User manual POLYCOM SOUNDPOINT IP 501 - Release Note

Release Notes SIP Application SoundPoint and SoundStation IP Version 1.6.7 7 July 2006 ® ® Part Number 3804-11530-167 Copyright © 2006 Polycom, Inc. All rights reserved. Release Notes - SIP Application Copyright © 2006 Polycom, Inc. All rights reserved. Release Notes - SIP Application Table of Contents Table of Contents 1. 2. GENERAL................................................................................................................................... 1 1.1 SYSTEM REQUIREMENTS........................................................................................................ 1 CHANGES................................................................................................................................... 1 2.1 VERSION 1.6.7 ....................................................................................................................... 1 2.1.1 Added or Changed Features ......................................................................................... 1 2.1.2 Removed Features......................................................................................................... 1 2.1.3 Corrections ................................................................................................................... 1 2.1.4 Configuration File Parameter Changes ....................................................................... 2 2.2 VERSION 1.6.6 C (LIMITED DISTRIBUTION) ........................................................................... 3 2.2.1 Added or Changed Features ......................................................................................... 3 2.2.2 Removed Features......................................................................................................... 3 2.2.3 Corrections ................................................................................................................... 3 2.2.4 Configuration File Parameter Changes ....................................................................... 3 2.3 VERSION 1.6.6 B .................................................................................................................... 3 2.3.1 Added or Changed Features ......................................................................................... 3 2.3.2 Removed Features......................................................................................................... 3 2.3.3 Corrections ................................................................................................................... 3 2.3.4 Configuration File Parameter Changes ....................................................................... 4 2.4 VERSION 1.6.6 ....................................................................................................................... 4 2.4.1 Added or Changed Features ......................................................................................... 4 2.4.2 Removed Features......................................................................................................... 5 2.4.3 Corrections ................................................................................................................... 5 2.4.4 Configuration File Parameter Changes ....................................................................... 6 2.5 VERSION 1.6.5 ....................................................................................................................... 6 2.5.1 Added or Changed Features ......................................................................................... 6 2.5.2 Removed Features......................................................................................................... 7 2.5.3 Corrections ................................................................................................................... 7 2.5.4 Configuration File Parameter Changes ....................................................................... 8 2.6 VERSION 1.6.4 ....................................................................................................................... 8 2.6.1 Added or Changed Features ......................................................................................... 8 2.6.2 Removed Features......................................................................................................... 8 2.6.3 Corrections ................................................................................................................... 8 2.6.4 Configuration File Parameter Changes ....................................................................... 9 2.7 VERSION 1.6.3 ....................................................................................................................... 9 2.7.1 Added or Changed Features ......................................................................................... 9 2.7.2 Removed Features....................................................................................................... 10 2.7.3 Corrections ................................................................................................................. 10 2.7.4 Configuration File Parameter Changes ..................................................................... 11 2.8 VERSION 1.6.2 ..................................................................................................................... 11 2.8.1 Added or Changed Features ....................................................................................... 11 2.8.2 Removed Features....................................................................................................... 11 2.8.3 Corrections ................................................................................................................. 11 2.8.4 Configuration File Parameter Changes ..................................................................... 11 2.9 VERSION 1.6.1 ..................................................................................................................... 11 Copyright © 2006 Polycom, Inc. Page i Release Notes - SIP Application Table of Contents 2.9.1 Added or Changed Features ....................................................................................... 11 2.9.2 Removed Features....................................................................................................... 12 2.9.3 Corrections ................................................................................................................. 12 2.9.4 Configuration File Parameter Changes ..................................................................... 12 2.10 VERSION 1.6.0 ..................................................................................................................... 12 2.10.1 Added or Changed Features ....................................................................................... 12 2.10.2 Removed Features....................................................................................................... 13 2.10.3 Corrections ................................................................................................................. 13 2.10.4 Configuration File Parameter Changes ..................................................................... 14 2.11 VERSION 1.5.2 ..................................................................................................................... 14 2.11.1 Added or Changed Features ....................................................................................... 14 2.11.2 Removed Features....................................................................................................... 15 2.11.3 Corrections ................................................................................................................. 15 2.11.4 Configuration File Parameter Changes ..................................................................... 16 2.12 VERSION 1.5.1 ..................................................................................................................... 16 2.12.1 Added or Changed Features ....................................................................................... 16 2.12.2 Removed Features....................................................................................................... 17 2.12.3 Corrections ................................................................................................................. 17 2.12.4 Configuration File Parameter Changes ..................................................................... 19 3. NOTES ....................................................................................................................................... 21 3.1 DISTRIBUTION FILES ............................................................................................................ 21 3.2 UPGRADING ......................................................................................................................... 22 3.2.1 From Version 1.6.6 to 1.6.7 ........................................................................................ 22 3.2.2 From Version 1.6.5 to 1.6.6 ........................................................................................ 22 3.2.3 From Version 1.6.4 to 1.6.5 ........................................................................................ 23 3.2.4 From Version 1.6.3 to 1.6.4 ........................................................................................ 23 3.2.5 From Version 1.6.2 to 1.6.3 ........................................................................................ 23 3.2.6 From Version 1.6.1 to 1.6.2 ........................................................................................ 24 3.2.7 From Version 1.6.0 to 1.6.1 ........................................................................................ 24 3.2.8 From Version 1.5.2 to 1.6.0 ........................................................................................ 24 3.2.9 From Version 1.5.1 to 1.5.2 ........................................................................................ 24 3.3 OUTSTANDING ISSUES.......................................................................................................... 25 4. REFERENCE DOCUMENTS................................................................................................. 27 Page ii Copyright © 2005 Polycom, Inc. Release Notes - SIP Application General 1. General These release notes apply to version 1.6.7 of the SoundPoint IP SIP application. For more information, refer to the documents listed in Section 4. 1.1 System Requirements Platform SoundPoint IP 300 SoundPoint IP 301 SoundPoint IP 430 SoundPoint IP 500 SoundPoint IP 501 SoundPoint IP 600 SoundPoint IP 601 SoundStation IP 4000 BootROM version 2.6.1 or greater 2.6.1 or greater 3.1.3 or greater 2.6.1 or greater 2.6.1 or greater 2.6.1 or greater 3.1.0 or greater 3.1.2 or greater 2. Changes 2.1 Version 1.6.7 2.1.1 Added or Changed Features · · · · · 15930: Added ability to set Ethernet link mode on SoundPoint IP 601 15981: Added 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 2.1.2 Removed Features None. 2.1.3 Corrections · 16027: When connecting to voicemail in specific scenario, phone may have no audio Copyright © 2006 Polycom, Inc. Page 1 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 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 voIpProt.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 2.1.4 Configuration File Parameter Changes .cfg Action File sip added Parameter voIpProt.SDP.answer.useLocalPreferences Description 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. Page 2 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application .cfg Action File sip added Changes Description 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 attempt to initiate a new outgoing call on the same SIP line that is currently in focus on the LCD (this was the behavior in SIP 1.6.5). 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). Parameter call.stickyAutoLineSeize 2.2 Version 1.6.6 C (Limited Distribution) 2.2.1 Added or Changed Features None. 2.2.2 Removed Features None. 2.2.3 Corrections · · 16250: Comfort noise received by phone is handled incorrectly. Fixed for SoundPoint IP 300, 301, 500, 501, 600 and 601 phones. 16388: DC bias should be removed from Tx signal on SoundPoint IP 300, 301, 500, 501, 600 and 601 phones 2.2.4 Configuration File Parameter Changes None. 2.3 Version 1.6.6 B 2.3.1 Added or Changed Features · Add Support for SoundPoint IP 430 hardware platform 2.3.2 Removed Features None. 2.3.3 Corrections None Copyright © 2006 Polycom, Inc. Page 3 Release Notes - SIP Application Changes 2.3.4 Configuration File Parameter Changes .cfg File sip Action added Parameter voice.gain.rx.analog.chassis.IP_430, voice.gain.rx.analog.ringer.IP_430, 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 voice.rxEq.hs.IP_430.preFilter.enable, voice.rxEq.hs.IP_430.postFilter.enable, 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 voice.txEq.hs.IP_430.preFilter.enable, voice.txEq.hs.IP_430.postFilter.enable, 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 voice.handset.rxag.adjust.IP_430, voice.handset.txag.adjust.IP_430, voice.handset.sidetone.adjust.IP_430, voice.headset.rxag.adjust.IP_430, voice.headset.txag.adjust.IP_430, voice.headset.sidetone.adjust.IP_430 font.IP_400.1.name bitmap.IP_400.61.name ind.anim.IP_400.38.frame.1.bitmap, ind.anim.IP_400.38.frame.1.duration ind.gi.IP_400... Description New gain parameters for SoundPoint IP 430 platform. sip added New Rx EQ parameters for SoundPoint IP 430 platform. sip added New Tx EQ parameters for SoundPoint IP 430 platform. sip added New handset and headset gain adjustments for SoundPoint IP 430 platform. sip sip sip sip added added added changed New dynamic font download parameter for SoundPoint IP 430 platform. New bitmap parameter for SoundPoint IP 430 platform. New animation parameters for SoundPoint IP 430 platform. Changed the values of some of these indicator parameters for the SoundPoint IP 430 platform. 2.4 Version 1.6.6 2.4.1 Added or Changed Features · · 15491: Added configurable option to enable phone with 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 Page 4 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Changes 2.4.2 Removed Features None. 2.4.3 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" softkey does not activate the first available line 14673: Special characters such as `@', `:' 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" softkey 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 credentials 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" softkey 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 Copyright © 2006 Polycom, Inc. Page 5 Release Notes - SIP Application · Changes 16161: Phone with a shared line displays the wrong softkey labels after attempting to hot dial when the remote shared line is in use 2.4.4 Configuration File Parameter Changes .cfg File sip Action added Parameter call.shared.exposeAutoHolds Description 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-INVITE will be sent to the server. Default is "0". 2.5 Version 1.6.5 2.5.1 Added or Changed Features · 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 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. 15012: Added a workaround to restart the application on the phone if many tasks get unrealistic task delays during startup (Outstanding issue 11653) Copyright © 2006 Polycom, Inc. · · · · · · · Page 6 Release Notes - SIP Application Changes 2.5.2 Removed Features None. 2.5.3 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 .cfg doesn't exist 14509: If 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 15162: Phone clears application log file during a warm boot even if the upload to the boot server failed Copyright © 2006 Polycom, Inc. Page 7 · · · · · · · · · Release Notes - SIP Application Changes 2.5.4 Configuration File Parameter Changes .cfg File sip Action added Parameter voIpProt.server.dhcp.available Description 1 = check with the DHCP server for SIP server IP address. 0 = do not check with DHCP server. Default = 0. Option to request from the DHCP server if voIpProt.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. 0 = IP address 1 = string Type to request from the DHCP server if voIpProt.server.dhcp.available = 1. There is no default value for this parameter, it must be filled in with a valid value. 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. sip added voIpProt.server.dhcp.option sip added voIpProt.server.dhcp.type sip added tcpIpApp.sntp.address.overrideDHCP and tcpIpApp.sntp.gmtOffset.overrideDHCP 2.6 Version 1.6.4 2.6.1 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 2.6.2 Removed Features None. 2.6.3 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" softkey when editing Forward contact 12626: Phone reboots on installation of a custom certificate Copyright © 2006 Polycom, Inc. Page 8 Release Notes - SIP Application · · · · · · · · · · · · · · Changes 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 2.6.4 Configuration File Parameter Changes None. 2.7 Version 1.6.3 2.7.1 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 Copyright © 2006 Polycom, Inc. Page 9 Release Notes - SIP Application · Changes 13931: Added notice of file system fix for bug 13361 to header of SoundStation IP 4000 binary image 2.7.2 Removed Features · 13025: Disabled url-dialing in main partner configuration files 2.7.3 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 changed to short date format after first call 13223: All user agent headers for SAS-VP v3 must include 13228: Audio lost for the first call after rejecting the second incoming call if headset or handsfree is used 13235: Repeatly 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 filesystem activity. Note: BootROM must be upgraded to version 3.1.2 as instructed in Technical Bulletin TB13361 · 13517: Handsfree dial-tone volume can become very quiet after significant volume adjustment · · · · · · · · · · Page 10 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Changes 2.7.4 Configuration File Parameter Changes .cfg File 000000000000 Action added Parameter CONTACTS_DIRECTORY, OVERRIDES_DIRECTORY Description New fields which can specify a directory on the boot server in which contact overrides (-directory.xml) and configuration overrides (-phone.cfg) should be stored. 0 or Null: New dialog event package draft is used (no SDP in dialog body). 1: For backwards compatibility, use this setting to send SDP in dialog body. 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. sip added voIpProt.SIP.dialog.useSDP sip changed feature.9.enabled 2.8 Version 1.6.2 2.8.1 Added or Changed Features None. 2.8.2 Removed Features None. 2.8.3 Corrections The following issues have been resolved with this release: · · · · · 9580: Changes in .cfg will not be detected during configuration polling 11190: Incorrect time zone is used for one to two minutes after a reboot 12552: Phone reboots if line keys on Expansion Module are pressed rapidly and continuously 12841: Far end phone continues to ring if near end phone ends call prior to far end answering in specific shared-line scenario 12951: Malformed RTP packets received by phone can cause it to crash 2.8.4 Configuration File Parameter Changes None. 2.9 Version 1.6.1 2.9.1 Added or Changed Features · · 12296: Pressing and holding unassigned line key adds a directory contact 12366: Application log is uploaded shortly after reboot Copyright © 2006 Polycom, Inc. Page 11 Release Notes - SIP Application Changes 2.9.2 Removed Features None. 2.9.3 Corrections The following issues have been resolved with this release: · · · · · · · · · 11388: Phone does not get a CDP response reliably in some scenarios 12208: 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 2.9.4 Configuration File Parameter Changes .cfg Action File sip changed Parameter voice.audioProfile.xxx parameter values and voice.gain.xxx parameter values Description Use the new values for these parameters. 2.10 Version 1.6.0 2.10.1 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 directory 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 © 2006 Polycom, Inc. · Page 12 Release Notes - SIP Application · Changes 11849: Added support for transfer dispatch during consultation call proceeding state New parameter for this is voIpProt.SIP.allowTransferOnProceeding which will normally not need to be changed. 12093: Added a Forward menu so that forwarding can be modified at any time · 2.10.2 Removed Features None. 2.10.3 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 voIpProt.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: Phone 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 Copyright © 2006 Polycom, Inc. Page 13 Release Notes - SIP Application · · · · · · 12284: Time handling for DHCP needs to be improved 12289: Common audio equalization tables should be grouped together Changes 12323: Exiting Display Diagnostics with termination key does not stop display diagnostics 12333: "Direct" and "Group" soft keys can appear when directed and group call pickup features are disabled 12370: Ringing can be heard during a connected call mixed with audio when there is a high number of unanswered incoming calls 12541: Error messages can appear in log file after putting two calls on hold 2.10.4 Configuration File Parameter Changes .cfg Action File sip added Parameter voIpProt.SIP.allowTransferOnProceeding Description 0 = don't allow transfer during consultation call proceeding state 1 = do allow it (1 is the default) Same function and possible values as existing voIpProt.server.x.transport parameter. Default is DNSnaptr. Gains specifically for the IP 601 platform. sip added voIpProt.SIP.outboundProxy.transport sip added sip sip sip sip sip changed changed added/ removed added/ removed added voice.gain.rx.analog.chassis.IP_601, voice.gain.rx.analog.ringer.IP_601, 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 voice.aec.xxx voice.ns.xxx voice.rxEq.xxx voice.txEq.xxx log.level.change.sotet, log.level.change.ttrs Changed parameter values. Do not modify these. Changed parameter values. Do not modify these. This whole section has changed and must be used. Do not modify these. This whole section has changed and must be used. Do not modify these. Added log level control for logging related to Expansion Module. 2.11 Version 1.5.2 2.11.1 Added or Changed Features · · 11356: Changed configuration of presence and instant messaging features to be disabled by default 11552: Added phone UI and web interface configuration support for lineKeys and callsPerLineKey Page 14 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Changes 2.11.2 Removed Features · 11816: Pressing a line key will no longer terminate a call 2.11.3 Corrections The following issues have been resolved with this release: · · · · · · · · · · · · · · · · · · · · · · · 9491: Empty "to" header may be sent in some cases 9776: Parsing errors when dealing with the override file 9817: Configuration override file gets unnecessary extra parameters 11189: User can corrupt the directory by editing it when "presence" feature is disabled 11343: Pressing handsfree or headset button activates handset if handset is off hook 11409: Provisioning may not work reliably with the proftpd FTP server on Linux 11417: Phone may not be able to boot from a remote subnet 11426: Secondary dial tone plays incorrectly on certain digit maps 11466, 11558: Provisioning may fail using HTTPS if a custom certificate is used 11556: Stored authentication key from a SAS-VP server is deleted when the phone is reset to factory defaults 11558: Provisioning may fail using HTTPS if a custom certificate is used 11575: SoundPoint IP300/301 doesn't give warning message if duplicate IP is detected by DHCP client 11584: Automatic key repeats do not work 11595: Phone displays URL encoded digits when dialing 11599: Check-sync and polled configuration change features do not work 11600: Phone ignores maximum password length parameters 11608: Disabling "presence" feature does not remove it from phone's menu 11609: Disabling "messaging" feature on SoundStation IP 4000 and SoundPoint IP30x disables voice message feature as well 11612: When Do Not Disturb per-registration is enabled, the Do Not Disturb "clear all" soft key is missing 11616: CANCEL requests include tag when they shouldn't 11633: Phone should use flash credentials when boot server URL lacks them 11641: Phone shows an error message on the display when Hold is invoked on the last available call appearance 11644: Join does not work from the last available call appearance Copyright © 2006 Polycom, Inc. Page 15 Release Notes - SIP Application · · · · · · · Changes 11665: Pressing the headset button in ringing state does not answer call when headset memory is enabled 11685: Line configuration cannot be changed using web server 11739: A call can be lost when Split is used under certain circumstances 11760: Custom certificate gets corrupted if SAS-VP is used 11788: Pressing "New Call" soft key auto dials the previous number entered using on-hook dialing if the previous call failed 11789: The "more" soft key for establishing a conference can disappear, hiding the "Join" soft key 11798: There is an incompatability when using EPSV with proftpd 2.11.4 Configuration File Parameter Changes .cfg File sip Action changed Parameter feature.1.enabled, feature.2.enabled changed from 1 to 0 voIpProt.server.x.transport reg.x.server.y.transport Description Presence and Instant Messaging are disabled by default. Explicitly set default to DNSnaptr Explicitly set default to DNSnaptr sip phone1 changed changed 2.12 Version 1.5.1 2.12.1 Added or Changed Features · · · · · · · 966: A single call will always show up in the first call appearance position 1509: Improved menu hierarchy 1842: Added visual "status" to contacts assigned to speed dial bins 3924: Added conference feature enhancement to "join" calls in progress 7204: Added flashing time/date until successful SNTP response 7663: Added ability to specify boot server address as URL per RFC 1738 This requires bootROM 3.0 or greater. 7894: Added support for having more than one line key associated with the same SIP identity This includes a new feature ­ pressing and holding down the line key provides call information about a call which is on hold on that line key. 7899: Added support for the application to provision its own configuration files 7900: Added application support for HTTP and HTTPS boot server transport This requires bootROM 3.0 or greater. For HTTPS, if the time on the phone is wrong the SSL certificate may be rejected. Configure SNTP to obtain an accurate time. 8055: Added support for SAS-VP v2 management This requires bootROM 3.0 or greater. Copyright © 2006 Polycom, Inc. · · · Page 16 Release Notes - SIP Application · · · · · · · · · · · · · · · · 8521: Added a menu entry to format the file system 8786: Added display of name and number on incoming caller ID 9053: Added support for displaying a useful CID when display name is uninformative 9096: Added customization options for SSL certificates 9299: Added allowing all files in .cfg to be full URL's Changes 9323: Removed requirement for at least two audio codecs to be configured 9496: Merged sip.cfg and ipmid.cfg configuration files into new sip.cfg file 9548: Added allowing user to disable time and date display 9579: Added allowing specific master configuration file to be specified in boot server URL 9588: Changed offering LED animation to continuous 2 Hz flash, rather than intermittent 9659: Added feature to split conferences and consultation calls into separate calls 9675: Added feature to allow conference initiation from call hold context 9694: Changed example directory file to no longer use silent ring type for contacts 9710: Changed default hold signaling to be the RFC 3261 style 10806: Added build ID to software revision stamps in User-Agent header 11235: Added support for arrow-key call-list shortcuts when phone is playing dial tone 2.12.2 Removed Features · 11973: Removed support for port mode FTP server configurations Use an FTP server/firewall that supports passive mode connections. 2.12.3 Corrections The following issues have been resolved with this release: · · · · · · · 737: Phone will not accept IP packets bigger than 38,000 bytes 2311: Line labels do not line up with line keys on SoundPoint IP 600 3707: Can't use speed dial when one call already on Hold 7952: FTP transfers should remove partially written files in a failure scenario 8050: Parameters which were not changed are saved in configuration override file 8333: Improve source data for random device 8416: Bridged Line second call appearance is incorrect in specific scenario Copyright © 2006 Polycom, Inc. Page 17 Release Notes - SIP Application · · · · · · · · · · · · · · · · · · · · · · · · · · Changes 8616: Incorrect message on display for incoming call on shared line on SoundPoint IP 4000 8674: Missing remote hold call appearance in specific Bridged Line scenario 8755: For TCP, the response to a request should try the remote port that sends the request first 8771: IP 4000 cannot download large directory file 8801: Phone ignores X-Syl-Line-ID and mixes call appearances 8873: When a new DHCP lease is obtained, the updated DNS information is not used 8962: Active Bridged Line cannot switch to incoming call 9090: Clock date menu choice ending in `YYYY' not displayed properly 9135: Random string for CNONCE value for digest authentication should be limited to the base64 character set 9187: GMT offset and SNTP address set in flash are ignored if parameters exist in configuration file but have no associated value (i.e. are empty) 9243: Web server buttons not labeled, and some labels are incorrect 9326: DST not working for Southern Hemisphere 9452: DTMF tones not recognized by specific IVR after shared line remote resume 9481: Phone will attempt to download files indefinitely if connection to FTP server lost 9482: Phone waits for an error response from the FTP server when none is forthcoming 9584: Call duration missing from placed call list items on IP 4000 9601: DNS resolution fails when downloading [mac]-phone.cfg 9735: Web interface of SoundStation IP 4000 phone edits some non-IP 4000 parameters 10825: Phone should not collect digits after dial tone has timed out 11285: SIP authentication password stored in [mac]-phone.cfg file 11303: SoundPoint IP 300 phone loses contrast settings during reboot 11348: Large DHCP messages get truncated 11350: SoundStation IP 4000 phone can lock up when a key is pressed 11458: Audio loss on one leg of conference after second conference automatically put on Hold (first conference is Resumed) 11516: Off-by-one error when ringTypes are saved 11548: Cannot change administrator password or user password on SoundPoint IP 300 Copyright © 2006 Polycom, Inc. Page 18 Release Notes - SIP Application · · · 11559: ACD login does not work on SoundStation IP 4000 Changes 11563: ACD available/unavailable functions work differently on Bridged and Private lines 11573: Pressing Handsfree button does not put you back to handset when handset is off hook 2.12.4 Configuration File Parameter Changes .cfg File ipmid Action removed Parameter All parameters Description The contents of this file have been added to sip.cfg and this file is no longer used. The contents of the old ipmid.cfg file have been added to sip.cfg. The number of calls or conferences which may be active or on hold per line key on the phone. For the IP 600, range is 1 to 24 and default is 24. For all other phones, range is 1 to 8 and default is 8. Changed the default value to "0" (it used to be "1") which means that RFC 3261-style hold signalling is the default. Changed name to voice.gain.rx.analog.chassis.IP_300 voice.gain.rx.analog.ringer.IP_300 This applies to SoundStation IP 4000 phones only in this build. For all other phones, one-touch resume is the default. In order to view call information about a call on hold on another phone with a shared line ­ press and hold down the line key for a few seconds. Changed values used for locating line key labels. Update this whole section. An incoming call causes the LED to flash continuously at 2Hz rather than flash intermittently. These parameters are no longer used. The ring type for each registration can be configured. Range is 1 to 22. Note: ring type number 1 is "silent ring". The number of line keys on the phone to be associated with registration `x'. Range is 1 to the maximum number of line keys on the phone (IP 300 = 2, IP 500 = 3, IP 600 = 6, IP 4000 = 1). Default is 1. sip sip added added All "ipmid.cfg" parameters call.callsPerLineKey sip changed voIpProt.SIP.useRFC2543hold sip changed voice.gain.rx.analog.chassis.IP300 voice.gain.rx.analog.ringer.IP300 call.shared.oneTouchResume sip changed sip sip changed removed ind.gi.IP_600.x... ind.pattern.8.step.3 to 6 sip phone1 removed added all .obs parameters from logging section reg.x.ringType phone1 added reg.x.lineKeys Copyright © 2006 Polycom, Inc. Page 19 Release Notes - SIP Application .cfg File phone1 Changes Description The number of calls or conferences which may be active or on hold per line key for a specific registration on the phone. This will override the global call.callsPerLineKey parameter in sip.cfg. Same range and defaults as call.callsPerLineKey above. The ipmid.cfg file is no longer used. Action added Parameter reg.x.callsPerLineKey 000000000000 removed ipmid.cfg from list of CONFIG_FILES Page 20 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Notes 3. Notes 3.1 Distribution Files The following files constitute the 1.6.7 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 sip.ld Description SIP application executable, App Version 1.6.7.0094 for SoundPoint IP 430 1.6.7.0098 for all other platforms IP 300 2345-11300-001: 1.6.7 IP 301 2345-11300-010: 1.6.7 IP 430 2345-11402-001: 1.6.7 IP 500 2345-11500-001: 1.6.7 2345-11500-010: 1.6.7 2345-11500-030: 1.6.7 2345-11500-020: 1.6.7 IP 501 2345-11500-040: 1.6.7 IP 600 2345-11600-001: 1.6.7 IP 601 2345-11605-001: 1.6.7 IP 4000 2201-06642-001: 1.6.7 main core and SIP configuration file example per-phone SIP configuration 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) sip.cfg phone1.cfg 000000000000.cfg 000000000000-directory~.xml Copyright © 2006 Polycom, Inc. Page 21 Release Notes - SIP Application Files SoundPointIP-dictionary.xml Notes Description dictionary files for multilingual support include (no IP 30X support): Chinese, China (for IP 60X 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 60X and IP 4000 only) Korean, Korea (for IP 60X and IP 4000 only) Norwegian, Norway Portuguese, Portugal Russian, Russia Spanish, Spain Swedish, Sweden start up welcome sound effect SoundPointIPWelcome.wav 3.2 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. 3.2.1 From Version 1.6.6 to 1.6.7 3.2.1.1 Mandatory Changes · 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. 3.2.1.2 Optional Changes · Overriding codec preferences received from far end To allow the phone to override the list of codec preferences received by the phone, set voIpProt.SDP.answer.useLocalPreferences to 1 in sip.cfg. 3.2.2 From Version 1.6.5 to 1.6.6 3.2.2.1 Mandatory Changes None. 3.2.2.2 Optional Changes · Sending re-INVITE to server during conference setup on BLA Set call.shared.exposeAutoHolds to 1 in sip.cfg Page 22 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Notes 3.2.3 From Version 1.6.4 to 1.6.5 3.2.3.1 Mandatory Changes · None. 3.2.3.2 Optional Changes · Getting SIP server address from DHCP The SIP server address can be obtained from a DHCP server if the new parameters voIpProt.server.dhcp.available, voIpProt.server.dhcp.option and voIpProt.server.dhcp.type are configured correctly. 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 tcpIpApp.sntp.address.overrideDHCP and tcpIpApp.sntp.gmtOffset.overrideDHCP to 1. 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. · · 3.2.4 From Version 1.6.3 to 1.6.4 3.2.4.1 Mandatory Changes None. 3.2.4.2 Optional Changes None. 3.2.5 From Version 1.6.2 to 1.6.3 3.2.5.1 Mandatory Changes · 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 voIpProt.SIP.dialog.useSDP parameter in sip.cfg to 1. 3.2.5.2 Optional Changes · Changing the destination of phone-specific override file uploads Use the new CONTACTS_DIRECTORY and OVERRIDES_DIRECTORY fields in 000000000000.cfg. Copyright © 2006 Polycom, Inc. Page 23 Release Notes - SIP Application · Notes Preventing IP address caller ID display when PSTN caller is unknown The "url-dialing" feature must be disabled in order for the IP address to be hidden. 3.2.6 From Version 1.6.1 to 1.6.2 3.2.6.1 Mandatory Changes None 3.2.7 From Version 1.6.0 to 1.6.1 3.2.7.1 Mandatory Changes · 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. 3.2.8 From Version 1.5.2 to 1.6.0 3.2.8.1 Mandatory Changes · Voice Configuration Parameters Updated Many parameters in the "voice" section of sip.cfg have been modified and this entire section is required when using SIP 1.6.0. Transfer On Proceeding Enabled by Default In SIP 1.5.2 there was no option to complete a transfer during the proceeding state of a consultation call. In SIP 1.6.0 this has been added and it is enabled by default. Set the parameter voIpProt.SIP.allowTransferOnProceeding to 0 if this feature is not wanted. Selecting the Transport for an Outbound Proxy The transport used by an outbound proxy is determined by the new parameter voIpProt.SIP.outboundProxy.transport. If this parameter is missing, the default of NAPTR will be used. In SIP 1.5.X the outbound proxy transport was determined by the voIpProt.server.1.transport or reg.x.server.1.transport parameters but these are no longer taken into account. · · 3.2.9 From Version 1.5.1 to 1.5.2 3.2.9.1 Mandatory Changes · Presence and Instant Messaging Disabled by Default These features have been disabled in sip.cfg by setting feature.1.enabled and feature.2.enabled to 0. If these features are required they must be enabled in sip.cfg. Page 24 Copyright © 2006 Polycom, Inc. Release Notes - SIP Application Notes 3.3 Outstanding Issues The following issues will be fixed in a subsequent release. · 4310: No 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. 5085: Cannot answer an incoming call while directory is being saved Workaround: None. 6527: Shared line does not ring if incoming call arrives when phone is playing dialtone then subsequently hangs up Workaround: None. 8532: 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 correct subnet mask. 8547: Local ringback is not played if far end does blind transfer without going on hold Workaround: None. 8921: Centralized conference fails due to RTP port being slow to open in some cases Workaround: None. 9176: Memory leak in phone if it tries to upload log files into a non-existent folder which is specified by LOG_FILE_DIRECTORY Workaround: Specify a valid folder destination in LOG_FILE_DIRECTORY. 9292: IP 4000 reboots upon downloading a wave file with a path containing `\' instead of `/' Workaround: Wave file paths must be specified using `/' e.g. "wavs/ring1.wav". 9709: RTCP not sent or received when calls are on hold Workaround: None. 11588: The local contact directory feature cannot be disabled Workaround: None. 12155: SoundPoint IP 300 and 301 phones have no "Exit" softkey during the ACD login process Workaround: Exit the display by pressing the Menu key or lifting and replacing the handset. 12455: On SoundPoint IP 601 phone, per-contact directory settings such as auto-divert do not work for calls arriving on lines 7 to 12 Workaround: None. 12492: SoundPoint IP 601 phone with Expansion Module(s) attached may fail to load the selected language after rebooting Workaround: Switch to English (Internal) and then back to the desired language after the reboot. Copyright © 2006 Polycom, Inc. Page 25 · · · · · · · · · · · · Release Notes - SIP Application · Notes 12616: Phone crashes after receiving high call rate (4 unanswered calls every 18 seconds) Workaround: Reduce the incoming call rate. 12647: Feature keys cannot be reconfigured to perform other functions Workaround: None. 12722: Stuttered dial tone does not work if first line is shared Workaround: Configure the first line on the phone as a private line 12952: There is no way to reset the user password back to the factory default password Workaround: None. 13076: Phone can pause at the "Welcome" screen for more than 5 minutes after being rebooted Workaround: Ensure that the boot server can handle the load of multiple phones rebooting. 13230: No audio on calls resumed from hold in some multiple call scenarios Workaround: None. 13412: Cannot edit the contact directory on the phone if the phone's directory file saved on the boot server has been corrupted Workaround: Correct the directory file on the boot server and reboot the phone. 13579: SDP parser applies wrong logic Workaround: Change the order of lines in the SDP. 13786: HTTP Digest Authentication does not work on IIS Workaround: Use a different form of authentication, a different protocol or a different server 14275: The call.callWaiting.prompt parameter does not have any effect Workaround: None. This functionality changed in SIP 1.5. 14400: Phone can take up to 30 minutes to boot when there are TCP timeouts Workaround: Ensure that the configured boot server is running correctly or do not use a boot server. 14466: 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. 14467: If a URL in .cfg specifies a protocol and user name but no password, the password in flash is not used Workaround: Specify the password in the configuration file 14624: Boot servers running explicit FTPS are not supported Workaround: Use implicit FTPS or HTTPS. 14844: A failed download of a pre-existing file causes that file to be deleted Workaround: None. 14937: Pattern generator for tones does not work well for the case of a single repeating chord Copyright © 2006 Polycom, Inc. · · · · · · · · · · · · · · · Page 26 Release Notes - SIP Application Reference Documents 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. · 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 Workaround: Include the password in the server address URL. 16041: After a reboot, a phone with a shared line is occasionally unable to seize the line Workaround: Reboot the phone again. 17102: IP430 locks up when performing a reboot on detection of a suspended task. Workaround: Manually reboot the phone. · · 4. Reference Documents · Administrator Guide ­ SoundPoint IP SIP ­ Version 1.6 Copyright © 2006 Polycom, Inc. Page 27