Phone Administration
OpenStage SIP V3R3 for OpenScape Voice
Administration Manual
A31003-S2030-M100-10-76A9
Our Quality and Environmental Management Systems are implemented according to the requirements of the ISO9001 and
ISO14001 standards and are certified by an external certification company.
Copyright © Unify GmbH & Co. KG 01/2014
Hofmannstr. 51, 81379 Munich/Germany
All rights reserved.
Reference No.: A31003-S2030-M100-10-76A9
The information provided in this document contains merely general descriptions or characteristics of performance which in case of actual use do not always apply as described or which may change as a result of further development of the products.
An obligation to provide the respective characteristics shall only exist if expressly agreed in the terms of contract.
Availability and technical specifications are subject to change without notice.
Unify, OpenScape, OpenStage and HiPath are registered trademarks of Unify GmbH & Co. KG.
All other company, brand, product and service names are trademarks or registered trademarks of their respective holders. unify.com
bkTOC.fm
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1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
1.1 Important Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
1.2 Maintenance Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.3 About the Manual. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.4 Conventions for this Document . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.5 The OpenStage Family . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
1.5.1 OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
1.5.2 OpenStage 40 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
1.5.3 OpenStage 40 US . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
1.5.4 OpenStage 20 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
1.5.5 OpenStage 15 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
1.6 Administration Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
1.6.1 Web-based Management (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
1.6.2 DLS (Deployment Service) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
1.6.3 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
2 Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.2 Assembling and Installing the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
2.2.1 Shipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
2.2.2 Connectors at the bottom side . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
2.2.3 Assembly. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
2.2.4 Connecting the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
2.3 Quick Start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
2.3.1 Accessing the Web Interface (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
2.3.2 Set the Terminal Number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
2.3.3 Basic Network Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
2.3.4 DHCP Resilience. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
2.3.5 Date and Time / SNTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
2.3.6 SIP Server Address. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
2.3.7 Extended Network Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
2.3.8 Vendor Specific: VLAN Discovery And DLS Address . . . . . . . . . . . . . . . . . . . . . . . 29
2.3.8.1 Using a Vendor Class. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
2.3.8.2 Using Option #43 "Vendor Specific" . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
2.3.9 Registering at Phone Administration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
2.4 Startup Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 44
2.5 Cloud Deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
2.5.1 Process of Cloud Deployment. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
2.5.2 Aborting cloud deployment process by User. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
2.5.3 Re-trigger cloud deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
2.5.4 Deployment errors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
3 Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
3.1 Access via Local Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
3.2 LAN Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
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3.2.1 LAN Port Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
3.2.2 VLAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
3.2.2.1 Automatic VLAN discovery using LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . . 57
3.2.2.2 Automatic VLAN discovery using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
3.2.2.3 Manual configuration of a VLAN ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
3.2.3 LLDP-MED Operation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
3.3 IP Network Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
3.3.1 Quality of Service (QoS). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
3.3.1.1 Layer 2 / 802.1p. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
3.3.1.2 Layer 3 / Diffserv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
3.3.2 Protocol Mode IPv4/IPv6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
3.3.3 Use DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
3.3.4 IP Address - Manual Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
3.3.4.1 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
3.3.5 Default Route/Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
3.3.6 Specific IP Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
3.3.7 DNS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
3.3.7.1 DNS Domain Name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75
3.3.7.2 DNS Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
3.3.7.3 Terminal Hostname . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
3.3.8 Configuration & Update Service (DLS). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
3.3.9 SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
3.4 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
3.4.1 Speech Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
3.4.1.1 General Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
3.4.1.2 MIKEY Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
3.4.1.3 SDES Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
3.4.2 Access Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
3.4.3 Security Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
3.4.4 Security-Related Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
3.4.5 Password Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
3.4.5.1 General Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
3.4.5.2 Admin Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
3.4.5.3 User Policy. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
3.4.5.4 Character Set. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
3.4.5.5 Change Admin and User password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
3.4.6 Certificate Policy. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
3.4.6.1 Online Certificate Check . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
3.4.6.2 Server Authentication Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
3.5 System Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
3.5.1 Terminal and User Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
3.5.1.1 Terminal Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
3.5.1.2 Display Identity. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
3.5.2 Emergency and Voice Mail. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
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3.5.3 Energy Saving (OpenStage 40/60/80) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
3.5.4 Call logging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
3.5.4.1 Logging of Missed Calls (via User menu) . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
3.5.5 Date and Time. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
3.5.5.1 SNTP is Available, but no Automatic Configuration by DHCP server. . . . . . . 103
3.5.5.2 No SNTP Server Available . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
3.5.6 SIP Addresses and Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
3.5.6.1 SIP Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
3.5.6.2 SIP Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
3.5.7 SIP Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
3.5.8 SIP Communication. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
3.5.8.1 Outbound Proxy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
3.5.8.2 SIP Transport Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
3.5.8.3 Media/SDP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
3.5.9 SIP Session Timer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114
3.5.10 Resilience and Survivability. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
3.5.10.1 TLS Connectivity Check . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 117
3.5.10.2 Response Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
3.5.10.3 Non-INVITE Transaction Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
3.5.10.4 Maximum Registration Backoff Timer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
3.5.10.5 Backup SIP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
3.6 Feature Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 124
3.7 Feature Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
3.7.1 Allow Refuse . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
3.7.2 Hot/Warm Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
3.7.3 Initial Digit Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
3.7.4 Group Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
3.7.4.1 Feature Code . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
3.7.4.2 Pickup alert. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
3.7.5 Call Transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
3.7.5.1 Transfer on Ring. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
3.7.5.2 Transfer on Hangup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
3.7.6 Callback URIs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
3.7.6.1 Call Completion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
3.7.7 Message Waiting Address. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139
3.7.8 Indicate Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 140
3.7.9 System Based Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142
3.7.10 Server Based Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 143
3.7.11 uaCSTA Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 145
3.7.12 Local Menu Timeout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
3.7.13 Call Recording. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148
3.8 Free Programmable Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 150
3.8.1 Clear (no feature assigned). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
3.8.2 Selected Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
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3.8.3 Repeat Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
3.8.4 Call Forwarding (Standard) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
3.8.5 Call Forwarding by Call Type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
3.8.6 Ringer Off . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
3.8.7 Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
3.8.8 Alternate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 155
3.8.9 Blind Call Transfer / Move Blind . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
3.8.10 Transfer Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
3.8.11 Deflect a Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
3.8.12 Shift Level. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
3.8.13 Phone-Based Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
3.8.14 Accept Call via Headset (OpenStage 40/60/80). . . . . . . . . . . . . . . . . . . . . . . . . 158
3.8.15 Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
3.8.16 Group Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
3.8.17 Repertory Dial. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 159
3.8.18 Hunt Group: Send Busy Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
3.8.19 Mobile User Logon . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
3.8.20 Directed Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
3.8.21 Callback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
3.8.22 Cancel Callbacks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
3.8.23 Consultation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
3.8.24 Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162
3.8.25 Call recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
3.8.26 Auto Answer With Zip Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
3.8.27 Server Feature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
3.8.28 BLF Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
3.8.29 Start Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165
3.8.30 Send Request via HTTP/HTTPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165
3.8.31 Built-in Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
3.8.32 2nd Alert . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
3.8.33 Start Phonebook (OpenStage 40/15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168
3.8.34 Show phone screen (OpenStage 15 and OpenStage 40 only) . . . . . . . . . . . . . 169
3.8.35 Mute (OpenStage 15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
3.8.36 Release (OpenStage 15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
3.9 Preset Function Keys (OpenStage 40 US only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
3.10 Fixed Function Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
3.10.1 Fixed Function Keys on OpenStage 40 US . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
3.11 Multiline Appearance/Keyset. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
3.11.1 Line key configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
3.11.2 Configure Keyset Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177
3.11.3 Line Preview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183
3.11.3.1 Preview and Preselection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
3.11.4 Immediate Ring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
3.11.5 Direct Station Select (DSS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
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3.11.5.1 General DSS Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186
3.11.5.2 Settings for a DSS key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
3.11.6 Distinctive Ringers per Keyset Lines. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
3.12 Key Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
3.13 Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194
3.13.1 Canonical Dialing Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 194
3.13.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 198
3.13.3 Phone location . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
3.13.4 Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
3.14 Ringer Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
3.14.1 Distinctive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 204
3.14.2 Map to Specials. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
3.14.3 Special Ringers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 207
3.15 Mobility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 210
3.16 Transferring Phone Software, Application, and Media Files . . . . . . . . . . . . . . . . . . . 212
3.16.1 FTP/HTTPS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 212
3.16.2 Common FTP/HTTPS Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 212
3.16.3 Phone Software. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
3.16.3.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 214
3.16.3.2 Download/Update Phone Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 217
3.16.4 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
3.16.4.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
3.16.4.2 Download Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
3.16.5 Picture Clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 221
3.16.5.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 221
3.16.5.2 Download Picture Clip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223
3.16.6 LDAP Template. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
3.16.6.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
3.16.6.2 Download LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 226
3.16.7 Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
3.16.7.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
3.16.7.2 Download Logo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
3.16.8 Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
3.16.8.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
3.16.8.2 Download Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 232
3.16.9 Ringer File. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
3.16.9.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 234
3.16.9.2 Download Ringer File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236
3.16.10 Dongle Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
3.16.10.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
3.16.10.2 Download Dongle Key File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 239
3.17 Corporate Phonebook: Directory Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
3.17.1 LDAP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
3.18 Speech. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 242
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3.18.1 RTP Base Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 242
3.18.2 Codec Preferences. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
3.18.3 Audio Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 245
3.19 Applications. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
3.19.1 XML Applications/Xpressions (OpenStage 60/80) . . . . . . . . . . . . . . . . . . . . . . . 246
3.19.1.1 Setup/Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
3.19.1.2 HTTP Proxy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
3.19.1.3 Modify an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 254
3.19.1.4 Remove an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 255
3.19.1.5 Application Start by Programmable Key . . . . . . . . . . . . . . . . . . . . . . . . . . . 255
3.20 Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 256
3.21 Troubleshooting: Lost Password. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
3.22 Restart Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
3.23 Factory Reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
3.24 SSH – Secure Shell Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
3.25 Display License Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 259
3.26 Diagnostics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260
3.26.1 Display General Phone Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 260
3.26.2 View Diagnostic Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 261
3.26.3 User Access to Diagnostic Information. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
3.26.4 Diagnostic Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
3.26.5 LAN Monitoring. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
3.26.6 LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266
3.26.7 IP Tests . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
3.26.8 Process and Memory Information. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
3.26.9 Fault Trace Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 271
3.26.10 EasyTrace Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
3.26.10.1 Bluetooth Handsfree . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
3.26.10.2 Bluetooth Headset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
3.26.10.3 Call Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 279
3.26.10.4 Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 280
3.26.10.5 Call Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 280
3.26.10.6 DAS Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
3.26.10.7 DLS Data Errors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 281
3.26.10.8 Help Application. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
3.26.10.9 Key Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
3.26.10.10 LAN Connectivity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
3.26.10.11 Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
3.26.10.12 Mobility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
3.26.10.13 Phone administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 284
3.26.10.14 LDAP Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
3.26.10.15 Local Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
3.26.10.16 Server based applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
3.26.10.17 Sidecar. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286
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3.26.10.18 SIP standard multiline. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
3.26.10.19 SIP standard singleline. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 287
3.26.10.20 Speech . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
3.26.10.21 Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 288
3.26.10.22 USB Backup/Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
3.26.10.23 Voice Dialling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
3.26.10.24 Web Based Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
3.26.10.25 802.1x problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
3.26.10.26 No Tracing for All Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 291
3.26.11 Bluetooth Advanced Traces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 292
3.26.12 QoS Reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 293
3.26.12.1 Conditions and Thresholds for Report Generation . . . . . . . . . . . . . . . . . . . 293
3.26.12.2 View Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 296
3.26.13 Core dump . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300
3.26.14 Remote Tracing – Syslog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
3.26.15 HPT Interface (For Service Staff) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 302
3.27 Bluetooth (OpenStage 60/80) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 303
3.28 MWI LED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 305
3.29 Missed Call LED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
3.30 Impact Level Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 308
4 Technical Reference. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
4.1 Menus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
4.1.1 Web Interface Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
4.1.1.1 Menu Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 311
4.1.1.2 Web Pages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 315
4.1.2 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 360
4.2 Default Port List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 372
4.3 Troubleshooting: Error Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 374
5 Examples and HowTos. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
5.1 Canonical Dialing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
5.1.1 Canonical Dialing Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 376
5.1.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 377
5.1.2.1 Conversion examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 378
5.2 How to Create Logo Files for OpenStage Phones. . . . . . . . . . . . . . . . . . . . . . . . . . . . 380
5.2.1 For OpenStage 40. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 380
5.2.2 For OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 381
5.3 How to Set Up the Corporate Phonebook (LDAP). . . . . . . . . . . . . . . . . . . . . . . . . . . . 384
5.3.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 384
5.3.2 Create an LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 385
5.3.3 Load the LDAP Template onto the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
5.3.4 Configure LDAP Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
5.3.5 Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
5.4 An LLDP-Med Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
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Nur für den internen Gebrauch
5.5 Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
5.5.1 Introduction. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
5.5.2 Dial Plan Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
5.5.3 How To Set Up And Deploy A Dial Plan. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 401
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408
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Overview
Important Notes
1 Overview
1.1
Important Notes
7
Do not operate the equipment in environments where there is a danger of explosions.
7
For safety reasons the phone should only be operating using the supplied plug in power unit.
7
Use only original accessories. The use of other accessories may be hazardous and will render the warranty, extended manufacturer’s liability and the CE marking invalid.
7
Never open the telephone or add-on equipment. If you encounter any problems, contact System Support.
Installation requirement for USA, Canada, Norway, Finland and Sweden: Connection to networks which use outside cables is prohibited. Only in-house networks are permitted.
7
For USA and Canada only:
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
• Reorient or relocate the receiving antenna.
• Increase the separation between the equipment and receiver.
• Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
• Consult the dealer or an experienced radio/TV technician for help.
This product is a UL Listed Accessory, I.T.E., in U.S.A. and Canada.
This equipment also complies with the Part 68 of the FCC Rules and the Industrie
Canada CS-03.
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Overview
Maintenance Notes
1.2
Maintenance Notes
7
Do not operate the telephone in environments where there is a danger of explosions.
7
7
Use only original accessories from Siemens Enterprise Communications GmbH &
Co. KG. Using other accessories may be dangerous, and will invalidate the warranty and the CE mark.
Never open the telephone or a key module. If you encounter any problems, contact
System Support.
1.3
About the Manual
The instructions within this manual will help you in administering and maintaining the Open-
Stage phone. The instructions contain important information for safe and proper operation of the phones. Follow them carefully to avoid improper operation and get the most out of your multi-function telephone in a network environment.
This guide is intended for service providers and network administrators who administer VoIP services using the OpenStage phone and who have a fundamental understanding of VoIP, SIP, and IP networking. The tasks described in this guide are not intended for end users. Many of these tasks affect the ability of a phone to function on the network and require an understanding of IP networking and telephony concepts.
These instructions are laid out in a user-oriented manner, which means that you are led through the functions of the OpenStage phone step by step, wherever expedient. For the users, a separate manual is provided.
You can find further information on the official Unify website ( http://www.unify.com/.
) and on the Unify Wiki ( http://wiki.unify.com/.
).
1.4
Conventions for this Document
The terms for parameters and functions used in this document are derived from the web interface (WBM). In some cases, the phone’s local menu uses shorter, less specific terms and abbreviations. In a few cases the terminologies differ in wording. If so, the local menu term is added with a preceding "/".
For the parameters described in this document, a WBM screenshot and the path in the local phone menu is provided. All WBM screenshots are taken from OpenStage 60/80. As some
WBM input masks have been changed with firmware updates, the screenshots are selected after the following rules:
• If a later version contains more or less parameters compared to previous software versions, the screenshot of the older version is shown.
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Overview
Conventions for this Document
• If the title of a mask (e.g. "Pixel saver" vs. "Energy saving") or the name of a parameter
(e.g. "Time Zone" vs. "DST zone") has changed, the later version is shown.
• If a parameter has moved from one mask to another, both older and later versions are shown. The same is true for the local menu paths.
This document describes the software version V3R3.
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Overview
The OpenStage Family
1.5
1.5.1
The OpenStage Family
OpenStage 60/80
1
9
2
6
3
5
7
8
4
1
1
2
3
4
5
6
7
8
9
10
Tabelle 1-1
With the handset, the user can pick up and dial calls in the usual manner.
The graphic display provides intuitive support for telephone operation.
The mode keys provide easy access to the phone’s applications.
With the TouchGuide, the user/administrator can navigate in the phone functions, applications, and configuration menus.
The free programmable keys enable the user to customize the telephone in line with his/her personal needs by assigning individual phone numbers and functions.
The fixed function keys provide access to frequently used telephony functions.
With the audio keys, the user can control the audio settings.
With the TouchSlider, the user can adjust the volume, e.g. of ringtones.
Inbound calls are visually signaled via the alert bar.
The keypad is used for entering phone numbers and text.
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Overview
The OpenStage Family
1.5.2
OpenStage 40
7
2
1
3
5
6
4
8
4
5
1
2
3
6
7
8
Tabelle 1-2
With the handset, the user can pick up and dial calls in the usual manner.
The graphic display provides intuitive support for telephone operation.
The fixed function keys provide access to frequently used telephony functions.
With the 5-way navigator, the user/administrator can navigate in the various phone functions, applications, and configuration menus.
The free programmable keys enable the user to customize the telephone in line with his/her personal needs by assigning individual phone numbers and functions.
With the audio keys, the user can control the audio settings.
Inbound calls are visually signaled via the alert bar.
The keypad is used for entering phone numbers and text.
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Overview
The OpenStage Family
1.5.3
OpenStage 40 US
7
2
3
5
6
4
8
4
5
1
2
3
6
7
8
Tabelle 1-3
With the handset, the user can pick up and dial calls in the usual manner.
The graphic display provides intuitive support for telephone operation.
The fixed function keys provide access to frequently used telephony functions.
With the 5-way navigator, the user/administrator can navigate in the various phone functions, applications, and configuration menus.
The OpenStage 40 US telephone comes with six programmable lit sensor keys, preset to the following factory settings:
• Shift
• Phonebook
• Group pickup
• Call Forward
• DND
• Show phone
The user can customize the telephone with his/her personal needs by assigning individual phone numbers and functions. After a factory reset, the system will be reset to these values.
With the audio keys, the user can control the audio settings.
Inbound calls are visually signaled via the alert bar.
The keypad is used for entering phone numbers and text.
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Overview
The OpenStage Family
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Overview
The OpenStage Family
1.5.4
OpenStage 20
1
3
2
4
5 6
1
2
3
4
5
6
Tabelle 1-4
With the handset, the user can pick up and dial calls in the usual manner.
The display provides intuitive support for telephone operation.
The fixed function keys provide access to frequently used telephony functions.
With the audio keys, the user can control the audio settings.
With the 3-way navigator, the user/administrator can navigate in the various phone functions, applications, and configuration menus.
The keypad is used for entering phone numbers and text.
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Overview
The OpenStage Family
1.5.5
OpenStage 15
1
2
3 4
5
5
6
7
3
4
1
2
Tabelle 1-5
6
With the handset, the user can pick up and dial calls in the usual manner.
The display provides intuitive support for telephone operation.
With the audio keys, the user can control the audio settings.
The fixed function keys provide access to frequently used telephony functions.
The keypad is used for entering phone numbers and text.
With the navigation keys, the user/administrator can navigate in the various phone functions, applications, and configuration menus.
The free programmable keys enable the user to customize the telephone in line with his/her personal needs by assigning individual phone numbers and functions.
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Overview
Administration Interfaces
1.6
Administration Interfaces
You can configure the OpenStage phone by using any of the methods described in this chapter.
1.6.1
Web-based Management (WBM)
This method employs a web browser for communication with the phone via HTTPS. It is applicable for remote configuration of individual IP phones in your network. Direct access to the phone is not required.
> To use this method, the phone must first obtain IP connectivity.
1.6.2
DLS (Deployment Service)
The Deployment Service (DLS) is an OpenScape Management application for administering phones and soft clients in communication networks. It has a Java-supported, web-based user interface, which runs on an internet browser. For further information, please refer to the Deployment Service Administration Guide.
1.6.3
Local Phone Menu
This method provides direct configuration of the OpenStage phone. Direct access to the phone is required.
> As long as the IP connection is not properly configured, you have to use this method to set up the phone.
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Startup
Prerequisites
2 Startup
2.1
Prerequisites
The OpenStage phone acts as an endpoint client on an IP telephony network, and has the following network requirements:
• An Ethernet connection to a network with SIP clients and servers.
7
Only use switches in the LAN to which the OpenStage phone is connected. An operation at hubs can cause serious malfunctions in the hub and in the whole network.
• Phone Administration server.
• An FTP Server for file transfer, e. g. firmware, configuration data, application software.
• A DHCP (Dynamic Host Configuration Protocol) server (recommended).
• DLS (Deployment Service) for advanced configuration and software deployment (recommended).
For additional information see : http://wiki.unify.com/wiki/IEEE_802.1x.
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Startup
Assembling and Installing the Phone
2.2
Assembling and Installing the Phone
2.2.1
Shipment
• Phone
• Handset
• Handset cable
• Subpackage:
• Document "Information and Important Operating Procedures"
• Emergency number sticker
2.2.2
Connectors at the bottom side
OpenStage 60
PC
Power supply USB
Stick
Switch
Headset
Key Module
Handset
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Startup
Assembling and Installing the Phone
OpenStage 40 (OpenStage 15 and 20 similar, except 1 )
[1] OpenStage 40 andd
Power supply PC
Switch
Headset
[1]
Key Module
[1]
Handset
2.2.3
Assembly
1. Handset
Insert the plug on the long end of the handset cable into the jack on the base of the telephone and press the cable into the groove provided for it. Next, insert the plug on the short end of the handset cable into the jack on the handset.
2. Emergency Number Sticker
Write your telephone number and those for the fire and police departments on the included label and attach it to the telephone housing underneath the handset (see arrow).
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Startup
Assembling and Installing the Phone
2.2.4
Connecting the Phone
1. Plug the LAN cable into the connector at the bottom of the telephone and connect the cable to the LAN resp. switch. If PoE (Power over Ethernet) is to be used, the PSE (Power
Sourcing Equipment) must meet the IEEE 802.3af specification.
For details about the required power supply, see the following table:
Model
OpenStage 15
1
OpenStage 15 G 1
OpenStage 20 E
OpenStage 20
OpenStage 20 G
OpenStage 40 2 , OpenStage 40 US 2
OpenStage 40 + 2nd Key Module
OpenStage 40 G 2
,OpenStage 40 G US 2
OpenStage 40 G or OpenStage 40 G US + 2nd Key
Module
OpenStage 60/80 3
OpenStage 60/80 + 2nd Key Module
OpenStage 60/80 G 3
OpenStage 60/80 G + 2nd Key Module
1 Includes 1 Key Module 15.
2 Includes 1 Key Module.
3 Includes 1 Key Module + USB-Extension with Acoustic Unit.
Power Consumption/Supply
Power Class 1
Power Class 2
Power Class 1
Power Class 1
Power Class 2
Power Class 2
Power Class 2
Power Class 3
Power Class 3
Power Class 3
Power Class 3
Power Class 3
External power unit required
2. Only if Power over Ethernet (PoE) is NOT supported:
7
The order no. for the plug-in power supply is region specific:
EU: C39280-Z4-C510
UK: C39280-Z4-C512
USA: C39280-Z4-C511
Plug the power supply unit into the mains. Connect the plug-in power supply unit to the jack at the bottom of the phone.
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Startup
Assembling and Installing the Phone
3. If applicable, connect the following optional jacks:
• LAN connection to PC
•
• Headset (accessory)
Connection to add-on device (accessory)
•
• Connection to external keyboard (accessory)
USB master for connection to a USB device (e. g. accessory USB Acoustic
Adapter)
7
To prevent damage on the OpenStage phone, connect an USB stick using the adapter cable C39195-Z7704-A5.
7
Do not connect a USB hub to the phone’s USB port, as this may lead to stability problems.
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Startup
Quick Start
2.3
Quick Start
This section describes a typical case: the setup of an OpenStage endpoint in an environment using a DHCP server and the web interface. For different scenarios, cross-references to the corresponding section of the administration chapter are given.
> Alternatively, the DLS (Deployment Service) administration tool can be used. Its
Plug & Play functionality allows to provide the phone with configuration data by assigning an existing data profile to the phone’s MAC address or E.164 number. For further information, see the Deployment Service Administration Manual.
> Any settings made by a DHCP server are not configurable by other configuration tools.
2.3.1
Accessing the Web Interface (WBM)
1. Open your web browser (MS Internet Explorer or Firefox) and enter the appropriate URL.
Example: https://192.168.1.15 or https://myphone.phones
For configuring the phone’s DNS name, please refer to Section 3.3.7.1, “DNS Domain
Name”.
If the browser displays a certificate notification, accept it. The start page of the web interface appears. In the upper right corner, the phone number, the phone’s IP address, as well as the DNS name assigned to the phone are displayed. The left corner contains the user menu tree.
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Startup
Quick Start
2. Click on the tab "Administrator Pages". In the dialog box, enter the admin password:
Admin Login
Enter Admin password:
Login Reset
3. The administration main page opens. The left column contains the menu tree. If you click on an item which is printed in normal style, the corresponding dialog opens in the center of the page. If you click on an item printed in bold letters, a sub-menu opens in the right column.
2.3.2
Set the Terminal Number
If the user and administrator menus are needed in the course of setup, the terminal number, which by default is identical with the phone number, must be configured first. When the phone is in delivery status, the terminal number input form is presented to the user/administrator right after booting, unless the Plug&Play facility of the DLS is used. For further information about this
setting, please refer to Section 3.5.1.1, “Terminal Identity”. With the WBM, the terminal number
is configured as follows:
In the left column, select System > System Identity to open the "System Identity" dialog. Enter the terminal number, i. e. the SIP name / phone number.
System Identity
Terminal number
Terminal name
Display identity
Enable ID
Web name
DNS name construction
Submit
4711 openstage
4711
;
Only number
Reset
2.3.3
Basic Network Configuration
For basic functionality, DHCP must provide the following parameters:
• IP Address: IP Address for the phone.
• Subnet Mask (option #1): Subnet mask of the phone.
• Default Route (option #3 "Router"): IP Address of the default gateway which is used for connections beyond the subnet.
• DNS IP Addresses (option #6 "Domain Server"): IP Addresses of the primary and secondary DNS servers.
If no DHCP server is present, see Section 3.3.4, “IP Address - Manual Configuration” for IP ad-
dress and subnet mask, and Section 3.3.5, “Default Route/Gateway” for the default route.
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Startup
Quick Start
2.3.4
DHCP Resilience
It is possible to sustain network connectivity in case of DHCP server failure. If DHCP lease re-
use is activated, the phone will keep its DHCP-based IP address even if the lease expires. To prevent address conflicts, the phone will send ARP requests in 5 second intervals. Additionally, it will send discovery messages periodically to obtain a new DHCP lease.
In the left column, select Network > IPv4 configuration to open the "System Identity" dialog. Select the check box to enable DHCP lease reuse.
IPv4 configuration
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
IP address
Subnet mask
Default route
Route 1 IP address
Route 1 gateway
192.168.1.235
255.255.255.0
192.168.1.2
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit Reset
2.3.5
Date and Time / SNTP
An SNTP (Simple Network Time Protocol) server provides the current date and time for network clients. The IP address of an SNTP server can be given by DHCP.
In order to provide the correct time, it is required to give the timezone offset, i.e. the shift in hours to be added to the UTC time provided by the SNTP server.
The following DHCP options are required:
• SNTP IP Address (option #42 "NTP Servers"): IP Address or hostname of the SNTP server to be used by the phone.
• Timezone offset (option #2 "Time Offset"): Offset in seconds in relationship to the UTC time provided by the SNTP server.
For manual configuration of date and time see Section 3.5.5, “Date and Time”.
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2.3.6
SIP Server Address
The IP Address or hostname of the SIP server can be provided by DHCP.
The option’s name and code are as follows:
• option #120 "SIP Servers DHCP Option"
For manual configuration of the SIP server address see Section 3.5.6.1, “SIP Addresses”.
2.3.7
Extended Network Configuration
To have constant access to other subnets, you can enter a total of two more network destinations. For each further domain/subnet you wish to use, first the IP address for the destination, and then that of the router must be given. The option’s name and code are as follows:
• option #33 "Static Routing Table"
For manual configuration of specific/static routing see Section 3.3.6, “Specific IP Routing”.
Also the DNS domain wherein the phone is located can be specified by DHCP. The option’s name and code are as follows:
• option #15 "Domain Name"
For manual configuration of the DNS domain name see Section 3.3.7.1, “DNS Domain Name”.
2.3.8
Vendor Specific: VLAN Discovery And DLS Address
>
The VLAN ID can also be configured by LLDP-MED (see Section 3.2.2.2, “Automatic
VLAN discovery using DHCP”).
If the phone is to be located in a VLAN (Virtual LAN), a VLAN ID must be assigned. In case the
VLAN shall be provided by DHCP, VLAN Discovery must be set to "DHCP" (see Section
3.2.2.2, “Automatic VLAN discovery using DHCP”).
If a DLS (Deployment Service) server is in use, its IP address must be provided. It is recommended to configure the DLS server address by DCHP, as this method enables full Plug &
Play: having received the DLS address from DHCP, the phone will contact the DLS during startup. Provided that the DLS is configured appropriately, it will send all necessary configuration data to the phone. Additionally, this method is relevant to security, as it ensures the authenticity of the DLS server.
For manual configuration of the DLS server address see Section 3.3.8, “Configuration & Update Service (DLS)”.
For the configuration of vendor-specific settings by DHCP, there are two alternative methods:
1) the use of a vendor class, or 2) the use of DHCP option 43.
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2.3.8.1
Using a Vendor Class
It is recommended to define a vendor class on the DHCP server, thus enabling server and phone to exchange vendor-specific data exclusively. The data is disclosed from other clients.
In the following, the configuration of vendor classes is explained both for a Windows DHCP
Server and for Unix/Linux.
Configuration of the Windows DHCP Server
> For DHCP servers on a pre-SP2 Windows 2003 Server:
Windows 2003 Server contains a bug that prevents you from using the DHCP console to create an option with the ID 1 for a user-defined vendor class. Instead, this entry must be created with the netsh tool in the command line (DOS shell).
You can use the following command to set the required option (without error message), so that it will appear in the DHCP console afterwards: netsh dhcp server add optiondef 1 "Optipoint element 001"
STRING 0 vendor=OptiIpPhone comment="Tag 001 for Optipoint"
The value "Siemens" for optiPoint Element 1 can then be re-assigned using the
DHCP console.
This error was corrected in Windows 2003 Server SP2.
1. In the Windows Start menu, select Start > Programs > Administrative Tools > DHCP.
2. In the DHCP console menu, right-click the DHCP server in question and select Define
Vendor Classes... in the context menu.
3. A dialog window opens with a list of the classes that are already available.
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4. Press Add... to define a new vendor class.
5. Enter "OptiIpPhone" as Display name and give a description of this class. Provide the class name proper by setting the cursor underneath ASCII and typing "OptiIpPhone". The binary value is displayed simultaneously.
Click OK to apply the changes. The new vendor class now appears in the list:
6. Exit the window with Close.
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7. In the DHCP console menu, right-click the DHCP server in question and select Set Pre-
defined Options from the context menu.
8. In the dialog, select the previously defined OptiIpPhone class and click on Add... to add a new option. (If the workaround for a pre-SP2 Windows 2003 Server has been applied, the first option will be there already.)
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9. In the following dialog, specify the option type as follows. (If the workaround for a pre-SP2
Windows 2003 Server has been applied, the option type dialog will be skipped for the first option.)
• Name: Free text, e. g. "OptiIpPhone element 01".
• Data type: "String".
• Code: "1".
• Description: Free text, e. g. "tag 1 for OptiIpPhone class".
Click OK to return to the previous window.
10. The newly created option is displayed now. Enter "Siemens" in the Value field.
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11. If the VLAN is to be provided by DHCP: Repeat step 7 and 8, and then specify the option type as follows. If you want to proceed to the configuration of the DLS address, continue with step 13.
• Name: Free text, e. g. "OptiIpPhone element 02"
• Data type: "Long"
• Code: "2"
• Description: Free text, e. g. "tag 2 for OptiIpPhone class".
Click OK to return to the previous window.
12. The newly created option is displayed now. Enter the VLAN ID as a hexadecimal number in the Value field. In the example, the VLAN ID is 10 (Hex: 2A).
If you do not intend to configure the DLS address, click OK and continue with step 15.
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13. If the DLS address is to be provided by DHCP: Repeat step 7 and 8, and then specify the option type as follows.
• Name: Free text, e. g. "OptiIpPhone element 03".
• Data type: "String".
• Code: "3".
• Description: Free text, e. g. "tag 3 for OptiIpPhone class".
Click OK to return to the previous window.
14. The newly created option is displayed now. Enter the DLS address in the Value field, using the following format:
<PROTOCOL>:://<IP ADDRESS OF DLS SERVER>:<PORT NUMBER>
In the example, the DLS address is "sdlp://192.168.3.30:18443".
Click OK.
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15. To define a scope, select the DHCP server in question, and then Scope, and right-click
Scope Options. Select Configure Options... in the context menu.
16. Select the Advanced tab. Under Vendor class, select the class that you previously defined (OptiIpPhone) and, under User class, select Default User Class.
Activate the check boxes for the options that you want to assign to the scope (in the example, 001, 002, and 003). Click OK.
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17. The DHCP console now shows the information that will be transmitted to the corresponding workpoints. Information from the Standard vendor is transmitted to all clients, whereas information from the OptiIpPhone vendor is transmitted only to the clients (workpoints) in this vendor class.
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Setup using a DHCP server on Unix/Linux
The following snippet from a DHCP configuration file (usually dhcpd.conf) shows how to set up a configuration using a vendor class and the "vendor-encapsulated-options" option.
class "OptiIpPhone" { option vendor-encapsulated-options
# The vendor encapsulated options consist of hexadecimal values for the option number (for instance, 01), the length of the value (for instance, 07), and the value (for instance, 53:69:65:6D:65:6E:73). The options can be written in separate lines; the last option must be followed by a ’;’ instead of a ’:’.
# Tag/Option #1: Vendor must be "Siemens"
#1 7 S i e m e n s
01:07:53:69:65:6D:65:6E:73:
# Tag/Option #2: VLAN ID
# 2 4 0 0 0 10
02:04:00:00:00:0A;
# Tag/Option #3: DLS IP Address (here: sdlp://192.168.3.30:18443)
# 3 25 s d l p : / / 1 9 2 .
1 6 8 .
3 . (...etc.)
03:19:73:64:6C:70:3A:2F:2F:31:39:32:2E:31:36:38:2E:33:2E:33:30:
3A:31:38:34:34:33; match if substring (option vendor-class-identifier, 0, 11) =
"OptiIpPhone";
}
2.3.8.2
Using Option #43 "Vendor Specific"
Alternatively, option #43 can be used for setting up the VLAN ID and DLS address. The following tags are used:
• Tag 1: Vendor name
• Tag 2: VLAN ID
• Tag 3: DLS address
Optionally, the DLS address can be given in an alternative way:
• Tag 4: DLS hostname
The Vendor name tag is coded as follows (the first line indicates the ASCII values, the second line contains the hexadecimal values):
Code
1
01
Length Vendor name
7 S i e m e n s
07 53 69 65 6D 65 6E 73
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The following example shows a VLAN ID with the decimal value "10". Providing:
Code
2
02
Length
4
04
VLAN ID
0
00
0
00
1
00
0
0A
For manual configuration of the VLAN ID see Section 3.2.2.3, “Manual configuration of a VLAN
ID”.
The DLS IP address tag consists of the protocol prefix "sdlp://", the IP address of the DLS server, and the DLS port number, which is "18443" by default. The following example illustrates the syntax:
Code Length DLS IP address
3
03
25
19 s d l p : / / 1 9 2 . 1 6 8 . 3 . 3 0 : 1 8 4 4 3
Setup using the Windows DHCP Server
1. In the Windows Start menu, select Start > Programs > Administrative Tools > DHCP.
2. Select the DHCP server and the scope. Choose Configure Options in the context menu using the right mouse button.
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3. Enter tag 1, that is the vendor tag. The value has to be "Siemens".
4. If the VLAN ID is to be provided by DHCP: Enter the hexadecimal value in Data entry. In the example, the VLAN ID is 10 (Hex: 0A).
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5. If the DLS address is to be provided by DHCP: Enter the DLS address in the Value field, using the following format:
<PROTOCOL>:://<IP ADDRESS OF DLS SERVER>:<PORT NUMBER>
> For ensuring proper functionality, the port number should not be followed by any character.
In the example, the DLS address is "sdlp://192.168.3.30:18443".
Note that the screenshot also shows the VLAN ID described in step 4.
Click OK.
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6. The DHCP console now shows the information that will be transmitted to the corresponding workpoints.
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2.3.9
Registering at Phone Administration
For registration at the OpenStage SIP V3R3 SIP server, a SIP user ID and password must be provided by the phone. The following procedure describes the configuration using the web in-
terface (see Section 2.3.1, “Accessing the Web Interface (WBM)”; if the web interface is not
applicable, please refer to Section 3.5.7, “Authenticated Registration”) for configuration via the
local menu.
1. In the administration menu, select System > Registration. The Registration dialog opens.
Registration
SIP Addresses
SIP server address
SIP registar Address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
HiQ8000
3600
UDP
;
Reset
2. Make sure that SIP server address and SIP registrar address contain the IP address of your OpenScape Voice server. If not provided by DHCP or DLS, enter the appropriate values. If the phone is to register with a gateway, enter the appropriate SIP Gateway ad-
dress.
3. In the Server type field, select "OS Voice".
4. In Realm, enter the SIP realm the targeted user/password combination refers to.
5. In the User ID and Password fields, enter the user name/password combination for the phone.
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Startup Procedure
2.4
Startup Procedure
The following flowchart shows the startup process for OpenStage phones:
Power Up
L2 interface configured
Yes No
802.1x
credential
Start 802.1x
Suppplicant in Proxy-EAPOL-
Logoff mode
Start 802.1x
Suppplicant in auth. mode
44
802.1x enabled on authenticator
Yes
No
Manual configured
VLAN-ID
No
LLDP-Med assigns
VLAN-ID
No
GuestVLAN or
AuthfailVLAN No
Yes
Yes
No networkaccess
No networkaccess
No
802.1x enabled on authenticator
802.1x
auth.
Yes
Yes
No
Yes Manual configured
VLAN-ID
No
LLDP-Med assigns
VLAN-ID
No
GuestVLAN or No untagged VLAN
Yes
Yes
DHCP assigns
VLAN-ID
No
Manual configured
IP addresses
No
DHCP assigned
IP addresses
Yes
Contact DLS and get conf. data
L2 conf. data changed
Yes
Yes No networkaccess
Description:
In case OpenStage is in state
„Proxy-EAPOL-Logoff“, it does not make sense to posess a VLAN-ID, because the OpenStage phone is not authorized for any authenticated VLAN.
No L3 not configured
No
Conf. finished
DHCP assigns
VLAN-ID
No
GuestVLAN
VoiceVLAN
Yes
OpenStage boot sequence using VoiceVLAN and no DHCP reuse
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Cloud Deployment
2.5
Cloud Deployment
This chapter describes how a phone progresses through the cloud deployment process from factory start-up until the cloud service provider considers it to be ready for use by its User.
The phone determines that a cloud deployment process is to be used based on the IP settings it receives from the DHCP at the customer site. The Unify Redirect server 1 requires a code to determine which cloud service provider is responsible for the phone. The code is provided as part of a pin supplied from the cloud organisation to the User. When the User enters the pin at the phone the Unify Redirect server redirects the phone to a DLS-WPI based management system operated by the cloud service provider, This management system completes the configuration of the phone with all the information required for it to be usable and may also customise the phone for the cloud service provider's 'house' style.
2.5.1
Process of Cloud Deployment
The following flow chart shows the way from a factory start-up until a user prepared OpenStage phone, deployed by a relevant DLS-WPI based management system.
Preconditions:
• Phone is not running
• Phone is set to factory default values
• The phone has a LAN connection
• The LAN connection provides access to the public internet
Start
Phone broadcasts a DHCP request
A DHCP server responds with IP addresses
The phone has all the information that it needs to contact a DNS server. A DLS address is not provided.
The phone detects that a cloud deployment is required
DHCP is available; IP address allocated to the phone; DNS address is available; Subnet mask is available;
Router address is available; no DLS address available; no SIP addresses available
1. The address for the Unify Redirect server is hardcoded as "'cloud-setup.com"
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46
The phone stars the cloud deployment process
The phone is locked so that the mode keys & FPKs etc cannot be used
The phone obtains the IP address of the Unify Redirect server ("cloud-setup.com") from the DNS
The phone gets the pin from the user
The User enters the pin and presses OK. The phone verifies that the pin is valid.
The User has the option to cancel
the process (see Kapitel 2, “Aborting cloud deployment process by
User”
Phone displays the Progress prompt
Phone contacts the Unify
Redirect server using DLS-
Phone receives configuration items from the Unify Redirect server
•
•
• DLS address (set to the name of the Deployment server)
Language optional)
DLS port (optional)
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Cloud Deployment
If the language is changed then the display is updated
Phone updates, and displays, the Progress prompt
The phone obtains the IP address of the Deployment server from the DNS by looking up the DLS address if appropriate.
The stored DLS address is not changed by the result of the DNS lookup
Phone contacts the Deployment server using DLS-WPI
The hash of the pin is provided as an Inventory item to the Deployment server
Deployment server configures the phone and the phone saves the changes
Deployment server terminates the DLS-WPI session
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Cloud Deployment
The phone exits the cloud deployment process and enables the mode keys & FPKs etc. to act as normal
Phone removes the
Progress prompt and displays a timed success popup, indicating that cloud deployment is done
The phone verifies that it now has an e.164 address registered
2.5.2
Aborting cloud deployment process by User
The phone detects that a cloud deployment is required and starts the cloud deployment process. The Phone expects the input of the PIN by the User. At this point the User has the option to cancel the process with Cancel. If the User confirms his decision, the deployment process is aborted.
2.5.3
Re-trigger cloud deployment
Cloud deployment may be restarted by triggering a Factory reset:
The DLS-WPI requests a restart to factory defaults of the phone. The phone restart should then trigger the cloud deployment process.
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2.5.4
Deployment errors
During deployment the display will always show deployment specific information. A persistent warning popup displays the information that will be shown in an idle screen error after deployment failed.
• It is shown to notify the phone User that deployment failed to complete as expected.
• It is a non-timed warning popup
• It is non-dismissible by user action
• It is shown over the idle screen only
• It is shown/re-shown whenever the idle screen is displayed or redisplayed to the user
• It is formatted as the warning icon followed by a warning text which ends in a code displayed in round brackets.
• The warning text is = "Deployment incomplete"
• It displays only the highest priority error condition should more than one error condition apply (note that priority 1 is the highest)
Code
AU
Priority
1
Cause
Abandoned by user
Occurs when the pin prompt is dismissed
RS 1
RN
RR
RU
RO
3
2
1
3
Unable to get the address for the Unify Redirect server
DNS lookup failed
Unable to establish contact with Unify Redirect server – no reply
Unable to establish contact with Unify Redirect server – refused
RV
DS
DN
2
1
3
Unable to establish contact with Unify Redirect server - unauthorised
Unable to establish contact with Unify Redirect server - no or invalid
OCSP response
Unable to establish contact with Unify Redirect server - certificate revoked
Unable to get the address for the Deployment server
DNS lookup failed
Unable to establish contact with Deployment server – no reply
DR 2 Unable to establish contact with Deployment server – refused
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Administration
Access via Local Phone
3 Administration
This chapter describes the configuration of every parameter available on the OpenStage phones. For access via the local phone menu, see the following; for access using the web in-
terface, please refer to Section 2.3.1, “Accessing the Web Interface (WBM)”.
3.1
Access via Local Phone
> The data entered in input fields is parsed and controlled by the phone. Thus, data is accepted only if it complies to the value range.
1. Access the Admin Menu
OpenStage 60/80:
The menu key v toggles between the Settings menu, the Applications menu, and the applications currently running. Press the v key repeatedly until the "Settings" tab is active.
OpenStage 40:
Press the keys D , l , and i consecutively to select Settings > Admin (the administration menu).
OpenStage 40 US:
Press the keys "Services", l , and i consecutively to select Settings > Admin (the administration menu).
2. Enter Password
When the Admin menu is active, you will be prompted to enter the administrator password.
The default admin password is "123456". It is highly recommended to change the pass-
word (see Section 3.20, “Password”) after your first login.
For entering passwords with non-numeric characters, please consider the following:
By default, password entry is in numeric mode. For changing the mode, press the # key once or repeatedly, depending on the desired character. The # key cycles around the input modes as follows:
(Abc) -> (abc) -> (123) -> (HEX) -> (ABC) -> back to start.
Usable characters are 0-9 A-Z a-z .*#,?!’+-()@/:_
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3. Navigate within the Admin Menu
OpenStage 60/80
Use the TouchGuide to navigate and execute administrative actions in the Admin menu.
Press the m
key briefly:
- scroll up
Hold down:
- scroll to top of list
Press the h
key:
- cancel a function
- delete character left of cursor
- up one level
Press the l
key briefly:
- scroll down
Hold down:
- scroll to end of list
Press the i
key:
- confirm entries
- perform an action
Press the g
key:
- open a context menu
- down one level
Run your finger around the inner sensor ring
W:
- browse lists and menus
- set up volume
OpenStage 40
Use the 5-way navigator to navigate and execute administrative actions in the administration menu.
Press the m key briefly:
- scroll up
Hold down:
- scroll to top of list
Press the h
key:
- cancel a function
- delete character left of cursor
- up one level
Press the l
key briefly:
- scroll down
Hold down:
- scroll to end of list
Press the i
key:
- confirm entries
- perform an action
Press the g
key:
- open a context menu
- down one level
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OpenStage 20
Use the 3-way navigator to navigate and execute administrative actions in the administration menu.
Press the m
key briefly:
- scroll up
Hold down:
- scroll to top of list
Press the l
key:
- confirm entries
- perform an action
Press the l
key briefly:
- scroll down
Hold down:
- scroll to end of list
OpenStage 15
Use the navigation keys to navigate and execute administrative actions in the administration menu.
Press the m
key briefly:
- scroll up
Press the i
key:
- confirm entries
- perform an action
Press the l
key:
In idle mode:
- Open idle menu
In lists and menus:
- scroll down
4. Select a parameter
If a parameter is set by choosing a value from a selective list, an arrow symbol appears in the parameter field that has the focus (on OpenStage 80,60, and 40 displays only). Press the i key to enter the selective list. Use the Sensor Wheel resp. the m and l key to scroll up and down in the selective list. To select a list entry, press the i key.
5. Enter the parameter value
For selecting numbers and characters, you can use special keys. See the following table:
Key
*
#
Function
Switch to punctuation and special characters.
Toggle between lowercase characters, uppercase characters, and digits in the following order:
(Abc) -> (abc) -> (123) -> (ABC) -> back to start.
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OpenStage 60/80
If a parameter is set by entering a number or character data, the onscreen keypad is used.
Press the i key to enter the editor. Within the editor, solely use the key numbers or the
Sensor Wheel for selecting numbers, characters, or groups of characters. The h key deletes one character in the input field, and the g key moves the cursor to the OK field.
The following figure describes the elements of the onscreen keypad and their functions:
Element with focus
Letters, digits, punctuation marks or special characters
Confirm
Cancel
Insert clipboard contents at cursor position
Copy contents of active field to clipboard
Move cursor left/right
Shift to punctuation and special characters
Shift to numeric entry Shift to punctuation and special characters Shift to numeric entry
Shift to upper/lower case Shift to upper/lower case
Additionally, you can use the following keys on the keypad as shortcuts for the selection of character groups
Element
*
#
Function
Switch to punctuation and special characters.
Toggle between lowercase characters, uppercase characters, and digits.
OpenStage 15/20/40
With the OpenStage 15/20/40, use the keypad for entering parameters. With the 3 way/5 way navigator, you can enter, delete, copy, and paste characters and numbers as well as navigate within an entry and toggle the input mode.
6. Save and exit
When you are done, select Save & exit and press .
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3.2
LAN Settings
3.2.1
LAN Port Settings
The OpenStage phone provides an integrated switch which connects the LAN, the phone itself and a PC port. By default, the switch will auto negotiate transfer rate (10/100 Mb/s, 1000 Mb/s with OpenStage 15/20/40/60/80 G) and duplex method (full or half duplex) with whatever equipment is connected. Optionally, the required transfer rate and duplex mode can be specified manually using the LAN port speed parameter.
> In the default configuration, the LAN port supports automatic detection of cable configuration (pass through or crossover cable) and will reconfigure itself as needed to connect to the network.
The PC Ethernet port is controlled by the PC port mode parameter. If set to "Disabled", the PC port is inactive; if set to "Enabled", it is active. If set to "Mirror", the data traffic at the LAN port is mirrored at the PC port. This setting is for diagnostic purposes. If, for instance, a PC running
Ethereal/Wireshark is connected to the PC port, all network activities at the phone’s LAN port can be captured.
> Removing the power from the phone, or a phone reset/reboot will result in the temporary loss of the network connection to the PC port.
When PC port autoMDIX is enabled, the switch determines automatically whether a regular
MDI connector or a MDI-X (crossover) connector is needed, and configures the connector accordingly.
Data required
• LAN port speed / LAN port type: Settings for the ethernet port connected to a LAN switch.
Value range: "Automatic," "10 Mbps half duplex", "10 Mbps full duplex", "100 Mbps half duplex", "100 Mbps full duplex".
Default: "Automatic"
• PC port speed / PC port type: Settings for the ethernet port connected to a PC.
Value range: "Automatic," "10 Mbps half duplex", "10 Mbps full duplex", "100 Mbps half duplex", "100 Mbps full duplex".
Default: "Automatic"
• PC port mode / PC port status: Controls the PC port.
Value range: "disabled", "enabled", "mirror".
Default: "disabled"
• PC port autoMDIX: Switches between MDI and MDI-X automatically.
Value range: "On", "Off"
Default: "Off"
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Administration via WBM
Network > Port configuration
SIP Server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port mode
PC port autoMDIX
Submit
Port configuration
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
Port Configuration
|---
LAN port type
|---
PC port status
|
|---
PC port type
---
PC port autoMDIX administration.fm
Administration
LAN Settings
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3.2.2
VLAN
VLAN (Virtual Local Area Network) is a technology that allows network administrators to partition one physical network into a set of virtual networks (or broadcast domains).
Partitioning a physical network into separate VLANs allows a network administrator to build a more robust network infrastructure. A good example is a separation of the data and voice networks into data and voice VLANs. This isolates the two networks and helps shield the endpoints within the voice network from disturbances in the data network and vice versa.
> The implementation of a voice network based on VLANs requires the network infrastructure (the switch fabric) to support VLANs.
In a layer 1 VLAN, the ports of a VLAN-aware switch are assigned to a VLAN statically. The switch only forwards traffic to a particular port if that port is a member of the VLAN that the traffic is allocated to. Any device connected to a VLAN-assigned port is automatically a member of this VLAN, without being a VLAN aware device itself. If two or more network clients are connected to one port, they cannot be assigned to different VLANs. When a network client is moving from one switch to another, the switches’ ports have to be updated accordingly by hand.
With a layer 2 VLAN, the assignment of VLANs to network clients is realized by the MAC addresses of the network devices. In some environments, the mapping of VLANs and MAC addresses can be stored and managed by a central database. Alternatively, the VLAN ID, which defines the VLAN whereof the device is a member, can be assigned directly to the device, e. g. by DHCP. The task of determining the VLAN for which an Ethernet packet is destined is carried out by VLAN tags within each Ethernet frame. As the MAC addresses are (more or less) wired to the devices, mobility does not require any administrator action, as opposed to layer 1 VLAN.
It is important that every switch connected to a PC is VLAN-capable. This is also true for the integrated switch of the OpenStage. The phone must be configured as a VLAN aware endpoint if the phone itself is a member of the voice VLAN, and the PC connected to the phone’s PC port is a member of the data VLAN.
There are 3 ways for configuring the VLAN ID:
• Manually
• By DHCP
• By LLDP-MED
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3.2.2.1
Automatic VLAN discovery using LLDP-MED
As an alternative, the VLAN ID can be configured by the network switch using LLDP-MED (Link
Layer Discovery Protocol-Media Endpoint Discovery). If this option is selected, and the switch provides an appropriate TLV (Type-Length-Value) element containing the VLAN ID, this VLAN
ID will be used. If no appropriate TLV is received, DHCP will be used for VLAN discovery.
Administration via WBM
Network > General IP configuration
To enable VLAN discovery via LLDP-MED, activate the LLDP-MED Enabled checkbox and select LLDP-MED in the VLAN discovery option. Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
;
;
LLDP-MED
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
To enable VLAN discovery via LLDP-MED, set the Use LLDP-MED option to Yes and select
LLDP-MED in the VLAN discovery option.
|
---
Admin
|
---
Network
|
---
General IP configuration
|
---
Protocol mode
|
---
Use LLDP-MED
|
---
Use DHCP
|
---
Use DHCPv6
|
---
VLAN discovery
|
---
VLAN ID
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3.2.2.2
Automatic VLAN discovery using DHCP
To automatically discover a VLAN ID using DHCP, the phone must be configured as DHCP enabled, and VLAN discovery mode must be set to "DHCP". LLDP-MED should be disabled.
The DHCP server must be configured to supply the Vendor Unique Option in the correct VLAN over DHCP format. If a phone configured for VLAN discovery by DHCP fails to discover its
VLAN, it will proceed to configure itself from the DHCP within the non-tagged LAN. Under these circumstances, network routing may probably not be correct.
Administration via WBM
Network > General IP configuration
To enable VLAN discovery via DHCP, activate the DHCPv6 Enabled checkbox and select
DHCP in the VLAN discovery option. Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
DHCP
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
To enable VLAN discovery via DHCP, activate the DHCPv6 Enabled checkbox and select
DHCP in the VLAN discovery option.
|
--Admin
|
--Network
|
--General IP configuration
|
--Protocol mode
|
--Use LLDP-MED
|
--Use DHCP
|
--Use DHCPv6
|
--VLAN discovery
|
--VLAN ID
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3.2.2.3
Manual configuration of a VLAN ID
To configure layer 2 VLAN manually, make sure that VLAN discovery is set to "Manual" and
LLDP-MED is disabled. Then, the phone must be provided with a VLAN ID between 1 and
4095. If you mis-configure a phone to an incorrect VLAN, the phone will possibly not connect to the network. In DHCP mode it will behave as though the DHCP server cannot be found, in fixed IP mode no server connections will be possible.
Administration via WBM
Network > General IP configuration
The phone must be provided with a VLAN ID between 1 and 4095. Set the VLAN discovery to
"Manual". Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
Manual
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
General IP Configuration
|
---
VLAN ID
To enable VLAN discovery by Manual, select Manual in the VLAN discovery option.
|
---
Admin
|
---
Network
|
---
General IP configuration
|
---
VLAN discovery
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3.2.3
LLDP-MED Operation
OpenStage phones support LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery) for auto-configuration and network management. The auto-configurable parameters
are VLAN ID (see Section 3.2.2, “VLAN”) and Quality of Service parameters (see Section 3.3.1,
“Quality of Service (QoS)”).
The data sent by a network device is stored in neighboring network devices in MIB (Management Information Base) format. In order to keep this information up-to-date, a specific TTL
(Time To Live) is specified in LLDP. This value tells a device how long the received information is valid. For OpenStage phones, the value range is 40, 60, 80, 100, 110, 120, 140, 180, 240,
320, 400.
An example for LLDP-MED operation an OpenStage phones can be found in Section 5.4, “An
LLDP-Med Example”.
Administration via WBM
Network > LLDP-MED operation
LLDP-MED operation
Time to live (seconds) 120
Submit Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--LLDP-MED operation
|
--Extended power
|
--Network policy (voice)
|
--LLDP-MED cap’s
|
--MAC-Phy config
|
--System Cap’s
|
--TTL
|
--TTL display only display only display only display only display only
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3.3
IP Network Parameters
3.3.1
Quality of Service (QoS)
The QoS technology based on layer 2 and the two QoS technologies Diffserv and TOS/IP Precedence based on layer 3 are allowing the VoIP application to request and receive predictable service levels in terms of data throughput capacity (bandwidth), latency variations (jitter), and delay.
>
Layer 2 and 3 QoS for voice transmission can be set via LLDP-MED (see Section
3.26.6, “LLDP-MED”). If so, the value can not be changed by any other interface.
3.3.1.1
Layer 2 / 802.1p
QoS on layer 2 is using 3 Bits in the 802.1q/p 4-Byte VLAN tag which has to be added in the
Ethernet header.
The CoS (class of service) value can be set from 0 to 7. 7 is describing the highest priority and is reserved for network management. 5 is used for voice (RTP-streams) by default. 3 is used for signaling by default.
Three Bits Used for CoS
(User Priority)
PREAM.
SFD DA SA
TAG
4 Bytes
PT DATA FCS
Data required
• Layer 2: Activates or deactivates QoS on layer 2.
Value range: "Yes", "No"
Default: "Yes"
• Layer 2 voice: Sets the CoS (Class of Service) value for voice data (RTP streams).
Value range: 0-7
Default: 5
• Layer 2 signalling: Sets the CoS (Class of Service) value for signaling.
Value range: 0-7
Default: 3
• Layer 2 video: Sets the CoS (Class of Service) value for video.
Value range: 0-7
Default: 4
• Layer 2 default: Sets the default CoS (Class of Service) value.
Value range: 0-7
Default: 0
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Administration via WBM
Network > QoS
QoS
Service
Layer 2
Layer 2 voice 5
Layer 2 signalling 3
Layer 2 video 4
Layer 2 default 0
;
Layer 3
Layer 3 voice
Layer 3 signalling
EF
AF31
Layer 3 video AF41
MLPP
Priority EF
Immediate EF
Flash EF
Flash override EF
Submit Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--QoS
|
--Service
|
|--Layer 2
|--Layer 2 voice
|--Layer 2 signalling
|
--Layer 2 video
--Layer 2 default
3.3.1.2
Layer 3 / Diffserv
Diffserv assigns a class of service to an IP packet by adding an entry in the IP header.
Traffic flows are classified into 3 per-hop behavior groups:
1. Default
Any traffic that does not meet the requirements of any of the other defined classes is placed in the default per-hop behaviour group. Typically, the forwarding has best-effort forwarding characteristics. The DSCP (Diffserv Codepoint) value for Default is "0 0 0 0 0 0".
2. Expedited Forwarding (EF referred to RFC 3246)
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Expedited Forwarding is used for voice (RTP streams) by default. It effectively creates a special low-latency path in the network. The DSCP (Diffserv Codepoint) value for EF is
"1 0 1 1 1 0".
3. Assured Forwarding (AF referred to RFC 2597)
Assured forwarding is used for signaling messages by default (AF31). It is less stringent than EF in a multiple dropping system. The AF values are containing two digits X and Y
(AFXY), where X is describing the priority class and Y the drop level.
Four classes X are reserved for AFXY: AF1Y (low priority), AF2Y, AF3Y and AF4Y (high priority).
Three drop levels Y are reserved for AFXY: AFX1 (low drop probability), AFX2 and AFX3
(High drop probability). In the case of low drop level, packets are buffered over an extended period in the case of high drop level, packets are promptly rejected if they cannot be forwarded.
Data required
• Layer 3: Activates or deactivates QoS on layer 3.
Value range: "Yes", "No"
Default: "Yes"
• Layer 3 voice: Sets the CoS (Class of Service) value for voice data (RTP streams).
Value range: "BE", "AF11", "AF12", "AF13", "AF21", "AF22", "AF23", "AF31", "AF32",
"AF33", "AF41", "AF42", "AF43", "EF", "CS7", "CS3", "CS4", "CS5", 0, 1, 2 ... through 63.
Default: "EF"
• Layer 3 signalling: Sets the CoS (Class of Service) value for signaling.
Value range: "BE", "AF11", "AF12", "AF13", "AF21", "AF22", "AF23", "AF31", "AF32",
"AF33", "AF41", "AF42", "AF43", "EF", "CS7", "CS3", "CS4", "CS5", 0, 1, 2 ... through 63.
Default: "AF31"
• Layer 3 video: Sets the CoS (Class of Service) value for video.
Value range: "BE", "AF11", "AF12", "AF13", "AF21", "AF22", "AF23", "AF31", "AF32",
"AF33", "AF41", "AF42", "AF43", "EF", "CS7", "CS3", "CS4", "CS5", 0, 1, 2 ... through 63.
Default: "AF41"
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Administration via WBM
Network > QoS
QoS
Service
Layer 2
Layer 2 voice 5
Layer 2 signalling 3
Layer 2 video 4
Layer 2 default 0
Layer 3
Layer 3 voice EF
Layer 3 signalling AF31
Layer 3 video AF41
MLPP
Priority EF
Immediate
Flash
EF
EF
Flash override EF
Submit
;
Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--QoS
|
--Service
|--Layer 3
|--Layer 3 voice
|
--Layer 3 signalling
|
--Layer 3 video
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3.3.2
Protocol Mode IPv4/IPv6
An IPv4 address consists of 4 number blocks, each between 0 and 255, separated by ".".
Example:
1.222.44.123
An IPv6 address consists of 8 hexadecimal number blocks, separated by ":".
Example:
2001:0db8:85a3:08d3:1319:8a2e:0370:7347 or, if not all blocks are used:
2000:1::3
Administration via WBM
Network > General IP configuration
Set the Protocol Mode to "IPv4" or "IPv6" or both (the default setting is IPv4_IPv6). Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
Manual
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
General IP Configuration
|
---
Protocol Mode
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3.3.3
Use DHCP
If this parameter is set to "Yes" (default), the phone will search for a DHCP server on startup and try to obtain IP data and further configuration parameters from that central server.
If no DHCP server is available in the IP network, please deactivate this option. In this case, the
IP address, subnet mask and default gateway/route must be defined manually.
> The phone is able to maintain its IP connection even in case of DHCP server failure.
For further information, please refer to Section 2.3.4, “DHCP Resilience”.
The following parameters can be obtained by DHCP:
Basic Configuration
• IP Address
• Subnet Mask
Optional Configuration
• Default Route (Routers option 3)
• IP Routing/Route 1 & 2 (Static Routes option 33, Classless static route option 121, Private/
Classless Static Route (Microsoft) option 249)
• SNTP IP Address (NTP Server option 42)
• Timezone offset (Time Server Offset option 2)
• Primary/Secondary DNS (DNS Server option 6)
• DNS Domain Name (DNS Domain option 15)
• SIP Addresses / SIP Server & Registrar (SIP Server option 120)
• VLAN ID, DLS address (Vendor specific Information option 43)
The following parameters can be obtained by DHCPv6:
Basic Configuration
• Global Address
• Global Address Prefix Length
Optional Configuration
• Primary/Secondary DNS (DNS recursive name server option 23)
• SNTP IP Address (Simple Network Time Protocol Server option 31)
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• SIP Addresses / SIP Server & Registrar (SIP Server Domain Name List option 21, SIP
Server IPv6 Address List option 22)
• VLAN ID, DLS address (Vendor specific Information option 17)
DHCPv6 options are preferred in Dual Stack Mode if a parameter is configured both via DHCP and via DHCPv6, for instance DNS or SNTP server addresses.
Administration via WBM
Network > General IP configuration
Set DHCP Enabled to selected. Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
DHCP
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--IPv4 configuration or/and
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Administration via WBM
Network > General IP configuration
Set DHCPv6 Enabled to selected (the default setting is Enabled). Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
DHCP
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--IPv6 configuration
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3.3.4
IP Address - Manual Configuration
3.3.4.1
Configuration
If not provided by DHCP dynamically, the phone’s IP address and subnet mask must be specified manually.
> IP addresses can be entered in the following formats:
– Decimal format. Example: 11.22.33.44 or 255.255.255.0 (no leading zeroes).
– Octal format. Example: 011.022.033.044 (leading zeroes must be used with every address block)
– Hexadecimal format. Example: 0x11.0x22.0x33.0x44 (prefix 0x must be used with every address block)
By default, IP configuration by DHCP and LLDP-MED is enabled. For manual IP configuration, please proceed as follows:
1. Navigate to Network > General IP configuration. Set DHCP Enabled, DHCPv6 Enabled and LLDP-MED to "not selected". Afterwards, click Submit.
2. Navigate to Network > General IP configuration> IPv4 configuration or IPv6 configu-
ration depending on settings in Section 3.3.2, “Protocol Mode IPv4/IPv6”. Enter the IP ad-
dress and the Subnet mask. If applicable, enter the Default route. Afterwards, click Sub-
mit.
IPv4 configuration
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
IP address
Subnet mask
Default route
Route 1 IP address
Route 1 gateway
192.168.1.235
255.255.255.0
192.168.1.2
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit Reset
IPv6 configuration
LLDP-MED Enabled
DHCPv6 Enabled
DHCPv6 lease reuse
Global Address
Global Address Prefix Len
Global Gateway
Link Local Address
Route 1 Dest.
Route 1 Prefix Len
Route 1 Gateway
Route 2 Dest.
Route 2 Prefix Len
Route 2 Gateway
Submit Reset
3. After the phone’s network service has restarted, the other IP parameters can be configured.
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General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
Manual
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
--Admin
|
--Network
General IP configuration
|--Use LLDP-MED
|
--Use DHCP
|
--Use DHCPv6
|
--Admin
|
--Network
IPv4 configuration
|--IP address
|
--Subnet mask
|
--Admin
|
--Network
IPv4 configuration
|--Global address
|
--Global Prefix Len
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3.3.5
Default Route/Gateway
If not provided by DHCP dynamically (see Section 3.3.3, “Use DHCP”), enter the IP address of
the router that links your IP network to other networks. If the value was assigned by DHCP, it can only be read.
Administration via WBM
Network > IPv4 configuration
Enter the IP address of the router that links your IP network to other networks. Afterwards, click
Submit.
IPv4 configuration
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
IP address
Subnet mask
Default route
Route 1 IP address
Route 1 gateway
;
192.168.1.235
255.255.255.0
192.168.1.2
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
IPv4 configuration
|
---
Route (Default)
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Administration via WBM
Network > IPv6 configuration
Enter the IP address of the Global Gateway that links your IP network to other networks. Afterwards, click Submit.
IPv6 configuration
LLDP-MED Enabled
DHCPv6 Enabled
DHCPv6 lease reuse
Global Address
Global Address Prefix Len
Global Gateway
Link Local Address
Route 1 Dest.
Route 1 Prefix Len
Route 1 Gateway
Route 2 Dest.
Route 2 Prefix Len
Route 2 Gateway
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
IPv6 configuration
|
---
Global Gateway
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3.3.6
Specific IP Routing
To have constant access to network subscribers of other domains, you can enter a total of two more network destinations, in addition to the default route/gateway. This is useful if the LAN has more than one router.
IPv4 Route Configuration
Data required
• Route 1/2 IP address: IP address of the selected route.
• Route 1/2 gateway: IP address of the gateway for the selected route.
• Route 1/2 mask: Network mask for the selected route.
Administration via WBM
Network > IPv4 configuration
Enter the IP address of the router that links your IP network to other networks. Click Submit.
IPv4 configuration
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
IP address
Subnet mask
Default route
Route 1 IP address
Route 1 gateway
;
192.168.1.235
255.255.255.0
192.168.1.2
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
IPv4 configuration
|---
Route 1 IP
|---
Route 1 gateway
|---
Route 1 mask
|---
Route 2 IP
|
|---
Route 2 gateway
---
Route 2 mask
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IPv6 Route Configuration
Data required
• Route 1/2 destination: IPv6 address of the selected route.
• Route 1/2 prefix len: Prefix length for the selected route.
• Route 1/2 gateway: IPv6 address of the gateway for the selected route.
Administration via WBM
Network > IPv6 configuration
Enter the IP address of the router that links your IP network to other networks. Afterwards, click
Submit.
IPv6 configuration
LLDP-MED Enabled
DHCPv6 Enabled
DHCPv6 lease reuse
Global Address
Global Address Prefix Len
Global Gateway
Link Local Address
Route 1 Dest.
Route 1 Prefix Len
Route 1 Gateway
Route 2 Dest.
Route 2 Prefix Len
Route 2 Gateway
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
IPv6 configuration
|---
Route 1 dest
|---
Route 1 prefix len
|---
Route 1 gateway
|---
Route 2 dest
|
|---
Route 2 prefix len
---
Route 2 gateway
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3.3.7
DNS
The main task of the domain name system (DNS) is to translate domain names to IP addresses. For some features and functions of the OpenStage phone, it is necessary to configure the
DNS domain the phone belongs to, as well as the name servers needed for DNS resolving.
3.3.7.1
DNS Domain Name
This is the name of the phone’s local domain.
Administration via WBM
Network > General IP configuration
Enter the DNS domain the phone belongs to. Afterwards, click Submit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
Manual
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
General IP configuration
|
---
DNS domain
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3.3.7.2
DNS Servers
If not provided by DHCP automatically, a primary and a secondary DNS server can be configured.
> Enhanced survivability using DNS SRV is available. To make use of it, a special con-
figuration is required. For details, please refer to Section 3.5.10, “Resilience and
Survivability”.
Data required
• Primary DNS: IP address of the primary DNS server.
• Secondary DNS: IP address of the secondary DNS server.
Administration via WBM
Network > General IP configuration
Enter the IP addresses of the primary and the secondary DNS server. Afterwards, click Sub-
mit.
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
Manual
;
192.168.1.105
192.168.1.2
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
General IP configuration
|
|---
Primary DNS
---
Secondary DNS
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3.3.7.3
Terminal Hostname
The phone’s hostname can be customized.
The corresponding DNS domain is configured in Network > IP configuration > DNS domain
(see Section 3.3.7.1, “DNS Domain Name”).
The current DNS name of the phone is displayed at the right-hand side of the banner of the admin and user web pages, under DNS name. To see configuration changes, the web page must be reloaded.
> It is recommended to inform the user about the DNS name of his/her phone. The complete WBM address can be found under User menu > Network information >
Web address.
The DNS name can be constructed from pre-defined parameters and free text. Its composition is defined by the DNS name construction parameter. The following options are available:
• "None":
• "MAC based": The DNS name is built from the prefix "OIP" followed by the phone’s MAC address.
• "Web name": The DNS name is set to the the string entered in Web name.
• "Only number": The DNS name is set to the Terminal number, that is, the phone’s call
number (see Section 3.5.1, “Terminal and User Identity”).
• "Prefix number": The DNS name is constructed from the the string entered in Web name, followed by the Terminal number.
Administration via WBM
System > System Identity
System Identity
Terminal number
Terminal name
Display identity
Enable ID
Web name
DNS name construction
Submit
4711 openstage
4711
;
Only number
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Identity
|
|---
Web name
---
DDNS hostname
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3.3.8
Configuration & Update Service (DLS)
The Deployment Service (DLS) is a HiPath Management application for administering workpoints in both HiPath and non-HiPath networks. Amongst the most important features are: security (e.g. PSS generation and distribution within an SRTP security domain), mobility for opti-
Point and OpenStage SIP phones, software deployment, plug&play support, as well as error and activity logging.
DLS address, i.e. the IP address or hostname of the DLS server, and Default mode port , i.e. the port on which the DLS server is listening, are required to enable proper communication between phone and DLS.
The Contact gap parameter is not used.
Set Revert to default security to disable mutual authentication and return to DEFAULT mode.
SECURE mode related settings are reset and certificates are removed.
The Mode determines whether the communication between the phone and the DLS is secure.
A secure connection is established by exchanging credentials between the DLS and the phone for mutual authentication. After this, the communication is encrypted, and a different port is used.
> It is possible to operate the DLS server behind a firewall or NAT (Network Address
Translation), which prevents the DLS from sending ContactMe messages directly to the phone. Only outbound connections from the phone are allowed. To overcome this restriction, a DLS Contact-Me proxy (DCMP) can be deployed. The phone periodically polls the DCMP (DLS Contact-Me Proxy), which is placed outside of the phone’s network, for pending contact requests from the DLS. If there are contact requests, the phone will send a request to the DLS in order to obtain the update, just as with a regular DLS connection.
The URI of the DCMP, as well as the polling interval, are configured by the DLS. For this purpose, it is necessary that the phone establishes a first contact to the DLS, e. g. by phone restart or local configuration change.
A Security PIN can be provided which is used for decrypting data provided by the DLS during bootstrap. For further information, please refer to the DLS documentation.
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Data required
• DLS address: IP address or hostname of the server on which the Deployment Service is running.
• DLS port: Port on which the DLS Deployment Service is listening.
Default: 18443
• Contact gap: The parameter is not used.
• Revert to default security: When set, security mode will be set to default. When using local phone administration, this will be set by selection option ’Default security’ after pressing Save&exit.
• Mode: Shows whether the communication between the phone and the DLS is secure.
Value range: "Default", "Secure", "Secure PIN"
This parameter is read-only.
• Security PIN : Used for enhanced security.
Administration via WBM
Network > Update Service (DLS)
Update Service (DLS)
DLS address
DLS port
Contact gap
Revert to default security
Mode
Security PIN
Submit
192.168.1.242
18443
300
Default
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
Update Service (DLS)
|---
DLS address
|---
DLS port
|---
Contact gap
|---
Mode
|
---
Security PIN
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3.3.9
SNMP
The Simple Network Management Protocol is used by network management systems for monitoring network-attached devices for conditions that warrant administrative attention. An SNMP manager surveys and, if needed, configures several SNMP elements, e.g. VoIP phones.
OpenStage phones support SNMPv1.
There are currently 4 trap categories that can be sent by the phones:
Standard SNMP traps
OpenStage phones support the following types of standard SNMP traps, as defined in RFC
1157:
• coldStart: sent if the phone does a full restart.
• warmStart: sent if only the phone software is restarted.
• linkUp: sent when IP connectivity is restored.
QoS Related traps
These traps are designed specifically for receipt and interpretation by the QDC collection system. The traps are common to SIP phones, HFA phones, Gateways, etc.
Traps for important high level SIP related problems
Currently, these traps are related to problems in registering with a SIP Server and to a failure in remotely logging off a mobile user. These traps are aimed at a non-expert user (e.g. a standard Network Management System) to highlight important telephony related problems.
Traps specific to OpenStage phones
Currently, the following traps are defined:
TraceEventFatal: sent if severe trace events occur; aimed at expert users.
TraceEventError: sent if severe trace events occur; aimed at expert users.
Data required
• Trap sending enabled: Enables or disables the sending of a TRAP message to the SNMP manager.
Value range: "Yes", "No"
Default: "No"
• Trap destination: IP address or hostname of the SNMP manager that receives traps.
• Trap destination port: Port on which the SNMP manager is receiving TRAP messages.
• Default: 162
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• Trap community: SNMP community string for the SNMP manager receiving TRAP messages.
Default: "snmp"
• Queries allowed: Allows or disallows queries by the SNMP manager.
• Query password: Password for the execution of a query by the SNMP manager.
• Diagnostic sending enabled: Enables or disables the sending of diagnostic data to the
SNMP manager.
Value range: "Yes", "No"
Default: "No"
• Diagnostic destination: IP address or hostname of the SNMP manager receiving diagnostic data.
• Diagnostic destination port: Port on which the SNMP manager is receiving diagnostic data.
• Diagnostic community: SNMP community string for the SNMP manager receiving diagnostic data.
• Diagnostic to generic destination / Diagnostic to generic device: Enables or disables the sending of diagnostic data to a generic destination.
Value range: "Yes", "No"
Default: "No"
• QoS traps to QCU: Enables or disables the sending of TRAP messages to the QCU server.
Value range: "Yes", "No"
Default: "No"
• QCU address: IP address or hostname of the QCU server.
• QCU port: Port on which the QCU server is listening for messages.
Default: 12010.
• QCU community: QCU community string.
Default: "QOSCD".
• QoS to generic destination: Enables or disables the sending of QoS traps to a generic destination.
Value range: "Yes", "No"
Default: "No"
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Administration via WBM
System > SNMP
SNMP
Generic traps
Traping sending enabled
Trap destination
Trap destination port
Trap community
Queries allowed
Query password
Diagnistic traps
Diagnostic sending enabled
Diagnostic destination
Diagnostic destination port
Diagnostic community
Diagnostic to generic destination
QoS report traps
QoS traps to QCU
QCU address
QCU port
QCU community
QoS togeneric destination
Submit
162
****
12010
*****
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--SNMP
|--Queries allowed
|--Query password
|--Trap sending enabled
|--Trap destination
|--Trap destination port
|--Trap community
|--Diag sending enabled
|--Diag destination
|--Diag destination port
|--Diag community
|--QoS traps to QCU
|--QCU address
|--QCU port
|--QCU community
|
--QoS to generic dest.
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Security
3.4
3.4.1
Security
Speech Encryption
3.4.1.1
General Configuration
OpenStage phones support secure (i.e. encrypted) speech transmission via SRTP. For enabling secure (encrypted) calls, a TLS connection to the Phone Administration server is required.
If Use secure calls is activated, the encryption of outgoing calls is enabled, and the phone is capable of receiving encrypted calls. When the phone is connected to an OpenScape Voice system, call security is communicated to the user as follows:
• An icon in the call view tells the user whether a call is secure (encrypted) or not.
• If an active call changes from secure to insecure, e. g. after a transfer, a popup window and an alert tone will notify the user.
> For secure (encrypted) calls, it is required that both endpoints support SRTP. The secure call indication tells the user that the other endpoint has acknowledged the secure connection.
> In order to use SRTP, the phone must be configured for NTP (for further information
please see Section 3.5.5, “Date and Time”). The reason is that the key generation
(MIKEY) uses the system time of the particular device as a basis. Thus, encryption will only work correctly if all devices have the same UTC time.
If SIP server certificate validation resp. Backup SIP server certificate validation is activated, the phone will validate the server certificate sent by the Phone Administration server in order to establish a TLS connection. The server certificate is validated against the root certificate from the trusted certificate authority (CA), which must be stored on the phone first. For delivering the root certificate, a DLS (OpenScape Deployment Service) server is required.
The SRTP type sets the key exchange method for SRTP.
When Use SRTCP is activated (together with Use secure calls), the phone will use SRTCP
(Secure RTCP) to transmit and receive RTP control packets.
>
If SRTP is enabled, ANAT interworking (see Section 3.5.8.3, “Media/SDP”) is only
possible if SDES is configured as the key exchange protocol for SRTP.
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Administration via WBM
System > Security > System
System
SIP server certificate validation
Use secure calls
SRTP type
Submit
Use SRTCP
MIKEY
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Security
|---
Server cerfificate
|---
Use secure calls
|
|---
SRTP type
---
Use SRTCP
3.4.1.2
MIKEY Configuration
MIKEY (Multimedia Internet KEYing) is a key management protocol that is intended for use with real-time applications. It can specifically be used to set up encryption keys for multimedia sessions that are secured using SRTP.
Use secure calls activates the encryption of outgoing calls, i.e. the phone is capable of receiving encrypted calls.
> For secure (encrypted) calls, it is required that both endpoints support SRTP. The secure call indication tells the user that the other endpoint has acknowledged the secure connection.
The SRTP type sets the key exchange method (negotiation method) for secure calls via SRTP.
The following encryption key exchange methods are available:
• MIKEY
•
SDES (see Section 3.4.1.3, “SDES Configuration”)
The SRTP Type and Use SRTCP options are only available for secure (encrypted) calls, i.e. these parameters are only enabled if Use secure calls is activated.
When Use SRTCP is activated (together with Use secure calls), the phone will use SRTCP
(Secure RTCP) to transmit and receive RTP control packets.
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>
If SRTP is enabled, ANAT interworking (see Section 3.5.8.3, “Media/SDP”) is only
possible if SDES is configured as the key exchange protocol for SRTP.
Administration via WBM
System > Security > System
System
SIP server certificate validation
Use secure calls
SRTP type
Use SRTCP
Submit
MIKEY
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Security
|--Server cerfificate
|--Use secure calls
|
|--SRTP type
--Use SRTCP
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3.4.1.3
SDES Configuration
When "SDES" is selected as SRTP negotiation method (see Section 3.4.1.1, “General Configuration”), it can be configured further.
The SDES status parameter enables or disables SDES, just like SRTP type in System > Se-
curity > System (see Section 3.4.1.1, “General Configuration”). When SDES is disabled,
MIKEY will be used.
The SDP negotiation parameter specifies whether the use of SRTP will be forced by the phone. The following choices are available:
• "RTP + SRTP" - Both non-encrypted (non-secure) and encrypted (secure) media connections are offered. Non-encrypted connections are preferred over encrypted connections, i.e. the phone uses the non-encrypted RTP connection if the remote party accepts it and only switches to SRTP if RTP is not accepted.
• With "SRTP only", only an encrypted (secure) media connection is allowed; if the remote party should not support SRTP, no connection will be established.
• With "SRTP + RTP", the phone will try to establish an SRTP connection, but fall back to
RTP if this should fail. This is the recommended option.
With SHA1-80 ranking and SHA1-32 ranking, the ranking for each crypto-suite for negotiation is defined. Additionally, each crypto-suite can be enabled or disabled.
Administration via WBM
System > Security > SDES config
SDES config
SDES status
SDP negotiation
SHA1-80 ranking
SHA1-32 ranking
Submit
Disabled
SRTP + RTP
X
X
Reset
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3.4.2
Access Control
The CCE access parameter controls TCP access for the CCE (CommsChannel Extender).
This affects the operation of the OpenStage Manager, local CTI access, and HPT access.
When "Disable" is selected, TCP is disabled. With "Enable", there are no restrictions.
With Factory reset claw, the ’hooded claw’ keypad mechanism to initiate a factory reset without requiring an authenticated access can be enabled or disabled.
The Serial port parameter controls access to the serial port. When set to "No password", a terminal connected to the port can interact with the phone’s operating system without restrictions.
When "Passwd reqd" is selected, the serial port requires a password for access (root user is not available). When "Unavailable" is chosen, the serial port is not accessible.
As a prerequisite, the root user needs to create a user and to define a password via Serial Access, so that access can be granted when the Password required prompt is issued.
Administration via WBM
System > Security > Access control
Access control
CCE access
Factory reset claw
Serial port
Submit
Enable
;
No Password
Reset
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3.4.3
Security Log
A circular security log is used to capture important security specific events. It can be exported as CSV data to an external application for analysis.
> The security log cannot be disabled.
The Max. lines parameter defines the maximum number of entry lines that can be kept in the security log before old entries are overwritten by new entries.
Archive to DLS controls whether the log is sent to the DLS. When activated, the DLS is used to automatically archive the security log so that no log entries will be lost.
With Archive when at, the trigger for log archiving is set. Automatic archiving of new security log entries will occur when the percentage of unarchived entries in the log is as specified or more. When set to disabled, every new entry will trigger a save (only possible via DLS). The possible values are "10%”, ”20%”, ”30%”, ”35%”, ”40%”, ”45%”, ”50%”, ”55%”, ”60%”, ”65%”,
”70%”, ”80%”, ”90%”.
The security log upload may be accomplished in two ways:
• If "Archive to DLS" is enabled, if the security log reaches the threshold % for unachieved entries, the phone will initiate an upload.
• If "Archive to DLS" is NOT enabled and the security log reaches the threshold % for unachieved entries, the phone only sets the "archive-me" flag, it does not initiate the archive.
It is up to the DLS to recognize the flag and initiate an upload.
Last archived shows the date when the security log was last archived to the DLS.
Administration via WBM
System > Security > Logging
Logging
Max. lines
Archive to DLS
Archive when at
Last archived
Submit
500
50%
20101105-0010
Reset
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3.4.4
Security-Related Faults
Security log entry shows the date and time of a loss of security log entries.
> The entries in this list are only displayed until they are reported to the DLS, which usually happens very fast. After that, the entries are automatically deleted from the phone. If the entries are not deleted automatically, they can be deleted manually by using the "Cancel faults" parameter.
OCSR failure shows the date and time when the phone was unable to connect to any certificate checking server for revoked certificates.
Admin access shows the date and time when the phone encountered multiple consecutive failures to enter the admin password.
User access shows the date and time when the phone encountered multiple consecutive failures to enter the user password.
Administration via WBM
System > Security > Faults
Faults
Security log entry
OCSR Failure
Admin access
User access
Cancel faults
Submit
20111009-2206
All
Reset
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3.4.5
Password Policy
3.4.5.1
General Policy
Expires after (days) sets the maximum validity period of a password.
Warn before (days) specifies when the user/admin is notified that his password will expire.
Force changed only affects the User password. When Force changed is activated, the user will be forced to change his/her password at next login. This only applies to users, not to administrators.
Tries allowed specifies the maximum number of password entry trials before the password is suspended. Values: 0 (no limits), 2, 3, 4, 5
No change for (hours) specifies a period before a password is allowed to be changed again.
Value range: 0 to 99
Suspended for (mins) defines how long a password will be suspended after the number of failed retries has exceeded. Value range: 0 to 99
History valid for (days) defines a period in days during which the history is valid. Passwords no longer used are kept in history lists for the user and admin passwords to prevent reuse of past passwords. This list is organised as FIFO (First In, First Out) so that it always contains the latest passwords.
Administration via WBM
Security and Policies > Password > Generic Policy
Generic policy
Expires after (days)
Warn before (days)
Force changed
Tries allowed
No change for (hours)
Suspended for (mins)
History valid for (days)
Submit
99
1
5
0
5
0
Reset
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3.4.5.2
Admin Policy
Expiry date shows the date and time when the admin password will expire.
Minimum length defines the minimum number of characters for the admin password.
Password history specifies the number of entries to be kept in the admin password history.
New passwords must not match any password in the history.
The Current status parameter determines the status for the admin password. When set to "Active", the admin password is available for use. With "Suspended", the admin password is not available for a period or until reset. When set to "Disabled", all access via the admin password is disabled. The status of the admin password can only be set via DLS/WPI. It is changed internally to "suspended" when the password has been entered incorrectly more times than allowed.
Administration via WBM
Security and Policies > Password > Admin Policy
Admin policy
Expiry date
Minimum length
Password history
Current status
Submit
2038-01-19T03:14:07+00:00
6
0
Active
Reset
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3.4.5.3
User Policy
Expiry date shows the date and time when the user password will expire.
Minimum length defines the minimum number of characters for the user password.
Password history specifies the number of entries to be kept in the user password history.
The Current status parameter determines the status for the user password. When set to "Active", the user password is available for use. With "Suspended", the user password is not available for a period or until reset. When set to "Disabled", all access via the user password is disabled.
Administration via WBM
Security and Policies > Password > User Policy
User policy
Expiry date
Minimum length
Password history
Current status
Submit
2038-01-19T03:14:07+00:00
6
0
Active
Reset
3.4.5.4
Character Set
The composition of the password can be configured in detail.
Ucase chars reqd. defines the minimum number of uppercase characters. Value range: 0 to
24
Lcase chars reqd. defines the minimum number of lowercase characters. Value range: 0 to 24
Digits required defines the minimum number of digits. 0 to 24
Special chars reqd defines the minimum number of special characters. The set of possible characters is ` - = [ ] ; ’ # \ , . / ¬ ! ” £ $ % ^ & * ( ) _ + { } : @ ~ | < > ?
Value range: 0 to 24
Bar repeat length specifies the maximum number of consecutive uses of a character. Value range: 0 to 24, but not 1 (with 1 set as value, no password would be valid, because it would be forbidden to use any character once).
Min char difference specifies the minimum number of characters by which a new password must differ from the previous password. Value range: 0 to 24
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Administration via WBM
Security and Policies > Password > Character set
Character set
Ucase chars reqd.
Lcase chars reqd.
Digits required
Special chars reqd
Bar repeat length
Min char difference
Submit
0
0
0
0
0
0
Reset
3.4.5.5
Change Admin and User password
The passwords for user and administrator can be set here. They have to be confirmed after entering. The factory setting for the Admin password is "123456"; it should be changed after
the first login (Password handling in previous versions see Section 3.20, “Password”).
Administration via WBM
Security and Policies > Password > Change Admin password
Change Admin password
Old password
New password
Confirm password
Submit Reset
Security and Policies > Password > Change User password
Change User password
Admin password
New password
Confirm password
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Security & policies
|
---
Password
|---
Change Admin password
|
|
|
|
|
|
---
Current password
---
New password
---
Confirm password
---
Change User password
|
---
Admin password
|
---
New password
|
---
Confirm password
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Administration
Security
3.4.6
Certificate Policy
3.4.6.1
Online Certificate Check
The Online Certificate Status Protocol (OCSP) is used to check if a certificate to be used has been revoked. This protocol is used to query an Online Certificate Status Responder (OCSR) at the point when the certificate is being validated. The address of an OCSR can be configured on the phone and can also be obtained from the certificate to be checked (which will have the priority).
When OCSP check is activated, the configured OCSR is requested to check if the certificate has been revoked.
OCSR 1 address specifies the IP address (or FQDN) of a primary OCSP responder.
OCSR 2 address specifies the IP address (or FQDN) of a secondary OCSP responder.
Administration via WBM
Security and Policies > Certificates > Generic
Generic
OCSP check
OCSR 1 address
OCSR 2 address
Submit Reset
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3.4.6.2
Server Authentication Policy
For individual certificates provided by specific servers, the level of authentication can be configured. When "None" is selected, no certificate check is performed. With "Trusted", the certificate is only checked against the signature credentials provided by the remote entity for signature, and the expiry date is checked. When "Full" is selected, the certificate is fully checked against the credentials provided by the remote entity for signature, the fields must match the requested subject/usage, and the expiry date is checked.
Secure file transfer sets the authentication level for the HTTPS server to be used (see Section
3.16.2, “Common FTP/HTTPS Settings”).
Secure send URL sets the authentication level for the server to which special HTTP requests
are sent on key press ("Send URL" function, see Section 3.8.30, “Send Request via HTTP/HT-
TPS”).
Secure SIP server sets the authentication level for the SIP server connected to the phone (see
Section 3.5.7, “SIP Registration”).
Secure 802.1x sets the authentication level for the 802.1x authentication server.
XML Applications sets the authentication level for the XML applications server (see Section
3.19, “Applications”).
Administration via WBM
Security and Policies > Certificates > Authentication policy
Authentication policy
Secure file transfer
Secure send URL
Secure SIP server
Secure 802.1x
XML Applications
Submit
None
None
None
None
None
Reset
Administration via Local Phone
|
--Admin
|
--Security & policies
|
--Certificates
|
--Authentication policy
|
|--Secure file transfer
--Secure send URL
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System Settings
3.5
3.5.1
System Settings
Terminal and User Identity
3.5.1.1
Terminal Identity
Within a SIP environment, both Terminal Number and Terminal Name may serve as a phone number. The values are used in the userinfo part of SIP URIs.
In order to register with a SIP registrar, the phone sends REGISTER messages to the registrar containing the contents of Terminal number.
Data required
• Terminal number: Number to be registered at the SIP registrar.
• Terminal name: Name to be registered at the SIP registrar.
Administration via WBM
System > System Identity
System Identity
Terminal number
Terminal name
Display identity
Enable ID
Web name
DNS name construction
Submit
4711 openstage
4711
;
Only number
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Identity
|
|---
Terminal number
---
Terminal name
3.5.1.2
Display Identity
If an individual name or number is entered as Display identity and Enable ID is activated, it is displayed in the phone’s status bar instead of the Terminal number.
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System > System Identity
System Identity
Terminal number
Terminal name
Display identity
Enable ID
Web name
DNS name construction
Submit
4711 openstage
4711
;
Only number
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Identity
|
|---
Display identity
---
Enable ID administration.fm
Administration
System Settings
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Administration
System Settings
3.5.2
Emergency and Voice Mail
It is important to have an Emergency number configured. If the phone is locked, a clickable area for making an emergency call is created.
> If more than one emergency number is needed, additional numbers can be configu-
red in the canonical dial settings (Section 3.13.1, “Canonical Dialing Configuration”).
If a mailbox located at a remote server shall be used, its Voice mail number must be entered.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
; alert-internal
;
Disabled
Off
Reset
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Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|
|---
Emergency number
---
Voicemail number
3.5.3
Energy Saving (OpenStage 40/60/80)
After the phone has been inactive within the timespan specified in Backlight time, the display backlight is switched off to save energy.
The possible values are: 1 minute, 5 minutes, 30 minutes, 60 minutes, 2 hours, 4 hours or 8 hours. Moreover, with OpenStage 40 and 60, this parameter can also be configured by the user.
Administration via WBM
Local functions > Energy saving
Energy saving
Backlight time
Submit
2 hours
Reset
Administration via Local Phone
|
---
Admin
|
---
Local Functions
|
---
Energy saving
|
---
Backlight time
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System Settings
3.5.4
Call logging
This configuration item allows the phone to detect if a number dialled by the User is likely to be a Feature Access Code (FAC) by comparing the start of the dialled number with the configured
FAC prefixes. If the dialled number does match a FAC prefix and the SIP server has provided a different number for the called party then the number shown in the Dialled tab list of Call Log is changed from the dialled number to the server provided number. If the new configuration item is left empty then the Dialled tab list display will remain as currently populated (i.e. the dialled number is shown in the list).
A further enhancement for an entry matched to a FAC in the Dialled tab list of Call Log is that the context menu for the list entry now provides both numbers from the last call associated with the entry as Dial options in the context menu for the list entry (similar to that already provided by the context menu for the Details form of such an entry). Note that the Call Log display on
OS15 has been simplified [3] so that an entry only displays a name or a number (not both) and there is no access to entry details. However this only limits the display and the default dialling number for an OS15 entry is determined as above.
Call Log entry grouping rules for the Dialled tab list remain unchanged, if multiple FACs all map to numbers associated with one contact then they are grouped together.
Data required
• FAC prefixes: A comma separated list of feature prefixes considered to represent feature codes configured at the SIP server for abbreviated dialling.
Administration via WBM
Local functions > Call logging
Call logging
FAC prefixes
Submit Reset
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3.5.4.1
Logging of Missed Calls (via User menu)
This feature allows the user to
• distinguish logged calls based on the device on which the calls were completed, and
• decide whether missed calls that were answered elsewhere shall be
– included into the call log, or
– excluded from the call log, i.e. not logged at all
• decide whether a number which also exists in missed calls tab of call log is to be deleted from call log when this number is called
– manually
– when called
In the Call Lists, missed calls that were completed elsewhere are marked with a check mark.
For details, please refer to the User manual.
Forwarded calls are not logged under "Missed calls", but under "Forwarded" in the call log.
Administration via WBM (User menu)
User > Configuration > Call logging > General
User > Configuration > Call logging > Missed calls
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System Settings
Administration via Local Phone (User menu)
|
---
User
|
---
Configuration
|
---
Call logging
|---
General
|
|
| ---
FAC prefixes
---
Missed calls
|---
Answered elsewhere
| |---
Include
|
---
Exclude
|
|
---
Delete entry
|---
Manually
|
---
When called
Answered elsewhere > Include: Calls completed elsewhere will be logged as missed calls. In the call log these calls are marked with a check mark.
Answered elsewhere > Exclude: Calls completed elsewhere will not be visible on phone; they will not be logged at all.
Delete entry > Manually : Call numbers remain in call log until they are deleted manually.
Delete entry > When called: Call numbers existing in missed call list are deleted automatically when they are called again.
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3.5.5
Date and Time
If the DHCP server in your network provides the IP address of the SNTP server, no manual configuration is necessary. If not, you have to set the SNTP IP address parameter manually.
For correct display of the current time, the Timezone offset must be set appropriately. This is the time offset from UTC (Universal Time Coordinated). If, for instance, the phone is located in
Munich, Germany, the offset is +1 (or simply 1); if it is located in Los Angeles, USA, the offset is -8. For countries or areas with half-our time zones, like South Australia or India, non-integer values can be used, for example 10.5 for South Australia (UTC +10:30).
If the phone is located in a country with DST (Daylight Saving Time), you can choose whether
DST is toggled manually or automatically. For manual toggling, disable Auto time change and enable or disable Daylight saving; the change will be in effect immediately. For automatical toggling, enable Auto time change; now, daylight saving is controlled by the DST zone / Time
zone parameter. This parameter determines when DST starts or ends, and must be set according to the location of the phone.
The Difference (minutes) parameter defines how many minutes the clock is put forward for
DST. In Germany, for instance, the value is +60.
> Please note that Difference (minutes) must be specified both for manual and automatic DST toggling.
3.5.5.1
SNTP is Available, but no Automatic Configuration by DHCP server
Data required
• SNTP IP address: IP address or hostname of the SNTP server.
• Timezone offset (hours): Shift in hours corresponding to UTC.
• Daylight saving: Enables or disables daylight saving time in conjunction with Auto time
change.
Value range: "Yes", "No"
Default setting for OpenStage 40 US is "Yes". After a factory reset, the system will be reset to this value.
• Difference (minutes): Time difference when daylight saving time is in effect.
Default setting for OpenStage 40 US is "60 (mins)". After a factory reset, the system will be reset to this value.
• Auto time change / Auto DST: Enables or disables automatic control of daylight saving time according to the Time zone.
Value range: "Yes", "No"
Default setting for OpenStage 40 US is "Yes". After a factory reset, the system will be reset to this value.
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System Settings
• Time zone / DST zone: Area with common start and end date for daylight saving time.
Value range: "Australia 2007 (ACT, South Australia, Tasmania, Victoria)", "Australia 2007
(New South Wales)", "Australia (Western Australia)", "Australia 2008+ (ACT, New South
Wales, South Australia, Tasmania, Victoria)", "Brazil", "Canada", "Canada (Newfoundland)", "Europe (Portugal, United Kingdom)", "Europe (Finland)", "Europe (Rest)", "Mexico", "United States", "New Zealand", "New Zealand (Chatham)".
Default setting for OpenStage 40 US is "United States". After a factory reset, the system will be reset to this value.
Administration via WBM
Date and Time
Time source
Date and time
SNTP IP address
Timezone offset (hours)
Daylight saving
Submit
Daylight saving
Difference (minutes)
Auto time change
DST zone Europe (Rest)
192.43.244.18
1
60
;
Reset
Administration via Local Phone
|
--Admin
|
--Date and Time
|
|--SNTP IP address
--Timezone offset
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System Settings
3.5.5.2
No SNTP Server Available
If no SNTP server is available, date and time must be set manually.
> The manual setting of time and date is located in the user menu, not in the administrator menu.
Data required
• Local time (hh:mm): Local time.
• Local date (day, month, year): Local date.
• Allow daylight saving: Defines whether there is daylight is set.
• Difference (minutes): Timezone offset in minutes.
• Auto time change / Auto DST: Enables or disables automatic control of daylight saving time according to the Time zone.
Value range: "Yes", "No"
Default setting for OpenStage 40 US is "Yes". After a factory reset, the system will be reset to this value.
Administration via WBM (User menu)
User > Date and time
Date and time
Local Time (hh:mm):
Local Date (day,month,year):
Allow daylight saving:
Difference (minutes):
Submit
30
15
November
: 44
87678
Reset
2006
Administration via Local Phone
|
---
Menu
|
---
Date and Time
|---
Time
|---
Date
|---
Daylight saving
|---
Difference (mins)
|
---
Auto DST
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System Settings
3.5.6
SIP Addresses and Ports
3.5.6.1
SIP Addresses
In this group of parameters, the IP addresses or host names for the SIP server, the SIP registrar, and the SIP gateway are defined.
SIP server address provides the IP address or host name of the SIP proxy server
(OpenScape Voice). This is necessary for outgoing calls. SIP registrar address contains the
IP address or host name of the registration server, to which the phone will send REGISTER messages. When registered, the phone is ready to receive incoming calls. SIP gateway ad-
dress gives the IP address or host name of the SIP gateway. If configured, the SIP gateway is used for outgoing calls; otherwise the server specified in SIP server address is used. A SIP gateway is able to perform a conversion of SIP to TDM, which enables to send calls directly into the public network.
> Enhanced survivability using DNS SRV is available. To make use of it, a special con-
figuration is required. For details, please refer to Section 3.5.10, “Resilience and
Survivability”.
Data required
• SIP server address: IP address or host name of the SIP proxy server.
• SIP registrar address: IP address or host name of the registration server.
• SIP gateway address: IP address or host name of the SIP gateway.
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System Settings
Administration via WBM
System > Registration
Registration
SIP Addresses
SIP server address
SIP registar address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
OS Voice
3600
UDP
;
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Registration
|
--SIP Addresses
|--SIP server
|--SIP registrar
|
--SIP gateway
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System Settings
3.5.6.2
SIP Ports
In this group of parameters, the ports for the SIP server, the SIP registrar, and the SIP gateway
are defined (for further information see Section 3.5.6.1, “SIP Addresses”), as well as the SIP
port used by the phone (SIP local).
Data required
• SIP server: Port of the SIP proxy server.
Default: 5060.
• SIP registrar: Port of the server at which the phone registers.
Default: 5060.
• SIP gateway: Port of the SIP gateway.
Default: 5060.
• SIP local: Port used by the phone for sending and receiving SIP messages.
Default: 5060.
> When changing the SIP Transport protocol from UDP/TCP to TLS, the SIP port now also have to be changed correspondingly (e.g. SIP port from 5060 to 5061) and on changing vice versa.
Administration via WBM
Network > Port configuration
SIP Server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port mode
PC port autoMDIX
Submit
Port configuration
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
Administration via Local Phone
|
--Admin
|
--Network
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System Settings
|
---
Port Configuration
|---
SIP server
|---
SIP registrar
|
|---
SIP gateway
---
SIP local
3.5.7
SIP Registration
Registration is the process by which centralized SIP Server/Registrars become aware of the existence and readiness of an endpoint to make and receive calls. The phone supports a number of configuration parameters to allow this to happen. Registration can be authenticated or un-authenticated depending on how the server and phone is configured.
For operation with an OpenScape Voice server, set Server type to "OS Voice". When HiQ8000 is to be used, set it to "HiQ8000". The expiry time of a registration can be specified by Regis-
tration timer.
Unauthenticated Registration
For unauthenticated registration, the following parameters must be set on the phone: Terminal
number or Terminal name (see Section 3.5.1.1, “Terminal Identity”), SIP server and SIP regis-
trar address (see Section 3.5.6.1, “SIP Addresses”).
In unauthenticated mode, the server must pre-authenticate the user. This procedure is server specific and is not described here.
Authenticated Registration
The phone supports the digest authentication scheme and requires some parameters to be configured in addition to those for unauthenticated registration. By providing a User ID and a
Password which match with a corresponding account on the SIP registrar, the phone authenticates itself. Optionally, a Realm can be added. This parameter specifies the protection domain wherein the SIP authentication is meaningful. The protection domain is globally unique, so that each protection domain has its own arbitrary usernames and passwords.
> A challenge from the server for authentication information is not only restricted to the
REGISTER message, but can also occur in response to other SIP messages, e. g.
INVITE.
> If registration has not succeeded at startup or registration fails after having been previously successfully registered the phone will try to re-register every 30 seconds.
This is not configurable.
If the registration is not answered at all, the phone will try to re-register every 60 sec-
onds by default. This is configurable (see Maximum Registration Backoff Timer
Æ page 120).
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Administration
System Settings
Data required
• Registration timer (seconds): Expiry time of the registration in seconds.
Default value: 3600.
• Server type: Type of server the phone will register to.
Value range: "Other", "OS Voice", "HiQ8000", "Genesys"
Default value: "OS Voice"
• Realm: Protection domain for authentication.
• User ID: Username required for an authenticated registration.
• Password: Password required for an authenticated registration.
Administration via WBM
System > Registration
Registration
SIP Addresses
SIP server address
SIP registar Address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
OS Voice
3600
UDP
;
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Registration
|
---
SIP Session
|--Registration timer
|--Server type
|--Realm
|--User ID
|
--Password
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3.5.8
SIP Communication
3.5.8.1
Outbound Proxy
If this option is set to "Yes", the phone routes outbond requests to the configured proxy. The outbound proxy will fulfill the task of resolving the domain contained in the SIP request. If "No" is set, the phone will attempt to resolve the domain by itself.
If a Default OBP (Outbound Proxy) domain is set and the number or name dialed by the user does not provide a domain, this value will be appended to the name or number. Otherwise, the domain of the outbound proxy will be appended.
Data required
• Outbound proxy: Determines whether an outbound proxy is used or not.
Value range: "Yes", "No"
Default: "Yes"; when System > Registration > Server type is set to "HiQ8000" (firmware version V3 onwards): "Yes"
• Default OBP domain: Alternative value for the domain that is given in the outbound request.
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
UDP
32000
32000
60
0
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
SIP Interface
|
|---
Outbound proxy
---
Default OBP domain
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System Settings
3.5.8.2
SIP Transport Protocol
Selects the transport protocol to be used for SIP messages. The values "UDP", "TCP", and
"TLS" are available. The default is "UDP"; default when System > Registration > Server type is set to "HiQ8000" (firmware version V3 onwards): "TLS".
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
UDP
32000
32000
60
0
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--SIP Interface
|
--SIP transport
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System Settings
3.5.8.3
Media/SDP
OpenStage phones support IPv4/IPv6 media address negotiation in SDP using ANAT (Alternative Network Address Types). ANAT allows for for the expression of alternative network addresses (e. g., different IP versions) for a particular media stream.
When Media negotation is set to "ANAT", ANAT is supported; the phone will re-register with the SIP server and advertise ANAT support in the SIP header. When set to "Single IP", ANAT support is disabled.
> If SRTP is enabled, ANAT interworking is only possible if SDES is configured as the
key exchange protocol for SRTP (see Section 3.4.1.1, “General Configuration”).
Media IP mode defines which IP version is to be used for voice transmission. With "IPv4", only
IPv4 is used; with "IPv6", only IPv6 is used; with "IPv4_IPv6", both IPv4 and IPv6 can be used, but IPv4 is preferred; with "IPv6_IPv4", both IPv6 and IPv4 can be used, but IPv6 is preferred.
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Keep alive format
Media Negotiation
Media IP Mode
Submit
;
UDP
32000
32000
60
0
Sequence
Single IP
IPv4
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
SIP Interface
|
|---
Media negotiation
---
Media IP mode
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System Settings
3.5.9
SIP Session Timer
Session timers provide a basic keep-alive mechanism between 2 user agents or phones. This mechanism can be useful to the endpoints concerned or for stateful proxies to determine that a session is still alive. This is achieved by the phone sending periodic re-INVITEs to keep the session alive. If no re-INVITE is received before the interval passes, the session is considered terminated. Both phones are supposed to terminate the call, and stateful proxies can remove any state for the call.
This feature is sufficiently backward compatible such that only one end of a call needs to implement the SIP extension for it to work.
The parameter Session timer enabled determines whether the mechanism shall be used, and
Session duration (seconds) sets the expiration time, and thus the interval between refresh re-INVITEs.
> Some server environments support their own mechanism for auditing the health of a session. In these cases, the Session timer must be deactivated.
For OpenScape Voice, the Session timer should be deactivated.
Data required
• Session timer enabled: Activates or deactivates the session timer mechanism.
Value range: "Yes", "No"
Default value: "No"
• Session duration (seconds): Sets the expiration time for a SIP session.
Default: 3600
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System Settings
Administration via WBM
System > Registration
Registration
SIP Addresses
SIP server address
SIP registar Address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
OS Voice
3600
UDP
;
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Registration
|
--SIP session
|
|--Session timer
--Session duration
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3.5.10
Resilience and Survivability
To allow for stable operation even in case of network or server failure, OpenStage phones have the capability of switching to a fallback system. The switchover is controlled by various configurable check and timeout intervals.
Survivability is achieved in two different ways:
1. DNS SRV can be used for enhanced survivability, either in a scenario with a survivability proxy, or in a scenario with multiple primary SIP servers. The DNS server provides the phone with a prioritized list of SIP servers via DNS SRV. The phone fetches this list periodically from the server, depending on the TTL (time to live) specified for the DNS SRV records.
To enable DNS SRV requests from the phone, please make the following settings:
• Specifiy the IP address of the DNS server that provides the server list via DNS SRV.
The web interface path is Network > IP configuration > Primary DNS. For details, see
Section 3.3.7.2, “DNS Servers”.
• Enable the use of an outbound proxy for routing outbound requests. The web interface
path is System > SIP interface > Outbound proxy. For details, see Section 3.5.8.1,
“Outbound Proxy”.
• Set the SIP gateway port to 0. The web interface path is Network > Port configuration
> SIP gateway. Alternatively, if the SIP server otherwise specified in System > Registration > SIP server address is to be configured by DNS SRV, set the SIP server port to 0. The web interface path is Network > Port configuration > SIP server. For details,
see Section 3.5.6.2, “SIP Ports”.
• As SIP gateway address, enter the DNS domain name for which the DNS SRV records are valid. The web interface path is System > Registration > SIP gateway address. Alternatively, if the SIP server otherwise specified in System > Registration > SIP server address is to be configured by DNS SRV, set the mentioned parmeter to the DNS do-
main name for which the DNS SRV records are valid. For details, see Section 3.5.6.1,
“SIP Addresses”.
A survivability proxy acts as a relay between the phone and the primary SIP server. Thus, the address of the survivability proxy is specified as gateway or SIP server at the phone
(see Section 3.5.7, “SIP Registration”). When the TLS connection between the survivability
proxy and the SIP server breaks down, e. g. because of server failure, the survivable proxy itself acts as a replacement for the primary SIP server. Vice versa, in case the phone can not reach the survivability proxy itself, it will register directly with the primary SIP server, provided that it is specified in the DNS SRV server list.
The survivability proxy notifies the phone whenever the survivability changes, so it can indicate possible feature limitations to the user. Furthermore, to enhance survivability, the phone will be kept up-to-date about the current survivability state even after a restart.
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Another way to realize survivability is the use of multiple, geographically separated SIP servers. Normally, the phone is registered with that server that has the highest priority in the DNS SRV server list. If the highest priority server fails to respond to the TLS connec-
tivity check (see Section 3.5.10.1, “TLS Connectivity Check”), the phone will register with
the server that has the second highest priority.
2. Use of a Backup SIP Server. Along with the registration at the primary SIP server, the phone is registered with a backup SIP server. In normal operation, the phone uses the primary server for outgoing calls. If the phone detects that the connection to the primary SIP server is lost, it uses the backup server for outgoing calls. This connection check is realized
by 2 timers; for details, see Section 3.5.10.2, “Response Timer” and Section 3.5.10.3,
“Non-INVITE Transaction Timer”. For configuring the backup server, please refer to Section 3.5.10.5, “Backup SIP Server”.
> In survivability mode, some features will presumably not be available. The user will be informed by a message in the Call View display.
3.5.10.1
TLS Connectivity Check
A regular check ensures that the TLS link to the main SIP server is active. When the Connec-
tivity check timer is set to a non-zero value, test messages will be sent at the defined interval.
If the link is found to be dead, the phone uses DNS SRV to find another SIP server. Certainly, the DNS SRV records must be properly configured in the DNS server.
If no other primary SIP server is found via DNS SRV, the phone will switch over to a backup
server for making receiving calls. For configuring the backup server, please refer to Section
3.5.10.5, “Backup SIP Server”.
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
Connectivity check timer (seconds)
Submit
TLS
3700
10
Reset
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3.5.10.2
Response Timer
The Response Timer resp. Call trans timer is started whenever the phone sends a new IN-
VITE message to the SIP server.
If the call transaction timer expires before the phone gets a response from the SIP server, the phone assumes that the server had died and then attempts to contact the backup server, if configured. If there is no backup server configured, the phone just tidies up internally.
The data is given in milliseconds. The default value is 32 000; for Phone Administration, the recommended setting is 3.7 seconds (3700 ms).
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
UDP
32000
32000
60
0
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
SIP Interface
|
---
Call trans. (ms)
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3.5.10.3
Non-INVITE Transaction Timer
The NonCall trans timer is started whenever the phone sends a non-INVITE message to the
SIP server. If the timer expires before the phone gets a response from the SIP server, the phone assumes that the server had died and then attempts to contact the backup server, if configured. If no backup server is configured, the phone will just tidy up internally.
The data is given in milliseconds. The default value is 32 000; for Phone Administration, the recommended setting is 6 seconds (6000 ms).
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
UDP
32000
32000
60
0
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
SIP Interface
|
---
NonCall transactions (ms)
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3.5.10.4
Maximum Registration Backoff Timer
If a registration attempt should result in a timeout, the phone waits a random time before sending another REGISTER message. The Reg. backoff (seconds) parameter determines the maximum waiting time.
Administration via WBM
System > SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
UDP
32000
32000
60
0
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
SIP Interface
|
---
Reg. backoff
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3.5.10.5
Backup SIP Server
The Backup registration flag indicates whether or not the phone treats the backup proxy server as a SIP registrar. If set to "Yes", the phone tries to register its SIP address with the server whose IP address or hostname is specified by Backup proxy address. Once an IP address has been entered, the SIP-UDP Port is opened, even if SIP-TLS is used for the OS Voice connection.
The Backup registration timer determines the duration of a registration with the backup SIP server.
The Backup transport option displays the current transport protocol used to carry SIP messages to the Backup proxy server.
The Backup OBP flag indicates whether or not the Backup proxy server is used as an outbound proxy.
Data required
• Backup registration allowed / Backup registration flag: Determines whether or not the backup proxy is used as a SIP Registrar.
Value Range: "Yes", "No"
Default: "Yes"
• Backup proxy address: IP address or hostname of the backup proxy server.
• Backup registration timer: Expiry time of the registration in seconds.
Default: 3600
• Backup transport: Transport protocol to be used for messages to the backup proxy.
Value range: "TCP", "UDP"
Default: "UDP"
• Backup OBP flag: Determines whether or not the backup proxy is used as an outbound proxy.
Value range: "Yes", "No"
Default: "No"
• Network > Port Configuration > Backup proxy: Port of the backup proxy server.
Default: 5060
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Administration via WBM
System > Registration
Registration
SIP Addresses
SIP server address
SIP registar Address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
HiQ8000
3600
UDP
;
Reset
Network > Port configuration
Port configuration
SIP Server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port mode
PC port autoMDIX
Submit
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
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---
Admin
|
---
System
|
---
Registration
|
---
SIP session
|
---
SIP survivability
|---
Backup reg. flag
|---
Backup proxy addr.
|---
Backup reg timer
|---
Backup transport
|
---
OBP flag
|
---
Admin
|
---
Network
|
---
Port Configuration
|
---
Backup proxy administration.fm
Administration
System Settings
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Feature Access
3.6
Feature Access
Certain OpenStage features and interfaces can be enabled or disabled:
•
Blind transfer (see Section 3.8.9, “Blind Call Transfer / Move Blind”)
• 3rd call leg (consultation from a second call; see user manual)
•
Callback busy (see Section 3.8.21, “Callback” and Section 3.7.6, “Callback URIs”)
•
Callback no reply (see Section 3.8.21, “Callback” and Section 3.7.6, “Callback URIs”)
•
Call pickup (see Section 3.8.20, “Directed Pickup”)
•
Group pickup (see Section 3.8.16, “Group Pickup”)
•
Call deflection (see Section 3.8.11, “Deflect a Call”)
•
Call forwarding (see Section 3.8.4, “Call Forwarding (Standard)”)
•
Do not disturb (see Section 3.8.15, “Do Not Disturb”)
•
Refuse call (see Section 3.7.1, “Allow Refuse”)
•
Repertory dial key (see Section 3.8.17, “Repertory Dial”)
•
Ext/int forwarding (see Section 3.8.5, “Call Forwarding by Call Type”)
• Phone book lookups (see user manual)
•
DSS feature (see Section 3.11.5, “Direct Station Select (DSS)”)
•
BLF feature (see Section 3.8.28, “BLF Key”)
• Line overview (see user manual)
• Video calls (see user manual)
•
Callback cancel (see Section 3.8.22, “Cancel Callbacks” and Section 3.7.6, “Callback
URIs”)
•
CTI control (see Section 3.7.11, “uaCSTA Interface”)
•
Bluetooth (see Section 3.27, “Bluetooth (OpenStage 60/80)”)
•
Web based manag. (see Section 1.6.1, “Web-based Management (WBM)”)
• USB device access (see user manual)
• Backup to USB (see user manual)
•
Feature toggle (see Section 3.8.18, “Hunt Group: Send Busy Status”)
• Phone lock (see user manual)
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System > Features > Feature access
Feature access
Call control
Blind transfer
3rd call leg
Call establish
Callback
Call pickup
Group pickup
Call deflection
Call forwarding
Do not disturb
Refuse call
Repertory dial key
Ext/int forwarding
;
;
;
;
;
;
;
;
;
;
;
Call associated
Phone book lookups
DSS feature
BLF feature
Line overview
Video calls
CTI
CTI control
Services
Bluetooth
Web based manag.
USB device access
Backup to USB
Feature toggle
Phone lock
Submit
;
;
;
;
;
;
Reset
;
;
;
;
;
;
Administration via Local Phone
|
--Admin
|
--System
|
--Features
|
--Feature access
|
--Call control
|
--Blind transfer
|
--3rd call leg
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Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Feature access
|
---
Call establish
|
---
Callback
|
---
Call pickup
|
---
Group pickup
|
---
Call deflection
|
---
Call forwarding
|
---
Do not disturb
|
---
Refuse call
|
---
Repertory dial key
|
---
Ext/int forwarding
|
---
Admin
|
---
System
|
---
Features
|
---
Feature access
|
---
Call associated
|
---
Phone book lookups
|
---
DSS feature
|
---
BLF feature
|
---
Line overview
|
---
Video calls
|
---
Admin
|
---
System
|
---
Features
|
---
Feature access
|
---
CTI
|
---
CTI control
|
---
Admin
|
---
System
|
---
Features
|
---
Feature access
|
---
Services
|
---
Bluetooth
|
---
Web based manag.
|
---
USB device access
|
---
Backup to USB
|
---
Feature toggle
|
---
Phone lock
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3.7
Feature Configuration
3.7.1
Allow Refuse
This parameter defines whether the Refuse Call feature is available on the phone. The possible values are "Yes" or "No". The default is "Yes".
> This parameter can also be configured under System > Features > Feature access
(see Section 3.6, “Feature Access”).
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
; alert-internal
;
Disabled
Off
Reset
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Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|
---
Allow refuse
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3.7.2
Hot/Warm Phone
If the phone is configured as hot phone, the number specified in Hot warm destination is dialed immediately when the user goes off-hook. For this purpose, Hot warm phone must be set to "Hot phone". If set to "Warm phone", the specified destination number is dialed after a delay
which is defined in Initial digit timer (seconds) (for details, see Section 3.7.3, “Initial Digit Timer”). During the delay period, the user can dial a number which will be used instead of the hot/
warm destination. In addition, the user will be provided with a dial tone during the delay period.
With the setting "No action", hot phone or warm phone functionality is disabled.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
; alert-internal
;
Disabled
Off
Reset
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Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|---
Hot / warm phone
|---
Hot / warm destination
|
---
Initial digit timer
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3.7.3
Initial Digit Timer
This timer is started when the user goes off-hook, and the dial tone sounds. When the user has not entered a digit until timer expiry, the dial tone is turned off, and the phone changes to idle mode. The Initial digit timer (seconds) parameter defines the duration of this timespan.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
; alert-internal
;
Disabled
Off
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Features
|
--Configuration
|
--General
|
--Initial digit timer
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3.7.4
Group Pickup
> This feature is only available when allowed under System > Features > Feature ac-
cess (see Section 3.6, “Feature Access”).
3.7.4.1
Feature Code
This feature allows a user to collect a call from any ringing phone that is in the same pickup group. To be a member of a Call Pickup group, the phone must be configured with the corresponding URI of the Call Pickup group service provided by the server. An example pickup URI is "**3".
Administration via WBM
> The BLF pickup code parameter is only relevant when the phone is connected to an
Asterisk server.
System > Features > Addressing
Addressing
MW server URI 192.168.1.2
Conference
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code *0
Submit Reset
3.7.4.2
Pickup alert
If desired, an incoming call for the pickup group can be indicated acoustically.
The Group pickup tone allowed parameter activates or deactivates the generation of an acoustic signal for incoming pickup group calls. The default is "Yes". If this is activated, Group
pickup as ringer determines whether the current ring tone or an alert beep is used. If set to
"Yes", a pickup group call will be signaled by a short ring tone; the currently selected rigtone is used. If set to "No", a pickup group call will be signaled by an alert tone. The default is "Yes".
Depending on the phone state and the setting for Group pickup as ringer, the group pickup tone comes from the loudspeaker, the handset, or the headset. The volumes can be set in the local user menu, under Audio > Volumes.
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The following table shows the group pickup alert behaviour for each possible scenario:
Phone State
Ringer on
Ringer off
Idle
In call
Idle
In call
Handset
Group pickup as ringer=yes
Ring tone
Speaker
Ring tone
Speaker
Handset
Open listening
Headset
Headset
Open listening
Hands-free
Beep
Handset and
Speaker
Ring tone
Speaker
Beep
Headset and
Speaker
Beep
Speaker
Nothing
Handset Nothing
Handset
Open listening
Headset
Headset
Open listening
Hands-free
Beep
Handset and
Speaker
Nothing
Beep
Headset and
Speaker
Beep
Speaker
Group pickup as ringer=no
Beep
Speaker
Beep
Handset
Beep
Handset and
Speaker
Beep
Headset
Beep
Headset and
Speaker
Beep
Speaker
Nothing
Beep
Handset
Beep
Handset and
Speaker
Beep
Headset
Beep
Headset and
Speaker
Beep
Speaker
Group pickup visual alert defines the user action required to accept a pickup call. If "Prompt" is selected, an incoming pickup call is signaled by an alert on the phone GUI. As soon as the user goes off-hook or presses the speaker key, the pickup call is accepted. Alternatively, the user can press the corresponding function key, if configured. If "Notify" is selected, an incoming pickup call is signaled by an alert on the phone GUI. To accept the call, the user must confirm the alert or press the corresponding function key, if configured.
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Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Group pickup
|---
Group pickup tone
|---
Group pickup as ringer
|
---
Group pickup visual
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Feature Configuration
3.7.5
Call Transfer
3.7.5.1
Transfer on Ring
If this function is active, a call can be transferred after the user has dialled the third participant’s number, but before the third party has answered the call. This feature is enabled or disabled in the User menu. The default is "Yes".
Administration via WBM (User menu)
User > Configuration > Outgoing calls
Outgoing calls
Autodial delay (seconds)
Allow callback
Allow busy when dialling
Allow transfer on ring
Allow immediate dialling
6
Submit
;
;
Reset
Administration via Local Phone (User menu)
|
--User
|
--Configuration
|
--Outgoing calls
|
--Transfer on ring
3.7.5.2
Transfer on Hangup
This feature applies to the following scenario: While A is talking to B, C calls A. A accepts the call, so B is on hold and the call between A and C is active. If Transfer on hangup is enabled, and A goes on-hook, B gets connected to C. If disabled, C will be released when A hangs up, and A has the possibility to reconnect to B. By default, the feature is disabled.
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Administration
Feature Configuration
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
Administration via Local Phone
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---
Admin
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System
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Features
|
---
Configuration
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General
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Transfer on hangup
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3.7.6
Callback URIs
The Callback option allows the user to request a callback on certain conditions. The callback request is sent to the SIP server. The Code for callback busy requests a callback if the line is busy, i. e. if there is a conversation on the remote phone. Code for callback no reply applies when the call is not answered, i. e. if nobody lifts the handset or accepts the call in another way.
The Code for callback cancel all all deletes all the callback requests stored previously on the telephone system/SIP server.
> The callback feature can be enabled or disabled under System > Features > Feature
access (see Section 3.6, “Feature Access”).
Data required
• Callback: FAC: Access code that si sent to the server for all kind of Callback .
• Code for callback cancel all / Callback: Cancel all: Access code for canceling all callback requests on the server.
Administration via WBM
System > Features > Addressing
Addressing
MW server URI 192.168.1.2
Conference
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code *0
Submit Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Features
|
--Addressing
|
|--Callback: FAC
--Callback: Cancel all
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Feature Configuration
3.7.6.1
Call Completion
Used with Asterisk only
Administration via WBM
System > Features > Call Completion
Call completion
Functional CCSS
Callback ringer
Allow after call (s)
; altert-internal
30
Max. callbacks 5
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Call completion
|---
Functional CCSS
|---
Callback ringer
|
---
Allow after call (s)
|
---
Max. callbacks
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3.7.7
Message Waiting Address
The MWI (Message Waiting Indicator) is an optical signal which indicates that voicemail messages are on the server. Depending on the SIP server / gateway in use, the Message waiting
server address, that is the address or host name of the server that sends message waiting notifications to the phone, must be configured.
With OpenScape Voice, this setting is not typically necessary for enabling MWI functionality.
Administration via WBM
System > Features > Addressing
Addressing
MW server URI 192.168.1.2
Conference
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code *0
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Addressing
|
---
MWI server URI
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Feature Configuration
3.7.8
Indicate Messages
The indication of old and new messages on the display can be configured. There are 4 categories of voicemail messages: new, new urgent, old, and old urgent. For each category, the administrator can define whether the message count is shown or hidden, and set a header for the category. If all four settings on the form are set to "Hide", the VoiceMail summary screen is not shown to the user. Any other permutation of "Show/Hide" settings must result in a VoiceMail summary. If theVoiceMail summary is not shown then calling the mailbox will be a single-step process on a suitably configured OS80/60 and at least a two-step process on OS40/20/15.
Data required
• New items: Determines whether new items are indicated.
Value range: "Show", "Hide"
• Alternative label: Label for new items.
• New urgent items: Determines whether new urgent items are indicated.
Value range: "Show", "Hide"
• Alternative label: Label for new urgent items.
• Old items: Determines whether new urgent items are indicated.
Value range: "Show", "Hide"
• Alternative label: Label for old items.
• Old urgent items: Determines whether old urgent items are indicated.
Value range: "Show", "Hide"
• Alternative label: Label for old urgent items.
Administration via WBM
Local functions > Messages settings
Messages settings
New items
Alternative label
Show
New urgent items Show
Alternative label
Old Items
Alternative label
Show
Old urgent items Show
Alternative label
Submit Reset
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---
Admin
|
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Locatl functions
|
---
Messages settings
|---
New items
|---
Alternative label
|---
New urgent items
|---
Alternative label
|---
Old items
|---
Alternative label
|
|---
Old urgent items
---
Alternative label administration.fm
Administration
Feature Configuration
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3.7.9
System Based Conference
The Conference URI provides the number/URI used for system based conferences, which can involve 3 to 16 members. This feature is not available with every system.
> It is recommended not to enter the full URI, but only the user part. For instance, enter
"123", not "123@<SIP SERVER ADDRESS>". A full address in this place might cause a conflict when Phone Administration uses multiple nodes.
Administration via WBM
System > Features > Addressing
Addressing
MW server URI 192.168.1.2
Conference
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code *0
Submit Reset
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3.7.10
Server Based Features
> Please note that the Server features parameter, despite the name similarity, is not
related to the Server feature functionality as described in Section 3.8.27, “Server
Feature”.
The use of server based call forwarding and server based DND is enabled or disabled here.
When phone based DND and phone based call forwarding are to be used, Server features must be deactivated. This is the default setting. For using server based Call Forwarding or server based DND, it must be activated.
> Server features is deactivated automatically if System > Registration > Server type
(see Section 3.5.7, “SIP Registration”) is set to "HiQ8000".
> Before switching Server features on or off, please ensure that both Call Forwarding and DND are not activated. Otherwise, the user will not be able to control the feature any more.
It is recommended to set Server features when setting up the phone, and avoid further changes, as possible.
>
To enable server based features, uaCSTA must be allowed (see Section 3.7.11,
“uaCSTA Interface”).
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Feature Configuration
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|
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Server features
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Feature Configuration
3.7.11
uaCSTA Interface
User Agent CSTA (uaCSTA) is a limited subset of the CSTA protocol, which allows external
CTI applications to interact with the phone.
> Access to the users “CTI calls” menu in User > Configuration > Incoming Calls can
be allowed or disallowed (see Section 3.6, “Feature Access”).
If Allow uaCSTA is enabled, applications which support the uaCSTA standard will have access to the OpenStage phone. The default is "Yes".
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Feature Configuration
Administration via Local Phone
|
---
Admin
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System
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Features
|
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Configuration
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General
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Allow uaCSTA
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3.7.12
Local Menu Timeout
The timeout for the local user and admin menu is configurable. When the time interval is over, the menu is closed and the administrator/user is logged out. The timeout may be helpful in case a user does a long press on a line key unintentionally, and thereby invokes the key configuration menu. The menu will close after the timeout, and the key will return to normal line key operation. The timeout ranges from 1 to 5 minutes. The default value is 2.
> The current position in the user or admin menu is kept in case the user/admin has exited the menu, e.g. for receiving a call. Thus, if the user/admin re-enters the menu, he is directed to exactly that submenu, or parameter, which he had been editing before.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Feature Configuration
Administration via Local Phone
|
---
Admin
|
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System
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Features
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Configuration
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General
|
---
Not used timeout
3.7.13
Call Recording
With firmware version V2R2, call recording is possible for OpenStage 15/20/40/60/80 using an
" ASC Voice Recorder". The implementation is similar to a local conference, with the recording device acting as the third conference member. To start recording, the phone calls the recording device and provides it with the mixed audio data. Unlike a true local conference, the call leg used for recording can not transport audio from the recording device to the phone.
With the Call recording mode/Recording Mode parameter, the behaviour of the feature is determined:
• "Disabled": Recording is not possible.
• "Manual": The user starts and stops recording manually using the menu or a free programmable key.
• "Auto-start": The recording starts automatically; besides, the user can stop and start the recording manually.
• "All Calls": The recording starts automatically for all recordable calls; the user can not stop or start the recording manually.
The Audible indication/Audible Notification parameter determines if and how the parties in a call are informed when a call is being recorded:
• "Off": No audible indication is given.
• "Single-shot": A single audible indication is given when recording commences or resumes.
• "Repeated": An audible indication is given when recording commences or resumes, and repeated periodically during the recording.
With the Recorder Address/Recorder number parameter, the SIP address of the call recorder is specified.
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Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
Call Recording
|---
Recorder number
|---
Recorder mode
|
---
Audible notification
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Free Programmable Keys
3.8
Free Programmable Keys
OpenStage 15/40/60/80 phones feature free programmable keys (FPKs) which can be associated with special phone functions. In the Administrator pages of the WBM, the programmable keys menu can be accessed via System > Features > Program keys.
At the phone, the configuration menu for a specific key is called by a long press on the related key. This feature can be disabled by deactivating the FPK program timer. When this parameter is disabled, it is not possible to enter programming mode by long key press. However, the other methods for key programming remain enabled. The functions available and their parameters are described in the following sub-sections. For keyset and DSS functionality, please re-
fer to Section 3.11, “Multiline Appearance/Keyset”.
Administration via WBM
System > Features > Configuration > General
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Administration via Local Phone
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---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|
---
FPK prog. timer
3.8.1
Clear (no feature assigned)
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys
Clear (no feature assigned)
Key.label 3
Submit Reset
3.8.2
Selected Dialing
On key press, a pre-defined call number is called.
The label displayed to the left of the key is defined in Key label <key number>.
The call number defined in the Dial number parameter is dialed on key press.
Administration via WBM
System > Features > Program keys > Selected dialling
Selected dialling
Key.label 4
Dial number
Submit
Selected dialling
Reset
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3.8.3
Repeat Dialing
On key press, the call number that has been dialed lastly is dialed again.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Repeat dialling
Repeat Dialling
Key.label 3
Submit
Repeat Dialling
Reset
3.8.4
Call Forwarding (Standard)
This key function controls phone based call forwarding. If forwarding is enabled, the phone will forward incoming calls to the predefined call number, depending on the current situation.
> To use phone based call forwarding, Server features must be switched off (see
Section 3.7.10, “Server Based Features”).
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
The Forwarding type parameter determines the forwarding behaviour. If "Unconditional" is selected, any incoming call will be forwarded. If "On no reply" is set, the call will be forwarded when the user has not answered within a specified timespan. The timespan is configured in the
WBM user pages under User > Configuration > Incoming calls > Forwarding > No reply delay
(seconds). If "On busy" is selected, incoming calls will be forwarded when the phone is busy.
Administration via WBM
System > Features > Program keys > Forwarding
Forwarding
Key.label 3 Forwarding
Forwarding type Unconditional
Destination
Submit Reset
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3.8.5
Call Forwarding by Call Type
This feature enhances the Call Forwarding (Standard) operation (see Section 3.8.4, “Call Forwarding (Standard)”) by adding support for additional Call Forwarding settings explicitly for Ex-
ternal and Internal calls, as well as the existing capability to forward any call, using functional menus that extend the existing Call Forwarding UI.
> To use extended call forwarding, Server features and Allow uaCSTA must be swit-
ched on (see Section 3.7.10, “Server Based Features”).
> This feature can be enabled or disabled under System > Features > Feature access
> Ext/int forwarding (see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>. It is possible to have an extra key defined for each Call Forwarding Call Type.
Data required
• Forwarding type: Determines forwarding behaviour.
Value range: „CF Unconditional any“,
„CF no reply - any“,
„CF busy - any“,
„CF unconditional - ext.“,
„CF unconditional - int.“,
„CF no reply - ext.“,
„CF no reply - int.“,
„CF busy - ext.“,
„CF busy - int“
Default: „CF Unconditional any“
• Destination: Destination number of call forwarding.
Administration via WBM
System > Features > Program keys > Forwarding
Forwarding
Key.label 3 Forwarding
Forwarding type CF Unconditional any
Destination
Submit Reset
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Administration via WBM (User menu)
User > Configuration > Incoming calls > Forwarding
Forwarding
Forwarding- Unconditional
Forward any call to
Destination
Forward external calls to
Destination
Forward internal calls to
Destination
2153
2102 not set
Forwarding- Busy
Forward any call to
Destination
Forward external calls to
Destination
Forward internal calls to
Destination
Forwarding- No reply
Forward any call to
Destination
Forward external calls to
Destination
Forward internal calls to
Destination
2152
2102
2102
2102
2102
2102
Forwarding Favourites
Submit
Forwarding Favorites
Reset
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3.8.6
Ringer Off
Turns off the ring tone. Incoming calls are indicated via LEDs and display only.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Ringer off
Ringer off
Key.label 3
Submit
Ringer off
Reset
3.8.7
Hold
The call currently selected or active is put on hold.
A held call can be retrieved by pressing the key a second time.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Hold
Hold
Key.label 3
Submit
Hold
Reset
3.8.8
Alternate
Toggles between two calls; the currently active call is put on hold.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Alternate
Alternate
Key.label 3
Submit
Alternate
Reset
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3.8.9
Blind Call Transfer / Move Blind
A call is transferred without consultation, as soon as the phone goes on-hook or the target phone goes off-hook.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Blind transfer
Blind transfer
Key.label 3
Submit
Blind transfer
Reset
3.8.10
Transfer Call
Call transfer, applicable when there is one active call and one call on hold. The active call and the held call are connected to each other, while the phone that has initiated the transfer is disconnected.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Transfer Call
Key.label 3
Submit
Join
Transfer Call
Reset
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3.8.11
Deflect a Call
On key press, an incoming call is deflected to the specified destination.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
The target destination is defined in the Destination parameter.
Administration via WBM
System > Features > Program keys > Deflect
Deflect
Key.label 3 Deflect
Destination 3335
Submit Reset
3.8.12
Shift Level
Shift the level for the programmable keys. When activated, the functions assigned to the shifted level are available on the keys.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Shift
Shift
Key.label 8
Submit
Shift
Reset
3.8.13
Phone-Based Conference
Establishes a three-party conference from an active call and held call.
The label displayed to the left of the key is defined in Key label <key number>.
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Administration via WBM
System > Features > Program keys > Conference
Conference
Key.label 3
Submit
Conference
Reset
3.8.14
Accept Call via Headset (OpenStage 40/60/80)
On key press, an incoming call is accepted via headset.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Headset
Headset
Key.label 3
Submit
Headset
Reset
3.8.15
Do Not Disturb
If this feature is activated, incoming calls will not be indicated to the user.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Do Not Disturb
Do Not Disturb
Key.label 3
Submit
Do Not Disturb
Reset
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3.8.16
Group Pickup
On key press, a call for a different destination within the same pickup group is answered.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Group pickup
Group pickup
Key.label 3
Submit
Group pickup
Reset
3.8.17
Repertory Dial
This feature is similar to the selected dialing function, but additionally, special calling functions are possible. The desired number and/or function is selected via the Dial string parameter.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The following call functions are available:
• "<" disconnect a call.
• "~" start a consultation call. Example: "~3333>"
• ">" (preceded by a call number) start a call. Example: "3333>"
• "-" enter a pause, e. g. for exit-code or international dialing. Example: "0-011511234567>"
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Repertory dial
Repertory dial
Key.label 3 Repertory dial
Use the following characters in the Dial string field
Release
Consult
<
~
Okay
Pause
Dial string
-
>
Submit Reset
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3.8.18
Hunt Group: Send Busy Status
This feature is relevant for hunt groups. If the user is a member of a hunt group and wants another member of the hunt group to pick up an incoming call, he can signal Busy status using the Feature toggle function.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
The Feature code parameter is the OpenScape Voice code for Busy status. In the Descrip-
tion field, an appropriate description for the feature can be entered.
Administration via WBM
System > Features > Program keys > Feature toggle
Feature toggle
Key.label 3
Feature code 0
Feature toggle
Description
Submit Reset
3.8.19
Mobile User Logon
The mobility feature enables users to transfer their personal settings, such as their key layout, or personal phonebook, from one phone to another. The data is stored and managed by the
DLS (Deployment Service).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Mobility
Mobility
Key.label 3
Submit
Mobility
Reset
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3.8.20
Directed Pickup
This feature enables the user to pick up a call which is ringing at another phone. On pressing the key, a menu opens which requests the call number of the target phone.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Directed pickup
Directed pickup
Key.label 3
Submit
Directed pickup
Reset
3.8.21
Callback
When the remote phone called is busy does not reply, the user can send a callback request to the server by pressing this key.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Callback
Callback
Key.label 3
Submit
Callback
Reset
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3.8.22
Cancel Callbacks
With this this function, the user can cancel all callback requests on the server.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Cancel callbacks
Cancel Callbacks
Key.label 3
Submit
Cancel Callbac
Reset
3.8.23
Consultation
When the phone is engaged in an active call, this function opens a dialing menu to make a consulation call.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Consultation
Consultation
Key.label 3
Submit
Consultation
Reset
3.8.24
Call Waiting
Enables or disables the call waiting feature. If enabled, calls from a third party are allowed during an active call.
> The Call Waiting feature cannot be disabled if System > Registration > Server type
(see Section 3.5.7, “SIP Registration””) is set to "HiQ8000".
The label displayed to the left of the key is defined in Key label <key number>.
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Administration via WBM
System > Features > Program keys > Call waiting
Call waiting
Key.label 3
Submit
Call waiting
Reset
3.8.25
Call recording
Starts or stops call recording (for configuring call recording, see Section 3.7.13, “Call Recording”).
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Call Recording
Call Recording
Key.label 3
Submit
Call Recording
Reset
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3.8.26
Auto Answer With Zip Tone
This feature is primarily designed for call centers. If activated, and a headset is used, the phone will automatically accept incoming calls without ringing and without the necessity to press a key. Moreover, additional signalling information from Phone Administration is not required.
To indicate a new call to the user, a zip tone is played through the headset when the call is accepted.
> The feature is available for OpenStage 40/60/80, which provide a headset jack; it only operates if the headset is plugged in. In case the key for feature activation has been pressed before the headset is connected, the feature will be automatically activated when the headset is plugged in.
Administration via WBM
System > Features > Program keys > AICS Zip tone
AICS ZIP
Key.label 1
Submit
AICS ZIP
Reset
3.8.27
Server Feature
Invokes a feature on the SIP server. The status of the feature can be monitored via the LED associated to the key.
> This function is intended primarily for operation with an Asterisk SIP server. For details, please refer to the Administration Manual for OpenStage 15/20/40/60/80 on
Asterisk.
3.8.28
BLF Key
This function offers the possibility to monitor another extension, and to pick up calls for the monitored extension.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
> This function is intended primarily for operation with an Asterisk SIP server. For details, please refer to the Administration Manual for OpenStage 15/20/40/60/80 on
Asterisk.
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3.8.29
Start Application
With this key, the user can start a pre-defined XML application (see Section 3.19, “Applications”). XML applications are available for OpenStage 60/80 phones.
The label displayed to the left of the key is defined in Key label <key number>.
The Application name parameter selects the XML application to be started.
Administration via WBM
System > Features > Program keys
Start Application
Key.label 3
Application name
Submit
App:
Example
Reset
3.8.30
Send Request via HTTP/HTTPS
With this function, the phone can send a specific HTTP or HTTPS request to a server. The function is available at any time, irrespective of registration and call state. Possible uses are HTTPcontrolled features on the system, or functions on a web server that can only be triggered by
HTTP/HTTPS request, e. g. login/logout for flexible working hours.
The Protocol parameter defines whether HTTP or HTTPS is to be used for sending the URL to the server.
The Web server address is the IP address or DNS name of the remote server to which the
URL is to be sent.
The Port is the target port at the server to which the URL is to be sent.
The Path is the server-side path to the desired function, i. e. the part of the URL that follows the IP address or DNS name. Example: webpage/checkin.html
In the Parameters field, one or more key/value pairs in the format "<key>=<value>" can be added to the request, separated by an ampersand (&).
Example: phonenumber=3338&action=huntGroupLogon
> The question mark will be automatically added between the path and the parameters. If a question mark has been entered at the start of the parameters, it will be stripped off automatically.
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The Method parameter determines the HTTP method to be used, which can either be GET or
POST. If GET is selected, the additional parameters (Parameters) and the user id/password
(Web server user ID/Web server password) are part of the URL. If POST is selected, these data form the body of the message.
In case the web server requires user authentication, the parameters Web server user ID and
Web server password can be used. If not null, the values are appended between the serverside path (Path) and the additional parameters (Parameter).
If the LED controller URI is given, the LED associated with this key indicates the state of the call number or SIP URI specified, provided the SIP server sends a notification:
• Busy notification: LED is glowing.
• Ringing notification: LED is blinking.
• Idle notification (state=terminated): LED is dark.
> When assigning the function described here to the release key s , please consider that this key has no LED.
With firmware version V2R2, the Push support parameter is available. If activated, the LED is controllable by a combination of an HTTP push request and an XML document. For further information, see the XML Applications Developer’s Guide.
> If you want to use the HTTP push solution, please ensure that the LED controller
URI field is empty. Otherwise, the phone will only use the SIP mechanism for LED control, and ignore the push request.
The Symbolic name is used to assign a push request from the application server to the appropriate free programmable key resp. fixed function key. This value must be unique for all keys involved.
Data required
• Key label <n>: Label for the key.
• Protocol: Transfer protocol to be used.
Value range: "HTTP", "HTTPS"
• Web server address: IP address or DNS name of the remote server.
• Port: Target port at the server.
• Path: Server-side path to the function.
• Parameters: Optional parameters to be sent to the server.
• Method: HTTP method used for transfer.
Value range: "GET", "POST"
• Web server user ID: User id for user authentication at the server.
• Web server password: Password for user authentication at the server.
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• LED controller URI: Indicates the state of the call number specified.
• Push support : Enables or disables LED control by push requests from the server.
• Symbolic name : Assigns a push request to the appropriate free programmable key resp. fixed function key.
Administration via WBM
System > Features > Program keys > Send URL
Send URL
Key label 2
Message details
Send URL
Protocol
Web server address
Port
HTTPS
Path
Parameters
(key1=value1&key2=value2)
Method GET
Authenticate phone
Web server user ID
Web server password
SIP response handling
LED controller URI
Push support
Push support
Symbolic name
Submit Reset
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3.8.31
Built-in Forwarding
As a programmable key function, this is relevant for OpenStage 15 phones, which have no fixed forwarding key.
System > Features > Program keys
Built in fwd
Key.label 3
Submit
Call forward
Reset
3.8.32
2nd Alert
This function allows for monitoring and accepting a second incoming call. When a call is ringing while the user is dialing, the LED will light up. As soon as the user presses the key, information about the incoming call is presented, and the user can accept the call. If a call is ringing, and another call starts ringing shortly after, the LED will light up, and the user has the possibility to toggle between these calls via key press.
System > Features > Program keys
2nd alert
Key.label 2
Submit
2nd alert
Reset
3.8.33
Start Phonebook (OpenStage 40/15 only)
These key functions opens a menu which enables the user to start the personal or the corporate phonebook. For further information about the personal and corporate phonebook, please refer to the user guide for OpenStage 40/15 phones. For more information about the corporate
phonebook, please see Section 3.17, “Corporate Phonebook: Directory Settings”.
Administration via WBM
System > Features > Program keys > Personal directory
Personal directory
Key.label 5
Submit
Personal
Reset
System > Features > Program keys > Corporate directory
Corporate directory
Key.label 5
Submit
Corporate
Reset
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3.8.34
Show phone screen (OpenStage 15 and OpenStage 40 only)
On pressing this key, the phone display switches to call view mode.
Administration via WBM
System > Features > Program keys > Show phone screen
Show phone screen
Key.label3
Submit
Show phone screen
Reset
3.8.35
Mute (OpenStage 15 only)
On pressing this key, the microphone is turned off. This programmable key function is available only for OpenStage 15 phones, which have no fixed mute key.
Administration via WBM
System > Features > Program keys > Mute
Key.label 3
Submit
Mute
Mute
Reset
3.8.36
Release (OpenStage 15 only)
On pressing this key, the current call is disconnected. This programmable key function is available only for OpenStage 15 phones, which have no fixed release key.
Administration via WBM
System > Features > Program keys > Release
Cancel/Release
Key.label 4
Submit
Cancel/Release
Reset
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3.9
Preset Function Keys (OpenStage 40 US only)
The OpenStage 40 US telephone comes with six programmable lit sensor keys, preset to the following factory settings:
• Shift
• Phonebook
• Group pickup
• Call Forward
• DND
• Show phone
All of this can be programmed on two separate levels but if you reset the phone, the keys in the first level will be reset to the default factory settings.
3.10
Fixed Function Keys
For
• OpenStage 60/80
– the forwarding key r , the release key s , and the voice recognition key q ,
• OpenStage 20 and OpenStage 40
– the release key s , the redial key , and the forwarding key r specific SIP or HTTP based functions can be defined. These functions can be employed as an alternative to the built-in functions.
3.10.1
Fixed Function Keys on OpenStage 40 US
For the Conference key, the Transfer key, and the Hold key, specific SIP or HTTP based functions can be defined. If you reset the phone, these three keys will be reset to the default factory settings.
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3.11
Multiline Appearance/Keyset
> This feature is available only on OpenStage 15, OpenStage 40 and OpenStage 60/
80 phones.
A phone that has more than one line associated to it, and therefore works as a multiline phone, is referred to as "keyset". The lines are assigned to the phone by setting up a separate line key for each line.
The multiline appearance feature allows for multiple lines to be assigned to a keyset and for a line to be assigned to multiple keysets. This feature requires configuration in Phone Administration and in the telephone, and is particularly useful for executive-assistant arrangements.
> In order to configure the phone as a keyset, it is required to
• use an outbound proxy (System > SIP interface > Outbound proxy, see
Section 3.5.8.1, “Outbound Proxy”), and
• set the server type to "OS Voice" (System > Registration > Server type, see
Section 3.5.7, “SIP Registration”).
For each keyset, a Primary Line/Main DN is required. The primary line is the dialing number for that keyset.
There are two types of line:
• Private line: A line with restricted line status signaling towards OSV.
• Shared line: A line that is shared between keysets.
3.11.1
Line key configuration
System > Features > Program keys
> It is recommended to configure primary lines only on keys 1 to 6, or 1 to 5, if a shift key is needed. This ensures that the lines are still accessible when the user migrates to a different phone with fewer keys via the mobility feature.
A line corresponds to a SIP address of record (AoR), which can have a form similar to an Email address, or can be a phone number. It is defined by the Address parameter. For registration of the line, a corresponding entry must exist on the SIP server resp. the SIP registrar server.
A label can be assigned to the line key by setting its Key label.
Every keyset must necessarily have a line key for the primary line. To configure the key of the primary line, set Primary line to "true".
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If Ring on/off is checked, the line will ring when an incoming call occurs, and a popup will appear on the display. If the option is not checked, the incoming call will be indicated only by the blinking of the key’s LED. If it is desired that the line ring with a delay, the time interval in seconds can be configured by Ring delay.
When the user lifts the handset in order to initiate a call, the line to be used is determined by selection rules. To each line, a priority is assigned by the Selection order parameter. A line with the rank 1 is the first line to be considered for use. If more than one line have the same rank, the selection is made according to the key number. Note that Selection order is a mandatory setting; it is also relevant to the Terminating line preference, as well as to other functions.
The Address (Address of Record) parameter is the phone number resp. SIP name corresponding to the entry in the SIP registrar at which the line is to be registered.
> For the configuration of line keys, the use of the DLS (Deployment Service) is recommended. For operating the DLS, please refer to the DLS user’s guide. Alternatively, the web interface or the local menu can be used. Note that the creation of a new line key and the configuration of some parameters can not be accomplished by the phone’s local menu.
Generally, it is advisable to restrict the user’s possibilities to modify line keys. This can be achieved solely by the DLS. For further instructions, see the DLS Administration Guide.
The Realm, a protection domain used for authenticated access to the SIP server, works as a name space. Any combination of user id and password is meaningful only within the realm it is assigned to. The other parameters necessary for authenticated access are User Identifier and
Password. For all three parameters, there must be corresponding entries on the SIP server.
The Shared type parameter determines whether the line is a shared line, i. e. shared with other endpoints, or a private line, i .e. available exclusively for this endpoint. A line that is configured as primary line on one phone can be configured as secondary line on other phones.
> Shared lines are not available if System > Registration > Server type (see Section
3.5.6, “SIP Registration”) is set to "HiQ8000".
When Allow in overview is set to "Yes", the line will be visible in the line overview on the phone’s display.
> Line overwiew can be enabled or disabled under System > Features > Feature ac-
cess (see Section 3.6, “Feature Access”)
If a line is configured as hot line, the number indicated in Hot warm destination is dialed immediately when the user goes off-hook. This number is configurerd in the user menu under
Configuration > Keyset > Lines > Hot/warm destination. To create a hot line, Hot warm ac-
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tion must be set to "hot line". If set to "Warm phone", the specified destination number is dialed
after a delay which is defined in Initial digit timer (seconds) (for details, see Section 3.7.3,
“Initial Digit Timer”). During the delay period, it is possible for the user to dial a different number
which will be used instead of the hot/warm line destination. In addition, the user will be provided with a dial tone during the delay period. With the setting "No action", the line key will not have hot line or warm line functionality.
System > Features > Program keys
Data required
• Key label <n>: Set the label of the line key with the key number <n>.
Default: "Line"
• Primary line: Determines whether the line is the primary line.
Value range: "Yes", "No"
Default: "No"
• Ring on/off: Determines whether the line rings on an incoming call.
Value range: "On", "Off"
Default: "On"
• Ring delay (seconds): Time interval in seconds after which the line starts ringing on an incoming call.
Default: 0
• Selection order: Priority assigned to the line for the selection of an outgoing line.
Default: 0
• Address: Address/phone number which has a corresponding entry on the SIP server/ registrar.
• Realm: Domain wherein user id and password are valid.
• User Identifier: User name for authentication with the SIP server.
• Password: Password for authentication with the SIP server.
• Shared type: Determines whether the line is a shared line (shared by multiple endpoints) or a private line (only available for this endpoint).
Value range: "shared", "private", "unknown".
Default: "shared"
• Allow in Overview: Determines whether the line appears in the phone’s line overview.
Value range: "Yes", "No"
Default: "Yes"
• Hot warm action : Determines if the line is a regular line, a hot line, or a warm line.
Value range: "No action", "hot line", "warm line"
• Hot warm destination : The destination to be dialed from the hot/warm line when the user goes off-hook.
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> A new line key can only be added by use of the WBM or, preferably, the DLS. Once a line key exists, it can also be configured by the local menu.
Administration via WBM
1. Invoke the "Phone keys" dialog and select "line" in the pulldown menu of the key you want to configure. Next, click the edit button.
System > Features > Program keys
Program keys
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Line
Label: Primary Line
Selected dialling
Label: Selected dialling
Hold
Label: Hold edit edit edit
1
2
3
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit
4
5
6
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Mobility
Label: Mobility
Clear (no feature assigned) edit edit
7
8
Clear (no feature assigned)
Clear (no feature assigned)
Shift
Label: Shift edit
9 Clear (no feature assigned) edit edit edit edit edit edit edit edit
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2. In the "Line" dialog, set the specific parameters for the line key.
Line
It is recommended that primary lines are only configured on keys 1 to 6.
This ensures compatibility with athe mobility feature, when using devices with 6 or fewer programmable feature keys.
Key label 1
Primary line
Ring on/off
Ring delay (seconds)
Selection order
Address
Realm
User Identifier
Password
Shared type
Allow Overview
Hot warm action
Hot warm destination
Submit
0
0 shared
No Action
Reset
3. (Only relevant if hot line / warm line is to be configured:) The destination for hot line or warm line is set in User menu > Configuration > Keyset > Lines:
Lines
Line
Key label 1
Allow in overview
Address
Primary line
Ring on/off
0
Selection order
Hot/warm line Hot line
Hot/warm destination 3333
;
3337
1
Submit Reset
In the local menu, the menu path is the same.
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Administration via Local Phone
The configuration of a line via Local phone is only possible when the line key has been created via Web interface or DLS before.
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
Keyset lines
|
---
Details For Keyset Line <xx>
|---
Address
|---
Ring on/off
|
|---
Selection order
---
Hot/warm action
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3.11.2
Configure Keyset Operation
The following parameters provide general settings which are common for all keyset lines.
The Rollover ring setting will be used when, during an active call, an incoming call arrives on a different line. If "no ring" is selected, the incoming call will not initiate a ring. If "alert ring" is selected, a 3 seconds burst of the configured ring tone is activated on an incoming call; "alert beep" selects a beep instead of a ring tone. "Standard ring tone" selects the default ringer.
LED on registration determines whether the line LEDs will be lit for a few seconds if they have been registered successfully with the SIP server on phone startup.
The Originating line preference parameter determines which line will be used when the user goes off-hook or starts on-hook dialing.
> When a terminating call exists, the terminating line preference takes priority over originating line preference.
The following preferences can be configured:
• "idle line": An idle line is selected. The selection is based on the Hunt ranking
parameter assigned to each line (see Section 3.11.1, “Line key configuration”).
• "primary": The designated Primary Line/Main DN is always selected for originating calls.
• "last": The line selected for originating calls is the line that has been used for the last call (originating or terminating).
• "none": The user manually selects a line by pressing its line key before going off-hook or by pressing the speaker key, to originate a call.
Manual line selection overrides automatic line preferences.
The Terminating line preference parameter decides which terminating line, i. e. line with an incoming call, is selected when the user goes off-hook.
The following preferences can be configured:
• "ringing line": The line in the alerting or audible ringing state is automatically selected when the user goes off-hook. In the case of multiple lines alerting or ringing, the lines are selected on the one that has been alerting the longest.
• "ringing PLP": The line in the alerting or audible ringing state is automatically selected when the user goes off-hook. However, if the prime line is alerting, it is given priority.
• "incoming": The earliest line to start audible ringing is selected, or else the earliest alerting (ringing suppression ignored) line is selected.
• "incoming PLP": The earliest line to start audible ringing is selected, or else the earliest alerting (ringing suppression ignored) line is selected. However, if the prime line is alerting, it is given priority.
• "none": To answer a call, the user manually selects a line by pressing its line key before going off-hook, or by pressing the speaker key.
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Manual line selection overrides automatic line preferences.
Line action mode determines the consequence for an established connection when the line key is pressed. If "hold" is selected, the call currently active is set to hold as soon as the line key is activated. The user has two options: 1) to reconnect to the remote phone by pressing the line key that corresponds to that call, or 2) to initiate another call from the newly selected line.
If "release" is selected, the previously established call is ended.
If Show focus is checked, the LED of a line key flutters when the line is in use. If it is not checked, the line key is lit steady when it is in use.
The Reservation timer sets the period after which the reservation of a line is canceled. A line is automatically reserved for the keyset whenever the user has selected a line for an outgoing call and hears a dial tone. The reservation of a line is accomplished by the Phone Administration server, which notifies all the endpoints sharing this line. If set to 0, the reservation timer is deactivated.
Forward indication activates or deactivates the indication of station forwarding, i. e. the forwarding function of Phone Administration. If Forward indication is activated and station forwarding is active for the corresponding line, the LED of the line key blinks.
Preselect mode determines the phone’s behaviour when a call is active, and another call is ringing. If the parameter is set to "Single button", the user can accept the call a single press on the line key. If it is set to "Preselection", the user must first press the line key to select it and then press it a second time to accept the call. In both cases, the options for a ringing call are presented to the user: "Accept", "Reject", "Deflect".
Preselect timer is relevant if Preselect mode is set to "Preselection". The parameter sets the timeout in seconds for the second key press that is required to accept the call. After the timeout has expired, the call is no longer available.
When Bridging enabled (Admin > Features > Configuration) is activated, the user may join into an existing call on a shared line by pressing the corresponding line key. On key press, the
Phone Administration builds a server based conference from the existing call parties and the user. If the call has already been in a server based conference, the user is added to this conference.
> When bridging shall be used, it is highly recommended to configure the phone for a
system based conference (see Section 3.7.9, “System Based Conference”). This
enables adding more users to a system based conference that has been initiated by bridging.
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Data required
• Rollover ring: Determines if a ring tone will signal an incoming call while a call is active.
Value range: "Standard ring", "No ring", "Alert beep", "Alert ring"
Default: "Alert beep"
• LED on registration: Determines if line LEDs will signal SIP registration.
Value range: "Yes", "No"
Default: "Yes"
• Originating line preference: Selects the line to be used for outgoing calls.
Value range: "Idle line", "Primary", "Last", "None"
Default: "Idle line"
• Terminating line preference: Determines which line with an incoming call shall be selected for answering.
Value range: "Ringing line", "Incoming", "Incoming PLP", "Ringing PLP", "None"
Default: "Idle line"
• Line action mode: Determines the consequence for an established connection when the line key is pressed.
Value range: "Hold", "Release"
Default: "Hold"
• Show focus: Determines whether the line key LED blinks or is steady when it is in use.
Value range: "Yes", "No"
Default: "Yes"
• Reservation timer: Sets the period in seconds after which a line reservation is cancelled.
If set to 0, the reservation timer is deactivated.
Default: 60
• Forward indication: Activates or deactivates the indication of station forwarding.
Value range: "Yes", "No"
Default: "No"
• Preselect mode: Determines whether an incoming call is accepted by a single press on the corresponding line key or a double press is needed.
Value range: "Single button", "Preselection"
Default: "Single button"
• Preselect timer: Sets the timeout in seconds for accepting an incoming call.
• Bridging enabled (see Admin > Features > Configuration) : When set to "Yes", the user is allowed to join a call on a shared line. For this purpose, a server based conference is established.
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System > Features > Keyset Operation
Keyset operation
Rollover ring
LED on registration
Originating line preference
Terminating line preference
Line action mode
Show focus
Reservation timer (seconds)
Forwarding indicated
Preselect mode
Preselect timer
Preview mode
Preview timer
Bridging priority
Submit alert beep
; idle line ringing line hold
60
; single button
8
Preview overwrites brid-
Reset
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System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset administration.fm
Administration
Multiline Appearance/Keyset
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Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Keyset operation
|---
Rollover ring
|---
LED on registration
|---
Originating line preference
|---
Terminating line preference
|---
Line action mode
|---
Show focus
|---
Reservation timer
|---
Forward indicated
|
|---
Preselect mode
---
Preselect timer
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
Configuration
---
General
|
---
Bridging enabled
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3.11.3
Line Preview
This key enables the preview mode, which allows the user to preview a line before using it.
When preview mode is active, the line keys behave similar to when the keyset configuration is
set to preselection for line keys (see Section 3.11.2, “Configure Keyset Operation”). On
pressing the line key (not DSS key!), the call activity on the corresponding line is shown. Unlike with a preselected line, there will be no change to the phone’s current line connections. The
LED indicates whether line preview is active or not.
The information shown to the user depends on the ring/alert configuration for the line in question. If the line is configured to alert only, the preview will only show the state of the call, not the identity of the call party. If the line is configured to ring, the identity of the call party will be revealed.
The preview mode can be configured as temporary or as permanent. If System > Features >
Keyset operation > Preview mode is disabled, preview mode will end when the user uses the previewed line, or a new call is started in any other way, or if the focus is changed away from call view. If the parameter is enabled, preview mode remains active until the user cancels it by pressing the key again.
The Preview timer parameter determines the timespan during which the line preview will remain on the screen.
The Bridging priority parameter affects the behavior of the line key (see Section 3.11.3.1,
“Preview and Preselection”). Precondition: Bridging is enabled (see Section 3.11.2, “Configure Keyset Operation”)
Data required
• Preview mode
Value range: "Yes", "No"
Default: "No"
• Prieview timer: When Prieview Mode is set, the timer controls preview.
Value range: 2, 3, 4, 6, 8, 10, 15, 20, 30, 40, 50, 60
Default: 8
• Bridging priority
Value range: "Prieview overrides bridging", "Bridging overrides preview"
Default: "Prieview overrides bridging"
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System > Features > Program keys > Preview
Preview
Key.label 5
Submit
Preview
Reset
3.11.3.1
Preview and Preselection
Precondition: Bridging is enabled
Action Preselect mode
Single button
Preselection
Preview mode
On (Lock
Prev.)
Off
(Temp
Prev.)
Bridging priority
Preview overrides bridging
Preview key is not pressed (LED off), only the Line key is pressed once or twice
Press busy second. line key 1x
9 n.rel.
n.rel.
n.rel.
Press busy second. line key 2x while line-view is displayed
9 n.rel.
n.rel.
n.rel.
Bridging overrides preview n.rel.
n.rel.
Press busy second. line key 1x
9 n.rel.
n.rel.
n.rel.
Preview key is pressed first (LED on) and the Line key is pressed once or twice
Press busy second. line key 1x n.rel.
n.rel.
9 n.rel.
9
Press busy second. line key 1x
Press busy second. line key 1x
Press busy second. line key 1x n.rel.
n.rel.
n.rel.
Press busy second. line key 2x while line-view is displayed n.rel.
Press busy second. line key 2x while line-view is displayed n.rel.
n.rel.
n.rel.
n.rel.
n.rel.
n.rel.
-
-
9
9
9
-
-
-
9
9
-
9
9
9
9
-
-
-
-
9
Result
Line status is displayed
(Preselect timer)
1 st
press: line status
2 nd
press: bridge (conference)
Bridge (conference)
Bridge (conference)
Preview LED -> off
Bridge (conference)
Preview LED remains on
Line status is displayed
(Preview timer)
Preview LED -> off
Line status is displayed
(Preview timer)
Preview LED remains on
1 st
press: line status
2 nd
press: bridge (conference)
Preview LED -> off
1 st
press: line status
2 nd
press: bridge (conference)
Preview LED remains on
In case the Preview key is not pressed, only the Preselection mode configuration is relevant:
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• if Preselection is selected then the line status is displayed after 1st line press
• if Single button is selected then bridging is invoked (if line busy, bridging enabled)
In case the Preview key is pressed first and then the line key, the Preselection mode configuration is not relevant.
• if Preview mode is activated then the Preview key LED remains on (preview mode must be deactivated manually by pressing the Preview key again).
• if Preview mode is deactivated then the Preview key LED is turned off after preview timer expiry
– if Bridging priority= Bridging overrides preview then pressing the busy line key invokes bridging
– if Bridging priority= Preview overrides bridging then pressing the line key displays the line status, however pressing the line key twice bridging will be invoked.
3.11.4
Immediate Ring
Enables or disables the preset delay for all line keys. This feature only applies to keyset lines.
The label displayed to the left of the key is defined in Key label <key number>.
Administration via WBM
System > Features > Program keys > Immediate ring
Immediate ring
Key.label 3
Submit
Immediate ring
Reset
3.11.5
Direct Station Select (DSS)
> This feature is available only on OpenStage 15/40/60/80, and requires OpenScape
Voice.
> This feature can be enabled or disabled under System > Features > Feature access
(see Section 3.6, “Feature Access”).
A DSS key is a special variant of a line key. It enables a direct connection to a target phone, allowing the user to pick up or forward a call alerting the DSS target and make/complete a call to the DSS target.
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3.11.5.1
General DSS Settings
These parameters define the behaviour of all DSS keys.
> Generally, it is advisable to restrict the user’s possibilities to modify line keys, including DSS keys. This can be achieved solely by the DLS. For further instructions, see the DLS Administration Guide.
If the user picks up an incoming call for the DSS target by pressing the associated DSS key, the call is forwarded to the user’s primary line. Thereafter, the user’s phone rings, and the user can accept the call.
> To enable the immediate answering of a call via the DSS key, Allow auto-answer in the user menu must be activated. The complete path on the WBM is:
User Pages > Configuration > Incoming calls > CTI calls > Allow auto-answer.
The value of Call pickup detect timer (seconds) determines the time interval in which the deflected call is expected at the primary line. When the call arrives whithin this interval, it is given special priority and handling. If a second call arrives on the primary line during this interval, it will be rejected. If a second call arrives outside the interval, it will be treated just like any other incoming call. The default is 3.
If Deflecting call enabled is checked, the user can forward an alerting call to the DSS target by pressing the DSS key. The default is "No".
>
This parameter is configured under System > Features > Feature access (see Section 3.6, “Feature Access”).
If Allow pickup to be refused is checked, the user is enabled to reject a call alerting on the line associated with the DSS key. The default is "No".
>
This parameter is configured under System > Features > Feature access (see Section 3.6, “Feature Access”).
The DSS key can be configured to indicate the call forwarding state of the number represented by the DSS key. This feature is activated when Forwarding shown is enabled.
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System > Features > DSS Settings
DSS settings
Call pickup detect timer (seconds)
Deflect alerting call enabled
Allow pickup to be refused
Forwarding shown
Submit
3
Reset
Administration via Local Phone
|
--Admin
|
--System
|
--Features
|
--DSS operation
|--Deflect to DSS
|--Refuse DSS pickup
|
--Forwarding shown
|
--Admin
|
--System
|
--Features
|
--Configuration
|
--General
|
--DSS Pickup timer
3.11.5.2
Settings for a DSS key
The Key label <n> parameter provides the DSS key with a label that is displayed on the graphic display on an OpenStage 40/60/80 phone. The label is also user configurable.
Address contains the call number of the line associated with the DSS key.
The Realm parameter stores the SIP Realm of the line associated with the DSS key.
User Identifier gives the SIP user ID of the line associated with the DSS key.
Password provides the password corresponding to the SIP user ID.
The Outgoing calls parameter determines the behaviour of a call over the DSS line at the target phone. If set to "Direct", any forwarding and Do not Disturb settings on the target phone will be overridden, so that a call will always alert. If set to Line type is set to "Normal", this is not the case, and the call will be treated like a regular call.
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Action on calls defines the handling of an active call when pressing the DSS key. If set to
"Consult", the user has an option to start a consultation with the DSS target. If set to "Transfer", the user can only transfer the call to the DSS target. If "No action" is selected, pressing the DSS key will have no effect.
When Allow in Overview is set to "Yes", the line corresponding to the DSS key will be visible in the line overview on the phone’s display.
Data required
• Key label <key number>: Label to be displayed on the display.
Default: "DSS"
• Address: SIP Address of Record of the destination that is assigned to the DSS key.
• Realm: SIP Realm of the DSS destination.
• User ID: SIP user ID of the DSS destination.
• Password: Password corresponding to the SIP user ID.
• Outgoing calls: Determines whether forwarding and DND at the target phone will be overridden on a DSS call.
Value range: "Normal", "Direct"
Default: "Normal"
• Action on calls: Handling of an active call when pressing the DSS key. "Consult": the user can start a consultation with the DSS target; "Transfer": the user can transfer the call to the
DSS target.
Value range: "Consult", "Transfer", "No action"
Default: "Consult"
• Allow in Overview: Determines whether the line appears in the phone’s line overview.
Value range: "Yes", "No"
Default: "Yes"
Administration via WBM
System > Features > Program keys > [edit]
DSS
Key label 2
Address
Realm
User Identifier
DSS
Password
Outgoing Calls Normal_
Action on Calls Consult
Allow in overview
Submit
;
Reset
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3.11.6
Distinctive Ringers per Keyset Lines
For implicit mapping of line ringer names following format is to be used:
"Line-<DN of line>-Reserved"
Thus for a line with DN=1234 the mapped distinctive ringer name is "Line-1234-Reserved".
(The name is case-sensitive, mind the uppercase L and R in name.)
The name needs to be manually constructed and configured by Admin as a new ringer name and each such name should be manually checked as being unique in the table.
> When using ’Distinctive Ringers per Keyset Lines’, it is not allowed to define
’bellcore_dr1’, ’bellcore_dr2’, and ’bellcore_dr3’ in the same distinctive ringer table.
Otherwise these settings will be used because of higher priority in SIP-INVITE header. MLPP and Lowel Impact Level calls are also with higher priority.
The "User>Configuration>Keyset>Lines" form has the ’Destination Number’ of the line being configured and this can be used to map directly to distinctive ringer names in the "Admin>Ringer setting" form. If a distinctive ringer with a matching name has not been configured into the table then the Ringer related items Ringer, Ringer tone melody, and Ringer sequence in the
"User>Configuration>Keyset>Lines" form will be absent. If a matching distinctive ringer name is found then the "Ringer" items are editable with the initially shown value being the same as the value in the "Admin>Ringer setting" form. Changes made to the "Ringer" values by the User will also change the matching distinctive ringer values in "Admin>Ringer setting".
Destinctive Ringers are not applicable for DSS Keys.
Data required
• Name: Destinctive ringer name .
Value Range: "Line-<Destination Number of line>-Reserved"
• Ringer sound: Specifies whether pattern, i. e. melody, or a specific sound file is used as ringer.
Default: ’Pattern’
• Pattern melody: Determines the melody pattern if Ringer sound is set to ’Pattern’.
Value Range: 1,...,8
• Pattern sequence: Determines the length and repetitions of pattern.
Value Range: "1":1 sec ON, 4 sec OFF,
"2": 1 sec ON, 2 sec OFF
"3": 0.7 sec ON, 0.7 sec OFF, 0.7 sec ON, 3 sec OFF
Default: "1"
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Administration via WBM
Admin > Ringer setting > Distinctive
Distinctive
This page allows you to set up interworking with other IP phone systems that support distinctive ringing
Name Ringer sound alert-internal alert-externa alert-recall alert-emerge
Line-3336-R
Line-3335-R
Ringer1.wav
Ringer2.mp4
Ringer3.mp3
Ringer5.mp3
Ringer2.mp3
Ringer4.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Submit
Pattern melody
3
3
3
3
3
3
3
3
8
4
3
3
8
3
3
2
2
2
2
2
2
2
2
1
2
2
2
2
2
2
Pattern sequence
60
60
60
60
60
60
60
60
0
60
60
60
60
60
60
Reset
Duration (sec) Audible
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
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|
---
Admin
|
---
Ringer setting
|
---
Distinctive
|
---
<1 .... 15>
|---
Name
|---
Ringer sound (= Ringer in UserMenu)
|---
Pattern melody (= Ringer melody in UserMenu)
|---
Pattern sequence (= Ringer tone sequence in User Menu)
|
|---
Duration
---
Audible
User menu > Configuration > Keyset > Lines
Lines
Mainline
0
;
3336
Ring delay (seconds)
Allow in overview
Address
Primary line
Ring on/off
Ringer melody
Ringer tone sequence
Ringer
Selection order
Hot line/Warm line
Hot/Warm destination
Submit
8
1.0 seconds ON, 2.0 seconds OFF
Ringer1.wav
1
Hot line
3333
Reset
Administration via Local Phone
The configuration of a line via Local phone is only possible when the line key has been created via WBM or DLS before by administrator.
|
--User Menu
|
--Configuration
|
--Keyset
|
--Lines <xx>
|--Ringer file(= Ringer sound in Admin Menu)
|--Ringer melody (= Pattern melody in Admin Menu)
|
--Ringer sequence (= Pattern sequence in Admin Menu)
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Key Modules
3.12
Key Modules
A key module provides 18 (OS 15) or 12 (OS 40/60/80) additional free programmable keys. Key modules are available for OpenStage 15/40/60/80 phones. The key module for the OpenStage
15 phone provides 18 programmable keys. A maximum of 2 key modules can be connected to one phone.
The following table shows which key modules can be connected to the particular phone types.
Phone Type
OpenStage 15
OpenStage 40
OpenStage 60/80 -
1
OpenStage Key Module 15 OpenStage Key Module
1 -
2
2
> Please note that OpenStage Key Modules (self-labeling) and
OpenStage Key Module 15 (paper label) can not be combined. For key labeling, a special tool is available; please refer to: http://wiki.siemens-enterprise.com/index.php/Key_Labelling_Tool .
The configuration of a key on the key module is exactly the same as the configuration of a phone key.
Administration via WBM
System > Features > Key module 1/2
Key Module 1
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit
3
4
1
2
5
6
7
8
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
9
10
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
11
12
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit edit edit edit edit
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Key Module 2
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit
3
4
1
2
5
6
7
8
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
9
10
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
11
12
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit edit edit edit edit administration.fm
Administration
Key Modules
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Administration
Dialing
3.13
Dialing
3.13.1
Canonical Dialing Configuration
Call numbers taken from a directory application, LDAP for instance, are mostly expressed in canonical format. Moreover, call numbers entered into the local phonebook are automatically converted and stored in canonical format, thereby adding "+", Local country code, Local na-
tional code, and Local enterprise number as prefixes. If, for instance, the user enters the extension "1234", the local country code is "49", the local national code is "89", and the local enterprise number is "722", the resulting number in canonical format is "+49897221234".
For generating an appropiate dial string, a conversion from canonical format to a different format may be required. The following parameters determine the local settings of the phone, like
Local country code or Local national code, and define rules for converting from canonical format to the format required by the PBX.
> To enable the number conversion, all parameters not marked as optional must be
provided, and the canonical dial lookup settings must be configured (see Section
3.13.2, “Canonical Dial Lookup”).
Data required
• Local country code: E.164 Country code, e.g. "49" for Germany, "44" for United Kingdom.
Maximum length: 5
• National prefix digit: Prefix for national connections, e.g. "0" in Germany and United Kingdom.
Maximum length: 5
• Local national code: Local area code or city code, e.g. "89" for Munich, "20" for London.
Maximum length: 6
• Minimal local number length: Minimum number of digits in a local PSTN number, e.g.
3335333 = 7 digits.
• Local enterprise number: Number of the company/PBX wherein the phone is residing.
Maximum length: 10 (Optional)
• PSTN access code: Access code used for dialing out from a PBX to a PSTN. Maximum length: 10 (Optional)
• International access code: International prefix used to dial to another country, e.g. "00" in Germany and United Kingdom. Maximum length: 5
• Operator codes: List of extension numbers for a connection to the operator. The numbers entered here are not converted to canonical format. Maximum length: 50 (Optional)
• Emergency number: List of emergency numbers to be used for the phone. If there are more than one numbers, they must be separated by commas. The numbers entered here are not converted to canonical format. Maximum length: 50 (Optional)
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These emergency numbers can also be dialed when the phone is locked, in line with the
emergency number configured in Features > Configuration (see Section 3.5.2, “Emergency and Voice Mail”).
• Initial extension digits / Initial digits: List of initial digits of all possible extensions in the local enterprise network. When a call number could not be matched as a public network number, the phone checks if it is part of the local enterprise network. This is done by comparing the first digit of the call number to the value(s) given here. If it matches, the call number is recognized as a local enterprise number and processed accordingly.
If, for instance, the extensions 3000-5999 are configured in Phone Administration, each number will start with 3, 4, or 5. Therefore, the digits to be entered are 3, 4, 5.
• Internal numbers
> To enable the phone to discern internal numbers from external numbers, it is
crucial that a canonical lookup table is provided (Section 3.13.2, “Canonical
Dial Lookup”).
• "Local enterprise form": Default value. Any extension number is dialled in its simplest form. For an extension on the local enterprise node, the node ID is omitted. If the extension is on a different enterprise node, then the appropriate node ID is prefixed to the extension number. Numbers that do not correspond to an enterprise node extension are treated as external numbers.
• "Always add node": Numbers that correspond to an enterprise node extension are always prefixed with the node ID, even those on the local node. Numbers that do not correspond to an enterprise node extension are treated as external numbers.
• "Use external numbers": All numbers are dialled using the external number form.
• External numbers
• "Local public form": Default value. All external numbers are dialled in their simplest form. Thus a number in the local public network region does not have the region code prefix. Numbers in the same country but not in the local region are dialled as national numbers. Numbers for a different country are dialled using the international format.
• "National public form": All numbers within the current country are dialled as national numbers, thus even local numbers will have a region code prefix (as dialling from a mobile). Numbers for a different country are dialled using the international format.
• "International form": All numbers are dialled using their full international number format.
• External access code
• "Not required": The access code to allow a public network number to be dialled is not required.
• "For external numbers": Default value. All public network numbers will be prefixed with the access code that allows a number a call to be routed outside the enterprise network. However, international numbers that use the + prefix will not be given access code.
• International gateway code:
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• "Use national code": Default value. All international formatted numbers will be dialled explicitly by using the access code for the international gateway to replace the "+" prefix.
• "Leave as +": All international formatted numbers will be prefixed with "+".
Administration via WBM
Local functions > Locality > Canonical dial settings
Submit
Canonical dial settings
Local country code
National prefix digit
Local national code
Minimum local number length
Local enterprise node
PSTN access code
International access code
Operator codes
Emergency numbers
Initial extension digits
49
0
89
4
723
0
00
1,2,3,4
Reset
Local functions > Locality > Canonical dial
Canonical dial
Internal numbers
External numbers
External access code
International gateway code
Submit
Local enterprise form
Local public form
Not required
Use national code
Reset
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|
---
Admin
|
---
Local Functions
|
---
Locality
|
---
Canonical dial settings
|---
Local country code
|---
National prefix digit
|---
Local national code
|---
Minimum local number length
|---
Local enterprise node
|---
PSTN access code
|---
International code
|---
Operator code
|
|---
Emergency number
---
Initial digits
|
---
Admin
|
---
Local Functions
|
---
Locality
|
---
Canonical dial
|---
Internal numbers
|---
External numbers
|
|---
External access
---
International access administration.fm
Administration
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3.13.2
Canonical Dial Lookup
The parameters given here are important for establishing outgoing calls and for recognizing incoming calls.
In the local phonebook, and, mostly, in LDAP directories, numbers are stored in canonical format. In order to generate an appropriate dial string according to the settings in Internal num-
bers and External numbers (-> Section 3.13.1), internal numbers must be discerned from ex-
ternal numbers. The canonical lookup table provides patterns which allow for operation.
Furthermore, these patterns enable the phone to identify callers from different local or international telephone networks by looking up the caller’s number in the phonebook. As incoming numbers are not always in canonical format, their composition must be analyzed first. For this purpose, an incoming number is matched against one or more patterns consisting of country codes, national codes, and enterprise nodes. Then, the result of this operation is matched against the entries in the local phonebook.
> To make sure that canonical dial lookup works properly, at least the following parameters of the phone must be provided:
•
Local country code (-> Section 3.13.1)
•
Local area code (-> Section 3.13.1)
•
Local enterprise code (-> Section 3.13.1)
Up to 5 patterns can be defined. The Local code 1 ... 5 parameters define up to 5 different local enterprise nodes, whilst International code 1... 5 define up to 5 international codes, that is, fully qualified E.164 call numbers for use in a PSTN.
Data required
• Local code 1 ... 5: Local enterprise code for the node/PBX the phone is connected to.
Example: "722" for Siemens Munich.
• International code 1 ... 5: Sequence of "+", local country code, local area code, and local enterprise node corresponding to to one or more phonebook entries.
Example: "+4989722" for Siemens Munich.
Administration via WBM
Locality > Canonical dial lookup
Canonical dial lockup
Local code 1:
Local code 2:
Local code 3:
Local code 4:
Local code 5:
Submit
International code 1:
International code 2:
International code 3:
International code 4:
International code 5:
Reset
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---
Admin
|
---
Local Functions
|
---
Locality
|
---
Canonical Dial Lookup
|---
Local code 1
|---
International code 1
|---
Local code 2
|---
International code 2
|---
Local code 3
|---
International code 3
|---
Local code 4
|---
International code 4
|
|---
Local code 5
---
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3.13.3
Phone location
This parameter specifies if phone location information are included in appropriate SIP messages or not included in any SIP messages but such information are allowed to be configured.
Data required
• Phone location: .
Value range: "Signalled", "Not signalled"
Default: "Signalled"
Administration via WBM
Locality > Phone Location
Phone location
Phone location
Submit
Signalled
Reset
Administration via Local Phone
|
--Admin
|
--Local Functions
|
--Locality
|
--Phone location
|
--Phone location
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3.13.4
Dial Plan
OpenStage phones may optionally use a dial plan residing on the phone. By means of the dial plan, the phone can infer from the digits entered by the user that a complete call number has been entered, or that a particular prefix has been entered. Thus, the dialing process can start without the need to confirm after the last digit has been entered, without delay or with a configurable delay. The standard timer, which is found on the WBM under User menu > Configuration
> Outgoing calls > Autodial delay (seconds), is overridden if a dial plan rule is matched.
A dial plan consists of rules defining patterns, timeouts and actions to be performed when a pattern is matched and/or a timeout has expired. The phone can store one dialplan, which can contain up to 48 different rules.
It is very important that the phone’s dial plan does not interfere with the dial plan in the SIP server, PBX, or public network.
The dial plan can be created and uploaded to the phone using the DLS (please refer to the Deployment Service Administration Manual). The DLS can also export and import dial plans in
.csv format. For details about the composition of a dial plan, please refer to Section 5.5, “Dial
Plan”.
The current dial plan, along with its status (enabled/disabled) and error status can be displayed on the WBM via Diagnostics > Fault trace configuration > Download dial plan file.
With software version V2R2, the Dial plan ID and the Dial plan status is displayed in the local menu.
To make use of the dial plan facility, the following requirements must be met:
• A correct dial plan is loaded to the phone.
• In the user menu, Allow immediate dialing is enabled.
• Dial plan enabled is checked.
Administration via WBM (User menu)
User > Configuration > Outgoing calls > Allow immediate dialing
Outgoing calls
Autodial delay (seconds)
Allow callback
Allow busy when dialling
Allow transfer on ring
Allow immediate dialling
6
Submit
;
;
Reset
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System > Features > Configuration > Dial plan enabled
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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---
User
|
---
Configuration
|
---
Outgoing calls
|
---
Immediate dialing
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
General
|
---
Dial plan
|
---
Admin
|---
General Information
|---
Dial plan ID
|
---
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Ringer Setting
3.14
Ringer Setting
3.14.1
Distinctive
The SIP server may provide information indicating a specific type of call within an incoming call.
The phone can use this information to choose a ring tone according to the call type from the
"Distinctive ringer table".
The relevant information is carried as a string in the SIP Alert-Info header. This string is configured in the OpenScape Voice system; please refer to the relevant OpenScape Voice documentation. When the string sent via alert-info matches the string specified in the Name parameter, the corresponding ringer is triggered. For instance, the OpenScape Voice system may send the string Bellcore-dr1 to indicate that a call is from within the same business group, and the Name parameter is set to "Bellcore-dr1". To select a specific ring tone for calls from the same business group, the other parameters corresponding to that Name must be set accordingly.
The Ringer sound parameter determines whether a pattern, i. e. melody, or a specific sound file shall be used as ringer.
Pattern melody selects the melody pattern that will be used if Ringer sound is set to "Pattern".
Pattern sequence determines the length for the melody pattern, and the interval between the repetitions of the pattern. There are 3 variants:
• "1": 1 sec ON, 4 sec OFF
• "2": 1 sec ON, 2 sec OFF
• "3": 0.7 sec ON, 0.7 sec OFF, 0.7 sec ON, 3 sec OFF
The Duration parameter determines how long the phone will ring on an incoming call. The range is 0-300 sec.
With the Audible parameter, the ringer can be muted. In this case, an incoming call will be indicated only visually.
Special Ringers can be configured for the following call types:
• Internal
• External
• Recall
• Emergency
• Special1
• Special2
• Special3
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> To make the special ringers available and configurable to the user, the administrator needs to map the call types to specific ringers via the Ringer setting mapping table in Admin > Ringer setting > Distinctive. Each call type can be mapped to a specific Ringer sound, Pattern Melody, and Pattern sequence.
Abstract names used for Special Ringers:
•"Bellcore-dr1" - normal (internal) alerting or ring-back;
•"Bellcore-dr2" - external alerting or ring-back;
•"Bellcore-dr3" - recall alerting or ring-back (e.g., following transfer).
•"alert-internal" - normal (internal) alerting or ring-back;
•"alert-external" - external alerting or ring-back;
•"alert-recall" - recall alerting or ring-back (e.g., following transfer)
•"alert-emergency" - emergency alerting or ring-back.
•"Line-<DN of Line>-Reserved" - distinctive alerting for a line with number <DN of Line>
Once made available (by the administrator) to the user, the Special Ringers for the call types
listed can be selected and configured via the User menu as shown in Section 3.14.3, “Special
Ringers”.
Administration via WBM
Admin > Ringer setting > Distinctive
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Ringer Setting
Name
Ringer setting
This page allows you to set up interworking with other IP phone systems that support distinctive ringing
Ringer sound
Bellcore-dr1
Impact-Level
Pattern
Ringer2.mp4
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Submit
Pattern melody
3
3
3
3
3
3
3
3
8
4
3
3
3
3
3
2
2
2
2
2
2
2
2
1
2
2
2
2
2
2
Pattern sequence
60
60
60
60
60
60
60
60
0
60
60
60
60
60
60
Reset
Duration (sec) Audible
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Administration via Local Phone
|
---
Admin
|
---
Ringer setting
|
---
Distinctive
|
---
<1 .... 15>
|---
Name
|---
Ringer sound
|---
Pattern melody
|---
Pattern sequence
|
|---
Duration
---
Audible
3.14.2
Map to Specials
The "Mapping table" is not accessible by local menu, WBM, or DLS but is predefined with Ringer name defaults. Only the special ringers for the default types will be shown in the local menu and WBM. If a default Ringer name is not configured in the "Distinctive ringer table" then the mapped entry in the "Special ringer table" will be greyed and read-only.
The "Mapping table" has been configured to identify the distinctive ringer names as a special ringer type and the User has access to configure a different audio file or pattern for this distinctive ringer via their "Special ringer table". Any change made by the User to this special ringer will be reflected in the "Distinctive ringer table" and any change made by Admin in the "Distinctive ringer table" will be reflected in the "Special ringer table".
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Administration via WBM
Admin > Ringer setting > Map To Specials
Submit
Map To Specials
Internal
External
Recall
Emergency
Special1
Special2
Special3
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3 alert-emerge
Reset
3.14.3
Special Ringers
Special Ringers can be configured via the User menu for the following call types:
• Internal
• External
• Recall
• Emergency
• Special1
• Special2
• Special3
Administration via WBM (User menu)
User > Audio > Special ringers
The Special ringers dialog allows the user to change the ring tones for the special call types listed below, provided that the call type is signaled to the phone.
> To make the special ringers available and configurable to the user, the administrator needs to map the call types to specific ringers via the Ringer setting mapping table in Admin > Ringer setting > Distinctive. Each call type can be mapped to a specific Ringer sound, Pattern melody, and Pattern sequence.
Special Ringer Call Types
• Internal
• External
• Recall
• Emergency
• Special1
• Special2
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• Special3
Special ringers
This page allows you to change the ringer played for a limited range of special incoming calls where the type of call has been signalled to the phone.
Call type Ringer sound
Pattern melody Pattern sequence
Internal
Extermal
Recall
Emergency
Special 1
Special 2
Special 3
Ringer2.mp3
Ringer3.mp3
Ringer4.mp3
Ringer5.mp3
Ringer1.mp3
Ringer1.mp3
Ringer1.mp3
Submit
2
2
1
2
2
1
1
1.0 sec. ON, 2.0 sec. OFF
1.0 sec. ON, 2.0 sec. OFF
1.0 sec. ON, 2.0 sec. OFF
1.0 sec. ON, 2.0 sec. OFF
1.0 sec. ON, 4.0 sec. OFF
1.0 sec. ON, 4.0 sec. OFF
1.0 sec. ON, 4.0 sec. OFF
Reset
Administration via Local Phone
|
--User
|
--Audio
|
--<Special Ringers>
|--Internal
|--External
|--Recall
|--Emergency
|--Special 1
|--Special 2
|
--Special 3
For each call type, the following parameters can be configured:
The Ringer sound parameter determines whether a pattern, i. e. melody, or a specific sound file shall be used as ringer.
Pattern melody selects the melody pattern that will be used if Ringer sound is set to "Pattern".
Pattern sequence determines the length for the melody pattern, and the interval between the repetitions of the pattern. There are 3 variants:
• "1": 1 sec ON, 4 sec OFF
• "2": 1 sec ON, 2 sec OFF
• "3": 0.7 sec ON, 0.7 sec OFF, 0.7 sec ON, 3 sec OFF
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Ringer Setting
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Mobility
3.15
Mobility
The Mobility feature requires the OpenScape Deployment Service (DLS). If the phone is mobility enabled by the DLS, a mobile user can log on to the phone and thereby have his own user settings transferred to the phone. These user data are stored in the DLS database and include, for instance, SIP registration settings, dialing properties, key layouts, as well as the user’s phonebook.
If the mobile user changes some settings, the changed data is sent to the DLS server. This ensures that his user profile is updated if necessary.
If Unauthorized Logoff Trap is set to "Yes", a message is sent to the SNMP server if an unauthorized attempt is made to log off the mobile user.
Logoff Trap Delay defines the time span in seconds between the unauthorized logoff attempt and the trap message to the SNMP server.
Timer Medium Priority determines the time span in seconds between a change of user data in the phone and the transfer of the changes to the DLS server.
The Mobility Feature parameter indicates whether the mobility feature is enabled by the DLS or not.
Data required
• Unauthorized Logoff Trap: An SNMP trap is sent on an unauthorized logoff attempt.
Value range: "Yes", "No"
Default: "No"
• Logoff Trap Delay: Time span in seconds between the unauthorized logoff attempt and the SNMP trap.
Default: 300
• Timer Medium Priority: Time span in seconds between a data change in the phone and its transfer to the DLS server.
Default: 60
• Mobility feature: Indicates whether the mobility feature is enabled.
• Managed Profile: Display only field.
• Error Count Local: Display only field.
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Mobility
Unauthorised Logoff Trap
Logoff Trap Delay
Timer Medium Priority
Mobility Feature
Managed Profile
Error Count Local
Submit
300
60
;
0
Reset
Administration via Local Phone
|
--Admin
|
--Mobility
|--Unauthorized Logoff Trap
|--Logoff Trap Delay
|--Timer Medium Priority
|--Mobility Feature
|
|--Managed Profile
--Error Count Local administration.fm
Administration
Mobility
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Transferring Phone Software, Application, and Media Files
3.16
Transferring Phone Software, Application, and Media Files
New software images, hold music, picture clips for phonebook entries, LDAP templates, company logos, screensaver images, and ring tones can be uploaded to the phone via DLS (Deployment Service) or WBM (Web Based Management).
If an incorrect software image is being attempted to be loaded onto the phone, the phone will reject the request and return to normal operation without reboot. As part of this security mechanism, new software binds are identified by a "Supported Hardware Level" information built into the header.
> For all user data, which includes files as well as phonebook content, the following amounts of storage place are available:
• OpenStage 15/20/40: 2,5 MB
• OpenStage 60/80: 8 MB
3.16.1
FTP/HTTPS Server
There are no specific requirements regarding the FTP server for transferring files to the Open-
Stage phone. Any FTP server providing standard functionality will do.
3.16.2
Common FTP/HTTPS Settings
For each one of the various file types, e.g. phone software, hold music, and picture clips, specific FTP/HTTPS access data can be defined. If some or all file types have the parameters
Download method, Server address, Server port, Account, Username, FTP path, and HT-
TPS base URL in common, they can be specified here. These settings will be used for a specific file type if its Use defaults parameter is set to "Yes".
> If Use defaults is activated for a specific file type, any specific settings for this file type are overridden by the defaults.
Additional log messages are issued for the following scenarios
• Update has been allowed due to override flag being set
• Whole part number is not recognized
• Block 4 of part number is not recognized
• Downloaded software does not have a hardware level included
Data required
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• FTP Server address: IP address or hostname of the FTP server in use.
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• FTP Server port: Port number of the FTP server in use. For HTTPS, port 443 is assumed, unless a different port is specified in the HTTPS base URL.
Default: 21
• FTP Account: Account at the server (if applicable).
• FTP Username: User name for accessing the server.
• FTP Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use. If no port number is specified here, port 443 is used. Only applicable if Download method is switched to
"HTTPS".
Administration via WBM
File transfer > Defaults
Defaults
Download method
FTP Server address
FTP
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
File Transfer
|
---
Defaults
|---
Download method
|---
Server
|---
Port
|---
Account
|---
Username
|---
Password
|
|---
FTP path
---
HTTPS base URL
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3.16.3
Phone Software
The firmware for the phone can be updated by downloading a new software file to the phone.
If an incorrect software image is being attempted to be loaded onto the phone, the phone will reject the request and return to normal operation without reboot. As part of this security mechanism, new software binds are identified by a "Supported Hardware Level" information built into the header.
Prerequisite: The phone knows its own hardware level (from the part number and/or by a dynamical check of its HW level).
When a new software bind is downloaded to the phone, the following verifcation is performed:
1. If new software bind has hardware level header included (in the bind header):
Hardware level of new bind is compared with phone’s hardware level a) If compatible (or if Override is set): Proceed with update b) If NOT compatible: Abandon update and return to original application
2. If new software bind does NOT have hardware level header included (in the bind header):
Software version of new bind is compared with minimum known supported SW level a) If compatible (or if Override is set): Proceed with update b) If NOT compatible: Abandon update and return to original application
7
Do not disconnect the phone from the LAN or power unit during software update. An active update process is indicated by blinking LEDs and/or in the display.
3.16.3.1
FTP/HTTPS Access Data
If the default FTP/HTTPS Access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use defaults must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use defaults: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No".
Default: "No".
• Filename: Specifies the file name of the phone software.
• After submit: Specifies action after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
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Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
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Administration via WBM
File transfer > Phone application
Phone application
Use defaults
Download method
FTP Server address
FTP
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Phone app
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename
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3.16.3.2
Download/Update Phone Software
If applicable, phone software should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
When the download has been successful, the phone will restart and boot up using the new software.
> When Phone Software was upgraded to V3R3 there may be displayed a downgrade protection message.
Start Download via WBM
Phone application
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path HFA/OpenStage
HTTPS base URL
Filename opera_bind.img
After submit start download
Submit Reset
In the File transfer > Phone application dialog, set After submit to "start download" and press the Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Phone app.
|
--Admin
|
--File Transfer
|
--Phone app
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.4
Music on Hold
If enabled by the user, the Music on Hold (MoH) sound file is played when a call is put on hold.
> The file size for a Music on Hold file is limited to 1MB. If the file is too large or the contents of the file are not valid, the file will not be stored in the phone.
The following formats for Music on Hold are supported:
• WAV format. The recommended specifications are:
• Audio format: PCM
• Bitrate: 16 kB/sec
• Sampling rate: 8 kHz
• Quantization level: 16 bit
• MIDI format
• MP3 format (OpenStage 60/80 only). A bitrate of 48 kB/sec is recommended.
3.16.4.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use Default smust be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use defaults: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies action after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
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• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
Administration via WBM
File transfer > Hold music
Hold music
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Hold Music
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename
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3.16.4.2
Download Music on Hold
If applicable, Music on Hold should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
Start Download via WBM
Hold music
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path media
HTTPS base URL
Filename hold_on.mp3
After submit start download
Submit Reset
In the File transfer > Hold music dialog, set After submit to "start download" and press the
Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Hold Music.
|
---
Admin
|
---
File Transfer
|
---
Hold Music
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.5
Picture Clips
> Picture clips are available only on OpenStage 60/80 phones.
> The file size for a picture clip is limited to 300 KB. If the file is too large or the contents of the file are not valid, the file will not be stored in the phone.
Picture Clips are small images used for displaying a picture of a person that is calling on a line.
The supported file formats for picture clips are JPEG and PNG (recommended).
3.16.5.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies action after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
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Administration via WBM
File transfer > Picture clip
Picture Clip
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Picture Clip
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename
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3.16.5.2
Download Picture Clip
The download can be triggered from the web interface or from the local phone menu.
Start Download via WBM
Picture Clip
Use defaults
Download method
FTP Server address
FTP
192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path media
HTTPS base URL
Filename einstein.jpg
After submit start download
Submit Reset
In the File transfer > Picture clip dialog, set After submit to "start download" and press the
Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Picture clip.
|
---
Admin
|
---
File Transfer
|
---
Picture clip
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.6
LDAP Template
> LDAP is available on OpenStage 15/20/40/60/80.
The LDAP template is an ASCII text file that uses an allocation list to assign directory server attributes to input and output fields on an LDAP client. The LDAP template must be modified correctly for successful communication between the directory server and the LDAP client.
> The OpenStage phones support LDAPv3.
3.16.6.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies actions after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
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File transfer > LDAP
LDAP
Use defaults
Download method
FTP Server address
FTP
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
File Transfer
|
---
LDAP
|---
Use default
|---
Download method
|---
Server
|---
Port
|---
Account
|---
Username
|---
Password
|---
FTP path
|
|---
HTTPS base URL
---
Filename administration.fm
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3.16.6.2
Download LDAP Template
If applicable, LDAP templates should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
> The OpenStage phone supports LDAPv3.
Start Download via WBM
LDAP
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path media
HTTPS base URL
Filename Idap_template.txt
After submit start download
Submit Reset
In the File transfer > LDAP dialog, set After submit to "start download" and press the Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to LDAP.
|
--Admin
|
--File Transfer
|
--LDAP
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.7
Logo
On OpenStage 40/60/80, a custom background image for the telephony interface can be supplied. In most cases, this will be the company logo.
On OpenStage 40, monochrome bitmap files (BMP) are supported. The ideal size is as follows:
• Width: 144 px
• Height: 32 px
On OpenStage 60/80, the supported file formats are JPEG and PNG. The ideal size values are is as follows:
OpenStage 60:
• Width: 240 px
• Height: 70 px
OpenStage 80:
• Width: 480 px
• Height: 142 px
If the size should deviate from these values, the image will appear skewed.
For guidance on creating a logo file for OpenStage 40/60/80, see Section 5.2, “How to Create
Logo Files for OpenStage Phones”.
3.16.7.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
• Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies actions after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
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Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
Administration via WBM
File transfer > Logo
Logo
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Logo
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename
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3.16.7.2
Download Logo
If applicable, logos should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
Start Download via WBM
Logo
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username
FTP password phone
FTP path media
HTTPS base URL
Filename company_logo.png
After submit start download
Submit Reset
In the File transfer > Logo dialog, set After submit to "start download" and press the Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Logo.
|
---
Admin
|
---
File Transfer
|
---
Logo
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.8
Screensaver
The screensaver is displayed when the phone is in idle mode. It performs a slide show consisting of images which can be uploaded using the web interface.
> Screensavers are available only on OpenStage 60/80 phones.
> The file size for a screensaver image is limited to 300 KB. If the file is too large or the contents of the file are not valid, the file will not be stored in the phone.
For screensaver images, the following specifications are valid:
• Data format: JPG or PNG. JPG is recommended.
• Screen format: 4:3. The images are resized to fit in the screen, so that images with a width/ height ratio differing from 4:3 will appear with deviant proportions.
• Resolution: The phone’s screen resolution is the best choice for image resolution:
• OpenStage 60: 320x240
• OpenStage 80: 640x480
3.16.8.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies actions after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
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• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
Administration via WBM
File transfer > Screensaver
Screensaver
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Screensaver
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|---
Password
|---
FTP path
|
|---
HTTPS base URL
---
Filename
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3.16.8.2
Download Screensaver
If applicable, screensavers should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
Start Download via WBM
Screensaver
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username
FTP password phone
FTP path media
HTTPS base URL
Filename seaside.jpg
After submit start download
Submit Reset
In the File transfer > Screensaver dialog, set After submit to "start download" and press the
Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Screensaver.
|
--Admin
|
--File Transfer
|
--Screensaver
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.9
Ringer File
Custom ring tones can be uploaded to the phone.
> The file size for a ringer file is limited to 1 MB. If the file is too large or the contents of the file are not valid, the file will not be stored in the phone. This limitation is only enforced on WBM. If a ringer file is downloaded via OpenStage Manager, this restriction does not apply.
The following file formats are supported:
• WAV format. The recommended specifications are:
• Audio format: PCM
• Bitrate: 16 kB/sec
• Sampling rate: 8 kHz
• Quantization level: 16 bit
• MIDI format.
• MP3 format (OpenStage 60/80 only). The OpenStage 60/80 phones are able to play MP3 files from 32 kbit/s up to 320 kbit/s. As the memory for user data is limited to 8 MB, a constant bitrate of 48 kbit/sec to 112 kbit/s and a length of max. 1 minute is recommended.
Although the phone software can play stereo files, mono files are recommended, as the phone has only 1 loudspeaker.
See the following table for estimated file size (mono files):
Length
0:15 min
0:30 min
0:45 min
1:00 min
64 kbit/s
0,12 MB
0,23 MB
0,35 MB
0,47 MB
80 kbit/s
0,15 MB
0,29 MB
0,44 MB
0,59 MB
96 kbit/s
0,18 MB
0,35 MB
0,53 MB
0,70 MB
112 kbit/s
0,21 MB
0,41 MB
0,62 MB
0,82 MB
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3.16.9.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to "Yes", and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: "Yes", "No"
Default: "No"
• Filename: Specifies the file name of the phone software.
• After submit: Specifies action after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: "FTP", "HTTPS"
Default: "FTP"
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to "HTTPS".
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File transfer > Ringer file
Ringer file
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Ringer
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename administration.fm
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3.16.9.2
Download Ringer File
If applicable, ring tone files should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
Start Download via WBM
Ringer file
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path media
HTTPS base URL
Filename ring.mp3
After submit start download
Submit Reset
In the File transfer > Ringer dialog, set After submit to "start download" and press the Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Ringer.
|
---
Admin
|
---
File Transfer
|
---
Ringer
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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3.16.10
Dongle Key
The HPT dongle key is a special file that contains a secret hash number which is required to connect the HPT tool to the phone. This testing tool is used exclusively by the service staff.
3.16.10.1
FTP/HTTPS Access Data
If the default FTP/HTTPS access settings (see Section 3.16.2, “Common FTP/HTTPS Settings”) are to be used, Use default must be set to „Yes“, and only the Filename must be spec-
ified.
Data required (in any case)
• Use default: Specifies whether the default FTP/HTTPS access settings shall be used.
Value range: „Yes“, „No“
Default: „No“
• Filename: Specifies the file name of the phone software.
• After submit: Specifies actions after submit button is pressed.
Value range: "do nothing", "start download".
Default: "do nothing".
Data required (if not derived from Defaults)
• Download method: Selects the protocol to be used.
Value range: „FTP“, „HTTPS“
Default: „FTP“
• Server address: IP address or hostname of the FTP/HTTPS server in use.
• Server port: Port number of the FTP/HTTPS server in use.
Default: 21
• Account: Account at the server (if applicable).
• Username: User name for accessing the server.
• Password: Password corresponding to the user name.
• FTP path: Path of the directory containing the files.
• HTTPS base URL: IP address or hostname of the HTTPS server in use; only applicable if
Download method is switched to „HTTPS“.
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Administration via WBM
File transfer > Dongle key
Dongle key
Use defaults
Download method FTP
FTP Server address
FTP Server port 21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit do nothing
Submit Reset
Administration via Local Phone
|
--Admin
|
--File Transfer
|
--Dongle key
|--Use default
|--Download method
|--Server
|--Port
|--Account
|--Username
|--Password
|--FTP path
|
|--HTTPS base URL
--Filename
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Administration
Transferring Phone Software, Application, and Media Files
3.16.10.2
Download Dongle Key File
If applicable, dongle key files should be deployed using the DLS (Deployment Service). Alternatively, the download can be triggered from the web interface or from the Local phone menu.
Start Download via WBM
Dongle key
Use defaults
Download method FTP
FTP Server address 192.168.1.150
FTP Server port 21
FTP account
FTP username phone
FTP password
FTP path media
HTTPS base URL
Filename dongle
After submit start download
Submit Reset
In the File transfer > Dongle key dialog, set After submit to „start download“ and press the
Submit button.
Start Download via Local Phone
1. In the administration menu, set the focus to Dongle key.
|
---
Admin
|
---
File Transfer
|
---
Dongle key
2. Press the g key. A context menu opens. In the context menu, select Download. The download will start immediately.
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Corporate Phonebook: Directory Settings
3.17
3.17.1
Corporate Phonebook: Directory Settings
LDAP
> LDAP is available on OpenStage 15/20/40/60/80 with SIP V3R3.
The Lightweight Directory Access Protocol enables access to a directory server via an LDAP client. Various personal information is stored there, e.g. the name, organization, and contact data of persons working in an organization. When the LDAP client has found a person’s data, e. g. by looking up the surname, the user can call this person directly using the displayed number.
> The OpenStage phones support LDAPv3.
For connecting the phone’s LDAP client to an LDAP server, the required access data must be configured. The parameter Server address specifies the IP address of the LDAP server. The parameter Transport defines whether the phone has to continue to use an unencrypted TCP connection to the LDAP server, or to use an encrypted TLS connection to a separate LDAPS port on the LDAP server. Depending on the setting of Transport the Secure Port (for TLS) or the Server port (for TCP) are to be defined. If the Authentication is not set to "Anonymous", the user must authenticate himself with the server by providing a User name and a corresponding Password. The user name and password are defined by the administrator. The user name is the string in the LDAP bind request, e. g. "C=GB,O=SIEMENS
COMM,OU=COM,L=NTH,CN=BAYLIS MICHAEL". The internal structure will depend on the specific corporate directory.
For a quick guide on setting up LDAP on an OpenStage phone, please refer to Section 5.3,
“How to Set Up the Corporate Phonebook (LDAP)”.
Data required
• Server address: IP address or hostname of the LDAP server.
• Transport: Defines Transport mode, whether LDAP interface uses TCP and is unencrypted, or uses TLS and is encrypted.
Value range: "TCP", "TLS"
Default: "TCP"
• Secure Port: Defines the port of the appropriate TLS interface on LDAP server when
Transport is set to TLS.
Default: "636"
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• Server port: Port on which the LDAP server is listening for requests, when Transport is set to TCP.
Default: 389
• Authentication: Authentication method used for connecting to the LDAP server.
Value range: "Anonymous", "Simple"
Default: "Anonymous"
• User name: User name used for authentication with the LDAP server in the LDAP bind request.
• Password: Password used for authentication with the LDAP server.
• Search trigger timeout: Timespan between entering the last character and search string submission to the LDAP server.
Administration via WBM
Local Functions > Directory settings
Directory settings
LDAP Server address
Transport TCP
Secure port 636
LDAP Server port 389
Authentication Anonymous
User Name
Password
Search trigger timeout 3
Submit Reset
Administration via Local Phone
|
---
Admin
|
---
Local Functions
|
---
LDAP
|---
Server address
|---
Transport
|---
LDAP secure port
|---
LDAP server port
|---
Authenticate
|---
User name
|
|---
Password
---
Search trigger (s)
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Speech
3.18
Speech
3.18.1
RTP Base Port
The port used for RTP is negotiated during the establishment of a SIP connection. The RTP base port number defines the starting point from which the phone will count up when negotiating. The default value is 5010.
The number of the port used for RTCP will be the RTP port number increased by 1.
Administration via WBM
Network > Port Configuration
SIP server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port mode
PC port autoMDIX
Submit
Port configuration
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
Administration via Local Phone
|
--Admin
|
--Network
|
--Port Configuration
|
--RTP base
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Speech
3.18.2
Codec Preferences
If Silence suppression is activated, the transmission of data packets is suppressed on no conversation, that is, if the user doesn’t speak.
The OpenStage phone provides the codecs G.711, G.722, and G.729. When a SIP connection is established between two endpoints, the phones negotiate the codec to be used. The result of the negotiation is based on the general availability and ranking assigned to each codec. The administrator can allow or disallow a codec as well as assign a ranking number to it.
The Packet size, i. e. length in milliseconds, of the RTP packets for speech data, can be set to 10ms, 20ms, 30ms, 60ms or to automatic detection.
Data required
• Silence suppression: Suppression of data transmission on no conversation.
Value range: "On", "Off"
Default: "Off"
• Allow "HD" icon: If "On" an additional icon is shown when codec G.722 is used.
Value range: "On", "Off"
Default: "On"
• Packet size: Size of RTP packets in milliseconds.
Value range: "10 ms", "20ms", "30ms", "60ms", "Automatic"
Default: "Automatic"
• G.711: Parameters for the G. 711 codec.
Value Range: "Choice 1", "Choice 2", "Choice 3", "Disabled", "Enabled"
Default: "Choice 1"
• G.729: Parameters for the G. 729 codec.
Value Range: "Choice 1", "Choice 2", "Choice 3", "Disabled", "Enabled"
Default: "Choice 2"
• G.722: Parameters for the G. 722 codec.
Value Range: "Choice 1", "Choice 2", "Choice 3", "Disabled", "Enabled"
Default: "Disabled"
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Administration via WBM
Speech > Codec preferences
Codec preferences
Silence suppression
Allow "HD" icon
Packet size Automatic
;
G.711 ranking X
G.729 ranking X
G.722 ranking
Submit Reset
Administration via Local Phone
|
--Admin
|
--Speech
|
--Codec Preferences
|--Silence suppression
|--Packet size
|--G.711
|--G.729
|
|--G.722
--Allow "HD" icon
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Speech
3.18.3
Audio Settings
The usage of microphone and speaker for speakerphone mode can be controlled by the administrator.
Both microphone and loudspeaker can be switched on or off separately. By default, both microphone and loudspeaker are switched on.
> The microphone control is not valid for OpenStage 20E, as this model has no builtin microphone.
Administration via WBM
Speech > Audio Settings
Audio settings
DTMF playback
Submit
Mute Settings Microphone ON-Loudspeaker ON
;
Reset
Administration via Local Phone
|
---
Admin
|
---
Speech
|
---
Audio Settings
|
|---
Disable microphone
|
---
Disable loudspeech
---
DTMF playback
The DTMF playback feature aims at the capability to play DTMF digits received using
RFC2833 coding (i.e. Rtp events) in the current active audio device (headset / loudspeaker / handset).
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Applications
3.19
3.19.1
Applications
XML Applications/Xpressions (OpenStage 60/80)
3.19.1.1
Setup/Configuration
The XML interface enables server-based applications with a set of GUI elements. The technologies commonly used in web applications can be used: Java Servlets, JSP, PHP, CGI etc., delivered by servers such as Tomcat, Apache, Microsoft IIS.
> A maximum number of 20 XML applications can be configured on OpenStage 60/80 phones.
There are several types of XML applications, which mainly differ in the way they are started and stopped:
• Regular XML applications are started by navigating to the applications menu using the v
key, or by pressing a programmable key (see Section 3.8.29, “Start Application”). They can
be stopped via the applications menu. Regular XML applications are configured via
Applications > XML applications > Add application.
• Xpressions is a special Unified Communications application which also uses the XML interface. Thus, the configuration is just the same as with other XML applications, except a few parameters, which are pre-configured. For details, please refer to the relevant Xpressions documentation. When configured on the phone, a press on the messages mode key x will invoke this application. Xpressions is configured via
Applications > XML applications > Xpressions.
• A messages application is configured like a regular application. It is started and stopped via the messages mode key x , thus enabling the deployment of an alternative voicemail server. From firmware version V2R1 onwards, the XML application can control the white
LED which indicates new messages. A messages application is configured via
Applications > XML applications > Add messages application.
• A phonebook application is configured like a regular application. It is started and stopped via the phonebook mode key u , thus enabling the deployment of a remote phonebook in place of the personal (local) or corporate (LDAP) phonebook. A phonebook application is configured via Applications > XML applications > Add phonebook application.
• A call log application is configured like a regular application. It is is started and stopped via the call log mode key w , thus enabling the deployment of a remote application that handles call history. From firmware version V2R1 onwards, the XML application can control the white LED which indicates missed calls. A call log application is configured via Applica-
tions > XML applications > Add call log application.
• A help application (OS60/80 only) is configured like a regular application. It is is started and stopped via the help mode key y , thus enabling the deployment of a remote help. A help application is configured via Applications > XML applications > Add help application.
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For detailed information about the OpenStage XML application interface, please see the Open-
Stage 60/80 - XML Applications Developer’s Guide. You can find the current version under http://wiki.unify.com/index.php/OpenStage_XML_Applications .
To set up a new XML application, enter the access data for the application on the server, which is described in the following.
The Display name can be defined freely. This name will appear in the applications tab once the application is configured, and it will appear in a newly created tab when the application is running. With Xpressions, this value is predefined as "Xpressions".
The Application name is used by the phone software to identify the XML application running on the phone. With Xpressions, this value is predefined as "Xpressions".
The HTTP Server address is the IP address or domain name of the server which hosts the remote program. HTTP Server port specifies the corresponding port.
The Protocol for exchanging XML data with the server-side program can be set to "HTTP" or
"HTTPS".
Program name on server specifies the relative path to the servlet or to the first XML page of the application on the server. The relative path refers to the root directory for documents on the web server. For instance, if an XML document is saved in:
C:\Program Files\Apache Group\Apache\htdocs\ipp\ippTest.xml
the entry is: ipp/ippTest.xml
.
The program name cannot be longer than 100 characters.
Auto start determines whether the application is started automatically on phone startup or on mobile user logon. Please note that, for being started on logon, the application must be part of the mobile user’s profile. When activated, the application will be ready without delay as soon as the user presses the corresponding start key or navigates to the application in the application menu.
Use proxy enables an HTTP/HTTPS proxy for communication with the server, if desired. If disabled, a direct connection is used.
XML trace enabled determines whether debugging information is sent to a special debugging program on the remote server. The relative path for the debugging program is given by the De-
bug program name parameter. When enabled, trace information about the XML elements and key internal objects is sent to the remote debug program.
Debug program name specifies the relative path to a special program on the same server as the program specified by Program name. This program must be able to receive the debug information sent by the phone as HTTP/HTTPS POST requests with Content-Type set to application/x-www-form-urlencoded
.
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XML applications can have internal tabs, if desired. The number of these tabs is specified in
Number of tabs.
> For an XML application with a number of tabs > 0, one of the entries between Tab
1 Application Name and Tab 3 Application Name must be set to the same value as the Application name that it is associated with. When the XML application is started, the tab which has the same name as the XML application is the tab that initially gets focus.
All tabs start determines whether all tabs of the application are started automatically when the application is started.
Tab 1...3 Display Name provides the label text for the corresponding tab.
Tab 1...3 Application Name is required if the application has internal tabs. This is a unique name for the specified tab. The remote program will use this name to provide the tab with specific content.
Auto restart / Restart after change : If checked, a running XML application is automatically restarted after it has been modified. This might be especially useful for special XML applications, like messages applications, or phonebook applications, as these cannot be stopped or restarted by the user. Please note that a restart will take place even if no changes have been made for the application selected in the Modify/Delete application mask, and Submit has been pressed. After the XML application has restarted, this option is automatically unchecked.
If the option is checked whilst the XML application is not running, there will be no restart, and the option is automatically unchecked.
Data required
• Display name: Program name to be displayed on the phone.
Value specifications:
• It must be unique on the phone.
• It cannot contain the ’^’ character.
• It cannot not be empty.
• Its length cannot not exceed 20 characters.
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• Application name: Used internally to identify the XML application running on the phone.
Value specifications:
• It must be unique on the phone.
• It cannot contain non-alphanumeric characters, spaces for instance.
• The first character must be a letter.
• It must not be empty.
• Its length must not exceed 20 characters.
• Protocol: Communication protocol for the data exchange with the server.
Value range: "HTTP", "HTTPS"
Default: "HTTPS"
• HTTP Server address: IP address or domain/host name of the server that provides the application or the XML document.
Examples: 192.168.1.133, backoffice.intranet
• Server port number: Number of the port that the server uses to provide the application or
XML document.
Examples: 80 (Apache default port), 8080 (Tomcat default port).
• Program name: Relative path to the servlet or to the first XML page of the application on the server. For instance, if an XML document is saved in:
C:\Program Files\Apache Group\Apache\htdocs\ipp\ippTest.xml
the entry is: ipp/ippTest.xml
The program name cannot be longer than 100 characters.
• Use proxy: Enables or disables an HTTP/HTTPS proxy for communication with the server.
Value range: "Yes", "No"
Default: "No"
• XML trace enabled: Enables or disables the debugging of the XML application.
Value range: "Yes", "No"
Default: "No"
• Debug program name: The relative path to a special servlet that receives the debug information.
Administration via WBM
A fixed function key can be defined as a start key for an XML application, in addition to the previously available start methods. Since the parameters are the same for those types of application, only the screenshot for a regular XML application is shown underneath.
Applications > XML Applications > Add application
Applications > XML Applications > Add messages application
Applications > XML Applications > Add phonebook application
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Applications > XML Applications > Add call log application
Applications > XML Applications > Add help application
Add application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
Applications > XML Applications > Modify/Delete application
Modify/Delete application
Select application
Modify testxml
Delete
Settings
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Mode key
Submit testxml testxml
192.168.1.150
8080 http testxml/servlet
;
No
No
0
0
Reset
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|
---
Admin
|
---
Applications
|
---
XML
|
---
Add application
|---
Display name
|---
Application name
|---
Server address
|---
Server port
|---
Protocol
|---
Program name
|---
Auto start
|---
Use proxy
|---
XML trace enabled
|---
All tabs start
|---
Debug program name
|---
Number of tabs
|---
Tab 1 display name
|---
Tab 1 application name
|---
Tab 2 display name
|---
Tab 2 application name
|---
Tab 3 display name
|---
Tab 3 application name
|
---
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Administration
Applications
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Applications
3.19.1.2
HTTP Proxy
For the HTTP data transfer between the phone and the server hosting the remote program, an
HTTP proxy can be used.
First, the proxy itself must be configured. Enter the IP address of the proxy it in the Network >
IP configuration > HTTP proxy parameter, and the corresponding port in the Network > Port configuration > HTTP proxy parameter.
Use proxy enables or disables the use of the proxy. If disabled, the phone connects directly to the server. By default, the use of a proxy is disabled.
Administration via WBM
Applications > XML Applications > Add application
Add application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Use proxy
XML Trace enabled
Debug program on server
Submit http
Yes
Yes
Reset
Applications > XML Applications > Modify/Delete application
Modify/Delete application
Select application
Modify
Weather
Delete
Settings
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Use proxy
XML Trace enabled
Debug program on server
Submit
Weather
Weather
87.106.21.36
8080 http
WR/WR
No
No
Reset
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Network > General IP configuration
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
DHCP
;
192.168.1.105
192.168.1.2
Reset
Network > Port configuration
SIP server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port modo
PC port autoMDIX
Submit
Port configuration
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
Administration via Local Phone
|
---
Admin
|
---
Network
|
---
General IP configuration
|
---
HTTP proxy
|
--Admin
|
--Network
|
--Port configuration
|
--HTTP proxy
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Applications
3.19.1.3
Modify an Existing Application
An existing application can be modified by changing its parameters. Please ensure to select the desired application before changing the parameters.
Administration via WBM
Applications > XML applications > Modify/Delete application
Modify/Delete application
Select application
Modify
Weather
Delete
Settings
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Use proxy
XML Trace enabled
Debug program on server
Submit
Weather
Weather
87.106.21.36
8080 http
WR/WR
No
No
Reset
Administration via Local Phone
|
---
Admin
|
---
Applications
|
---
XML
|
---
<Application to be modified>
|---
Display name
|---
Application name
|---
Server address
|---
Server port
|---
Protocol
|---
Program name
|
|---
XML trace enabled
---
Debug program name
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Applications
3.19.1.4
Remove an Existing Application
An existing application can be removed. Please ensure to select the desired application before changing the parameters.
Administration via WBM
Applications > XML applications > Modify/Delete application
Modify/Delete application
Select application
Modify
Weather
Delete
Settings
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Use proxy
XML Trace enabled
Debug program on server
Submit
Weather
Weather
87.106.21.36
8080 http
WR/WR
No
No
Reset
Administration via Local Phone
Select the application to be deleted, and, in the context menu, select Remove & exit.
|
--Admin
|
--Applications
|
--XML
|
--<Application to be deleted>
3.19.1.5
Application Start by Programmable Key
To offer more convenience to the user, a previously configured application can be started by a free programmable key. For this purpose, the appropriate Application name and a Key label must be entered.
Administration via WBM
System > Features > Program keys
Start Application
Key label 4 App:
Application name IppKeySaller
Submit Reset
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Password
3.20
Password
The passwords for user and administrator can be set here. They have to be confirmed after entering. The factory setting is "123456"; it should be changed after the first login (Password
handling V2R2 onwards see Section 3.4.5.5, “Change Admin and User password”). Usable
characters are 0-9 A-Z a-z .*#,?!’+-()@/:_
Administration via WBM
Security and Policies > Password > Change Admin password
Change Admin password
Current password
New password
Confirm password
Submit Reset
Security and Policies > Password > Change User password
Change User password
Admin password
New password
Confirm password
Submit Reset
Administration via Local Phone
|
--Admin
|
--Security & policies
|
--Password
|--Change admin password
|--Change user password
|
--Confirmation
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Troubleshooting: Lost Password
3.21
Troubleshooting: Lost Password
If the administration and/or user password is lost, and there is no DLS available, new passwords must be provided. For this purpose, a factory reset is necessary. Take the following steps to initiate a factory reset:
1. Press the number keys 2-8-9 simultaneously. The factory reset menu opens.
>
The Factory reset claw option needs to be enabled for this to work - see Section
3.4.2, “Access Control”.
2. In the input field, enter the special password for factory reset: "124816".
3. Confirm by pressing i .
3.22
Restart Phone
If necessary, the phone can be restarted from the administration menu.
Administration via WBM
Maintenance > Restart Phone
Restart Phone
Confirm Restart
3.23
Factory Reset
This function resets all parameters to their factory settings. A special reset password is required for this operation: "124816".
Administration via WBM
Maintenance > Factory reset
Factory reset
Factory reset password
Submit Reset
Administration via Local Phone
|
--Admin
|
--Maintenance
|
--Factory reset
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SSH – Secure Shell Access
3.24
SSH – Secure Shell Access
The phone’s operating system can be accessed via SSH for special troubleshooting tasks. As of V3, administration via DLS is also supported. Hereby, the administrator is enabled to use the built-in Linux commands. As soon as SSH access has been enabled using the WBM, the system can be accessed by the user "admin" for a specified timespan. When this timespan has expired, no connection is possible any more. The user "admin" has the following permissions:
• Log folder and files: read only
• User data folder and files: read/write access
• Opera deploy folders and files: read only
• Version folder: read/write access; version files: read only
> It is not possible to logon as root via SSH.
When Enable access is enabled, and the parameters described underneath are specified,
SSH access is activated. By default, SSH access is disabled.
With the Session password parameter, a password for the "admin" user is created. This password is required. It will be valid for the timespan specified in the parameters described underneath.
Access minutes defines the timespan in minutes within which the SSH connection must be established. After it has expired, a logon via SSH is not possible. The possible values (as of
V3) are: 1, 2, 3, 4, 5, 6, 7, 8, 9, 10.
Session minutes defines the maximum length in minutes for an SSH connection. After it has expired, the "admin" user is logged out. The possible values are 5, 10, 20, 30, 60.
Administration via WBM
Maintenance > Secure Shell
Secure Shell
Enable access
Session password
Access minutes 1
Session minutes 1
Submit Reset
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Display License Information
3.25
Display License Information
The license information for the OpenStage phone software currently loaded can be viewed via the local menu.
> The license information can also be viewed by users who logged on using the User login if logging on as Admin is not permitted.
Administration via Local Phone
|
---
Admin
|
---
Licence information
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Diagnostics
3.26
Diagnostics
> Some of the diagnostic tools and functions may reveal personal data of the user, such as caller lists. Thus, with regards to data privacy, it is recommended to inform the user when diagnostic functions are to be executed.
3.26.1
Display General Phone Information
General information about the status of the phone can be displayed if desired.
Displayed Data
• MAC address: Shows the phone’s MAC address.
• Software version: Displays the version of the phone’s firmware.
• Last restart: Shows date and time of the last reboot.
• Backlight type : Indicates whether the phone has a backlight, and, if applicable, the type of backlight.
Value range: 0 (no backlight); 1 (cathode tube backlight); 2 (LED backlight)
Display on the WBM
General information
General information
MAC address: 0001e323f9a1
Software version: 0.7.5.0004-061027
Last restart: 2014-02-18T13:30
Backlight type 1
Display on the Local Phone
|
---
Admin
|
---
General Information
|---
MAC address
|---
Software version
|---
Last restart
|---
Dial plan ID
|
---
Dial plan status
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3.26.2
View Diagnostic Information
In addition to the general phone information (see Section 3.26.1, “Display General Phone Information”), extended data can be viewed.
> The Diagnostic Information can also be viewed by the administrator on the local phone by selecting Diagnostic information > View.
Display on the WBM
Diagnostics > Diagnostic information > View
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View
2011-10-16 20:22:33
00 Terminal number:
01 SIP server:
02 SIP port:
03 SIP registrar:
04 SIP registrar port
05 SIP gateway:
06 SIP gateway port
07 SIP transport:
08 SIP local:
09 Server features:
10 DNS results:
11 Multiline:
12 Registered lines:
13 Backup active:
14 Backup proxy:
15 Use secure calls:
16 SDES status:
17 Secure SIP server:
18 Software version:
19 Display message:
20 Last restart:
21 Memory free:
22 Protocol mode:
23 IP4 address:
24 IP4 subnet mask:
25 IP4 default route:
26 Primary DNS:
27 Secondary DNS:
28 IP4 route 1 IP:
29 IP4 route 1 gateway:
30 IP4 route 1 mask:
31 IP4 route 2 IP:
32 IP4 route 2 gateway:
33 IP4 route 2 mask:
34 IP6 address:
35 IP6 prefix length:
36 IP6 global gateway:
37 IP6 link local addr:
38 IP6 route 1 dest:
39 IP6 route 1 pref len:
40 IP6 route 1 gateway:
41 IP6 route 2 dest:
42 IP6 route 2 pref len:
43 IP6 route 2 gateway:
44 MAC address:
45 Discovery mode:
46 DHCP re-use:
47 DHCPv6:
48 DHCPv6 re-use:
49 LAN port type:
50 PC port status:
51 PC port type:
52 PC port autoMDIX:
53 VLAN ID:
54 QoS Layer 2:
55 QoS Layer 2 voice:
56 QoS Layer 2 signalling:
57 QoS Layer 2 default:
58 QoS Layer 3:
59 QoS Layer 3 voice:
60 QoS Layer 3 signalling:
61 LLDP-MED operation:
62 XML application:
None
63 XML app config:
None
None
None
None
None
None
None
None
None
None
None
None
0001e325eaca
Manual
No
Yes
No
0
None
0
No
None
None
5
None
0
Yes
EF / 46
AF31 / 26
None
3339
192.168.1.230
5060
192.168.1.230
5060
192.168.1.230
5060
UDP
5060
No
5060
No
5060
Yes
192.168.1.148
No
0
0
V3R0.50.0 110924
None
2011-10-10T23:59:01
65733K free
IPv4
192.168.1.235
255.255.255.0
192.168.1.2
192.168.1.105
192.168.1.2
None
None
None
None
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3.26.3
User Access to Diagnostic Information
If this option is enabled, extended phone data is also displayed to the user. To view the data, the user must click on the "Diagnostic information" link in the user menu.
> The Diagnostic Information can also be viewed by the user on the local phone by selecting User > Diagnostic information.
Administration via WBM
Diagnostics > Diagnostic information > User access
User access
User Access
Submit
;
Reset
3.26.4
Diagnostic Call
The feature "Rapid Status Diagnostic Call" will provide the possibility to place a diagnostic call, for example by the user, which starts call related tracing on the phone and on involved Open-
Scape Voice and collect these traces at OpenScape Voice Trace Manager (OSVTM). With all these traces available, a call can be followed throughout the voice system and a possible problem can be detected faster. As all traces from all involved components are available at the first level support, the analysis of a possible problem can be started immediately.
A so-called diagnostic scenario will enable traces on all involved SIP components of the OSC
Voice solution and store all traces at a central server. A tool will help service to follow a call through the traces and determine the point of problem.
The approach is to use a SIP Header ([1]) to indicate, whether a call is a diagnostic call or not.
Presence of this header will mean that related call is a diagnostic call. Absence of this field means a non-diagnostic call. This header will either switch on traces in the solution component or be ignored, if it isn't supported. If the call is recognized as a diagnostic call, the traces will be sent to DLS as a first step and then DLS will forward them to OSVTM. Collected traces will either be sent after a successful end of diagnostic scenario or trace file is full.
For enabling tracing on all involved solution components, a call must be recognized to be a "diagnostic" call. Therefore, a special SIP header will be added to the signalling messages. All components which are able to support such a call will then switch on traces and send the traces to DLS server (which will forward them to a pre-defined OSVTM server).
A dial-prefix has been chosen, as the dialled number should be identical to a number, where the user identified a possible problem. This prefix will be filtered before placing a call, so that the SIP messages will be similar to the ones for the problematic destination.
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The SIP header "X-Siemens-Trace-ID" has been chosen, as this is a special SIP field created for this feature. Existence of the diagnostic call, start and finish of a diagnostic call can be determined via this field [1].
Trace id will be unique throughout the system and the following format will be used to generate trace id:
TraceId: <UNIX_Timestamp>_<Last 6 bytes of MAC Address>
If related calls (diagnostic or not) are established following the start of the diagnostic call, then it turns to be a diagnostic scenario. Related calls become diagnostic (if they are not already) and traces are collected until the last diagnostic call ends plus a predefined timer. This timer guarantees capturing related information regarding to a problematic scenario.
The diagnostic call can only be determined during the call so initial traces might get lost. For this reason, user may need to do additional call. This is completely user related and user should be informed about the process. There will not be any restriction to prevent user to dial the prefix.
If the prefix is configured by admin, user can always dial the prefix and start a diagnostic call.
The prefix has to consist of the leading asterisk followed by three digits and the hash. Example:
*333#.
Administration via WBM
Maintenance > Diagnostic call
Diagnostic call
Prefix Code
Submit
*333#
Reset
Administration via Local Phone
|
--Admin
|
--Maintenance
|
--Diagnostic Call
Admin will not be able to change trace settings or can not clear the existing phone traces during an active diagnostic tracing. If admin tries to change trace configuration or delete existing traces this will not be allowed and admin will get the following error: Change not allowed: Diag- nostic tracing is active!
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3.26.5
LAN Monitoring
The LAN port mirror facility allows for monitoring all network traffic at the phone’s LAN port.
Additionally, there is a possibility to monitor LAN traffic and port settings in the Local user menu:
|
---
User
|
---
Network information
|---
Phone address
|---
Web address
|---
IPv4 address
|---
IPv6 Global Address
|---
IPv6 Linklocal Address
|---
LAN RX
|---
LAN TX
|---
PC RX
|---
PC TX
|---
LAN autonegotiated
|---
LAN information
|---
PC autonegotiated
|
---
PC information
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3.26.6
LLDP-MED
When the phone is connected to a switch with LLDP-MED capabilities, it can receive a VLAN
ID and QoS parameters and advertise its own network-related properties. The data is exchanged in TLV (Type-Length-Value) format.
Both sent and received LLDP-MED data can be monitored at the administrator interface.
> For details on LLDP-MED, please refer to the ANSI/TIA-1057 standard.
For a network configuration example that shows LLDP-MED in operation, please refer to
Section 5.4, “An LLDP-Med Example”.
Displayed Data
• Extended Power: Power Consumption; relevant for PoE.
• Network policy (voice): VLAN ID and QoS (Quality of Service) parameters for voice transport.
• Network policy (signalling): VLAN ID and QoS (Quality of Service) parameters for signalling.
• LLDP-MED capabilities: The LLDP-MED TLVs supported by the phone and the switch as well as the specific device class they belong to.
• MAC_Phy configuration: Identifies the possible duplex and bit-rate capability of the sending device, its current duplex and bit-rate capability, and whether theses settings are the result of auto-negotiation during the initialization of the link, or of manual set override actions.
• System capabilities: The devices advertise their potential and currently enabled functions, e. g. "Bridge", "Telephone".
• TTL: Time To Live. This parameter determines how long the TLVs are valid. When expired, the device will send a new set of TLVs.
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View Data From WBM
Diagnostics > LLDP-MED TLVs
LLDP-MED TLV’s
Sent
Sent: Mon Oct 27 10:41:14 2013
Received
Received: Mon Oct 27 10:41:14 2013
Channel ID TLV Data
.ID = 163.165.2.105
Channel ID TLV Data
.ID = 00:3E:37:01:20:01
TTL TLV Data
.records = 120
TTL TLV Data
.records = 120
System Caps TLV Data
.Supported = Bridge, Telephone
.Enabled = Telephone
System Caps TLV Data
.Supported = Other, Repeater
.Enabled = Other, Repeater
View Data From Local Menu
If both sent and received values are concordant, OK is appended to the parameter. If not, an error message is displayed.
|
---
Admin
|
---
Network
|
---
LLDP-MED operation
|---
Extended Power
|---
Network policy (voice)
|---
LLDP-MED cap’s
|---
MAC_Phy config
|
|---
System cap’s
---
TTL
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3.26.7
IP Tests
For network diagnostics, the OpenStage phone can ping any host or network device to determine whether it is reachable. Additionally, the IP route to a host or network device can be traced using the traceroute tool contained in the phone software.
The Pre Defined Ping tests provide pinging for a pre-defined selection of servers: DLS, SIP server, and SIP registrar.
Ping tests enables the pinging of a random IP address.
The Pre Defined Trace tests provide traceroute tests for a pre-defined selection of servers:
DLS, SIP server, and SIP registrar.
Traceroute enables traceroute tests for a random IP address.
Administration via WBM
Diagnostics > Miscellaneous > IP tests
IP tests
Pre Defined Ping tests
Ping DLS
Ping tests
Ping
Ping
Pre Defined Trace tests
Traceroute DLS
Traceroute
Traceroute
Traceroute
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3.26.8
Process and Memory Information
The processes currently running on the phone’s operating system as well as their CPU and memory usage can be monitored here. 100 processes are monitored on the web page. For further information, please refer to the manual of the "top" command for Unix/Linux systems, or to related documentation.
The amount of free memory is checked on a regular basis in order to prevent problems caused by low memory. This check determines whether a recovery is necessary.
When Disable reboot is checked, no reboot will take place when a memory problem has been found. However, recovery requires a reboot.
The recovery process will be triggered when the available main memory (RAM) falls below a given threshold value. As memory consumption is assumed to be higher during working hours, two thresholds are configurable. The High Threshold (MBs) parameter defines the threshold for off-time. For OpenStage 15/20/40, the default value is 10 MB, and for OpenStage 60/80, it is 30 MB. With Low Threshold (MBs), the threshold for off-time is defined. For OpenStage 15/
20/40, the default value is 8 MB, and for OpenStage 60/80, it is 20 MB.
The beginning and end of the working hours are defined in 24 hours format with Working Hour
Start (Default: 5) and Working Hour End (Default: 24).
When memory shortage has occured, information about the incident is written to a log file which can be viewed via the Download memory info file link. If there has been a previous case of memory shortage, the corresponding log file can be viewed via Download memory info file.
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Diagnostics > Miscellaneous > Memory information
Memory information
Memory Monitor Configuration
Disable Reboot
High Threshold(MBs)
Low Threshold(MBs)
Working Hour Start
Working Hour End
Download memory info file
Submit
Device Memory Information
10
8
5
24
Download old memory info file
Reset
Mem: 111336K used, 12380K free, 0K shrd, 0K buff, 55084K cached
CPU: 5% usr 15% sys 5% nic 25% idle 0% io 0% irq 50% sirq
Load average: 0.14 0.13 0.09 1/196 6098
PID PPID USER STAT VSZ %MEM %CPU COMMAND
6098 1908 root R 1420 1% 40% /bin/busybox top -d 0 -a -n 1 -l 600 -b
2063 1876 root S N 34148 28% 10% PhoneletLauncher desktopphonelet.phd V3 R0.50.0 SIP 110924 WP4 Siemens SIP DE de DD.MM.YYYY
24HR 0 NO_APP_PROP
3664 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
3665 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1929 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2515 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1902 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2992 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1876 1855 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1880 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2057 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1881 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
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3.26.9
Fault Trace Configuration
Error tracing and logging can be configured separately for all components, i. e. the services and applications running on the OpenStage phone. The resulting files can be viewed in the WBM web pages over the Download links.
The File size (bytes) parameter sets the maximum file size. When it is reached, the data is saved as old file, and a new file is generated. From then on, the trace data is written to the new file. When the maximum file size is reached again, the data is saved as old file once more, thereby overwriting the previous old file. The default value is 65536.
> The absolute maximum file size is 6290000 bytes. However, on OpenStage 15/20/
40 phones, a maximum size no greater than 1000 000 bytes is recommended due to the amount of available memory.
The Trace timeout (minutes) determines when to stop tracing. When the timeout is reached, the trace settings for all components are set to OFF, but ERROR and STATUS messages are still written to the trace file ad infinitum. When the trace file has reached its maximum size, the data is saved, and a new file is created (for more information, see File size (bytes) above). If the value is 0, the trace data will be written without time limit.
If Automatic clear before start is checked, the existing trace file will be deleted on pressing the Submit button, and a new, empty trace file will be generated. By default, it is unchecked.
You can read the log files by clicking on the appropriate hyperlinks (the hyperlinks work only if the file in question has been created). The following logs can be viewed:
• Download trace file
The trace data according to the settings specified for the services.
• Download old trace file
The trace file is stored in permanent memory. When the file has reached its size limit, it will be saved as old trace file, and the current exception file is emptied for future messages.
The old trace file can be viewed here.
• Download saved trace file
Normally, the trace file is saved only in the phone RAM. When the phone restarts in a controlled manner, the trace file will be saved in permanent memory.
• Download syslog file
Messages from the phone’s operating system, including error and exception messages.
• Download old syslog file
Old messages from the phone’s operating system.
• Download saved syslog file
Saved messages from the phone’s operating system.
• Download exception file
If an exceptions occurs in a process running on the phone, a message is written to this file.
These messages are incorporated in the syslog file (see Download syslog file also).
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• Download old exception file
The exception file is stored permanent memory. When the file has reached its size limit, it will be saved as old exception file, and the current exception file is emptied for future messages. The old exception file can be viewed here.
• Download upgrade trace file
The trace log created during a software upgrade.
• Download upgrade error file
The error messages created during a software upgrade. These messages are incorporat-
ed in the syslog file (see Download syslog file also).
• Download dial plan file
If a dial plan has been uploaded to the phone, it is displayed here, along with its status (en-
abled/disabled) and error status. For details, please refer to Section 3.13.4, “Dial Plan” and
Section 5.5, “Dial Plan”.
• Download Database file
Configuration parameters of the phone in SQLite format.
• Download HPT remote service log file
Log data from the HPT service.
• Download security log file
Log data from the Security Log Service.
By pressing Submit, the trace settings are submitted to the phone. With Reset, the recent changes can be canceled.
The following trace levels can be selected:
• OFF: Default value. Only error messages are stored.
• FATAL: Only fatal error messages are stored.
• ERROR: Error messages are stored.
• WARNING: Warning messages are stored.
• LOG: Log messages are stored.
• TRACE: Trace messages are stored. These contain detailed information about the processes taking place in the phone.
• DEBUG: All types of messages are stored.
Brief Descriptions of the Components/Services
• Administration
Deals with the changing and setting of parameters within the phone database, from both the User and Admin menus.
• AGP Phonelet
Deals with AGP Phonelet.
• Application framework
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All applications within the phone, e.g. Call view, Call log, or Phonebook, are run within the application framework. It is responsible for the switching between different applications and bringing them into and out of focus as appropriate.
• Application menu
This is where applications to be run on the phone can be started and stopped.
• Bluetooth service
Handles the Bluetooth interactions between external Bluetooth devices and the phone.
Bluetooth is available only on OpenStage 60/80 phones.
• Call log
The Call log application displays the call history of the phone.
• Call view
Handles the representation of telephony calls on the phone screen.
• Certificate management
Handles the verification and exchange of certificates for security and verification purposes.
• Clock service
Handles the phone’s time and date, including daylight saving and NTP functionality.
• Communications
Involved in the passing of call related information and signaling to and from the CSTA service.
• Component registrar
Handles data relating to the type of phone, e.g. OpenStage 20/40 HFA/SIP, OpenStage
60/80 HFA/SIP.
• CSTA service
Any CSTA messages are handled by this service. CSTA messages are used within the phone by all services as a common call progression and control protocol.
• Data Access service
Allows other services to access the data held within the phone database.
• Desktop
Responsible for the shared parts of the phone display. Primarily these are the status bar at the top of the screen and the FPK labels.
• Digit analysis service
Analyses and modifies digit streams which are sent to and received by the phone, e.g. canonical conversion.
• Directory service
Performs a look up for data in the phonebook, trying to match incoming and outgoing numbers with entries in the phonebook.
• DLS client management
Handles interactions with the DLS (Deployment Service).
• Health service
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Monitors other components of the phone for diagnostic purposes and provides a logging interface for the services in the phone.
• Help
Handles the help function.
• HTTP Service
Handles the HTTP Service messages.
• Instrumentation service
Used by the Husim phone tester to exchange data with the phone for remote control, testing and monitoring purposes.
• Journal service
Responsible for saving and retrieving call history information, which is used by the Call log application.
• Media control service
Provides the control of media streams (voice, tones, ringing etc. ) within the phone.
• Media processing service
This is a layer of software between the media control service, the tone generation, and voice engine services. It is also involved in the switching of audio devices such as the handset and loudspeaker.
• Media recording service
Logs the data flow generated with call recording.
• Mobility service
Handles the mobility feature whereby users can log onto different phones and have them configured to their own profile.
• OBEX service
Involved with Bluetooth accesses to the phone.
Bluetooth is available only on OpenStage 60/80 phones.
• OpenStage client management
Provides a means by which other services within the phone can interact with the database.
• Password management service
Verifies passwords used in the phone.
• Phonebook
Responsible for the phonebook application.
• Performance Marks
Aid for measuring the performance of the phone. For events triggered by the user, a performance mark is written to the trace file, together with a timestamp in the format hh:mm:ss yyyy.milliseconds, and information about the event. The timespan between two performance marks is an indicator for the performance of the phone.
> The trace level must be set to "TRACE" or "DEBUG".
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• Physical interface service
Handles any interactions with the phone via the keypad, mode keys, fixed feature buttons, clickwheel and slider.
• Security Log Service
Handles Security Log Service messages.
• Service framework
This is the environment within which other phone services operate. It is involved in the starting and stopping of services.
• Service registry
Keeps a record of all services currently running inside the phone.
• Sidecar service
Handles interactions between the phone and any attached sidecars.
• SIP call control
Contains the call model for the phone and is associated with telephony and call handling.
• SIP messages
Traces the SIP messages exchanged by the phone.
> After changing the level for the tracing of SIP messages, the phone must be rebooted. Otherwise the changes would have no effect.
• SIP signalling
Involved in the creation and parsing of SIP messages. This service communicates directly with the SIP stack.
• Team service
Primarily concerned with keyset operation.
• Tone generation service
Handles the generation of the tones and ringers on the phone.
• Transport service
Provides the IP (LAN) interface between the phone and the outside world.
• USB backup service
Used to make backup/restore to/from USB stick by using password. This item is available in the phone GUI.
• vCard parser service
Handles parsing and identification of VCard information while sending or getting VCards via Bluetooth.
• Voice engine service
Provides a switching mechanism for voice streams within the phone. This component is also involved in QDC, Music on hold and voice instrumentation.
• Voice mail
Handles the voice mail functionality.
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• Web server service
Provides access to the phone via web browser.
• 802.1x service
Provides authentication to devices attached to a LAN port, establishing a point-to-point connection or preventing access from that port if authentication fails. The service is used for certain closed wireless access points.
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Administration via WBM
Diagnostics > Fault trace configuration
File size (bytes)
Fault trace configuration
65536 Trace timeout (minutes) 0
Trace levels for components
Administration
Application menu
Call Log
Certificate management
Component registrar
Data Access service
Digit analysis service
DLS client management
Help
Java
Media control service
Mobility service
OpenStage client management
Pot service
Physical interface service
Service registry
SIP call control
SIP signalling
Tone geberation service vCard parser service
Voice mail
USB backup service
802.1x service
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
Application framework
Bluetooth service
Call View
Communications
CSTA service
Desktop
Directory service
Health service
Instrumentation service
Journal service
Media processing service
OBEX service
Phonebook
Password management service
Service framework
Sidecar service
SIP messages
Team service
Transport service
Voice engine service
Web server service
Video service engine
Clock Service
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
SIP messaging traces are enabled after reboot
Download trace file
Download syslog file
Download saved trace file
Download old syslog file
Download upgrade error file Download HPT remote service log file
Submit
Download upgrade trace file
Download saved syslog file
Download dial plan file
Automatic clear before start
Download old trace file
Download Database file
Reset
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Administration
Diagnostics
3.26.10
EasyTrace Profiles
In order to simplify tracing for a specific problem, the tracing levels can be adjusted using predefined settings. The EasyTrace profiles provide settings for a specific area, e. g. call connection. On pressing Submit, those pre-defined settings are sent to the phone. If desired, the settings can be modified anytime using the general mask for trace configuration under
Diagnostics > Fault Trace Configuration (see Section 3.26.9, “Fault Trace Configuration”).
If desired, the tracing for all services can be disabled (see Section 3.26.10.26, “No Tracing for
All Services”).
The following sections describe the EasyTrace profiles available for the phone.
3.26.10.1
Bluetooth Handsfree
Diagnostics > EasyTrace Profiles > Bluetooth handsfree profile
Bluetooth handsfree profile
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Data Access service
Media control service
OpenStage client management
Physical interface service
Voice engine service
Media processing service
Bluetooth service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
DEBUG
TRACE
TRACE
TRACE
Download saved trace file
Reset
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3.26.10.2
Bluetooth Headset
Diagnostics > EasyTrace Profiles > Bluetooth headset profile
Bluetooth headset profile
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Data Access service
Media control service
OpenStage client management
Voice engine service
Media processing service
Bluetooth service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset
3.26.10.3
Call Connection
Diagnostics > EasyTrace Profiles > Call connection
Call connection
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Service registry
SIP signalling
SIP call controll
Call View
Communications
CSTA service
SIP messages
Download trace file
Submit
TRACE
LOG
TRACE
DEBUG
DEBUG
TRACE
TRACE
TRACE
DEBUG
Download saved trace file
Reset
> This EasyTrace profile contains the tracing of SIP messages. Please note that after changing the level for the tracing of SIP messages, the phone must be rebooted.
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3.26.10.4
Call Log
Diagnostics > EasyTrace Profiles > Call log problems
Call log problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Call Log
Component registrar
Health service
Application framework
Desktop
Journal service
Download trace file
Submit
1048576
0
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset
3.26.10.5
Call Recording
Diagnostics > EasyTrace Profiles > Call recording
Call recording
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Call View
Communications
SIP call control
Media recording service
Download trace file
Submit
DEBUG
DEBUG
DEBUG
DEBUG
Download saved trace file
Reset
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3.26.10.6
DAS Connection
Diagnostics > EasyTrace Profiles > DAS connection
DAS connection
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
1048576
0
Certificate management
Component registrar
Health service
DLS client management
Service framework
Download trace file
Submit
LOG
TRACE
LOG
TRACE
TRACE
Download saved trace file
Reset
3.26.10.7
DLS Data Errors
Diagnostics > EasyTrace Profiles > DLS data errors
DLS data errors
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Certificate management
Component registrar
Data Access service
Health service
DLS client management
OpenStage client management
Service framework
Download trace file
Submit
1048576
0
LOG
TRACE
TRACE
LOG
TRACE
LOG
TRACE
Download saved trace file
Reset administration.fm
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Diagnostics
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3.26.10.8
Help Application
Diagnostics > EasyTrace Profiles > Help application problems
Help application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Application menu
Component registrar
Health service
Application framework
Help
Web server service
Download trace file
Submit
LOG
TRACE
LOG
TRACE
DEBUG
TRACE
Download saved trace file
Reset
3.26.10.9
Key Input
Diagnostics > EasyTrace Profiles > Key input problems
Key input problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Physical interface service
Download trace file
Submit
TRACE
LOG
DEBUG
Download saved trace file
Reset
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3.26.10.10 LAN Connectivity
Diagnostics > EasyTrace Profiles > LAN connectivity problems
LAN connectivity problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Transport service
Download trace file
Submit
TRACE
LOG
TRACE
Download saved trace file
Reset
3.26.10.11 Messaging
Diagnostics > EasyTrace Profiles > Messaging application problems
Messaging application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Application framework
Call View
Communications
CSTA service
Desktop
SIP signalling
Download trace file
Submit
TRACE
LOG
TRACE
TRACE
TRACE
TRACE
TRACE
DEBUG
Download saved trace file
Reset administration.fm
Administration
Diagnostics
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3.26.10.12 Mobility
Diagnostics > EasyTrace Profiles > Mobility problems
Mobility problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Administration
Data Access service
DLS client management
Mobility service
Download trace file
Submit
1048576
0
TRACE
TRACE
LOG
TRACE
Download saved trace file
Reset
3.26.10.13 Phone administration
Diagnostics > EasyTrace Profiles > Phone administration problems
Phone administration problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Administration
Health service
OpenStage client management
Application framework
Communications
CSTA service
Desktop
Download trace file
Submit
TRACE
WARNING
LOG
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
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3.26.10.14 LDAP Phonebook
Diagnostics > EasyTrace Profiles > Phonebook (LDAP) problems
Phonebook (LDAP) problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Application menu
Component registrar
Directory service
Health service
Application framework
Desktop
Journal service
Transport service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
LOG
Download saved trace file
Reset
3.26.10.15 Local Phonebook
Diagnostics > EasyTrace Profiles > Phonebook (local) problems
Phonebook (local) problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Application menu
Component registrar
Health service
Application framework
Desktop
Journal service
Download trace file
Submit
LOG
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset administration.fm
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Diagnostics
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3.26.10.16 Server based applications
Diagnostics > EasyTrace Profiles > Server based application problems
Server based application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
AGP Phonelet
Download trace file
Submit
LOG
Download saved trace file
Reset
3.26.10.17 Sidecar
Diagnostics > EasyTrace Profiles > Sidecar problems
Sidecar problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Sidecar service
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
Download saved trace file
Reset
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3.26.10.18 SIP standard multiline
Diagnostics > EasyTrace Profiles > SIP standard multiline
SIP standard multiline
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Call View
Communications
CSTA service
Team Service
SIP signalling
SIP call control
SIP messages
Download trace file
Submit
1048576
0
TRACE
WARNING
LOG
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
3.26.10.19 SIP standard singleline
Diagnostics > EasyTrace Profiles > SIP standard singleline
SIP standard singleline
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Call View
Communications
CSTA service
SIP signalling
SIP call control
SIP messages
Download trace file
Submit
TRACE
LOG
TRACE
TRACE
DEBUG
DEBUG
Download saved trace file
Reset administration.fm
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3.26.10.20 Speech
Diagnostics > EasyTrace Profiles > Speech problems
Speech problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Voice engine service
Media processing service
SIP signalling
SIP call control
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
TRACE
DEBUG
DEBUG
Download saved trace file
Reset
3.26.10.21 Tone
Diagnostics > EasyTrace Profiles > Tone problems
Tone problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Tone generation service
Media processing service
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
TRACE
Download saved trace file
Reset
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3.26.10.22 USB Backup/Restore
Diagnostics > EasyTrace Profiles > USB backup/restore
USB backup/restore
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
1048576
0
Administration
Component registrar
Physical interface service
USB backup service
Download trace file
Submit
TRACE
TRACE
DEBUG
DEBUG
Download saved trace file
Reset
3.26.10.23 Voice Dialling
Diagnostics > EasyTrace Profiles > Voice recognition problems
Voice recognition problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Media control service
Voice engine service
Call View
Media processing service
Voice recognition
Phonebook
Download trace file
Submit
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset administration.fm
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Diagnostics
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3.26.10.24 Web Based Management
Diagnostics > EasyTrace Profiles > Web based management
Web based management
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Data Access service
OpenStage client management
Web server service
USB backup service
802.1x service
Voice recognition
Download trace file
Submit
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
3.26.10.25 802.1x problems
Diagnostics > EasyTrace Profiles > 802.1x problems
802.1x problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Certificate management
Component registrar
Data Access service
Download trace file
Submit
1048576
0
LOG
TRACE
TRACE
Download saved trace file
Reset
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3.26.10.26 No Tracing for All Services
Diagnostics > EasyTrace Profiles > Clear all profiles
Clear all profiles
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Administration
Call Log
Call View
Phonebook
Help
Application menu
Certificate management
Communications
Component registrar
CSTA service
Data Access service
Digit analysis service
Digital data service
Directory service
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
DLS client management
Health service
Instrumentation service
Journal service
Media control service
Media processing service
Mobility service
OBEX service
OpenStage client management
Performance Marks
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
Password management service
Physical interface service
Sidecar service
Team service
Tone generation service
Transport service
Voice engine service
Web server service
SIP signalling
SIP call control
SIP messages
Application framework
Desktop
AGP Phonelet
Service framework
Service registry
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
Bluetooth service vCard parser service
Voice mail
OFF
OFF
OFF
USB backup service OFF
802.1x service
Voice recognition
Download trace file
OFF
OFF
Download saved trace file
Submit Reset
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3.26.11
Bluetooth Advanced Traces
For OpenStage 60/80 phones, low level Bluetooth traces can be controlled and viewed via web
interface, in addition to the tracing facilities available in previous firmware versions (see Section
3.26.9, “Fault Trace Configuration”). Internally, the phone uses the hcdump utility for creating
the traces. It is also possible to run the trace from the shell via SSH (for information about the
SSH access, please refer to Section 3.24, “SSH – Secure Shell Access”).
If Automatic clear before start is enabled, the log file will be emptied before the Start button is pressed, so that the log file will only containd newly created entries. By default, this parameter is enabled.
The File size (Max 6290000 bytes) parameter determines the maximum size of the log file. If this value is exceeded, no more data will be written to the file. The default value is 265536.
If Extended dump is enabled, all hexadecimal and ASCII data is displayed for each packet. If disabled, only the packet type is displayed. By default, this parameter is enabled.
If Verbose decoding is enabled, the packets are decoded in a more verbose way. By default, this parameter is enabled.
With the Start/Stop button, tracing is started or halted. The label depends on whether tracing is active or not.
On clicking the Download trace file link, the trace file is displayed.
With Submit, the changes on the parameters described above are sent to the phone.
With Reset, parameter changes that have been made in the form, but not yet sent to the phone, are cancelled.
Administration via WBM
Bluetooth Advanced Traces
Automatic clear before start
;
File size (Max 6290000 bytes)
Extended dump
Verbose decoding
256000
;
;
Tracing is stopped
Start
Download trace file
Submit Reset
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3.26.12
QoS Reports
3.26.12.1
Conditions and Thresholds for Report Generation
> For details about the functionality, please refer to the release notes.
The generation of QoS (Quality of Service) reports which are sent to a QCU server (see Section
3.3.9, “SNMP”) is configured here.
Data required
• Report mode: Sets the conditions for generating a QoS report.
Value range:
• "OFF": No reports are generated.
• "EOS Threshold exceeded": Default value. A report is created if a) a telephone conversation longer than the Minimum session length has just ended, and b) a threshold value has been exceeded during the conversation.
• "EOR Threshold exceeded": A report is created if a) the report interval has just passed, and b) a threshold value has been exceeded during the observation interval.
• "EOS (End of Session)": A report is created if a telephone conversation longer than the Minimum session length has just ended.
• "EOR (End of Report Interval)": A report is created if the report interval has just passed.
• Report interval (seconds): Time interval between the periodical observations.
Default: 60
• Observation interval (seconds): During this time interval, the traffic is observed.
Value: 10
• Minimum session length (100 millisecond units): When the Report mode is set to "EOS
Threshold exceeded" or "EOS (End of Session)", a report can be created only if the duration of the conversation exceeds this value.
Default: 20
• Maximum jitter (milliseconds): When the jitter exceeds this value, a report is generated.
Default: 20
• Average round trip delay (milliseconds): When the average round trip time exceeds this value, a report is generated.
Default: 100
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Non-compressing/ Compressing codecs threshold values:
• Lost packets (per 1000 packets): When the number of lost packets exceeds this maximum value during the observation interval, a report is created.
Default: 10
• Consecutive lost packets: When the number of lost packets following one another exceeds this maximum value during the observation interval, a report is created.
Default: 2
• Consecutive good packets: When the number of good packets following one another falls below this minimum value, a report is created.
Default: 8
• Resend last report: If checked, the previous report is sent once again on pressing Sub-
Submit.
Value range: "Yes", "No"
Default: "No"
The transmission of report data can be triggered manually by pressing Send now in the local menu.
Administration via WBM
Diagnostics > QoS Reports > Generation
Generation
Report mode
Report interval (seconds)
Observation interval (seconds)
Minimum session length (100 millisecond units)
Codec independent threshold values
Maximum jitter (milliseconds)
Average round trip delay (milliseconds)
Non-compressing codec threshold values
Lost packets (per 1000 packets)
Consecutive lost packets
Consecutive good packets
Compressing codec threshold values
Lost packets (per 1000 packets)
Consecutive lost packets
Consecutive good packets
Resend last report
Submit
EOS Threshold exceeded
60
10
20
20
100
10
2
8
10
2
8
Reset
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Administration via Local Phone
|
---
Admin
|
---
Network
|
---
QoS
|
---
Reports
|---
Generation
| |---
Mode
| |---
Report interval
| |---
Observe interval
|
|
---
Minimum session length
|---
Send now
|
---
Thresholds
|---
Maximum jitter
|---
Round-trip delay
|---
Non-compressing:
|---
...Lost packets (K)
|---
...Lost consecutive
|---
...Good consecutive
|---
Compressing:
|---
...Lost packets (K)
|
|---
...Lost consecutive
---
...Good consecutive administration.fm
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3.26.12.2
View Report
OpenStage phones generate QoS reports using a HiPath specific format, QDC (QoS Data Collection). The reports created for the last 6 sessions, i. e. conversations, can be viewed on the
WBM.
To enable the generation of reports, please ensure that:
•
the switch QoS traps to QCU (System > SNMP) is activated (see Section 3.3.9, “SNMP”);
•
the conditions for the generation of reports are set adequately (see Section 3.26.12.1,
“Conditions and Thresholds for Report Generation”).
For details about QoS reports on OpenScape devices, see the HiPath QoS Data Collection V
1.0 Service Manual.
A QoS report contains the following data:
• Start of report period - seconds: NTP time in seconds for the start of the report period.
• Start of report period - fraction of seconds: Additional split seconds to be added to the seconds for an exact start time.
• End of report period - seconds: NTP time in seconds for the end of the report period.
• End of report period - fraction of seconds: Additional split seconds to be added to the seconds for an exact end time.
• SNMP specific trap type: The trap type is a 5 bit value calculated from a list of thresholdexceeding bits. Every time a threshold is exceeded, the associated bit is set, otherwise it is cleared.
The trace type bits are defined as follows:
• Bit 0: Jitter threshold was exceeded.
• Bit 1: Delay threshold was exceeded.
• Bit 2: Threshold for lost packets was exceeded.
• Bit 3: Threshold for consecutive lost packets was exceeded.
• Bit 4: Threshold for consecutive good packets was exceeded.
• IP address (local): IP address of the local phone.
• Port number (local): RTP receiving port of the local phone.
• IP address (remote): IP address of the remote phone that took part in the session.
• Port number (remote): RTP sending port of the local phone.
• SSRC (receiving): RTP Source Synchronization Identifier of the local phone.
• SSRC (sending): RTP Source Synchronization Identifier of the remote phone.
• Codec: Number of the Payload Type applied in the session; see RFC 3551 (Table 4 and
5).
• Maximum packet size: Maximum size (in ms) of packets received during the report interval.
• Silence suppression: Number of silence suppression activation objects found in the RTP stream received. A silence suppression activation object is defined as a period of silence when no encoded voice signals were transmitted by the sender.
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• Count of good packets: Total amount of good packets.
• Maximum jitter: Maximum jitter (in ms) found during the report interval.
• Maximum inter-arrival jitter: Maximum of the interarrival jitter values (in ms). The interarrival jitter is the smoothed absolute value of the jitter measurements. It is calculated continuously. For details about the calulation, see RFC 3550.
• Periods jitter threshold exceeded: Number of observation intervals in which the threshold for maximum jitter was exceeded.
• Round trip delay: Average value of delay calculated for each RTCP packet. The first value is available after about 15 sec.
• Round trip delay threshold exceeded: Set to "true" if the average round trip delay threshold value was exceeded in the report interval.
• Count of lost packets: Number of packets lost in the course of speech decoding.
• Count of discarded packets: Number of the packets discarded without transferring the contents.
• Periods of lost packets: Number of observation intervals in which the threshold for lost packets was exceeded.
• Consecutive packet loss (CPL): List of sequences consecutive packets that were all lost, grouped according to the amount of packets per sequence. The first number in the list counts single lost packets, the second number counts sequences of two lost packets, and so on. The last number counts sequences of more than 10 lost packets.
• Periods of consecutive lost packets: Number of observation intervals in which the threshold for consecutive lost packets was exceeded.
• Consecutive good packets (CGP): List of sequences consecutive packets that were all processed, grouped according to the amount of packets per sequence. The first number in the list counts single good packets, the second number counts sequences of two good packets, and so on. The last number counts sequences of more than 10 good packets. All values are reset to 0 after an interval without packet loss.
• Periods of consecutive good packets: Number of intervals in which the count of lost packets went below the threshold.
• Count of jitter buffer overruns: Number of packets rejected because the jitter buffer was full.
• Count of jitter buffer under-runs: Increased by one whenever the decoder requests new information on decoding and finds an empty jitter buffer.
• Codec change on the fly: The value is 1, if there has been a codec or SSRC change during the observation period, and 0, if there has been no change.
• Periods with at least one threshold exceeded: Number of observation intervals with at least one threshold exceedance. If there is no data, the value is 255. The threshold values included are:
• maximum jitter;
• lost packets;
• consecutive lost packets;
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• consecutive good packets.
• HiPath Switch ID: Unique number identifying the HiPath switch to which the endpoints are assigned.
• LTU number: In HiPath 4000 only, the shelf identification is taken from the shelf containing a gateway.
• Slot number: The slot number where the phoneis connected in the shelf.
• Endpoint type: Type of the local phone.
• Version: Software version of the local phone.
• Subscriber number type: Type of subscriber number assigned to the local phone. The possible types are:
• 1: local number, extension only
• 2: called number, network call
• 3: E.164 number of the local phone
• Subscriber number: Subscriber number of the local phone.
• Call ID: SIP call id.
• MAC address: MAC address of the local phone.
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Data viewing via WBM
Diagnostics > QoS reports > View Session Data
View Session Data
Select a report to view
Submit
Start of report period - seconds
End of report period - seconds
SNMP specific trap type
IP address (local)
Port number (local)
IP address (remote)
Port number (remote)
SSRC (receiving)
SSRC (sending)
Codec
Maximum packet size
Silence suppression
Count of good packets
Maximum jitter
Maximum inter-arrival jitter
Periods jitter threshold exceeded
Round trip delay
Round trip delay threshold exceeded
Count of lost packets
Count of discarded packets
Periods of lost packets
Consecutive packet loss (CPL)
Periods of consecutive lost packets
Consecutive good packets (CGP)
Periods of consecutive good packets
Count of jitter buffer overruns
Count of jitter buffer under-runs
Codec change on the fly
Periods with at least one threshold exceeded
HiPath Switch ID
LTU number
Slot number
Endpoint type
Version
Subscriber number type
Subscriber number
Call ID
MAC address
QoS Statistics 1
2011/10/16 21:51:29 UTC
2011/10/16 21:56:36 UTC
2
192.168.1.235
5012
192.168.1.202
5010
1481715715
3244864262
G.711 PCMU
20
0
15203
2
0
0
433
0
0
0
255,255,255,255,255,255,255,255,255,255,255
255
255,255,255,255,255,255,255,255,255,255,255
255
0
0
0
Asterisk PBX 1.6.2.19
255
255
OpenStage 80
V3 R0.50.0 SIP 110924
0
3339
05b4445aeaf00008
0001e325eaca administration.fm
Administration
Diagnostics
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Diagnostics
3.26.13
Core dump
If Enable core dump is checked, a core dump will be initiated in case of a severe error. The core dump will be saved to a file. By default, this function is activated.
If Delete core dump is activated, the current core dump file is deleted on Submit. By default, this is not activated.
If one or more core dump file exist, hyperlinks for downloading will be created automatically.
Administration via WBM
Diagnostics > Miscellaneous > Core Dump
Core Dump
Enable core dump*
Delete core dump
;
*Changes to this item do not take effect until the phone is restarted
Submit Reset
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Diagnostics
3.26.14
Remote Tracing – Syslog
All trace messages created by the components of the phone software can be sent to a remote server using the syslog protocol. This is helpful especially for long-term observations with a greater number of phones.
To enable remote tracing, Remote trace status must be set to "Enabled". Furthermore, the IP address of the server receiving the syslog messages must be entered in Remote ip, and the corresponding server port must be given in Remote port.
With version V2, the User notification parameter controls whether the user is notified about the remote tracing or not. If user notification is enabled, a blinking symbol ( on OpenStage
60/80; on OpenStage 15/20/40) will inform the user when remote tracing is active, that is, when Remote trace status is set to "Enabled".
Administration via Local Phone
|
---
Admin
|
---
Maintenance
|
---
Remote trace
|---
Remote trace status
|---
User notification
|
|---
Remote ip
---
Remote port
Administration via WBM
Remote trace
Remote Trace Status
Use Notification
Disabled
Enabled
Remote Server
Remote Server Port 514
Submit Reset
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Diagnostics
3.26.15
HPT Interface (For Service Staff)
For special diagnosis and maintenance tasks, the service staff may employ the HPT tool, which is able to control and observe an OpenStage phone remotely. For security reasons, this tool
can only be used when a dongle key file is uploaded to the phone (see Section 3.16.10, “Dongle
Key”). This key is accessable to the service staff only. It is specific for a particular SIP firmware
version, but it will also be valid for previous versions.
There are 2 types of HPT sessions, control session and observation session.
A control session allows for activating phone functions remotely. When a control session is established, the following changes will occur:
• The display shows a message indicating that remote service is active.
• Handset, microphone, speaker, headset, and microphone are disabled.
An observation session allows for supervising events on the phone, like, for instance, pressing a key, incoming calls or navigating in the menus. Before an observation session is started, the user is prompted for allowing the observation. During an observation session, the phone operates normally, including loudspeaker, microphone and ringer. Thus, the local user can demonstrate an error towards the service staff that is connected via HPT.
The HPT interface is enabled by downloading the dongle key file to the phone (see Section
3.16.10, “Dongle Key”). It can be disabled via local menu or WBM. Thereby, the dongle key file
is deleted. To enable the HPT interface again, the file must be downloaded anew.
The session data is written to a log file on the phone. It can be downloaded from the Diagnostics
> Fault trace configuration menu (see Section 3.26.9, “Fault Trace Configuration”).
Administration via WBM (Disable)
Maintenance > HPT interface
HPT interface
Disable HPT
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Bluetooth (OpenStage 60/80)
3.27
Bluetooth (OpenStage 60/80)
The Bluetooth interface can be enabled or disabled in the admin menu. By default, it is enabled.
If Bluetooth is enabled, the user has the possibility to activate or deactivate it via the user menu.
> This parameter can also be configured under System > Features > Feature access
> Bluetooth (see Section 3.7, “Feature Configuration”) or Bluetooth > Enable Blue-
tooth interface.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Bluetooth (OpenStage 60/80)
Administration via Local Phone
Bluetooth can be enabled or disabled via the local admin menu:
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
Bluetooth
|
---
Enable or
|
---
Admin
|
---
System
|
---
Features
|
---
Feature Access
|
---
Services
|
---
Bluetooth
|
---
Allow
|
---
Disallow
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Administration
MWI LED
3.28
MWI LED
This configurable item is added to the Administrator settings to allow the Administrator to control how new VoiceMails are indicated to the user; via the "Envelope" mode key LED only, via the Top LED only or via both LEDs.
The selection field offers the choice between:
• "Key only" (default)
• "Key & AlertBar"
• "AlertBar only"
Default setting for OpenStage 40 US is "AlertBar only". After a factory reset, the system will be reset to this value.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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MWI LED
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
MWI LED
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Missed Call LED
3.29
Missed Call LED
This configurable item is added to the Administrator settings to allow the Administrator to control how new Missed Calls are indicated to the user; via the "Envelope" mode key LED only, via the Top LED only, via both LEDs or no LED.
The selection field offers the choice between:
• "Key only" (default)
• "Key & AlertBar"
• "AlertBar only"
• "No LED"
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
5
On
;
Prompt
Beep
;
;
;
;
;
Disabled
Off
Reset
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Administration
Impact Level Notification
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
Missed call LED
3.30
Impact Level Notification
Communications for the Public Sector Network (PSN) is seen as originating from or terminating to zones with differing 'impact' levels (the impact level indicates how the phone user should handle the call conversation). The purpose is to notify the OpenStage phone users when they are connecting or in a call where another party in the call is in a lower Impact Level (IL) zone.
This feature uses a UI mechanism to notify/remind the phone user that the call may require special treatment. This involves special icons, text indications, and special audio (ringer or tone as appropriate). There are no restrictions on call handling as a result of any special status for the call.
Thus the Impact Level Notification feature only involves UI changes that are triggered by receiving new SIP headers and affects the following:
• Prompts presented to alert for incoming calls
• Prompts presented to monitor progress for outgoing calls
• Connected call displays
• Call scenarios involving multiple calls
• Retrieving a held call
However, since there are no call restrictions explicit for the Impact Level Notification feature the solution needs to consider some additional scenarios:
• Group pickup
• Directed pickup
• Callback
• CTI action
• Shared lines on a Keyset
This feature cannot be turned off at the phone since it is driven solely by the OSV.
The OSV is responsible for being aware of the impact level of the phone (the phone does not have control of its own level) and the impact levels of all other endpoints that are participating in a call with the phone. The OSV uses this information to signal (via a new SIP header) the phone when the call is to be treated as from a lower impact level. It does this during the start of a call or anytime during a call.
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Impact Level Notification
Data required
• Impact level ringer: Identifies one of the named distinctive ringers to be used in place of the normal ringer for calls from a lower impact level.
Value range: the offered values are those definded in "Ringer Settings" > Distinctive", e.g.
"Bellcore-dr1" or any arbitrary name
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
No action
;
30
5
On
;
Prompt
Beep
;
;
;
; alert-internal
None
;
Disabled
Off
Reset
The phone plays the configured Impact Level Notification ringer when the call is from a lower
impact level. The ringer has to be configured in the ringer setting table (see Section 3.14, “Ringer Setting”).
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Impact Level Notification
Administration via Local Phone
|
---
Admin
|
---
System
|
---
Features
|
---
Configuration
|
---
Audio
|
---
Lower IL ringer
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Technical Reference
Menus
4
4.1
Technical Reference
Menus
> This section describes the structure of the administration menus of the OpenStage phone. For information on user menus, please refer to the user manual.
4.1.1
Web Interface Menu
4.1.1.1
Admin Login
Menu Structure
Applications (OpenStage 60/80)
XML applications
Add application
Modify/Delete application
Xpressions
Add messages application
Add messages applicationAdd phonebook application
Add call log application
Add help application
Bluetooth
Network
General IP configuration
IPv4 configuration
IPv6 configuration
Update Service (DLS)
QoS
Port configuration
LLDP-MED operation
System
System Identity
SIP interface
Registration
SNMP
Features
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Technical Reference
Menus
Configuration
DSS settings
Program keys > Line (V2 on OpenStage 15/40/60/80)
Key Module 1
Key Module 2
Fixed keys
Keyset operation
Addressing
Call completion
Feature access
Security
System
SDES Config
Access control
Logging
Faults
File transfer
Defaults
Phone application
Hold music
Picture Clip (OpenStage 60/80)
LDAP (OpenStage 60/80)
Logo (OpenStage 40/60/80)
Screensaver (OpenStage 60/80)
Ringer file
Dongle key
Local functions
Directory settings
Messages settings
Locality
Canonical dial settings
Canonical dial lookup
Canonical dial
Phone location
Energy saving
Call logging
Date and time
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Speech
Codec preferences
Audio settingsAudio settings
General information
Security and Policies
Password
Generic policy
Admin policy
User policy
Character set
Change Admin password
Change User password
Certificates
Generic
Authentication policy
Ringer Setting
Distinctive
Map To Specials
Mobility
Diagnostics
Diagnostic information
View
User acess
LLDP-MED TLVs
Fault trace configuration
EasyTrace Profiles
Bluetooth handsfree profile (OpenStage 60/80)
Bluetooth headset profile (OpenStage 60/80)
Call connection
Call log problems
Call Recording
DAS connection
DLS data errors
Help application problems
Key input problems
LAN connectivity problems
Messaging application problems
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Mobility problems
Phone administration problems
Phonebook (LDAP) problems (OpenStage 60/80)
Phonebook (local) problems (OpenStage 60/80)
Server based application problems (OpenStage 60/80)
Sidecar problems
SIP standard multiline
SIP standard singleline
Speech problems
Tone problems
USB backup/restore
Voice recognition problems (OpenStage 60/80)
Web based management
802.1x problems
Clear all profiles
Bluetooth Advanced Traces
QoS Reports
Generation
View Session Data
Miscellaneous
IP tests
Memory information
Core Dump
Maintenance
Remote trace
Restart Phone
Factory reset
HPT interface
Secure Shell
Diagnostic call
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Technical Reference
Menus
4.1.1.2
Web Pages
Admin Login
Admin Login
Enter Admin password:
Login Reset
Add application
Add application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
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Menus
Modify/Delete application
Modify/Delete application
Select application
Modify testxml
Delete
Settings
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Mode key
Submit testxml testxml
192.168.1.150
8080 http testxml/servlet
;
No
No
0
0
Reset
Xpressions
Xpessions
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs Start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit
Xpressions
Xpressions https
No
No
3
Voice Mail
Xpressions
Inbox
XprInbox
Outbox
Xproutbox
Delete
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Add messages application
Add messages application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs Start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
XML Phonebook (up to V2R0)
XML Phonebook
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit
XMLPhonebook
XMLPhonebook http
Yes
Yes
0
Reset
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Menus
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Menus
Add phonebook application
Add phonebook application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs Start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
Add call log application
Add call log application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs Start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
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Add help application
Add help application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
Program name on server
Auto start
Use proxy
XML Trace enabled
Debug program on server
Number of tabs
All tabs Start
Tab 1 Display Name
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit http
Yes
Yes
0
Reset
Bluetooth
Bluetooth
Enable Bluetooth interface
Submit
;
Reset
General IP configuration
General IP configuration
Protocol Mode
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
VLAN ID
DNS domain
Primary DNS
Secondary DNS
HTTP proxy
Submit
IPv4_IPv6
;
Manual
192.168.1.105
192.168.1.2
Reset
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Menus
IPv4 configuration
IPv4 configuration
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
IP address
Subnet mask
Default route
Route 1 IP address
Route 1 gateway
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit
;
192.168.1.235
255.255.255.0
192.168.1.2
Reset
IPv6 configuration
IPv6 configuration
LLDP-MED Enabled
DHCPv6 Enabled
DHCPv6 lease reuse
Global Address
Global Address Prefix Len
Global Gateway
Link Local Address
Route 1 Dest.
Route 1 Prefix Len
Route 1 Gateway
Route 2 Dest.
Route 2 Prefix Len
Route 2 Gateway
Submit
;
Reset
Update Service (DLS)
Update Service (DLS)
DLS address
DLS port
Contact gap
Revert to default security
Mode
Security PIN
Submit
192.168.1.242
18443
300
Default
Reset
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QoS
QoS
Service
Layer 2
Layer 2 voice
Layer 2 signalling
Layer 2 default
Layer 3
Layer 3 voice
Layer 3 signalling
Layer 3 video
MLPP
Priority
Immediate
Flash
Flash override
Submit
5
3
0
EF
AF31
AF41
;
EF
EF
EF
EF
Reset
Port configuration
Port configuration
SIP Server
SIP registrar
SIP gateway
SIP local
Backup proxy
RTP base
Download server (default)
LDAP server
HTTP proxy
LAN port speed
PC port speed
PC port mode
PC port autoMDIX
Submit
5060
5060
5060
5060
5060
5010
21
389
0
Automatic
Automatic disabled
Reset
LLDP-MED operation
LLDP-MED operation
Time to live (seconds)
Submit
120
Reset
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Menus
System Identity
System Identity
Terminal number
Terminal name
Display identity
Enable ID
Web name
DNS name construction
Submit
4711 openstage
4711
;
Only number
Reset
SIP interface
SIP interface
Outbound proxy
Default OBP domain
SIP transport
Response timer (ms)
NonCall trans. (ms)
Reg. backoff (seconds)
Connectivity check timer (seconds)
Keep alive format
Media Negotiation
Media IP Mode
Submit
;
UDP
32000
32000
60
0
Sequence
Single IP
IPv4
Reset
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Registration
Registration
SIP Addresses
SIP server address
SIP registar address
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
Registration timer (seconds)
Server type
Realm
User ID
Password
MLPP base
MLPP Domain
Other Domain
SIP Survivability
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
Backup OBP flag
Submit
192.168.1.165
192.168.1.165
3600
3600
HiQ8000
Local dsn+uc
;
3600
UDP
Reset referenz.fm
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Menus
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Menus
SNMP
SNMP
Generic traps
Traping sending enabled
Trap destination
Trap destination port
Trap community
Queries allowed
Query password
Diagnistic traps
Diagnostic sending enabled
Diagnostic destination
Diagnostic destination port
Diagnostic community
Diagnostic to generic destination
QoS report traps
QoS traps to QCU
QCU address
QCU port
QCU community
QoS togeneric destination
Submit
162
****
12010
*****
Reset
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Configuration
Configuration
General
Emergency number
Voice Mail number
MWI LED
Missed call LED
Allow refuse
Hot/warm phone
Hot/warm destination
Initial digit timer (seconds)
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
Audio
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
3335
Key & AlertBar
Key only
;
No action
30
;
5
;
;
On
Prompt
;
;
Beep alert-internal
Impact-Level
;
Disabled
Off
Reset
DSS settings
DSS settings
Call pickup detect timer (seconds)
Deflect alerting call enabled
Allow pickup to be refused
Forwarding shown
Submit
3
Reset
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Program keys
Program keys
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Line
Label: Primary Line
Selected dialling
Label: Selected dialling edit edit
1
2
Clear (no feature assigned)
Clear (no feature assigned)
Hold
Label: Hold
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit
3
4
5
6
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Mobility
Label: Mobility
Clear (no feature assigned)
Shift
Label: Shift edit edit edit
7
8
9
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit
Line (V2 on OpenStage 15/40/60/80)
Line
It is recommended that primary lines are only configured on keys 1 to 6.
This ensures compatibility with athe mobility feature, when using devices with 6 or fewer programmable feature keys.
Key label 1
Primary line
Ring on/off
Ring delay (seconds)
Selection order
Address
Realm
User Identifier
Password
Shared type
Allow Overview
Hot warm action
Hot warm destination
Submit
0
0 shared
No Action
Reset
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Key Module 1
Key Module 1
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit
1
2
3
4
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit 5 Clear (no feature assigned)
Clear (no feature assigned) edit 6 Clear (no feature assigned)
Clear (no feature assigned) edit 7 Clear (no feature assigned)
Clear (no feature assigned) edit 8 Clear (no feature assigned)
Clear (no feature assigned) edit 9 Clear (no feature assigned)
Clear (no feature assigned) edit 10 Clear (no feature assigned)
Clear (no feature assigned) edit 11 Clear (no feature assigned)
Clear (no feature assigned) edit 12 Clear (no feature assigned) edit edit edit edit edit edit edit edit edit edit edit edit
Key Module 2
Key Module 2
To assign a new function to a key, select from the drop down list box. To view or modify the Parameters associated with the key, use the Edit button.
Normal Key Shifted
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit
1
2
3
4
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
5
6
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit 7 Clear (no feature assigned)
Clear (no feature assigned) edit 8 Clear (no feature assigned)
Clear (no feature assigned) edit 9 Clear (no feature assigned)
Clear (no feature assigned) edit Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned) edit edit
10
11
12
Clear (no feature assigned)
Clear (no feature assigned) edit edit edit edit edit edit edit edit edit edit edit edit
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Fixed keys
Fixed Keys
To assign a new function to a key, select from the drop down list box. To view or modify the parameters associated with the key, use the
Edit button.
Release key
Forwarding key
Voice recognition key
Built-in release
Built-in forwarding
Send URL
Edit
Edit
Edit
Keyset operation
Keyset operation
Rollover ring
LED on registration
Originating line preference
Terminating line preference
Line action mode
Show focus
Reservation timer (seconds)
Forwarding indicated
Preselect mode
Preselect timer
Preview mode
Preview timer
Bridging priority
Submit alert beep
; idle line ringing line hold
60
; single button
8
Preview overwrites brid-
Reset
Addressing
Addressing
MW server URI
Conference
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code
Submit
192.168.1.2
*0
Reset
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Call completion
Call completion
Functionnal CCSS
Callback ringer
Allow after call (s)
Max. callbacks
Submit
30
5
Reset
Feature access
Feature access
Call control
Blind transfer
3rd call leg
Call establish
Callback
Call pickup
Group pickup
Call deflection
Call forwarding
Do not disturb
Refuse call
Repertory dial key
Ext/int forwarding
;
;
;
;
;
;
;
;
;
;
;
Call associated
Phone book lookups
DSS feature
BLF feature
Line overview
Video calls
CTI
CTI control
Services
Bluetooth
Web based manag.
USB device access
Backup to USB
Feature toggle
Phone lock
Submit
;
;
;
;
;
;
;
;
;
;
;
;
Reset
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System
System
SIP server certificate validation
Use secure calls
SRTP type
Use SRTCP
Submit
SDES Config
SDES config
SDES status
SDP negotiation
SHA1-80 ranking
SHA1-32 ranking
Submit
Disabled
SRTP + RTP
X
X
Reset
MIKEY
Reset
Access control
Access control
CCE access
Factory reset claw
Serial port
Submit
Enable
;
No Password
Reset
Logging
Logging
Max. lines
Archive to DLS
Archive when at
Last archived
Submit
500
50%
20101105-0010
Reset
Faults
Faults
Security log entry
OCSR Failure
Admin access
User access
Cancel faults
Submit
20111009-2206
All
Reset
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Defaults
Defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Submit
FTP
21
Reset
Phone application
Phone application
Use defaults
Download method FTP
FTP Server address
FTP Server port
FTP account
FTP username
21
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit do nothing
Reset
Hold music
Hold music
Use defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit
FTP
21 do nothing
Reset
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Picture Clip
Picture Clip
Use defaults
Download method FTP
FTP Server address
FTP Server port
FTP account
FTP username
21
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit do nothing
Reset
LDAP
LDAP
Use defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit
FTP
21 do nothing
Reset
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Logo
Logo
Use defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit
FTP
21 do nothing
Reset
Screensaver
Screensaver
Use defaults
Download method
FTP Server address
FTP
FTP Server port
FTP account
FTP username
21
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit do nothing
Reset
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Ringer file
Ringer file
Use defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit
FTP
21 do nothing
Reset
Dongle key
Dongle key
Use defaults
Download method FTP
FTP Server address
FTP Server port
FTP account
21
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit do nothing
Reset
Directory settings (OpenStage 40/20/15 )
Directory settings
LDAP Server address
LDAP Server port
Authentication
User name
Password
Submit
389
Anonymous
Reset
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Directory settings
Directory settings
LDAP Server address
Transport
LDAP Secure port
LDAP Server port
Authentication
User Name
Password
Search trigger timeout
TCP
636
389
Anonymous
3
Submit Reset
Messages settings
Messages settings
New items
Alternative label
New urgent items
Alternative label
Old Items
Alternative label
Old urgent items
Alternative label
Show
Show
Show
Show
Submit Reset
Canonical dial settings
Canonical dial settings
Local country code
National prefix digit
Local national code
Minimum local number length
Local enterprise node
PSTN access code
International access code
Operator codes
Emergency numbers
Initial extension digits
Submit
49
0
89
4
723
0
00
1,2,3,4
Reset
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Canonical dial lookup
Local code 1:
Local code 2:
Local code 3:
Local code 4:
Local code 5:
Submit
Canonical dial lockup
Canonical dial
Canonical dial
Internal numbers
External numbers
External access code
International gateway code
Submit
Local enterprise form
Local public form
Not required
Use national code
Reset
International code 1:
International code 2:
International code 3:
International code 4:
International code 5:
Reset
Phone location
Phone location
Phone location
Submit
Signalled
Reset
Energy saving
Energy saving
Backlight time
Submit
2 hours
Reset
Call logging
Call logging
FAC prefixes
Submit Reset
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Date and time
Time source
Date and time
SNTP IP address
Timezone offset (hours)
Daylight saving
Daylight saving
Difference (minutes)
Auto time change
DST zone
Submit
192.43.244.18
1
Europe (Rest)
;
60
Reset
Codec preferences
Codec preferences
Silence suppression
Allow "HD" icon
Packet size Automatic
;
G.711 ranking X
G.729 ranking X
G.722 ranking
Submit Reset
Audio settings
Audio settings
Mute Settings
DTMP playback
Microphone ON-Loudspeaker ON
;
Submit Reset
General information
General information
MAC address: 0001e323f9a1
Software version: 0.7.5.0004-061027
Last restart:
Backlight type 1
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Generic policy
Generic policy
Expires after (days)
Warn before (days)
Force changed
Tries allowed
No change for (hours)
Suspended for (mins)
History valid for (days)
Submit
1
99
5
0
5
0
Reset
Admin policy
Expiry date
Minimum length
Password history
Current status
Submit
Admin policy
2038-01-19T03:14:07+00:00
6
0
Active
Reset
User policy
Expiry date
Minimum length
Password history
Current status
Submit
User policy
2038-01-19T03:14:07+00:00
6
0
Active
Reset
Character set
Character set
Ucase chars reqd.
Lcase chars reqd.
Digits required
Special chars reqd
Bar repeat length
Min char difference
Submit
0
0
0
0
0
0
Reset
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Change Admin password
Change Admin password
Current password
New password
Confirm password
Submit Reset
Change User password
Change User password
Admin password
New password
Confirm password
Submit Reset
Generic
Generic
OCSP check
OCSR 1 address
OCSR 2 address
Submit Reset
Authentication policy
Authentication policy
Secure file transfer
Secure send URL
Secure SIP server
Secure 802.1x
XML Applications
Submit
None
None
None
None
None
Reset
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Distinctive
Name Ringer sound
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3 alert-emerge
Pattern
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Ringer2.mp3
Submit
Distinctive
This page allows you to set up interworking with other IP phone systems that support distinctive ringing
Pattern melody
3
3
3
3
3
3
3
3
3
3
3
3
3
8
3
Pattern sequence
2
2
2
2
2
2
2
2
2
2
2
2
2
1
2
Duration (sec) Audible
60
60
60
60
60
60
60
60
60
60
60
0
60
60
60
Reset
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Ring
Map To Specials
Map To Specials
Internal
External
Recall
Emergency
Special1
Special2
Special3
Bellcore-dr1
Bellcore-dr2
Bellcore-dr3 alert-emerge
Submit Reset
Mobility
Mobility
Unauthorised Logoff Trap
Logoff Trap Delay
Timer Medium Priority
Mobility Feature
Managed Profile
Error Count Local
Submit
300
60
;
0
Reset
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View
View
2011-10-16 20:22:33
00 Terminal number:
01 SIP server:
02 SIP port:
03 SIP registrar:
04 SIP registrar port
05 SIP gateway:
06 SIP gateway port
07 SIP transport:
08 SIP local:
09 Server features:
10 DNS results:
11 Multiline:
12 Registered lines:
13 Backup active:
14 Backup proxy:
15 Use secure calls:
16 SDES status:
17 Secure SIP server:
18 Software version:
19 Display message:
20 Last restart:
21 Memory free:
22 Protocol mode:
23 IP4 address:
24 IP4 subnet mask:
25 IP4 default route:
26 Primary DNS:
27 Secondary DNS:
28 IP4 route 1 IP:
29 IP4 route 1 gateway:
30 IP4 route 1 mask:
31 IP4 route 2 IP:
32 IP4 route 2 gateway:
33 IP4 route 2 mask:
34 IP6 address:
35 IP6 prefix length:
36 IP6 global gateway:
37 IP6 link local addr:
38 IP6 route 1 dest:
39 IP6 route 1 pref len:
40 IP6 route 1 gateway:
41 IP6 route 2 dest:
42 IP6 route 2 pref len:
43 IP6 route 2 gateway:
44 MAC address:
45 Discovery mode:
46 DHCP re-use:
47 DHCPv6:
48 DHCPv6 re-use:
49 LAN port type:
50 PC port status:
51 PC port type:
52 PC port autoMDIX:
53 VLAN ID:
54 QoS Layer 2:
55 QoS Layer 2 voice:
56 QoS Layer 2 signalling:
57 QoS Layer 2 default:
58 QoS Layer 3:
59 QoS Layer 3 voice:
60 QoS Layer 3 signalling:
61 LLDP-MED operation:
62 XML application:
None
63 XML app config:
None
None
None
None
None
None
None
None
None
None
None
None
0001e325eaca
Manual
No
Yes
No
0
None
0
No
None
None
5
None
0
Yes
EF / 46
AF31 / 26
None
3339
192.168.1.230
5060
192.168.1.230
5060
192.168.1.230
5060
UDP
5060
No
5060
No
5060
Yes
192.168.1.148
No
0
0
V3R0.50.0 110924
None
2011-10-10T23:59:01
65733K free
IPv4
192.168.1.235
255.255.255.0
192.168.1.2
192.168.1.105
192.168.1.2
None
None
None
None
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User acess
User access
User Access
Submit
;
Reset
LLDP-MED TLVs
Sent
Sent: Mon Oct 27 10:41:14 2013
Channel ID TLV Data
.ID = 163.165.2.105
TTL TLV Data
.records = 120
System Caps TLV Data
.Supported = Bridge, Telephone
.Enabled = Telephone
LLDP-MED TLV’s
Received
Received: Mon Oct 27 10:41:14 2013
Channel ID TLV Data
.ID = 00:3E:37:01:20:01
TTL TLV Data
.records = 120
System Caps TLV Data
.Supported = Other, Repeater
.Enabled = Other, Repeater
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Fault trace configuration
Fault trace configuration
File size (Max 6290000 bytes)
Trace levels for components
Administration
Application framework
Bluetooth service
Call View
Clock Service
Component registrar
Data Access service
Digit analysis service
DLS client management
Help
Instrumentation service
Media control service
Media recording service
OBEX service
Password management service
Performance Marks
Security Log Service
Service registry
SIP call control
SIP signalling
Tone generation service
USB backup service
Voice engine service
Web server service
1048576
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
DEBUG
OFF
OFF
OFF
OFF
OFF
OFF
Trace timeout (minutes) 0
AGP Phonelet
Application menu
Call Log
Certificate management
Communications
CSTA service
Desktop
Directory service
Health service
HTTP Service
Journal service
Media processing service
Mobility service
OpenStage client management
Phonebook
Physical interface service
Service framework
Sidecar service
SIP messages
Team service
Transport service vCard parser service
Voice mail
802.1x service
OFF
OFF
OFF
OFF
DEBUG
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
DEBUG
DEBUG
OFF
OFF
OFF
OFF
OFF
OFF
Automatic clear before start
Download trace file
Download syslog file
Download exception file
Download dial plan file
SIP messaging traces are enabled after reboot
Download old trace file
Download old syslog file
Download old exception file
Download Database file
Download saved trace file
Download saved syslog file
Download upgrade trace file
Download HPT remote service log file
Submit
Download upgrade error file
Download security log file
Reset
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Bluetooth handsfree profile
Bluetooth handsfree profile
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Data Access service
Media control service
OpenStage client management
Physical interface service
Voice engine service
Media processing service
Bluetooth service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
DEBUG
TRACE
TRACE
TRACE
Download saved trace file
Reset
Bluetooth headset profile
Bluetooth headset profile
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
1048576
0
Component registrar
Data Access service
Media control service
OpenStage client management
Voice engine service
Media processing service
Bluetooth service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset
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Call connection
Call connection
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Service registry
SIP signalling
SIP call controll
Call View
Communications
CSTA service
SIP messages
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
DEBUG
DEBUG
TRACE
TRACE
TRACE
DEBUG
Download saved trace file
Reset
Call log problems
Call log problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Call Log
Component registrar
Health service
Application framework
Desktop
Journal service
Download trace file
Submit
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset
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Call Recording
Call recording
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Call View
Communications
SIP call control
Media recording service
Download trace file
Submit
DEBUG
DEBUG
DEBUG
DEBUG
Download saved trace file
Reset
DAS connection
DAS connection
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Certificate management
Component registrar
Health service
DLS client management
Service framework
Download trace file
Submit
1048576
0
LOG
TRACE
LOG
TRACE
TRACE
Download saved trace file
Reset
DLS data errors
DLS data errors
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Certificate management
Component registrar
Data Access service
Health service
DLS client management
OpenStage client management
Service framework
Download trace file
Submit
LOG
TRACE
TRACE
LOG
TRACE
LOG
TRACE
Download saved trace file
Reset
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Help application problems
Help application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Application menu
Component registrar
Health service
Application framework
Help
Web server service
Download trace file
Submit
LOG
TRACE
LOG
TRACE
DEBUG
TRACE
Download saved trace file
Reset
Key input problems
Key input problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Physical interface service
Download trace file
Submit
1048576
0
TRACE
LOG
DEBUG
Download saved trace file
Reset
LAN connectivity problems
LAN connectivity problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Transport service
HTTP Service
TRACE
LOG
TRACE
TRACE
Download trace file
Submit
Download saved trace file
Reset
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Messaging application problems
Messaging application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Application framework
Call View
Communications
CSTA service
Desktop
SIP signalling
Download trace file
Submit
TRACE
LOG
TRACE
TRACE
TRACE
TRACE
TRACE
DEBUG
Download saved trace file
Reset
Mobility problems
Mobility problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Administration
Data Access service
DLS client management
Mobility service
Download trace file
Submit
1048576
0
TRACE
TRACE
LOG
TRACE
Download saved trace file
Reset
348
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Technical Reference
Menus
Phone administration problems
Phone administration problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Administration
Health service
OpenStage client management
Application framework
Communications
CSTA service
Desktop
Download trace file
Submit
TRACE
WARNING
LOG
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
Phonebook (LDAP) problems
Phonebook (LDAP) problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Application menu
Component registrar
Directory service
Health service
Application framework
Desktop
Journal service
Transport service
Download trace file
Submit
TRACE
TRACE
TRACE
LOG
TRACE
TRACE
TRACE
LOG
Download saved trace file
Reset
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Menus
Phonebook (local) problems
Phonebook (local) problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
1048576
0
Application menu
Component registrar
Health service
Application framework
Desktop
Journal service
Download trace file
Submit
LOG
TRACE
LOG
TRACE
TRACE
TRACE
Download saved trace file
Reset
Server based application problems
Server based application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
AGP Phonelet
Download trace file
Submit
LOG
Download saved trace file
Reset
Sidecar problems
Sidecar problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Component registrar
Health service
Sidecar service
Download trace file
Submit
TRACE
LOG
TRACE
Download saved trace file
Reset
350
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SIP standard multiline
SIP standard multiline
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
1048576
0
Call View
Communications
CSTA service
Team service
SIP signalling
SIP call control
SIP messages
Download trace file
Submit
DEBUG
DEBUG
DEBUG
DEBUG
DEBUG
DEBUG
DEBUG
Download saved trace file
Reset
SIP standard singleline
SIP standard singleline
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Call View
Communications
CSTA service
SIP signalling
SIP call control
SIP messages
Download trace file
Submit
DEBUG
DEBUG
DEBUG
DEBUG
DEBUG
DEBUG
Download saved trace file
Reset referenz.fm
Technical Reference
Menus
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Menus
Speech problems
Speech problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Voice engine service
Media processing service
SIP signalling
SIP call control
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
TRACE
DEBUG
DEBUG
Download saved trace file
Reset
Tone problems
Tone problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Component registrar
Health service
Tone generation service
Media processing service
Download trace file
Submit
1048576
0
TRACE
LOG
TRACE
TRACE
Download saved trace file
Reset
USB backup/restore
USB backup/restore
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Administration
Component registrar
Physical interface service
USB backup service
Download trace file
Submit
1048576
0
TRACE
TRACE
DEBUG
DEBUG
Download saved trace file
Reset
352
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Voice recognition problems
Voice recognition problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Media control service
Voice engine service
Call View
Media processing service
Voice recognition
Phonebook
Download trace file
Submit
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
Web based management
Web based management
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
1048576
0
Trace levels for components
Data Access service
OpenStage client management
Web server service
USB backup service
802.1x service
Voice recognition
Download trace file
Submit
TRACE
LOG
TRACE
OFF
OFF
OFF
Download saved trace file
Reset referenz.fm
Technical Reference
Menus
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Menus
802.1x problems
802.1x problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Certificate management
Clock service
Component registrar
CSTA service
Data Access service
DLS client management
Mobílity service
SIP call control
SIP messages
SIP signalling
802.1x service
Download trace file
Submit
1048576
0
LOG
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
354
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Clear all profiles
Clear all profiles
File size (Max 6290000 bytes)
Trace timeout (minutes)
Automatic clear before start
Trace levels for components
Administration
Call Log
Call View
Phonebook
Help
Application menu
Certificate management
Communications
Component registrar
CSTA service
Data Access service
Digit analysis service
Digital data service
Directory service
DLS client management
Health service
Instrumentation service
Jounal service
Media control service
Media processing service
Mobility service
OBEX service
OpenStage client management
Performance Marks
Password management service
Physical interface service
Sidecar service
Team service
Tone generation service
Transport service
Voice engine service
Web server service
SIP signalling
SIP call control
SIP messages
Application framework
Desktop
AGP Phonelet
Service framework
Service registry
Bluetooth service vCard parser service
Voice mail
USB backup service
802.1x service
Voice recognition
1048576
0
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
OFF
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Menus
Bluetooth Advanced Traces
Bluetooth Advanced Traces
Automatic clear before start ;
File size (Max 6290000 bytes)
Extended dump
Verbose decoding
256000
;
;
Start Tracing is stopped
Download trace file
Submit Reset
Generation
Generation
Report mode
Report interval (seconds)
Observation interval (seconds)
Minimum session length (100 millisecond units)
Codec independent threshold values
Maximum jitter (milliseconds)
Average round trip delay (milliseconds)
Non-compressing codec threshold values
Lost packets (per 1000 packets)
Consecutive lost packets
Consecutive good packets
Compressing codec threshold values
Lost packets (per 1000 packets)
Consecutive lost packets
Consecutive good packets
Resend last report
Submit
EOS Threshold exceeded
60
10
20
20
100
10
2
8
10
2
8
Reset
356
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View Session Data
View Session Data
Codec change on the fly
Start of report period - seconds
End of report period - seconds
SNMP specific trap type
IP address (local)
Port number (local)
IP address (remote)
Port number (remote)
SSRC (receiving)
SSRC (sending)
Codec
Maximum packet size
Silence suppression
Count of good packets
Maximum jitter
Maximum inter-arrival jitter
Periods jitter threshold exceeded
Round trip delay
Round trip delay threshold exceeded
Count of lost packets
Count of discarded packets
Periods of lost packets
Consecutive packet loss (CPL)
Periods of consecutive lost packets
Consecutive good packets (CGP)
Periods of consecutive good packets
Count of jitter buffer overruns
Count of jitter buffer under-runs
Submit
QoS Statistics 1
2011/10/16 21:51:29 UTC
2011/10/16 21:56:36 UTC
2
192.168.1.235
5012
192.168.1.202
5010
1481715715
3244864262
G.711 PCMU
20
0
15203
2
0
0
433
0
0
0
255,255,255,255,255,255,255,255,255,255,255
255
255,255,255,255,255,255,255,255,255,255,255
255
0
0
IP tests
IP tests
Pre Defined Ping tests
Ping DLS
Ping tests
Pre Defined Trace tests
Traceroute DLS
Traceroute
Ping
Ping
Traceroute
Traceroute
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Memory information
Memory information
Memory Monitor Configuration
Disable Reboot
High Threshold(MBs)
Low Threshold(MBs)
Working Hour Start
Working Hour End
10
8
5
24
Download old memory info file
Reset
Download memory info file
Submit
Device Memory Information
Mem: 111336K used, 12380K free, 0K shrd, 0K buff, 55084K cached
CPU: 5% usr 15% sys 5% nic 25% idle 0% io 0% irq 50% sirq
Load average: 0.14 0.13 0.09 1/196 6098
PID PPID USER STAT VSZ %MEM %CPU COMMAND
6098 1908 root R 1420 1% 40% /bin/busybox top -d 0 -a -n 1 -l 600 -b
2063 1876 root S N 34148 28% 10% PhoneletLauncher desktopphonelet.phd V3 R0.50.0 SIP 110924 WP4 Siemens SIP DE de DD.MM.YYYY
24HR 0 NO_APP_PROP
3664 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
3665 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1929 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2515 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1902 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2992 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1876 1855 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1880 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
2057 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
1881 1877 root S 44712 36% 0% SvcConfig services.conf -startLogDaemon -logAll V3 R0.50.0 SIP 110924
Core Dump
Core Dump
Enable core dump*
Delete core dump
;
*Changes to these items do not take effect until the phone is restarted
Submit Reset
Remote trace
Remote trace
Remote Trace Status
User Notification
Remote Server IP
Remote Server Port
Submit
;
Reset
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Menus
Restart Phone
Restart Phone
Confirm Restart
Factory reset
Factory reset
Factory reset password
Submit Reset
HPT interface
HPT interface
Disable HPT
Secure Shell
Secure Shell
Enable access
Session password
Access minutes
Session minutes
Submit
1
1
Reset
Diagnostic call
Diagnostic call
Prefix Code
Submit
*333#
Reset
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4.1.2
Local Phone Menu
Menu
|---
Administration
|
|
|
|
|
|
| |---
Applications (OpenStage 60/80 only)
|
|
|
|
|
|
|
---
XML applications
|
|---
New phonebook application
|---
Add application: New call log application
|---
Add application: New messages application
|---
Add Xpressions: Xpressions application
---
Add
A
F
F
?
| |
| |---
Network
| | |---
General IP configuration
| | | |---
Protocol mode
| | | |---
Use LLDP-Med
| | | |---
Use DHCP
| | | |---
Use DHCPv6
| | | |---
VLAN discovery
| | | |---
VLAN ID
| | | |---
DNS domain
| | | |---
Primary DNS
| | | |---
Secondary DNS
| | |
|
---
HTTP Proxy (OpenStage 60/80 only)
| | |---
IPv4 configuration
| | | |---
Use LLDP-Med
| | | |---
Use DHCP
| | | |---
DHCP reuse
| | | |---
IP address
| | | |---
Subnet mask
| | | |---
Route (default)
| | | |---
Route 1 IP
| | | |---
Route 1 gateway
|
|
| | | |---
Route 1 mask
| | | |---
Route 2 IP
|
|
|
|
|
|--Route 2 gateway
--Route 2 mask
| | |--IPv6 configuration
| | | |--Use LLDP-Med
| | | |--Use DHCPv6
| | | |--DHCPv6 reuse
| | | |--Global address
| | | |--Global prefix len
| | | |--Global gateway
| | | |--Link local address
| | | |--Route 1 dest.
| | | |--Route 1 prefix len.
|
| | | |--Route 1 gateway
| | | |--Route 2 dest.
|
| | |
|
|
|--Route 2 prefix len.
--Route 2 gateway
| | |
360
Further information ...
-> Section 3.19.1.1
-> Section 3.19.1.1
-> Section 3.19.1.1
-> Section 3.19.1
-> Section 3.19.1.1
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-> Section 3.3.2
-> Section 3.2.2
-> Section 3.3.3
-> Section 3.3.3
-> Section 3.2.2.2
-> Section 3.2.2.3
-> Section 3.3.7.1
-> Section 3.3.7.2
-> Section 3.3.7.2
-> Section 3.19.1.2
-> Section 3.2.2
-> Section 3.3.3
-> Section 2.3.4
-> Section 3.3.4
-> Section 3.3.4
-> Section 3.3.5
-> Section 3.3.7
-> Section 3.3.7
-> Section 3.3.7
-> Section 3.3.7
-> Section 3.3.7
-> Section 3.3.7
-> Section 3.2.2
-> Section 3.3.3
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
-> Section 3.3.4.1
Menu
| | |---
Update Service (DLS)
| | | |---
DLS address
| | | |---
DLS port
| | | |---
Contact gap
|
| | | |---
Mode
| |
|
---
Security PIN
| | |
| | |---
QoS
| | | |---
Service
| | | | |---
Layer 2
| | | | |---
Layer 2 voice
| | | | |---
Layer 2 signalling
| | | | |---
Layer 2 default
| | | | |---
Layer 3
|
|
|
| | | | |---
Layer 3 voice
| | | |
|
---
Layer 3 signalling
| | | |---
MLPP (not used with OpenScape Voice)
|
|
|
|
|
|
|
---
Reports
|---
Generation
| |---
Mode
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |---
Report interval
| |---
Observe interval
|
---
Minimum session
|---
Send now
|
---
Thresholds
|---
Max jitter
|---
Round-trip delay
|---
Non-compressing:
|---
...Lost packets (K)
|---
...Lost consecutive
|---
...Good consecutive
|---
Compressing:
|---
...Lost packets (K)
|
|
| | |
|
|
|
|
|
|---
...Lost consecutive
--...Good consecutive
| | |
| | |--Port configuration
| | | |--SIP server
| | | |--SIP registrar
| | | |--SIP gateway
| | | |--SIP local
| | | |--Backup proxy
| | | |--RTP base
| | | |--LDAP Server port
| | | |--LAN port type
| | | |--PC port status
| | | |--PC port type
|
|
| | |
| |
|
|
|
|
|
|--PC port autoMDIX
--HTTP proxy
--LLDP-MED operation
|--Extended Power
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Further information ...
-> Section 3.3.8
-> Section 3.3.8
-> Section 3.3.8
-> Section 3.3.8
-> Section 3.3.8
-> Section 3.3.1.1
-> Section 3.3.1.1
-> Section 3.3.1.1
-> Section 3.3.1.1
-> Section 3.3.1.2
-> Section 3.3.1.2
-> Section 3.3.1.2
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12.1
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.26.12
-> Section 3.5.6.2
-> Section 3.5.6.2
-> Section 3.5.6.2
-> Section 3.5.6.2
-> Section 3.5.10.5
-> Section 3.18.1
-> Section 3.17.1
-> Section 3.2.1
-> Section 3.2.1
-> Section 3.2.1
-> Section 3.2.1
-> Section 3.19.1.2
-> Section 3.26.6
361
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Technical Reference
Menus
Menu
|
|
|
|
| |
|
|
|
| |---
Network policy (voice)
|---
LLDP-MED cap’s
|---
MAC_Phy config
|---
System cap’s
|
---
TTL - OK
| |
| |---
System
| | |---
Identity
| | | |---
Terminal number
| | | |---
Terminal name
| | | |---
Display identity
| | | |---
Enable ID
| | | |---
Web name
| | |
|
---
DDNS hostname
| | |---
SIP Interface
| | | |---
Outbound proxy
| | | |---
Default OBP domain
| | | |---
SIP transport
| | | |---
Call trans (ms) / Response timer (ms)
| | | |---
NonCall trans (ms)
| | | |---
Registration backoff
| | | |---
Connectivity timer (ms)
| | | |---
Keep alive format (not used with OS Voice)
| | | |---
Media negotiation
| | |
|
---
Media IP mode
| | |---
Registration
| | | |---
SIP addresses
| | | | |---
SIP server
| | | | |---
SIP registrar
|
| | | | ---
SIP gateway
| | | |---
SIP session
| | | | |---
Session timer
| | | | |---
Session duration (s)
|
|
|
|
| | | | |---
Registration timer (s)
| | | | |--Server type
| | | | |--Realm
| | | | |--User ID
| | | | |--Password
| | | | |--MLPP base (not used with OpenScape Voice)
| | | | |--MLPP Domain (not used with OpenScape Voice)
| | | |
|
|
|
|
|
|
|
|
|
--Other Domain (not used with OpenScape Voice)
|
--SIP survivability
|--Backup registration flag
|--Backup proxy address
|--Backup registration timer (s)
|--Backup transport
|
| | |
| |
|
--OBP flag
| | |
| | |--SNMP
| | | |--Queries allowed
| | | |--Query password
Further information ...
-> Section 3.26.6
-> Section 3.26.6
-> Section 3.26.6
-> Section 3.26.6
-> Section 3.2.3
-> Section 3.5.1.1
-> Section 3.5.1.1
-> Section 3.5.1.2
-> Section 3.5.1.2
-> Section 3.3.7.3
-> Section 3.3.7.3
-> Section 3.5.8.1
-> Section 3.5.8.1
-> Section 3.5.8.2
-> Section 3.5.10.2
-> Section 3.5.10.3
-> Section 3.5.10.4
-> Section 3.5.10.1
-> Section 3.5.8.3
-> Section 3.5.8.3
-> Section 3.5.6.1
-> Section 3.5.6.1
-> Section 3.5.6.1
-> Section 3.5.9
-> Section 3.5.9
-> Section 3.5.7
-> Section 3.5.7
-> Section 3.5.7
-> Section 3.5.7
-> Section 3.5.7
-> Section 3.5.10.5
-> Section 3.5.10.5
-> Section 3.5.10.5
-> Section 3.5.10.5
-> Section 3.5.10.5
-> Section 3.3.9
-> Section 3.3.9
362
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Menus
Menu
| | | |---
Trap sending enabled
| | | |---
Trap destination
| | | |---
Trap destination port
| | | |---
Trap community
| | | |---
Diagnostic sending enabled
| | | |---
Diagnostic destination
| | | |---
Diagnostic destination port
| | | |---
Diagnostic community
| | | |---
QoS traps to QCU
| | | |---
QCU address
|
| | | |---
QCU port
| | | |---
QCU community
| |
|
---
QoS to generic destination
| | |
| | |---
Features
| | | |---
Configuration
| | | | |---
General
| | | | | |---
Emergency number
| | | | | |---
Voicemail number
| | | | | |---
MWI LED
| | | | | |---
Missed Call LED
| | | | | |---
Allow refuse
| | | | | |---
Hot / warm phone
| | | | | |---
Hot / warm destination
| | | | | |---
Initial digit timer
| | | | | |---
Allow uaCSTA
| | | | | |---
Server features
| | | | | |---
Transfer on hangup
| | | | | |---
Not used timeout
| | | | | |---
DSS Pickup timer
| | | | | |---
Bridging enabled
| | | | | |---
Dial plan
|
| | | | | ---
FPK prog. timer
| | | | |---
Audio
| | | | | |--Pickup tone allowed
| | | | | |--Pickup as ringer
|
| | | | | |--Pickup visual alert
| | | | | |--BLF alerting
| | |
| | | | |
|
|
|
|---
---
MLPP ringer (not used with OpenScape Voice)
Callback ringer
--Lower IL ringer
|
| | | | |
| | | | |--Keyset Lines (OpenStage 15/40/60/80)
| | | | |
| | | |
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--Details For Keyset Line <n>
|--Address
|--Ring on/off
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--Call recording
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|---
|---
|
|--Selection order
--Hot/warm action
Recorder number
Recording mode
--Audible notification
Further information ...
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.3.9
-> Section 3.5.2
-> Section 3.5.2
-> Section 3.28
-> Section 3.29
-> Section 3.7.1
-> Section 3.7.2
-> Section 3.7.2
-> Section 3.7.3
-> Section 3.7.11
-> Section 3.7.10
-> Section 3.7.5.2
-> Section 3.7.12
-> Section 3.11.5.1
-> Section 3.11.2
-> Section 3.13.4
-> Section 3.8
-> Section 3.7.4.2
-> Section 3.7.4.2
-> Section 3.7.4.2
-> Section 3.7.4.2
-> Section 3.7.6.1
-> Section 3.30
-> Section 3.11.1
-> Section 3.11.1
-> Section 3.11.1
-> Section 3.11.1
-> Section 3.7.13
-> Section 3.7.13
-> Section 3.7.13
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual 363
referenz.fm
Technical Reference
Menus
Menu
| | | |---
Keyset operation (OpenStage 15/40/60/80 only)
| | | | |---
Rollover ring
| | | | |---
LED on registration
| | | | |---
Originating line preference
| | | | |---
Terminating line preference
| | | | |---
Line action mode
| | | | |---
Show focus
| | | | |---
Reservation timer
| | | | |---
Forwarding indicated / Forwarding shown
| | | | |---
Preselect mode
| | | | |---
Preselect timer
| | | | |---
Preview mode
| | | | |---
Preview timer
|
| | | | ---
Bridging priority
| | | |---
DSS operation (OpenStage 15/40/60/80 only)
| | | | |---
Deflect to DSS
| | | | |---
Refuse DSS pickup
| | | |
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---
Forwarding shown
| | | |---
Addressing
| | | | |---
MWI server URI
| | | | |---
Conference
| | | | |---
Group pickup URI
| | | | |---
Callback FAC
| | | | |---
Callback: busy (upto V2R2)
| | | | |---
Callback: no reply (upto V2R2)
| | | | |---
Callback: cancel all
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| | | | ---
BLF pickup code
| | | |---
Call completion
| | | | |---
Functional CCSS
| | | | |---
Callback ringer
| | | | |---
Allow after call (s)
|
| |
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---
Max. callbacks
---
Feature Access
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|---
Call control
|
|---
---
Blind transfer
3rd call leg
|--Call establish
| |--Callback
| |--Call pickup
| | |
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|
| |--Group pickup
| |--Call deflection
| |--Call forwarding
| |--Do not disturb
| |--Refuse call
| |--Repertory dial key
|
--Ext/int forwarding |
|--Call associated
| |--Phone book lookups
| |--DSS feature
|
|
|
|--BLF feature
--Line overview
364
Further information ...
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.2
-> Section 3.11.3
-> Section 3.11.3
-> Section 3.11.3
-> Section 3.11.5.1
-> Section 3.11.5.1
-> Section 3.11.5.1
-> Section 3.7.7
-> Section 3.7.9
-> Section 3.7.4
-> Section 3.7.6
-> Section 3.7.6
-> Section 3.7.6
-> Section 3.7.6
-> Section 3.7.4
-> Section 3.7.6.1
-> Section 3.7.6.1
-> Section 3.7.6.1
-> Section 3.7.6.1
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
Menu
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---
Video calls
|---
CTI
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---
CTI control
---
Services
|---
Bluetooth
|---
Web based mang.
|---
USB device access
|---
Backup to USB
|---
Feature togggle
|
---
Phone lock
|
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---
Security
|---
System
| |---
Server certificate
|---
Use secure calls
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|---
SRTP type
---
Use SRTCP
|---
SDES config
|
|
|---
SDES status
|---
SDP negotiation
|---
SHA1-80 crypto
| ---
SHA1-32 crypto
|---
Access control
|
|
|
|
|---
CCE access
|---
Factory reset claw
|
---
Serial port
|
|---
Logging
|---
Max. lines
|---
Archive to DLS
|
|
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|
|---
Last archived
---
Archive when at
---
Faults
|---
Security log entry
|---
OCSR failure
|
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| |
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|--Admin access
--User access
| |
| |--File Transfer
| | |--Defaults
| | | |--Download method
| | | |--Server
| | | |--Port
| | | |--Account
| | | |--Username
| | | |--Password
| | | |--FTP path
|
--HTTPS base URL | | |
| | |--Phone app
| | | |--Use default
| | | |--Download method
| | | |--Server
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual referenz.fm
Technical Reference
Menus
Further information ...
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.6
-> Section 3.4
-> Section 3.4
-> Section 3.4
-> Section 3.4
-> Section 3.4.1.3
-> Section 3.4.1.3
-> Section 3.4.1.3
-> Section 3.4.1.3
-> Section 3.4.2
-> Section 3.4.2
-> Section 3.4.2
-> Section 3.4.3
-> Section 3.4.3
-> Section 3.4.3
-> Section 3.4.3
-> Section 3.4.4
-> Section 3.4.4
-> Section 3.4.4
-> Section 3.4.4
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.2
-> Section 3.16.3
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
365
referenz.fm
Technical Reference
Menus
Menu
| | | |---
Port
| | | |---
Account
| | | |---
Username
| | | |---
Password
| | | |---
FTP path
| | | |---
HTTPS base URL
| | |
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---
Filename
| | |---
Hold music
| | | |---
Use default
| | | |---
Download method
| | | |---
Server
| | | |---
Port
| | | |---
Account
| | | |---
Username
| | | |---
Password
| | | |---
FTP path
| | | |---
HTTPS base URL
| | |
|
---
Filename
| | |---
Ringer
| | | |---
Use default
| | | |---
Download method
| | | |---
Server
| | | |---
Port
| | | |---
Account
| | | |---
Username
| | | |---
Password
| | | |---
FTP path
| | | |---
HTTPS base URL
| | |
|
---
Filename
| | |---
Picture clip (OpenStage 60/80 only)
| | | |---
Use default
| | | |---
Download method
| | | |---
Server
| | | |---
Port
| | | |--Account
| | | |--Username
| | | |--Password
| | | |--FTP path
| |
| | |
|
|
|--HTTPS base URL
--Filename
| | |--LDAP
| | | |--Use default
| | | |--Download method
| | | |--Server
| | | |--Port
| | | |--Account
| | | |--Username
| | | |--Password
| | | |--FTP path
| | | |--HTTPS base URL
| | |
|
--Filename
366
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
Further information ...
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.3.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.4.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.5.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
-> Section 3.16.6.1
Menu
| | |---
Logo (OpenStage 40/60/80 only)
| | | |---
Use default
| | | |---
Download method
| | | |---
Server
| | | |---
Port
| | | |---
Account
| | | |---
Username
| | | |---
Password
| | | |---
FTP path
| | | |---
HTTPS base URL
| | |
|
---
Filename
| | |---
Screensaver (OpenStage 60/80 only)
| | | |---
Use default
| | | |---
Download method
| | | |---
Server
| | | |---
Port
| | | |---
Account
| | | |---
Username
| | | |---
Password
| | | |---
FTP path
|
|
| | | |---
HTTPS base URL
|
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---
Filename
---
HPT dongle
|
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| |---
Use default
|---
Download method
|---
Server
|---
Port
|---
Account
|---
Username
|---
Password
|---
FTP path
|
|---
HTTPS base URL
---
Filename
| |
| |--Local Functions
| | |--Directory Settings / LDAP
| | | |--(LDAP) server address
| | | |--(LDAP) server port
| | | |--(LDAP) Authenticate / Authentication
| | | |--(LDAP) User name
| | | |--(LDAP) Password
| | |
|
--Timeout (sec) for / Search Trigger (s)
| | |--Locality
| | | |--Canonical settings
| | | | |--Local country code
| | | | |--National prefix digit
| | | | |--Local national code
| | | | |--Minimum local number length
| | | | |--Local enterprise node
| | | | |--PSTN access code
| | | | |--International access code
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual referenz.fm
Technical Reference
Menus
Further information ...
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.7.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.8.1
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.16.10
-> Section 3.17.1
-> Section 3.17.1
-> Section 3.17.1
-> Section 3.17.1
-> Section 3.17.1
-> Section 3.17.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
367
referenz.fm
Technical Reference
Menus
Menu
| | | | |---
Operator code
| | | | |---
Emergency number
| | | |
|
---
Initial digits
| | | |---
Canonical lookup
| | | | |---
Local code 1
| | | | |---
International 1
| | | | |---
Local code 2
| | | | |---
International 2
| | | | |---
Local code 3
| | | | |---
International 3
| | | | |---
Local code 4
| | | | |---
International 4
| | | | |---
Local code 5
| | | |
| | |
|
---
|
---
International 5
Canonical dial
| | | | |---
Internal numbers
| | | | |---
External numbers
| | | | |---
External access
| | | |
|
---
International gateway / International access
| | |---
Energy saving (OpenStage 40/60/80 only)
|
---
Backlight timeout
| | |
| | |---
Messages settings
| | | |---
New items
| | | |---
Alternative label
| | | |---
New urgent items
| | | |---
Alternative label
| | | |---
Old items
| | | |---
Alternative label
|
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|
| | | |---
Old urgent items
|
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---
Alternative label
---
Call logging
|
---
FAC prefixes
|
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| |
| |---
Date and Time
| | |--Time source
| | | |--SNTP IP address
|
| | |
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|
--Timezone offset
--Daylight saving
|--Daylight saving
|--Difference (mins)
|--Auto DST
|
--DST zone
|
|
| |
| |--Speech
| | |--Codec preferences
| | | |--Silence suppression
| | | |--Packet size
| | | |--G.711
| | |
|
|
|
|
|
|
|---
---
G.729
G.722
--Allow “HD” icon
368
Further information ...
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.2
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.13.1
-> Section 3.5.3
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.7.8
-> Section 3.5.4
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.5.5.1
-> Section 3.18.2
-> Section 3.18.2
-> Section 3.18.2
-> Section 3.18.2
-> Section 3.18.2
-> Section 3.18.2
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
Menu
|
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|
---
Audio settings
|---
Disable microphone
|---
Disable loudspeech
|
---
DTMF playback
| |
| |---
General information
| | |---
MAC address
| | |---
Software version
|
| | |---
Last restart
| | |---
Dial plan ID
|
|
---
Dial plan status
| |
| |---
Licence information
| |
| |---
Password (up to V2R1)
| | |---
Admin
| | |---
Confirm admin
|
| | |---
User
|
|
---
Confirm user
| |
| |---
Security & policies
| | |---
Password
| | | |---
Generic policy
| | | | |---
Expires after (days)
| | | | |---
Warn before (days)
| | | | |---
Force changed
| | | | |---
Tries allowed
| | | | |---
No change for (hours)
| | | | |---
Suspended for (mins)
|
---
History valid for (days)
| | | |
| | | |---
Admin policy
| | | | |---
Expiry date
| | | | |---
Minimum length
| | | | |---
Password history
| | | |
|
--Current status
| | | |--User policy
| | | | |--Expiry date
| | | | |--Minimum length
| | | | |--Password history
| | | |
|
--Current status
| | | |--Character set
| | | | |--Ucase chars reqd.
| | | | |--Lcase chars reqd.
| | | | |--Digits required
| | | | |--Special chars reqd.
| | | | |--Bar repeat length
| | | |
|
--Min char difference
| | | |--Change admin password
| | | | |--Current password
| | | | |--New password
| | | |
|
--Confirm password
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual referenz.fm
Technical Reference
Menus
Further information ...
-> Section 3.18.3
-> Section 3.18.3
-> Section 3.18.3
-> Section 3.26.1
-> Section 3.26.1
-> Section 3.26.1
-> Section 3.13.4
-> Section 3.13.4
-> Section 3.25
-> Section 3.20
-> Section 3.20
-> Section 3.20
-> Section 3.20
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.1
-> Section 3.4.5.2
-> Section 3.4.5.2
-> Section 3.4.5.2
-> Section 3.4.5.2
-> Section 3.4.5.3
-> Section 3.4.5.3
-> Section 3.4.5.3
-> Section 3.4.5.3
-> Section 3.4.5.4
-> Section 3.4.5.4
-> Section 3.4.5.4
-> Section 3.4.5.4
-> Section 3.4.5.4
-> Section 3.4.5.4
-> Section 3.4.5.5
-> Section 3.4.5.5
-> Section 3.4.5.5
369
referenz.fm
Technical Reference
Menus
Menu
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| | |
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| | |
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|
---
Change User password
|---
Admin password
|---
New password
|
---
Confirm password
---
Certificates
|---
Generic
|
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| |---
OCSP check
| |---
OCSR 1 address
|
---
OCSR 2 address
---
Authentication policy
|---
Secure file transfer
|---
Secure send URL
|---
Secure SIP server
|
|---
Secure 802.1x
---
Secure XML appl.
|
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| |
| |---
Ringer setting
|
| | |---
Distinctive
| |
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|
---
<1 .... 15>
|---
Name
|---
Ringer sound
|---
Pattern melody
|---
Pattern sequence
|---
Duration
|
---
Audible
|
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| | |
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|
---
Map to Specials
|---
Internal
|---
External
|---
Recall
|---
Emergency
|---
Special1
|
|
|
|---
Special2
---
Special3
|
| |
| |--Mobility
| | |--Unauthorized logoff trap
| | |--Logoff trap delay
| | |--Timer med priority
| | |--Mobility feature
| |
|
|
|--Managed profile
--Error count local
| |
| |--Diagnostic information
|
|
|
|
| | |--View
| |
|
--Configure
--Allow user
|
| |
|
--Maintenance
|--Factory reset
|--Disable HPT
|--Remote trace
370
Further information ...
-> Section 3.4.5.5
-> Section 3.4.5.5
-> Section 3.4.5.5
-> Section 3.4.5.5
-> Section 3.4.6.1
-> Section 3.4.6.1
-> Section 3.4.6.1
-> Section 3.4.6.2
-> Section 3.4.6.2
-> Section 3.4.6.2
-> Section 3.4.6.2
-> Section 3.4.6.2
-> Section 3.14
-> Section 3.14
-> Section 3.14
-> Section 3.14
-> Section 3.14
-> Section 3.14
-> Section 3.14.2
-> Section 3.15
-> Section 3.15
-> Section 3.15
-> Section 3.15
-> Section 3.26.2
-> Section 3.26.3
-> Section 3.23
-> Section 3.26.15
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Trace status
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User notification
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Remote server
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Remote port
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Memory monitor
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Disable reboot
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High threshold
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Low threshold
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Working Hour start
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Working Hour end
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Diagnostic call
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Technical Reference
Menus
Further information ...
-> Section 3.26.14
-> Section 3.26.14
-> Section 3.26.14
-> Section 3.26.14
-> Section 3.26.8
-> Section 3.26.8
-> Section 3.26.8
-> Section 3.26.8
-> Section 3.26.8
-> Section 3.26.4
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Technical Reference
Default Port List
4.2
Default Port List
The following table contains all default ports, resp. port ranges, and protocols used by the services running on OpenStage SIP phones.
Service
Payload transport (VoIP)
Payload transport (VoIP)
SIP subscriber - TCP is used
SIP subscriber - TLS is used
SIP subscriber - UDP is used
XML applications in phone, connecting to an application server
XML Push service
XML Push service
Directory access via LDAP
Directory access via LDAP
DHCP Client
DNS Client
DLS contact me service - workpoint side
Default communication with the DLS workpoint interface
Secure communication with the DLS workpoint interface
Connection to the control port of FTP server
FTP client; uses the FTP server in active mode
HTTPS file download server
8085
443
---
---
---
---
8085
Server Default Port
Client Default
Port
5010 - 5059 5010 - 5059
5010 - 5059 5010 - 5059
5060
5061
5060
---
32786 - 61000
32786 - 61000
5060
32786 - 61000
---
---
21
443
---
---
32786 - 61000
32786 - 61000
68
1024 - 65535
---
18443
18444
32786 - 61000
32786 - 61000 20
32786 - 61000
Client application which sends QDC data to the QCU
---
Part of SNMP-Agent - sending Traps ---
Part of SNMP-Agent - receive Set/
Get commands
161
32786 - 61000
32786 - 61000
---
Protocol Stack
RTP - RTCP
SRTP - SRTCP
SIP / TCP
SIP / TLS
SIP / UDP
HTTP / TCP
HTTPS / TCP-TLS
HTTP / TCP
HTTPS / TCP-TLS
TCP
TCP- SSL/TLS
DHCP / UDP
DNS / TCP_UDP
HTTP / TCP
HTTPS / TCP -
SSL / TLS
HTTPS / TCP -
SSL / TLS
FTP / TCP
FTP / TCP
HTTPS / TCP -
SSL/TLS
SNMP / UDP
SNMP / UDP
SNMP / UDP
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Technical Reference
Default Port List
Service Server Default Port
--SNTP client - queries time information in unicast operation
SNTP client - receives time information in broadcast operation
Web server for WBM access
123
8085
Secure Web Server for WBM access 443
OpenStage Phone Manager
Remote Trace
HPT- debug IF (Available only if a dongle file is present on phone.)
65532
---
65532
SSH (Secure Shell Remote Login) 22
Syslog Client (sends Traces to Syslog Server)
---
Video H.263
Secure Video H.263
5050-5059
5050-5059
Client Default
Port
123
---
---
---
---
514
---
---
32786 - 61000
5050-5059
5050-5059
Protocol Stack
SNTP / UDP
SNTP / UDP
HTTP / TCP
HTTPS / TCP -
SSL / TLS
TCP - SSL/TLS
UDP
TCP - SSL/TLS
TCP
UDP
RTP - RTCP
SRTP - SRTCP
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Technical Reference
Troubleshooting: Error Codes
4.3
Troubleshooting: Error Codes
For a set of error cases, specific error codes are defined. These error codes are shown in brackets on the display, following a general error note. Example: „No Telephony possible
(LP1)“.
Problem
Network Problem
Not Initialised
Description
No network connection
Waiting for data
Unable to use LAN 802.1x error
Unable to use LAN Physical connection missing
Unable to Register Server timeout
Unable to Register Server failed
Unable to Register Authentication failed
Unable to Register No number configured
Unable to Register No server configured
Unable to Register No registrar configured
Unable to Register No DNS domain configured
Unable to Register Rejected by server
Unable to Register No phone IP address set
Survivability Backup route active
Survivability
Survivability
Survivability
Cloud-Deployment abandoned by user
Backup not configured
Backup timeout
Backup authentication failed
Cloud-Deployment abandoned by user
Occurs when the pin prompt is dismissed
RA2
RN2
RS2
RG2
RD2
RR2
RI2
B8
Error code
LI1
I1
LX1
LP1
RT2
RF2
RS8
RT8
RA8
AU
Cloud-Deployment: unable to get the address for the SEN
Redirect server
Unable to establish contact with SEN
Redirect server - no reply
Cloud-Deployment: unable to get the address for the SEN Redirect server. DNS lookup failed
Cloud-Deployment: unable to establish contact with SEN Redirect server - no reply
RS
RN
Tabelle 4-1 Troubleshooting Error Codes
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Troubleshooting: Error Codes
Problem
Unable to establish contact with SEN
Redirect server - refused
Unable to establish contact with SEN
Redirect server - unauthorised
Unable to establish contact with SEN
Redirect server - no or invalid OCSP response
Unable to establish contact with SEN
Redirect server - certificate revoked
Unable to get the address for the Deployment server
Unable to establish contact with Deployment server
Unable to establish contact with Deployment server - refused
Description
Cloud-Deployment: unable to establish contact with SEN Redirect server - refused
Cloud-Deployment: unable to establish contact with SEN Redirect server - unauthorised
Cloud-Deployment: unable to establish contact with SEN Redirect server - no or invalid
OCSP response
Cloud-Deployment: unable to establish contact with SEN Redirect server - certificate revoked
Cloud-Deployment: unable to get the address for the Deployment server. DNS lookup failed
Cloud-Deployment: unable to establish contact with Deployment server - no reply
Cloud-Deployment: unable to establish contact with Deployment server - refused
Error code
RR
RU
RO
RV
DS
DN
DR
Tabelle 4-1 Troubleshooting Error Codes
> A special “fast-busy” tone (also called congestion tone) is played if a temporary network problem causes a user-initiated call action to fail. Typical call actions: making an outgoing call; picking up a call from Manual Hold; or Group pickup. Phone users include keyset users and mobile users logged on to the phone. The special tone is triggered if one of the following SIP response codes is received from the server:
606, 408, or 503.
http://wiki.unify.com/wiki/OpenStage_SIP_FAQ#Error_codes
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Examples and HowTos
Canonical Dialing
5 Examples and HowTos
5.1
Canonical Dialing
5.1.1
Canonical Dialing Settings
The following example shows settings suitable for the conversion of given dial strings to canonical format. The example phone is located in Nottingham, UK.
Parameter
Local country code
National prefix digit
Local national code
Minimum local number length
Local enterprise node
PSTN access code
Operator codes
Emergency numbers
Initial extension digits
Example value Explanation
44
0
International country code for the UK.
Used in front of national codes when dialled without international prefix.
115
7
780
9
0,7800
Area code within the UK (here: Nottingham).
Minimum number of digits in a local PSTN number (e. g. 3335333 = 7 digits).
Prefix to access Nottingham numbers from within the Siemens network.
Prefix to make an international call in the UK.
999,555
2,3,4,5,6,8
Set of numbers to access the local operators.
(No blank after comma, or else the subsequent entry is ignored.)
Set of numbers to access emergency services. (No blank after comma, or else the subsequent entry is ignored.)
1 st
digits of numbers that are used for extension numbers on the local node. (No blank after comma, or else the subsequent entry is ignored.)
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Canonical Dialing
5.1.2
Canonical Dial Lookup
The following example shows settings suitable for recognizing incoming numbers and assigning them to entries in the local phonebook, and for generating correct dial strings from phone book entries, depending on whether the number is internal or external.
Parameter
Local code <1>
International code <1>
Local code <2>
International code <2>
Example value Explanation
780
+44115943
Enterprise node prefix (here: Nottingham).
Equivalent prefix to access numbers on this node from the PSTN. Here, the prefix used by the PSTN (DID/DDI: direct inward dialing) is
943, which differs from the enterprise node prefix used within the enterprise network.
7007
+49897007
Enterprise node prefix (here: Munich).
Equivalent prefix to access numbers on this node from the PSTN. Here, the prefix used by the PSTN for direct inward dialing is identical to the enterprise node prefix.
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Canonical Dialing
5.1.2.1
Conversion examples
In the following examples, numbers entered into the local phonebook by the user are converted according to the settings given above.
Example 1: Internal number, same node as the local phone
User entry
External numbers
External access code
International gateway code
Number stored in the phonebook
Dial string sent when dialing from the phonebook
Internal numbers = Local enterprise form
Internal numbers = Always add node
Internal numbers = Use external numbers
Example 2: Internal number, different node
2345
Local public form
Not required
Use national code
+441159432345
1234
7802345
9432345
User entry
External numbers
External access code
International gateway code
Number stored in the phonebook
Dial string sent when dialing from the phonebook
Internal numbers = Local enterprise form
Internal numbers = Always add node
Internal numbers = Use external numbers
70072345
Local public form
Not required
Use national code
+498970072345
2345
7802345
9432345
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Example 3: External number, same local national code as the local phone
User entry
External numbers
External access code
International gateway code
Number stored in the phonebook
Dial string sent when dialing from the phonebook
External numbers = Local public form
External numbers = National public form
External numbers = International form
011511234567
Local public form
Not required
Use national code
+4411511234567
234567
011511234567
004411511234567
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Examples and HowTos
How to Create Logo Files for OpenStage Phones
5.2
How to Create Logo Files for OpenStage Phones
5.2.1
For OpenStage 40
1. Create a New Image
Create an image with the following specifications:
• Width: 144 px
• Height: 32 px
• Color Mode: 1 bit (monochrome)
Adobe Photoshop:
2. Insert the Logo
Place the logo image on the background, e.g. by copying it from a source file. Due to the size and color specifications, some adaptations may be necessary.
Adobe Photoshop Example:
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3. Save the Image
Finally, save the image in BMP format. You can now upload the logo file to the phone as
described in Section 3.16.7, “Logo”.
5.2.2
For OpenStage 60/80
In the following, the creation of a transparent image suitable for use as a logo in OpenStage
60/80 is described. This description is based on Adobe Photoshop, but any similar graphics software can be used as well.
> Because of performance issues, half transparency in the alpha channel of the PNG files is not allowed on OpenStage phones. Therefore only 100% transparency or no transparency is used in the phone’s UI elements.
1. Select the Background Color
For production purposes, we set the background color to the background color of the skin currently selected on the phone. Later, the background color will be replaced by transparency, which facilitates placing a logo on a gradient background. The following table lists the hexadecimal values, as used in HTML:
Phone Type Skin
OpenStage 60 Crystal Sea
OpenStage 60 Warm Grey
OpenStage 80 Crystal Sea
Color Code
#E7E7E7
#424242
1
#E6EBEF
OpenStage 80 Warm Grey #3A3D3A
1 The background color on WP4 - skin 1 is a gradient; the colour listed here is an average value.
Adobe Photoshop:
Click on the Background Color icon on the Color palette group, then type the color code without leading "#" into the # field)
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2. Create a New Image
Create an image with the size according to the phone type:
Phone Type
OpenStage 60
OpenStage 80
Adobe Photoshop:
Size (px)
240 x 70
480 x 142
3. Insert the Logo
Place the logo image on the background, e.g. by copying it from a source file.
Adobe Photoshop Example:
4. Merge Layers
Merge the two layers to one.
Adobe Photoshop:
In the Panel, select both the background layer and the new layer containing the inserted logo. Afterwards, go to Layer in the Menu bar, and select Merge Layers.
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5. Background Transparency
Delete the background colour so that only the exact former background colour is 100% transparent.
Adobe Photoshop:
Make sure that the background color is selected by clicking on the Background Color icon.
In the Tool palette, click on the Eraser symbol with the right Mouse button and select the
Magic Eraser Tool. After this, got to the Menu bar and set the Tolerance field to "0".
6. Save the Image
Finally, save the image in PNG format. You can now upload the logo file to the phone as
described in Section 3.16.7, “Logo”.
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Examples and HowTos
How to Set Up the Corporate Phonebook (LDAP)
5.3
How to Set Up the Corporate Phonebook (LDAP)
The Corporate Phonebook function is based on an LDAP client that can be connected to the company’s LDAP service. A variety of LDAP servers can be used, for instance Microsoft Active
Directory, OpenLDAP, or Apache Directory Server.
> The Corporate Phonebook is available on OpenStage 15/20/40/60/80 phones with firmware version V3R3 onwards.
5.3.1
Prerequisites
1. An LDAP server is present and accessible to the phone’s network. The standard Server port for LDAP is 389, the standard transport for LDAP is TCP.
2. Query access to the LDAP server must be provided. Unless anonymous access is used, a user name and password must be provided. It might be feasible to use a single login/password for all OpenStage phones.
3. To enable dialing internal numbers from the corporate phonebook, an LDAP entry must be provided that contains the proper number format required by Phone Administration.
In Microsoft Active Directory, the standard LDAP attribute telephoneNumber is typically populated as follows: +1<area code><call number>. However, in a standard configuration, Phone Administration will not handle this dial string correctly, due to the +1 prefix.
Therefore, it is recommended to use the ipPhone field, which is typically unused in Active
Directory. It can be found in the Telephones tab of the Active Directory User Manager.
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How to Set Up the Corporate Phonebook (LDAP)
5.3.2
Create an LDAP Template
The user interface of the corporate phonebook application provides a form which is used both for search and retrieval.
The task of an LDAP template is to map the phone’s search and display fields to LDAP attributes that can be delivered by the server. In the LDAP template, the fields are represented by hard-coded names: ATTRIB01, ATTRIB02, and so on. These field names are assigned to
LDAP attributes, as appropriate.
Wildcard Quick Search
Supporting a wildcard search for a quick search (V3.3 and upwards) an additional attribute
"ATTRIB12" is to be set in LDAP template. So it is possible to search for a pattern within a word and not only at the beginning, e.g. ’lea’ finds ’project lea der’ and ’team lea der’. The addtional attribute "ATTRIB12" could be mapped to any attribute of the LDAP server. Therefore the name of any LDAP attribute can be assigned to "ATTRIB12" in LDAP template.
The "ATTRIB12" is optional, the existing templates remain unchanged, when no wildcard
search is desired. Search via FIND-mask is not affected by "ATTRIB12". See example
Æ
LDAP
Template for wildcard search example
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The following examples show the relations between GUI field names, the attribute labels used in the template, and exemplary mappings to LDAP attributes.
> In an LDAP template for OpenStage 15/20/40, the entries must be sorted according to the sequential number of the template labels, as shown in the example underneath.
Generic Example (Standard Attributes)
OpenStage Field LDAP
Template
Lables
Last name
ATTRIB01
LDAP Attribute sn
First name
Business 1
Business 2
Mobile
ATTRIB02
ATTRIB03
ATTRIB04
ATTRIB05 givenName telephoneNumber facsimileTelephoneNumber mobile
Example
Value
Doe
John
9991234
9992345
017711223344
Private
Company
Address 1
Address 2
Job function
Email
ATTRIB06
ATTRIB07
ATTRIB08
ATTRIB09
ATTRIB10
ATTRIB11 homePhone o departmentNumber title mail
441274333444
Example Inc.
0815
Product Manager [email protected]
ATTRIB12 any LDAP attribute optional for wildcard search
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Given "example.com" as the LDAP subtree to be searched, the LDAP template file looks like this:
OpenStage LDAP TEMPLATE (v.1)
SEARCHBASE="dc=example,dc=com"
ATTRIB01="sn"
ATTRIB02="givenname"
ATTRIB03="telephoneNumber"
ATTRIB04="facsimileTelephoneNumber"
ATTRIB05="mobile"
ATTRIB06="homePhone"
ATTRIB07="o"
ATTRIB08="departmentNumber"
ATTRIB09="l"
ATTRIB10="title"
ATTRIB11="mail"
EOF
Microsoft Active Directory Specific Example
OpenStage Field LDAP
Template
Attribute
Last name
First name
ATTRIB01
ATTRIB02
LDAP Attribute sn givenName
Business 1
Business 2
Mobile
Private
Company
Address 1
ATTRIB03
ATTRIB04
ATTRIB05
ATTRIB06
ATTRIB07
ATTRIB08 ipPhone otherTelephone mobile homePhone company department
Address 2
Job function
Email
ATTRIB09
ATTRIB10
ATTRIB11 title mail
ATTRIB12 any LDAP attribute
Example
Value
Doe
John
9991234
9992345
017711223344
441274333444
Example Inc.
Administration
Product Manager [email protected]
optional for wildcard search
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Given "example.com" as the LDAP subtree to be searched, the LDAP template file looks like this:
OpenStage LDAP TEMPLATE (v.1)
SEARCHBASE="dc=example,dc=com"
ATTRIB01="sn"
ATTRIB02="givenName"
ATTRIB03="ipPhone"
ATTRIB04="otherTelephone"
ATTRIB05="mobile"
ATTRIB06="homePhone"
ATTRIB07="company"
ATTRIB08="department"
ATTRIB09="l"
ATTRIB10="title"
ATTRIB11="mail"
EOF
LDAP Template for wildcard search example
Given "example.com" as the LDAP subtree to be searched with ’wildcard search’ quick search for LDAP attribute ’info’, the LDAP template file looks like this:
OpenStage LDAP TEMPLATE (v.1)
SEARCHBASE="dc=example,dc=com"
ATTRIB01="sn"
ATTRIB02="givenName"
ATTRIB03="ipPhone"
ATTRIB04="otherTelephone"
ATTRIB05="mobile"
ATTRIB06="homePhone"
ATTRIB07="company"
ATTRIB08="department"
ATTRIB09="l"
ATTRIB10="title"
ATTRIB11="mail"
ATTRIB12="sn"
EOF
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5.3.3
Load the LDAP Template onto the Phone
When you have configured the LDAP template, you can upload it to the phone:
1. Save the template under a suitable name, for example, ldap-template.txt.
2. Copy the template file to the FTP server designated for deploying LDAP templates.
3. Upload the file using the WBM (see Section 3.16.6, “LDAP Template”), or, alternatively,
the Local menu, or the DLS (see the Deployment Service Administration Manual). For an example configuration, see the following WBM screenshot (path: File transfer > LDAP):
LDAP
Use defaults
Download method
FTP Server address
FTP Server port
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
Submit
FTP
21 phone media ldap-template.txt
do nothing
Reset
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How to Set Up the Corporate Phonebook (LDAP)
5.3.4
Configure LDAP Access
To enter the access data using the WBM, take the following steps:
1. Navigate to Local Functions > Directory Settings.
2. Enter the following parameters:
• LDAP Server address (IP address or hostname of the LDAP server)
• Transport (allows the LDAP interface to be encrypted using TLS (via LDAPS) or unencrypted using TCP, typically TCP)
• LDAP Secure port (port used by the LDAP for encrypted (TLS) transport, typically
636)
• LDAP Server port (port used by the LDAP for unencrypted (TCP) transport, typically
389)
• Authentication (authentication method for the connection to the LDAP server)
• User name (only required if simple authentication is selected); Password (relating to the user name).
Directory settings
LDAP Server address
Transport
LDAP Secure port
LDAP Server port
Authentication
User Name
Password
Search trigger timeout
TCP
636
389
Anonymous
3
Submit Reset
3. Press Submit.
5.3.5
Test
If everything went well, you can run a test query on your OpenStage phone.
1. To navigate to the phone's corporate phonebook use the following keys: Press the g button until the corporate directory tab is shown (OS60/OS80). Press the "Settings" key or u on OS15/OS20, or OS40 and use page up/page down to select the corporate phonebook
2. In the query mask, select the entry to be searched, for instance Last Name. Press i to open the onscreen keypad for text input.
3. Enter the text to be searched. For information on using the onscreen keypad, see Section
3.1, “Access via Local Phone”, step 5.
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4. Navigate to the Find option and press i . If the query was successful, at least one entry will be listed in the following manner:
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5. Navigate to the desired entry and press g on the TouchGuide to open the context menu.
You can select one of the following options:
• Dial the Business 1 number.
• Dial the Mobile number.
• Have the entry’s details, that is, all attributes displayed.
• Start a new search.
• Clear the list of search results.
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An LLDP-Med Example
5.4
An LLDP-Med Example
The following example illustrates the mode of operation of LLDP-MED. In order to evoke a reaction from LLDP-MED, the LAN switch has been set to auto-negotiation, whereas the phone’s
LAN port (see Section 3.2.1, “LAN Port Settings”) is set to 100Mbit/s, hence a fixed value. This
configuration error is discovered by LLDP-MED. The following sceenshots from the phone’s local menu will show the error messages.
This screenshot shows the LLDP-MED operation submenu (see Section 3.2.3, “LLDP-MED
Operation”). Please note the status of MAC_Phy config.
When MAC_Phy config is selected, the details are displayed.
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An LLDP-Med Example
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Examples and HowTos
Dial Plan
5.5
Dial Plan
5.5.1
Introduction
A dial plan is a set of rules that determine the phone’s behaviour on digit entry by the user. Up to 48 rules are possible. With OpenStage phones, a dial plan rule is constructed from 9 parameters. In the following, the setup of a dial plan is explained.
The dial plan entries are preceded by a title line. This is a free format string, e. g. a descriptive name or version number, which can be used by the administrator for version control purposes.
5.5.2
Dial Plan Syntax
> The phone will not perform any checking on the title; ensuring that different dial plans are given different titles is part of the administration process.
A dial plan rule is built from the parameters described underneath.
• Digit string: A pattern of digits or "*", "#", or "x" characters that is to be matched for starting an action. The maximum length is 24 characters. The "x" character is a wildcard character that represents any of the other digits (it may be upper or lower case).
• Action : The action to be taken when the criteria are met. The following options are available:
• "S" (Send digits): The digits entered are sent to the server when one of the following three conditions is satisfied: a) the maximum digits have been received, or b) the timer expires after the minimum digits have been received, or c) on receipt of the terminator after the minimum digits.
• "C" (Check for other actions): If the the digit sequence entered by the user matches
Digit string, Maximum length, and Minimum length, the timer starts. On timer expiry, the digit string will be sent to the server. If further digits are received before timer expiry, further entries will be checked.
If the timer is set to 0, the dial string will be sent immediately.
This option is used when there are more than one rules which start with the same digits.
• Minimum length: The dial plan rule will not initiate the sending of digits until at least this number of digits have been entered. However, the digits will be sent after the delay configured in User menu > Configuration > Outgoing calls > Autodial delay (seconds).
• Maximum length: Automatic sending will occur when this number of digits have been dialed. If not specified, then the digits will be sent when the timer expires, or a terminating character is entered.
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Dial Plan
• Timer: This indicates the timeout to be used for subsequent digit handling. If not specified, the default timer value is used (User menu > Configuration > Outgoing calls > Autodial delay (seconds)).
• Terminating character: A "*" or "#" character which indicates that the preceding digits should be considered complete, even though the maximum length may not be reached.
However, the reach the minimum length must be reached by the string built from the digits entered and the terminating characters.
• Special indication:
• "E" (Emergency): If this character is entered here, the digits matching this rule will be sent even if the phone is locked. The number will be dialed immediately even when immediate dialing is disabled, and the phone is on-hook.
• "b" (bypass): The phone lock is bypassed. The number will be dialed immediately even when immediate dialing is disabled, if the phone is off-hook.
• Comment: A remark on this dial plan entry.
• Terminator sent: If set to true, the terminating character is sent to the server along with the dial string proper. If set to false, the dial string is sent without the terminating character.
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Examples and HowTos
Dial Plan
5.5.3
How To Set Up And Deploy A Dial Plan
For creating and deploying a dial plan to an OpenStage phone, a working installation of the
DLS (version V2R4 onwards) is required. This HowTo describes the creation of a simple dial plan for OpenStage phones by example. Unless otherwise stated, the actions described underneath are made in the DLS.
1. Log on to the DLS with an account that has suitable rights for deploying a dial plan. For details, please refer to the Deployment Service Administration Manual.
2. Navigate to IP Devices > IP Phone Configuration > Features > "Dialplan" tab.
3. Check Dialplan, if not checked already.
4. Enter a suitable Dialplan ID.
5. Click on to create the first dial plan rule.
6. Enter the following data:
Parameter
Digit string
Action
Value
3
S
Description/Remarks
This rule matches numbers beginning with 3. For instance, theses might be internal numbers.
When all criteria are met, the number is sent to the server.
This rule matches numbers with a length of 4 digits.
Minimum length 4
Maximum length 4
Timer 0 The specified Action will take place without delay when all other criteria are met.
Summary: This rule determines that digit strings which begin with 3 and have a length of 4 digits are sent to the server without delay after the last digit has been entered.
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Dial Plan
7. Click on to create the second dial plan rule.
8. Enter the following data:
Parameter
Digit string
Action
Value
0
C
Description/Remarks
This rule matches numbers beginning with 0. In the
USA, this number calls the operator.
When Minimum length, Maximum length, and the length of the digit string entered by the user match, the
Timer is started. When it expires, the digits are sent to the server. When another digit is entered before expiry, the next dial plan entry will come into operation.
This rule matches numbers with a length of 1 digits.
Minimum length
1
Maximum length 1
Timer
1
The phone waits 1 second for further digits. If the user does not enter any further digits, the action specified in
Action is initiated.
Summary: When 0 is entered as first digit, the phone will wait 1 second. After this, 0 will be sent to the server, which might result in a call to an operator, for instance. When further digits are entered during the 1 second timespan, the next dial plan rule will take control.
9. Click on to create the third dial plan rule.
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Dial Plan
10. Enter the following data:
Parameter
Digit string
Action
Minimum length
Terminating
Character
Value
011
S
4
Maximum length 13
Timer
3
#
Description/Remarks
This rule matches numbers beginning with 011. In the
USA, this digit string is the prefix international calls.
When the entered digit string reaches the Minimum
length, the Timer is started. On expiry, the digit string is sent.
When the length of the digit sequence entered by the user reaches this value, the Timer is started.
When the length of the digit sequence entered by the user reaches this value, the digits are sent to the server immediately. The Timer is overridden.
When the length of the digit sequence entered by the user reaches the Minimum length, the phone waits 3 seconds for further digits. If the user does not enter any further digits, the Action is triggered.
When this character is entered, the digits are sent to the server immediately, regardless of the criteria contained in this rule.
Summary: Any numbers that start with 011 and have a length of 13 digits are sent to the server immediately. Shorter numbers with a length from 4 digits onwards are sent after a
3 seconds delay.
11. The example dial plan is completed; it should look like this:
12. You can check the dial plan using the phone’s web interface; navigate to Diagnostics >
Fault trace configuration > Download dial plan file.
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Glossary
Glossary
X
A
Address of Record (AoR)
A ->SIP ->URI that represents the "public address" of a SIP user resp. a phone or line. The
format is similar to an E-mail address: "username@hostname". (for a definition, see RFC
3261)
ADPCM
Adaptive Differential Pulse Code Modulation. A compressed encoding method for audio
signals which are to be transmitted by a low bandwidth. As opposed to regular ->PCM, a
sample is coded as the difference between its predicted value and its real value. As this difference is usually smaller than the real, absolute value itself, a lesser number of bits can be used to encode it.
C
CSTA
Computer Supported Telecommunications Applications. An abstraction layer for telecom-
munications applications allowing for the interaction of ->CTI computer applications with
telephony devices and networks.
CTI
Computer Telephony Integration. This term denotes the interaction of computer applications with telephony devices and networks.
D
DFT
Digital Feature Telephone. A phone with no line keys.
DHCP
Dynamic Host Configuration Protocol. Allows for the automatic configuration of network endpoints, like IP Phones and IP Clients.
DiffServ
Differentiated Services. Specifies a layer 3 mechanism for classifying and managing net-
work traffic and providing quality of service (->QoS) guarantees on ->IP networks. DiffServ
can be used to provide low-latency, guaranteed service for e. g. voice or video communication.
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DLS
The Deployment Service (DLS) is a OpenScape management application for the administration of workpoints, i. e. IP Phones and IP Clients, in both HiPath- and non-HiPath networks.
DNS
Domain Name System. Performs the translation of network domain names and computer
hostnames to ->IP addresses.
DTMF
Dual Tone Multi Frequency. A means of signaling between a phone and e. g. a voicemail facility. The signals can be transmitted either in-band, i. e. within the speech band, or outband, i. e. in a separate signaling channel.
E
EAP
Extensible Authentication Protocol. An authentication framework that is frequently used in
WLAN networks. It is defined in RFC 3748.
F
FTP
File Transfer Protocol. Used for transferring files in networks, e. g., to update telephone software.
G
G.711
ITU-T standard for audio encoding, used in ISDN and ->VoIP. It requires a 64 kBit/s band-
width.
G.722
ITU-T standard for audio encoding using split band ->ADPCM. The audio bandwidth is 7
kHz at a sampling rate of 16 kHz. There are several transfer rates ranging from 32 to 64 kBit/s, which correspond to different compression degrees. The voice quality is very good.
G.729
ITU-T standard for audio encoding with low bandwidth requirements, mostly used in VoIP.
The standard bitrate is 8 kBit/s. Music or tones such as ->DTMF or fax tones cannot be
transported reliably with this codec.
Gateway
Mediation components between two different network types, e. g., ->IP network and ISDN
network.
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Glossary
GUI
Graphical User Interface.
H
HTTP
Hypertext Transfer Protocol. A standard protocol for data transfer in ->IP networks.
I
IP
Internet Protocol. A data-oriented network layer protocol used for transferring data across a packet-switched internetwork. Within this network layer, reliability is not guaranteed.
IP address
The unique address of a terminal device in the network. It consists of four number blocks of 0 to 255 each, separated by a point.
J
Jitter
Latency fluctuations in the data transmission resulting in distorted sound.
L
LAN
Local Area Network. A computer network covering a local area, like an office, or group of buildings.
Layer 2
2nd layer (Data Link Layer) of the 7-layer OSI model for describing data transmission interfaces.
Layer 3
3rd layer (Network Layer) of the 7-layer OSI model for describing the data transmission interfaces.
LCD
Liquid Crystal Display. Display of numbers, text or graphics with the help of liquid crystal technology.
LDAP
Lightweight Directory Access Protocol. Simplified protocol for accessing standardized directory systems, e.g., a company telephone directory.
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LED
Light Emitting Diode. Cold light illumination in different colours at low power consumption.
LLDP
Link Layer Discovery Protocol (IEEE Standard 802.1AB). Provides a solution for the discovery of elements on a data network and how they are connected to each other.
M
MAC Address
Media Access Control address. Unique 48-bit identifier attached to network adapters.
MDI-X
Media Dependent Interface crossover (X). The send and receive pins are inverted. This
MDI allows the connection of two endpoints without using a crossover cable. When Auto
MDI-X is available, the MDI can switch between regular MDI and MDI-X automatically, depending on the connected device.
MIB
Management Information Base. A type of database used to manage the devices in a communications network.
MWI
Message Waiting Indicator. A signal, typically a LED, to notify the user that new mailbox messages have arrived.
O
OSC
OpenScape Phone
P
PBX
Private Branch Exchange. Private telephone system that connects the internal devices to each other and to the ISDN network.
PCM
Pulse Code Modulation. A digital representation of an analog signal, e. g. audio data, which consists of quantized samples taken in regular time intervals.
PING
Packet Internet Gro(u)per. A program to test whether a connection can be made to a defined IP target. Data is sent to the target and returned from there during the test.
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Glossary
PoE
Power over Ethernet. The IEEE 802.3af standard specifies how to supply power to compliant devices over Ethernet cabling (10/100Base-T).
Port
Ports are used in ->IP networks to permit several communication connections simulta-
neously. Different services often have different port numbers.
PSTN
Public Switched Telephone Network. The network of the world's public circuit-switched telephone networks.
Q
QoS
Quality of Service. The term refers to control mechanisms that can provide different priority to different users or data flows, or guarantee a certain level of performance to a data flow in accordance with requests from the application program. The OpenStage phone allows for the setting of QoS parameters on layer 2 and layer 3 (DiffServ).
QDC
QoS Data Collection. A HiPath IP service that is used to collect data from HiPath products in order to analyze their voice and network quality.
QCU
Quality of Service Data Collection Unit. A service tool that collects QoS report data from IP endpoints.
QoS
Quality of Service. Provides different priority to different users or data flows, or guarantee a certain level of performance to a data flow.
R
RAM
Random Access Memory. Memory with read / write access.
ROM
Read Only Memory. Memory with read only access.
RTCP
Realtime Transport Control Protocol. Controls the ->RTP stream and provides information
about the status of the transmission, like QoS parameters.
RTP
Realtime Transport Protocol. This application layer protocol has been designed for audio
and video communication. Typically, the underlying protocol is ->UDP.
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S
SDP
Session Description Protocol. Describes and initiates multimedia sessions, like web con-
ferences. The informations provided by SDP can be processed by ->SIP.
SIP
Session Initiation Protocol. Signaling protocol for initialising and controlling sessions, used
e. g. for ->VoIP calls.
SNMP
Simple Network Management Protocol. Used for monitoring, controlling, and administration of network and network devices.
SNTP
Simple Network Time Protocol. Used to synchronize the time of a terminal device with a timeserver.
Subnet Mask
To discern the network part from the host part of an ->IP address, a device performs an
AND operation on the IP address and the network mask. The network classes A, B, and C each have a subnet mask that demasks the relevant bits: 255.0.0.0 for Class A,
255.255.0.0 for Class B and 255.255.255.0 for Class C. In a Class C network, for instance,
254 IP addresses are available.
Switch
Network device that connects multiple network segments and terminal devices. The for-
warding of data packets is based on ->MAC Addresses: data targeted to a specific device
is directed to the switch port that device is attached to.
T
TCP
Transfer Control Protocol. The protocol belongs to the transport layer and establishes a connection between two entities on the application layer. It guarantees reliable and in-or-
der delivery of data from sender to receiver, as opposed to ->UDP.
TLS
Transport Layer Security. Ensures privacy between communicating applications. Typically, the server is authenticated, but mutual authentication is also possible.
U
UDP
User Datagram Protocol. A minimal message-oriented transport layer protocol used espe-
cially in streaming media applications such as ->VoIP. Reliability and order of packet de-
livery are not guaranteed, as opposed to ->TCP, but ->UDP is faster and more efficient.
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Glossary
URI
Uniform Resource Identifier. A compact string of characters used to identify or name a resource.
URL
Uniform Resource Locator. A special type of ->URI which provides means of acting upon
or obtaining a representation of the resource by describing its primary access mechanism or network location.
V
VLAN
Virtual Local Area Network. A method of creating several independent logical networks within a physical network. For example, an existing network can be separated into a data and a voice VLAN.
VoIP
Voice over IP. A term for the protocols and technologies enabling the routing of voice con-
versations over the internet or through any other ->IP-based network
W
WAP
Wireless Application Protocol. A collection of protocols and technologies aiming at enabling access to internet applications for wireless devices. WAP can also be used by the
OpenStage phone.
WBM
Web Based Management. A web interface which enables configuration of the device using a standard web browser.
WML
Wireless Markup Language. An XML-based markup language which supports text,
graphics, hyperlinks and forms on a ->WAP-browser.
WSP
Wireless Session Protocol. The protocol is a part of the ->WAP specification. Its task is to
establish a session between the terminal device and the WAP gateway.
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Index
Zahlen
2nd Alert 1-168
A
Access Control 1-87
Address of Record (AoR) 1-401
Admin Menu (Local Menu) 1-50, 1-51
Admin Password 1-91
Allow in overview 1-172
Alternate 1-155
Application
Modify 1-254
Remove 1-255
Start 1-255
Audio Keys 1-14, 1-15, 1-16, 1-18, 1-19
Audio Settings 1-245
Authenticated Registration 1-109
Automatic VLAN discovery 1-57
B
Backlight time 1-99
Backup SIP Server 1-121
Blind Transfer 1-156
Bluetooth 1-303
Advanced Traces 1-292
Bridging enabled 1-179
Bridging priority 1-183
Built-in Forwarding 1-168
Busy Status 1-160
C
Call
Accept via Headset 1-158
Call Forwarding 1-152, 1-153
Call Recording 1-148
Call Transfer 1-135
Call Waiting 1-162
Callback 1-137, 1-161
Cancel Callbacks 1-162
Canonical Dial Lookup 1-198
Z
Canonical Dialing 1-194
CCE access 1-87
Certificate Policy 1-94
Character Set 1-92
Cloud deployment 1-45
Cloud service provider, 1-45
Codec Preferences 1-243
Conference
Phone-Based 1-157
System based 1-142
Connectivity Check (TLS) 1-117
Connectors 1-22
Consult 1-162
Consultation 1-162
Core dump 1-300
Corporate Phonebook 1-240
CSTA 1-145, 1-401
CTI 1-401
D
Date and Time (SNTP) 1-28, 1-103
Daylight Saving 1-103
Default Route 1-71
Deflect a Call 1-157
Deployment errors 1-49
DFT (Digital Feature Telephone) 1-401
DHCP 1-28, 1-66, 1-401
Diagnostic 1-261
Dial Plan 1-201, 1-395
Dialing
Repeat 1-152
Selected 1-151
Diffserv 1-62
Direct Station Select (DSS) 1-185
Directory Settings 1-240
Display Identity 1-96
Distinctive Ringing 1-204
DLS (Deployment Service) 1-20, 1-78, 1-402
DLS Address 1-29
DNS 1-75, 1-402
Domain Name 1-75
Primary/Secundary 1-76
Servers 1-76
Do Not Disturb (DND) 1-158
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Index
Dongle Key (Download) 1-237
Download 1-217
DSS key settings 1-187
DST Zone (Daylight Saving Time Zone) 1-
104
DTMF playback 1-245
E
Easy Trace Profiles 1-278
802.1x 1-290
Bluetooth
Headset 1-279
Bluetooth Handsfree 1-278
Call Connection 1-279
Call Log 1-280
DAS Connection 1-281
DLS Data Errors 1-281
Help Application 1-282
Key Input 1-282
LAN Connectivity 1-283
LDAP Phonebook 1-285
Local Phonebook 1-285
Mobility 1-284
No Tracing for All Services 1-291
Phone administration 1-284
Server based applications 1-286
Sidecar 1-286
Speech 1-288
Tone 1-288
USB Backup/Restore 1-289
Voice Dialling 1-289
Web Based Management 1-290
Emergency Number 1-98, 1-194
Energy Saving 1-99
Error Codes 1-374
External Access Code 1-195
External Numbers 1-195
F
Factory Reset 1-257
Factory reset 1-87
Factory reset claw 1-87
Fault Trace Configuration 1-271
Feature Access 1-124
Features
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Server Based 1-143
Fixed Function Keys 1-170
Forward indication 1-178
Forwarding 1-152, 1-153
FPK program timer 1-150
FTP Settings 1-212
Function Keys 1-14, 1-18, 1-19
G
G.711 1-243
G.722 1-243
G.729 1-243
Gateway 1-71
General Configuration 1-83
General Information 1-260
General IP configuration 1-69
Graphics Display 1-14, 1-15, 1-16
Group Pickup 1-132
H
Handset 1-14, 1-15, 1-16, 1-18, 1-19
Hold 1-155
Hostname 1-77
Hot Phone 1-129
Hot warm
action 1-172 destination 1-172
HPT Interface 1-302
HTTP Proxy 1-252
HTTPS Settings 1-212
Hunt Group 1-160
I
Identity
Display 1-96
Terminal and User 1-96
Immediate Ring 1-185
Initial Digit Timer 1-131
Initial digit timer 1-129
Initial Digits 1-195
Internal Numbers 1-195
International Code (Local Country Code) 1-
194
International Gateway Code 1-195
International Prefix (International Access Co-
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de) 1-194
IP 1-403
Address 1-27, 1-69
Specific Routing 1-73
IPv4/IPv6 configuration 1-69
J
Join Two Calls 1-156
K
Key Modules 1-192
Keypad 1-14, 1-15, 1-16, 1-18, 1-19
Keys
programmable 1-150
Keyset Operation 1-177
L
LAN 1-403
Monitoring 1-265
Port 1-54
Layer 2 1-61
Layer 3 1-62
LDAP 1-240, 1-384, 1-403
LDAP Template (Download) 1-224
License Information 1-259
Line action mode 1-178
Line Key Configuration 1-171
Line Preview 1-183
LLDP-MED 1-57, 1-60, 1-266
Local Area Code (Local National Code) 1-
194
Local Country Code (International Code) 1-
194
Local Enterprise Number 1-194
Local National Code (Local Area Code) 1-
194
Local Phone Menu 1-360
Logo (Create) 1-380
Logo (Download) 1-227
M
MAC Address 1-404
MDI-X 1-54, 1-404
Media/SDP 1-113
Memory Information 1-269
410
Messages settings 1-140
MIB 1-404
MIKEY (Multimedia Internet KEYing) 1-84
Missed Call LED 1-307
Mobile User 1-160
Mobility 1-210
Monitoring 1-265
Multiline / Keyset 1-171
Multiline Appearance/Keyset 1-171
Music on Hold (Download) 1-218
Mute 1-169
MWI 1-139
(Message Waiting Indicator) 1-404
MWI LED 1-305
N
National Prefix (Trunk Prefix) 1-194
Navigation keys 1-19, 1-52
Navigator 1-18, 1-51
Network port configuration 1-55
NonCall trans 1-119
Non-INVITE 1-119
O
OCSP 1-94
OCSR failure 1-89
OpenScape Voice (Registration) 1-43
Operator Code 1-194
Originating line preference 1-177
Outbound Proxy 1-111
P
Password
Admin 1-91
Change 1-256
Enter 1-50
Lost 1-257
Policy 1-90
User 1-92
PBX 1-404
PC port 1-54
Phone
Restart 1-257
Software (Download) 1-214
Phone Menu 1-360
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Index
Phonebook 1-240
Pickup alert 1-132
Picture Clips (Download) 1-221
PoE (Power over Ethernet) 1-24, 1-405
Port configuration 1-55
Port List 1-372
Power Consumption/Supply 1-24
Preselect
timer 1-178
Preselect mode 1-178
Preview and Preselection 1-184
Preview mode 1-183
Preview timer 1-183
Process
Information 1-269
Program timer (FPK) 1-150
Programmable Keys 1-150
Protocol Mode IPv4/IPv6 1-65
PSTN 1-405
PSTN Aaccess Code 1-194
Q
QCU 1-81
QoS 1-61
QoS Reports 1-293
Quick Start 1-26
R
Realm 1-172
Refuse 1-127
Registration 1-43
Authenticated 1-109
Registration Backoff Timer 1-120
Release 1-169
Remote Tracing – Syslog 1-301
Repeat Dialing 1-152
Repertory Dial 1-159
Reservation timer 1-178
Reset Factory 1-257
Resilience 1-116
Response Timer 1-118
Restart Phone 1-257
Ringer
Off 1-155
Ringer File 1-233
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RTP 1-405
Base Port 1-242
S
Screensaver (Download) 1-230
SDES
Configuration 1-86
SDES status 1-86
SDP negotiation 1-86
Secure
file transfer 1-95
SIP server 1-95
Secure calls 1-83, 1-84
Security
log entry 1-89
Security Log 1-88
Selected Dialing 1-151
Send Request 1-165
Server
Authentication Policy 1-95
Server Based Features 1-143
Shared type 1-172
Shift Level 1-157
Shipment 1-22
Show Focus 1-178
Show phone screen 1-169
Silence suppression 1-243
SIP
Addresses 1-106
Ports 1-106
Registration 1-109
Server Address 1-29
Server Addresses 1-106
Server Ports 1-108
Session Timer 1-114
Transport Protocol 1-112
SNMP 1-80, 1-406
SNTP 1-103
SRTP Type 1-84
SRTP type 1-83, 1-84
SSH – Secure Shell Access 1-258
Start Phonebook 1-168
Startup Procedure 1-44
Subnet Mask 1-27
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Survivability 1-116
T
TCP 1-406
Terminal
Hostname 1-77
Number 1-27, 1-96
Terminal Identity 1-96
Terminating line preference 1-177
Timeout (Not used) 1-147
Timer
FPK programming 1-150
Timezone Offset 1-28, 1-103
TLS 1-406
Connectivity Check 1-117
TouchGuide 1-14, 1-15, 1-16, 1-51
TouchSlider 1-14
Trace Configuration 1-271
Trace Profiles 1-278
Transaction timer 1-119
Transfer on hangup 1-135
Transfer on Ring 1-135
Traps 1-80
Trunk Prefix (National Prefix) 1-194
U
uaCSTA 1-145
UDP 1-406
Unauthenticated RegistrationRegistration
Unauthenticated 1-109
Update Service 1-78
Use SRTCP 1-84
User Identifier 1-172
User Identity 1-96
User Password 1-92
V
Vendor Class (DHCP) 1-30
View Report 1-296
VLAN 1-29, 1-56
VLAN ID configuration 1-59
Voice Mail Number 1-98
W
Warm Phone 1-129
412
WBM (Web Based Management) 1-20, 1-26,
1-407
Web Interface Menu 1-311
X
XML ApplicationsApplications
XML 1-246
Xpressions 1-246
Z
Zip Tone 1-164
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
