Polycom Version 1.4.x 17 3.1.21 Bridged Line Appearances

Administrator Guide - SoundPoint® IP / SoundStation® IP Features

32 Copyright © 2004 Polycom, Inc.

3.1.21 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 which 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 which binds the appearances together logi-

cally and looks after the necessary state notifications and performs an access control

Local

Web Se rve r

(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 via the

Reset User Settings 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 .......................................................... 17 Page 4 Optimization ....................................................57 5 Session Initiation Protocol (SIP)........................... 131 Page Page Administrator Guide - SoundPoint IP / SoundStation IP Overview Copyright 2004 Polycom, Inc. 1 1 Overview PC PC Modem Bank Hub Voice Bridge 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 Local User Interface Setup Menus 2.2.1.2.1 Main Menu The DHCP and Server sub-menus may be accessed from the main setup menu. Name Possible ValuesaDescription 2.2.1.2.2 DHCP Menu Name Possible Values Description 2.2.1.2.3 Server Menu Name Name Possible Values Description Possible Values Description 2.2.1.3 Reset to Factory Defaults 2.2.2 Application Configuration 2.2.2.1 Centralized Configuration 2.2.2.1.1 FTP vs. TFTP 2.2.2.1.2 Configuration Files Page 2.2.2.1.2.2 Application Configuration Files Note Category Description Example 2.2.2.1.3 Deploying a Boot Server for the Phones Step: Instructions: 2.2.2.2 Local Phone Configuration Step: Instructions: 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 Administrator Guide - SoundPoint IP / SoundStation IP Features * SoundPoint IP 300 SIP Key Layout 121110 * SoundPoint IP 500 SIP Key Layout * Copyright 2004 Polycom, Inc. 21 SoundPoint IP 600 SIP Key Layout SoundStation IP 4000 SIP Key Layout Key ID IP 300 Function IP 500 Function IP 600 Function 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 Administration: Distinctive Incoming Call Treatment For more information, see 3.1.16 Local Contact Directory on page 27. 3.1.13 Distinctive Ringing There are three aspects to Distinctive Ringing: 3.1.13 Distinctive Call Waiting 3.1.14 Do-Not-Disturb 3.1.15 Handset, Headset, and Speakerphone 3.1.16 Local Contact Directory 3.1.16.1 Local Contact Directory File Format An example local contact directory is shown. Look to the table for an explanation of each element. Local Contact Directory File example: Element Permitted Values Interpretation 3.1.17 Local Digit Map 3.1.18 Microphone Mute 3.1.19 Multiple Call Appearances 3.1.20 Shared Call Appearances 3.1.21 Bridged Line Appearances 3.1.22 Customizable Fonts and Indicators 3.1.23 Soft Key-Driven User Interface The user interface makes extensive use of intuitive, context-sensitive soft key menus. 3.1.24 Speed Dial 3.1.25 Time and Date Display 3.1.26 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 Local Conference Mixing 3.3.4 Voice Activity Detection (VAD) 3.3.5 DTMF Tone Generation 3.3.6 DTMF Event RTP Payload 3.3.7 Acoustic Echo Cancellation (AEC) Algorithm MIME Type Ref. Bit Rate 3.3.8 Audio Codecs The following table summarizes the phones audio codec support: Sample Rate Frame Size 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 Name Range Note 3.5.2 Downloadable Fonts 3.5.3 Synthesized Call Progress Tones Note 3.6 Advanced Server Features 3.6.1 Voicemail Integration waiting signal from a voicemail server will trigger the message-waiting indicator to flash. 3.6.2 Multiple Registrations use for outgoing calls or which registration to use when initiating new instant message dialogs. 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 3.7 Accessory Internet Features 3.7.1 MicroBrowser 3.8 Security Features 3.8.1 Local User and Administrator Privilege Levels 3.8.2 Incoming Signaling Validation Three optional levels of security are provided for validating incoming network signal- ing: source IP address validation digest authentication both Page 4 Optimization 4.1 Ethernet Switch 4.2 Application Network Setup 4.2.1 RTP Ports 4.2.2 Working with Network Address Translation (NAT) 4.3 Updating and Rebooting 4.4 Event Logging Log files uploaded in this manner are named: <Ethernet address>-now-boot.log <Ethernet address>-now-app.log 4.5 Audio Quality Issues and VLANs 4.5.1 IP TOS 4.5.2 IEEE 802.1p/Q Page Page 4.6 Configuration Files 4.6.1 Core Configuration - ipmid.cfg 4.6.1.1 Localization <localization/> 4.6.1.1.1 Multilingual <multilingual/> Follow these steps to add new languages to those included with the distribution: 4.6.1.1.2 Date and Time <datetime/> 4.6.1.1.3 Call Progress Tones <callProgTones> 4.6.1.2 User Preferences <user_preferences/> 4.6.1.3 Tones <tones/> This section describes configuration items for the tone resources available in the phone. 4.6.1.3.1 Dual Tone Multi-Frequency <DTMF/> 4.6.1.3.2 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 In the following table, x is the chord-set number and cat is one of callProg, ringer, or misc. 4.6.1.4 Sampled Audio for Sound Effects <sampled_audio/> The following sampled audio WAVE file (.wav) formats are supported: mono 8 kHz G.711 -Law G.711 A-Law L16/1600 (16-bit, 16 kHz sampling rate, mono) Permitted The following table defines the default usage of the sampled audio files with the phone: 4.6.1.5 Sound Effects <sound_effects/> Sampled Audio File Default use within phone (pattern reference) Permitted 4.6.1.5.1 Patterns <patterns/> Instruction Meaning Example 4.6.1.5.1.1 Call Progress Patterns The following table maps call progress patterns to their usage within the phone. Call progress pattern number Use within phone 4.6.1.5.1.2 Ringer Patterns The following table maps ringer pattern numbers to their default descriptions. Ringer pattern number Default description 4.6.1.5.2 Ring type <ringType/> 4.6.1.6 Voice Settings <voice/> 4.6.1.6.1 Voice Coding Algorithms <codecs/> Algorithm The following voice codecs are supported: MIME Type Label Bit Rate 4.6.1.6.1.1 Codec Preferences <preferences/> 4.6.1.6.1.2 Codec Profiles <profiles/> Permitted 4.6.1.6.2 Volume Persistence <volume/> 4.6.1.6.3 Gains <gains/> 4.6.1.6.4 Acoustic Echo Cancellation <AEC/> These settings control the performance of the speakerphone acoustic echo canceller. 4.6.1.6.5 Acoustic Echo Suppression <AES/> 4.6.1.6.6 Background Noise Suppression <NS/> These settings control the performance of the transmit background noise suppression feature. 4.6.1.6.7 Automatic Gain Control <AGC/> These settings control the performance of the transmit automatic gain control feature.7 4.6.1.6.8 Receive Equalization <RXEQ/> These settings control the performance of the handsfree receive equalizaton feature. 4.6.1.6.9 Transmit Equalization <TXEQ/> These settings control the performance of the handsfree transmit equalization feature. 4.6.1.6.10 Voice Activity Detection <VAD/> 4.6.1.7 Quality of Service <QOS/> 4.6.1.7.1 Ethernet IEEE 802.1p/Q <Ethernet/> These default parameter values are used for all packets which are not set explicitly. 4.6.1.7.2 IP TOS <IP/> These parameters apply to RTP packets. These settings control the type of service field in outgoing packets. These parameters apply to call control packets, such as the network protocol signaling. 4.6.1.8 Basic TCP/IP <TCP_IP/> 4.6.1.8.1 Network Monitoring <netMon/> 4.6.1.8.2 Time Synchronization <SNTP/> stop DST on the last Sunday in October at 2 am. Permitted 4.6.1.8.3 port <port/> 4.6.1.8.3.1 RTP <RTP/> Permitted 4.6.1.9 Web Server <HTTPD/> 4.6.1.9.1 Configuration <cfg/> 4.6.1.10 Call Handling Configuration <call/> 4.6.1.10.1 Hold, Local Reminder <localReminder/> 4.6.1.11 Directory <directory/> 4.6.1.12 Fonts <font/> Page 4.6.1.12.1 IP_500 font <IP_500/> 4.6.1.12.2 IP_600 font <IP_600/> 4.6.1.13 Keys <keys/> These settings control the scrolling behavior of keys and can be used to change key functions. The following table lists the functions that are available: Page Page 4.6.1.14 Bitmaps <bitmaps/> Bitmaps used by the phone are defined in this section. 4.6.1.14.1 Platform <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> In the following table, x=IP_300, IP_500, IP_600, or IP_4000, y is the bitmap number. 4.6.1.15 Indicators <indicators/> 4.6.1.15.1 Animations <Animations/> <IP_300/>, <IP_500/>, <IP_600/> and <IP_4000/> 4.6.1.15.2 Patterns <Patterns/> 4.6.1.15.3 Classes <Classes/> 4.6.1.15.4 Assignments <Assignments/> 4.6.1.16 Event Logging <logging/> The event logging system supports the following classes of events: Warning! Level Interpretation Permitted 4.6.1.16.1 Basic Logging <level/><change/> and <render/> Three formats are available for the event timestamp: 4.6.1.16.2 Scheduled Logging Parameters <scheduled/> 4.6.1.17 Security <security/> These settings affect security aspects of the phone. 4.6.1.17.1 Password Lengths <pwd/><length> 4.6.1.18 Provisioning <provisioning/> These settings control aspects of the phones boot server provisioning system. 4.6.1.19 RAM Disk <RAMdisk/> 4.6.1.20 Request <request/> 4.6.1.20.1 Delay <delay/> 4.6.1.21 Feature <feature/> 4.6.1.22 Resource <resource/> These settings control the maximum size or an external resource retrieved at run time. 4.6.1.22.2 quotas <quotas/> 4.6.1.22.1 finder <finder/> 4.6.1.23 MicroBrowser <microbrowser/> 4.6.1.23.1 Main Browser <main/> This setting controls the home page used by the MicroBrowser when that function is selected. 4.6.1.23.2 Browser Limits <limits/> Page 4.6.2 SIP Configuration - sip.cfg 4.6.2.1 Protocol <volpProt/> 4.6.2.1.1 Local <local/> 4.6.2.1.2 Server <server/> 4.6.2.1.3 SIP <SIP/> 4.6.2.1.3.1 Outbound Proxy <outboundProxy/> 4.6.2.1.3.2 Alert Information <alertInfo/> Permitted 4.6.2.1.3.3 Request Validation <requestValidation/> Page 4.6.2.1.4 Dial Plan <dialplan/> 4.6.2.1.4.1 Digit Map <digitmap/> 4.6.2.1.4.2 Routing <routing/> Permitted 4.6.2.1.4.2.1 Server <server/> 4.6.2.1.5 SIP Logging <logging/> <level/> <change/> 4.6.2.1.4.2.2 Emergency <emergency/> Permitted 4.6.3 Per-phone Configuration - phone1.cfg 4.6.3.1 Registration <reg/> Page 4.6.3.2 Calls <call/> 4.6.3.2.1 Do Not Disturb <donotdisturb/> 4.6.3.2.2 Automatic Off-hook Call Placement <autoOffHook/> 4.6.3.2.3 Missed Call Configuration <serverMissedCall/> In the following table, x is the registration number. IP 300: x=1-2; IP 500: x=1-3; IP 4.6.3.3 Diversion <divert/> 4.6.3.3.1 Forward All <fwd/> 4.6.3.3.2 Busy <busy/> The phone can automatically divert calls after a period of ringing. Calls can be automatically diverted when the phone is busy. 4.6.3.3.3 No Answer <noanswer/> 4.6.3.4 Dial Plan <dialplan/> 4.6.3.4.1 Digit Map <digitmap/> 4.6.3.4.2 Routing <routing/> 4.6.3.4.2.1 Server <server/> Permitted 4.6.3.4.2.2 Emergency <emergency/> 4.6.3.5 Messaging <msg/> Message-waiting indication is supported on a per-registration basis. Permitted 4.6.3.5.1 Message Waiting Indicator <mwi/> 4.6.3.6 Network Address Translation <nat/> Page 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 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 Page 5.2 Protocol Extensions The phone supports the following SIP protocol extensions. 5.2.2 Request Support 5.2.1 RFC and Internet Draft Support ID Title 5.2.3 SIP for Instant Messaging and Presence Leverag- ing Extensions (SIMPLE) 5.2.4 Shared Call Appearance Signaling 5.2.5 Bridged Line Appearance Signaling