Polycom 1725-11530-200 Rev A1 3.1.23 Bridged Line Appearances

Administrator’s Guide - SoundPoint® IP / SoundStation® IP Features

Copyright © 2006 Polycom, Inc. 45

3.1.23 Bridged Line Appearances

Calls and lines on multiple phones can be logically related to each other. A call that is

active on one phone will be presented visually to phones that share that line. Mutual

exclusion features emulate traditional PBX or key system privacy for shared calls.

Incoming calls can be presented to multiple phones simultaneously. This feature is

dependent on support from a SIP server that binds the appearances together logically

and looks after the necessary state notifications and performs an access control func-

Local

Web Se rver

(if enabled) Specify line-seize subscription period. Navigate to:

http://<phoneIPAddress>/appConf.htm#se

Specify standard or non-standard behavior for processing

line-seize subscription for mutual exclusion feature. Nav-

igate to:

http://<phoneIPAddress>/appConf.htm#ls

Specify per-registration line type (private or shared) and

line-seize subscription period if using per-registration

servers, and whether diversion should be disabled on

shared lines. Navigate to:

http://<phoneIPAddress>/reg.htm

Changes are saved to local flash and backed up to <Ether-

net address>-phone.cfg on the boot server. They will per-

manently override global settings unless deleted through

the Reset Local Config menu selection.

Local Phone User

Interface Specify per-registration line type (private or shared) using

the SIP Configuration menu. Either the Web Server or the

boot server configuration files or the local phone user

interface should be used to configure registrations, not a

mixture of these options. When the SIP Configuration

menu is used, it is assumed that all registrations use the

same server.

Contents

Main Page Table of Contents 3 Features .......................................................... 27 Page 4 Optimization ....................................................77 Page 5 Session Initiation Protocol (SIP)........................... 163 6 Appendix 1...................................................... 173 7 Appendix 2...................................................... 177 Page 1 Overview Page 2 Installation and Operation 2.1 Installation Models 2.2 Installation Process 2.2.1 Basic Network Setup 2.2.1.1 DHCP or Manual TCP/IP Setup 2.2.1.2 Provisioning File Transfer Page 2.2.1.3 Local User Interface Setup Menus 2.2.1.3.1 Main Menu The DHCP and Server sub-menus may be accessed from the main setup menu. Name Possible ValuesaDescription 2.2.1.3.2 DHCP Menu Name Possible Values Description Name Possible Values Description 2.2.1.3.3 Server Menu 2.2.1.3.4 Ethernet Menu 2.2.1.4 Reset to Factory Defaults 2.2.2 Application Configuration 2.2.2.1 Centralized Configuration 2.2.2.1.1 Configuration Files Page Page Page Page Page 2.2.2.1.2 Boot Server Deployment for the Phones Page Page 2.2.2.2 Local Phone Configuration 2.2.3 Management of File Encryption and Decryption 2.2.3.1 Changing the Key on the Phone Page Page 3 Features 3.1 Basic Features 3.1.1 Call Log 3.1.2 Call Timer 3.1.3 Call Waiting 3.1.4 Called Party Identification 3.1.5 Calling Party Identification 3.1.6 Missed Call Notification 3.1.7 Configurable Feature Keys * 1 2 31 29 32 23 7 5 89 SoundPoint IP 430 Key Layout 2635 192021 * 131415 Administrators Guide - SoundPoint IP / SoundStation IP Features SoundPoint IP 600 and 601 Key Layout 41 35 SoundPoint IP 4000 Key Layout Key ID IP 300 & 301 IP 430 Function IP 500 & 501 IP 600 & 601 IP 4000 Function 3.1.8 Connected Party Identification 3.1.9 Context Sensitive Volume Control 3.1.10 Customizable Audio Sound Effects 3.1.11 Message Waiting Indication 3.1.12 Distinctive Incoming Call Treatment 3.1.13 Distinctive Ringing 3.1.14 Distinctive Call Waiting 3.1.15 Do-Not-Disturb 3.1.16 Handset, Headset, and Speakerphone 3.1.17 Local Contact Directory 3.1.17.1 Local Contact Directory File Format Local Contact Directory File example: An example local contact directory is shown. Look to the table for an explanation of each element. Element Permitted Values Interpretation 3.1.18 Local Digit Map Element Permitted Values Interpretation 3.1.19 Microphone Mute 3.1.20 Multiple Line Keys per Registration 3.1.21 Multiple Call Appearances 3.1.22 Shared Call Appearances Page 3.1.23 Bridged Line Appearances tion. For more information, refer to 5.2.5 Bridged Line Appearance Signaling on page 172. 3.1.24 Busy Lamp Field 3.1.25 Customizable Fonts and Indicators 3.1.26 Soft Key-Driven User Interface 3.1.27 Speed Dial 3.1.28 Time and Date Display Page 3.1.29 Idle Display Animation 3.2 Call Management Features 3.2.1 Automatic Off-hook Call Placement The phone supports an optional automatic off-hook call placement feature for each registration. 3.2.2 Call Hold 3.2.3 Call Transfer 3.2.4 Three-Way Conference, Local or Centralized 3.2.5 Call Diversion (Call Forward) 3.2.6 Directed Call Pick-up 3.2.7 Group Call Pick-up 3.2.8 Call Park / Retrieve 3.2.9 Last Call Return 3.3 Audio Processing Features 3.3.1 Low-Delay Audio Packet Transmission 3.3.2 Jitter Buffer and Packet Error Concealment 3.3.3 Voice Activity Detection 3.3.4 DTMF Tone Generation 3.3.5 DTMF Event RTP Payload 3.3.6 Acoustic Echo Cancellation (AEC) 3.3.7 Audio Codecs The following table summarizes the phones audio codec support: Algorithm MIME Type Ref. Bit Rate Sample Rate Frame Size Effective audio band- width 3.4 Presence and Instant Messaging Features 3.4.1 Presence 3.4.2 Instant Messaging 3.5 Localization Features 3.5.1 Multilingual User Interface Page 3.5.2 Downloadable Fonts 3.5.3 Synthesized Call Progress Tones 3.6 Advanced Server Features 3.6.1 Voice Mail Integration Page 3.6.2 Multiple Registrations Page 3.6.3 ACD login / logout 3.6.4 ACD agent available / unavailable 3.6.5 Server Redundancy 3.6.5.1 DNS SIP Server Name Resolution 3.6.6 Microsoft Office Live Communications Server 2005 Integration Page 3.6.6.1 Configuration File Changes Modify the phone1.cfg configuration file as follows: 3.7 Accessory Internet Features 3.7.1 MicroBrowser 3.8 Security Features 3.8.1 Local User and Administrator Privilege Levels 3.8.2 Custom Certificates 3.8.3 Incoming Signaling Validation Three optional levels of security are provided for validating incoming network signal- ing: source IP address validation digest authentication both 3.8.4 Configuration File Encryption Page 4 Optimization 4.1 Ethernet Switch 4.2 Application Network Setup 4.2.1 Real-Time Transport Protocol Ports 4.2.2 Working with Network Address Translation 4.3 Updating and Rebooting 4.4 Event Logging 4.5 Audio Quality Issues and VLANs 4.5.1 IP TOS 4.5.2 IEEE 802.1p/Q The phone will tag all Ethernet packets it transmits with an 802.1Q VLAN header for one Page 4.6 Configuration Files 4.6.1 SIP Configuration - sip.cfg 4.6.1.1 Protocol <volpProt/> 4.6.1.1.1 Local <local/> 4.6.1.1.2 Server <server/> Page Page 4.6.1.1.3 SDP <SDP/> 4.6.1.1.4 SIP <SIP/> Page Page 4.6.1.1.4.2 Alert Information <alertInfo/> Permitted 4.6.1.1.4.3 Request Validation <requestValidation/> Page 4.6.1.2 Dial Plan <dialplan/> 4.6.1.2.1 Digit Map <digitmap/> 4.6.1.2.2 Routing <routing/> 4.6.1.2.2.1 Server <server/> 4.6.1.2.2.2 Emergency <emergency/> 4.6.1.3 Localization <localization/> 4.6.1.3.1 Multilingual <multilingual/> To add new languages to those included with the distribution: 4.6.1.3.2 Date and Time <datetime/> 4.6.1.4 User Preferences <user_preferences/> 4.6.1.5 Tones <tones/> This section describes configuration items for the tone resources available in the phone. 4.6.1.5.1 Dual Tone Multi-Frequency <DTMF/> 4.6.1.5.2 Chord Sets <chord_sets/> All three blocks use the same chord set specification format. callProg (used for call progress sound effect patterns) ringer misc (miscellaneous) Permitted 4.6.1.6 Sampled Audio for Sound Effects <sampled_audio/> In the following table, x is the sampled audio file number. The following table defines the default usage of the sampled audio files with the phone: Sampled Audio File Default use within phone (pattern reference) 4.6.1.7 Sound Effects <sound_effects/> 4.6.1.7.1 Patterns <patterns/> Permitted Instruction Meaning Example Instruction Meaning Example Permitted 4.6.1.7.1.1 Call Progress Patterns The following table maps call progress patterns to their usage within the phone. 4.6.1.7.1.2 Ringer Patterns The following table maps ringer pattern numbers to their default descriptions. Call progress Ringer pattern number Default description The following table maps miscellaneous patterns to their usage within the phone. Miscellaneous Ringer pattern number Default description 4.6.1.7.2 Ring type <ringType/> In the following table, x is the ring class number. The x index needs to be sequential. Miscellaneous 4.6.1.8 Voice Settings <voice/> 4.6.1.8.1 Voice Coding Algorithms <codecs/> The following voice codecs are supported: 4.6.1.8.1.2 Codec Profiles <profiles/> Permitted 4.6.1.8.2 Volume Persistence <volume/> 4.6.1.8.3 Gains <gains/> Page 4.6.1.8.4 Acoustic Echo Cancellation <AEC/> These settings control the performance of the speakerphone acoustic echo canceller. 4.6.1.8.5 Acoustic Echo Suppression <AES/> These settings control the performance of the speakerphone acoustic echo suppressor. 4.6.1.8.6 Background Noise Suppression <NS/> These settings control the performance of the transmit background noise suppression feature. These settings control the performance of the transmit automatic gain control feature.7 4.6.1.8.7 Automatic Gain Control <AGC/> 4.6.1.8.8 Receive Equalization <RXEQ/> These settings control the performance of the receive equalizaton feature. 4.6.1.8.9 Transmit Equalization <TXEQ/> These settings control the performance of the hands-free transmit equalization feature. 4.6.1.8.10 Voice Activity Detection <VAD/> These parameters apply to RTP packets. 4.6.1.9 Quality of Service <QOS/> These settings control the Quality of Service (QOS) options. 4.6.1.9.1 Ethernet IEEE 802.1p/Q <Ethernet/> These settings control the 802.1p/Q user_priority field. These parameters apply to call control packets, such as the network protocol signaling. 4.6.1.9.2 IP TOS <IP/> These settings control the type of service field in outgoing packets. 4.6.1.9.2.2 Call Control <CallControl/> These parameters apply to call control packets, such as the network protocol signaling. Permitted 4.6.1.10 Basic TCP/IP <TCP_IP/> 4.6.1.10.1 Network Monitoring <netMon/> 4.6.1.10.2 Time Synchronization <SNTP/> Page Page 4.6.1.10.3 port <port/> 4.6.1.10.3.1 RTP <RTP/> Permitted 4.6.1.11 Web Server <HTTPD/> 4.6.1.12 Call Handling Configuration <call/> 4.6.1.11.1 Configuration <cfg/> 4.6.1.12.1 Shared Calls <shared/> 4.6.1.12.2 Hold, Local Reminder <hold/><localReminder/> 4.6.1.13 Directory <directory/> 4.6.1.14 Presence <presence/> 4.6.1.15 Fonts <font/> Page 4.6.1.15.1 IP_400 font <IP_400/> 4.6.1.15.2 IP_500 font <IP_500/> 4.6.1.15.3 IP_600 font <IP_600/> 4.6.1.16 Keys <keys/> 4.6.1.17 Bitmaps <bitmaps/> Function Function Bitmaps used by the phone are defined in this section. 4.6.1.17.1 Platform <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/> and <IP_4000/> 4.6.1.18 Indicators <indicators/> 4.6.1.18.1 Animations <Animations/> <IP_300/>, <IP_400/>, <IP_500/>, <IP_600/> and <IP_4000/> 4.6.1.18.2 Patterns <Patterns/> 4.6.1.18.3 Classes <Classes/> 4.6.1.18.4 Assignments <Assignments/> In the following table, x is the LED number. 4.6.1.19 Event Logging <logging/> The event logging system supports the following classes of events: Each event in the log contains the following fields separated by the | character: Three formats are available for the event timestamp: 4.6.1.19.1 Basic Logging <level/><change/> and <render/> Type Example Permitted Example: time stamp ID event class missed events text Page 4.6.1.19.2 Scheduled Logging Parameters <scheduled/> These settings affect security aspects of the phone. 4.6.1.20 Security <security/> 4.6.1.20.1 Encryption <encryption/> 4.6.1.20.2 Password Lengths <pwd/><length/> 4.6.1.21 Provisioning <provisioning/> These settings control aspects of the phones boot server provisioning system. 4.6.1.22 RAM Disk <RAMdisk/> 4.6.1.23 Request <request/> 4.6.1.23.1 Delay <delay/> 4.6.1.24 Feature <feature/> 4.6.1.25 Resource <resource/> These settings control the maximum size or an external resource retrieved at run time. 4.6.1.25.2 quotas <quotas/> 4.6.1.25.1 finder <finder/> 4.6.1.26 MicroBrowser <microbrowser/> 4.6.1.26.1 Idle Display <idleDisplay/> 4.6.1.26.2 Main Browser <main/> This setting controls the home page used by the MicroBrowser when that function is selected. 4.6.1.26.3 Browser Limits <limits/> Page 4.6.2 Per-phone Configuration - phone1.cfg 4.6.2.1 Registration <reg/> Page Page Page 4.6.2.2 Calls <call/> These sections describe call-oriented per-phone configuration items. 4.6.2.2.1 Do Not Disturb <donotdisturb/> 4.6.2.2.2 Automatic Off-hook Call Placement <autoOffHook/> 4.6.2.2.3 Missed Call Configuration <serverMissedCall/> 4.6.2.3 Diversion <divert/> 4.6.2.3.1 Forward All <fwd/> 4.6.2.3.2 Busy <busy/> The phone can automatically divert calls after a period of ringing. Calls can be automatically diverted when the phone is busy. 4.6.2.3.3 No Answer <noanswer/> 4.6.2.4 Dial Plan <dialplan/> 4.6.2.4.1 Digit Map <digitmap/> 4.6.2.4.2 Routing <routing/> 4.6.2.4.2.1 Server <server/> 4.6.2.4.2.2 Emergency <emergency/> 4.6.2.5 Messaging <msg/> Message-waiting indication is supported on a per-registration basis. 4.6.2.5.1 Message Waiting Indicator <mwi/> 4.6.2.6 Network Address Translation <nat/> 4.6.2.7 Attendant <attendant/> 4.6.2.8 Roaming Buddies <roaming_buddies/> This attribute is used in conjunction with Microsoft Office Live Communications Server 2005 only. 4.6.2.9 Roaming Privacy <roaming_privacy/> This attribute is used in conjunction with Microsoft Office Live Communications Server 2005 only. 5 Session Initiation Protocol (SIP) 5.1 Basic Protocols 5.1.2 Request Support 5.1.1 RFC and Internet Draft Support ID Title 5.1.3 Header Support Method Supported Notes Page 5.1.4 Response Support 5.1.4.2 2xx Responses - Success 5.1.4.1 1xx Responses - Provisional 5.1.4.3 3xx Responses - Redirection 5.1.4.4 4xx Responses - Request Failure Response Supported Notes 5.1.4.5 5xx Responses - Server Failure 5.1.4.6 6xx Responses - Global Failure 5.1.5 Hold Implementation 5.1.6 Reliability of Provisional Responses 5.1.7 Transfer 5.1.8 Third Party Call Control 5.2 Protocol Extensions The phone supports the following SIP protocol extensions. 5.2.1 RFC and Internet Draft Support ID Title 5.2.2 Request Support 5.2.3 SIP for Instant Messaging and Presence Leverag- ing Extensions 5.2.4 Shared Call Appearance Signaling 5.2.5 Bridged Line Appearance Signaling 6 Appendix 1 6.1 Trusted Certificate Authority List Page 6.2 Miscellaneous Administrative Tasks 6.2.1 Adding a Background Logo Page 7 Appendix 2 7.1 Third Party Software Attribution