Polycom SoundPoint IP 601 Administrator's Manual

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Administrator Guide

SoundPoint

/SoundStation

IP SIP

Version 1.6.x

18 July 2005

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Summary of Contents for Polycom SoundPoint IP 601

Page 2 Notices 1. Specifications subject to change without notice. Polycom, Inc. 1565 Barber Lane, Milpitas CA 95035, USA www.polycom.com Part Number: 1725-11530-160 Rev A Copyright © 2005 Polycom, Inc. All rights reserved.
Page 3: Table Of Contents
3.1.13 Distinctive Ringing........................27 3.1.14 Distinctive Call Waiting ......................... 27 3.1.15 Do-Not-Disturb..........................28 3.1.16 Handset, Headset, and Speakerphone ................... 28 3.1.17 Local Contact Directory ......................... 29 3.1.17.1 Local Contact Directory File Format..................31 3.1.18 Local Digit Map ..........................32 Copyright © 2005 Polycom, Inc.
Page 4 3.3.9 Background Noise Suppression (BNS)....................51 3.3.10 Comfort Noise Fill......................... 51 3.3.11 Automatic Gain Control (AGC)....................... 51 3.4 Presence and Instant Messaging Features ....51 3.4.1 Presence ............................51 3.4.2 Instant Messaging .......................... 52 3.5 Localization Features ..........52 Copyright © 2005 Polycom, Inc.
Page 5 4.5.2 IEEE 802.1p/Q..........................68 4.5.3 RTCP Support ..........................69 4.6 Configuration Files...........70 4.6.1 SIP Configuration - sip.cfg ......................70 4.6.1.1 Protocol <volpProt/>......................71 4.6.1.2 Dial Plan <dialplan/>......................77 4.6.1.3 Localization <localization/> ....................78 4.6.1.4 User Preferences <user_preferences/> ..................82 Copyright © 2005 Polycom, Inc.
Page 6 5.1.4.3 3xx Responses - Redirection....................145 5.1.4.4 4xx Responses - Request Failure....................145 5.1.4.5 5xx Responses - Server Failure ....................146 5.1.4.6 6xx Responses - Global Failure ....................146 5.1.5 Hold Implementation........................147 5.1.6 Reliability of Provisional Responses....................147 Copyright © 2005 Polycom, Inc.
Page 7 5.2.4 Shared Call Appearance Signaling ....................149 5.2.5 Bridged Line Appearance Signaling ....................149 6 Appendix 1 ............151 6.1 Trusted Certificate Authority List ........151 7 Appendix 2 ............155 7.1 Third Party Software Attribution ........155 Copyright © 2005 Polycom, Inc.
Page 9: Overview
These phones are endpoints in the overall network topology designed to interoperate with other compatible equip- ment including application servers, media servers, internetworking gateways, voice bridges, and other endpoints. Copyright © 2005 Polycom, Inc.
Page 11: Installation And Operation
A boot server allows global and per-phone configuration to be managed centrally via text XML-format configuration files that are downloaded by the phones at boot time. The boot server also facilitates automated application upgrades, diagnostics, and a measure of fault tolerance. Copyright © 2005 Polycom, Inc.
Page 12: Installation Process
If file uploads are permitted, the boot server allows these local changes to be backed up automatically. Polycom recommends the boot server central provisioning model for installations involving more than a few phones. The investment required is minimal in terms of time and equipment, and the benefits are significant.
Page 13: Dhcp Or Manual Tcp/Ip Setup
2.2.1.1 DHCP or Manual TCP/IP Setup Basic network settings can be derived from DHCP or entered manually using the phone’s LCD-based user interface. Polycom recommends using DHCP where possi- ble to eliminate repetitive manual data entry. The following table shows the manually entered networking parameters that may be...
Page 14 SNTP be functional. Provisioning of configuration files is done by the application instead of the bootROM and this transfer can use a secure protocol. 1. HTTP is supported on all phones to download ringer wave files. Copyright © 2005 Polycom, Inc.
Page 15: Local User Interface Setup Menus
Note: Disabled when DHCP client is enabled. Subnet Mask dotted-decimal subnet Phone’s subnet mask. mask Note: Disabled when DHCP client is enabled. IP Gateway dotted-decimal IP Phone’s default router. address Server Menu See 2.2.1.3.3 Server Menu on page 10. Copyright © 2005 Polycom, Inc.
Page 16 The DHCP menu is accessible only when the DHCP client is enabled. DHCP config- uration parameters are described in the following table: Possible Name Values Description Timeout 1 through 600 Number of seconds the phone waits for secondary DHCP Offer messages before selecting an offer. Copyright © 2005 Polycom, Inc.
Page 17 DHCP private option value. VLAN ID Option 128 through 254 The DHCP private option value (when VLAN Discov- (Cannot be the ery is set to Custom). Default is 129. same as Boot Server Option) Copyright © 2005 Polycom, Inc.
Page 18: Server Menu
The server user name and password should be changed from the default values. Note that for insecure protocols the user chosen should have very few privileges on the server. ® ® b. Not available on SoundPoint IP 300 and SoundPoint IP 500 phones. Copyright © 2005 Polycom, Inc.
Page 19: Reset To Factory Defaults
XML-format configuration files. In the centrally provisioned model, these files are stored on a boot server and cached in the phone. If the boot server is available at boot time, the phone will automatically synchronize its configuration cache with the boot Copyright © 2005 Polycom, Inc.
Page 20: Configuration Files
If a phone does not find its own <Ethernet address>.cfg file, it will use this one, and establish a baseline configuration. This file is part of the standard Polycom distribution of configuration files. It should be used as the template for the <Ethernet address>.cfg files.
Page 21 2.2.2.1.1.2 Application Configuration Files Typically, the files are arranged in the following manner although parameters may be moved around within the files and the file names themselves can be changed as needed. Per-phone settings phoneXXXX.cfg Application settings sip.cfg Copyright © 2005 Polycom, Inc.
Page 22 Each phone in an installation usually has its own customized ver- sion of user files derived from Polycom templates. These application configuration files dictate the behavior of the phone once it is run- ning the executable specified in the master configuration file.
Page 23 Edit contents of phoneXXXX.cfg as appropriate. For example, edit the registration parameters. Edit the CONFIG_FILES attribute of the <Ethernet address>.cfg files so that it references the appropriate phoneXXXX.cfg file. (Replace the reference to phone1.cfg with phoneXXXX.cfg.) Copyright © 2005 Polycom, Inc.
Page 24 • time and date format Decide on boot server security pol- Polycom recommends allowing file uploads to the icy: boot server where the security environment permits. This allows event log files to be uploaded and changes made by the phone user to the configuration (via the web server and local user interface) and changes made to the directory to be backed up.
Page 26 If the boot server permits uploads, these override setting will be saved in a file called <Ethernet address>-phone.cfg on the boot server. Important Local configuration changes will continue to override the boot server-derived configuration until deleted via the Reset User Settings menu selection. Copyright © 2005 Polycom, Inc.
Page 27: Features
125. Web Server None. (if enabled) Local Local Telephone None. User Interface 3.1.2 Call Timer A call timer is provided on the display. A separate call timer is maintained for each distinct call in progress. Copyright © 2005 Polycom, Inc.
Page 28: Call Waiting
Missed Calls list. The types of calls which are counted as “missed” can be configured per registration. Remote missed-call notification can be used to notify the phone when a call originally destined for it is diverted by another entity such as a SIP server. Copyright © 2005 Polycom, Inc.
Page 29: Configurable Feature Keys
Local Telephone None. User Interface The following diagrams and table show the default SIP key layouts for SoundPoint ® IP 300, IP 301, IP 500, IP 501, IP 600, IP 601 and SoundStation ® 4000 models. Copyright © 2005 Polycom, Inc.
Page 30 OPER Key ID SoundPoint IP 500 and IP 501 SIP Key Layout Conference Directories Services Menu Call Lists Messages Conference Do Not Disturb Transfer PQRS WXYZ Redial Hold Hold PQRS WXYZ OPER OPER Key ID Copyright © 2005 Polycom, Inc.
Page 31 SoundPoint IP 600 and IP 601 SIP Key Layout Menu Directories Services Messages Conference Do Not Disturb Transfer Redial PQRS WXYZ Hold PQRS WXYZ Hold OPER OPER Key ID SoundStation IP 4000 SIP Key Layout Copyright © 2005 Polycom, Inc.
Page 32 Dialpad0 Dialpad1 Dialpad6 Dialpad6 DialpadPound Dialpad3 Dialpad3 Redial Redial Dialpad2 Dialpad2 Dialpad1 Dialpad1 SoftKey3 SoftKey4 SoftKey4 Menu MicMute SoftKey3 SoftKey3 Exit SoftKey2 SoftKey2 SoftKey2 SoftKey1 SoftKey1 SoftKey1 SoftKey1 SoftKey2 ArrowDown Conference Services SoftKey3 CallHistory Directories Copyright © 2005 Polycom, Inc.
Page 33: Connected Party Identification
<volume/> on page 94. 3.1.10 Customizable Audio Sound Effects Audio sound effects used for incoming call alerting and other indications are customi- zable. Sound effects can be composed of patterns of synthesized tones or sample Copyright © 2005 Polycom, Inc.
Page 34: Message Waiting Indication
The distinctive treatment that can be applied includes customizable alerting sound effects and automatic call diversion or rejection. Call attributes that can trigger ® ® 2. L16/16000 is not supported on SoundPoint IP 300, 301 and SoundStation IP 4000 phones. Copyright © 2005 Polycom, Inc.
Page 35: Distinctive Ringing
Reset User Settings menu selection. 3.1.14 Distinctive Call Waiting The SIP Alert-Info field can be used to map calls to distinct call waiting types, cur- rently limited to two styles. Copyright © 2005 Polycom, Inc.
Page 36: Do-Not-Disturb
(not supplied). The SoundPoint IP 500, 501, 600 and 601 ® phones have full-duplex speakerphones. The SoundPoint IP 300 and 301 phones ® have a listen-only speakerphone. The SoundPoint phones provide dedicated keys for Copyright © 2005 Polycom, Inc.
Page 37: Local Contact Directory
The directory is the central database for several other features including speed-dial, distinctive incoming call treatment, presence, and instant messaging. See the following table for further information. Copyright © 2005 Polycom, Inc.
Page 38 Local backed up to the boot server copy of <Ethernet address>-directory.xml if this is configured. When the phone boots, the boot server copy of the directory, if present, will overwrite the local copy. Copyright © 2005 Polycom, Inc.
Page 39: Local Contact Directory File Format
<sd>3</sd> <rt>3</rt> <dc /> <ad>0</ad> <ar>0</ar> <bw>0</bw> <bb>0</bb> </item> </item_list> </directory> Element Permitted Values Interpretation UTF-8 encoded string of up to first name 40 bytes UTF-8 encoded string of up to last name 40 bytes Copyright © 2005 Polycom, Inc.
Page 40: Local Digit Map
Instead, as soon as a digit pattern matching the digit map is found, the call setup process will complete automatically. This feature is similar to the digit map feature of the Media Gateway Control Protocol (MGCP) and Copyright © 2005 Polycom, Inc.
Page 41: Microphone Mute
The number of line keys allocated per registration is configurable. Configuration file: Specify the number of line keys to assign per registration. Central phone1.cfg (boot • For more information, see 4.6.2.1 Registration <reg/> server) on page 128. Copyright © 2005 Polycom, Inc.
Page 42: Multiple Call Appearances
Specify per-registration the number of calls which can be server) phone1.cfg active or on hold per line key assigned to that registration. This will override the default value specified in sip.cfg. • For more information, see 4.6.2.1 Registration <reg/> on page 128. Copyright © 2005 Polycom, Inc.
Page 44 • For more information, see 4.6.2.1 Registration <reg/> on page 128. Specify per-registration whether diversion should be dis- abled on shared lines. • For more information, see 4.6.2.3 Diversion <divert/> on page 133. Copyright © 2005 Polycom, Inc.
Page 45: Bridged Line Appearances
SIP server which binds the appearances together logi- cally and looks after the necessary state notifications and performs an access control function. For more information, see 5.2.5 Bridged Line Appearance Signaling on page 149. See the following table for further information. Copyright © 2005 Polycom, Inc.
Page 46: Customizable Fonts And Indicators
The bitmaps and bit- map animations used for graphic icons on the display can be changed and reposi- tioned. LED flashing sequences and colors can be changed. See the following table for further information. Copyright © 2005 Polycom, Inc.
Page 47: Soft Key-Driven User Interface
If Presence watching is enabled for speed dial entries, their status will be shown on the idle display if the SIP server supports this feature. See 3.4.1 Presence on page 51. See the following table for further information. Copyright © 2005 Polycom, Inc.
Page 48: Time And Date Display
The clock and calendar must be synchronized to a remote SNTP timeserver. The time and date displayed on the phone will flash continuously until a successful SNTP response is Copyright © 2005 Polycom, Inc.
Page 49: Idle Display Animation
All phones except the SoundPoint IP 300 and SoundPoint IP 301 can display a cus- tomized animation on the idle display in addition to the time and date. For example, a company logo could be displayed. Copyright © 2005 Polycom, Inc.
Page 50: Call Management Features
(boot • For more information, see 4.6.2.2.2 Automatic Off- server) hook Call Placement <autoOffHook/> on page 131. Web Server None. (if enabled) Local Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 51: Call Hold
A dials C’s number and hears ring-back and decides to complete the transfer before C answers. This option can be disabled. • True consultation transfers: User A dials C’s number and consults privately with C after the call is answered and then completes the transfer. Copyright © 2005 Polycom, Inc.
Page 52: Three-Way Conference, Local Or Centralized
The call diversion feature works in conjunction with the distinctive ® 3. On SoundStation IP 4000, conferences are not available if the G.729 codec is enabled on the phone. This restriction will be removed in future releases. Copyright © 2005 Polycom, Inc.
Page 53: Directed Call Pick-Up
This feature depends on support from a SIP server. Configuration file: Turn this feature on or off. Central sip.cfg (boot • For more information, see 4.6.1.23 Feature <feature/> server) on page 125. Web Server None. (if enabled) Local Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 54: Group Call Pick-Up
This feature depends on support from a SIP server. Configuration file: Turn this feature on or off. Central sip.cfg (boot • For more information, see 4.6.1.23 Feature <feature/> server) on page 125. Web Server None. (if enabled) Local Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 55: Last Call Return
(lost or excessively delayed by the network) packets. The jitter buffer is adaptive and config- urable for different network environments. When packets are lost, a concealment algorithm minimizes the resulting negative audio consequences. Copyright © 2005 Polycom, Inc.
Page 56: Local Conference Mixing
For more information, see 4.6.1.8.10 Voice Activity server) Detection <VAD/> on page 102. ® 4. On SoundStation IP 4000, conferences are not available if the G.729 codec is enabled on the phone. Copyright © 2005 Polycom, Inc. This restriction will be removed in future releases.
Page 57: Dtmf Tone Generation
SDP offers. (boot • For more information, see 4.6.1.5.1 Dual Tone Multi- server) Frequency <DTMF/> on page 83. Web Server None. (if enabled) Local Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 58: Acoustic Echo Cancellation (Aec)
Changes are saved to local flash and backed up to <Ethernet Local address>-phone.cfg on the boot server. Changes will per- manently override global settings unless deleted via the Reset User Settings menu selection. Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 59: Background Noise Suppression (Bns)
5. AGC support will be available in a subsequent release. 6. Notification when a change in monitored status occurs will be available in a subsequent release. Copyright © 2005 Polycom, Inc.
Page 60: Instant Messaging
System Administrator or the user can choose the language. Support for major western European languages is included and additional languages can be easily added. Sup- port for Asian languages (Chinese, Japanese, and Korean) is also included but will Copyright © 2005 Polycom, Inc.
Page 62: Downloadable Fonts
<font/> on page 111. 3.5.3 Synthesized Call Progress Tones In order to emulate the familiar and efficient audible call progress feedback generated by the PSTN and traditional PBX equipment, call progress tones are synthesized dur- Copyright © 2005 Polycom, Inc.
Page 63: Advanced Server Features
Voicemail access can be configured to be via a single key press if only one registration has voicemail set up and the phone has a dedicated function key ® for this purpose (for example the Messages key on the SoundPoint IP 500, 501, 600 Copyright © 2005 Polycom, Inc.
Page 64: Multiple Registrations
IP 300 and 301 support ® a maximum of two registrations, the SoundPoint IP 500 and 501 support three and the SoundPoint ® IP 600 and 601 support six. With the attachment of one or more Copyright © 2005 Polycom, Inc.
Page 65 The optional array of serv- ers and their associated parameters will override the servers specified in sip.cfg if non-Null. • For more information, see 4.6.2.1 Registration <reg/> on page 128. Copyright © 2005 Polycom, Inc.
Page 66 Reset User Settings menu selection. • For more information on the fields in this menu, see 4.6.1.1.1 Local <local/> on page 71, 4.6.1.1.2 Server <server/> on page 71 and 4.6.2.1 Registration <reg/> on page 128. Copyright © 2005 Polycom, Inc.
Page 67: Acd Login / Logout
The phone can be configured with multiple SIP servers, one primary and one or more backup. The phone will switch to a backup server when the current primary server fails. Backup server configuration can be static or can use advanced DNS methods. In Copyright © 2005 Polycom, Inc.
Page 68: Dns Sip Server Name Resolution
Services key. Configuration file: Specify the Services browser home page, a proxy to use, and Central sip.cfg size limits. (boot • For more information, see 4.6.1.25 MicroBrowser server) <microbrowser/> on page 127. Copyright © 2005 Polycom, Inc.
Page 69: Security Features
See 6.1 Trusted Certificate Authority List on page 151. In addition, custom certificates can be added to the phone. This is done by using the SSL Security menu on the phone to pro- Copyright © 2005 Polycom, Inc.
Page 70: Incoming Signaling Validation
Central cases. (boot server) • For more information, see 4.6.1.1.3.3 Request Valida- tion <requestValidation/> on page 75. Web Server (if None. enabled) Local Local Phone User None. Interface Copyright © 2005 Polycom, Inc.
Page 71: Optimization
RTP endpoints will negotiate the respective destination IP addresses and ports. This allows RTCP to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unautho- rized sources can be rejected. Copyright © 2005 Polycom, Inc.
Page 73: Updating And Rebooting
Manual Reboot: Menu Option or Key Presses To reboot phones manually, a menu option can be selected or a key combination can be used. The menu option is called Restart Phone and it is found in the Settings menu. Copyright © 2005 Polycom, Inc.
Page 74: Event Logging
The boot log file is uploaded to the boot server after every reboot. The application log file is uploaded periodically or when the local copy reaches a pre- determined size. 7. Note: HTTP and TFTP don’t support append mode unless server settings are changed for this. Copyright © 2005 Polycom, Inc.
Page 75: Audio Quality Issues And Vlans
The “type of service” field in an IP packet header consists of four TOS bits and a 3-bit precedence field. Each TOS bit can be set to either 0 or 1. The precedence field can Copyright © 2005 Polycom, Inc.
Page 76: Ieee 802.1P/Q
Phase 1: bootRom - Navigate to: SETUP menu during auto- Local boot countdown. Phase 2: Application - Navigate to: Menu>Set- tings>Advanced>Admin Settings>Network Configuration • For more information, see 2.2.1 Basic Network Setup on page 4. Copyright © 2005 Polycom, Inc.
Page 77: Rtcp Support
4.5.3 RTCP Support The phone supports RTCP per RFC 1889. For each RTP stream, which, by conven- tion, uses even ports only, the next higher odd port is used to send and receive RTCP reports. Copyright © 2005 Polycom, Inc.
Page 78: Configuration Files
Settings include the local port used for SIP signaling, the address and ports of a cluster of SIP servers, and other parameters. The following sec- tions describe each of these parameters. Copyright © 2005 Polycom, Inc.
Page 79: Protocol
IP address, there is no DNS lookup and 5060 is used for the port but it is not advertised in signaling. If port is 1 to 65535: This value is used and it is advertised in signaling. Copyright © 2005 Polycom, Inc.

Page 80 Requested line-seize subscrip- minimum 10 tion period. a. This is the phone’s requested registration period. The period negotiated with the server may be different. The phone will attempt to re-register when half the negotiated period has expired. Copyright © 2005 Polycom, Inc.
Page 81 If set to 1, a transfer can be completed Proceeding during the proceeding state of a consul- tation call. This is the default. If set to 0, a transfer is not allowed dur- ing the proceeding state of a consulta- tion call. Copyright © 2005 Polycom, Inc.
Page 82 (x=1, 2, ..., N) and if a INVITE match is found, the behavior requests described in the correspond- ing ring class (see 4.6.1.7.2 voIpProt.SIP.alertInfo.x.class positive Null Ring type <ringType/> on integer page 91) will be applied. Copyright © 2005 Polycom, Inc.
Page 83 A valid string PolycomSPIP Determines string used for tion.digest.realm Realm. a. WARNING: Intensive request validation may have a negative performance impact due to the additional signaling required in some cases, therefore, use it judiciously. Copyright © 2005 Polycom, Inc.
Page 84 If set to some value, conferences are set up by the server using the conferencing agent specified by this address. The acceptable values depend on the confer- encing server implementation policy. Copyright © 2005 Polycom, Inc.
Page 85: Dial Plan
Timeout in seconds for ‘T’ feature of digitmap. 4.6.1.2.2 Routing <routing/> This configuration section allows the user to create a specific routing path for outgoing SIP calls independent of other ‘default’ configuration. Copyright © 2005 Polycom, Inc.

Page 86: Localization
The call progress tones can also be customized. For more information, see 4.6.1.3.3 Call Progress Tones <callProgTones> on page 81, 4.6.1.5.2 Chord Sets <chord_sets/> on page 84, and 4.6.1.7.1.1 Call Progress Patterns on page 89. Copyright © 2005 Polycom, Inc.

Page 87 The call progress tone index to be associated with this language. See 4.6.1.3.3 Call Progress Tones <call- ProgTones> on page 81. lcl.ml.lang.clock.x.24HourClock If attribute present, overrides lcl.datetime.time.24HourClock; If 1, display time in 24-hour clock mode rather than am/pm. Copyright © 2005 Polycom, Inc.
Page 88 4. Add a lcl.ml.lang.clock.menu.x attribute to the configuration file. 5. Add lcl.ml.lang.clock.x.24HourClock, lcl.ml.lang.clock.x.format, lcl.ml.lang.clock.x.longFormat and lcl.ml.lang.clock.x.dateTop attributes and set them according to the regional preferences. 6. (Optional) Set lcl.ml.lang to be the new language_region string. Copyright © 2005 Polycom, Inc.
Page 89 String to specify the country or region such as Italy. Multiple lcl.cpt.menu.x strings are sup- ported, the strings are displayed in the Call Progress Tones menu. The lcl.cpt.menu.x attributes must be sequential (lcl.cpt.menu.1, lcl.cpt.menu.2, lcl.cpt.menu.3, ..., lcl.cpt.menu.N) with no gaps. Copyright © 2005 Polycom, Inc.
Page 90: User Preferences
If set to 1, play welcome sound effect after a reboot. up.welcomeSoundOnWarm- 0, 1 If set to 1, play welcome sound effect on BootEnabled warm as well as cold boots, otherwise only a cold boot will trigger the wel- come sound effect. Copyright © 2005 Polycom, Inc.

Page 91: Tones
This prevents DTMF digits being broadcast to other surrounding telephony devices or being inadvert- ently transmitted in-band due to local acoustic echo. Note: tone.dtmf.chassis.masking should only be enabled when tone.dtmf.viaRtp is disabled. Copyright © 2005 Polycom, Inc.

Page 92 There are three blocks of chord sets: • callProg: used for call progress sound effect patterns • ringer • misc (miscellaneous) All three blocks use the same chord set specification format. Copyright © 2005 Polycom, Inc.
Page 93: Sampled Audio For Sound Effects
Files will be truncated to a maximum size of 300 kilo- bytes. ® ® 8. L16/16000 is not supported on SoundPoint IP 300, 301 and SoundStation IP 4000 phones. Copyright © 2005 Polycom, Inc.

Page 94 Welcome Sound Effect (se.pat.misc.7) Ringer 13 (se.pat.ringer.13) Ringer 14 (se.pat.ringer.14) Ringer 15 (se.pat.ringer.15) Ringer 16 (se.pat.ringer.16) Ringer 17 (se.pat.ringer.17) Ringer 18 (se.pat.ringer.18) Ringer 19 (se.pat.ringer.19) Ringer 20 (se.pat.ringer.20) Ringer 21 (se.pat.ringer.21) Ringer 22 (se.pat.ringer.22) 12-24 Not used. Copyright © 2005 Polycom, Inc.
Page 95: Sound Effects
2000 milliseconds) silence (d) Play silence for d se.pat.callProg.x.inst.y.type = “silence” (silence instruc- milliseconds (Rx tion type) audio is not muted) se.pat.callProg.x.inst.y.value = “300” (specifies silence is to last 300 milliseconds) Copyright © 2005 Polycom, Inc.

Page 97 The following table maps ringer pattern numbers to their default descriptions. Ringer pattern number Default description Silent Ring Low Trill Low Double Trill Medium Trill Medium Double Trill High Trill High Double Trill Highest Trill Copyright © 2005 Polycom, Inc.
Page 98 The following table maps miscellaneous patterns to their usage within the phone. Miscellaneous pattern number Use within phone new message waiting indication new instant message Not used. local hold notification positive confirmation negative confirmation welcome (boot up) Copyright © 2005 Polycom, Inc.
Page 99 The duration of the ring in milliseconds relevant if the type is before the call is auto answered. If this field set to ‘ring-answer’. is omitted or is left blank, a value of 2000 is Default value is 2000. used. Copyright © 2005 Polycom, Inc.
Page 100: Voice Settings
IP 500, 501, 600 and voice.codecPref.G711A 601 platforms. voice.codecPref.G729AB 1 = highest 3 = lowest Null = do not use Give each codec a unique priority, this will dictate the order used in SDP negotiations. Copyright © 2005 Polycom, Inc.

Page 102 If set to 1, the receive volume will be remembered between calls. voice.volume.persist.headset 0, 1 If set to 0, the receive volume will be voice.volume.persist.handsfree 0, 1 reset to nominal at the start of each call. Copyright © 2005 Polycom, Inc.
Page 103 Do not alter these values. Attribute Default voice.gain.rx.analog.handset voice.gain.rx.analog.headset voice.gain.rx.analog.chassis voice.gain.rx.analog.chassis.IP_300 voice.gain.rx.analog.chassis.IP_4000 voice.gain.rx.analog.chassis.IP_601 voice.gain.rx.analog.ringer voice.gain.rx.analog.ringer.IP_300 voice.gain.rx.analog.ringer.IP_4000 voice.gain.rx.analog.ringer.IP_601 voice.gain.rx.digital.handset voice.gain.rx.digital.headset voice.gain.rx.digital.chassis voice.gain.rx.digital.chassis.IP_4000 voice.gain.rx.digital.chassis.IP_601 voice.gain.rx.digital.ringer voice.gain.rx.digital.ringer.IP_4000 voice.gain.rx.digital.ringer.IP_601 voice.gain.rx.analog.handset.sidetone voice.gain.rx.analog.headset.sidetone voice.gain.tx.analog.handset voice.gain.tx.analog.headset voice.gain.tx.analog.chassis voice.gain.tx.analog.chassis.IP_300 voice.gain.tx.analog.chassis.IP_4000 voice.gain.tx.analog.chassis.IP_601 voice.gain.tx.digital.handset Copyright © 2005 Polycom, Inc.
Page 104 These settings control the performance of the speakerphone acoustic echo canceller. Do not alter these values. Attribute Default voice.aec.hs.enable voice.aec.hs.lowFreqCutOff voice.aec.hs.highFreqCutOff 7000 voice.aec.hs.erlTab_0_300 voice.aec.hs.erlTab_300_600 voice.aec.hs.erlTab_600_1500 voice.aec.hs.erlTab_1500_3500 voice.aec.hs.erlTab_3500_7000 voice.aec.hd.enable voice.aec.hd.lowFreqCutOff voice.aec.hd.highFreqCutOff 7000 voice.aec.hd.erlTab_0_300 voice.aec.hd.erlTab_300_600 voice.aec.hd.erlTab_600_1500 voice.aec.hd.erlTab_1500_3500 voice.aec.hd.erlTab_3500_7000 Copyright © 2005 Polycom, Inc.
Page 105 ® ® Administrator Guide - SoundPoint IP / SoundStation Optimization Attribute Default voice.aec.hf.enable voice.aec.hf.lowFreqCutOff voice.aec.hf.highFreqCutOff 7000 voice.aec.hf.erlTab_0_300 voice.aec.hf.erlTab_300_600 voice.aec.hf.erlTab_600_1500 voice.aec.hf.erlTab_1500_3500 voice.aec.hf.erlTab_3500_7000 Copyright © 2005 Polycom, Inc.
Page 107 4.6.1.8.7 Automatic Gain Control <AGC/> These settings control the performance of the transmit automatic gain control feature. Do not alter these values. Attribute Default voice.agc.hs.enable voice.agc.hd.enable voice.agc.hf.enable 9. Automatic Gain Control will be implemented in a future release. Copyright © 2005 Polycom, Inc.
Page 108 These settings control the performance of the receive equalizaton feature. Do not alter these values. Attribute Default voice.rxEq.hs.IP_500.preFilter.enable voice.rxEq.hs.IP_600.preFilter.enable voice.rxEq.hs.IP_601.preFilter.enable voice.rxEq.hs.IP_500.postFilter.enable voice.rxEq.hs.IP_600.postFilter.enable voice.rxEq.hs.IP_601.postFilter.enable voice.rxEq.hd.IP_500.preFilter.enable voice.rxEq.hd.IP_600.preFilter.enable voice.rxEq.hd.IP_601.preFilter.enable voice.rxEq.hd.IP_500.postFilter.enable voice.rxEq.hd.IP_600.postFilter.enable voice.rxEq.hd.IP_601.postFilter.enable voice.rxEq.hf.IP_500.preFilter.enable voice.rxEq.hf.IP_600.preFilter.enable voice.rxEq.hf.IP_601.preFilter.enable voice.rxEq.hf.IP_4000.preFilter.enable voice.rxEq.hf.IP_500.postFilter.enable voice.rxEq.hf.IP_600.postFilter.enable voice.rxEq.hf.IP_601.postFilter.enable voice.rxEq.hf.IP_4000.postFilter.enable voice.rxEq.hf.IP_500.preFilter... Do not change these values. Copyright © 2005 Polycom, Inc.
Page 109 These settings control the performance of the hands-free transmit equalization feature. Do not alter these values. Attribute Default voice.txEq.hs.IP_500.preFilter.enable voice.txEq.hs.IP_600.preFilter.enable voice.txEq.hs.IP_601.preFilter.enable voice.txEq.hs.IP_500.postFilter.enable voice.txEq.hs.IP_600.postFilter.enable voice.txEq.hs.IP_601.postFilter.enable voice.txEq.hd.IP_500.preFilter.enable voice.txEq.hd.IP_600.preFilter.enable voice.txEq.hd.IP_601.preFilter.enable voice.txEq.hd.IP_500.postFilter.enable voice.txEq.hd.IP_600.postFilter.enable voice.txEq.hd.IP_601.postFilter.enable voice.txEq.hf.IP_500.preFilter.enable voice.txEq.hf.IP_600.preFilter.enable voice.txEq.hf.IP_601.preFilter.enable voice.txEq.hf.IP_4000.preFilter.enable voice.txEq.hf.IP_500.postFilter.enable voice.txEq.hf.IP_600.postFilter.enable voice.txEq.hf.IP_601.postFilter.enable voice.txEq.hf.IP_4000.postFilter.enable voice.txEq.hf.IP_500.postFilter... Do not change these values. Copyright © 2005 Polycom, Inc.
Page 110: Quality Of Service
User-priority used for RTP packets. 4.6.1.9.1.2 Call Control <CallControl/> These parameters apply to call control packets, such as the network protocol signaling. Permitted Attribute Values Default Interpretation qos.ethernet.callControl.user_priority User-priority used for call con- trol packets. Copyright © 2005 Polycom, Inc.

Page 111 IP TOS field of the IP header, or else don’t set it. qos.ip.rtp.precedence If set to 1, set precedence bits in the IP TOS field of the IP header, or else don’t set them. Copyright © 2005 Polycom, Inc.
Page 112: Basic Tcp/Ip
The defaults shown will enable daylight savings time for North America. Daylight savings defaults: • do not use fixed day, use first or last day of week in the month, • start DST on the first Sunday in April at 2 am, Copyright © 2005 Polycom, Inc.

Page 113 Week.lastInMonth Enable=0, start DST on the last day of the week (specified by dayOf- Week) in the month, rather than the first in the month. tcpIpApp.sntp.daylightSavings.stop.month 1-12 Month to stop DST. 1=Jan, 2=Feb, ..., 12=Dec Copyright © 2005 Polycom, Inc.
Page 114 If set to 1, reject RTP pack- ets arriving from (sent from) a non-negotiated (via SDP) IP address. tcpIpApp.port.rtp.filterByPort 0, 1 If set to 1, reject RTP pack- ets arriving from (sent from) a non-negotiated (via SDP) port. Copyright © 2005 Polycom, Inc.
Page 115: Web Server
0, 1 If set to 1, the HTTP server will be enabled. 4.6.1.11.1 Configuration <cfg/> Permitted Attribute Values Default Interpretation httpd.cfg.enabled 0, 1 If set to 1, the HTTP server configuration interface will be enabled. Copyright © 2005 Polycom, Inc.

Page 116: Call Handling Configuration
This is the number of calls or conferences which may be active or on hold per line key on the phone. Note that this may be overridden by the per-registration attribute of reg.x.callsPer- LineKey. See 4.6.2.1 Registration <reg/> on page 128. Copyright © 2005 Polycom, Inc.

Page 117 Time in seconds between subse- integer quent reminders. call.hold.localReminder.startDelay non-negative Time in seconds to wait before integer the initial reminder. Copyright © 2005 Polycom, Inc.
Page 118: Directory
1 to 100 Maximum size in Kbytes of volatile storage that the direc- tory will be permitted to con- sume. 10.The phone could have 2 megabytes or 4 megabytes of flash memory depending on the hardware model. Copyright © 2005 Polycom, Inc.

Page 119: Fonts
600 is a large font, larger than the one used to display the date, and if you overwrite this default font with a unique font, you lose this size aspect. 11..fon file format is a collection of .fnt fonts mangled together within a single file. Copyright © 2005 Polycom, Inc.

Page 120 When defining a single .fon file, there is a need for a “font delimiter”, currently “Copyright Polycom Canada Ltd” is used as an embedded delimiter, but this can be configured using “font.delimiter”. The font delimiter is important to retrieve the dif- ferent mangled .fnt blocks.
Page 121: Keys
IP 300 and 301 phones, and IP_500 parameters ® affect SoundPoint IP 500 and 501 phones. IP 300: y=1-35; IP 500: y=1-40; IP 600: y=1-42 Permitted Attribute Values Interpretation key.x.y.function.prim Functions listed Sets the function for key y on platform x. below. Copyright © 2005 Polycom, Inc.

Page 122 The following table lists the functions that are available: Function ArrowDown ArrowLeft ArrowRight ArrowUp BuddyStatus CallList Conference Delete Dialpad0 Dialpad1 Dialpad2 Dialpad3 Dialpad4 Dialpad5 Dialpad6 Dialpad7 Dialpad8 Dialpad9 DialpadStar DialpadPound Directories DoNotDisturb Handsfree Headset Hold Copyright © 2005 Polycom, Inc.
Page 123: Bitmaps
Line3 Line4 Line5 Line6 Messages Menu MicMute MyStatus Null Offline Redial Select Setup SoftKey1 SoftKey2 SoftKey3 SoftKey4 SpeedDial SpeedDialMenu Transfer VolDown VolUp 4.6.1.16 Bitmaps <bitmaps/> Bitmaps used by the phone are defined in this section. Copyright © 2005 Polycom, Inc.

Page 124: Indicators
This section defines bitmap animations composed of bitmap/duration couples. In the following table, x=IP_300, IP_500, IP_600 or IP_4000, y is the animation number, z is the step in the animation. Note that IP_300 parameters affect SoundPoint ® IP 300 and Copyright © 2005 Polycom, Inc.

Page 125 For LED type indicators, index refers to the pattern index, such as index x in the <Patterns/> tag above. For GraphicIcon type indicators, index refers to the animation index, such as index y in the <Anima- tions/> tag above. Copyright © 2005 Polycom, Inc.
Page 126 IP 300: 0-3 For GraphicIcon type indicators, this is the y-axis loca- tion of the upper left corner of the indicator measured in IP 500: 0-79 pixels from top to bottom. IP 600: 0-159 IP 4000: 0-67 Copyright © 2005 Polycom, Inc.
Page 127: Event Logging
IP 4000: 1-68 4.6.1.18 Event Logging <logging/> Logging parameter changes can impair system operation. Do not change any logging parameters Warning! without prior consultation with Polycom. The event logging system supports the following classes of events: Level Interpretation Debug only...

Page 128 See above table for timestamp type. log.render.realtime 0, 1 Set to 1. Do not change. log.render.stdout 0, 1 Set to 1. Do not change. log.render.file 0, 1 Set to 1. Do not change. Copyright © 2005 Polycom, Inc.
Page 129 The phone can be configured to schedule certain advanced logging tasks on a periodic basis. These attributes should be set in consultation with Polycom. Each scheduled log task is controlled by a unique attribute set starting with log.sched.x where x identi- fies the task.
Page 130: Security
(Ethernet address) via protocol signaling. 4.6.1.19.1 Password Lengths <pwd/><length/> Permitted Attribute Values Default Interpretation sec.pwd.length.admin 0-32 Password changes will need to be at least this long. Use 0 to allow null passwords. sec.pwd.length.user 0-32 Copyright © 2005 Polycom, Inc.

Page 131: Provisioning
Permitted Attribute Values Default Interpretation ramdisk.enable 0, 1 If set to 1, RAM disk will be available. The RAM disk is used to cache downloaded wave files, and other resources for the user interface. Copyright © 2005 Polycom, Inc.

Page 133: Feature
“acd-agent-available” is the ACD agent avail- able/unavailable feature feature.x.enabled 0 or 1 (default) If set to 0, the feature will be disabled. If set to 1, the feature will be enabled and usable by the local user. Copyright © 2005 Polycom, Inc.

Page 134: Resource
If downloading a resource would make the quota exceeded for that category, the resource will not be downloaded and a predefined default will be used instead. a. This is available on MGCP application, but not on SIP application. Copyright © 2005 Polycom, Inc.

Page 135: Microbrowser
XML parser will handle. This limits the amount of memory used by complicated pages. A maximum total of 500 (256 each) is recommended. Increasing this value may have a detrimental effect on performance of the phone. Copyright © 2005 Polycom, Inc.

Page 136: Per-Phone Configuration - Phone1.Cfg
In the following table, x is the registration number. IP 300 and 301: x=1-2; IP 500 and 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1. Permitted Attribute Values Default Interpretation reg.x.displayName UTF-8 encoded Null Display name used for local user inter- string face as well as SIP signaling. Copyright © 2005 Polycom, Inc.
Page 137 Null Password to be used for authentication challenges for this registration. If non- Null, will override the “Reg Password x” parameter entered into the Authenti- cation submenu off of the Settings menu on the phone. Copyright © 2005 Polycom, Inc.
Page 138 = 6 on IP 600, max = 24 on IP 601 (without any Expansion Modules attached, only 6 line keys are available) The number of line keys on the phone to be associated with registration ‘x’. Copyright © 2005 Polycom, Inc.
Page 139: Calls
If set to 1, the DND feature will allow selection of DND on a per-registration basis. 4.6.2.2.2 Automatic Off-hook Call Placement <autoOffHook/> An optional per-registration feature is supported which allows automatic call place- ment when the phone goes off-hook. Copyright © 2005 Polycom, Inc.

Page 140 501: x=1-3; IP 600: x=1-6; IP 601: x=1-12; IP 4000: x=1 Permitted Values Attribute Default Interpretation call.serverMissedCall.x.enabled 0, 1 If set to 0, all missed-call events will increment the counter. If set to 1, only missed-call events sent by the server will increment the counter. Copyright © 2005 Polycom, Inc.
Page 141: Diversion
4.6.2.3.2 Busy <busy/> Calls can be automatically diverted when the phone is busy. Attribute Permitted Values Default Interpretation divert.busy.x.enabled 0, 1 If set to 1, calls will be forwarded on busy to the contact specified below. Copyright © 2005 Polycom, Inc.
Page 142 Forward-to contact used digits (the user part of a SIP URL) for calls forwarded due to or a string that constitutes a valid DND status, if Null SIP URL (6416 or 6416@poly- divert.x.contact will be com.com) used. Copyright © 2005 Polycom, Inc.
Page 143: Dial Plan
For more information, see bytes and 20 segments; a 4.6.1.2.1 Digit Map <digit- comma is also allowed; map/> on page 77. when reached in the digit map, a comma will turn dial tone back on. Copyright © 2005 Polycom, Inc.

Page 144 4.6.2.4.2.2 Emergency <emergency/> In the following attributes, y is the index of the emergency entry description and z is the index of the server associated with the emergency entry y. For each emergency Copyright © 2005 Polycom, Inc.
Page 145: Messaging
Messages key. The phone will act as if “Message Center” was chosen. See 3.6.1 Voicemail Integration on page 55. Instant Messages will still be accessible from the Main Menu. Copyright © 2005 Polycom, Inc.

Page 146: Network Address Translation
NAT device can be sent to port 5432 on an internal device, but not port 1234. Permitted Attribute Values Default Interpretation nat.ip dotted-deci- Null IP address to advertise within SIP signaling - mal IP should match the external IP address used by the address NAT device. Copyright © 2005 Polycom, Inc.

Page 147 Prot.local.signalPort in sip.cfg. nat.mediaPortStart 1024 to Null If non-Null, this attribute will be used to set the 65536 initially allocated RTP port, overriding the value set for tcpIpApp.port.rtp.mediaPortRangeStart in sip.cfg. See 4.6.1.10.3.1 RTP <RTP/> on page 106. Copyright © 2005 Polycom, Inc.
Page 149: Session Initiation Protocol (Sip)
Session Initiation Protocol (SIP) - Specific Event Notification RFC 3515 The Session Initiation Protocol (SIP) Refer Method draft-ietf-sip-cc-transfer-05.txt SIP Call Control - Transfer draft-ietf-sip-replaces-03.txt The Session Initiation Protocol (SIP) “Replaces” Header 5.1.2 Request Support Method Supported Notes REGISTER INVITE CANCEL Copyright © 2005 Polycom, Inc.
Page 150: Header Support
In the following table, a “Yes” in the Supported column means the header is sent and properly parsed. Header Supported Notes Accept Accept-Encoding Accept-Language Alert-Info Allow Allow-Events Authentication-Info Authorization Call-ID Call-Info Contact Content-Disposition Content-Encoding Content-Language Content-Length Content-Type CSeq Date Diversion Copyright © 2005 Polycom, Inc.
Page 151 Session Initiation Protocol (SIP) Header Supported Notes Error-Info Event Expires From In-Reply-To Max-Forwards Min-Expires Min-SE MIME-Version Organization P-Asserted-Identity P-Preferred-Identity Priority Proxy-Authenticate Proxy-Authorization Proxy-Require RAck Record-Route Refer-To Referred-By Remote-Party-ID Replaces Reply-To Require Retry-After Route RSeq Server Session-Expires Subject Subscription-State Copyright © 2005 Polycom, Inc.
Page 152: Response Support
5.1.4.1 1xx Responses - Provisional Response Supported Notes 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress 5.1.4.2 2xx Responses - Success Response Supported Notes 200 OK 202 Accepted In REFER transfer. Copyright © 2005 Polycom, Inc.
Page 153: 3Xx Responses - Redirection
407 Proxy Authentication Required 408 Request Timeout 410 Gone 413 Request Entity Too Large 414 Request-URI Too Long 415 Unsupported Media Type 416 Unsupported URI Scheme 420 Bad Extension 421 Extension Required 423 Interval Too Brief Copyright © 2005 Polycom, Inc.
Page 154: 5Xx Responses - Server Failure
502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported 513 Message Too Large 5.1.4.6 6xx Responses - Global Failure Response Supported Notes 600 Busy Everywhere 603 Decline 604 Does Not Exist Anywhere Copyright © 2005 Polycom, Inc.
Page 155: Hold Implementation
The phone supports transfer using the REFER method specified in draft-ietf-sip-cc- transfer-05 and RFC 3515. 5.1.8 Third Party Call Control The phone supports the delayed media negotiations (INVITE without SDP) associated with third party call control applications. Copyright © 2005 Polycom, Inc.
Page 156: Protocol Extensions
RFC 2976, the phone does not generate INFO requests, but will issue a final response upon receipt. No INFO message bodies are parsed. MESSAGE Final response is sent upon receipt. Message bodies of type text/plain are sent and received. UPDATE Copyright © 2005 Polycom, Inc.
Page 157: Sip For Instant Messaging And Presence Leveraging Extensions (Simple)
The phone supports bridged line appearances (BLA) using the SUBSCRIBE-NOTIFY method in the “SIP Specific Event Notification” framework (RFC 3265). The events used are: • “dialog” for bridged line appearance subscribe and notify Copyright © 2005 Polycom, Inc.
Page 159: Appendix 1
Equifax Premium CA Equifax Secure CA GTE CyberTrust Global Root GTE CyberTrust Japan Root CA GTE CyberTrust Japan Secure Server CA GTE CyberTrust Root 2 GTE CyberTrust Root 3 GTE CyberTrust Root 4 GTE CyberTrust Root 5 Copyright © 2005 Polycom, Inc.
Page 160 Verisign Class 2 Public Primary Certification Authority - G2 Verisign Class 2 Public Primary Certification Authority - G3 Verisign Class 3 Public Primary Certification Authority Verisign Class 3 Public Primary Certification Authority - G2 Verisign Class 3 Public Primary Certification Authority - G3 Copyright © 2005 Polycom, Inc.
Page 161 ® Administrator Guide - SoundPoint IP / SoundStation Appendix 1 Verisign Class 4 Public Primary Certification Authority - G2 Verisign Class 4 Public Primary Certification Authority - G3 Verisign/RSA Commercial CA Verisign/RSA Secure Server CA Copyright © 2005 Polycom, Inc.
Page 163: Appendix 2
3. All advertising materials mentioning features or use of this software must display the following acknowledgment: "This product includes software developed by the OpenSSL Project for use in the OpenSSL Toolkit. (http://www.openssl.org/)" 4. The names "OpenSSL Toolkit" and "OpenSSL Project" must not be used to endorse Copyright © 2005 Polycom, Inc.
Page 164 2. Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. 3. All advertising materials mentioning features or use of this software must display Copyright © 2005 Polycom, Inc.
Page 165 2. Altered source versions must be plainly marked as such, and must not be misrepre- sented as being the original software. 3. This notice may not be removed or altered from any source distribution. Jean-loup Gailly Mark Adler jloup@gzip.org madler@alumni.caltech.edu Copyright © 2005 Polycom, Inc.
Page 166 Except as contained in this notice, the name of a copyright holder shall not be used in advertising or otherwise to promote the sale, use or other dealings in this Software without prior written authorization of the copyright holder. Copyright © 2005 Polycom, Inc.

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Polycom SoundPoint IP 601 Administrator's Manual

1 Overview

2 Installation and Operation

3 Features

4 Optimization

5 Session Initiation Protocol (SIP)

6 Appendix 1

7 Appendix 2

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