Transfer Transfer Call screen only Unavail serverACDAgentUnavailable Video Video Polycom VVX 1500 only VolDown VolDown VolUp VolUp Administrator s Guide SoundPoint IP SoundStation IP VVX Assigning a VLAN ID Using DHCP Note To assign a VLAN ID to a phone using DHCP gt gt Inthe DHCP menu of the Main setup menu set VLAN Discovery to Fixed or Custom When set to Fixed the phone will examine DHCP options 128 144 157 and 191 in that order for a valid DVD string When set to Custom the value set in VLAN ID Option will be examined for a valid DVD string DVD string in the DHCP option must meet the following conditions to be valid Must start with VLAN A case sensitive Must contain at least one valid ID VLAN IDs range from 0 to 4095 Each VLAN ID must be separated by a character The string must be terminated by a Allcharacters after the will be ignored There must be no white space before the VLAN IDs may be decimal hex or octal For example The following DVD strings will result in the phone using VLAN 10 VLAN A 10 VLAN A 0x0a VLAN A 012 If a VLAN tag is assigned by CDP DHCP VLAN tags will be ignored Miscellaneous Administrative Tasks The following figure shows the phone s processing to determine if the VLAN ID is valid DHCP Discover no VLAN tag
297. l 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 VVX family is for administrators who need to configure customize manage and troubleshoot SoundPoint IP SoundStation IP VVX phone systems This guide covers the SoundPoint IP 320 321 330 331 430 450 550 560 650 and 670 desktop phones the SoundStation IP 6000 and 7000 conference phones and the Polycom VVX 1500 business media phone The following related documents for SoundPoint IP SoundStation IP VVX family are available e Quick Start G
298. lag to determine if the dial plan applies to for calls made through the Polycom HDX or SoundStructure systems If set to 1 the dial plan applies If set to 0 or Null the dial plan does not apply This attributes also includes A 22 Digit Map lt digitmap gt Routing lt routing gt Configuration Files Digit Map lt digitmap gt A digit map is defined either by a string or by a list of strings Each string in the list is an alternative numbering scheme specified either as a set of digits or timers or as an expression over which the gateway will attempt to find a shortest possible match Digit map extension letter R indicates that certain matched strings are replaced Digit map timer letter T indicates a timer expiry Digit map protocol letters S and H indicate the protocol to use when placing a call The following examples shows the semantics of the syntax R RRxxxxxxx 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 seven 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 R9R6O4Rxxxxxxx Replaces 9 with 604 e R999R911R Convert 999 to 911 e xxR601R600Rxx When applied on 1160122 gives 1160022 e xR60xR600Rxxxxxxx Any 60x will be replaced with 6
299. lcl ml lang tags x in Multilingual lt ml gt on page A 26 volpProt SIP header warning enable Oori If set to 1 the warning header is displayed if received If set to 0 or Null the warning header is not displayed volpProt SIP acd signalingMethod Oori If set to 0 or Null the SIP B signaling is supported This is the older ACD functionality If set to 1 the feature synchronization signaling is supported This is the new ACD functionality Configuration Files Attribute volpProt SIP tcpFastFailover Permitted Values Oor1 Default Null Interpretation If set to 1 failover occurs based on the values of reg x server y retryMaxCount volpProt server x retryTimeOut If set to 0 this is old behavior If reg x tcpFastFailover is Null this attribute is checked If volpProt SIP tcpFastFailover is Null then this feature is disabled If both attributes are set the value of reg x tcpFastFailover takes precedence volpProt SIP strictReplacesHeader Oor1 Null This parameter applies only to directed call pick up attempts initiated against monitored BLF resources If set to 1 or Null the phone requires call id to tag and from tag to perform a directed call pickup when call directedCallPickupMethod is configured as native If set to 0 all that is required to perform the directed call pick up is a call id volpProt SIP use486forReject Oor1
300. ld code and length of the first sub option 1 1 octets If this is a field you want to parse save the data Skip to the start of the next sub option on kk w Repeat steps 3 to 5 until you have all the data or you encounter the End of Suboptions code OxFF For example the following is a sample decode of a packet from an IP601 3c 74 Option 60 length of Option data part of the DHCP spec 00 00 36 3d Polycom signature always 4 octects 6f Length of Polycom data 01 07 50 6 6c 79 63 6f 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 Miscellaneous Administrative Tasks Product Model and Part Number
301. le the display of specific standard keys the soft keys that are displayed on SoundStation IP SoundStation IP and Polycom VVX 1500 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 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 32x 33x Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note e Callers appears on the SoundPoint IP 32x 33x 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
302. le to reboot the phones You must review the provisioning server configuration make the correction and reapply the configuration file by restarting the phones This error also happens when phone does a restart not a reboot and finds a newer version of BootROM or application this triggers a reboot Usually this error is self recoverable Network link is down Since the SoundPoint IP SoundStation IP VVX 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 Administrator s Guide SoundPoint IP SoundStation IP VVX Status Menu the phone is not in the menu system and will remain on screen until the link problem is resolved Call related functions for example soft keys and feature keys disabled when the network is down however the menu works Status When the phone is unable to register with the call control server the icon a isshown outline Once the phone is registered the icon mis shown solid On the SoundStation IP 7000 the icons are and On the Polycom 1500 the icons are and i Flashing 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 of
303. ll 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 32x 33x 430 450 550 560 650 and 670 phones only The Busy Lamp Field BLF attendant console feature enhances support for a phone based attendant console A 145 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation attendant uri string Null For attendant console busy lamp field BLF feature This specifies the list SIP URI on the server If this is just a user part the URI is constructed with the server host name IP Note If attendant uri is set then the individally addressed users configured by attendant resourceList and attendant beh
304. ll 1 to 389 TCP This parameter is used to specify 65535 636 TLS the port to connect to on the server if a full URL is not provided dir corp transport TCP TLS Null TCP This parameter is used to specify whether a TCP or TLS connection is made with the server if a full URL is not provided dir corp baseDN UTF 8 encoded Null The base domain name is the string starting point for making queries on the LDAP server dir corp user UTF 8 encoded Null The username used to authenticate string to the LDAP server 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 pase 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Aitribute dir corp attribute x name Permitted Values UTF 8 encoded string Default Null Interpretation The name of the attribute to match on the server Each name must be unique however an LDAP entry can have multiple attributes with the same name
305. ll need to re enable the Screen Capture feature after any phone restart or reboot 3 Using your chosen browser do the following a To get your phone s IP address press the Menu key and then select Status gt Platform gt Phone Scroll down to see the IP address b As the browser address enter http lt phone s IP address gt captureScreen The current screen that is shown on the phone is shown in the browser window The image can be saved as a BMP or JPEG file LLDP and Supported TLVs The Link Layer Discovery Protocol LLDP is a vendor neutral Layer 2 protocol that allows a network device to advertise its identity and capabilities on the local network The protocol was formally ratified as IEEE standard 802 1AB 2005 in May 2005 Refer to section 10 2 4 4 of the LLDP MED standard at http www tiaonline org standards technology voip documents ANSI TIA 1057_final_for_publication pdf Administrator s Guide SoundPoint IP SoundStation IP VVX The LLDP feature added in SIP 3 2 0 supports VLAN discovery and LLDP power management but not power negotiation LLDP has a higher priority than CDP and DHCP VLAN discovery The following Type Length Values TLVs are supported e Mandatory Chassis ID Must be first TLV Port ID Must be second TLV Time to live Must be third TLV set to 120 seconds End of LLDPDU Must be last TLV LLDP MED Capabilities LLDP MED Network Policy VLAN L2 QoS L3 QoS
306. lletin 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 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 replaced 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 Configuration Files Quotas lt quotas gt This configu
307. ls the TX frame size If set to 1 CIF is used provided the far end supports CIF 1 otherwise QCIF is used This value is H263 1998 90000 format parameter QCIF used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 SqcifMpi 1 default to 32 This value is H263 1998 90000 format parameter SQCIF used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 annexF Oor1 This value is H263 1998 90000 format default Null parameter ANNEXF used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 annexl Oor1 This value is H263 1998 90000 format default Null parameter ANNEXI used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 annexJ Oor1 This value is H263 1998 90000 format default Null parameter ANNEXJ used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 annexT Oor1 This value is H263 1998 90000 format default Null parameter ANNEXT used to signal Polycom VVX 1500 receiving capability in the SDP Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute video profile H2631998 annexK Permitted Values 0 or 1 default 2 3 4 Interpretation This value is H263 1998 90000 format parameter ANNEXK used to signal Polycom VVX 1500 receiving capability in the SDP video profile H2631998 annexN 0 o
308. lude hangup hu hold h e waitconnect wc pause lt number of seconds gt p lt num sec gt where the maximum value is 10 T lt type gt The embedded action type Multiple actions can be defined Supported action types include invite e 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 dependent and may require the addition of star codes Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note Name Interpretation 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 execution of the action string fails P lt prompt num gt N lt num The user input prompt string digits gt 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 18 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 P
309. lycom VVX 1500 37 APP 13 1 a 3 20 14 40 34 28 A 45 HAS S 22 42 36 30 23 17 72 5 R y Key ID Key ID Function Key ID Function Key ID Function Key ID Function 1 Messages 12 MicMute 23 Headset 34 Dialpad5 2 ArrowLeft 13 Directories 24 n a 35 Dialpad8 3 Select 14 Redial 25 Menu 36 Dialpad0 4 ArrowRight 15 Conference 26 n a 37 Applications 5 Delete 16 DoNotDisturb 27 Dialpad3 38 n a 6 n a 17 Handsfree 28 Dialpad6 39 Dialpad1 7 n a 18 VolUp 29 Dialpad9 40 Dialpad4 8 ArrowUp 19 n a 30 DialpadPound 41 Dialpad7 9 ArrowDown 20 Video 31 n a 42 DialpadStar 10 n a 21 Transfer 32 n a 11 n a 22 Hold 33 Dialpad2 Internal Key Functions C 18 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 The Label value is case sensitive 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 32x 33x phone because it does not display as many soft keys Miscellaneous Administrative Tasks On the Sound Point IP 32x 33x phone CallPickup and ParkedPickup refer to the same function On other phones CallPickup refers to the soft key function that
310. mation 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 gl and received calls appear as f 5 The call list feature can be disabled on all SoundPoint IP and SoundStation IP platforms except the SoundPoint IP 32x 33x and SoundStation IP 7000 Configuration changes can be performed centrally at the provisioning server Central provisioning server sip cfg Configuration File Enable or disable all call lists or individual call lists e For more information refer to Feature lt feature gt on page A 110 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 th
311. mation on or off provisioning sip cfg e For more information refer to Indicators lt ind gt on page A 95 server To replace the animation used for the idle display For more information refer to Animations lt anim gt lt IP_330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt on page A 96 To change the position of the idle display animation e For more information refer to Graphic Icons lt gi gt lt IP_330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt on page A 98 Ethernet Switch The SoundPoint IP phones except the SoundPoint IP 32x and the Polycom VVX 1500 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 phone can be powered through a local 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
312. max reliability bit in the IP TOS field of the IP header or else don t set it qos ip rtp video min_cost Oor1 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 video 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 Call Control lt callControl gt These parameters apply to call control packets such as the network protocol signaling Permitted Attribute Values Default Interpretation qos ip callControl dscp 0 to 63 or Null This parameter allows the DSCP of EF or packets to be specified If set to a any of value this will override the other AF11 AF12 gos ip callControl AF13 AF21 parameters Default of Null which AF22 AF23 means the other AF31 AF32 gos ip callControl AF33 AF41 parameters will be used AF42 AF43 qos ip callControl min_delay Oor1 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 Oor1 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 Oor1 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 Oor1 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
313. may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller The SoundPoint IP 650 and 670 and the Polycom VVX 1500 phones can be configured to allow recording of audio calls on a supported USB device The filenames of the recorded wav files will include a date time stamp for example 20A pr2007_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 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 Supported USB Devices for SoundPoint IP 650 and 670 and Polycom VVX 1500 Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuration changes can be performed centrally at the provisioning server Central provisioning server Configuration file Turn this featur
314. me sound reboot A 35 Index Index 7 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Index 8 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 other 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 no
315. mes but there are cases where it is impossible to move backward Make special note of these cases before upgrading For the latest information refer to the latest Release Notes Multiple Key Combinations C 10 Note 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 SoundStation IP or Polycom VVX phones refer to Quick Tip 18298 Resetting and Rebooting SoundPoint IP SoundStation IP VVX Phones at http www polycom com support voice Rebooting the Phone For the key combination press and hold certain key combinations depending on the phone model simultaneously until a confirmation tone is heard or for about three seconds e IP 32x 33x Volume Volumet Hold and Hands free e IP 430 Volume Volumet Hold and Messages e IP 450 550 560 600 601 and 650 and 670 Volume Volume Mute and Messages e IP 6000 Volumet and Select e IP 7000 Volume and Volume e VVX 1500 Delete Volume Volume and Select As of SIP 3 2 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 Miscellaneous Administrative Tasks Restoring Factory Defaults For the key combination press and hol
316. mily LCD Power Saving Note Note 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 depends on your call server 1 Update sip cfg as follows softkey 2 label ToVMail softkey 2 action 55S P1N10S S Tinvite softkey 2 use alerting 1 2 Reboot the phone 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 This feature is only supported on the Polycom VVX 1500 This feature applies during configured non working hours and when the phone is idle Working hours are defined in the configuration files and users can change the default values through the phone s menu to accommodate their individual schedules The Polycom VVX 1500 phone enters power saving mode after it has been idle for a certain period of time and its camera doesn t detect motion The phone s ability to detect the users presence is biased for easy detection during office hours and for difficult detection during off hours Configuration changes can be performed centrally at the provisioning server Central provisioning server Configuration file Turn this feature on or off and configure how it works sip cfg e For more information refer to LCD Power Saving lt powerSaving gt on page A 126 Shared Call Appearances Calls and lines on multiple
317. ministrator 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 e Check same status of handset headset if connected and through the Hands Free Speakerphone Outbound or inbound calling is unsuccessful Do one of the following Place a call to the phone under investigation Check that the display indicates incoming call information Lift the handset Ensure dial tone is present and place a call to another extension or number Check that the display changes in response The line icon shows an unregistered line icon The phone line is unregistered Contact your system administrator 5 14 Displays Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones Symptom Problem Corrective Action There is no display The display is incorrect The display has bad contrast Power is not correctly applied to the SoundPoint IP family SIP phone The contrast needs adjustment Do one of the following 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 Use the screen capture feature Refer to Capturing
318. mitted 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note Name Interpretation 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 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 This parameter is not required if there are no efk efklist entries In SIP 3 0 1 is the only supported version In SIP 3 1 or later 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 or ASCII 0x27
319. mple 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 Sie Lee ht Park success 801 Veena 7 ie 2a Configuration 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 11 Quick Setup of SoundPoint IP SoundStation IP VVX Phones In the SIP 3 1 2 release a Quick Setup feature was added to simplify the process of entering the provisioning boot server parameters from the phone s user interface This feature is designed to make it easier for on site out of the box provisioning of SoundPoint IP SoundStation IP and VVX phones When enabled this feature will present a QSetup soft key to the user When t
320. 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 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 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 registe
321. n Menu Setting up Your System 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 8 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 10 SNTP Address dotted decimal IP address Simple Network Time Protocol SNTP server from OR which the phone will obtain the current time domain name string GMT Offset 13 through 12 Offset of the local time zone from Greenwich Mean Time GMT in half hour increments DNS Server dotted decimal IP address Primary server to which the phone directs Domain Name 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 12 EM Power Enabled Disabled This parameter is relevant if
322. n file refer to the Configuration File Management on SoundPoint IP Phones whitepaper at www polycom com support voice 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 110 Configuration file Specify two calls per line key phonet cfg 7 For more information refer to Registration lt reg gt on page A 128 xml 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 Macro names must be embedded into the contact cn fields with the P 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 11 Useful Tips The following information should be noted Activation of the enhanced feature key will fail if configured values are invalid except where noted in previous sections All failures are logged at level 4 minor If two 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 A sequence of characters prefixed with The exception is the speed dial reference which starts with conta
323. n parameter in its SDP If sec srtp sessionParams noAuth require s setto 1 sec srtp sessionParams noAuth offer is logically set to 1 no matter what the value in the configuration file If set to 0 or Null authentication is required noEncrypRTCP offer sec srtp sessionParams Oor1 Null If set to 1 no encryption of RTCP is offered A session description that includes the UNENCRYPTED_SRTCP session parameter is sent when initiating a call If set to 0 or Null encryption of RTCP is offered A 106 Configuration Files Attribute sec srtp sessionParams noEncrypRTCP require Permitted Values Oor1 Default Null Interpretation If set to 1 no encryption of RTCP is required A call placed to a phone configured with noAuth require must offer the UNENCRYPTED_SRITCP session parameter in its SDP If sec srtp sessionParams noEncryptRTICP requi re is set to 1 sec srtp sessionParams noEncryptRICP offer is logically set to 1 no matter what the value in the configuration file If set to 0 or Null encryption of RTCP is required sec srtp sessionParams noEncrypRTP offer Oor1 Null If set to 1 no encryption of RTP is offered A session description that includes the UNENCRYPTED_SRTP session parameter is sent when initiating a call If set to 0 or Null encryption of RTP is offered sec srtp sessionParams noEncrypRTP require Oor1 Null If set to 1 no en
324. n 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 provisioning server and it can encrypt files before uploading them to the provisioning server Custom Certificates When trying to establish a connection to a provisioning server for application provisioning the phone trusts certificates issued by widely recognized certificate authorities CAs Incoming Signaling Validation Levels of security are provided for validating incoming network signaling Secure Real Time Transport Protocol Encrypting audio streams to avoid interception and eavesdropping Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 2 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 4 Any new features introduced after SIP 2 1 4 are not supported Refer to the S P 2 1 Administrator Guide which is available at http Awww polycom com global documents support setup_maintenance products v oice sip_2 1_addendum_to_sip_2 0_administrator 27s_guide pdf The SoundPoint IP 301 501 600 and 601 and the SoundStation IP 4000
325. n place of normal dial tone to indicate that one or more messages voice mail are waiting at the message center se appLocalEnabled Oor1 1 If set to 1 local user interface sound effects such as confirmation error tones will be enabled This attribute also includes e Patterns lt pat gt e Ring type lt rt gt Patterns lt pat gt Patterns use a simple script language that allows different chord sets or wave files to be strung together with periods of silence The script language uses the following instructions Instruction Meaning Example sampled n Play sampled audio file se pat callProg x inst y type Sampled sampled audio n file instruction type se pat callProg x inst y value 3 specifies sampled audio file 3 chord n d Play chord set n d is se pat callProg x inst y type chord chord set optional and allows the instruction type chord set ON duration to se pat callProg x inst y value 3 specifies call 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 br
326. n upload overrides Oori If set to 0 the phone specific configuration override file lt Ethernet Address gt phone cfg is uploaded unencrypted regardless of how it was downloaded This will replace the override file on the server even if it is encrypted If set to 1 the phone specific configuration override file is uploaded encrypted regardless of how it was downloaded This will replace the override file on the server even if it is unencrypted Password Lengths lt pwd gt lt length gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation sec pwd length admin 0 32 1 Password changes will need to be at least this long Use 0 to allow null passwords sec pwd length user 0 32 2 A 104 Note SRTP lt srtp gt Configuration Files As per RFC 3711 you cannot turn off authentication of RTCP This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation sec srtp enable Oor1 Null If set to 1 or Null the phone accepts SRTP offers If set to 0 the phone always declines SRTP offers sec srtp offer Oori Null If set to 1 or Null the phone includes a secure media stream description along with the usual non secure media description in the SDP of a SIP INVITE This is for the phone initiating offering a phone call If set to 0 no secure media stre
327. nNn 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 If the maximum number of characters is greater than 32 or less than one macro execution fails 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 Macros 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 11 Note Configuring Your System 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 SIP 3 2 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 configuratio
328. nabled Disabled Refer to the VLAN Filtering parameter in Ethernet Menu on page 3 12 device net etherStormFilter Enabled Disabled Refer to the Storm Filtering parameter in Ethernet Menu on page 3 12 device net etherModeLAN 1to5 Refer to the LAN Port Mode parameter in Ethernet Menu on page 3 12 device net etherModePC 1to5 Refer to the PC Port Mode parameter in Ethernet Menu on page 3 12 device serial enable 0 1 Enables the debug serial port The default value is 1 device sec SSL certList all custom default The type of certificate list device sec SSL customCert X 509 certificate The certificate value device net lldpEnabled Oor1 If set to 1 the phone will attempt to determine its VLAN ID and negotiate power through LLDP If set to 0 the phone will not attempt to determine its VLAN ID or power management through LLDP 154 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 Ext
329. national 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 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
330. nction 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 SoundPoint IP 430 Miscellaneous Administrative Tasks 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 13 Administrator s Guide SoundPoint IP SoundStation IP VVX SoundPoint IP 450 se C y 7 d do do da is Qs o OF Qe OIO O OLORE OO G CeO
331. nd 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 Configuring Your System 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 Recording and Playback of Audio Calls Note Note This feature requires a license key for activation except for the Polycom VVX 1500 Using this feature
332. ndStation IP VVX Family we POLYCOM 3 Copy all files from the distribution zip file to the phone home directory Maintain the same folder hierarchy There are two distribution zip files The combined image file contains sip ld sip cfg phonel cfg 000000000000 cfg 000000000000 directory xml SoundPointIP dictionary xml one of each supported language SoundPointIPWelcome wav The split image file contains individual sip Id files for each model as well as the configuration files and dictionary files Refer to the latest Release Notes for a detailed description of each file in the distribution and further information on determining which distribution to use Provisioning Server Security Policy You must decide on a provisioning server security policy Polycom recommends allowing file uploads to the provisioning server where the security environment permits This allows event log files to be uploaded and changes made by the phone user to the configuration through the web server and local user interface and changes made to the directory to be backed up This greatly eases our ability to provide customer support in diagnosing issues that may occur with the phone operation For organizational purposes configuring a separate log file directory override directory contact directory and license directory is recommended but not required The different directories can have different access permissions
333. ndling of missedCalls depends on how the server is configured A 135 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 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 table x is the registration number IP 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Attribute Permitted Values Default Interpretation divert x contact ASCII encoded string Null The forward to contact used for containing digits the user all automatic call diversion part of a SIP URL or a string features unless overridden by a that constitutes a valid SIP specific contact of a per call URL 6416 or diversion feature refer to 6416 polycom com below divert x autoOnSpecificCaller Oor1 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 param
334. nencrypted file and using the SDK to facilitate key generation refer to Encrypting Configuration Files on page C 4 Configuration changes can be performed centrally at the provisioning server Configuration File Specify the phone specific contact directory and the sip cfg phone specific configuration override file e For more information refer to Encryption lt encryption gt on page A 104 Configuration file Change the encryption key lt device gt cfg e For more information refer to refer to Flash Parameter Configuration on page A 151 Digital Certificates Starting in May 2009 Polycom is installing a digital certificate on certain SoundPoint IP phone models at the manufacturing facility Over time other SoundPoint IP phone models as well as all SoundStation IP and Polycom VVX phone models will have a digital certificate Refer to Technical Bulletin 37148 Digital Certificates on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html This X 509 digital certificate is signed by the Polycom Root CA and may be used for a server to authenticate the phone when initiating Transport Layer Security TLS based communications such as those used for HTTPS provisioning and TLS SIP signalling encryption The Polycom Root CA can be downloaded from http pki polycom com pki Polycom Root CA crt The X 509 digital cert
335. nes can also be customized For more information refer to Chord Sets lt chord gt on page A 33 and Call Progress Patterns on page A 37 This attribute includes e Multilingual lt ml gt e Date and Time lt datetime gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Multilingual lt ml gt The multilingual feature is based on string dictionary files downloaded from the provisioning server These files are encoded in standalone XML format Several eastern European and Asian languages are included with the distribution Attribute Permitted Values Interpretation Icl ml lang Null If Null the default internal language US OR English will be used otherwise the An exact match for one of the label names stored in lcl ml lang menu x label language to be used may be specified in the format of lcl ml lang menu x label For example to get the phone to boot up in German 1cl ml lang Deutsch de de Icl ml lang menu x Icl ml lang menu x label Icl ml lang clock x 24HourClock String in the format language_region 0 1 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 pr
336. ng MGCP described in 2 1 5 0 2 9poxorxxxxx 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 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 24 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 V
337. ng packets Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify protocol specific IP TOS settings provisioning sip cfg For more information refer to IP TOS lt IP gt on page A 68 server Local Web Server Specify IP TOS settings if enabled Navigate to http lt phonelPAddress gt netConf htm qo IEEE 802 1p Q The phone will tag all Ethernet packets it transmits with an 802 10 VLAN header for one of the following reasons e When it has a valid VLAN ID set in its network configuration Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 8 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 Configuration changes can be performed centrally at the provisioning server or locally Central Configuration file Specify default and protocol specific 802 1p Q settings provisioning sip cfg For more information refer to Ethernet IEEE 802 1p Q server lt ethernet gt on page A 67 Local Web Server Specify 802 1p Q settings i
338. nguages 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 file in UTF 8 but also ensuring the UTF 8 characters chosen are within the Unicode character ranges indicated in the tables below Place the file in an appropriately named folder according to the format language_region parallel to the other dictionary files under the SoundPointIPLocalization folder on the provisioning server Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 4 Addaicl ml lang clock menu x attribute to the configuration file Add lcl ml lang clock x 24HourClock Ilcl ml lang clock x format lcl
339. nother contact and the contact to call when the user accesses voice mail For more information refer to Messaging lt msg gt on page A 143 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 430 450 550 560 650 and 670 and the MSG key on the Polycom VVX 1500 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 provisioning server Changes will permanently override global settings unless deleted through the Reset Local Config menu selection Multiple Registrations The SoundPoint IP 32x 33x and 430 support a maximum of two registrations the SoundPoint IP 450 supports three registrations the SoundPoint IP 550 and 560 support four and the SoundPoint IP 650 and 670 and the Polycom VVX 1500 support 6 Up to three SoundPoint IP Expansion Modules can be added to a single host SoundPoint IP 650 and 670 phone in
340. nt 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 Santa 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
341. ntains the following parameters Name mname Interpretation 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 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 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 pag
342. ntil 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 32x 33x 430 450 550 560 650 and 670 and SoundStation IP 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 for the SoundPoint IP SoundStation IP VVX Phones Polycom does not recommend the remapping for keys POLYCOM In the following table x IP_ IP_330 IP 430 IP_450 IP_550 IP_650 and IP_4000 and IP_7000 and y is the key number Note that IP_330 parameters affect SoundPoint IP 32x 33x phones IP_430 parameters affect SoundPoint IP 430 phones IP_550 parameters affect SoundPoint IP 550 and 560 phones IP_650 parameters affect SoundPoint IP 650 and 670 phones IP_4000 parameters affect the SoundStation IP 6000 phones and IP_7000 parameters affect the SoundStation IP 7000 phones IP 330 y 1 34 IP 430 y 1 35 IP_550 y 1 40 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
343. nto the network configuration SIP is the Internet Engineering Task Force IETF standard for multimedia communications 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 VVX phones to successfully operate as a SIP endpoint in your network it must meet the following requirements e A working IP network is established e Routers are configured for VoIP e VoIP gateways are configured for SIP e The latest or compatible SoundPoint IP SoundStation IP VVX phone SIP application image is available e Acall 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 VVX Phones Fit e Session Initiation Protocol Application Architecture e Available Features e New Features in SIP 3 2 To install your SoundPoint IP SoundStation IP VVX phones on the network refer to
344. o 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 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 38 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 37 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 val
345. oint 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 from computer applications such as Microsoft Office Communicator Administrator s Guide SoundPoint IP SoundStation IP VVX 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 11 and Registration lt reg gt on page A 128 and needs to be activated by a feature application key SIP for Instant Messaging and Presence Leveraging Extensions The phone is compatible wi
346. oint IP SoundStation IP VVX Family c Edit contents of phone MACaddress cfg if desired For example edit the parameters 2 Create new configuration file s in the style of sip cfg by performing the following steps a Create application sipXXXX cfg file by using the sip cfg file from the distribution as templates For more information on the sip cfg file refer to Application Configuration on page A 5 Edit contents of sipXXXX cfg if desired For example edit the parameters 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 Optional Disable the local web HTTP server or change its signalling port if local security policy dictates refer to Web Server lt httpd gt on page A 75 Change the default location settings for user interface language and time and date format refer to Localization lt Icl gt on page A 25 3 Create a master configuration file by performing the following steps a Create per phone or per platform lt MACaddress gt cfg files by using the 00000000000 cfg and files from the distribution as templates For more information refer to Master Configuration Files on page A 2 Edit the CONFIG_FILES attribute of
347. ointIP SPIP_560 Note Miscellaneous Administrative Tasks Model Model Name IP 650 SoundPointIP SPIP_650 IP 670 SoundPointIP SPIP_670 IP 6000 SoundStationIP SSIP_6000 IP 7000 SoundStationIP SSIP_7000 VVX 1500 VVX VVX_1500 PMD Advertise and Operational MAU PMD Advetise Mode Speed Capability Bit Operational MAU Type 10BASE T half duplex 1 10 mode 10BASE T full duplex 2 11 mode 100BASE T half duplex 4 15 mode 100BASE T full duplex 5 16 mode 1000BASE T half duplex 14 29 mode 1000BASE T full duplex 15 30 mode Unknown 0 0 By default all phones have the PMD Advertise Capability set for 10HD 10FD 100HD and 100FD bits For SoundPoint IP 560 and IP 670 and Polycom VVX 1500 phones that have Gigabit Ethernet support PMD Advertise Capability also contains set 1000FD bit 37 Administrator s Guide SoundPoint IP SoundStation IP VVX Power Values Power Value Sent in LLDP MED Extended Model Power Usage Watts Power Via MDI TLV IP 320 330 4 5 45 IP 321 331 4 5 45 IP 430 4 5 45 IP 450 4 5 45 IP 550 6 60 IP 560 8 80 IP 650 with EM 12 120 IP 670 with EM 14 140 IP 6000 10 5 105 IP 7000 10 5 105 VVX 1500 14 140 Note By default the power values for the SoundPoint IP 650 and 670 are sent for the phone and the Expansion Module s The values are not adjusted when the
348. on as there is no active audio on the phone independently of any call state If set to call a request can be executed as soon as there are no calls in any state on the phone Feature lt feature gt 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 presence is the presence feature including management of buddies and own status messaging messaging is the instant messaging feature directory directory is the local directory feature calllist calllist is the locally controlled call lists Note The call list feature can be disabled on all SoundPoint IP and SoundStation IP models except the SoundPoint IP 32x 33x and SoundStation IP 7000 ring download ring download is run time downloading of ringers calllist received calllist received is the received calls list feature the calllist feature must be enabled for this feature to be available calllist placed calllist placed is the placed calls list feature the calllist feature must be enabled for this feature to be available calllist missed calllist missed is the missed calls list feature the calllist feature must be enabled for this feature to be available u
349. on the SoundPoint IP 32x 33x 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 Aspeed dial Anenhanced feature key operation Configuring Your System 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 be performed centrally at the provisioning server Central provisioning server Configuration File Set the key scrolling timeout key functions and sub pointers for each sip cfg key usually not necessary For more information refer to Keys lt key gt on page A 89 For more information on the default feature key layouts refer to Default Feature Key Layouts on page C 12 Multiple Line Keys per Registration More than one Line Key can be allocated to a single registration phone number or line on SoundPoint IP and Polycom VVX 1500 phones The number of Line Keys allocated per registration is configurable Configuration changes can be performed centrally at the provisioning server or locally Central provision
350. or the following advanced features e Configurable Feature Keys e Multiple Line Keys per Registration e Multiple Call Appearances e Customizable Fonts and Indicators e Instant Messaging e Multilingual User Interface e Downloadable Fonts e Synthesized Call Progress Tones e Microbrowser e Application Launch Pad e Real Time Transport Protocol Ports e Network Address Translation e Corporate Directory e Recording and Playback of Audio Calls e Digital Picture Frame e Enhanced Feature Keys e Configurable Soft Keys Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family LCD Power Saving This section also provides information for making configuration changes for the following advanced call server features Shared Call Appearances Bridged Line Appearance Busy Lamp Field Voice Mail Integration Multiple Registrations SIP B Automatic Call Distribution Feature Synchronized Automatic Call Distribution Server Redundancy Presence Microsoft Live Communications Server 2005 Integration Access URL in SIP Message Static DNS Cache Display of Warnings from SIP Headers Quick Setup of SoundPoint IP SoundStation IP VVX Phones 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 6000 and 7000 and the Polycom VVX 1500 can be remapped Since there is no Redial key
351. ormal push messages 3 Both Allows both critical and normal push messages 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 for the SoundPoint IP SoundStation IP VVX Phones 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 standAlone masterOnly masterPreferred slaveOnly slavePreferred Null The role of the SoundStation IP 7000 when communicating with
352. ormally AES should be used whenever AEC is used for handsfree or handset and both are enabled by default for those terminations 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 voice aes hd enable oO 1oO N voice aes hd duplexBalance e voice aes hf enable voice aes hf duplexBalance 0 voice aes hf duplexBalance 1 voice aes hf duplexBalance 2 voice aes hf duplexBalance 3 voice aes hf duplexBalance 4 voice aes hf duplexBalance 5 voice aes hf duplexBalance 6 voice aes hf duplexBalance 7 mo wo A AJ AJ oJ oO NIN voice aes hf duplexBalance 8 oO voice aes hf duplexBalance IP_4000 0 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 Jol NI oa voice aes hf duplexBalance IP_4000 5 Configuration Files Attribute Default voice aes hf duplexBalance IP_4000 6 4 voice aes hf duplexBalance IP_4000 7 3 voice aes hf duplexBalance IP_4000 8 2 Background Noise Suppression lt ns gt These settings control the performance of the transmit background noise suppression feature Polycom recommends that you do not change these val
353. ormation refer to Missed Call Configuration lt serverMissedCall gt on page A 134 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 For example Bob places a call to Alice but he ends up connected to Fred 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 for speakerphone handset and headset separately While transmit levels are fixed according to the TIA EIA 810 A standard receive volume is adjustable For SoundPoint IP 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 be performed centrally at the provisioning server Central Configuration file Adjust receive and handset headset volume provisioning sip cfg e For more information refer to Volume Persistence lt volume gt on server page A 47 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Cu
354. ost often this will be used to set the VLAN that the phone should use for voice traffic It also reports power management to the switch The default value is Enabled If the switch does not support it VLAN Discovery is used Refer to DHCP Menu on page 3 8 There are four ways to get VLAN on the phone and they can all be turned on but the VLAN used is chosen by priority of each method The priority is 1 LLDP 2 CDP 3 DVD VLAN Via DHCP 4 Static VLAN ID entered in config menu For more information refer to LLDP and Supported TLVs on page C 29 CDP Compatibility Enabled Disabled If enabled the phone will use CDP compatible signalling to communicate with the network switch for certain network parameters Most often this will be used to set the VLAN that the phone should use for Voice Traffic and for the phone to communicate its PoE power requirements 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 Setting up Your System Name Possible Values Description Storm Filtering Enabled Disabled Filter received Ethernet packets so that the TCP IP s
355. ostname 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Alert Information lt alertInfo gt 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 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 40 will be INVITE 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 Defa
356. other Polycom devices If the attribute is not defined or is null the default value is auto meaning that the configuration of the peer role is automatic The other value definitions are 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 e slaveOnly IP 7000 is always the slave 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 pnet hdx ext string Null The HDX Extension Number to be displayed on the IP 7000 when it is connected to an HDX system callProgAtten pnet remoteCall 60 to 0 Null The attenuation applied to tones played by the IP 7000 for POTS calls when it is connected to an HDX system when the HDX is the active speaker If set to Null the default is 15 localDialTone pnet remoteCall Oor1 Null A flag to determine whether or not a dialtone is played when the IP 7000 makes an outgoing POTS call when it is connected to an HDX If set to 1 a dial tone is played If set to 0 or Null a dial tone is not played DNS Cache lt dns gt A 120 In the tables below a maximum of 12 entries of NAPTR SRV and A re
357. oundPoint 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 WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT INDIRECT INCIDENTAL SPECIAL EXEMPLARY OR CONSEQUENTIAL DAMAGES INCLUDING BUT NOT LIMITED TO PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES LOSS OF USE DATA OR PROFITS OR BUSINESS INTERRUPTION HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY WHETHER IN CONTRACT STRICT LIABILITY OR TORT INCLUDING NEGLIGENCE OR OTHERWISE ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE The licence and distribution terms for any publicly available version or derivative of this code cannot be changed i e this code cannot simply be copied and put under anot
358. oundPoint IP 32x 33x 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 Polycom VVX 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 64 Configuration changes can performed centrally at the provisioning server or locally Central XML file The lt sd gt x lt sd gt element in the lt Ethernet address gt directory xml provisioning lt Ethernet file links a directory entry to a speed dial resource within the phone server address gt directory Speed dial entries are mapped automatically to unused line keys line xml keys are not available on the SoundStation IP 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 10 Local Local Phone User The next available Speed Dial Index is assigned to new directory 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 provision
359. our SoundPoint IP SoundStation IP VVX Phones 1 6 2 Overview 69 os ke oe hs Se hee een 22l Where SoundPoint IP SoundStation IP VVX Phones Fit 2 2 Session Initiation Protocol Application Architecture 2 3 BootROM 0 een i bce es ee ee Eee eee eden A 2 3 Application ssri ettepet bodice EEE bd E A E as 2 4 COMMUTATION esssisii tisi iai i kna i pai aiis 2 5 Resource Files scc 5 00 ereere bebe tebe tia a dees 2 8 Available Features 5 02 ccs 5 s osesendes bees EENE nE eats eases 2 8 New Features in SIP 3 2 2 0 eee eee 2 14 3 Setting up Your System cc cee eee eee eee Ord Setting Up the Network 0 02 eee eee ee 3 2 DHCP or Manual TCP IP Setup 0 eee eee ee eee 3 2 Supported Provisioning Protocols 0 00 008 3 4 Modifying the Network Configuration 000 3 6 Setting Up the Provisioning Server 000 00 000 3 14 Deploying Phones From the Provisioning Server 3 17 Upgrading SIP Application 0 eee eee 3 21 Supporting SoundPoint IP SoundStation IP and Polycom VVX PHONES s2 cncodin oti wh berrdigindeee enti Limba ames aes 3 22 Supporting SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones 0 0 cece eee eee 3 23 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 4 Configuring Your System
360. override global settings unless deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the provisioning server Local Phone User Specify per registration line type private or shared using the SIP Interface Configuration menu Either the Web Server or the provisioning 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 Incoming calls can be presented to multiple phones 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 Bridged line appearances and shared call appearances are two different signalling methods of implementing a feature whereby more than one phone can share the same line or registration These implementations are dependent on the SIP server The methods are mutually exclusive and you should confirm with the call server vendor which if any method is supported
361. override the servers TCPpreferred or tr a act specified in sip cfg in Server lt server gt on UDPOnly or A 8 TLS or page a 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 Oor1 Null page A 8 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 reg x server y retry TimeOut Null or Null 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 Oor 1 0 This attribute overrides the reg x 1cs If set to 1 the Microsoft Live Communications Server is supported for registration x 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 133 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Do Not Disturb lt donotdisturb gt This configuration attrib
362. ovisioning server for the phone to be able to locate the dictionary file For example lcl ml lang menu 8 German_Germany lcl ml lang menu 8 label Deutsch de de 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 Oor1 If attribute present overrides lcl datetime date longFormat If 1 display the day and month in long format Friday November otherwise use abbreviations Fri Nov Icl ml lang clock x dateTop Oor1 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 la
363. ownloadable Fonts New fonts can be loaded onto the phone Enhanced Busy Lamp Field Allows an attendant to see a remote line that is Ringing and answer a remote ringing call using a single key press Also allows the attendant to view the caller id of remote active and ringing calls This feature may require call server support Enhanced Feature Keys Allows customers to redefine soft keys to suit their needs In SIP 3 0 this feature required a license key Instant Messaging Supports sending and receiving instant text messages Microbrowser The SoundPoint IP 430 450 550 560 600 601 650 and 670 desktop phones the SoundStation IP 6000 and 7000 conference phones and the Polycom VVX 1500 phones support an XHTML microbrowser The Polycom VVX 1500 phones also support the Application Launch Pad 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 have multilingual user interfaces Multiple Call Appearances The phone supports multiple concurrent calls The hold feature can be used to pause activity on one call and switch to another call Multiple Line Keys per Registration
364. owser 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 Info 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 for the SoundPoint IP SoundStation IP VVX Family Static DNS Cache 4 72 e An expires parameter is defined to indicate
365. pair This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation tcplpApp keepalive tcp idleTransmitinterval 10 to 7200 Null After idle x seconds the keep alive message is sent to the call server If set to Null the default value is 30 seconds Note If this parameter is set to a value that is out of range the default value is used tcplpApp keepalive tcp 5 to 120 Null If no response is received to noResponseTrasmitinterval 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 Oor1 0 If set to 1 enable TCP keep alive for SIP signalling connections that use TLS transport If set to 0 disable TCP keep alive for SIP signalling connections that use TLS 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones This configuration attribute is defined as follows
366. 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 powerful and should be used with caution For example an incorrect setting could set the IP Address of multiple phones to the same value Note that some parameters may be ignored for example if DHCP is enabled it will still override the value set with device net ipAddress Individual parameters are checked to see whether they are in range however the interaction between parameters is not checked If a parameter is out of range an error message will appear in the log file and parameter will not be used Incorrect configuration could cause phones to get into a reboot loop For example server A has a configuration file that specifies that server B should be used which has a configuration file that specifies that server 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 A 151 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones The flash attributes are defined as follows Name Possible Values Description device set Oor1 If set
367. 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 128 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation call stickyAutoLineSeize Null 0 or 1 0 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 Dialing through the call list when there is no active call will use the line index for the previous call Dialing through the call list when there is an active call will use the current active call line index Dialing through the contact directory will use the current active call line index If set to O or Null the feature is disabled this was the behavior in SIP 1 6 6 Dialing through the call list will use the line index for the previous call Dialing through the contact directory will use a random line index Note This may fail due to glare issues in which case the phone may select a different available line for the call call stickyAutoLineSeize Null 0 or 1 Null If call stickyAutoLineSeize is set to 1 this onHookDialing parameter has no effect The regular stickyAutoLineSeize behavior is followed If call stic
368. 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 for the SoundPoint IP SoundStation IP VVX Family 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 through 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
369. phone should include the mode 30 FMTP attribute in SDP offers If voice codecPref iLBC 13_33kbps S set and voice codecPref iLBC 15_2kbps S Null If voice codecPref iLBC 13_33kbps and voice codecPref iLBC 15_2kbps are both set but iLBC 13 33 kbps codec is set to a higher preference If set to 0 the phone should not include the mode 30 FTMP attribute in SDP offers even if iLBC 13 33 kbps codec is being advertised Refer to Codec Preferences lt codecPref gt on page A 42 volpProt SDP early answerOrOffer Oori Null If set to 1 an SDP offer or answer is generated in a provisional reliable response and PRACK request and response If set to 0 an SDP offer or answer is not generated Note An SDP offer or answer is not generated if the user reg x is configured for the Music On Hold Refer to Music on Hold lt musicOnHold gt on page A 20 Configuration Files Outbound Proxy lt outboundProxy gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation volpProt SIP outboundProxy address dotted deci Null IP address or host name and port of a SIP mal IP server to which the phone shall send all address or requests host name volpProt SIP outboundProxy port 0 to 65535 5060 volpProt SIP outboundProxy DNSnapitr or DNSnapt If set to Null or DNSnaptr transport TCPpreferre r If volpProt SIP outboundProxy address is a dor h
370. 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 Note Shared call appearances and bridged line appearances are two different signalling methods of implementing a feature whereby more than one phone can share the same line or registration These implementations are dependent on the SIP server The methods are mutually exclusive and you should confirm with the call server vendor which if any method is supported Configuration changes can be performed centrally at the provisioning server or locally Central provisioning 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 80 Specify line seize subscription period e For more information refer to Server lt server gt on page
371. pired Device Certificate Not Installed is displayed if the certificate is not available in flash memory or the flash memory location where the device certificate is to be stored is blank Device Certificate Invalid is displayed if the certificate is not valid if any of the fields listed above are not correct Mutual TLS Authentication Mutual Transport Layer Security TLS authentication is a process in which both entities in a communications link authenticate each other In a network environment the phone authenticates the server and vice versa In this way phone users can be assured that they are doing business exclusively with legitimate entities and servers can be certain that all would be users are attempting to gain access for legitimate purposes This feature requires that the phone being used has a Polycom factory installed device certificate Refer to the previous section Digital Certificates Prior to SIP 3 2 and in cases where the phones do not have factory installed device certificates the phone will authenticate to the server as part of the TLS authentication but the server cannot cryptographically authenticate the phone This is sometime referred to as Server Authentication or single sided Authentication Mutual TLS authentication is optional and is initiated by the server When the phone acts as a TLS client and the server is configured to require mutual TLS the server will request and then
372. ple of a configuration file and the resulting log file e og sched 1 name iog sched 1 level og sched 1 period og sched 1 startMode og sched 1 startTime log sched 1 startDay og sched 2 name og sched 2 level og sched 2 period log sched 2 startMode og sched 2 startTime og sched 2 startDa 6 amp amp showCpuLoad 4 16 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163033 slog 0522163048 slog 0522163048 slog 0522163048 slog 0522163048 slog 0522163048 slog 0522163048 slog 0522163048 slog 0522163048 slog 14 01 Running showCpuLoad 4 01 Cpu load is 6 0 and the average is 57 6 4 O1 H FRRERHRHRRRBRRRRRRRERRRRRRR RRR RR ER ER RR ES 4 O1 H HBRRHEHRRRARARRRRRR ERR RBBB RRERERERER ES 4 01 Running memShow 4 01 status bytes blocks avg block max block 4 OL 4 01 current 14 01 free 9410608 65 144778 9257824 14 01 alloc 11147888 31569 353 14 01 cumulative 14 01 alloc 18961376 58186 325 4 O1 H AHRHRHRHRAHRAHRHRRRRR RAHA RRR RRRR RR RR ER RR ES 4 O1 H HRRHHHRHREARRRRRRRERRRRRRRER RRR RR ERR ES 4 01 Running showCpuLoad 4 01 Cpu load is 6 0 and the average is 47 1 4 O1 H HHRRHHRRRAHRRHRRRRRRRRRRRRRRR RRR ER RR RRR ER
373. port To disable the PC Ethernet port refer to Disabling PC Ethernet Port on page C 26 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 toVoice Settings lt voice gt on page A 41 and Video Settings lt video gt on page A 61 Graphic Display Backgrounds You can set up a picture or design to be displayed on the background of the graphic display of all SoundPoint IP 450 550 560 650 and 670 and Polycom VVX 1500 phones Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note Note 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 For SoundPoint IP 450 550 560 650 and 670 phones There are a number of default backgrounds both solid color and p
374. provides the menu with separate soft keys for parked pickup 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 on older platforms On newer platforms use line key functions The table below shows only Line1 to Line6 functions For the SoundPoint IP 650 and 670 phones with attached Expansion Modules Line7 to Line48 functions are also supported Label Function Notes ACDAvailable ACDAvailableFromldle ACDLogin ACDLoginLogout ACDLogout ACDLoginLogout ACDUnavailable ACDAvailableFromldle Answer Answer Call screen only Applications Main Browser ArrowDown ArrowDown ArrowLeft ArrowLeft ArrowRight ArrowRight ArrowUp ArrowUp ASignIn serverACDSignIn ASignOut serverACDSignOut Avail serverACDAgentAvailable Bargeln BarglnShowAppearances Bargeln Call screen only BuddySitatus Buddy Status Callers Callers CallList Call Lists CallPark ParkEntry Call screen only CallPickup CallPickupEntry Call screen only Conference ConferenceCall Call screen only Delete Delete Administrator s Guide SoundPoint IP SoundStation IP VVX
375. ps to syslog severity as follows 0 gt SeverityDebug 7 1 gt SeverityDebug 7 2 gt Severitylnformational 6 3 gt Severitylnformational 6 4 gt SeverityError 3 5 gt SeverityCritical 2 6 gt SeverityEmergency 0 7 gt SeverityNotice 5 For more information refer to Syslog Menu on page 3 13 log render type 0 2 2 Refer to above table for timestamp type log render realtime Oor1 1 Set to 1 Note Polycom recommends that you do not change this value log render stdout Oor 1 1 Set to 1 Note Polycom recommends that you do not change this value log render file Oori 1 Set to 1 Note Polycom recommends that you do not change this value A 101 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 provisioning server and the local copy is erased log render file upload period positive 172800 Time in seconds between log file integer uploads to the provisioning 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 Oor1 1 If set to 1 use append mode when uploading log files to server Note HTTP and TFTP don t support append mode unles
376. quired to modify this feature Central provisioning 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 104 Configuration file Change the encryption key lt device gt cfg For more information refer to Flash Parameter Configuration on page A 151 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 Administrator s Guide SoundPoint IP SoundStation IP VVX 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 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 provisioning 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
377. r s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 Oor1 1 If set to 1 reject RTP packets arriving from sent from a non negotiated through SDP IP address tcplpApp port rtp filterByPort Oor1 0 If set to 1 reject RTP packets arriving from sent from a non negotiated through SDP port tcplpApp port rtp forceSend Null Null When non Null send all RTP 1024 65534 packets to and expect all RTP packets to arrive on the specified port Note both tcplpApp port rtp filterBylp and tcplpApp port rtp filterByPort must be enabled for this to work tcplpApp port rtp mediaPortRangeStart Null even Null If set to Null the value 2222 will integer from be used for the first allocated 1024 65534 RTP port otherwise the 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
378. r improvements have been made and Polycom recommends that existing configuration files be reviewed and updated Customers replacing legacy telephony PBX or key system would like to get equivalent functionality from their new VoIP telephony system The enhanced feature key capability is designed to allow system administrators to program the speed dials and soft keys 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 32x 33x 430 450 550 560 650 and 670 desktop phones and Polycom VVX 1500 business media phones The enhanced feature key functionality is implemented using Star Code sequences and SIP messaging Configuring Your System The enhanced feature key macro language was designed to follow current configuration file standards and to be extensible It is described in more detail in Enhanced Feature Key Definition Language 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 The enhanced feature key capability can be used to provide a customized interactive user interface by mapping functions from speed dial keys soft keys and re mapped hard function keys This section provides detailed information on e Enhanced Feature Key Definition Language e Macro Definition e Configur
379. r takes precedence If set to 1 forces phone to wait for 200 OK on registration x when receiving a TRYING notify If set to 0 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 configuration file reg x tcpFastFailover Oor1 Null If set to 1 failover occurs based on the values of reg x server y retryMaxCount volpProt server x retryTimeOut If set to 0 or Null this is old behavior If this parameter is Null volpProt SIP tcpFastFailover Is checked If both parameters are set this parameter takes precedence A 132 Configuration Files Permitted Attribute Values Default Interpretation reg x server y address dotted decimal IP Null Optional IP address or host name port address or host transport registration period fail over name parameters and 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 yo te z tOr fault toierance If specified these servers may
380. r 1 default 2 3 4 This value is H263 1998 90000 format parameter ANNEXN used to signal Polycom VVX 1500 receiving capability in the SDP Camera Controls lt camera gt These settings control the performance of the camera These configuration attributes are defined as follows Attribute Permitted Values Default Interpretation video camera flickerAvoidance Oto2 Null Set flicker avoidance If set to 0 or Null flicker avoidance is automatic If set to 1 50hz AC power frequency flicker avoidance Europe Asia If set to 2 60hz AC power frequency flicker avoidance North America video camera frameRate 5 to 30 frames per second Null Set target frame rate Values indicate a fixed frame rate from 5 least smooth to 30 most smooth If set to Null the value 25 is used video camera brightness Oto6 Null Set brightness level The value range is from 0 Dimmest to 6 Brightest If set to Null the value 3 is used video camera saturation 0to6 Null Set saturation level The value range is from 0 Lowest to 6 Highest If set to Null the value 3 is used 66 Configuration Files Permitted Attribute Values Default Interpretation video camera contrast 0 to 4 Null Set contrast level The value range is from 0 No contrast increase to 3 Most contrast increase and 4 Noise reduction contrast If set
381. r System Configuration File Examples For more examples refer to Technical Bulletin 42250 Using Enhanced Feature Keys and Configurable Soft Keys on SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at 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 configuration 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 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 l action S1 Tinvite softkey 1l use idle 1 4 Reboot the phone If you press the soft key ChainAct the 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Fa
382. rage for phone resident copy of the directory to allow for larger size dir local nonVolatile maxSize 8meg 1 to 100 100 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 dir local readonly Oor1 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 Oor1 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 Configuration Files 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 or corporate directory For example FQDN host domain com dir corp port 0 Nu
383. rate directory entries up one TouchVoiceMail Oori 0 If set to 1 the voice mail summary display is bypassed and voice mail is dialed directly if configured Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation up welcomeSoundEnabled Oor1 1 If set to 1 play welcome sound effect after a reboot up welcomeSoundOnWarmBootE 0 or 1 0 If set to 1 play welcome sound effect on warm nabled and cold boots If set to 0 only a cold reboot will trigger the welcome sound effect up localClockEnabled Oori 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 ae 3 The default value is low high Note If idleintensity 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
384. rate with internally saved parameters This is useful for occasions when the provisioning 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 VVX phone refer to Quick Tip 44011 Register Standalone SoundPoint IP SoundStation IP and Polycom VVX 1500 Phones at http www polycom com usa en support voice soundpoint_ip VolP_Technical_Bulle tins_pub html Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 SIP application uses the network to query the provisioning 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 MACaddress 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 t
385. ration attribute is defined as follows Permitted Attribute Values Interpretation res quotas x name 1 tone The name of the sub application for which the particular quota 2 bitmap will apply 3 font tone relates to all downloaded tones and sound effects 5 background bitmap relates to all downloaded bitmaps font relates to all downloaded fonts background relates to all downloaded backgrounds res quotas x value positive integer When a particular resource one of category font bitmap or font is downloaded to the phone a quota equal to this value 1024 bytes of compound data size is applied for that category If downloading a resource would exceed the quota for that category the resource will not be downloaded and a predefined default will be used instead For res quotas x value the default is 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 430 450 550 560 601 650 and 670 the SoundStation IP 6000 an
386. ration line type private or shared and the shared line third party name using the SIP Configuration menu Either the Web Server or the provisioning 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 430 450 550 560 600 601 650 and 670 phones Other SoundPoint IP phone models may be monitored but cannot be configured to monitor other phones Some aspects of this feature are dependent on the SIP server signaling You should consult your SIP server partner or Polycom Channel partner for information as needed 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 was updated for the following e Visual and audible indication when a remote line is in an alerting state e Display of the caller ID of calls on remotely monitored lines Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note we POLYCOM Note e Single button Directed Call Pickup on a remote line In the SIP 3 2 release the BLF feature is updated for t
387. re 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 be performed centrally at the provisioning server Central Configuration file Specify the user s preferences for treble and bass provisioning sip cfg e For more information refer to User Preferences lt up gt on page server A 29 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Setting Up Video Features Video Transmission 4 86 The Polycom VVX 1500 phone supports transmission and reception of high quality video images The video is compatible with RFC 3984 RTP Payload Format for H 264 Video RFC 4629 RTP Payload Format for ITU T Rec H 263 Video and RFC 5168 XML Schema for Media Control This section provides information for making configuration changes for the following video related features e Video Transmission e Video Codecs By default at the start of a video call the Polycom VVX 1500 phone transmits an RTP encapsulated video stream with images captured from the local camera Users can stop and start video transmission by pressing the Video key and then selecting the appropriate soft k
388. re 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 configured to use a DNS server For more information refer to DNS Cache lt dns gt on page A 120 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
389. reFilter enable 0 voice txEq hs IP_430 preFilter enable 0 voice txEq hs IP_450 preFilter enable 0 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute Default voice txEq hs IP_650 preFilter enable 1 voice txEq hs VVX_1500 preFilter enable 0 voice txEq hs IP_330 postFilter enable voice txEq hs IP_430 postFilter enable voice txEq hs IP_450 postFilter enable voice txEq hs IP_650 postFilter enable voice txEq hs VVX_1500 postFilter enable voice txEq hd IP_330 preFilter enable voice txEq hd IP_430 preFilter enable voice txEq hd IP_450 preFilter enable ojoj o voice txEq hd IP_650 preFilter enable voice txEq hd VVX_1500 preFilter enable voice txEq hd IP_330 postFilter enable voice txEq hd IP_430 postFilter enable voice txEq hd IP_450 postFilter enable voice txEq hd IP_650 postFilter enable voice txEq hd VVX_1500 postFilter enable voice txEq hf IP_330 preFilter enable voice txEq hf IP_430 preFilter enable voice txEq hf IP_450 preFilter enable o o 1ao o o o o o oa voice txEq hf IP_650 preFilter enable voice txEq hf IP_6000 preFilter enable voice txEq hf IP_7000 preFilter enable voice txEq hf VVX_1500 preFilter enable ojojo voice txEq hf IP_330 postFilter enable voice txEq hf IP_430 postFilter enable voice txEq hf IP_450 postFilter enable
390. reboots When the reboot is complete the PC Ethernet port is disabled s Configuration Using the Web Interface You can make changes to the configuration files through the web interface to the phone Miscellaneous Administrative Tasks To configure your phone through the web interface gt gt Using your chosen browser do the following a To get your phone s IP address press the Menu key and then selecting Status gt Platform gt Phone Scroll down to see the IP address b Enter your phone s IP address as the browser address A web page similar to the one shown below appears x0 D 80 SoundPoint IP Configuration Home General Network SIP Lines Welcome to the SoundPoint IP Configuration Utility Select an area to configure from the menu above c Select SIP from the menu tab You will be prompted for the SIP username and password A web page similar to the one shown below appears SIP Configuration Parameters Servers Local Settings Outbound Proxy Address Port 5060 Transport DNSnaptr Server 1 Address Port 5060 Transport DNSnaptr Expires Register d Make the desired configuration changes e Scroll down to the bottom of the Servers section Administrator s Guide SoundPoint IP SoundStation IP VVX f Select the Submit button A web page similar to the one shown below appears POLYCOM Soun
391. 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Permitted Attribute Values Default Interpretation bg medRes gray bm x name any string Graphic files for display on the phone and Expansion Module and also the brightness bg medRes gray bm x em any string adjustment to the graphic name For x 1 bg medRes gray bm x adj integer e name is Leaf256x116 jpg adjustment is 0 For x 2 name is Sailboat256x116 jpg adjustment is 3 For x 3 name is Beach256x116 jpg adjustment is 0 For x 4 name is Palm256x116 jpg adjustment is 3 For x 5 name is Jellyfish256x116 jpg adjustment is 2 For x 6 name is Mountain256x116 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 color selection x y any string The label color for soft keys and line key labels modify associated with the defined colored backgrounds These values can be modified locally by the user The format is rgoHILo lt parameter list g
392. red 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 2008 The OpenSSL Project All rights reserved Redistribution and use in source and binary forms with or without modification are permitted provided that the following conditions are met 1 Redistributions of source code must retain the above copyright notice this list of conditions and the following disclaimer 2 Redistributions in binary form must reproduce the above copyright notice this list of conditions and the following disclaimer in the documentation and or other materials provided with the distribution 3 All advertising materials mentioning features or use of this software must display the following acknowledgment This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit http www openssl org 4 The names OpenSSL Toolkit and OpenSSL Project must not be used to Third Party Software
393. rences lt codecPref gt Permitted Attribute Values Default Interpretation voice codecPref G711Mu Null 1 3 1 Specifies the codec preferences for SoundPoint IP 32x 33x and 430 voice codecPref G711A 2 platforms voice codecPref G729AB 3 1 highest 3 lowest voice codecPref iLBC 13_33kbps Null Null do not use voice codecPref iLBC 15_2kbps Null Give each codes a Unique priority Hits will dictate the order used in SDP negotiations Note iLBC is not supported on the SoundPoint IP 430 voice codecPref IP_650 G711Mu Null 1 4 2 Specifies the codec preferences for the SoundPoint IP 450 550 560 650 and voice codecPref IP_650 G711A 3 670 platform Interpretation as above voice codecPref IP_650 G729AB 4 voice codecPref IP_650 G722 1 voice codecPref iLBC IP_650 Null 13 33kbps voice codecPref iLBC IP_650 15_ 2kbps Null Configuration Files Attribute voice codecPref IP_6000 G711Mu voice codecPref IP_6000 G711A voice codecPref IP_6000 G722 voice codecPref IP_6000 G7221 16kbps voice codecPref IP_6000 G7221 24kbps voice codecPref IP_6000 G7221 32kbps voice codecPref IP_6000 G729AB voice codecPref IP_6000 G7221C 24kbps voice codecPref IP_6000 G7221C 32kbps voice codecPref IP_6000 G7221C 48kbps voice codecPref IP_6000 Siren14 24kbps voice codecPref IP_6000 Siren14 32kbps voice codecPr
394. ries 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 encrypted using the old key refer to Changing the Key on the Phone on page C 5 At a later date new phones from the factory will have a key pre loaded in them This key will be changed at regular intervals to enhance security Note Note Miscellaneous Administrative Tasks 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 m
395. rks Use larger values to minimize packet loss on networks with large jitter 3000 ms voice audioPrrofile x jitterBufferMax gt The largest jitter buffer depth to be supported jitterBufferMin in milliseconds Jitter above this size will multiple of 10 always cause lost packets This parameter lt 300 for IP should be set to the smallest possible value 32x 33x 430 that will support the expected network jitter 550 600 and 650 voice audioProfile x payloadType 96 127 The codec payload encoding in the dynamic default range to be used in SDP offers Volume Persistence lt volume gt Configuration Files The user s selection of the receive volume during a call can be remembered between calls This can be configured per termination handset headset and hands free chassis In some countries regulations exist which dictate that receive volume should be reset to nominal at the start of each call on handset and headset Permitted Attribute Values Default Interpretation voice volume persist handset Oor1 0 If set to 1 the receive volume will be i thead Toa R remembered between calls R Z If set to 0 the receive volume will be reset voice volume persist handsfree Oor1 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
396. rl dialing url dialing controls whether URL name dialing is available from a private line it is never available from a shared line 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 call park call park is the call park and park retrieve features Configuration Files Attribute Permitted Values Interpretation feature x name continued group call pickup group call pickup is the group call pickup feature directed call pickup directed call pickup is the directed call pickup feature last call return last call return is the last call return feature acd login logout acd login logout is the ACD login logout feature acd agent available acd agent available is the ACD agent available unavailable feature nway conference nway conference is the conference managing feature 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 Ifsetto 1 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 i
397. rmed centrally at the provisioning server Central provisioning server Configuration file Turn this feature on or off sip cfg e For more information refer to Feature lt feature gt on page A 110 Call Forward 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 7 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 disables local Call Forward and DND features unless configured otherwise Server based call forwarding will behave the same as per SIP 2 1 feature with the following exception Cen
398. rovide 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 Productivity Suite 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 VVX Error Messages If a problem 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 booting Some of these errors are fatal meaning that the phone will not able to boot until this issue has been resolved and some are recoverable meaning the phone will continue booting after the error but the configuration of the phone may not be what you were expecting BootROM Error Messages Most of these errors are also logged on the phone s boot log however if
399. rovisioning 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 For information on configuring Polycom VoIP phones please refer to the gt lt Configuration File Management white paper available from gt lt http www polycom com common documents whitepapers configuration_file _management_on_soundpoint_ip_phones pdf gt lt SRCSfile 000000000000 cfg v Revision 1 21 gt lt APPLICATION APP_FILE_PATH sip MACADDRESS 1d CONF IG_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 provisioning 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 For information on configuring Polycom VoIP phones please refer to the gt lt Configuration File Management white paper available from gt lt http www polycom com common documents whitepapers configuration_file _management_on_soundpoint_ip_phones pdf gt lt SR
400. rporate directories that have server side sorting Polycom recommends that you consult your LDAP Administrator when making any configuration changes for this feature The corporate directory can be browsed or searched Entries retrieved from the LDAP server can be saved to the local contact directory on the phone Phone calls can be placed based on the phone number contained in the LDAP entry The corporate directory interface is read only so that editing or deleting existing directory entries as well as adding new directory entries from the phone is not be possible There is no matching of first and last names in the corporate directory to incoming calls caller identification display and in the call lists 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 detailed examples for all currently supported LDAP directories refer to Technical Bulletin 41137 Best Practices When Using Corporate Directory on SoundPoint IP SoundStation IP VVX Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuration changes can be performed centrally at the provisioning server or locally Central provisioning server sip cfg Configuration file Sp
401. rpretation 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 voice 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
402. s e Central Report Collector lt collector gt e Alert Reports lt alert gt e Server lt server gt e RTCP XR lt rtcpxr gt Central Report Collector lt collector gt This configuration attribute is defined as follows Permitted Attribute Values Default Interpretation voice qualityMonitoring collector enable 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 qualityMonitoring collector period 5 to 20 20 The time interval between successive periodic quality reports A 58 Alert Reports lt alert gt Configuration Files This configuration attribute is defined as follows Aitribute Permitted Values Default Inte
403. s Protection against recursive macro calls exists the enhanced feature keys fails once 50 macro substitutions is reached Examples 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 name enhanced feature keys feature 18 enabled 1 gt Action String Example The action string SChangup 444 SPIN4 TinviteS Cwaitconnect P2N3 Cpause2 Tdt mf Changup is executed as follows 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 Once connected there is a 2 second pause and then the string 567 is sent using DTMF dialing on the active call The active call is disconnected Speed Dial Example Configuring Your System 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 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 efk efklist 2 e
404. s gt on page A 97 for this indicator ind led x physNum This maps the logical index to a specific physical LED Graphic Icons lt gi gt lt IP_330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt In the following table x IP_330 IP_400 IP_500 IP_600 IP_4000 or IP_7000 y is the graphic icon number Note that IP_330 parameters affect SoundPoint IP 32x 33x phones IP_400 parameters affect SoundPoint IP 430 phones IP_450 parameters affect SoundPoint IP 450 phones and IP_600 parameters affect SoundPoint IP 550 560 600 601 650 and 670 phones IP_4000 parameters affect SoundStation IP 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 97 for this indicator ind gi x y physX IP 330 0 101 IP 400 0 122 IP 450 0 238 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 A 98 Configuration Files Attribute ind gi x y physY Permitted Values IP 330 0 19 IP 400 0 45 IP 450 0 89 IP 600 0 159 IP 4000
405. s Guide for the SoundPoint IP SoundStation IP VVX Phones Codec Profiles lt audioProfile gt The following profile attributes can be adjusted for each of the five supported codecs In the table x G711Mu G711A G722 G7221 G7221C G729AB Lin16 Siren14 Siren22 and iLBC 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 G719 G7221 G7221C Siren14 Siren22 and iLBC 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 before 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 netwo
406. s always disabled on the SoundPoint IP 32x 33x and 430 phone The manage conference feature is always enabled on the SoundStation IP 7000 and the Polycom VVX 1500 phone call recording call recording is the call recording and playback feature 3 enhanced feature keys enhanced feature keys is the enhanced feature keys feature corporate directory corporate directory is the corporate directory feature picture frame picture frame is the digital picture frame feature Note feature 20 name picture frame is only supported on the Polycom VVX 1500 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 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 However these feature are included on the Polycom VVX 1500 A 111 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 Bu
407. s 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 Modifying Phone s Configuration Using the Web Interface Capturing Phone s Current Screen LLDP and Supported TLVs 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 Administrator s Guide SoundPoint IP SoundStation IP VVX e American Express Global CA e BelSign Object Publishing CA e BelSign Secure Server CA 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 Equifax Secure eBusiness CA 1 e Equifax Secure eBusiness CA 2 e Equifax Secure Global eBusiness CA 1 e GeoTrust Primary Certi
408. s do not have the capability to determine their physical location automatically or provision to a statically configured location Because of these limitations the SoundPoint IP SoundStation IP VVX phones will not transmit Location Identification TLV in the LLDP frame However the location information from the switch is decoded and displayed on the phone s menu For more information on configuration parameters refer to Flash Parameter Configuration on page A 151 This is the basic TLV format TLV Type 7 bits 0 6 TLV Length 9 bits TLV Information 0 511 7 15 bytes The following is a list of supported TLVs Org Version Type Length Unique 7 bits 9 bits Type Code Sub No Name 0 6 7 15 Length 3 bytes Type Information 1 Chassis ld 1 6 0x0206 5 IP address of phone 4 bytes Note 0 0 0 0 is sent until the phone has a valid IP address 2 Port Id 2 7 0x0407 3 MAC address of phone 6 bytes 3 TTL 3 2 0x0602 TTL value is 120 0 sec 4 Port 4 1 0x0801 Port description 1 description 5 System 5 min len gt Refer to System Names on name 0 max len page C 36 lt 255 Administrator s Guide SoundPoint IP SoundStation IP VVX No Name Type 7 bits 0 6 Length 9 bits 7 15 Type Length Org Unique Code 3 bytes Version Sub Type Information System description min len gt 0
409. s 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 96 127 127 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 cadence A chord set can contain up to four frequency components generated simultaneously each with its own level There are three blocks of chord sets e callProg used for call progress sound effect patterns e ringer e misc miscellaneous All three blocks use the same chord set specification format Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones 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 pos
410. s menu has a series of real time graphs for CPU network and memory utilization that can be helpful in diagnosing performance issues Sound Point IP and SoundStation IP phones will log various events to files stored in the flash file system and will periodically upload these log files to the provisioning server The files are stored in the phone s home directory or a user configurable directory You can also configure a phone to send log messages to a syslog server 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 provisioning server with the MAC address of the phone prepended to the file name For example 0004 200360b boot log and 0004f200360b app log are the files associated with MAC address 00f4f200360b The bootROM log file is uploaded to the provisioning server after every reboot The application log file is uploaded periodically or when the local copy reaches a predetermined size Refer to Basic Logging lt level gt lt change gt and lt render gt on page A 101 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 unless they are empty The amount of logging that the phone performs can be tuned for the application to provide more or less de
411. s that are needed for the operation of the phone 6 Examines the master configuration file for the name of the application file and then looks for this file on the provisioning server If the copy on the provisioning server is different than the one stored in flash memory or there is no file stored in flash memory the application file is downloaded 7 Extracts the application from flash memory Installs the application into RAM then uploads 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 manages everything to do with the phone s operation The application is a single file binary image and contains a digital signature to prevent tampering or loading rogue software images There is a new image file in each release of software The application performs the following tasks in order 1 Downloads system per phone configuration and resource files These files are called sip cfg and phonel cfg by default You can customize the filenames Controls all aspects of the phone 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 Current build archives have both split and combined images
412. s the server is set up for this log render file upload append sizeLimit positive 512 Maximum log file size on integer provisioning server in Kbytes log render file upload append limitMode delete stop delete Behavior when server log file has reached its limit delete delete file and start over stop stop appending to file Scheduled Logging Parameters lt sched gt The phone can be configured to schedule certain advanced logging tasks ona periodic basis These attributes should be set in consultation with Polycom Technical Support Each scheduled log task is controlled by a unique attribute set starting with log sched x where x identifies the task 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 A 102 Configuration Files 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 w
413. s to appear in the Proxy Require header If Null no Proxy Require will be sent reg x serverFeatureControl cf Oor1 If set to 1 server based call forwarding is enabled The call server has control of call forwarding If set to 0 server based call forwarding is not enabled This is the old behavior If reg x serverFeatureControl cf is not Null this attribute overrides the global server based call forwarding flag in the sip cfg configuration file reg x serverFeatureControl dnd Oor1 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 A 131 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute reg x auth optimized InFailover reg x strictLineSeize Permitted Values Oori Oori 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 paramete
414. 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 other 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 inter
415. 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 volpProt SIP Ics Oori 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 Oor1 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 sendCompactHdrs Oor1 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Attribute
416. set sidetone adjust IP_650 3 Acoustic Echo Cancellation lt aec gt Configuration Files These settings control the performance of the speakerphone acoustic echo canceller Polycom recommends that you do not change these values POLYCOM Attribute Default voice aec hs enable 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 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Acoustic Echo Suppression lt aes gt Acoustic Echo Suppression AES provides non linear processing of the microphone signal to remove any residual echo remaining after linear AEC processing Because AES depends on AEC AES should only be enabled when AEC is also enabled N
417. set to 0 call transfer blindPreferred 0 1 Null If set to 1 the blind transfer is the default mode The Normal soft key is available to switch to a consulatative transfer If set to O or Null the consultative transfer is the default mode The Blind soft key is available to switch to a blind transfer Note This parameter is supported on the SoundPoint IP 330 320 only call directedCallPickupString star code 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 call directedCallPickupMethod native or Null legacy The method the phone will use to perform a directed call pick up of a BLF resource s inbound ringing call native indicates the phone will use a native protocol method in this case SIP INVITE with the Replaces header 4 legacy indicates the phone will use the method specified in call directedCallPickupString call parkedCallRetrieveMethod native or Null legacy The method the phone will use to retrieve a BLF resource s call which has dialog state confirmed native indicates the phone will use a native protocol method in this case SIP INVITE with the Replaces header 4 legacy indicates the phone will use the method specified in call parkedCallRetrieveString call parkedCallRetrieveString star code N
418. so it up to the administrator which model to support Using split files saves a lot of internal network traffic during reboots and updates Configuration Warning Note Overview The SoundPoint IP SoundStation IP VVX phones can be configured automatically through files stored on a central provisioning server manually through the phone s local UI or web interface or a combination of the automatic and manual methods The recommended method for configuring phones is automatically through a central provisioning server but if one is not available the manual method will allow changes to most of the key settings 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 You can make changes to the configuration files through the web interface to the phone Using your chosen browser enter the phone s IP address as the browser address For more information refer to Modifying Phone s Configuration Using the Web Interface on page C 26 Changes made through the web interface are written to the override file highest priority These changes remain active and will take precedence over the configuration files stored on the provisioning server until Reset Local Config is performed
419. ss deleted through the Reset Local Config menu selection and the lt Ethernet address gt phone cfg is removed from the provisioning server Distinctive Call Waiting The voIpProt SIP alertInfo x value and volpProt SIP alertiInfo x class fields can be used to map calls to distinct call waiting types currently limited to two styles This feature requires server support Configuration changes can be performed centrally at the provisioning server Central Configuration file Specify the mapping of Alert Info strings to call waiting types provisioning sip cfg e For more information refer to Alert Information lt alertInfo gt on server page A 18 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 22 Server based DND is active if the feature is enabled on both the phone and the server and the phone is registered The server based DND feature is applicable for all registrations on the phone no per registration mode and it disables local Call Forward and DND features unless configured otherwise Configuring Your System Server based DND will behave the s
420. ssible 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 call server for example shared lines presence and Message Waiting Indicator Polycom phones support configuration of multiple servers per SIP registration for this purpose In some cases a combination of the two may be deployed Your SIP server provider should be consulted for recommended methods of configuring phones and servers for fail over configuration Prior to SIP 2 1 the reg x server y parameters refer to Registration lt reg gt on page A 128 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
421. stem level parameters letting you override that parameter for a given user For example it might be desirable to set the default CODEC for a remote user differently than for all the users who reside in the head office By adding the CODEC settings to a particular 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 The following figure shows one possible layout of the central provisioning method Boot Server config overrides event log contact directory files g004F2002779 phone cfg g004f2002979 directory cfg master config file application binary config files dictionary files user interface resource files license files SoundPoint IP SIP Local User Interface Local MAC 00 04 f2 00 29 99 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 provisioning 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family This can create
422. stomizable 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 Note 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 provisioning server or the Internet for downloading at boot time Configuration changes can be performed centrally at the provisioning server or locally Central Configuration File Specify patterns used for sound effects and the individual tones or provisioning sip cfg sampled audio files used within them server e For more information refer to Sampled Audio for Sound Effects lt saf gt on page A 34 or Sound Effects lt se gt on page A 35 Local Web Server Specify sampled audio wave files to replace the built in defaults if enabled Navigate to http lt phonelPAddress gt coreConf htm sa Changes are saved to local flash and backed up to lt Ethernet address gt phone cfg on the provisioning server Changes will permanentl
423. string containing Null Contact to call when retrieving digits the user part of a SIP messages for this registration URL or a string that constitutes a valid SIP URL 6416 or 6416 polycom com Network Address Translation lt nat gt 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 A 144 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 signaling overriding the value set for volpProt local Port 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 74 nat keepalive interval 0 to 3600 Nu
424. support from a SIP server With many SIP servers directed call pick up is implemented using a particular star code sequence With some SIP servers specific network signaling is used to implement this feature Configuration changes can be performed centrally at the provisioning server server Central Configuration file Turn this feature on or off provisioning sip cfg For more information refer to Feature lt feature gt on page A 110 Determine the type of directed call pickup e For more information refer to Call Handling Configuration lt call gt on page A 76 Determine the type of SIP header to include For more information refer to Protocol lt volpProt gt on page A 7 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 With many SIP servers group call pick up is implemented using a particular star code sequence With some SIP servers specific network signaling is used to implement this feature Configuration changes can be performed centrally at the provisioning server provisioning sip ctg server Central Configuration file Turn this feature on or off For more information refer to Feature lt feature gt on page A 110 Call Park Retrieve An active call can be parked and the parked call can be retrieved by another phone This
425. t For example rbgHiLo 51 255 68 255 0 119 is the default button color associated with the built in background button gray selection x y any string The label color for soft keys and line key labels modify associated with the defined gray 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 Bitmaps lt bitmap gt Configuration Files The bitmaps used by each phone model are defined in this section Platform lt IP 330 gt lt IP_400 gt lt IP_450 gt lt IP_600 gt lt IP_4000 gt and lt IP_7000 gt In the following table x IP_330 IP_400 IP_450 IP_600 IP_4000 or IP_7000 and y is the bitmap number Note that IP_330 parameters affect SoundPoint IP 32x 33x phones IP_400 parameters affect SoundPoint IP 430 phones IP_450 parameters affect SoundPoint IP 450 phones IP_600 parameters affect SoundPoint IP 550 560 600 601 and 650 and 670 phones IP_4000 parameters affect SoundStation IP 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 provisioning server its name must Be different from any name already in use in sip cfg
426. t 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 17 upgrading 3 21 SIP basic protocols header support B 4 SIP header diversion A 13 warning A 14 SIP headers warnings 4 76 SIP See also Session Initiation Protocol sip cfg A 5 SIP lt SIP gt A 11 SIP B automatic call distribution 4 59 soft keys lt softkey gt A 123 sound effects lt se gt A 35 SoundPoint IP Phones applications 4 32 configuring phones locally 4 94 digital certificates 4 91 features list of 1 6 supported languages 4 29 Index 6 SoundPoint IP SoundStation IP VVX phones features overview 2 8 introduction 1 1 network 2 2 new features overview 2 14 SoundPoint IP 32x 33x switching text entry mode 3 8 SoundPoint IP 650 playback 4 37 A 111 recording 4 37 A 111 SoundPoint IP 670 playback 4 37 A 111 recording 4 37 A 111 SoundStation IP Phones applications 4 32 configuring phones locally 4 94 features list of 1 6 supported languages 4 29 SoundStation IP 7000 treble bass controls 4 85 speed dial 4 15 SRTP See also secure real time transport protocol static DNS cache 4 72 status menu 5 4 supported LDAP directories 4 35 T text entry mode switching 3 8 time and date display 4 15 time synchronization A 71 TLS See also transport layer security TLVs See also type length values transmit equalization lt txEq gt A 55 transport layer se
427. t IP 330 320 430 450 550 560 650 and 670 desktop phones The Feature Synchronized ACD feature is distinct from the existing SIP B Automatic Call Distribution functionality which was added in SIP 1 6 For details on how to configure SoundPoint IP SoundStation IP and VVX phones for Feature Synchronized ACD refer to Technical Bulletin 34787 Using Feature Synchronized Automatic Call Distribution with SoundPoint IP and Polycom VVX 1500 Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuration changes can be performed centrally at the boot server boot server sip cfg Central Configuration file Enable or disable Feature Synchronized ACD e For more information refer to SIP lt SIP gt on page A 11 Turn this feature on or off For more information refer to Feature lt feature gt on page A 110 Configuration file Set the registration to be used for Feature Synchronized ACD and the phonet cfg users sign in state For more information refer to Automatic Call Distribution lt acd gt on page A 150 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 between the phone and the server fails Note Warning Configuring Your System Two types of redundancy are po
428. t gt A 40 ringer patterns A 38 roaming buddies lt roaming_buddies gt A 148 roaming privacy lt roaming_provacy gt A 148 routing lt routing gt A 141 routing server lt server gt A 25 A 142 RTP lt RTP gt A 68 A 69 A 74 S sampled audio files A 35 sampled audio for sound effects lt saf gt A 34 SCA See also shared call appearances scheduled logging parameters A 102 screen capture phone A 31 SDP lt SDP gt A 16 secure real time transport protocol 4 89 security lt sec gt A 103 server menu 3 10 server redundancy 4 60 server based call forwarding See also call forwarding server based DND See also do not disturb Services key See also Applications key Index 5 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Session Initiation Protocol setting up advanced features 4 25 audio features 4 78 basic features 4 1 boot server 3 14 network 3 2 security features 4 88 shared call appearance signaling B 10 shared call appearances shared calls lt shared gt A 80 shared lines barge in 4 53 A 130 SIP 1xx Responses Provisional B 6 2xx Responses Success B 7 3xx Responses Redirection B 7 4xx Responses Request Failure B 7 5xx Responses Server Failure B 8 6xx Responses Global Failure B 9 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 suppor
429. t 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 provisioning 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 Provisioning Server on page 3 17 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 The Server User and Server Password parameters should be changed from the default values Note that for insecure protocols the user chosen should have very few privileges on the server Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Ethernet Menu The following Ethernet configuration parameters can be modified on the Ethernet menu Name Possible Values Description LLDP Enabled Disabled If enabled the phone will use the LLDP protocol to communicate with the network switch for certain network parameters M
430. t 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 T port IE D or SuN 65535 If volpProt server x address ISa hostname and volpProt server x transport is setto DNSnaptr do NAPTR then SRV lookups If volpProt server x transport is set to TCPpreferred or UDPOnly then use 5060 and don t advertise the port number in signalling If volpProt server x address is an IP address there is no DNS lookup and 5060 is used for the port but it is not advertised in signaling If port is 1 to 65535 This value is used and it is advertised in signaling Note If the reg x server y address parameter in Registration lt reg gt on page A 128 is non Null all of the reg x server y xxx parameters will override the volpProt server parameters Configuration Files 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
431. t ramdisk gt 0 00 ccc ce cee A 109 Request lt request gt 2 0 0 eee cnet A 109 Feature lt feature gt 2 0 0 ccc ccc ene teens A 110 Resource lt res gt 2 6 nnne erennere errre A 112 Microbrowser lt mb gt 1 0 0 0000s ccc e cee cnet eens A 113 Applications lt apps gt 00 e cece A 117 Peer Networking lt pnet gt 00 0 000008 A 120 DNS Cache lt dns gt u n1i154540eeneeatnasatneiaeadwaeeawae A 120 Soft Keys lt softkey gt 2 0 0 0 cee eee ER A 123 LCD Power Saving lt powerSaving gt susrss rnrn A 126 Per Phone Configuration 0 0 0 eee eee eee eee A 127 Registration lt reg gt 00 A 128 Calls lt call gt eersten ea Ae R hn ee BAe dh aks eaten A 133 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Diversion lt divert gt 2 0 0 c ccc cece eee n eee A 136 Dial Plan lt dialplan gt 0 0 ee eee A 139 Messaging lt msg gt 000 A 143 Network Address Translation lt nat gt 0000 eee eee A 144 Attendant lt attendant gt 0 c ccc cece tenes A 145 Roaming Buddies lt roaming_buddies gt A 148 Roaming Privacy lt roaming_privacy gt 008 A 148 User Preferences lt up gt 0 0c e eee cece eens A 149 Automatic Call Distribution lt acd gt 0 000 eee A 150 Flash Parameter Configuration
432. t 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 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
433. t specialEvent gt on page A 19 The phones can be configured to periodically poll the provisioning 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 108 Supporting SoundPoint IP 300 301 500 501 600 and 601 and SoundStation IP 4000 Phones With enhancements available since BootROM 4 0 0 and SIP 2 1 2 you can modify the 000000000000 cfg or lt MACaddress 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 3 2 0 or later software distributions contain only the new distribution files for the new release You must rename the sip ld sip cfg and phonel cfg from a previous 2 1 x distribution that is compatible with SoundPoint IP 300 and 500 phones or a previous 3 1 y distribution that is compatible with SoundPoint IP 301 501 600 and 601 SoundStation IP 4000 phones The following procedure must be used for upgrading to SIP 3 2 0 or later for installations that have SoundPoint IP 300 301 500 501 600 601 and SoundStation IP 4000 phones deployed It is also recommended that this same approach be followed even if these phones are not part of the deployment as it will simplify management of phone systems with future software releases To upgrade your SIP application 1
434. tack 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 ea The default value is Auto 3 j 100HD 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 pti The default value is Auto 3 5 100HD HD means half duplex 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 applies to all phones supported by SIP 3 2 The PC Port Mode parameters are only available on phones with a second Ethernet port Only the SoundPoint IP 560 and 670 and Polycom VVX 1500 phones supports the LAN Port Mode and PC Port Mode setting of 1000FD The 1000BT LAN Clock and 1000BT PC Clock parameters are only available on SoundPoint IP 560 and 670 phones Syslog Menu Syslog is a standard for forwarding log messages in an IP network The term syslog is often used for both the actual syslog protocol as well as the application or library sending syslog messages The syslog protocol is a very simplistic protocol the syslog sender sends a small textual message less than 1024 bytes to the syslog receiver The receiver is commonly called syslogd syslog daemon
435. tail on specific components of the phone s software For example if you are troubleshooting a SIP signaling issue you are not likely interested in DSP events Logging levels are adjusted in the configuration files or via the web interface You should not modify the default logging levels unless directed to by Polycom Technical Support Inappropriate logging levels can cause performance issues on the phone In addition to logging events the phone can be configured to automatically execute command line instructions at specified intervals that output run time information such as memory utilization task status or network buffer contents to the log file These techniques should only be used in consultation with Polycom Technical Support Application Logging Options Each of the components of the application software is capable of logging events of different severity This allows you to capture lower severity events in one part of the application while still only getting high severity event for other components Administrator s Guide SoundPoint IP SoundStation IP VVX 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 types that can be adjusted to assist with the investigation of different problems When testing is complete remember to r
436. th 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 NOTIFY method in the SIP Specific Event Notification framework RFC 3265 The events used are e call info for call appearance state notification e line seize for the phone to ask to seize the line Bridged Line Appearance Signaling 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 Thi
437. the number of SRTP packets If set to 0 or Null the master key lifetime is not set If set to 1 or greater master key lifetime is set The default setting should be suitable for most installations When the lifetime is set greater than 0 a re invite with a new key will be sent when the number of SRTP packets sent for an outgoing call exceeds half the value of the master key lifetime Note Setting this parameter to a non zero value may affect performance of the phone sec srtp mki enabled Oor1 Null The master key identifier MKI is an optional parameter for the cryptographic attribute in the SDP that uniquely identifies the SRTP stream within an SRTP session MKI is expressed as a pair of decimal numbers in the form mki mki_length where mki is the MKI value and mki_length its length in bytes If set to 1 a four byte MKI parameter is sent within the SDP message of the SIP INVITE 200 OK If set to 0 or Null the MKI parameter is not sent noAuth offer noAuth require sec srtp sessionParams sec srtp sessionParams Oor1 Oor1 Null Null If set to 1 no authentication of RTP is offered A session description that includes the UNAUTHENTICATED_SRTP session parameter is sent when initiating a call If set to 0 or Null authentication is offered If set to 1 no authentication of RTP is required A call placed to a phone configured with noAuth require must offer the UNAUTHENTICATED_SRTP sessio
438. through their Polycom VVX 1500 phone or through a computer using http myinfoportal apps polycom com When they sign in they will be asked to accept the Polycom End User Licensing Agreement EULA The Application Launch Pad is enabled by default This means that the Microbrowser configuration that is standard on the SoundPoint IP and SoundStation IP phones will not work on the Polycom VVX 1500 If you want to use the Microbrowser you must add the Microbrowser to the Application Launch Pad For more information refer to Microbrowser lt mb gt on page A 113 To get the My Info Portal to appear in the phone s idle browser set mb idleDisplay home to http idle myinfoportal apps polycom com idle and mb idleDisplay refresh to 600 Configuring Your System Configuration changes can be performed centrally at the provisioning server Central provisioning server Configuration file Turn this feature on or off and configure how it appears sip cfg For more information refer to Microbrowser lt mb gt on page A 113 For more information refer to Web Server lt httpd gt on page A 75 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 w
439. tion refer to Network Address Translation lt nat gt on page A 144 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 provisioning 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 provisioning server Corporate Directory Note This feature requires a license key for activation except on the Polycom VVX 1500 Using this feature may require purchase of a license key or activation by Polycom channels For more information contact your Certified Polycom Reseller POLYCOM Configuring Your System The SoundPoint IP SoundStation IP and Polycom VVX phones can be configured to interface with a corporate directory server that supports the Lightweight Directory Access Protocol LDAP version 3 Currently the following LDAP servers are supported e Microsoft Active Directory 2003 e Sun ONE Directory Server 5 2 p6 e Open LDAP Directory Server 2 4 12 e Microsoft Active Directory Application Mode ADAM 1 0 SP1 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 co
440. 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 Oori 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 Oor1 0 If set to 1 the headset will be selected as the preferred transducer after its first use until the headset key is pressed again otherwise hands free will be selected preferentially over the headset up useDirectoryNames Oori 0 If set to 1 the name fields of the local contact 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 Note There is no matching of outgoing calls There is no matching to corpo
441. tored 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 Setting up Your System 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 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 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
442. tral provisioning server Configuring Your System 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 Note Server based and local call forwarding are disabled if Shared Call Appearance or Bridged Line Appearance is enabled 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 be performed centrally at the provisioning server or locally Configuration file Enable or disable server based call forwarding sip cfg e For more information refer to SIP lt SIP gt on page A 11 Enable or disable local call forwarding behavior when server based enabled For more information refer to SIP lt SIP gt on page A 11 Enable or disable display of Diversion header and the order in which to display the caller ID and number For more information refer to SIP lt SIP gt on page A 11 Configuration file Enable or disable server based call forwarding as a per registration phonet cfg feature For more information refer to Registrat
443. tralized 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 local contact directory that can be downloaded from the provisioning server and edited locally Any edits to the Contact Directory made on the phone are saved to the provisioning server as a backup 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 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 Access URL in SIP Message A
444. trally at the provisioning server Central Configuration file Specify the basic tone frequencies levels and basic repetitive provisioning sip cfg cadences server For more information refer to Chord Sets lt chord gt on page A 33 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 34 Specify patterns For more information refer to Patterns lt pat gt on page A 36 and Call Progress Patterns on page A 37 Microbrowser Note The Sound Point IP 430 450 550 560 650 and 670 phones the SoundStation IP 6000 and 7000 phones and the Polycom VVX 1500 phones support an XHTML Microbrowser This can be launched by pressing the Applications key or it can be accessed through the Menu key by selecting Applications On some older phones the Applications key is labelled Services 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 which can be found at http www polycom com voicedocumentation Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Configuration changes can be performed centrally at the provisioning server or locally Central Config
445. tring 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 23 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 separated by 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 23 Routing lt routing gt This attribute includes e Server lt server gt e Emergency lt emergency gt This attribute allows specific routing paths for outgoing SIP calls to be configured independent of other default configuration A 141 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Server lt server gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation dialplan x routing server y dotted decimal IP address Null IP address or host name and address or host name port of a SIP server that will 7 7 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
446. ts of the solution are SRTP capable and some are not In SIP 3 2 sec srtp requireMatchingTag was added to sip cfg as a flag to force a check of the tag value in the crypto attribute in an SDP answer For detailed configuration instructions refer to Technical Bulletin 25751 Secure Real Time Transport Protocol on SoundPoint IP Phones at http www polycom com support voice soundpoint_ip VoIP_Technical _Bulletins_pub html Configuration changes can be performed centrally at the boot server Central Configuration File Specify the parameters to enable and disable SRTP boot server sip cfg e For more information refer to SRTP lt srtp gt on page A 105 Configuration File Encryption Configuration files excluding the master configuration file contact directories and configuration override files can all be encrypted 4 90 Note Central provisioning server Configuring Your System Encrypted configuration files can be decrypted on the SoundPoint IP 32x 33x 430 450 550 560 650 and 670 the SoundStation IP 6000 and 7000 and the Polycom VVX 1500 phones The master configuration file cannot be encrypted on the provisioning server This file is downloaded by the bootROM that does not recognize encrypted files For more information refer to Master Configuration Files on page A 2 For more information on encrypting configuration files including determining whether an encrypted file is the same as an u
447. ture depends on support from a SIP server Configuration changes can be performed centrally at the provisioning server Central Configuration file Turn this feature on or off provisioning sip cfg e For more information refer to Feature lt feature gt on page A 110 server Configuration file phone1 cfg Enable this feature per registration For more information refer to Registration lt reg gt on page A 128 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Feature Synchronized Automatic Call Distribution Note As of SIP 3 1 2 you can use your SoundPoint IP phones in a call center agent supervisor role on a supported call server When this feature is enabled the phone will indicate the ACD Call Center Agent state as directed by the call server The call center agent is provided with an entry method to initiate Sign In Sign Out and other ACD states through soft keys however the phone state will only change once the server has acknowledged that the phone can move into that new state in this way the ACD state is maintained in synchronization with the call server and any ACD computer based soft clients The SIP signaling used for this implementation is described in the Device Key Synchronization Requirements Document Release R14 sp2 Document version 1 6 Contact Polycom Product Management for more information The Feature Synchronized ACD feature is supported on the SoundPoin
448. uch 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 Overview Call Timer A separate call timer in hours minutes and seconds is maintained for each distinct call in progress Call Transfer Call transfer allows the user to transfer a call in progress to some other destination Call Waiting When an incoming call arrives while the user is active on another call the incoming call is presented to the user visually on the display and a configurable sound effect will be mixed with the active call audio Called Party Identification The phone displays and logs the identity of the party specified for outgoing calls Calling Party Identification The phone displays the caller identity derived from the network signalling when an incoming call is presented if information is provided by the call server Connected Party Identification The identity of the party to which the user has connected is displayed and logged if the name is provided by the call server Context Sensitive Volume Control The volume of user interface sound effects such as the ringer and the receive volume of call audio is adjustable Customizable Audio Sound Ef
449. ue 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 Interpretation voice txPacketFilter Oor1 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 O or Null no Tx filtering is performed This attribute includes e Voice Coding Algorithms lt codecs gt e Volume Persistence lt volume gt e Gains lt gain gt e Acoustic Echo Cancellation lt aec gt e Acoustic Echo Suppression lt aes gt e Background Noise Suppression lt ns gt e Automatic Gain Control lt agc gt e Receive Equalization lt rxEq gt Transmit Equalization lt txEq gt e Voice Activity Detection lt vad gt e Quality Monitoring lt quality monitoring gt Voice Coding Algorithms lt codecs gt These codecs include e Codec Preferences lt codecPref gt e Codec Profiles lt audioProfile gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Codec Prefe
450. ues we 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 voice ns hf enable 1 voice ns hf signalAttn 6 voice ns hf silenceAttn 9 voice ns hf IP_4000 enable 1 voice ns hf IP_4000 signalAttn 6 voice ns hf IP_4000 silenceAttn 9 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones amp 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 x Polycom recommends that you do not change these values POLYCOM Attribute Default voice rxEq hs IP_330 preFilter enable 1 voice rxEq hs IP_430 preFilter enable 1 voice rxEq hs IP_450 preFilter enable 1 voice rxEq hs IP_650 preFilter enable 1 voice rxEq hs VVX_1500 preFilter enable 1 voice rxEq hs IP_330 postFilter enable 0 voice rxEq hs IP_430 postFilter enable 0 voice rxEq hs IP_450 postFilter enable 0 voice rxEq hs IP_650 postFilter enable 0 voic
451. uides 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 VVX phone s Microbrowser e Technical Bulletins which describe workarounds to existing issues and provide expanded descriptions and examples 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 voicedocumentation 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 VVX models BootROM version SIP Application version Partner Platform Administrators Guide for the SoundPoint IP SoundStation IP VVX Family Contents About This Guide ccc cece cece cece eeece H 1 Introducing the SoundPoint IP SoundStation IP VVX POM asker A E E E E SoundPoint IP Desktop Phones 0 cece cece eee ee 1 1 SoundStation IP Conference Phones 0 0 cece eee eee 1 4 Polycom VVX 1500 Business Media Phone 00055 1 6 Key Features of Y
452. ull The star code used to initiate retrieve of a parked call This attribute also includes e Shared Calls lt shared gt e Hold Local Reminder lt hold gt lt localReminder gt Administrator s Guide for the SoundPoint IP SoundStation IP VVX Phones Shared Calls lt shared gt This configuration attribute is defined as follows Attribute Permitted Values Default Interpretation call shared disableDivert Oori 1 If set to 1 disable diversion feature for shared lines Note This feature is disabled on most call servers call shared seizeFailReorder Oori If set to 1 play re order tone locally on shared line seize failure call shared oneTouchResume Oori 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 exposeAutoHolds Oor1 If set to 1 on a shared line when setting
453. ult 150ms tterBufferMin always cause lost packets This parameter 500ms to should be set to the smallest possible value 2500ms that will support the expected network jitter default 2000ms video profile H2631998 jitterBufferMin 33ms to The smallest jitter buffer depth in milliseconds 1000ms that must be achieved before 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 Configuration Files Attribute video profile H2631 998 jitterBufferShrink Permitted Values 33ms to 1000ms default 70ms Interpretation The absolute minimum duration time in milliseconds of RTP packet Rx with no packet loss between jitter buffer size shrinks Use smaller values 33 ms to minimize the delay on known good networks Use larger values 1000ms to minimize packet loss on networks with large jitter 3000 ms video profile H2631998 payloadType 96 default to 127 RTP payload format type for H263 1998 90000 MIME type video profile H2631998 CifMpi video profile H2631998 QcifMpi 1 default to 32 1 default to 32 This value is H263 1998 90000 format parameter CIF used to signal Polycom VVX 1500 receiving capability in SDP This value also contro
454. ult 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 methods Configuration Files 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 S
455. undSync Oor1 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 Configuration Files Attribute dir corp backGroundSync period Permitted Values 3600 to 604800 seconds Default 86400 Interpretation 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 Oor1 If set to 0 the browse position in the data on the LDAP server and the attribute filters are reset for subsequent usage of the corporate directory If set to 1or Null the browse position in the data and the attribute filters are retained for subsequent usage of the corporate directory dir corp cacheSize 8 to 256 128 The maximum number of entries that can be cached locally on the phone dir corp pageSize 8 to 64 32 The maximum number of entries requested from the corporate directory server with each query dir corp viv allow Oor1 A flag to determine whether or not VLV queries can be made if the LDAP server supports VLV If set to 0 VLV queries are disabled If set to 1 or Null VLV queries are enabled Note If VLV is enabled dir corp attribute x searchab le is ignored dir corp vlv sort
456. 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 Oor1 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 Configuration Files 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 provisioning server that allows uploads is available to store a back up copy of the directory or its contents will be lost when the phone reboots or loses power Permitted Attribute Values Default Interpretation dir local volatile 2meg Oor1 0 Applies to platforms with 2 Mbytes of flash memory If set to 1 us
457. uration file Specify the Application browser home page a proxy to use and size provisioning sip cfg limits server For more information refer to Microbrowser lt mb gt on page A 113 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 117 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 provisioning 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 provisioning server Application Launch Pad Note Note Note This feature is only supported on the Polycom VVX 1500 You can configure a page similar to the Menu page where users can launch any applications that run on phone There are two built in applications the Digital Picture Frame and the My Info Portal There are four additional entries that you can configure for any of your company s applications For more information on application development refer to the Web Application Developer s Guide which can be found at http www polycom com voicedocumentation Users can sign up for access to My Info Portal
458. urned on and or the volume is correctly adjusted There are audio and echo issues on the headset Possible issues include Echo on external calls through a gateway 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 Productivity Suite Symptom Problem Corrective Action A user is trying to access one of the following features but it is not available on their phone e Corporate Directory e Recording and Playback of Audio Calls e Managing Conferences The license is not installed on the phone or it has expired Do the following Press the Menu key then select Status gt Licenses e Using the arrow keys verify that the feature in question has a valid license If no licenses are installed the No license installed message appears 5 16 Upgrading Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones Symptom Problem Corrective Action SoundPoint IP 300 301 500 501 600 and or 601 and or SoundStation IP 4000 behave incorrectly or do not display new features New features are not supported on the SoundPoint IP 300 301 500 501 600 and 601
459. us enter the user or administrator passwords The administrator password can be used anywhere that the user password is used Factory default passwords are User password 123 Administrator password 456 Passwords Administrator Network 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 provisioning server If the provisioning server permits uploads these override setting will be saved ina file called lt Ethernet address gt phone cfg on the provisioning server as well as in flash memory Warning Local configuration changes will continue to override the provisioning server derived configuration until deleted through the Reset Local Config menu selection or configured using the device set procedure For more information refer to Modifying Phone s Configuration Using the Web Interface on page C 26 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones This chapter provides you with some tools and techniques for troubleshooting SoundPoint IP SoundStation IP VVX phones and installations The phone can p
460. ute is defined as follows Permitted Attribute Values Default Interpretation call donotdisturb perReg Oor1 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 11 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 32x 33x 430 x 1 2 IP 450 x 1 3 IP 550 560 x 1 4 VVX 1500 x 1 6 IP 650 670 x 1 34 IP 6000 x 1 IP 7000 x 1 Attribute Permitted Values Default Interpretation call autoOffHook x enabled Oori 0 If set to 1 a call will be automatically placed to the contact specified upon call autoOffHook x contact ASCII encoded string Null going off hook on this registration containing digits the user part of a SIP URL or a string that constitutes a valid SIP URL 6416 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 A 134 Configuration Files In the following table x is the registration number IP 32x 33x 430 x 1 2 IP 450 x
461. validate the client certificate during the handshake If the server is configured to require mutual TLS a device certificate and an associated private key must be loaded on the phone The digital certificate stored on the phone is used by e HTTPS device configuration if the server is configured for Mutual Authentication e SIP signaling when the selected transport protocol is TLS and the server is configured for Mutual Authentication e Syslog when the selected transport protocol is TLS and the server is configured for Mutual Authentication Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Note e Corporate Directory when the selected transport protocol is TLS and the server is configured for Mutual Authentication At this time the user will not be able to modify or update the digital certificate or the associated private key stored on the phone during manufacturing The Polycom Root CA can be downloaded from http pki polycom com pki Polycom Root CA crt The location of the Certificate Revocation List CRL a list of all expired certificates signed by the Polycom Root CA is part of the Polycom Root CA digital certificate If Mutual TLS is enabled the Polycom Root CA must be downloaded onto the HTTPS server At this time the following operating systems web servers combinations are supported e Microsoft Internet Information Services 6 0 on Microsoft Windows Server 2003 e Apac
462. ve FTP is still supported Note Only implicit FTPS is supported Setting up Your System 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 provisioning 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 provisioning 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 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 attemp
463. veral 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 server refer to Server Menu on page 3 10 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 MACaddress 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 10 Protocol used by Protocol used by bootROM SIP Application IP 32x 33x 430 IP 32x 33x 430 450 550 560 650 450 550 560 650 Specified 670 6000 7000 670 6000 7000 Protocol VVX 1500 VVX
464. ware gt gt Press Menu and then select Status gt Diagnostics gt Test Hardware gt Audio Diagnostics Keypad Diagnostics or Display Diagnostics 5 10 Power and Startup Troubleshooting Your SoundPoint IP SoundStation IP VVX Phones Symptom Problem Corrective Action There are power issues The SoundPoint IP SoundStation IP VVX 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 Administrator s Guide SoundPoint IP SoundStation IP VVX Controls Symptom Problem Corrective Action The dial pad does not work The dial pad on the SoundPoint IP SoundStation IP VVX family SIP phone does not respond Do one of the following Check for a response from other feature keys or from the dial pad Place a call to the phone from a known working telephone Check for display updates Press the Menu key followed by System Status and Server Status to check if the telephone is
465. will first use the custom option if present or use Option 66 if the custom option is not present If the DHCP server sends nothing the following scenarios are possible Ifa boot server value is stored in flash memory and the value is not 0 0 0 0 then the value stored in flash is used e Otherwise the phone sends out a DHCP INFORM query Ifa single alternate DHCP server responds this is functionally equivalent to the scenario where the primary DHCP server responds with a valid boot server value The phone prefers the custom option value over the Option 66 value but if no custom option is given the phone will use the Option 66 value If no alternate DHCP server responds the INFORM query process will retry and eventually time out Boot Server Option 128 through 254 Cannot be the same as VLAN ID Option When the boot server parameter is set to Custom this parameter specifies the DHCP option number in which the phone will look for its boot server Boot Server Option Type 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 Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family Name VLAN Discov
466. will take the user back through the browser history This is the null default behavior for backward compatibility If set to 0 the phone will not provide a Back soft key All soft keys will be created and controlled by the application mb main x url string Null The internal external URI icon and associated an text for at most six applications mb main x icon Null for x 1 to 6 mb main x text Null The default values for x 1 is mb main 1 url PicFrame mb main 1 icon http 127 0 0 1 launch pad geticonPicFrame mb main 1 text Picture Frame The default values for x 2 is mo main 1 url http myinfoportal apps polycom com mb main 1 icon http 127 0 0 1 launch pad geticonMyPortal mo main 1 text My Info Portal Note This feature is supported on the Polycom VVX 1500 only 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
467. xercise caution when doing this to ensure that the phone and network load generated by registration refresh of multiple registrations do not become excessive This would be of particularly concern if a phone had multiple registrations with multiple servers per registration and it is expected that some of these servers will be unavailable Phone Operation for Registration After the phone has booted up it will register to all the servers that are configured Server 1 is the primary server and supports greater SIP functionality than any of servers For example SUBSCRIBE NOTIFY services used for features such as shared lines presence and BLF will only be established with Server 1 Upon registration timer expiry of each server registration the phone will attempt to re register If this is unsuccessful normal SIP re registration behavior typically at intervals of 30 to 60 seconds will proceed and continue Administrator s Guide for the SoundPoint IP SoundStation IP VVX Family 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 w
468. y DTMF tones in response to user dialing on the dial pad Dynamic Noise Reduction Provides maximum microphone sensitivity while automatically reducing background noise on SoundStation IP 7000 conference phones IEEE 802 1p Q The phone will tag all Ethernet packets it transmits with an 802 10 VLAN header IP Type of Service Allows for the setting of TOS settings Jitter Buffer and Packet Error Concealment Employs a high performance jitter buffer and packet error concealment system designed to mitigate packet inter arrival 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 Treble Bass Controls Equalizes the tone of the high and low frequency sound from the speakers on SoundStation IP 7000 conference phones 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 ow
469. y override global settings unless deleted through the Reset Local Config menu selection Message Waiting Indication Configuring Your System The phone will flash a message waiting indicator MWI LED when instant messages and voice messages are waiting Configuration changes can be performed centrally at the provisioning server Central Configuration file provisioning phonet cfg server Specify per registration whether the MWI LED is enabled or disabled e For more information refer to Message Waiting Indicator lt mwi gt on page A 143 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 29 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 For related configuration changes refer to Local Contact Directory on page 4 10 Distinctive Ringing There are three options for distinctive ringing 1 The user can select the ring type for each line by pressing the Menu key and then selecting Settings gt Basic gt Ring Type This option has t
470. y used features Up to four context sensitive soft keys for further menu driven activities Introducing the SoundPoint IP SoundStation IP VYX Family 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 Field upgradeable Upgrade SoundPoint IP SoundStation IP VVX as standards develop and protocols evolve Extends the life of the phone to protect your investment Application flexibility for call management and new telephony applications 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 102 x 23 pixel graphical LCD for the SoundPoint IP 320 321 330 331 256 x 116 pixel graphical grayscale LCD for the SoundPoint IP 450 320 x 160 pixel graphical grayscale LCD for the SoundPoint IP 550 560 650 supports Asian characters 320 x 160 pixel graphical color LCD for the SoundPoint IP 670 supports Asian characters 248 x 68 pixel graphical LCD for the SoundStation IP 6000 256 x 128 pixel graphical grayscale LCD for the SoundStation IP 7000 800 x 480 pixel graphical color LCD for the Polycom VVX 1500 touch screen Dual auto sensing 10 100 1000baseT Ethernet ports Leverages existing infrastructur

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