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Polycom 3725-23487-003/A Computer Hardware User Manual

1. Clg Action Parameter Description File sip added volpProt SIP useSendonlyHold Can be set to 0 or 1 Null default is 0 Default in sip cfg is 1 If set to 1 the phone will send a reinvite with a stream mode attribute of sendonly when a call is put on hold This is the same as the previous behavior If set to 0 the phone will send a reinvite with a stream mode attribute of inactive when a call is put on hold Note The phone will ignore the value of this parameter if set to 1 when the parameter volpProt SIP useRFC2543hold is also set to 1 default is O sip added dialplan apply ToUserSend 1 Refer to Technical Bulletin 11572 dialplan applyToUserDial 1 dialplan applyToCallListDial 0 dialplan applyToDirectoryDial 0 sip changed dialplan digitmap timeOut 3 to Refer to Technical Bulletin 11572 3 3 3 3 3 3 sip changed tcplpApp sntp daylightSavings start mo Changes to support new daylight savings nth 4 to 3 time rules sip changed tcplpApp sntp daylightSavings start dat e 1 to 8 sip changed tcplpApp sntp daylightSavings stop mon th 10 to 11 sip changed tcplpApp sntp daylightSavings stop day OfWeek lastInMonth 1 to 0 sip added call stickyAutoLineSeize onHookDialing Refer to Administrator s Guide Addendum for SIP 2 1 sip changed voice gain rx digital chassis IP_650 9 Gain changes required to match new to 6 software load sip changed voice gain rx digital ringer
2. 2 19 4 Configuration File Parameter Changes None 2 20 Version 1 6 3 2 20 1 Added or Changed Features 11358 Added configurable subdirectories for configuration and contact directory override files 12761 Added support for setting flash parameters from configuration file 13029 Added support for new dialog event package draft draft ietf sipping dialog package 06 txt 13030 Added support for new BLA draft draft anil sipping bla 02 txt 13222 Changed maximum number of XML retries for SAS VP to be equal to 7 days Page 48 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 13931 Added notice of file system fix for bug 13361 to header of SoundStation IP 4000 binary image 2 20 2 Removed Features 13025 Disabled url dialing in main partner configuration files 2 20 3 Corrections The following issues have been resolved with this release 11271 Phone repeatedly tries to upload log file when log render file parameter disabled 12449 Shared line continues to ring after receiving a CANCEL event in some scenarios 12470 Misplaced comma in date display for two possible date formats 12748 Caller ID shows IP address when PSTN caller is unknown Note The url dialing feature must be disabled in order for the IP address to be hidden 12842 Some characters sent in the dial string should be escaped but are not 13089 Outbound proxy port greater than 6535 does not work 13198 Long date format gets cha
3. 2007 Polycom Inc Page 19 Release Notes SIP Application Changes Clg Action Parameter Description File sip added voice handset rxag adjust IP 330 New parameters to support SoundPoint voice handset txag adjust IP 330 IP 320 and 330 platforms which will be voice handset sidetone adjust IP 330 supported in a future software release Do voice headset rxag adjust IP 330 not change these values voice headset txag adjust IP 330 voice headset sidetone adjust IP 330 dir search field font IP_330 1 name bitmap IP_330 1 name to bitmap IP_330 66 name ind idleDisplay mode ind anim IP_330 38 frame 1 bitmap ind anim IP_330 38 frame 1 duration ind gi IP_330 1 index to ind gi IP_330 10 index ind gi IP_330 1 class to ind gi IP_330 10 class ind gi IP_330 1 physX to ind gi IP_330 10 physX ind gi IP_330 1 physY to ind gi IP_330 10 physY ind gi IP_330 1 physW to ind gi IP_330 10 physW ind gi IP_330 1 physH to ind gi IP_330 10 physH 2 7 Version 2 1 0 2 7 1 Added or Changed Features e 5844 Enhanced support for server fall back configurations e 7275 Microbrowser should auto navigate to first selectable item e 7444 Added table support to microbrowser e 8452 Added microbrowser support to the SoundStation IP 4000 e 9268 Added unique prompt for billing code entry e 9649 Enhanced global prefix character for E 164 user parts in sip URIs e 11572 Added ability to strip or insert leading digits for outgoing c
4. We POLYCOM Helease Notes SIP Application SoundPoint and SoundStation IP Version 2 2 2 2 December 2007 Part Number 3804 11530 222 Copyright 2007 Polycom Inc All rights reserved Release Notes SIP Application Copyright 2007 Polycom Inc All rights reserved Release Notes SIP Application Table of Contents Table of Contents EN GENERAL aate A EEE 1 1 1 IMPORTANT NOTES aiaia a a iie der 1 1 2 SYSTEM REQUIREMBNES td ots reg occas ic pee Ni peu ea UA a ple ded 1 1 3 DISTRIBUTION FILES sis 2 NS A E A EEEE EE 3 2 1 Ka 3 2 1 1 Added or Changed EE 3 2 1 2 Removed PESTE iu iter D et E A ST 3 2 4 3 COITECHONS EE 3 2 1 4 Configuration File Parameter Changes a de uw ene eua una ae did 4 2 2 VERSION 2 2 1 LIMITED RELEASE ccccccccccccscssssssececesccecessessaecesecceeceessnsecececeescsesessaaeeeseeeens 4 2 2 Added or Changed Features uu tbe dei ee ix a a S I ied d Ced d bed a 4 2 2 2 Removed Features a EAS aa E ai 4 2 2 3 OIA LAA ee J 2 2 4 Configuration File Parameter Changes 3 esta et t tes 2 2 3 VERSION 2 2 0 E ias 5 2 3 1 Added or E 5 2 3 2 Removed Features oleren td ax tacos a 7 2 3 3 E WES 7 2 3 4 Configuration File Parameter Changes ie tette tases dansa ek Aen 10 23 VERSION Li op vale sait dede e Cotidie icd dua A A E AAA 14 2 4 1 Added or Changed Features ld lado aiti Loeb de di laa Md 14 2 4 2 Removed EE 14 2 4 3 Corrections a 14 2 4 4 Configuration F
5. sip added voice txPacketFilter See Administrator s Guide for SIP 2 2 0 for details Page 10 Copyright 2007 Polycom Inc Release Notes SIP Application Changes cfg File Action Parameter Description sip added voice codecPref IP_7000 xxx Not currently used will be used ina future release sip added voice audioProfile Lin1 6 frequency Not currently used will be used in a voice audioProfile G 7221 xxx future release voice audioProfile G7221C xxx voice audioProfile Siren14 xxx voice audioProfile Siren22 xxx sip added Several gain and other voice The entire gain section in sip cfg must parameters have been added be updated Failure to do this will affect the audio performance of the phone sip added voice rxEq hf IP_7000 xxx Not currently used will be used ina voice txEq hf IP_7000 future release sip added call dialtoneTimeOut See Administrator s Guide for SIP 2 2 0 for details sip added call disableAutoResumeCentralConf Not currently used will be used in a erence future release sip added call singleKeyPressConference Not currently used will be used in a future release sip added call transfer blindPreferred See Administrator s Guide for SIP 2 2 0 for details Sip added call cellPhoneAutoBridging Not currently used will be used in a future release Sip added bitmap IP_7000 xxx Not currently used will be used in a future release Sip added log level change srtp See Admin
6. Messages Waiting is set to no 35692 Functionality breaks down on pressing conference gt gt cancel soft keys after transfer try is rejected Phone reboots 36566 Microbrowser Left arrow when on first field in a form makes cursor turn invisible 36786 Changing audio modes e g handsfree to handset during call set up mode may not work correctly in some circumstances 37284 During a Blind Transfer the phone should terminate the call on receipt of a 180 Ringing Response 37313 RTP packet size incorrect when SRTP authentication turned off 37316 Authentication failing when phones have different payload size 37334 Disabling CDP from the phone menu causes an unnecessary reboot 37709 SoundPoint IP 330 320 phones using the idle micro browser may re boot after several days due to low memory 38112 Logging message indicates that default cert bundle in use when custom only has been selected Copyright 2007 Polycom Inc Page 3 Release Notes SIP Application Changes e 38344 If URL dialing is disabled in the configuration file the phone shows Number ServerlP for caller ID This issue occurs on SIP 2 2 0 and SIP 2 2 1 releases only e 38430 In a BLA configuration attempting to make a call on a remotely busy shared line may cause the phone to re boot instead of displaying Service Unavailable Occurs on SoundPoint IP 330 320 430 550 650 phones e 38435 When the phone s local directory is writable unable to
7. Phone does not clear indicators if BLF removed on server Copyright 2007 Polycom Inc Page 29 Release Notes SIP Application Changes 16311 Phone with maximum number of line keys configured may have its line key labels overwritten by roaming buddy records 16373 Local conference host cannot end conference if one leg is put on hold by far end 16562 Expansion Module may reboot if the Do Not Disturb key on the phone is pressed multiple times while the Expansion Module is booting up 16577 Local conference host cannot end conference if first leg was put on hold by far end when conference was created 16659 To and Refer to domains incorrect during failover 16681 In some scenarios a phone may initiate a call using TCP but send an ACK using UDP 16768 Inconsistent backlight behavior on SoundStation IP 4000 when resuming a call or conference 16904 Excessive logging from soem module at boot time in some scenarios involving Expansion Module 17009 Non numeric characters or an invalid IP address when dialing by IP may cause the phone to reboot 17068 If the silent ringer is selected an incoming call can only be answered in hands free mode 17102 SoundPoint IP 430 phone locks up instead of rebooting after detecting an operating system suspended task bug 17037 17188 Time information in placed call list contains incorrect data after a transfer has been done 17257 Phone loses audio when there is an active ca
8. SIP Application Notes 32476 IP601 does not work correctly when Presence feature is enabled with LCS server without using Roaming Buddies Workaround Enable roaming buddies by setting roaming buddies reg to the LCS registration number 32611 BLA line can not place and hold more than 10 calls Workaround For BLA lines ensure that call callsPerLineKey is set to 10 or lower 32816 Phone crashes on subsequent call if using NTLM and received transfer from non NTLM phone Workaround Ensure that all phones involved in a transfer use NTLM or do not use NTLM authentication 32994 SoundPoint IP 650 phone may have an incomplete display with only shades of grey after booting up Workaround Cycle power to the phone to make it boot again 33063 Active FTP mode is not supported for phone provisioning Workaround Configure the ftp server for Passive FTP operation 33445 LCS Presence and dialing from Buddy Lists does not work across Federations Workaround To dial contacts across federations program a speed dial with the SIP URI of the contact There is no workaround for watching Federated Buddy status from the phone 33593 Shared line does not show remote active for the second incoming call if callsPerLineKey parameter is set to 1 Workaround Set callsPerLineKey parameter to a value greater than 1 34454 If microbrowser is enabled and refreshes are too frequent and pages contain large images the phone may crash Workaround
9. qos ip rtp max reliability iv qos ip rtp min cost v qos ip rtp precedence 2 Similarly when qos ip callControl dscp has a valid value then it overrides qos ip callControl min delay etc 3 1 13 From Version 1 6 6 to 1 6 7 3 1 13 1 Mandatory Changes e Selecting sticky line seize behavior To have the same line seize behavior as SIP 1 6 5 set call stickyAutoLineSeize to 1 in sip cfg 3 1 13 2 Optional Changes e Overriding codec preferences received from far end To allow the phone to override the list of codec preferences received by the phone set volpProt SDP answer useLocalPreferences to 1 in sip cfg 3 1 14 From Version 1 6 5 to 1 6 6 3 1 14 1 Mandatory Changes None 3 1 14 2 Optional Changes e Sending re INVITE to server during conference setup on BLA Set call shared exposeAutoHolds to 1 in sip cfg 3 1 15 From Version 1 6 4 to 1 6 5 3 1 15 1 Mandatory Changes e None 3 1 15 2 Optional Changes e Getting SIP server address from DHCP The SIP server address can be obtained from a DHCP server if the new parameters volpProt server dhcp available volpProt server dhcp option and volpProt server dhcp type are configured correctly e Using configuration file values for SNTP parameters instead of DHCP values If the configuration file settings for the SNTP server address or GMT offset should be used instead of the values obtained from a DHCP server set one or both of the new parameters tcplpApp sntp address overrideDHCP an
10. should not be available in buddy list if roaming buddy feature is disabled e 36072 On SoundPoint IP 320 and 330 phones the digit map is not applied to numbers selected from a call list when in the dial tone state e 36074 On SoundPoint IP 320 and 330 phones the digit map is not correctly applied when using hot dialing from the second line key e 36225 Phone may reboot if several voicemail NOTIFY messages are received from the server in a short interval e 36233 Specially crafted Via header in an INVITE can crash the phone e 36504 A call is dropped if a blind transfer to an invalid number is attempted e 36581 SoundPoint IP 320 and 330 phones cannot send nn codes e 36753 One phone drops the call when 2nd party attempts another blind transfer to an invalid number e 36877 All microbrowser text regardless of which tag is used except for href is dim on SoundPoint IP 550 and 650 phones 2 4 4 Configuration File Parameter Changes cfg File Action Parameter Description sip added volpProt SIP authOptimizedInFail This parameter controls failover over behavior during authentication signaling 0 default behavior which obeys the RFC 1 optimization enabled phone first retries a SIP transaction with the server that has just sent a 401 or 407 response Copyright 2007 Polycom Inc Page 15 Release Notes SIP Application Changes cfg File Action Parameter
11. volpProt server dhcp available 1 Allowable range is 128 255 There is no default value for this parameter it must be filled in with a valid value sip added volpProt server dhcp type 0 IP address 1 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 sip added tcplpApp sntp address overrideDHCP and tcplpApp sntp gmtOffset overrideDHCP These parameters determine whether configuration file parameters override DHCP parameters for the SNTP server address and GMT offset The default is 0 which means that DHCP values will override configuration file parameters A value of 1 means that configuration file parameters will override DHCP values 2 19 Version 1 6 4 2 19 1 Added or Changed Features 12278 Added support for SAS VP v3 XML configuration transactions 12883 Added sending and processing the early only flag in the replaces header to support RFC 3891 in call pickup 12890 Added accepting SDP with telephone event on the first line 13492 Disabled CA certificate expiry checking when SNTP has not been configured 2 19 2 Removed Features None 2 19 3 Corrections The following issues have been resolved with this release 7707 LED which shows mute and incoming call and message waiting status can show incorrect state 8598 There is no 1 A a s
12. 1 For backwards compatibility use this setting to send SDP in dialog body sip changed feature 9 enabled The url dialing feature must be disabled by setting feature 9 enabled 0 in order to prevent unknown callers from being identified on the display by an IP address 2 21 Version 1 6 2 2 21 1 Added or Changed Features None 2 21 2 Removed Features None 2 21 3 Corrections The following issues have been resolved with this release e 9580 Changes in Ethernet address gt cfg will not be detected during configuration polling e 11190 Incorrect time zone is used for one to two minutes after a reboot e 12552 Phone reboots if line keys on Expansion Module are pressed rapidly and continuously e 12841 Far end phone continues to ring if near end phone ends call prior to far end answering in specific shared line scenario e 12951 Malformed RTP packets received by phone can cause it to crash 2 21 4 Configuration File Parameter Changes None 2 22 Version 1 6 1 2 22 1 Added or Changed Features e 12296 Pressing and holding unassigned line key adds a directory contact e 12366 Application log is uploaded shortly after reboot Page 50 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 22 2 Removed Features None 2 22 3 Corrections The following issues have been resolved with this release 11388 Phone does not get a CDP response reliably in some scenarios 1
13. 2 22 4 Configuration File Parameter Changes eese eee eene eene corn nr cren nete en 51 2 23 VERSION 1 6 0 BETA ON 51 2 23 1 Added or Changed enuresis 51l 223 2 Removed Features E A A EE EE 52 223 3 Corrections i eenen t A A Ss CEA gE 52 2 23 4 Configuration File Parameter Changes eee ee e eee esee ee eene eee tnnt 53 Dist AAA E 54 3 1 UPGRADING EE 54 3 1 1 from Version 2 O a ge a dev Pede e eb terc pe veto Eee dee 54 3 1 2 From Version2 2 0 10 2 2 1 5 e o T eate etie 54 3 1 3 From Version 2 1 2 to 2 EE 54 3 1 4 From Versiom 2 LL C 1021 2 i etd tsi deter 55 3 1 5 From Version EE E siet tte initi aR 55 3 1 6 From Version 2 10 to 2T T cci lege EO c ges erede 55 3 1 7 From Version 20 310 2 10 i aaa e de 56 3 1 6 From Version 2 0 3 to 2 0 3 B iiie tees se ri eet II ai iae Ives eines 56 3 1 9 From Version 20 32207 56 3 1 10 From Version 2 0 1 to 2 0 2 ccccccccccscccccccccccsssssssecesececssessssseceeeseceeeesssseaeeeeeceesesessseaeeeeees 57 JI FromVersion 2 00 t0 2 01 s e e cte e i Ota ist e eese d tes an lechuga 57 3 142 Prom Version OF 10 2 0 0 5 i rana ote tiem estes 57 3 1 13 From Version Jooro o aa aeiaai Si SR Eies 58 3 1 14 From Version 1 6 5 to 1 6 6 essent eene tnnt esses nenne teta nenne seen 56 SATS From Version LOA LO iia 56 346 From Versto E EE 59 3 1 17 From Version Ioirol oin n a E E E E N 59 3 1 18 From Version 1 6 1 to OA a a a e A E a a RA aS ess 59 SAAD Prom
14. 4 1 Mandatory Changes e Adding logging of version information for configuration files In order for this new feature to work the latest version of all configuration files must be used 3 1 4 2 Optional Changes e Using different versions of configurable items in Ethernet address gt cfg for different phone models or platforms Different phone models or platforms can be configured to use different application files configuration files log file directory etc See technical bulletin TB35361 for details e Optimizing failover behavior for authentication signaling Use the new parameters volpProt SIP authOptimizedlInFailover in sip cfg and reg x auth optimizedInFailover in phone1 cfg to change the phone s failover behavior during authentication signaling if desired e Viewing message waiting indicators while still retaining one touch voicemail access when multiple lines are configured If a phone has multiple lines with just one registration set to have msg mwi x callBackMode registration and all others set to have msg mwi x callBackMode disabled but it is desirable to be able to see message waiting indicators for all lines and still retain one touch voicemail access set the new parameter up mwiVisible to 1 in sip cfg 3 1 5 From Version 2 1 1 to 2 1 1 C 3 1 5 1 Mandatory Changes None 3 1 5 2 Optional Changes None 3 1 6 From Version 2 1 0 to 2 1 1 3 1 6 1 Mandatory Changes None 3 1 6 2 Optional Changes e Using URI
15. Description sip added up mwiVisible 0 same behavior as SIP 2 1 1 this is the default behavior 1 2 if msg mwi x callBackMode parameter is set to disabled message waiting indicator is displayed but voicemail cannot be accessed sip changed Changed file header from This is required to support the new Revision Date feature 36681 described above to RCSfile sip cfg v Revision phone added reg x auth optimizedInFailover If this parameter is set it overrides the global volpProt SIP authOptimizedInFailover parameter x is the registration index See the description for volpProt SIP authOptimizedlInFailover phone1 changed Changed file header from This is required to support the new Revision Date feature 36681 described above to RCSfile phone1 cfg v Revision 000000000000 changed Changed file header from This is required to support the new Revision Date feature 36681 described above to RCSfile 000000000000 cfg v Revision 000000000000 changed Changed file header from This is required to support the new directory ml Revision Date feature 36681 described above to RCSfile 000000000000 directory xml v Revision 2 5 Version 2 1 1 C 2 5 1 Added or Changed Features e 32146 Added support for SoundPoint IP 330 e 33391 Added support for SoundPoint IP 320 e 35415 Added translations for new phrases needed for SoundPoint IP 320 and 330 phones 2 5 2 Removed Feature
16. Do not refresh microbrowser too frequently in configuration settings or by rapidly pressing the Refresh softkey Design the pages so that the content is within reasonable limits 34743 A phone may freeze when it receives a check sync if the resources on the phone are heavily used by downloaded wave files or large or complex microbrowser pages Workaround Reduce the RAM disk size configured in sip cfg this will reduce the amount of space available for downloaded wave files and other resources by setting ramdisk nBlocks to 3072 Design web pages used by the microbrowser carefully 36969 SoundStation IP 4000 phone does not display Japanese language properly Workaround None 37391 The Phone may fail to boot if the contact directory contains improper XML syntax Workaround Ensure that the contact directory is in a proper XML format 37449 The phone may re boot when the user tries to end a local conference if the call server does not respond to the REFER message Workaround Ensure that the server is configured to respond to the REFER that ends the conference Copyright 2007 Polycom Inc Page 61 Release Notes SIP Application Reference Documents e 37391 Brief audio noise due to SRTP encryption key change Workaround To minimize the frequency of occurrence configure the sec srtp key lifetime as long as possible e 37437 When SRTP is used with both Authentication and Encryption enabled on SoundPoint IP 301 501
17. E 42 2 16 44 Configuration File Parameter Changes aii 43 Daley VERSION O A Sa ee ee even ae Sule acne 43 2 17 1 Added or EE 43 2449 Removed COMES At tt dit 44 KK OR EE 44 2 17 4 Configuration File Parameter Changes eee eee eese ee eeee conan tnnt that ena 45 DAP MER ODE OS ec Trpo 45 2 18 1 Added or Changed Peaturesau oet eie ita 45 2 13 2 Removed FCO gereegelt 46 Page ii Copyright 2007 Polycom Inc Release Notes SIP Application Table of Contents 2 18 3 e 46 2 18 4 Configuration Pile Parameter E 47 ATI VERSION DOSE ii dido 47 2 19 1 Added OF Changed Features 1 d tiani AAA AA ei reddo 47 2 19 2 Removed Features i ee pto a te Tert ER E tee P esi Leda 47 GN E Ee e 47 2 19 4 Configuration File Parameter Changes 4 cies sies cest custo eee Queue au e Puy anao 46 2 20 MERSION 1 6 3 1 dnce a i od cci eds Uode EE EES 48 2 20 1 Added or CHANCES ARA tes Disi etii A ud etu da 48 2 20 2 Removed e E Deele e tee e oC E EE 49 AA 0 Eo EG e 49 2 20 4 Configuration File Parameter Changes stc ltda 50 2 21 VERSION LO Za di 50 2 21 1 Added or Changed eene tatis eu t tenian AE uie 50 2 21 2 Removed Eeatures isi eee Base eae te etie ree Ha e c pese re etae 50 2 21 3 EE 50 2 21 4 Configuration File Parameter Changes rusia 50 2 229 WBRSION Eege 50 2 224 Added or Changed COMES cce tot aet side dd 50 2 22 2 Removed a cios e ivt t Se est e Teile a iiis 51 2 22 94 COLECCION asta tea uide eM ors E et t 51
18. IP_650 21 to 12 sip changed voice handset sidetone adjust IP_430 12 to 13 sip added volpProt server x transport and Added TCPOnly as a possible value for volpProt SIP outboundProxy transport these existing parameters 2 8 Version 2 0 3 B 2 8 1 Added or Changed Features 14874 Added support for SoundPoint IP 650 platform 15775 Added support for LCD backlight on SoundPoint IP 650 15852 Added support for 32 MB of memory on SoundPoint IP 650 15853 Added support for G 722 audio code on SoundPoint IP 650 16335 Added support for 8 MB of flash on SoundPoint IP 650 Page 24 Copyright 2007 Polycom Inc Release Notes SIP Application Changes e 16686 Added support for USB diagnostics e 17132 Added visual indication of wideband audio 2 8 2 Removed Features None 2 8 3 Corrections The following issues have been resolved with this release None 2 8 4 Configuration File Parameter Changes None 2 9 Version 2 0 3 2 9 1 Added or Changed Features None 2 9 2 Removed Features None 2 9 3 Corrections The following issues have been resolved with this release e 17981 DHCP initialization incorrect for SoundStation IP 4000 which may cause boot time problems on some servers e 18491 Network load reported by SoundPoint IP 430 phones is affected by traffic which is not destined for the phone e 18692 Presence subscribe has application pidf xml in Accept header although it is not
19. URLs 13579 SDP parser applies wrong logic 13793 cnonce generated by the phone is not random 13933 Directory menu display is not perfectly cleaned up after deleting all contacts 14069 Phone may behave incorrectly if an incoming call is answered on a shared line when another phone sharing the line has Do Not Disturb enabled 14083 Wrong expire time might be used when there are multiple contact header lines 14126 If a call is placed to a phone with an unread IM the message waiting indicator LED stops flashing 14172 Phone will reboot when a contact is added to the contact directory which already contains over 40 contacts which are being watched 14390 Changing the DNS server configuration via the phone s menu does not have any effect 14400 Phone can take up to 30 minutes to boot when there are TCP timeouts Copyright 2007 Polycom Inc Page 35 Release Notes SIP Application Changes 14408 Soft key labels do not get updated correctly after hot dial attempt when remote shared line is busy 14467 If a URL in Ethernet Address cfg specifies a protocol and user name but no password the password in flash is not used 14635 No welcome sound effect is played on SoundStation IP 4000 phone 14664 SoundPoint IP 301 and 501 and SoundStation IP 4000 phones fail during a reboot if 12 SAS VP appearances are configured 14781 Cannot use special characters for filenames with TFTP boot server 14844 A failed download of a pr
20. Verston 16 010 LO T ie eile eats i exeo bo Mee est ETARE 59 Copyright 2007 Polycom Inc Page iii Release Notes SIP Application Table of Contents 3 2 OUTSTANDING ISSUES 2 3 occi epe A te Ue e bovet occa tea ios aute eese a Fe thee Poe e ea Pee EE DU 60 4 REFERENCE DOCUMENTS eee eee eese eee eee eee e eee eee e eee e eee e eee eee eee eee eee e eee eee e ee eee 62 Page iv Copyright 2007 Polycom Inc Release Notes SIP Application General 1 General These release notes apply to version 2 2 2 of the SoundPoint IP SIP application This release is a patch release that replaces the 2 2 0 release as the latest generally available GA release For more information refer to the documents listed in Section 4 1 1 Important Notes e This software release does not include images for the SoundPoint IP 300 and 500 phone models If deployments utilize a mix of IP 300 and IP 500 phones along with newer models the steps detailed in technical bulletin 35311 must be followed The technical bulletin is available from www polycom com support voip Search the Knowledge Base for 35311 e Support for encrypted media using SRTP is available in this release Due to the significant inter operability needs when deploying SRTP this feature is available when the phones are used with particular call servers and network infra structure only Please contact your solutions provider to establish whether they offer this feature Any
21. added voice codecPref IP 650 G711Mu These parameters allow the voice codec voice codecPref IP_650 G711A preference list to be set for the SoundPoint voice codecPref IP 650 G729AB IP 650 phone By default the G 722 codec is voice codecPref IP_650 G722 the first choice The use of these parameters is the same as other voice codecPref parameters sip added voice audioProfile G722 payloadSize These parameters configure the G 722 voice audioProfile G722 jitterBufferMin voice codec The use of them is the same voice audioProfile G722 jitterBufferMin as the other voice audioProfile parameters voice audioProfile G722 jitterBufferMin sip added voice gain rx analog chassis IP_650 These parameters control gain settings voice gain rx analog ringer IP_650 which are specific to the SoundPoint IP 650 voice gain rx digital chassis IP_650 phone The values should not be modified voice gain rx digital ringer IP_650 voice gain tx analog chassis IP_650 voice gain tx digital chassis P 650 sip added voice rxEq hs IP 650 preFilter enable These parameters control equalization voice rxEq hs IP_650 postFilter enable voice rxEq hd IP_650 preFilter enable voice rxEq hd IP_650 postFilter enable voice rxEq hf IP_650 preFilter enable voice rxEq hf IP_650 postFilter enable voice txEq hs IP_650 preFilter enable voice txEq hs IP_650 postFilter enable voice txEq hd IP_650 preFilter enable voice txEq hd IP_650 postFilter enable voice txEq hf IP_650 preFilter enable v
22. browser configuration 14029 Lowered CPU load associated with RTP processing 14209 Added support for getting buddy lists from Microsoft Live Communications Server 2005 14322 Added per registration Ics parameters 14323 Added per registration outbound proxy parameters 14348 Added support for connection reuse draft 14496 Added presence support with Windows Messenger 5 1 Office Communicator in Microsoft Live Communications Server 2005 context 14498 Added Windows Messenger 5 1 Office Communicator compatible presence and IM support in peer to peer mode 14556 Added support for roaming access control lists 14610 Added ability to store resource files listed in MISC FILES field in Ethernet Address gt cfg in flash file system For example a dictionary file can be listed which should be used if the phone reboots when the boot server is unavailable 14628 Added support for populating the speed dial list from a roaming buddies list sent by a Microsoft Live Communications Server 2005 Copyright 2007 Polycom Inc Page 33 Release Notes SIP Application Changes 14638 Changed source port for TCP TLS connection to be a random value above 32766 after each reboot 15180 Added configurable maximum number of servers for redundant boot server feature 11785 15363 Changed call timer format 15644 Added a configuration parameter to choose the name of pval field in Dialog 15987 Reduced default resource quota limits f
23. codec in a conference call with SIP 1 6 6 C software 16660 Failover to backup SIP server does not occur when hostname of primary cannot be resolved via DNS 16691 Dialog does not get removed after its expiration time in some scenarios This addresses 16374 and 16480 16813 Going on and off hook repeatedly on a shared line may result in the line showing an active call state when the handset is physically on hook 16915 Phone sends SIP requests to port 5060 regardless of volpProt SIP outboundProxy port configuration setting 17014 When a shared line call is on hold using on hook dialing seizes the last used line instead of the first available line 17284 An unnecessary ACK is sent by the phone if no reply is received within 32 seconds 2 14 4 Configuration File Parameter Changes cfg Action Parameter Description File sip added volpProt SDP answer useLocalPreferences Can be 0 or 1 Use this new parameter to have the phone use its own preference list when deciding which codec to use rather than the preference list in the offer Null default 0 disabled Copyright 2007 Polycom Inc Page 41 Release Notes SIP Application Changes cfg Action Parameter Description File sip added call stickyAutoLineSeize Can be 0 or 1 Set to 1 to make the phone use sticky line seize behavior This will help with features that need a second call object to work with The phone will attem
24. e 19754 Do Not Disturb key cannot be remapped to Null e 19827 Phone using Bridged Line Appearance can send corrupt message header in SUBSCRIBE message e 19875 Phone should use NTP time to check validity of SSL server certificate 19876 Phone will lose some memory if microbrowser displays Cache bounced error message due to unresponsive server e 19883 Handset sidetone level is 3dB too hot on SoundPoint IP 430 e 35063 Power levels reported via CDP for SoundPoint IP 650 are too low e 35068 Power levels reported via CDP for SoundPoint IP 601 with EM Power option enabled are too high 2 7 4 Configuration File Parameter Changes Cfg Action Parameter Description File phone added reg x server y lcs Refer to Technical Bulletin 5844 phone added dialplan x applyToUserSend 1 Refer to Technical Bulletin 11572 dialplan x apply ToUserDialz 1 dialplan x applyToCallListDial 0 dialplan x apply ToDirectoryDial 0 phone added reg x server y transport and Added TCPOnly as a possible value for reg x outboundProxy transport these existing parameters phone changed msg mwi x callBackMode disabled to msg mwi x callBackMode registration for x 2 3 4 5 6 changed for bug 13818 sip added volpProt server 1 lcs Refer to Technical Bulletin 5844 Copyright 2007 Polycom Inc Page 23 Release Notes SIP Application Changes
25. extinguish LED s lit as a result of previous active dialogs 35049 DSP load graph on SoundPoint IP 550 shows slightly incorrect value 35228 Phone may have one way audio when SDP is received with c line below m line 35293 Soft keys have some missing pixels on the SoundPoint IP 430 when the microbrowser is accessed 35308 A known problem in the SoundPoint IP 430 processor may cause the phone to reboot with a DSP assertion failure instead of restarting the affected driver 35477 When handset AEC is enabled on SoundPoint IP 50X and 60X phones echo may occur on speaker phone when switching between handset and speaker phone 35533 The phone s web server shows the DST start and stop days as Monday by default instead of Sunday 35537 A saturated transmit signal may cause SoundPoint IP 430 phone to reboot 35573 After selecting the Russian language and accessing the microbrowser the phone may freeze 36012 Conference host may indicate phone is muted but audio is heard by far end after one leg ends call Page 18 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 6 4 Configuration File Parameter Changes Clg Action Parameter Description File sip added volpProt SIP useContactInReferTo 0 default behavior which is the same as previous behavior use URI from initial call s To header in REFER s refer to header 1 use URI from initial call s Contact header in REFER s refer to header when setti
26. fully supported e 18766 Ethernet transmit level is low on SoundPoint IP 430 phone e 18790 Some shared line scenarios do not work with Broadsoft R14 and R13 MP13 releases e 18919 11981 18997 Time stamp in RTCP packets is incorrect e 19016 SDP containing two a lines causes transfer from a private line to a shared line to fail e 19082 Phone seizes wrong line making outbound call to FAC 55 e 19210 Too many messages are logged when so is set to level 2 Copyright 2007 Polycom Inc Page 25 Release Notes SIP Application Changes 2 9 4 Configuration File Parameter Changes The following configuration file changes have been included in this build in preparation for future inclusion of the IP 650 platform in a software release Support for the IP 650 is not currently included in this release cfg Action Parameter Description File sip added up backlight onIntensity This parameter controls the intensity of the LCD backlight when it turns on during normal use of the phone Possible values are 0 1 2 or 3 0 off 1 to 3 low medium high Null default is 3 high sip added up backlight idlelntensity This parameter controls the intensity of the LCD backlight when the phone is idle Possible values are 0 1 2 or 3 0 off 1 to 3 low medium high Null default is 1 low Note If idlelntensity is set higher than onintensity it will be replaced with the onintensity value sip
27. in Configuration menu on SoundPoint IP 500 and 501 phones when it exceeds a certain length Page 34 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 12155 SoundPoint IP 300 and 301 phones have no Exit soft key during the ACD login process 12308 Cannot place a call from the second line on the phone if the first line is a shared unregistered line 12492 SoundPoint IP 601 phone with Expansion Module s attached may fail to load the selected language after rebooting 12630 When a shared line is being used on another phone pressing the line key for that line can cause the display to show Enter number briefly 12711 Phone should play default ring tone if Alert Info URL is invalid 12952 There is no way to reset the user password back to the factory default password 13230 No audio on calls resumed from hold in some multiple call scenarios 13253 An unregistered SoundStation IP 4000 may reboot if an invalid number is dialed 13320 When the micro browser fetches SSL data this can interrupt audio transmitted by the phone 13358 My Status menu has two offline entries 13477 Pressing Hold Resume soft key twice quickly results in three effective state changes 13500 Phone does not use FTP password stored in flash when OVERRIDES DIRECTORY and CONTACTS DIRECTORY are configured in this format FTP usr IP directory 13512 Parsing of URLs in configuration files does not work for some categories of
28. or Null a crypto line with the AES CM 128 HMAC SHA1 80 crypto suite will be included in offered SDP If set to O the crypto line is not included sip added sec srtp offer HMAC SHA1 32 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 2 3 Version 2 2 0 2 3 1 Added or Changed Features 22532 When there has been no activity in a menu for a configurable period of time the phone returns to the idle display This does not happen if the user is entering data using a menu 25274 Added sending vendor identifier information through DHCP 25702 Added microbrowser support for accepting and displaying a URL that points directly to a BMP image previously it was necessary to embed BMP images in an XHTML document Copyright 2007 Polycom Inc Page 5 Release Notes SIP Application Changes 27040 Added new configurable ring while busy options 28029 Added microbrowser support for two dimensional table navigation using all four arrow keys 28747 Added a general flash file system caching mechanism so that downloaded resources can be stored in non volatile memory 29030 Added automatic provisioning support for individual image files 29854 Added support for tracking of missed calls to be configurable on a per line basis 31558 Added synchronization of local DND CF features with serve
29. reported via CDP to platform specific values In order for these CDP power requirements to be reported at boot time as well bootROM version 3 1 3 is required Copyright 2007 Polycom Inc Page 45 Release Notes SIP Application Changes 15012 Added a workaround to restart the application on the phone if many tasks get unrealistic task delays during startup Outstanding issue 11653 2 18 2 Removed Features None 2 18 3 Corrections The following issues have been resolved with this release 11264 SoundStation IP 4000 hangs when booting if custom DHCP option 150 of type String is used 11302 SoundPoint IP 300 and 301 incorrectly truncate displayed line label if the reg x label field is empty and reg x address is longer than 4 characters 13904 SoundStation IP 4000 always shows LAN Mode as half duplex 14077 Under certain DNS failover conditions the phone stops sending DNS and SIP requests 14110 Phone does not reset to using All Certificates for CA Certificates after the user chooses the Reset Device Settings menu option 14163 Phone incorrectly updates Placed Calls list with an empty entry after New Call then End Call are pressed 14166 Calls answered on a phone with a shared line are incorrectly logged in the Received Calls list of another phone sharing that line 14474 Phone won t upload all log files to TFTP boot server if LOG FILE DIRECTORY specified in Ethernet Address gt cfg doesn t exist 14509 If
30. sip added voice rxEq hs IP_430 preFilter enable New Rx EQ parameters for SoundPoint IP voice rxEq hs IP_430 postFilter enable 430 platform voice rxEq hd IP_430 preFilter enable voice rxEq hd IP_430 postFilter enable voice rxEq hf IP_430 preFilter enable voice rxEq hf IP_430 postFilter enable sip added voice txEq hs IP_430 preFilter enable New Tx EQ parameters for SoundPoint IP voice txEq hs IP_430 postFilter enable 430 platform voice txEq hd IP_430 preFilter enable voice txEq hd IP_430 postFilter enable voice txEq hf IP_430 preFilter enable voice txEq hf IP_430 postFilter enable sip added voice handset rxag adjust IP_430 New handset and headset gain adjustments voice handset txag adjust IP_430 for SoundPoint IP 430 platform voice handset sidetone adjust P 430 voice headset rxag adjust P 430 voice headset txag adjust IP 430 voice headset sidetone adjust IP 430 sip added font IP_400 1 name New dynamic font download parameter for SoundPoint IP 430 platform sip added bitmap IP_400 61 name New bitmap parameter for SoundPoint IP 430 platform sip added ind anim IP_400 38 frame 1 bitmap New animation parameters for SoundPoint ind anim IP_400 38 frame 1 duration IP 430 platform sip changed ind gi IP_400 Changed the values of some of these indicator parameters for the SoundPoint IP 430 platform 2 17 Version 1 6 6 2 17 1 Added or Changed Features 15491 Added configurable option to enable phone wit
31. 060 SoundPoint IP PolycomSoundPointlP SPIP 650 libcurl 7 12 1 r n Example showing format of user agent in HTTP GET s now with security sec tagSerialNo set to 1 User Agent Microbrowser 1 1 PolycomSoundPointIP SPIP_430 UA 2 1 0 2643 SN 0004f210013a 19111 Added TCPOnly as a transport option 19425 Added microbrowser support for form input elements with checked true attribute 19443 Added microbrowser support for forms within tables 19572 Added configurable sticky line seize behavior only for on hook dialing 2 7 2 Removed Features None 2 7 3 Corrections The following issues have been resolved with this release 7301 Phone doesn t ring if one line has Do Not Disturb enabled 16354 Inconsistent error message given when attempting to make a call on an unregistered line using different methods when call enableOnNotRegistered is set to 0 16477 When phone is configured for NAPTR transport but server does not contain NAPTR and SRV the phone may do SRV lookups for A records or A lookups for SRV records Copyright 2007 Polycom Inc Page 21 Release Notes SIP Application Changes 16899 Phone can send a malformed target URI in some NOTIFY messages in certain scenario 17179 Transfer may fail in some scenarios if the Transfer softkey is pressed before the second party answers 17318 Phone does not update presence status e g to offline when reboot initiated 17422 When using a bridged line if a c
32. 1 From Version 2 2 1 to 2 2 2 3 1 1 1 Mandatory Changes e None 3 1 1 2 Optional Changes TCP Keep Alive message when using TLS Configure the tcplpApp KeepAlive parameters as detailed in Section 2 1 4 if using TLS and there is a risk of the TCP connection being improperly terminated e Read only Contact Directory If it is desired to centrally manage the phones directory the user can be restricted from making any changes To enable this capability set dir local read only 1 e Disable Presence MyStat and Buddies soft keys when using the Presence feature signalling Some call servers use the phones Presence feature for controlling BLF capability but don t implement the full suite of Presence options To avoid giving the user visibility to this setting the idle soft keys may be removed from the phone UI by setting pres idleSoftKeys 0 3 1 2 From Version 2 2 0 to 2 2 1 3 1 2 1 Mandatory Changes e None 3 1 2 2 Optional Changes e None 3 1 3 From Version 2 1 2 to 2 2 0 3 1 3 1 Mandatory Changes e New configuration file settings for audio The entire voice section in the latest sip cfg must be used to ensure good audio quality e New configuration file settings for indicators The entire indicators section in the latest sip cfg must be used to ensure correct icons on the display Page 54 Copyright 2007 Polycom Inc Release Notes SIP Application Notes 3 1 4 From Version 2 1 1 C to 2 1 2 3 1
33. 2208 Indicator for watched contact remains red if speed dial line removed 12247 Two stage dialing user interface not correct 12348 Handsfree and handset buttons do not work correctly to answer call when silent ringer is selected 12364 Cannot establish a centralized conference from one of the conference legs 12475 One Touch Voicemail dialing does not support multiple lines correctly 12506 INVITE message never tried on backup proxy when primary server fails over 12640 CDP word on SoundPoint IP 601 needs to advertise maximum power to Cisco switch 12775 Phone cannot join more than two legs to centralized conference 2 22 4 Configuration File Parameter Changes cfg Action Parameter Description File sip changed voice audioProfile xxx parameter values and Use the new values for these voice gain xxx parameter values parameters 2 23 Version 1 6 0 Beta only 2 23 1 Added or Changed Features 4614 Added display of date and time during a call 9046 Added support for SoundPoint IP Expansion Module 9108 10480 Added support for SoundPoint IP 601 hardware platform 9660 Pressing and holding an assigned speed dial line key opens the contact directory to that entry 11540 Improved speed dial key assignment When perusing the contact directory pressing and holding an unassigned line key assigns the in focus directory entry to that key as a speed dial A confirmation beep is heard When a new director
34. 600 and 601 platforms and three way conferencing is enabled the phone will re boot when a local conference is attempted Workaround Disable local conferencing by setting sec srtp leg allowLocalConf 0 this is the default setting or disable SRTP Authentication See Technical Bulletin 25751 for details e 38279 If a 403 response is received by the phone when attempting to complete a call transfer in the proceeding state the phone may re boot Workaround Set allowTransferOnProceeding 0 which prevents a transfer from occurring during the proceeding state e 39419 Maximum Backlight Intensity setting has very little effect on SoundPoint IP 560 phones Workaround None e 39490 In some call scenarios the phone may not display the SRTP secure line icon even though the call is encrypted Workaround None Note The phone does not ever indicate that a call is encrypted when it is not e 39630 Using SoundPoint IP 330 320 phone with LCS2005 Blocking a roaming buddy from the Privacy list also prevents the user from viewing the Blocked buddy s status Workaround Do not block user s from viewing your status if you wish to view their s 4 Reference Documents e Administrator Guide SoundPoint IP SIP Version 2 2 0 e Technical Bulletins 5844 11572 35311 35361 may be obtained from the Polycom web site Support Knowledge Base www polycom com support voip e Technical Bulletin 25751 is available from the Polycom PRC Page 62 Copyrigh
35. Default 0 Can be 0 or 1 0 Support for MS forking is disabled 1 Support for MS forking is enabled and the phone will reject all Instant Message INVITEs This parameter is relevant for LCS server installations Note that if any endpoint registered to the same account has MS forking disabled all other endpoints default back to non forking mode Windows Messenger does not use MS forking so be aware of this behavior if one of the endpoints is Windows Messenger sip added volpProt SIP dialog usePvalue Default 0 Can be 0 or 1 0 Phone uses pval field name in Dialog This obeys the draft ietf sipping dialog package 06 txt draft 1 Phone uses a field name of pvalue sip added volpProt SIP connectionReuse useAli as Default 0 Can be 0 or 1 0 old behaviour 1 Phone uses the connection reuse draft which introduces alias sip added se pat callProg 15 name secondary dial se pat callProg 15 inst 1 type chord se pat callProg 15 inst 1 value 1 Same configuration method as primary dial tone Allows a different tone to be configured for secondary dial tone Copyright O 2007 Polycom Inc Page 37 Release Notes SIP Application Changes Clg Action Parameter File Description sip added qos ip rtp dscp This parameter allows the DSCP of packets to be specified If set to a value this will override the other qos ip rtp parame
36. Emergency Call Routing does not work correctly if multiple numbers are configured in a single entry in the configuration file e g dialplan 1 routing emergency 1 value 911 9911 34649 First call after a reboot may demonstrate one way audio if phones have different codec preferences and volpProt SDP answer useLocalPreferences parameter is set to default 34891 SoundStation IP 4000 loudness does not decrease for bottom six volume settings 35320 If two function keys are remapped to dial specific speed dial numbers only the first one will work 35480 SoundPoint IP 320 and 330 phones allow watching only 7 buddies instead of 8 and may crash when an 8 watched buddy is added Copyright 2007 Polycom Inc Page 9 Release Notes SIP Application Changes 35490 SoundPoint IP 320 and 330 phones do not display SAS VP failure messages during boot up 36031 If a phone is configured to use TLS for the 2 line and TCP for the 15 the 2 line does not register 36107 SoundStation IP 4000 phone drops maximum size packets when VLAN is enabled 36477 Configuring the nat signalPort parameter may cause the phone to crash 36775 Route Set susceptible to change mid dialog in certain situations 36882 Selecting a speed dial number using the nn key sequence does not work on SoundPoint IP 320 and 330 phones when the phone is unregistered or is using URL dialing mode 36905 CDP packet always advertises LAN duplex mode as Duplex Fu
37. IP 6XX IP 550 560 and IP 4000 only Danish Denmark Dutch Netherlands English Canada English United Kingdom English United States French France German Germany Italian Italy Japanese Japan for IP 6XX IP 550 and IP 4000 only Korean Korea for IP 6XX IP 550 and IP 4000 only Norwegian Norway Portuguese Portugal Russian Russia Spanish Spain Swedish Sweden SoundPointIPWelcome wav start up welcome sound effect Page 2 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 Changes 2 1 Version 2 2 2 2 1 1 2 1 2 2 1 3 Added or Changed Features 35534 De couple Presence Signaling from Idle Screen Soft key UI 36931 Add support for SoundPoint IP 560 product 37053 Add ability to make local contact directory read only from the phone 38328 Add check for local contact directory changes during configuration change checks 38357 Add ability to adjust the maximum brightness of the SoundPoint IP 550 and 650 phones 38371 Allow for TCP keep alive on SIP signaling TLS connections 38654 Add support for SoundPoint IP 320 Part Number 2345 12200 005 and SoundPoint IP 330 Part Number 2345 12200 004 for China market 38888 Add ability to adjust the maximum brightness of SoundPoint IP Backlit Expansion Modules Removed Features 38813 Remove 1000 half duplex as a valid ethernet configuration Corrections 34800 MWI Notify Message Waiting Counts are ignored if
38. ITE will be sent to the server Default is 0 2 18 Version 1 6 5 2 18 1 Added or Changed Features 8072 Added support for Nortel MCP NAT traversal 11805 Changed behavior when a local conference is terminated The remote conference legs are transferred so that the remote parties can continue the conversation 13193 Added configuration options to allow configuration file parameters to override DHCP values for SNTP server address and GMT offset 13527 Added support for setting SIP server address from DHCP option 151 13509 Added allowing reg x address to contain host part instead of being a user part only 13492 CA certificate expiry is no longer checked if SNTP has not been configured 14052 Added flash parameter for SoundPoint IP 601phones to toggle power requirements in CDP between 5W no Expansion Modules can be connected and 12W three Expansion Modules can be connected with a default setting of 5W This EM Power flash parameter is accessible when the SIP application is running under the Network Configuration menu Note that no Expansion Modules can be connected to the phone when the EM Power parameter is disabled The default setting for this parameter is Enabled i e 12W power requirement In order for the correct CDP power requirements to be reported at boot time as well bootROM version 3 1 3 is required See Tech Bulletin TB14052 for details on how to use this feature 14886 Changed power
39. ON 2 0 l Beo ici ella it diles 28 2 11 1 Added or Changed Feat ina ean Corax PS Ter E RU 28 Dou Removed ECOLE OS etia dd taa 28 Dod Teo COREA EE 28 2 11 4 Configuration File Parameter Changes diia 28 2 12 C VERSIONO E 29 2 12 1 Added OF Changed Features uas dee ntque e rice eiae as Sh aq ug e ade tesla 29 222 2 Removed Feat res A a ea ti dpa SE qud a Ls aia bud a d 29 Dal Zid CORTES da iad pO Ka PUOI EET a ul IA AAA Lea OY a SOT RR RA ta TRU 29 212 4 Configuration File Parameter Changes A a ents 31 2 13 VERSION 2 0 0 BETA RELEASE ONLY 32 2 13 1 Added or Changed Features uec eor acis era cop AA ecce Di ees Ree uode 32 243 2 Removed Features iacit utu tin Er b an EE as ees FOR qaia 34 DASS e 34 2 13 4 Configuration File Parameter Changes a Dd deci d cu ue DH ake 37 2 14 VERSION TAO E 40 2 14 1 Added or Changed Features ia id 40 2 14 2 Removed Features iussu oou edet red x EI sagen upon ex s Leave ec expe 40 Z2uldiJ EE 40 2 14 4 Configuration File Parameter Changes aos ds ass 41 2 15 VERSION 1 6 6 C LIMITED DISTRIBUTION ssssssessessssssoseeeessssssesereesessssesroersessssoserrreesssss gt 42 2 15 1 Added or Changed Features Read 42 DAD Removed Feat te S oaa ASA E d e AA A AAA A 42 EE Ne EE 42 2 15 44 Configuration File Parameter Changes iicet dt eee Id ven RUE e dud Na n 42 2 10 VERSION 1 6 6 EE 42 2 16 1 Added or Changed Features is o iR A RES 42 2 10 2 UROIHOVOH P EUTANASIA AA E AA AA 42 LAIA
40. SIP 2 2 0 for details sip added feature 16 name feature 16 enabled Not currently used will be used in a future release sip added mb main idleTimeout See Administrator s Guide for SIP 2 2 0 for details sip added mb main statusbar See Administrator s Guide for SIP 2 2 0 for details sip added pnet role Not currently used will be used in a future release Page 12 Copyright 2007 Polycom Inc Release Notes SIP Application Changes Cfg File Action Parameter Description sip changed tone chord ringer 46 offDur 200 to o tone chord ringer 46 repeat 2 to Sch sip changed se pat ringer 12 inst 1 type silence to chord se pat ringer 12 inst 1 value 100 to 46 se pat ringer 12 inst 2 type chord to silence se pat ringer 12 inst 2 value 46 to 200 se pat ringer 12 inst 3 type silence to chord se pat ringer 12 inst 3 value 2000 to 46 se pat ringer 12 inst 4 type branch to silence se pat ringer 12 inst 4 value 2 to 2000 Note also added se pat ringer 12 inst 5 type branch and se pat ringer 12 inst 5 value 4 sip changed voice audioProfile G722 jitterBufferS hrink 500 to 1500 voice audioProfile G722 jitterBufferM ax 160 to 200 Audio performance tuning sip changed Several gain and other voice parameter
41. When SIP over TLS is configured the phone will send TCP Keep Alive messages to the SIP server every 30 seconds and will retry 3 times at 20 seconds before resetting RST the connection if no response is received 2 2 2 Removed Features None Page 4 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 2 3 Corrections 36557 When SRTP is enabled and so logging level is set to 1 the RTCP sender report displays encrypted values in the log file 37651 RTP Timestamp not updated correctly for silence packets 37690 Phone does not retry ACK when receiving duplicate 200 OK 37708 Phones fail SIP TLS registration when SNTP server is not configured 37851 SRTP phone doesn t include Crypto Suite in Group Pickup signaling 37873 Crypto line in answer does not have correct tag field 37878 Multiple crypto suites not handled when there is a re INVITE 37879 SRTCP packets have invalid authentication tags 37968 Phone with multiple lines using TLS not re registering on loss of connection 38110 Far end hears noise when an SRTP call is taken off hold with some SIP Servers 38249 SRTP lifetime value cannot be parsed correctly by the called party 38384 During a local SRTP conference a far end holding then resuming may result in one way audio or noise with some SIP servers 2 2 4 Configuration File Parameter Changes cfg File Action Parameter Description sip added sec srtp offer HMAC SHA1 80 If set to 1
42. add a new contact by selecting new entry on SoundPoint IP 330 320 phones e 38666 If a call is initiated in hands free mode and the Ringback Tone is server generated the far end user may experience echo when they answer the call If the originating phone is switched to handset mode and back to hands free mode the echo goes away Occurs on SoundPoint IP 330 320 430 550 650 phones e 38678 In a particular network configuration when using BLA the bridged line indication does not light up properly due to a missing NOTIFY from the phone 2 1 4 Configuration File Parameter Changes cfg File Action Parameter Description sip added tcplpApp keepalive tcp Sets the interval of the TCP keep idleTransmitlnterval alive packets sip added tcplpApp keepalive tcp Set the retransmission interval when noResponseTrasmitInterval the server fails to acknowledge the TCP keep alive sip added tcplpApp keepalive tcp sip tls Enables sending a TCP keep alive enable packet from the phone to the server The server is expected to respond with a TCP keep alive ack This is only used with TLS sessions sip added dir local readonly When set to 1 the contact directory cannot be changed and MACADDRESS directory xml is not uploaded sip added pres idleSoftKeys If set to 0 appearance of presence idle soft keys is disabled 2 2 Version 2 2 1 Limited Release 2 2 1 Added or Changed Features 38371
43. all is transferred to an invalid number it cannot be retrieved 17614 Setting the phone s own status through MyStat does not work properly 17868 Boot server password is displayed in Configuration menu if boot server is specified as a full URL including user name and password 17911 Per registration DND does not work on SoundPoint IP 430 17918 call enableOnNotRegistered parameter is not working correctly 17920 Incorrect icon displayed for offline status when using peer to peer presence 18078 When using an LCS server contacts cannot be added on the phone when the contact list is empty 18147 Expansion modules may display solid background if SoundPoint IP 601 or 650 has maximum number of registrations configured and maximum number of roaming buddies enabled 18198 Value of reg x callsPerLineKey parameter is not taken into account when additional calls are placed using hot static dialing 18297 VAD CNG Rx synthesis not working on SoundPoint IP 650 18333 Received data on any socket resets timeout of all sockets 18393 DTMF levels 3dB lower than configured level when RFC 2833 disabled 18501 Incoming call is sent to wrong line in some scenarios when the phone has an active call and reg x lineKeys gt 1 18688 Value of reg 1 callsPerLineKey parameter is not taken into account when two lines are configured and reg 2 callsPerLineKey is set to default and there is a call on hold on both lines 18772 SoundPoint IP 650 phone does
44. alls e 13497 Updated default daylight savings time rules e 13818 Added ability to disable message waiting indication on a line by line basis e 13882 Added support for setting RTP streams to inactive when on hold e 14485 Increased maximum number of digit map segments to 30 e 14733 Improved text entry efficiency in the microbrowser e 14740 Improved visibility of cursor in text entry fields of microbrowser Page 20 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 14759 Added microbrowser support to the SoundPoint IP 501 platform 14760 Added microbrowser support to the SoundPoint IP 430 platform 14900 Changed line seize subscription failure handling to be biased towards providing dial tone 15934 Added more low end dynamic range to volume control 16110 Added support for SoundPoint IP 550 platform 16515 Improved aresDnsLookup time out on socket select log message 16527 Added a debugging command to display cached DNS NAPTR records 17124 Added support for SYSLOG reporting of system status and errors 18434 Changed call timer clock display to have no leading colon 18966 Added support for adding phone serial number Ethernet address to user agent string in HTTP GET s used by microbrowser and modified format of user agent string used during provisioning process and used by microbrowser Example showing format of user agent in HTTP GET s previously User Agent Polycom Microbrowser 1 0 SIP 2 0 2 0
45. cation executables for SoundPoint IP 320 Version 2 2 2 0084 SIP application executables for SoundPoint IP 330 Version 2 2 2 0084 2345 11402 001 sip ld SIP application executable for SoundPoint IP 430 Version 2 2 2 0084 2345 11500 030 sip ld 2345 11500 040 sip ld SIP application executables for SoundPoint IP 501 Version 2 2 2 0084 2345 12500 001 sip ld SIP application executable for SoundPoint IP 550 Version 2 2 2 0084 2345 12560 001 sip ld SIP application executable for SoundPoint IP 560 Version 2 2 2 0084 2345 11600 001 sip ld 2345 11605 001 sip ld SIP application executable for SoundPoint IP 600 Version 2 2 2 0084 SIP application executable for SoundPoint IP 601 Version 2 2 2 0084 2345 12600 001 sip ld SIP application executable for SoundPoint IP 650 Version 2 2 2 0084 2201 06642 001 sip ld SIP application executable for SoundStationt IP 4000 Version 2 2 2 0084 000000000000 directory xml sip cfg main core and SIP configuration file phonet cfg example per phone SIP configuration sip ver Text file detailing build id s for the release 000000000000 cfg example master configuration file example per phone local contact directory XML file edit and then remove from name to seed phones which have no directory SoundPointIP dictionary xml dictionary files for multilingual support include no IP 30X support Chinese China for
46. cenarios phone adds itself to its own buddy list when using the LCS server 17976 NTLM signature should include full From URI 2 12 4 Configuration File Parameter Changes Cfg Action Parameter Description File sip removed call callWaiting prompt sip removed sec srtp offer sec srtp require sec srtp key lifetime sip added volpProt SIP pingInterval This parameter is used together with reg x proxyRequire It specifies the number of seconds between PING messages sent by the phone Default 0 disabled Possible range is 0 to 3600 Note Server support is required before this feature can be used sip added res finder minFree This parameter is used to ensure that the phone will not download resources which could leave it with insufficient memory to function correctly A resource will not be downloaded if the phone has less memory free than res finder minFree kBytes This parameter can have the values 1 to 2048 The recommended configuration file value is 1200 If the parameter is left empty the default is 800 Notes Setting this value too small may affect functionality of the phone Setting this value too large may mean that some resources are not downloaded at boot time Copyright 2007 Polycom Inc Page 31 Release Notes SIP Application Changes Cfg File Action Parameter Description phone added reg x proxyRequire This parameter is used together with v
47. d tcplpApp sntp gmtOffset overrideDHCP to 1 Page 58 Copyright 2007 Polycom Inc Release Notes SIP Application Notes e Reducing the power requirements reported via CDP for a SoundPoint IP 601 A new flash parameter EM Power is available under the Network Configuration menu of SoundPoint IP 601 phones If this is set to Enabled the phone will report power requirements of 12W which is sufficient to power three Expansion Modules If the parameter is set to Disabled the phone will report power requirements of 5W and no Expansion Modules can be connected to the phone By default this parameter will be set to Enabled when the phone is upgraded to 1 6 5 BootROM version 3 1 3 is required in order for the same power requirements to be reported at boot time Please refer to Tech Bulletin TB14052 for details on upgrade downgrade process with respect to this parameter 3 1 16 From Version 1 6 3 to 1 6 4 3 1 16 1 Mandatory Changes None 3 1 16 2 Optional Changes None 3 1 17 From Version 1 6 2 to 1 6 3 3 1 17 1 Mandatory Changes e Dialog event package draft backwards compatibility If the old dialog event package draft behavior is desired SDP is sent in dialog body set the new volpProt SIP dialog useSDP parameter in sip cfg to 1 3 1 17 2 Optional Changes e Changing the destination of phone specific override file uploads Use the new CONTACTS DIRECTORY and OVERRIDES DIRECTORY fields in 000000000000 cfg e Preventing IP a
48. dded menu options for setting Ethernet link mode on SoundPoint IP 601 16376 Improved response time of phone to SIP messages 16482 Added option for phone to be more assertive in negotiating the preferred codec 16500 Added configurable line seize behavior 2 14 2 Removed Features None 2 14 3 Corrections 16027 When connecting to voicemail in specific scenario phone may have no audio Page 40 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 16075 Phone plays re order tone when taking call off hold in specific scenario 16100 BLA line key status is not maintained in specific scenario 16116 Cannot register lines 7 to 12 from SIP configuration menu 16149 Line key LEDs for BLA lines can switch from one line key to another in specific scenario 16250 Comfort noise received by phone is handled incorrectly 16374 Phone keeps sending NOTIFY if 481 received in early NOTIFY 16388 Removed DC bias from Tx signal 16429 Web interface does not have configuration options for lines 7 to 12 16459 Phone is unable to park a call that is received via ACD final destination 16480 BLA Led gets stuck and there is a phantom NOTIFY from the phone in a particular scenario 16485 Notify Talk is ignored if interval between it and 180 is too brief 16565 Dialed digits can be lost if they are dialed too quickly after selecting an SCA line 16599 SoundPoint IP 300 and 301 phones reboot when using G 729
49. ddress caller ID display when PSTN caller is unknown The url dialing feature must be disabled in order for the IP address to be hidden 3 1 18 From Version 1 6 1 to 1 6 2 3 1 18 1 Mandatory Changes None 3 1 19 From Version 1 6 0 to 1 6 1 3 1 19 1 Mandatory Changes e Voice Configuration Parameters Updated Some parameters in the voice section of sip cfg have been modified and this entire section is required when using SIP 1 6 1 Copyright 2007 Polycom Inc Page 59 Release Notes SIP Application Notes 3 2 Outstanding Issues The following issues will be fixed in a subsequent release Note Polycom has switched to a different issue tracking system which has caused the reference numbers in these release notes to be different to earlier versions When the issues are addressed the numbers in this release note can be used to track in which version the issue is addressed 24398 No Layer 2 QoS support for signaling protocol TCP Workaround The default QOS parameters will still be used for TCP signaling packets and these may be specified in the sip cfg configuration file Layer3 QoS settings are supported 24805 Cannot answer an incoming call while directory is being saved Workaround None 26615 Subnet mask forces all packets through gateway when not using DHCP and when using the wrong subnet mask for the network class in use for example using 192 168 X X addresses with a 255 255 0 0 subnet mask Workaround Use the corr
50. ded reg x outboundProxy port Same interpretation as voipProt SIP outboundProxy port for registration x phone added reg x outboundProxy transport Same interpretation as voipProt SIP outboundProxy transport for registration x phone added attendant uri For attendant console 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 phone added attendant reg 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 phone added roaming buddies reg Specifies the line registration number which has roaming buddies support enabled Default is empty which means roaming buddies is disabled If value 1 then value is replaced with 1 This parameter is relevant for LCS server installations phone added roaming privacy reg Specifies the line registration number which has roaming privacy support enabled Default is empty which means roaming privacy is disabled If value 1 then value is replaced with 1 This parameter is relevant for LCS server installations 2 14 Version 1 6 7 2 14 1 Added or Changed Features 15930 Added ability to set Ethernet link mode on SoundPoint IP 601 15981 A
51. dentials as the INVITE which is a failure to comply with RFC 3261 15419 Blind transfer doesn t work for URL calling 15568 A comma in quotes in SIP address headers should be interpreted correctly 15596 Remote phone can force local conference host to resume call unexpectedly in specific scenario 15615 When a shared line call is on hold lifting the handset seizes the last used line instead of the first available line 14939 Shared line user must press Answer soft key twice to answer an incoming call in some scenarios 15907 After a reboot a phone may show 1 new missed call which can t be cleared until another call is missed 15982 The SDP session identifier should not be changed on each re INVITE 16021 FTP downloads may fail because incorrect timeouts are used 16141 Phone with a shared line loses hot dialed digits when remote shared line changes state such as placing an active call on hold Page 44 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 16161 Phone with a shared line displays the wrong soft key labels after attempting to hot dial when the remote shared line is in use 2 17 4 Configuration File Parameter Changes cfg Action Parameter Description File sip added call shared exposeAutoHolds call shared exposeAutoHolds 1 means that on a shared line when setting up a conference a re INVITE will be sent to the server call shared exposeAutoHolds 0 means no re INV
52. ding 0 don t allow transfer during consultation call proceeding state 1 do allow it 1 is the default sip added volpProt SIP outboundProxy transport Same function and possible values as existing volpProt server x transport parameter Default is DNSnaptr sip added voice gain rx analog chassis IP_601 Gains specifically for the IP 601 voice gain rx analog ringer IP_601 platform voice gain rx digital chassis IP_601 voice gain rx digital ringer IP_601 voice gain tx analog chassis IP_601 voice gain tx digital chassis IP_601 voice gain tx analog preamp chassis IP_601 sip changed voice aec xxx Changed parameter values Do not modify these sip changed voice ns xxx Changed parameter values Do not modify these sip added voice rxEq xxx This whole section has changed and removed must be used Do not modify these sip added voice txEq xxx This whole section has changed and removed must be used Do not modify these sip added log level change sotet Added log level control for logging log level change ttrs related to Expansion Module Copyright 2007 Polycom Inc Page 53 Release Notes SIP Application Notes 3 Notes 3 1 Upgrading This section lists the changes that should be made to configuration files when using the centralized boot server provisioning model For general guidelines see the Updating and Rebooting information in Section 4 3 of the Administrator Guide 3 1
53. e existing file causes that file to be deleted 14858 Phone reboots if idle micro browser is running and the Status Platform Application menu is displayed 15007 If the server address flash parameter is a URL which specifies a protocol and user name but not password the password in flash is not used 15101 Provisioning of phone stalled forever in specific scenario 15145 SAS VP feature does not work correctly when the filename parameter is empty 15154 Phone does not behave correctly when it is disconnected from the network and is using SAS VP 15185 Editing problems exist with long strings 15214 Headset memory indicator is not restored after adjusting volume on some platforms 15269 When tcplpApp sntp gmtOffset overrideDHCP is set but no override value is given the DHCP based offset is not applied 15351 Blind transfer does not drop unless server sends signaling to drop the call on the originator s phone Problem will occur in pure proxy scenarios only 15368 Character appears to be deleted during editing 15412 TFTP URL of configuration file name in log file may be truncated 15455 Phone should not reboot if parameters are missing from flash file system 15463 Phone s presence status is not displayed on Ul on SoundPoint IP 300 and 301 phones 15554 Problems with password entry for very long passwords 15561 Phone may reboot after entering a long incorrect password 15571 Phone cannot recover in several scenario
54. e holds the call when a fourth party is added to a centralized conference 11946 Some clock date format selections do not work 12032 Pressing headset button in ringing state does not answer call when headset memory is enabled 12066 After editing contact directory items the Save soft key can get relabeled as Search 12191 The menu produced when the Directories key is pressed should not include the Messages option 12221 1 displayed as number of different priority messages for voice message feature when data is missing 12227 Phone attempts to forward a call to a shared line if Auto Divert is enabled for the contact making the call 12247 Two stage dialing does not work Page 52 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 12284 Time handling for DHCP needs to be improved e 12289 Common audio equalization tables should be grouped together e 12323 Exiting Display Diagnostics with termination key does not stop display diagnostics e 12333 Direct and Group soft keys can appear when directed and group call pickup features are disabled e 12370 Ringing can be heard during a connected call mixed with audio when there is a high number of unanswered incoming calls e 12541 Error messages can appear in log file after putting two calls on hold 2 23 4 Configuration File Parameter Changes cfg Action Parameter Description File sip added volpProt SIP allowTransferOnProcee
55. e restricted to TCP This means the phone will not attempt to fail over to UDP if TCP fails Anew TCPOnly option has been added to all parameters which control the transport used by the phone e Adding sticky line seize behavior for hot dial on hook dialing If sticky behavior is desired for hot dialing this can be configured using the new call sticky AutoLineSeize onHookDialing parameter Hot dialing sticky behavior can be configured to be different than normal new call sticky behavior Stickiness refers to using the same line for a new call as the last used line when a call has been put on hold 3 1 8 From Version 2 0 3 to 2 0 3 B 3 1 8 1 Mandatory Changes None 3 1 8 2 Optional Changes None 3 1 9 From Version 2 0 2 to 2 0 3 3 1 9 1 Mandatory Changes None 3 1 9 2 Optional Changes None Page 56 Copyright 2007 Polycom Inc Release Notes SIP Application Notes 3 1 10 From Version 2 0 1 to 2 0 2 3 1 10 1 Mandatory Changes None 3 1 10 2 Optional Changes None 3 1 11 From Version 2 0 0 to 2 0 1 3 1 11 1 Mandatory Changes None 3 1 11 2 Optional Changes e Using template support in master configuration file The master configuration file may contain the string MACADDRESS This will be replaced with the MAC address of the phone For example the file 000000000000 cfg may refer to MACADDRESS phone cfg which will be replaced with something like 000412100137phone cfg This can make pro
56. ecommended that this value should not be modified The allowed range for this parameter is 5 to 512 and the default is 512 sip added usb enable This parameter enables or disables the USB port on the phone It can be set to 0 or 1 The Null default is 0 sip added usb bulkDrive enable This parameter enables or disables support for a USB bulk drive memory stick connected to the USB port on the phone It can be set to 0 or 1 The Null default is 0 sip added usb bulkDrive name This parameter is a string which specifies the name of the mounted USB drive The Null default is usbDrive sip changed dir local volatile maxSize For the SoundPoint IP 650 platform only prov fileSystem rfs0 minFreeSpace the values specified by these parameters ramdisk bytesPerBlock are replaced internally with double the res finder sizeL imit value This is because the SoundPoint IP res finder minFree 650 platform has 32 Mbytes of memory res quotas x value instead of 16 Mbytes mb limits nodes mb limits cache 2 10 Version 2 0 2 2 10 1 Added or Changed Features e 8428 Split call signaling processing from lamp management processing Copyright 2007 Polycom Inc Page 27 Release Notes SIP Application Changes e 18356 Emergency routing is not supported on shared lines 2 10 2 Removed Features None 2 10 3 Corrections The following issues have been resolved with this release e 6527 Shared line does not ring if incoming ca
57. ect subnet mask 26920 Centralized conference fails due to RTP port being slow to open in some cases Workaround None 27469 Local Conferencing on IP4000 phones is disabled if G 729 is in the Codec preference list Workaround Disable G 729 as a Codec option on the phone by setting voice codecPref IP_4000 G729AB 28508 Phone crashes after receiving high call rate 4 unanswered calls every 18 seconds Workaround Reduce the incoming call rate 29344 HTTP Digest Authentication does not work on IIS Workaround Use a different form of authentication a different protocol or a different server 29946 Log files are not uploaded if an Apache 2 0 X boot server requires authentication Workaround Turn off authentication or use version 1 3 3X of the Apache server 30086 Boot servers running explicit FTPS are not supported Workaround Use implicit FTPS or HTTPS 30371 Pattern generator for tones does not work well for the case of a single repeating chord Workaround Start the pattern with a short period of silence then the desired initial chord Loop back to the desired initial chord instead of the initial silence 30903 Packet Loss statistics jump if calls are transferred Workaround lf using the packet loss statistics for troubleshooting purposes make a note of the Packet Loss value after the transfer and apply a correction based on this to subsequent calculations Page 60 Copyright 2007 Polycom Inc Release Notes
58. ent call if using NTLM and received transfer from a non NTLM phone 33105 Hold does not work if selected just before a Conference is completed 33748 Web server has vulnerability to DOS attacks 33931 Not all keys on phone can be remapped to Null 34089 SoundPoint IP 430 phone keeps rebooting if a function key is remapped to null in the configuration files 34196 Phone keeps rebooting when SIP server address is not a fully qualified domain name and primary DNS server replies to queries with ICMP destination unreachable packets due to service being turned off and secondary DNS server is not configured with NAPTR and SRV entries for the SIP server 34237 Default directory file 000000000000 directory xml is not downloaded by the phone when the lt Ethernet address gt directory xml file does not exist on the boot server 34258 Log file is deleted when it reaches the configured size limit even though log render file upload append limitMode is set to stop 34271 SoundPoint IP 430 550 650 phones may reboot when microbrowser XHTML page contains combined FORM and TABLE elements 34460 Local directory file larger than 10kB is downloaded by phone once but on subsequent reboots the phone freezes 34578 Phones may crash when downloading a directory file which contains an empty contact field 34636 Call on a shared line may lose audio when cancelling a transfer after the far end has already cancelled a transfer or conference 34641
59. et ingress filtering for DoS suppression and VLAN filtering 36277 Added ability to delete the contact number entered in the Forward menu 36531 Updated all translation dictionary files to rename Services menu entry to Applications Removed Features 36079 Removed support for the SoundPoint IP 300 and 500 phones Corrections 24021 Call display gets corrupted in IP dialed call if caller presses a digit then puts call on hold 25744 Spaces go missing in text in microbrowser occasionally 26110 Volume level cannot be changed in audio diagnostics mode 26231 ACD login failure should cause busy tone to be played Copyright 2007 Polycom Inc Page 7 Release Notes SIP Application Changes 26389 Forward contact which has been disabled is not displayed after a reboot 26935 ACD icon not suppressed if feature is disabled in sip cfg but activated in phone1 cfg 27105 The idle browser occasionally displays when the menu is being updated 27958 Phone hears busy tone for 2 seconds after far end hangs up and another call is already in the incoming state and has triggered the call waiting alert 28419 Divert settings for lines 7 to 12 are not used 28503 When in the held state a shared line hears ring tone instead of call waiting tone when another call comes in 28570 Stuttered dial tone indicating voice mail waiting does not work on shared line 28622 Some UNICODE ranges are not properly mapped 28681 Forward i
60. f 2 2 0 for details phone1 added call missedCallTracking x enabled See Administrator s Guide for SIP 2 2 0 for details phone1 added call callWaiting ring See Administrator s Guide for SIP 2 2 0 for details 000000000000 added LICENSE DIRECTORY See Administrator s Guide for SIP 000000000000 added APP FILE PATH SPIP300 sip 21 2 10 CONFIG_FILES_SPIP300 phonet _212 cfg sip 212 cfg 2 2 0 for details These are samples of the new fields which can specify application images and configuration files for specific hardware platforms in this case the SoundPoint IP 300 See Administrator s Guide for SIP 2 2 0 for details 000000000000 added APP FILE PATH SPIP500 2 sip 21 2 ld CONFIG_FILES_SPIP500 phone1 _212 cfg sip 212 cfg 2 4 Version 2 1 2 2 4 1 Added or Changed Features These are samples of the new fields which can specify application images and configuration files for specific hardware platforms in this case the SoundPoint IP 500 See Administrator s Guide for SIP 2 2 0 for details e 35361 Added ability for parameters in Ethernet address gt cfg to be overridden by model or platform specific versions e 35969 Changed behavior of the select button or right arrow button in call lists and contact directory on SoundPoint IP 320 and 330 to give contact information instead of acting the same as the dial key e 36538 Added configurable failover behavior for authentication signaling to specify that the phone fi
61. from IdleDefault ro sip changed HEADSET MEM IP 300 indicator to use indicator 50 HEADSET MEM IP 500 indicator to use indicator 50 ind class 4 state 6 index from 48 to 50 Removed compiled in Polycom idle display indicator bitmap Changed due to rearrangement of other indicators sip changed ind anim IP_400 38 frame 1 bitmap from IdleDefault to ind anim IP_500 38 frame 1 bitmap from IdleDefault to ind anim IP_500 39 frame 1 bitmap from IdleDefault to ind anim IP 600 38 frame 1 bitmap from IdleDefault to ind anim IP 600 39 frame 1 bitmap from IdleDefault to ind anim IP_4000 38 frame 1 bitmap from IdleDefault to ind anim IP_4000 39 frame 1 bitmap from IdleDefault to Removed compiled in Polycom idle display indicator bitmap sip changed res quotas 1 value from 2000 to 600 Reduced default resource quota limits for tones phone added reg x lcs Default 0 Can be 0 or 1 If set to 1 the LCS server is supported for registration x phone added reg x server y expires overlap Same interpretation as volpProt server y expires overlap for registration x phone added reg x outboundProxy address Same interpretation as voipProt SIP outboundProxy address for registration x Copyright 2007 Polycom Inc Page 39 Release Notes SIP Application Changes cfg File Action Parameter Description phone ad
62. from call s contact header in refer to header Set the parameter volpProt SIP useContactInReferTo to 1 in sip cfg if the URI from the initial call s Contact header should be used in REFER s refer to header when setting up a transfer The previous and default behavior is to use the URI from the initial call s To header e Supporting G 729 Annex B SDP signalling per RFC 3555 If the new parameter voice vad signalAnnexB in sip cfg is set to 1 a new attribute Copyright 2007 Polycom Inc Page 55 Release Notes SIP Application Notes line will be added to SDP See details in 2 6 4 Configuration File Parameter Changes 3 1 7 From Version 2 0 3 to 2 1 0 3 1 7 1 Mandatory Changes e Using a Microsoft LCS Server It may be required to set the new parameters volpProt server x lcs in sip cfg and reg x server y lcs in phone1 cfg if the phone registers to a Microsoft LCS server 3 1 7 2 Optional Changes e Using inactive stream mode attribute when a call is put on hold The default behavior is for the sendonly stream mode attribute to be used when a call is put on hold This behavior can be changed to use the inactive attribute In order to configure this behavior the parameter volpProt SIP useSendonlyHold must be set to O e Digit map extension support The digit map can be configured to remove add or replace digits For details see Technical Bulletin 11572 e Restricting transport to TCP The transport used by the phone can b
63. h BLA to send re INVITE during conference setup 13315 Increased the maximum number of buddies to 8 for all platforms except SoundPoint IP 600 and 601 which can watch 48 buddies Copyright 2007 Polycom Inc Page 43 Release Notes SIP Application Changes 2 17 2 Removed Features None 2 17 3 Corrections The following issues have been resolved with this release 11658 Phone continues to append to log file on FTP boot server after that file has reached its configured size limit 12613 SoundPoint IP600 and 601 phones may establish a call with no audio after holding resuming and ending multiple calls 12949 If the phone s first line is a shared line and cannot obtain dial tone pressing the NewCall soft key does not activate the first available line 14673 Special characters such as o and are not accepted as part of the FTP or HTTP password 14968 If the phone reboots the app log size can increase past the size limit 15002 If the phone s first line is unregistered pressing the NewCall soft key does not activate another line 15127 Phone may have one way audio in a call after multiple transfers have been done 15218 If multiple contact header fields contain multiple expire values the phone does not always pick the lowest non zero value 15235 Phone will freeze if the SAS VP server becomes unavailable when the phone application is starting 15339 ACK lacks the same authorization cre
64. ile Parameter Changes EEN 15 2 5 VERSION DL L diet 16 Zod Added or Changed Features A es Ux PL ce ute el Ma edo 16 29 2 Removed Features A A AA AE AA e 16 2 943 Corrections A dior dc ud 16 2 5 4 Configuration File Parameter Changes eese eese eene enne enne 17 2 6 VERSION JT E 17 2 6 1 Added or Changed Features aie eui te e eite esl OU 17 2 6 2 Removed ENTES id I7 2 6 3 Eeer 17 2 6 4 Configuration File Parameter Changes esee esee nennen nennen nnns 19 2 7 VERSION 2 EE 20 2 7 1 Added PTA E CEE CAES A A A E 20 2 7 2 Rem ved Features ee daa 21 2 75 COIPCCHOBS ut basa AA IS bof AH EE 21 2 7 4 Configuration File Parameter Changes incita 23 2 8 VERSION 2 0 9 B sii eile week eR e d 24 2 8 1 Added or Changed Features a uud sace e el Bia A A OT ad caque en BS 24 2 8 2 Removed E EME A A ea 25 2 8 3 OVC CLOVIS usta OR A A oe LU QUIS A LUUD de A 25 Copyright 2007 Polycom Inc Page i Release Notes SIP Application Table of Contents 2 8 4 Configuration File Parameter Changes it a aida 25 29 MERSION MAU E A Er 23 2 9 1 Added OF EE 25 2 9 2 REMOVER FEVER DO A bU as 25 2 9 3 COPE CCE MMC 25 2 9 4 Configuration File Parameter Changes eese eese eene enne enne 26 2107 VERSION ZIDANE 27 2 10 1 Added or Changed Features edel 27 210 2 EE 28 210 3 CONFECCION E tee CGU Ced muet Na tu E 28 240 4 Configuration File Parameter Changes iue e etin ene e tue td nun endete ti ee ed eu eus 28 Pau VERSI
65. is not refreshed Note If an HTTP Refresh header is detected it will be respected even if this parameter is set to 0 The use of this parameter in combination with the Refresh HTTP header may cause the idle display to refresh at unexpected times sip removed volpProt SIP WM50 For selecting between Windows Messenger 4 7 and 5 0 no longer supported Page 38 Copyright 2007 Polycom Inc Release Notes SIP Application Changes Clg Action Parameter Description File sip removed Icl ml lang cpt x Removed the parameters used to configure Icl cpt the call progress tone localization menu Icl cpt menu x Icl cpt chord cp x y freq z feature 10 name cpt settings feature 10 enabled 1 In order to still use this feature the old configuration parameters should be added to the sip cfg file and a new parameter feature cpt enabled must be added and set to 1 sip changed tone chord ringer 46 offDur from 200 Changes to make ring type 12 work as to 0 expected tone chord ringer 46 repeat from 1 to 2 Settings for se pat ringer 12 sip changed voice gain tx digital chassis IP_430 Gain corrections for SoundPoint IP 430 from 3 to 0 platform voice handset txag adjust IP 430 from 24 to 21 sip changed bitmap IP 400 61 name from IdleDefault to bitmap IP_500 61 name from IdleDefault ro bitmap IP_600 65 name from IdleDefault to bitmap P_4000 66 name
66. istrator s Guide for SIP 2 2 0 for details Sip added log level change clink Not currently used will be used ina log level change pnetm log level change peer future release Copyright 2007 Polycom Inc Page 11 Release Notes SIP Application Changes cfg File Action Parameter Description Sip added sec srtp enable sec srtp leg enable sec srtp offer sec srtp require sec srtp key lifetime sec srtp mki enabled sec srtp sessionParams noAuth offe r sec srtp sessionParams noAuth req uire sec srtp sessionParams noEncrypR TP offer sec srtp sessionParams noEncrypR TP require sec srtp sessionParams noEncrypR TCP offer sec srtp sessionParams noEncrypR TCP require sec srtp sessionParams leg noAuth offer sec srtp sessionParams leg noAuth r equire sec srtp sessionParams leg noEncry pRTP offer sec srtp sessionParams leg noEncry pRTP require sec srtp sessionParams leg noEncry pRTCP offer sec srtp sessionParams leg noEncry pRTCP require sec srtp sessionParams IP 4000 no Auth offer sec srtp sessionParams IP 4000 no Auth require sec srtp sessionParams IP 4000 no EncrypRTP offer sec srtp sessionParams IP 4000 no EncrypRTP require sec srtp sessionParams IP 4000 no EncrypRTCP offer sec srtp sessionParams IP 4000 no EncrypRTCP require sec srtp leg allowLocalConf See Technical Bulletin 25751 for details sip added license polling time See Administrator s Guide for
67. ll 36948 On SoundPoint IP 320 and 330 phones if the Dial and Menu keys are pressed at the same time after entering digits from the idle display incorrect soft keys are displayed 36967 If the phone receives an INVITE with SDP which contains video information it returns a malformed response 37086 Phone ignores expiration date of CA certificate if SNTP is only set via DHCP 37632 Out of order SCA signaling can lead to improper handling of Shared Lines in some situations 37646 DNS SRV querying after A record cache makes registration fail 2 3 4 Configuration File Parameter Changes Cfg File Action Parameter Description sip added volpProt SIP csta Not currently used will be used in a future release sip added volpProt SIP serverFeatureControl d See Administrator s Guide for SIP nd 2 2 0 for details sip added volpProt SIP serverFeatureControl c See Administrator s Guide for SIP f 2 2 0 for details sip added up toneControl bass Not currently used will be used in a future release sip added up toneControl treble Not currently used will be used in a future release sip added up audioSetup auxlnput Not currently used will be used in a future release sip added up audioSetup auxOutput Not currently used will be used in a future release sip added up idleTimeout See Administrator s Guide for SIP 2 2 0 for details sip added se pat ringer 12 inst 5 type branch se pat ringer 12 inst 5 valuez 4
68. ll arrives when phone is playing dial tone then subsequently hangs up e 8542 Phone does not display second call appearance in specific bridged line scenario e 8547 Local ringback is not played if far end does blind transfer without going on hold e 15671 Pressing a line key of a shared line when a call is remote busy ends the call e 16662 Shared line can not establish a call if there are two simultaneous incoming calls e 18435 If two INVITE s come close together with SDP containing a ptime the phone will crash e 18471 Setting NAT IP address causes truncation or corruption of IP address in VIA e 18747 INVITE failover does not work 2 10 4 Configuration File Parameter Changes None 2 11 Version 2 0 1 B 2 11 1 Added or Changed Features None 2 11 2 Removed Features None 2 11 3 Corrections The following issues have been resolved with this release e 18358 Malformed RTCP packets can crash Cisco gateways 2 11 4 Configuration File Parameter Changes None Page 28 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 12 Version 2 0 1 The 2 0 1 Release includes all the changes and corrections from Releases 1 6 6 and 1 6 7 2 12 1 Added or Changed Features 8072 Added Nortel MCP NAT traversal parameters to config files 11678 Added template support in master configuration file 16399 Changed behavior when there is an incoming call on a phone idle dial digits are no longer clea
69. ll on headset and another incoming call is rejected 17206 Local conference host cannot end conference if both legs are put on hold by far ends 17242 Local conference host s state changes to held when second leg holds and invalid soft keys are displayed 17271 Phone will not accept a call with a codec with a dynamic payload identifier 17308 Phone displays In a meeting status as Away when using LCS server 17362 Add or edit directory speed dial contact crashes phone when configured for roaming buddies 17370 Phone may reboot if LCS server is used and presence is enabled without having roaming buddies enabled Note If the LCS server is used the roaming buddies parameter should be enabled 17457 Phone may display incorrect soft keys if a digit is pressed then Menu Directories or Messages is selected then de selected Page 30 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 17573 In some scenarios phone sends 603 Decline after 2 rings on SCA line 17639 Expansion Module updates should be continuously done in the background 17656 Phone does not handle outbound fragmented packets that are tagged for VLAN 17706 Phone may freeze after regaining connection with LCS server 17783 PRACK message goes directly between phones instead of via LCS server because of no record route 17797 In some scenarios phone sets its own presence status to Away when using the LCS server 17831 In some s
70. ng up a transfer sip added voice d in rv analog Chassis IP 330 New parameters to support SoundPoint voice gain rx analog ringer IP 330 IP 320 and 330 platforms which will be voice gain rx digital chassis IP 330 supported in a future software release Do voice gain rx digital ringer IP 330 not change these values voice gain tx analog chassis P 330 voice gain tx digital chassis IP 330 voice rxEq hs IP_330 preFilter enable voice rxEq hs IP_330 postFilter enable voice rxEq hd IP_330 preFilter enable voice rxEq hd IP_330 postFilter enable voice rxEq hf IP_330 preFilter enable voice rxEq hf IP_330 postFilter enable voice txEq hs IP_330 preFilter enable voice txEq hs IP_330 postFilter enable voice txEq hd IP_330 preFilter enable voice txEq hd IP_330 postFilter enable voice txEq hf IP_330 preFilter enable voice txEq hf IP_330 postFilter enable sip added voice vad signalAnnexB A new line can be added to SDP depending on the setting of this parameter and the voice vadEnable parameter Default behavior is the same as voice vad signalAnnexB 0 No change to the SDP voice vad signalAnnexB 1 If voice vadEnable 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 0 add attribute line a fmtp 18 annexb no below a rtpmap attribute line where 18 could be replaced by another payload Copyright
71. nged to short date format after first call 13223 All user agent headers for SAS VP v3 must include Ethernet address gt 13228 Audio lost for the first call after rejecting the second incoming call if headset or hands free is used 13235 Repeatedly holding and resuming a call can result in no audio when the call is resumed 13258 Frequent registration retry to an inactive server after server failover can result in the phone being unable to put a call on hold 13285 Unverified SSL connections were allowed to SAS VP server 13289 Long date format does not work if a shared line calls itself 13361 IP 4000 security certificate HTTPS and SAS VP provisioning can become corrupt after file system activity Note BootROM must be upgraded to version 3 1 2 as instructed in Technical Bulletin TB13361 13517 Hands free dial tone volume can become very quiet after significant volume adjustment Copyright 2007 Polycom Inc Page 49 Release Notes SIP Application Changes 2 20 4 Configuration File Parameter Changes cfg File Action Parameter Description 000000000000 added CONTACTS DIRECTORY New fields which can specify a directory on OVERRIDES DIRECTORY the boot server in which contact overrides Ethernet address directory xml and configuration overrides Ethernet address gt phone cfg should be stored sip added volpProt SIP dialog useSDP 0 or Null New dialog event package draft is used no SDP in dialog body
72. not show HD animation when a wide band call is transferred to it 18773 After a transfer a SoundPoint IP 650 phone may incorrectly display the HD animation when the call is no longer a wide band call 18785 After receiving a transferred call which is not a wide band call a SoundPoint IP 650 phone may incorrectly display the HD animation 18985 The log render level for the sip module cannot be changed 19113 Phone sends incorrect authorization header in some hold scenarios Page 22 Copyright 2007 Polycom Inc Release Notes SIP Application Changes e 19124 Setting codec preferences using web interface does not work correctly for SoundPoint IP 650 e 19252 Phone does not send a final NOTIFY to initiator of transfer if the phone cancels the transfer before it completes e 19292 SoundPoint IP 650 phone may freeze after restarting after configuration changed using one of the menus e 19427 Phone can display Cache bounced error message when submitting forms from the microbrowser e 19524 Problems resuming a call which is on hold on a remote bridged line for a specific SIP server e 19605 Phone may continue to send INVITE s in specific scenario if a call is initiated then ended but the SIP servers are not reachable e 19664 Phone may reboot in some scenarios with log file showing a Null pointer in a specific task e 19702 Receipt of a re transmitted invalid SIP ACK message may cause phone to reboot
73. oft key when editing Forward contact Copyright 2007 Polycom Inc Page 47 Release Notes SIP Application Changes 12626 Phone reboots on installation of a custom certificate 12882 Display of time and date on SoundStation IP 4000 gets truncated during a call if the line label is 10 digits long 13034 Phone should stop sending further NOTIFY messages if 481 response received 13318 SoundStation IP 4000 file system is smaller than it should be 13440 Changes in APP FILE PATH cause unnecessary application changes Note This fix requires bootROM version 3 1 2 13507 The phone at times incorrectly maintains two SUBSCRIBEs for call info 13533 The phone doesn t upload directory or configuration override files to a TFTP server unless they already exist on the server 13553 The entity field in a dialog for private lines can be improperly formatted 13554 A phone in the offering state should send a NOTIFY response to a dialog SUBSCRIBE request for all lines except Bridged Lines 13582 Supported header in INVITE should contain replaces instead of replace 13699 VLAN from CDP may work intermittently on SoundStation IP 4000 14116 After a blind transfer fails the call cannot be retrieved 14219 RTP sequence numbering starts at wrong value after a call is resumed from hold 14220 Lost packets statistics are incorrect after far end resumes a call 14387 A display name containing a is not displayed in some scenarios
74. oice txEq hf IP_650 postFilter enable settings which are specific to the SoundPoint IP 650 phone The values should not be modified Page 26 Copyright 2007 Polycom Inc Release Notes SIP Application Changes cfg Action Parameter Description File sip added voice handset rxag adjust P 650 These parameters control gain settings voice handset txag adjust IP 650 which are specific to the SoundPoint IP 650 voice handset sidetone adjust IP_650 phone The values should not be modified voice headset rxag adjust IP_650 voice headset txag adjust IP_650 voice headset sidetone adjust IP_650 sip added dir local volatile 8meg This parameter applies only to platforms with 8 Mbytes of flash memory It can be set to 0 or 1 and is 0 by default If set to 1 use volatile storage for phone resident copy of the directory to allow for larger size sip added dir local nonVolatile maxSize 8meg This parameter applies only to platforms with 8 Mbytes of flash memory It can be set from 1 to 100 The units are Kbytes and the default is 100 This is the maximum size of non volatile storage that the directory will be permitted to consume sip added log level change usb This parameter is used to set the logging detail level for the usb module sip added prov fileSystem ffs0 8meg minFreeSpac The minimum free space in Kbytes to e reserve in the file system when downloading files from the boot server It is r
75. olpProt SIP pinglnterval It specifies the string which is put in the Proxy Require header Default is an empty string which means no Proxy Require will be sent Note Server support is required before this feature can be used phone added nat keepalive interval This parameter is used to set the interval in seconds 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 set up initially Default value is 0 which means the feature is disabled The allowable range is 0 to 3600 2 13 Version 2 0 0 Beta Release Only Note The 2 0 0 Release does not include the changes and corrections from SIP releases 1 6 6 and 1 6 7 2 13 1 Added or Changed Features 2236 Added support for TLS protocol 2307 When the phone reboots due to a fatal error it should first log any useful information 5403 Added support for the NTLM authentication protocol 5404 Added support for Microsoft Live Communications Server authentication schemes 8817 Added support for BLF SCA mode 9110 Added support for platform specific override strings in dictionaries to allow abbreviated strings for certain platforms 9734 Added option to select which registration to use for presence signaling 11646 Added IP QoS support for DSCP DiffServ 11785 Added support for multiple redundant provisioning servers 12270 SIP re registration inter
76. one wishing to use this feature for inter operability testing should contact Polycom to receive technical bulletin 25751 for details on how to enable this feature e This is the first GA release to support the SoundPoint IP 560 product platform 1 2 System Requirements Platform BootROM version SoundPoint IP 301 2 6 1 or greater SoundPoint IP 320 3 2 3RevB or greater SoundPoint IP 330 3 2 3RevB or greater SoundPoint IP 430 3 1 3 or greater SoundPoint IP 501 2 6 1 or greater SoundPoint IP 550 3 2 3 or greater SoundPoint IP 560 4 0 1 or greater SoundPoint IP 600 2 6 1 or greater SoundPoint IP 601 3 1 0 or greater SoundPoint IP 650 3 2 2RevB or greater SoundStation IP 4000 3 1 2 or greater Copyright 2007 Polycom Inc Page 1 Release Notes SIP Application General 1 3 Distribution Files The following files constitute the 2 2 2 distribution of the SoundPoint SoundStation IP SIP application For centrally provisioned systems copy these files to the boot server maintaining the folder hierarchy present in the zip file Some of the configuration files must be modified Refer to the Administrator Guide for details Files Description sip ld Concatenated SIP application executable Version 2 2 2 0084 for all platforms 2345 11300 010 sip Id SIP application executable for SoundPoint IP 301 Version 2 2 2 0084 2345 12200 002 sip ld 2345 12200 005 sip ld 2345 12200 001 sip ld 2345 12200 004 sip ld SIP appli
77. or tones 16047 Added configurable ms forking support and reject IM when it is enabled 2 13 2 Removed Features 12109 Removed configuration parameters for localized call progress tones menu In order to still use this feature see details in 3 1 Upgrading 13447 Removed presence and IM support for Windows Messenger 4 6 4 7 and 5 0 12350 Removed compiled in Polycom idle display indicator bitmap 2 13 3 Corrections The following issues have been resolved with this release 6078 Cannot adjust the volume of the reorder tone when attempting to seize a shared line which is remotely active 7084 Transducer indicator is not cleared after blind transfer on some platforms 9292 IP 4000 reboots upon downloading a wave file with a path containing Y instead of 9709 RTCP not sent or received when calls are on hold 9815 SoundStation IP 4000 cannot change language after already changing language 10 to 12 times 11177 Fast Busy sound effect sequencing wrong in specific scenario when call on hold 11588 The local contact directory feature cannot be disabled 11952 If destination phone rejects a blind transferred call the far end does not hear a busy tone 12020 Bridged line with multiple line keys may have one line indicator left in the remote active state if a peer bridged line hosts a centralized conference 12043 Label of CPU Load graph does not change when DSP load is displayed 12106 Address of boot server is truncated
78. pt 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 This may fail due to glare issues in which case the phone may select a different available line for the call Null default 0 disabled this was the behavior in SIP 1 6 6 2 15 Version 1 6 6 C Limited Distribution 2 15 1 Added or Changed Features None 2 15 2 Removed Features None 2 15 3 Corrections e 16250 Comfort noise received by phone is handled incorrectly Fixed for SoundPoint IP 300 301 500 501 600 and 601 phones e 16388 DC bias should be removed from Tx signal on SoundPoint IP 300 301 500 501 600 and 601 phones 2 15 4 Configuration File Parameter Changes None 2 16 Version 1 6 6 B 2 16 1 Added or Changed Features e Add Support for SoundPoint IP 430 hardware platform 2 16 2 Removed Features None 2 16 3 Corrections None Page 42 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 16 4 Configuration File Parameter Changes cfg Action Parameter Description File sip added voice gain rx analog chassis IP 430 New gain parameters for SoundPoint IP 430 voice gain rx analog ringer IP_430 platform voice gain rx digital chassis IP_430 voice gain rx digital ringer IP_430 voice gain tx analog chassis IP_430 voice gain tx digital chassis IP_430 voice gain tx analog preamp chassis IP _ 430
79. r based DND CF features 31840 Set transfer time out for image file download to worst case scenario 32259 Added microbrowser support for recognizing mime types 32648 Reformatted call list entries 33616 Added configuration option for default transfer type for SoundPoint IP 320 and 330 phones 33748 Improved resistance to denial of service attacks aimed at phone s web server 34131 Changed URL dialing terminology from Name to URL 34434 Implemented 300Hz high pass transmit filter to reduce low frequency noise noise creates problems in some network line echo cancellers This can be enabled or disabled 34573 Added support for re establishing a TLS connection if the connection closes 34625 Added ability to discover provisioning server address using DHCPINFORM 34651 Added phone serial number MAC address to user agent string HTTP Gets 34685 Renamed Services menu entry to Applications 34705 Added support in microbrowser for form functionality when embedded in tbody or out of tbody 34707 Added low delay handset acoustic echo canceller for SoundPoint IP 320 330 430 550 and 650 phones This can be enabled or disabled 34874 If all DNS servers are found to be unreachable the phone suppresses DNS queries for 5 minutes as per RFC 2308 Sec 7 1 34998 Increased maximum number of registrations on SoundPoint IP 650 phones to 34 35039 Pressing Exit soft key when using the microbrowser should return user to
80. red when an incoming call is received 16645 Added support for NAT keep alive 17412 Added ability to set Ethernet link mode to SoundPoint IP 430 17413 Added ability to set Ethernet link mode to SoundStation IP 4000 2 12 2 Removed Features 14275 call callWaiting prompt has no effect This parameter has been removed from the configuration files because it is no longer used 2 12 3 Corrections The following issues have been resolved with this release 7723 Name of net logging module is sometimes corrupted in log file 12337 Display of SoundPoint IP 430 flickers under fluorescent lights and may be shifted vertically by a few pixels 12382 The phone will freeze if the DNS server address is all zeroes and the phone uses a FQDN server name 12647 Feature keys cannot be reconfigured to perform other functions 12749 Phone locks up during CERT PROTOS testing 15138 Text in line labels on SoundPoint IP 430 should be moved one pixel left 15227 Phone model of SoundPoint IP 430 is incorrect in CDP packets 15311 Contrast adjustment range on the SoundPoint IP 430 is unsuitable 15729 Phone does not retry connecting to boot server in specific scenario 15731 Phone should use Office Communicator model to update LCS presence status when multiple endpoints share same registration 15812 Phone doesn t handle simultaneous 200 OK and CANCEL race condition 16069 When using Russian dictionary phone reboots after exiting the DHCP Menu 16073
81. rst retries a SIP transaction with the server that has just sent a 401 or 407 response Uses new parameters volpProt SIP authOptimizedInFailover and or reg x auth optimizedlnFailover e 36647 Added configurable option allowing message waiting indicator to be displayed although voicemail cannot be accessed Uses new parameter up mwiVisible e 36681 Added logging of version information for configuration files 2 4 2 Removed Features None 2 4 3 Corrections e 34899 Phone may continuously reboot if a configuration change is made then power is disconnected and the provisioning server is unavailable Page 14 Copyright 2007 Polycom Inc Release Notes SIP Application Changes e 35873 Registration expiry period is limited to 65535 seconds e 35914 Scheduled logging stops after 99 days e 35961 Cannot use call group directed pickup on SoundPoint IP 320 and 330 phone while a call is incoming or the phone is off hook e 35974 SoundPoint IP 320 and 330 phones do not show status for watched contacts until after the next reboot e 35979 SoundPoint IP 320 and 330 phones reboot while trying to use call pickup on a remote hold BLA call e 36011 After changing termination while in a local conference the first time the volume is adjusted the volume slider shows minimum e 36044 Downloadable character sets are not working correctly in certain scenarios e 36053 On SoundPoint IP 320 and 330 phones Add and Delete soft keys
82. s None 2 5 3 Corrections The following issues have been resolved with this release Page 16 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 35913 SoundPoint IP430 550 650 phones may reboot while in a call under certain network conditions 2 5 4 Configuration File Parameter Changes None 2 6 Version 2 1 1 2 6 1 Added or Changed Features e 33263 Added support for G 729 Annex B SDP signalling per RFC 3555 Note New parameter voice vad signalAnnexB has been added to support this e 35268 Added support for 16 levels of gray on the LCD of SoundPoint IP 550 and 650 phones e 35643 Added support for new SoundPoint IP 320 and 330 phones in the configuration files to allow easier addition of these phones in a future software release 2 6 2 Removed Features None 2 6 3 Corrections The following issues have been resolved with this release e 32273 Failure of call park action results in a dropped call e 32609 Heavy call volume may cause phone to reject calls due to resource depletion e 33390 35392 35482 Voice activity detection VAD comfort noise generation CNG packets can be discarded by the jitter buffer or interpreted as out of order packets which may result in delayed receive audio when the G 729B codec is in use e 33586 The To URI is used in a refer to header instead of the contact URI Note New parameter volpProt SIP useContactInReferTo has been added to sip cfg to control the so
83. s have been changed The entire gain section in sip cfg must be updated Failure to do this will affect the audio performance of the phone voice rxEq hd IP 650 preFilter enabl sip changed e 1 to 0 voice txEq hs IP_650 preFilter enabl e 1 to 0 voice txEq hd IP_650 preFilter enabl e 1 to 0 voice txEq hf IP_650 preFilter enabl e 1 to 0 Audio performance tuning sip changed voice handset txag adjust IP_430 2 4 to g voice handset sidetone adjust IP_43 0 13 to o Audio performance tuning sip changed Multiple parameters in the ind anim xxx ind class xxx and ind gi xxx sections The entire indicator section in sip cfg must be updated Failure to do this will affect the appearance of the display sip changed res finder minFree 1200 to 600 sip sip removed removed ind anim xxx parameters from CTX_CUSTOM1 to CTX_CUSTOM8 and CTX_UNASSIGNED for all platforms usb enable These parameters were not used These parameters were not used usb bulkDrive enable usb bulkDrive name phone added reg x csta Not currently used will be used in a future release Copyright 2007 Polycom Inc Page 13 Release Notes SIP Application Changes Cfg File Action Parameter Description phone1 added reg x serverFeatureControl dnd See Administrator s Guide for SIP reg x serverFeatureControl c
84. s involving transferring mixed URL and E 164 calls 15603 The sip field name which appears when using IP dialing should not be deletable Page 36 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 15679 Ring Type 12 Ringback style sounds incomplete after the first ring 15694 Phone crashes and reboots when Exit is pressed from Network Configuration menu in Korean Language 15730 If a menu is displayed when a call is missed on the SoundPoint IP 300 and 301 phones the missed call count is not updated on the idle display 15766 Display is incorrect after selecting name dialing then entering and exiting a call list while dial tone is playing 15781 After putting a local conference on hold then splitting the calls then joining them the first call may remain on hold 15855 In the Instant Msg menu of the SoundPoint IP 300 and 301 phones x Ascii is not displayed after pressing the 1 A a softkey 2 13 4 Configuration File Parameter Changes cfg File Action Parameter Description sip sip added added volpProt server x expires overlap volpProt SIP ms forking The number of seconds before the expiration time returned by server x at which the phone should try to re register The phone will try to re register at half the expiration time returned by the server if that value is less than the configured overlap value Default 60 Minimum 5 maximum 65535
85. s not removed from menu when function disabled 29014 Cannot edit the local directory on the phone if the file is corrupt on the server 29358 Phone may crash if the specified DNS server is down and an invalid SNTP address is configured 29470 Cursor is in wrong position when performing a factory reset on the SoundPoint IP 301 phone 29573 Phone may freeze if a DNS server address is all zeroes 29966 Phone may reboot if incorrect information is entered in the menu for custom CA certificate 30880 Phone may crash when editing a server address which is 255 characters long 30902 Auto reject or divert settings changed in a contact after entering contact directory by pressing and holding a speed dial line key are not correctly displayed when next pressing and holding that speed dial line key 31019 There is no confirmation pop up message after choosing to reset the local security key 31326 Transferring a call to windows messenger or office communicator may leave the phone in a frozen state 31886 Remote resume does not work on BLA line when call between two other phones sharing the same line has been put on hold 31994 Trying to delete a null unicode character in the contact list causes the phone to crash Page 8 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 32179 When SAS VP provisioning is used the boot server password is visible in the application log file 32816 Phone may crash on subsequ
86. t 2007 Polycom Inc
87. telephony application Page 6 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 2 3 2 2 3 3 35040 Added configurable timeout parameter to allow microbrowser to return to telephony application after a period of inactivity in the microbrowser 35043 Added configurable option to display or hide browser status messages in microbrowser 35087 Changed boot up behaviour so that idle browser only starts about 2 minutes after the phone has booted up this is to optimize memory use 35099 Added support for TLS transport to Syslog 35199 Improved some translations in Norwegian XML dictionary file 35296 Added support for managing TLS custom certificates via the configuration file system 35311 Added support for specifying different versions of the application executable and configuration files in the Ethernet address gt cfg file on the boot server 35372 Pressing the Exit function key on the SoundStation IP 4000 phone when using the microbrowser should return user to telephony application 35373 Changed appearance of soft keys when running microbrowser so that they look the same as when running the telephony application 35419 Added user interface for configuring no answer and busy forwarding behavior 35481 Added support for Backlit Expansion Module 35507 Adding configuration parameter to control the timeout back to the idle display after a period of inactivity in a menu 36030 Implemented Ethern
88. ters Default is Null which means the other qos ip rtp parameters will be used Possible values are 0 to 63 EF AF11 AF12 AF13 AF21 AF22 AF23 AF31 AF32 AF33 AF41 AF42 or AF43 sip added qos ip callControl dscp This parameter allows the DSCP of packets to be specified If set to a value this will override the other qos ip callControl parameters Default is Null which means the other qos ip callControl parameters will be used Possible values are 0 to 63 EF AF11 AF12 AF13 AF21 AF22 AF23 AF31 AF32 AF33 AF41 AF42 or AF43 sip added pres reg Default 1 Can be 1 2 8 Must be a valid line registration number If the number is not a valid line registration number it is ignored Specifies the line registration number used to send SUBSCRIBE for presence sip added mb idleDisplay home mb idleDisplay home can be empty or any fully formed valid HTTP URL Length up to 255 characters Default is empty This specifies the URL used for the microBrowser idle display home page Example http www example com xhtml frontpage cgi page home If empty there will be no micro Browser idle display feature sip added mb idleDisplay refresh Can be 0 or an integer greater than 5 Values from 1 to 4 will be ignored and 5 will be used instead Default 0 This specifies the period in seconds between refreshes of the microBrowser idle display content 0 the idle display microBrowser
89. the SAS VP xml response has a blank or missing contactaddr element the phone does not use the username field for the contact address and may lock up during reboot 14510 The username field in a SAS VP xml response is not used as the SIP login name for authentication of SIP messages 14557 The SAS VP key is cleared if the user chooses the Reset Device Settings menu option 14634 Blind transfer fails with certain devices due to NOTIFY behavior 14684 Problems with text entry interface in custom certificate installation display 14805 Shared lines behave incorrectly if the line registration contains a 14935 Phone begins to ring when there is no incoming call in specific shared line scenario 15104 SoundStation IP 4000 CDP does not advertise new link duplex levels correctly 15122 Time displayed on phone changes from correct to incorrect shortly after a reboot in some scenarios Page 46 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 15162 Phone clears application log file during a warm boot even if the upload to the boot server failed 2 18 4 Configuration File Parameter Changes cfg File Action Parameter Description sip added volpProt server dhcp available 1 check with the DHCP server for SIP server IP address 0 do not check with DHCP server Default 0 sip added volpProt server dhcp option Option to request from the DHCP server if
90. urce of the URI used in the refer to header e 33647 The phone may reboot because it detects a suspended task even though that task may have been suspended intentionally e 33967 An error message is logged if a daylight savings time DST start or stop time of 0 12am is selected although the selection is correctly used e 34325 Microbrowser display is closed when shared line is opened on other phone e 34431 When changing the configuration of a phone via the web interface the phone may lock up Copyright 2007 Polycom Inc Page 17 Release Notes SIP Application Changes 34443 A remote on hold call on a line is not picked up by the first press of the line key with some SIP servers 34508 In a G 729 call SoundPoint IP 50X and 60X phones may reboot with a DSP assertion failure This problem is more likely in conference calls and can be reliably reproduced within 20 minutes of the call start 34723 RTCP transmission interval is not consistent with industry norms 34772 The value of the DLSR field in RTCP sent by the phone can be wrong by up to about one second 34827 There are two places to configure the microbrowser from the phone web server 34882 The configuration page on the phone web server has two Event 2 entries in the Global Log Level Limit drop down list 34906 NOTIFY request without dialog content an empty NOTIFY request such as you would get with a subscription renewal when the line is idle does not
91. val is now configurable 12419 Added support for Broadsoft attendant console BLF feature 12426 Added support for peer to peer calls using Microsoft Live Communications Server 2005 Page 32 Copyright 2007 Polycom Inc Release Notes SIP Application Changes 12427 Added support for calling to and from Windows Messenger 5 1 and Office Communicator using Microsoft Live Communications Server 2005 12938 Added caching of the state of the message waiting indicator LED across controlled reboots 13038 Changed DNS Lookup name to Transport in SIP Configuration menu and on web interface to match parameter name in sip cfg 13080 Added new consultative transfer behavior so that transfer automatically completes when originator hangs up 13100 Added support for individual configuration of secondary dial tone 13315 Increased the maximum number of buddies to 8 for all platforms except SoundPoint IP 600 and 601 which can watch 48 buddies 13317 Increased speed dial menu size limit to 99 for all platforms 13463 Added IM support with Office Communicator and Windows Messenger 5 1 in Microsoft Live Communications Server 2005 context 13509 Added support for reg x address configuration parameter to contain host part 13552 Improved boot up logging 13613 Improved support for multiple m lines in SDP 13813 Added the ability for file transfers to attempt to contact multiple IP addresses per DNS name 13893 Re enabled idle micro
92. visioning more efficient e Adding Nortel MCP NAT traversal The new parameters volpProt SIP pingInterval and reg x proxyRequire should be configured if this feature is needed e Adding NAT keepalive If NAT keepalive is required the new parameter nat keepalive interval should be set to a non zero value 3 1 12 From Version 1 6 7 to 2 0 0 3 1 12 1 Mandatory Changes e Using the phone s menu to select call progress tones This feature has been removed from the default configuration of the phone In order to still use this feature the old configuration parameters should be added to the sip cfg file and a new parameter feature cpt enabled must be added and set to 1 Old configuration parameters are feature 10 name cpt settings feature 10 enabled 1 and the entire localization multilingual language callProgTones section and the entire localization callProgTones section 3 1 12 2 Optional Changes e Adding IP QoS support for DSCP DiffServ Add the parameters qos ip rtp dscp and qos ip callContol dscp for DSCP A valid value is either a number or string as follows 1 Any number from 0 to 63 2 EF 3 Any of AF11 AF12 AF13 AF21 AF22 AF23 AF31 AF32 AF33 AF41 AF42 AF43 Copyright 2007 Polycom Inc Page 57 Release Notes SIP Application Notes The rules are 1 When qos ip rtp dscp has a valid value then it overrides the following i qos ip rtp min delay ii qos ip rtp max throughput iil
93. y entry is added the speed dial index is automatically assigned the next available value 11731 Calls from more than one SIP registration line can be joined Copyright 2007 Polycom Inc Page 51 Release Notes SIP Application Changes 11849 Added support for transfer dispatch during consultation call proceeding state New parameter for this is volpProt SIP allowTransferOnProceeding which will normally not need to be changed 12093 Added a Forward menu so that forwarding can be modified at any time 2 23 2 Removed Features None 2 23 3 Corrections The following issues have been resolved with this release 7521 Transfer from a shared line can be interrupted 8507 Directory search does not produce all matches for some last names 9790 Outbound proxy transport selection should be clear New parameter for this is volpProt SIP outboundProxy transport 9827 A keypad initiated reboot waits for dial tone to time out before starting 11583 Phone does not upload log file when it exceeds render file size 11738 Audio Diagnostics don t work for headset mode 11762 Headset indicator icon can blink during a call between two phones using the same bridged line which have headset memory enabled 11790 Multi tap entry doesn t work for the very first character entered for URL dialing 11846 484 response should be treated as an error in ringback state 11848 No stuttered dial tone when a line has a message waiting 11940 Phon

Polycom 3725-23487-003/A Computer Hardware User Manual

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