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
Nur für den internen Gebrauch
1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.1 Important Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.2 Maintenance Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.3 About the Manual. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.4 Conventions for this Document . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5 The OpenStage Family . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5.1 OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5.2 OpenStage 40 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5.3 OpenStage 40 US . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5.4 OpenStage 20 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.5.5 OpenStage 15 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.6 Administration Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.6.1 Web-based Management (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.6.2 DLS (Deployment Service) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
1.6.3 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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2 Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2 Assembling and Installing the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2.1 Shipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2.2 Connectors at the bottom side . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2.3 Assembly. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.2.4 Connecting the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3 Quick Start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.1 Accessing the Web Interface (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.2 Set the Terminal Number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.3 Basic Network Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.4 DHCP Resilience. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.5 Date and Time / SNTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.6 SIP Server Address. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.7 Extended Network Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.8 Vendor Specific: VLAN Discovery And DLS Address . . . . . . . . . . . . . . . . . . . . . . .
2.3.8.1 Using a Vendor Class . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.8.2 Using Option #43 "Vendor Specific" . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.3.9 Registering at Phone Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.4 Startup Procedure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5 Cloud Deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5.1 Process of Cloud Deployment. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5.2 Aborting cloud deployment process by User. . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5.3 Re-trigger cloud deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
2.5.4 Deployment errors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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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 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.2.2 VLAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.2.2.1 Automatic VLAN discovery using LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . .
3.2.2.2 Automatic VLAN discovery using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.2.2.3 Manual configuration of a VLAN ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.2.3 LLDP-MED Operation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3 IP Network Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.1 Quality of Service (QoS). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.1.1 Layer 2 / 802.1p . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.1.2 Layer 3 / Diffserv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.2 Protocol Mode IPv4/IPv6 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.3 Use DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.4 IP Address - Manual Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.4.1 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.5 Default Route/Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.6 Specific IP Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.7 DNS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.7.1 DNS Domain Name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.7.2 DNS Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.7.3 Terminal Hostname . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.8 Configuration & Update Service (DLS). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.3.9 SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.1 Speech Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.1.1 General Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.1.2 MIKEY Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.1.3 SDES Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.2 Access Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.3 Security Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.4 Security-Related Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5 Password Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5.1 General Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5.2 Admin Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5.3 User Policy. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5.4 Character Set. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.5.5 Change Admin and User password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.6 Certificate Policy. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.6.1 Online Certificate Check . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.4.6.2 Server Authentication Policy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.5 System Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.5.1 Terminal and User Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.5.1.1 Terminal Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.5.1.2 Display Identity. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.5.2 Emergency and Voice Mail. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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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 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.4 Call Forwarding (Standard) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.5 Call Forwarding by Call Type . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.6 Ringer Off . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.7 Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.8 Alternate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.9 Blind Call Transfer / Move Blind . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.10 Transfer Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.11 Deflect a Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.12 Shift Level. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.13 Phone-Based Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.14 Accept Call via Headset (OpenStage 40/60/80). . . . . . . . . . . . . . . . . . . . . . . . .
3.8.15 Do Not Disturb . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.16 Group Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.17 Repertory Dial. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.18 Hunt Group: Send Busy Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.19 Mobile User Logon . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.20 Directed Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.21 Callback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.22 Cancel Callbacks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.23 Consultation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.24 Call Waiting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.25 Call recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.26 Auto Answer With Zip Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.27 Server Feature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.28 BLF Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.29 Start Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.30 Send Request via HTTP/HTTPS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.31 Built-in Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.32 2nd Alert . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.33 Start Phonebook (OpenStage 40/15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.34 Show phone screen (OpenStage 15 and OpenStage 40 only) . . . . . . . . . . . . .
3.8.35 Mute (OpenStage 15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.8.36 Release (OpenStage 15 only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.9 Preset Function Keys (OpenStage 40 US only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.10 Fixed Function Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.10.1 Fixed Function Keys on OpenStage 40 US . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11 Multiline Appearance/Keyset. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.1 Line key configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.2 Configure Keyset Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.3 Line Preview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.3.1 Preview and Preselection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.4 Immediate Ring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.5 Direct Station Select (DSS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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3.11.5.1 General DSS Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.5.2 Settings for a DSS key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.11.6 Distinctive Ringers per Keyset Lines. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.12 Key Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.13 Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.13.1 Canonical Dialing Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.13.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.13.3 Phone location . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.13.4 Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.14 Ringer Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.14.1 Distinctive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.14.2 Map to Specials. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.14.3 Special Ringers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.15 Mobility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16 Transferring Phone Software, Application, and Media Files . . . . . . . . . . . . . . . . . . .
3.16.1 FTP/HTTPS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.2 Common FTP/HTTPS Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.3 Phone Software. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.3.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.3.2 Download/Update Phone Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.4 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.4.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.4.2 Download Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.5 Picture Clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.5.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.5.2 Download Picture Clip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.6 LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.6.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.6.2 Download LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.7 Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.7.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.7.2 Download Logo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.8 Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.8.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.8.2 Download Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.9 Ringer File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.9.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.9.2 Download Ringer File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.10 Dongle Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.10.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.16.10.2 Download Dongle Key File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.17 Corporate Phonebook: Directory Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.17.1 LDAP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.18 Speech. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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3.18.1 RTP Base Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.18.2 Codec Preferences. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.18.3 Audio Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19 Applications. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19.1 XML Applications/Xpressions (OpenStage 60/80) . . . . . . . . . . . . . . . . . . . . . . .
3.19.1.1 Setup/Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19.1.2 HTTP Proxy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19.1.3 Modify an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19.1.4 Remove an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.19.1.5 Application Start by Programmable Key . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.20 Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.21 Troubleshooting: Lost Password. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.22 Restart Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.23 Factory Reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.24 SSH – Secure Shell Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.25 Display License Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26 Diagnostics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.1 Display General Phone Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.2 View Diagnostic Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.3 User Access to Diagnostic Information. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.4 Diagnostic Call . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.5 LAN Monitoring. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.6 LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.7 IP Tests . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.8 Process and Memory Information. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.9 Fault Trace Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10 EasyTrace Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.1 Bluetooth Handsfree . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.2 Bluetooth Headset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.3 Call Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.4 Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.5 Call Recording . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.6 DAS Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.7 DLS Data Errors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.8 Help Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.9 Key Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.10 LAN Connectivity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.11 Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.12 Mobility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.13 Phone administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.14 LDAP Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.15 Local Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.16 Server based applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.17 Sidecar. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
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260
260
261
263
263
265
266
268
269
271
278
278
279
279
280
280
281
281
282
282
283
283
284
284
285
285
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3.26.10.18 SIP standard multiline. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.19 SIP standard singleline . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.20 Speech . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.21 Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.22 USB Backup/Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.23 Voice Dialling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.24 Web Based Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.25 802.1x problems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.10.26 No Tracing for All Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.11 Bluetooth Advanced Traces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.12 QoS Reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.12.1 Conditions and Thresholds for Report Generation . . . . . . . . . . . . . . . . . . .
3.26.12.2 View Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.13 Core dump . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.14 Remote Tracing – Syslog . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.26.15 HPT Interface (For Service Staff) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.27 Bluetooth (OpenStage 60/80) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.28 MWI LED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.29 Missed Call LED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
3.30 Impact Level Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
287
287
288
288
289
289
290
290
291
292
293
293
296
300
301
302
303
305
307
308
4 Technical Reference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.1 Menus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.1.1 Web Interface Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.1.1.1 Menu Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.1.1.2 Web Pages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.1.2 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.2 Default Port List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
4.3 Troubleshooting: Error Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
311
311
311
311
315
360
372
374
5 Examples and HowTos. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.1 Canonical Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.1.1 Canonical Dialing Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.1.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.1.2.1 Conversion examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.2 How to Create Logo Files for OpenStage Phones. . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.2.1 For OpenStage 40. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.2.2 For OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3 How to Set Up the Corporate Phonebook (LDAP). . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3.2 Create an LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3.3 Load the LDAP Template onto the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3.4 Configure LDAP Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.3.5 Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.4 An LLDP-Med Example . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
376
376
376
377
378
380
380
381
384
384
385
389
390
390
393
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5.5 Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.5.1 Introduction. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.5.2 Dial Plan Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
5.5.3 How To Set Up And Deploy A Dial Plan. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
395
395
395
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
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.
7
1.3
About the Manual
The instructions within this manual will help you in administering and maintaining the OpenStage 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.
12
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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
The OpenStage Family
1.5.1
OpenStage 60/80
9
5
2
1
6
3
7
8
4
1
1
With the handset, the user can pick up and dial calls in the usual manner.
2
The graphic display provides intuitive support for telephone operation.
3
The mode keys provide easy access to the phone’s applications.
4
With the TouchGuide, the user/administrator can navigate in the phone functions, applications, and configuration menus.
5
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.
6
The fixed function keys provide access to frequently used telephony functions.
7
With the audio keys, the user can control the audio settings.
8
With the TouchSlider, the user can adjust the volume, e.g. of ringtones.
9
Inbound calls are visually signaled via the alert bar.
10
The keypad is used for entering phone numbers and text.
Tabelle 1-1
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Overview
The OpenStage Family
1.5.2
OpenStage 40
7
5
2
1
6
3
4
8
1
With the handset, the user can pick up and dial calls in the usual manner.
2
The graphic display provides intuitive support for telephone operation.
3
The fixed function keys provide access to frequently used telephony functions.
4
With the 5-way navigator, the user/administrator can navigate in the various
phone functions, applications, and configuration menus.
5
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.
6
With the audio keys, the user can control the audio settings.
7
Inbound calls are visually signaled via the alert bar.
8
The keypad is used for entering phone numbers and text.
Tabelle 1-2
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Overview
The OpenStage Family
1.5.3
OpenStage 40 US
7
5
2
1
6
3
4
8
1
With the handset, the user can pick up and dial calls in the usual manner.
2
The graphic display provides intuitive support for telephone operation.
3
The fixed function keys provide access to frequently used telephony functions.
4
With the 5-way navigator, the user/administrator can navigate in the various
phone functions, applications, and configuration menus.
5
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.
6
With the audio keys, the user can control the audio settings.
7
Inbound calls are visually signaled via the alert bar.
8
The keypad is used for entering phone numbers and text.
Tabelle 1-3
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Overview
The OpenStage Family
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Overview
The OpenStage Family
1.5.4
OpenStage 20
2
1
3
5
4
6
1
With the handset, the user can pick up and dial calls in the usual manner.
2
The display provides intuitive support for telephone operation.
3
The fixed function keys provide access to frequently used telephony functions.
4
With the audio keys, the user can control the audio settings.
5
With the 3-way navigator, the user/administrator can navigate in the various
phone functions, applications, and configuration menus.
6
The keypad is used for entering phone numbers and text.
Tabelle 1-4
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Overview
The OpenStage Family
1.5.5
OpenStage 15
1
2
3
4
5
6
7
1
With the handset, the user can pick up and dial calls in the usual manner.
2
The display provides intuitive support for telephone operation.
3
With the audio keys, the user can control the audio settings.
4
The fixed function keys provide access to frequently used telephony functions.
5
The keypad is used for entering phone numbers and text.
6
With the navigation keys, the user/administrator can navigate in the various
phone functions, applications, and configuration menus.
7
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.
Tabelle 1-5
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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.
>
1.6.2
To use this method, the phone must first obtain IP connectivity.
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.
>
20
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
Power supply
USB
Stick
PC
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
1.
Assembly
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
1.
Connecting the Phone
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
Power Consumption/Supply
OpenStage 151
OpenStage 15
Power Class 1
G1
Power Class 2
OpenStage 20 E
Power Class 1
OpenStage 20
Power Class 1
OpenStage 20 G
OpenStage
402,
Power Class 2
OpenStage 40
US2
Power Class 2
OpenStage 40 + 2nd Key Module
Power Class 2
OpenStage 40 G2,OpenStage 40 G US2
Power Class 3
OpenStage 40 G or OpenStage 40 G US + 2nd Key Power Class 3
Module
OpenStage 60/803
Power Class 3
OpenStage 60/80 + 2nd Key Module
OpenStage 60/80 G
3
Power Class 3
OpenStage 60/80 G + 2nd Key Module
1
2
3
2.
Power Class 3
External power unit required
Includes 1 Key Module 15.
Includes 1 Key Module.
Includes 1 Key Module + USB-Extension with Acoustic Unit.
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.
24
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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
1.
Accessing the Web Interface (WBM)
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:
Reset
Login
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
4711
Display identity
4711
Enable ID
Web name
DNS name construction
Submit
2.3.3
openstage
;
Only number
Reset
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 address and subnet mask, and Section 3.3.5, “Default Route/Gateway” for the default route.
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2.3.4
DHCP Resilience
It is possible to sustain network connectivity in case of DHCP server failure. If DHCP lease reuse 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 192.168.1.235
Subnet mask 255.255.255.0
Default route 192.168.1.2
Route 1 IP address
Route 1 gateway
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Submit
2.3.5
Reset
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 Predefined 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
Length
Vendor name
1
7
S
i
e
m
e
n
s
01
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
Length
VLAN ID
2
4
0
0
1
0
02
04
00
00
00
0A
For manual configuration of the VLAN ID see Section 3.2.2.3, “Manual configuration of a VLAN
ID”.
Code
Length DLS IP address
3
25
s d l
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
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:
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.
42
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 interface (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
192.168.1.165
SIP registar Address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
HiQ8000
Realm
User ID
Password
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
3600
Backup transport
UDP
Backup OBP flag
Submit
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 address.
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
No
802.1x
credential
Yes
Start 802.1x
Suppplicant
in Proxy-EAPOLLogoff mode
802.1x enabled on
authenticator
Start 802.1x
Suppplicant
in auth. mode
No
802.1x enabled on
authenticator
Yes
Yes
No
Manual
configured
VLAN-ID
No
Yes
No networkaccess
802.1x
auth.
Yes
No
LLDP-Med
assigns
VLAN-ID
Yes
No networkaccess
YesManual
configured
VLAN-ID
No
GuestVLAN
or
AuthfailVLAN No
No
LLDP-Med
assigns
VLAN-ID
DHCP
assigns
VLAN-ID
Yes
No
Yes
No
No networkaccess
GuestVLAN
orNo
untagged VLAN
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.
Manual
configured
IP addresses
DHCP
assigns
VLAN-ID
Yes
No
No
DHCP
assigned
IP addresses
Yes
No
L3 not
configured
GuestVLAN
Yes
Contact DLS
and
get conf. data
L2
conf. data
changed
VoiceVLAN
Conf. finished
No
OpenStage boot sequence using VoiceVLAN
and no DHCP reuse
Yes
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Startup
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 server1 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
The phone has all the information
that it needs to contact a DNS server. A DLS address is not provided.
A DHCP server responds
with IP addresses
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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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
46
•
•
DLS address (set to the name of
the Deployment server)
Language optional)
DLS port (optional)
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Phone saves the
configuration data
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.
Phone contacts the Deployment server using DLS-WPI
The stored DLS address is not
changed by the result of the DNS
lookup
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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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
Priority
Cause
AU
1
Abandoned by user
Occurs when the pin prompt is dismissed
RS
1
Unable to get the address for the Unify Redirect server
DNS lookup failed
RN
3
Unable to establish contact with Unify Redirect server – no reply
RR
2
Unable to establish contact with Unify Redirect server – refused
RU
1
Unable to establish contact with Unify Redirect server - unauthorised
RO
3
Unable to establish contact with Unify Redirect server - no or invalid
OCSP response
RV
2
Unable to establish contact with Unify Redirect server - certificate revoked
DS
1
Unable to get the address for the Deployment server
DNS lookup failed
DN
3
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 interface, 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 password (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 i key:
- confirm entries
- perform an action
Press the h key:
- cancel a function
- delete character left
of cursor
- up one level
Press the g key:
- open a context menu
- down one level
Press the l key briefly:
- scroll down
Hold down:
- scroll to end of list
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 mkey 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 i key:
- confirm entries
- perform an action
Press the g key:
- open a context menu
- down one level
Press the l key briefly:
- scroll down
Hold down:
- scroll to end of list
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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:
52
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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LAN Settings
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
•
•
•
•
54
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
Port configuration
SIP Server
5060
SIP registrar
5060
SIP gateway
5060
SIP local
5060
Backup proxy
5060
RTP base
5010
Download server (default)
LDAP server
HTTP proxy
21
389
0
LAN port speed
Automatic
PC port speed
PC port mode
PC port autoMDIX
Automatic
Submit
disabled
Reset
Administration via Local Phone
|---
Admin
|--- Network
|--- Port Configuration
|--- LAN port type
|--- PC port status
|--- PC port type
|--- PC port autoMDIX
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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 IPv4_IPv6
LLDP-MED Enabled
;
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery LLDP-MED
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
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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LAN Settings
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery DHCP
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
Reset
Administration via Local Phone
To enable VLAN discovery via DHCP, activate the DHCPv6 Enabled checkbox and select
DHCP in the VLAN discovery option.
|---
58
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery Manual
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
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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LAN Settings
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
|---
60
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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IP Network Parameters
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.
>
3.3.1.1
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.
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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IP Network Parameters
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 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 EF
Flash EF
Flash override EF
Submit
Reset
Administration via Local Phone
|---
64
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery Manual
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
Reset
Administration via Local Phone
|
--- Admin
|
--- Network
|--- General IP Configuration
|--- Protocol Mode
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IP Network Parameters
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
•
•
66
Primary/Secondary DNS (DNS recursive name server option 23)
SNTP IP Address (Simple Network Time Protocol Server option 31)
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IP Network Parameters
•
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery DHCP
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery DHCP
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
Reset
Administration via Local Phone
|---
68
Admin
|--- Network
|--- IPv6 configuration
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IP Network Parameters
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 configuration depending on settings in Section 3.3.2, “Protocol Mode IPv4/IPv6”. Enter the IP address and the Subnet mask. If applicable, enter the Default route. Afterwards, click Submit.
IPv6 configuration
IPv4 configuration
LLDP-MED Enabled
LLDP-MED Enabled
DHCP Enabled
DHCP lease reuse
DHCPv6 Enabled
DHCPv6 lease reuse
IP address 192.168.1.235
Global Address
Subnet mask 255.255.255.0
Default route 192.168.1.2
Global Address Prefix Len
Route 1 Dest.
Route 1 gateway
Route 1 Prefix Len
Route 1 mask
Route 1 Gateway
Route 2 IP address
Route 2 Dest.
Route 2 gateway
Route 2 Prefix Len
Route 2 mask
Submit
3.
Global Gateway
Link Local Address
Route 1 IP address
Reset
Route 2 Gateway
Submit
Reset
After the phone’s network service has restarted, the other IP parameters can be
configured.
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General IP configuration
Protocol Mode IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
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VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
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 192.168.1.235
Subnet mask 255.255.255.0
Default route 192.168.1.2
Route 1 IP address
Route 1 gateway
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
|---
72
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 192.168.1.235
Subnet mask 255.255.255.0
Default route 192.168.1.2
Route 1 IP address
Route 1 gateway
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 IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery Manual
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
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 configuration 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 Submit.
General IP configuration
Protocol Mode IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
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VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Submit
Reset
Administration via Local Phone
|---
76
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
4711
Display identity
4711
Enable ID
Web name
DNS name construction
Submit
openstage
;
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 optiPoint 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
192.168.1.242
18443
300
Revert to default security
Mode
Security PIN
Submit
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
•
•
•
•
80
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
162
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
12010
*****
QoS togeneric destination
Submit
Reset
Administration via Local Phone
|---
82
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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3.4
Security
3.4.1
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
MIKEY
Use SRTCP
Submit
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
MIKEY
Use SRTCP
Submit
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 > Security > 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
SRTP + RTP
SHA1-80 ranking
X
SHA1-32 ranking
Submit
86
Disabled
SDP negotiation
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
88
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 20111009-2206
OCSR Failure
Admin access
User access
Cancel faults All
Submit
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)
99
1
Force changed
Tries allowed
5
No change for (hours)
0
Suspended for (mins)
5
History valid for (days)
0
Submit
90
Reset
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Security
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 2038-01-19T03:14:07+00:00
Minimum length
Password history
Current status
Submit
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 2038-01-19T03:14:07+00:00
Minimum length
6
Password history
0
Current status
Submit
3.4.5.4
Active
Reset
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.
0
Lcase chars reqd.
0
Digits required
0
Special chars reqd
0
Bar repeat length
0
Min char difference
0
Reset
Submit
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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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
94
Reset
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Security
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/HTTPS”).
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
None
Secure send URL
None
Secure SIP server
None
Secure 802.1x
None
XML Applications
None
Submit
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
System Settings
3.5.1
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
4711
Display identity
4711
Enable ID
Web name
DNS name construction
Submit
openstage
;
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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Administration via WBM
System > System Identity
System Identity
Terminal number
Terminal name
4711
Display identity
4711
Enable ID
Web name
DNS name construction
Submit
openstage
;
Only number
Reset
Administration via Local Phone
|---
Admin
|--- System
|
--- Identity
|--- Display identity
|--- Enable ID
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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 configured 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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
alert-internal
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
98
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
100
Reset
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System Settings
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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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.
>
3.5.5.1
Please note that Difference (minutes) must be specified both for manual and automatic DST toggling.
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
Date and time
Time source
SNTP IP address
192.43.244.18
Timezone offset (hours)
1
Daylight saving
Daylight saving
Difference (minutes)
Auto time change
DST zone
60
;
Europe (Rest)
Submit
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): 30
Allow daylight saving:
Difference (minutes):
: 44
15
November
2006
87678
Submit
Reset
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 address 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 configuration is required. For details, please refer to Section 3.5.10, “Resilience and
Survivability”.
Data required
•
•
•
106
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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Administration via WBM
System > Registration
Registration
SIP Addresses
SIP server address
192.168.1.165
SIP registar address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
OS Voice
Realm
User ID
Password
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
3600
Backup transport
UDP
Backup OBP flag
Submit
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
Port configuration
SIP Server
5060
SIP registrar
5060
SIP gateway
5060
SIP local
5060
Backup proxy
5060
RTP base
5010
Download server (default)
LDAP server
HTTP proxy
21
389
0
LAN port speed
Automatic
PC port speed
PC port mode
PC port autoMDIX
Automatic
Submit
disabled
Reset
Administration via Local Phone
|---
Admin
|--- Network
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|---
3.5.7
Port Configuration
|--- SIP server
|--- SIP registrar
|--- SIP gateway
|--- SIP local
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 Registration 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 registrar 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 seconds by default. This is configurable (see Maximum Registration Backoff Timer
Æ page 120).
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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
192.168.1.165
SIP registar Address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
OS Voice
Realm
User ID
Password
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
3600
Backup transport
UDP
Backup OBP flag
Submit
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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
60
0
Reset
Administration via Local Phone
|
--- Admin
|
--- System
|
--- SIP Interface
|--- Outbound proxy
|
--- Default OBP domain
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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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
60
0
Reset
Administration via Local Phone
|---
Admin
|--- System
|--- SIP Interface
|--- SIP transport
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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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
60
0
Keep alive format
Sequence
Media Negotiation
Single IP
Media IP Mode
Submit
IPv4
Reset
Administration via Local Phone
|---
Admin
|--- System
|--- SIP Interface
|--- Media negotiation
|--- Media IP mode
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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
•
•
114
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
192.168.1.165
SIP registar Address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
OS Voice
Realm
User ID
Password
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
3600
Backup transport
UDP
Backup OBP flag
Submit
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 domain 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 connectivity 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 Connectivity 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 INVITE 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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
Submit
60
0
Reset
Administration via Local Phone
|---
Admin
|--- System
|--- SIP Interface
|--- NonCall transactions (ms)
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System Settings
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
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
Connectivity check timer (seconds)
60
0
Submit
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
192.168.1.165
SIP registar Address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
HiQ8000
Realm
User ID
Password
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
3600
Backup transport
UDP
Backup OBP flag
Reset
Submit
Network > Port configuration
Port configuration
SIP Server
5060
SIP registrar
5060
SIP gateway
5060
SIP local
5060
Backup proxy
5060
RTP base
5010
Download server (default)
LDAP server
HTTP proxy
0
LAN port speed
Automatic
PC port speed
PC port mode
PC port autoMDIX
Automatic
Submit
122
21
389
disabled
Reset
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Administration via Local Phone
|---
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
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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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Administration via WBM
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
Line overview
Video calls
;
;
;
;
;
CTI control
;
Phone book lookups
DSS feature
BLF feature
CTI
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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Feature Configuration
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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
alert-internal
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
alert-internal
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
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Feature Configuration
Administration via Local Phone
|---
Admin
|--- System
|--- Features
|--- Configuration
|--- General
|--- Hot / warm phone
|--- Hot / warm destination
|--- Initial digit timer
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Feature Configuration
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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
alert-internal
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
Administration via Local Phone
|
--- Admin
|
--- System
|
--- Features
|
--- Configuration
|
--- General
|
--- Initial digit timer
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Feature Configuration
3.7.4
Group Pickup
>
This feature is only available when allowed under System > Features > Feature access (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
3.7.4.2
Reset
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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Feature Configuration
The following table shows the group pickup alert behaviour for each possible scenario:
Phone State
Ringer on
Group pickup as Group pickup as
ringer=yes
ringer=no
Idle
In call
Ringer off
Ring tone
Speaker
Beep
Speaker
Handset
Ring tone
Speaker
Beep
Handset
Handset
Open listening
Beep
Handset and
Speaker
Beep
Handset and
Speaker
Headset
Ring tone
Speaker
Beep
Headset
Headset
Open listening
Beep
Headset and
Speaker
Beep
Headset and
Speaker
Hands-free
Beep
Speaker
Beep
Speaker
Nothing
Nothing
Handset
Nothing
Beep
Handset
Handset
Open listening
Beep
Handset and
Speaker
Beep
Handset and
Speaker
Headset
Nothing
Beep
Headset
Headset
Open listening
Beep
Headset and
Speaker
Beep
Headset and
Speaker
Hands-free
Beep
Speaker
Beep
Speaker
Idle
In call
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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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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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
Submit
6
;
;
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 via WBM
System > Features > Configuration
Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
Administration via Local Phone
|
--- Admin
|
--- System
|
--- Features
|
--- Configuration
|
--- General
|
--- Transfer on hangup
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Feature Configuration
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)
Max. callbacks
Submit
altert-internal
30
5
Reset
Administration via Local Phone
|---
Admin
|--- System
|--- Features
|--- Call completion
|--- Functional CCSS
|--- Callback ringer
|--- Allow after call (s)
|--- Max. callbacks
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Feature Configuration
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 Show
Alternative label
New urgent items Show
Alternative label
Old Items Show
Alternative label
Old urgent items Show
Alternative label
Submit
140
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Administration via Local Phone
|---
Admin
|--- Locatl functions
|--- Messages settings
|--- New items
|--- Alternative label
|--- New urgent items
|--- Alternative label
|--- Old items
|--- Alternative label
|--- Old urgent items
|--- Alternative label
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Feature Configuration
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
142
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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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
Administration via Local Phone
|---
Admin
|--- System
|--- Features
|--- Configuration
|--- General
|--- Server features
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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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
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Administration via Local Phone
|---
Admin
|--- System
|--- Features
|--- Configuration
|--- General
|--- 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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
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Administration via Local Phone
|---
Admin
|--- System
|--- Features
|--- Configuration
|--- 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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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 refer to Section 3.11, “Multiline Appearance/Keyset”.
Administration via WBM
System > Features > Configuration > General
Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
150
Disabled
Off
Reset
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Free Programmable Keys
Administration via Local Phone
|---
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
3.8.2
Reset
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
Selected dialling
Dial number
Submit
Reset
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Free Programmable Keys
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
Repeat Dialling
Reset
Submit
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
152
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Free Programmable Keys
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 External 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 switched 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 type
Forwarding
CF Unconditional any
Destination
Submit
Reset
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Free Programmable Keys
Administration via WBM (User menu)
User > Configuration > Incoming calls > Forwarding
Forwarding
Forwarding- Unconditional
Forward any call
to
2153
Destination
Forward external calls
to
2102
Destination
Forward internal calls
to
not set
Destination
Forwarding- Busy
Forward any call
to
2152
Destination
Forward external calls
to
2102
Destination
Forward internal calls
to
2102
Destination
Forwarding- No reply
Forward any call
to
2102
Destination
Forward external calls
to
2102
Destination
Forward internal calls
to
2102
Destination
Forwarding Favourites
Forwarding Favorites
Submit
154
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
Ringer off
Reset
Submit
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
Hold
Reset
Submit
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
Blind transfer
Reset
Submit
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
Join
Key.label 3
Submit
156
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
Deflect
Key.label 3
3335
Destination
Reset
Submit
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
3.8.13
Shift
Reset
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
Conference
Key.label 3
Reset
Submit
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
Headset
Submit
3.8.15
Reset
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
158
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
Group pickup
Key.label 3
Reset
Submit
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 Description field, an appropriate description for the feature can be entered.
Administration via WBM
System > Features > Program keys > Feature toggle
Feature toggle
Feature toggle
Key.label 3
Feature code
0
Description
Reset
Submit
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
160
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
Directed pickup
Reset
Submit
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
Cancel Callbac
Reset
Submit
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 Consultation
Submit
3.8.24
Reset
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 Call waiting
Submit
3.8.25
Reset
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 Call Recording
Submit
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 AICS ZIP
Submit
3.8.27
Reset
Server Feature
Invokes a feature on the SIP server. The status of the feature can be monitored via the LED
associated to the key.
>
3.8.28
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.
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
3.8.30
App:
Example
Reset
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
•
•
•
•
•
•
•
•
•
166
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
Send URL
Key label 2
Message details
Protocol
HTTPS
Web server address
Port
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 Call forward
Submit
3.8.32
Reset
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 2nd alert
Submit
3.8.33
Reset
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 Personal
Submit
Reset
System > Features > Program keys > Corporate directory
Corporate directory
Key.label 5 Corporate
Submit
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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 Show phone screen
Reset
Submit
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
Mute
Key.label 3 Mute
Submit
3.8.36
Reset
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 Cancel/Release
Submit
Reset
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Preset Function Keys (OpenStage 40 US only)
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 access (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
Line
Key
edit
Label: Primary Line
Selected dialling
Label: Selected dialling
edit
Hold
edit
Label: Hold
Shifted
1
Clear (no feature assigned)
edit
2
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
Mobility
Label: Mobility
edit
7
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
8
Clear (no feature assigned)
edit
Shift
Label: Shift
edit
9
174
Clear (no feature assigned)
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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
0
0
Address
Realm
User Identifier
Password
Shared type
Allow Overview
Hot warm action
Hot warm destination
shared
No Action
Reset
Submit
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
0
;
3337
Ring on/off
1
Selection order
Hot/warm line
Hot/warm destination
Submit
Hot line
3333
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.
>
178
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)
alert beep
;
idle line
ringing line
hold
;
60
Forwarding indicated
Preselect mode
Preselect timer
Preview mode
Preview timer
Bridging priority
Submit
180
single button
8
Preview overwrites bridReset
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System > Features > Configuration
Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
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|---
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 Preview
Submit
3.11.3.1
Reset
Preview and Preselection
Precondition: Bridging is enabled
Action
Preselect mode
Single but- Preseton
lection
Preview mode
On (Lock
Prev.)
Off
(Temp
Prev.)
Bridging priority
Preview
overrides
bridging
Result
Bridging
overrides
preview
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.
n.rel.
Line status is displayed
(Preselect timer)
Press busy second.
line key 2x while
line-view is displayed
-
9
n.rel.
n.rel.
n.rel.
n.rel.
1st press: line status
Press busy second.
line key 1x
9
2nd press: bridge (conference)
-
n.rel.
n.rel.
n.rel.
n.rel.
Bridge (conference)
9
Bridge (conference)
Preview key is pressed first (LED on) and the Line key is pressed once or twice
Press busy second.
line key 1x
n.rel.
Press busy second.
line key 1x
n.rel.
Press busy second.
line key 1x
n.rel.
n.rel.
-
9
-
Preview LED -> off
n.rel.
9
-
-
9
Bridge (conference)
Preview LED remains on
n.rel.
-
9
9
-
Line status is displayed
(Preview timer)
Preview LED -> off
Press busy second.
line key 1x
n.rel.
n.rel.
9
-
9
-
Line status is displayed
(Preview timer)
Preview LED remains on
Press busy second.
line key 2x while
line-view is displayed
n.rel.
n.rel.
-
9
9
-
1st press: line status
2nd press: bridge (conference)
Preview LED -> off
Press busy second.
line key 2x while
line-view is displayed
n.rel.
n.rel.
9
-
9
-
1st press: line status
2nd 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 Immediate ring
Submit
3.11.5
Reset
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)
3
Deflect alerting call enabled
Allow pickup to be refused
Forwarding shown
Submit
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
DSS
Address
Realm
User Identifier
Password
Outgoing Calls
Normal_
Action on Calls
Consult
Allow in overview
Submit
188
;
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
Pattern melody
Pattern
sequence
alert-internal Ringer1.wav
8
1
0
Ring
alert-externa Ringer2.mp4
4
2
60
Ring
alert-recall
Ringer3.mp3
3
2
60
Ring
alert-emerge Ringer5.mp3
3
2
60
Ring
Line-3336-R
Ringer2.mp3
8
2
60
Ring
Line-3335-R
Ringer4.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Name
Ringer sound
Submit
190
Duration (sec) Audible
Ring
Reset
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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
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
0
;
3336
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 Key Module 15 OpenStage Key Module
OpenStage 15
1
-
OpenStage 40
1
2
OpenStage 60/80
-
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
edit
1
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
2
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
Clear (no feature assigned)
Clear (no feature assigned)
edit
7
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
8
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
9
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
10
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
11
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
12
Clear (no feature assigned)
edit
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Key Modules
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
edit
1
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
2
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
7
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
8
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
9
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
10
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
11
Clear (no feature assigned)
edit
edit
12
Clear (no feature assigned)
edit
Clear (no feature assigned)
Clear (no feature assigned)
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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 national 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
•
•
•
•
•
•
•
•
•
194
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
Canonical dial settings
Local country code
49
National prefix digit
0
Local national code
89
Minimum local number length
4
Local enterprise node
723
PSTN access code
0
International access code
00
Operator codes
Emergency numbers
Initial extension digits
1,2,3,4
Reset
Submit
Local functions > Locality > Canonical dial
Canonical dial
Internal numbers
External numbers
External access code
Local enterprise form
Local public form
Not required
International gateway code
Use national code
Submit
Reset
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Administration via Local Phone
|---
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
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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 numbers and External numbers (-> Section 3.13.1), internal numbers must be discerned from external 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:
International code 1:
Local code 2:
International code 2:
Local code 3:
International code 3:
Local code 4:
International code 4:
Local code 5:
International code 5:
Submit
198
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Administration via Local Phone
|---
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
|--- International 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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Dialing
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
Submit
6
;
;
Reset
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System > Features > Configuration > Dial plan enabled
Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
202
Disabled
Off
Reset
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Administration via Local Phone
|---
User
|---
Admin
|--- System
|--- Features
|--- Configuration
|--- General
|--- Dial plan
|---
Admin
|--- General Information
|--- Dial plan ID
|--- Dial plan status
|---
Configuration
|--- Outgoing calls
|--- Immediate dialing
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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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Ringer Setting
>
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
Ringer setting
This page allows you to set up interworking with other IP phone
systems that support distinctive ringing
Name
Ringer sound
Pattern melody
Pattern
sequence
Bellcore-dr1
Pattern
Duration (sec) Audible
8
1
0
Ring
Impact-Level Ringer2.mp4
4
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Reset
Submit
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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Ringer Setting
Administration via WBM
Admin > Ringer setting > Map To Specials
Map To Specials
Internal
Bellcore-dr1
External
Bellcore-dr2
Recall
Bellcore-dr3
Emergency
Special1
Special2
Special3
Reset
Submit
3.14.3
alert-emerge
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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Ringer Setting
•
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
Pattern sequence
melody
Internal
Extermal
Ringer2.mp3
2
1.0 sec. ON, 2.0 sec. OFF
Ringer3.mp3
2
1.0 sec. ON, 2.0 sec. OFF
Recall
Emergency
Ringer4.mp3
2
1.0 sec. ON, 2.0 sec. OFF
Ringer5.mp3
2
1.0 sec. ON, 2.0 sec. OFF
Special 1
Ringer1.mp3
1
1.0 sec. ON, 4.0 sec. OFF
Special 2
Ringer1.mp3
1
1.0 sec. ON, 4.0 sec. OFF
Special 3
Ringer1.mp3
1
1.0 sec. ON, 4.0 sec. OFF
Submit
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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Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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
•
•
•
•
•
•
210
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
Administration via WBM
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
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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.
>
3.16.1
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
FTP/HTTPS Server
There are no specific requirements regarding the FTP server for transferring files to the OpenStage 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 HTTPS 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
•
•
212
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
FTP Server address
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
2.
a)
If compatible (or if Override is set): Proceed with update
b)
If NOT compatible: Abandon update and return to original application
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
3.16.3.1
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.
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 specified.
Data required (in any case)
•
•
•
214
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
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
|--- 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.
|---
2.
In the administration menu, set the focus to Phone app.
Admin
|--- File Transfer
|--- Phone app
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 specified.
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)
•
•
•
•
•
•
218
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.
220
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 specified.
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
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 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.
|---
2.
In the administration menu, set the focus to Picture clip.
Admin
|--- File Transfer
|--- Picture clip
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.
>
3.16.6.1
The OpenStage phones support LDAPv3.
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 specified.
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)
•
•
•
•
•
•
•
•
224
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 > LDAP
LDAP
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
|--- LDAP
|--- Use default
|--- Download method
|--- Server
|--- Port
|--- Account
|--- Username
|--- Password
|--- FTP path
|--- HTTPS base URL
|--- Filename
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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.
226
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 specified.
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 phone
FTP password
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.
|---
2.
In the administration menu, set the focus to Logo.
Admin
|--- File Transfer
|--- Logo
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 specified.
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)
•
•
230
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 phone
FTP password
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.
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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
64 kbit/s
80 kbit/s
96 kbit/s
112 kbit/s
0:15 min
0,12 MB
0,15 MB
0,18 MB
0,21 MB
0:30 min
0,23 MB
0,29 MB
0,35 MB
0,41 MB
0:45 min
0,35 MB
0,44 MB
0,53 MB
0,62 MB
1:00 min
0,47 MB
0,59 MB
0,70 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 specified.
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)
•
•
•
•
•
•
•
•
234
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 > 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
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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.
236
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 specified.
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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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.
|---
2.
In the administration menu, set the focus to Dongle key.
Admin
|--- File Transfer
|--- Dongle key
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
Corporate Phonebook: Directory Settings
3.17.1
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
•
•
•
240
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
Port configuration
SIP server
SIP registrar
5060
SIP gateway
5060
SIP local
5060
Backup proxy
RTP base
5010
Download server (default)
LDAP server
HTTP proxy
5060
5060
21
389
0
LAN port speed
Automatic
PC port speed
PC port mode
PC port autoMDIX
Automatic
Submit
disabled
Reset
Administration via Local Phone
|---
Admin
|--- Network
|--- Port Configuration
|--- RTP base
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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
G.711 ranking
Automatic
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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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
Mute Settings Microphone ON-Loudspeaker ON
;
DTMF playback
Submit
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
Applications
3.19.1
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 Applications > 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 OpenStage 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 Debug 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
•
248
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
http
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
0
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
Reset
Submit
Applications > XML Applications > Modify/Delete application
Modify/Delete application
Select application
testxml
Delete
Modify
Settings
Display name
testxml
Application name
testxml
HTTP Server address
HTTP Server port
Protocol
Program name on server
192.168.1.150
8080
http
testxml/servlet
;
Auto start
Use proxy
No
XML Trace enabled
No
Debug program on server
Number of tabs
0
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
250
0
Reset
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Administration via Local Phone
|---
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
|--- Restart after change
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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
http
Program name on server
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Reset
Submit
Applications > XML Applications > Modify/Delete application
Modify/Delete application
Select application
Weather
Delete
Modify
Settings
Display name
Weather
Application name
Weather
HTTP Server address
HTTP Server port
Protocol
Program name on server
87.106.21.36
8080
http
WR/WR
Use proxy
No
XML Trace enabled
No
Debug program on server
Submit
252
Reset
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Applications
Network > General IP configuration
General IP configuration
Protocol Mode IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
;
VLAN discovery DHCP
VLAN ID
DNS domain
Primary DNS 192.168.1.105
Secondary DNS 192.168.1.2
HTTP proxy
Reset
Submit
Network > Port configuration
Port configuration
SIP server
SIP registrar
5060
SIP gateway
5060
SIP local
5060
Backup proxy
RTP base
5010
Download server (default)
LDAP server
HTTP proxy
5060
5060
21
389
0
LAN port speed
Automatic
PC port speed
PC port modo
PC port autoMDIX
Automatic
Submit
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
Weather
Delete
Modify
Settings
Display name
Weather
Application name
Weather
HTTP Server address
HTTP Server port
Protocol
Program name on server
87.106.21.36
8080
http
WR/WR
Use proxy
No
XML Trace enabled
No
Debug program on server
Submit
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
Weather
Select application
Delete
Modify
Settings
Display name
Weather
Application name
Weather
87.106.21.36
HTTP Server address
HTTP Server port
8080
http
Protocol
Program name on server
WR/WR
Use proxy
No
XML Trace enabled
No
Debug program on server
Reset
Submit
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
258
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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Diagnostics
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:
262
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
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
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Diagnostics
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
3.26.4
;
Reset
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 OpenScape 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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Diagnostics
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 *333#
Submit
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: Diagnostic tracing is active!
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Diagnostics
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
•
•
•
•
•
•
•
266
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
Received
Sent
Sent: Mon Oct 27 10:41:14 2013
Received: Mon Oct 27 10:41:14 2013
Channel ID TLV Data
Channel ID TLV Data
.ID = 163.165.2.105
.ID = 00:3E:37:01:20:01
TTL TLV Data
TTL TLV Data
.records = 120
.records = 120
System Caps TLV Data
System Caps TLV Data
.Supported = Bridge, Telephone
.Supported = Other, Repeater
.Enabled = Telephone
.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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Diagnostics
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
Ping tests
Ping
Pre Defined Trace tests
Traceroute DLS
Traceroute
Traceroute
Traceroute
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Diagnostics
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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Administration via WBM
Diagnostics > Miscellaneous > Memory information
Memory information
Memory Monitor Configuration
Disable Reboot
High Threshold(MBs)
10
Low Threshold(MBs)
8
Working Hour Start
5
Working Hour End
24
Download memory info file
Download old memory info file
Submit
Reset
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
6098
1908 root
R
2063 1876 root
24HR 0 NO_APP_PROP
VSZ %MEM %CPU COMMAND
1420
1%
S N
34148
28%
40% /bin/busybox top -d 0 -a -n 1 -l 600 -b
10% PhoneletLauncher desktopphonelet.phd V3 R0.50.0
SIP
110924 WP4 Siemens SIP DE de DD.MM.YYYY
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 incorporated 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 (enabled/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
•
•
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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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•
•
•
•
•
•
•
•
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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.
>
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The trace level must be set to "TRACE" or "DEBUG".
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•
•
•
•
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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
Fault trace configuration
File size (bytes)
Trace timeout (minutes)
65536
0
Automatic clear before start
Trace levels for components
Administration
OFF
Application framework
OFF
Application menu
OFF
Bluetooth service
OFF
Call Log
OFF
Call View
OFF
Certificate management
OFF
Communications
OFF
Component registrar
OFF
CSTA service
OFF
Data Access service
OFF
Desktop
OFF
Digit analysis service
OFF
Directory service
OFF
DLS client management
OFF
Health service
OFF
OFF
Help
OFF
Instrumentation service
Java
OFF
Journal service
OFF
Media control service
OFF
Media processing service
OFF
Mobility service
OFF
OBEX service
OFF
OpenStage client management
OFF
Phonebook
OFF
Pot service
OFF
Password management service
OFF
Physical interface service
OFF
Service framework
OFF
Service registry
OFF
Sidecar service
OFF
SIP call control
OFF
SIP messages
OFF
SIP signalling
OFF
Team service
OFF
Tone geberation service
OFF
Transport service
OFF
vCard parser service
OFF
Voice engine service
OFF
Voice mail
OFF
Web server service
OFF
USB backup service
OFF
Video service engine
OFF
802.1x service
OFF
Clock Service
OFF
SIP messaging traces are enabled after reboot
Download trace file
Download saved trace file
Download upgrade trace file
Download old trace file
Download syslog file
Download old syslog file
Download saved syslog file
Download Database file
Download upgrade error file
Download HPT remote service log file
Download dial plan file
Submit
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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)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Data Access service
TRACE
Media control service
TRACE
OpenStage client management
DEBUG
Voice engine service
TRACE
Media processing service
TRACE
Bluetooth service
Download trace file
Submit
278
LOG
Physical interface service
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)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Data Access service
TRACE
Media control service
TRACE
OpenStage client management
LOG
Voice engine service
TRACE
Media processing service
TRACE
Bluetooth service
Download trace file
TRACE
Download saved trace file
Reset
Submit
3.26.10.3
Call Connection
Diagnostics > EasyTrace Profiles > Call connection
Call connection
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
LOG
Service registry
TRACE
SIP signalling
DEBUG
SIP call controll
DEBUG
Call View
TRACE
Communications
TRACE
CSTA service
TRACE
SIP messages
Download trace file
Submit
>
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)
1048576
0
Automatic clear before start
Trace levels for components
Call Log
TRACE
Component registrar
TRACE
Health service
LOG
Application framework
TRACE
Desktop
TRACE
Journal service
Download trace file
TRACE
Download saved trace file
Reset
Submit
3.26.10.5
Call Recording
Diagnostics > EasyTrace Profiles > Call recording
Call recording
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Call View
DEBUG
Communications
DEBUG
SIP call control
DEBUG
Media recording service
Download trace file
Submit
280
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)
1048576
0
Automatic clear before start
Trace levels for components
Certificate management
Component registrar
Health service
DLS client management
Service framework
Download trace file
LOG
TRACE
LOG
TRACE
TRACE
Download saved trace file
Reset
Submit
3.26.10.7
DLS Data Errors
Diagnostics > EasyTrace Profiles > DLS data errors
DLS data errors
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Certificate management
LOG
Component registrar
TRACE
Data Access service
TRACE
Health service
DLS client management
OpenStage client management
Service framework
Download trace file
Submit
LOG
TRACE
LOG
TRACE
Download saved trace file
Reset
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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)
1048576
0
Automatic clear before start
Trace levels for components
Application menu
Component registrar
Health service
Application framework
LOG
TRACE
LOG
TRACE
Help
DEBUG
Web server service
TRACE
Download trace file
Download saved trace file
Reset
Submit
3.26.10.9
Key Input
Diagnostics > EasyTrace Profiles > Key input problems
Key input problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
Physical interface service
Download trace file
Submit
282
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)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
Transport service
Download trace file
TRACE
LOG
TRACE
Download saved trace file
Reset
Submit
3.26.10.11 Messaging
Diagnostics > EasyTrace Profiles > Messaging application problems
Messaging application problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
TRACE
LOG
Application framework
TRACE
Call View
TRACE
Communications
TRACE
CSTA service
TRACE
Desktop
TRACE
SIP signalling
Download trace file
Submit
DEBUG
Download saved trace file
Reset
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3.26.10.12 Mobility
Diagnostics > EasyTrace Profiles > Mobility problems
Mobility problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Administration
TRACE
Data Access service
TRACE
DLS client management
Mobility service
Download trace file
LOG
TRACE
Download saved trace file
Reset
Submit
3.26.10.13 Phone administration
Diagnostics > EasyTrace Profiles > Phone administration problems
Phone administration problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Administration
TRACE
Health service
WARNING
OpenStage client management
TRACE
Communications
TRACE
CSTA service
TRACE
Desktop
Download trace file
Submit
284
LOG
Application framework
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)
1048576
0
Automatic clear before start
Trace levels for components
Application menu
TRACE
Component registrar
TRACE
Directory service
TRACE
Health service
LOG
Application framework
TRACE
Desktop
TRACE
Journal service
TRACE
Transport service
Download trace file
LOG
Download saved trace file
Reset
Submit
3.26.10.15 Local Phonebook
Diagnostics > EasyTrace Profiles > Phonebook (local) problems
Phonebook (local) problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Application menu
Component registrar
Health service
Application framework
LOG
TRACE
LOG
TRACE
Desktop
TRACE
Journal service
TRACE
Download trace file
Submit
Download saved trace file
Reset
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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)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
AGP Phonelet
Download trace file
LOG
Download saved trace file
Reset
Submit
3.26.10.17 Sidecar
Diagnostics > EasyTrace Profiles > Sidecar problems
Sidecar problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
Sidecar service
Download trace file
Submit
286
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)
1048576
0
Automatic clear before start
Trace levels for components
Call View
TRACE
WARNING
Communications
CSTA service
LOG
Team Service
TRACE
SIP signalling
TRACE
SIP call control
TRACE
SIP messages
Download trace file
TRACE
Download saved trace file
Reset
Submit
3.26.10.19 SIP standard singleline
Diagnostics > EasyTrace Profiles > SIP standard singleline
SIP standard singleline
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Call View
Communications
TRACE
LOG
CSTA service
TRACE
SIP signalling
TRACE
SIP call control
DEBUG
SIP messages
DEBUG
Download trace file
Submit
Download saved trace file
Reset
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3.26.10.20 Speech
Diagnostics > EasyTrace Profiles > Speech problems
Speech problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
TRACE
LOG
Voice engine service
TRACE
Media processing service
TRACE
SIP signalling
DEBUG
SIP call control
DEBUG
Download trace file
Download saved trace file
Reset
Submit
3.26.10.21 Tone
Diagnostics > EasyTrace Profiles > Tone problems
Tone problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Component registrar
Health service
Tone generation service
Media processing service
Download trace file
Submit
288
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)
1048576
0
Automatic clear before start
Trace levels for components
Administration
TRACE
Component registrar
TRACE
Physical interface service
DEBUG
USB backup service
Download trace file
DEBUG
Download saved trace file
Reset
Submit
3.26.10.23 Voice Dialling
Diagnostics > EasyTrace Profiles > Voice recognition problems
Voice recognition problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Media control service
TRACE
Voice engine service
TRACE
Call View
TRACE
Media processing service
TRACE
Voice recognition
TRACE
Phonebook
TRACE
Download trace file
Submit
Download saved trace file
Reset
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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)
1048576
0
Automatic clear before start
Trace levels for components
Data Access service
TRACE
OpenStage client management
TRACE
Web server service
TRACE
USB backup service
TRACE
802.1x service
TRACE
Voice recognition
TRACE
Download trace file
Download saved trace file
Reset
Submit
3.26.10.25 802.1x problems
Diagnostics > EasyTrace Profiles > 802.1x problems
802.1x problems
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Certificate management
TRACE
Data Access service
TRACE
Download trace file
Submit
290
LOG
Component registrar
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)
1048576
0
Automatic clear before start
Trace levels for components
Administration
OFF
Call Log
OFF
Call View
OFF
Phonebook
Help
OFF
Application menu
OFF
Certificate management
OFF
Communications
OFF
Component registrar
OFF
CSTA service
OFF
Data Access service
OFF
Digit analysis service
OFF
Digital data service
OFF
OFF
Directory service
OFF
DLS client management
OFF
Health service
OFF
Instrumentation service
OFF
Journal service
OFF
Media control service
OFF
Media processing service
OFF
Mobility service
OFF
OBEX service
OFF
OpenStage client management
OFF
Performance Marks
OFF
Password management service
OFF
Physical interface service
OFF
Sidecar service
Team service
OFF
Tone generation service
OFF
OFF
Transport service
OFF
Voice engine service
OFF
Web server service
OFF
SIP signalling
SIP call control
OFF
SIP messages
OFF
OFF
Application framework
OFF
Desktop
OFF
AGP Phonelet
OFF
Service framework
Service registry
OFF
OFF
Bluetooth service
OFF
vCard parser service
OFF
Voice mail
OFF
USB backup service
OFF
802.1x service
OFF
Voice recognition
OFF
Download trace file
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
Tracing is stopped
256000
;
;
Start
Download trace file
Submit
292
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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 SubSubmit.
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)
EOS Threshold exceeded
60
10
Observation interval (seconds)
Minimum session length (100 millisecond units)
20
Codec independent threshold values
Maximum jitter (milliseconds)
Average round trip delay (milliseconds)
20
100
Non-compressing codec threshold values
Lost packets (per 1000 packets)
10
Consecutive lost packets
2
Consecutive good packets
8
Compressing codec threshold values
Lost packets (per 1000 packets)
10
Consecutive lost packets
2
Consecutive good packets
8
Resend last report
Submit
294
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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
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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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•
•
•
•
•
•
•
•
•
298
•
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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Administration
Diagnostics
Data viewing via WBM
Diagnostics > QoS reports > View Session Data
View Session Data
Select a report to view
QoS Statistics 1
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
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
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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
300
Reset
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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
Disabled
Use Notification
Enabled
Remote Server
Remote Server Port
Submit
514
Reset
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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 Bluetooth interface.
Administration via WBM
System > Features > Configuration
Configuration
General
Emergency number
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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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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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
Disabled
Off
Reset
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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
3335
Voice Mail number
MWI LED
Missed call LED
Key & AlertBar
Key only
;
Allow refuse
Hot/warm phone
No action
Hot/warm destination
Initial digit timer (seconds)
30
;
Allow uaCSTA
Server features
Not used timeout (minutes)
Transfer on hangup
Bridging enabled
Dial plan enabled
FPK program timer
5
;
;
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
Impact level ringer
alert-internal
None
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Audible Notification
Submit
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
Technical Reference
4.1
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
Menu Structure
Admin Login
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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Technical Reference
Menus
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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Menus
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:
Reset
Login
Add application
Add application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
http
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
0
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
Reset
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Menus
Modify/Delete application
Modify/Delete application
Select application
testxml
Delete
Modify
Settings
Display name
testxml
Application name
testxml
HTTP Server address
HTTP Server port
192.168.1.150
8080
http
Protocol
Program name on server
testxml/servlet
Auto start
;
Use proxy
No
XML Trace enabled
No
Debug program on server
Number of tabs
0
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
0
Reset
Submit
Xpressions
Xpessions
Display name
Application name
Xpressions
Xpressions
HTTP Server address
HTTP Server port
Protocol
https
Program name on server
Auto start
Use proxy
No
XML Trace enabled
No
Debug program on server
Number of tabs
All tabs Start
3
Tab 1 Display Name
Voice Mail
Tab 1 Application Name
Tab 2 Display Name
Tab 2 Application Name
Tab 3 Display Name
Tab 3 Application Name
Restart after change
Submit
316
Xpressions
Inbox
XprInbox
Outbox
Xproutbox
Delete
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Menus
Add messages application
Add messages application
Display name
Application name
HTTP Server address
HTTP Server port
http
Protocol
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
All tabs Start
0
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
Reset
Submit
XML Phonebook (up to V2R0)
XML Phonebook
Display name
XMLPhonebook
Application name
XMLPhonebook
HTTP Server address
HTTP Server port
Protocol
http
Program name on server
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
0
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
Reset
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Menus
Add phonebook application
Add phonebook application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
http
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
All tabs Start
0
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
Reset
Submit
Add call log application
Add call log application
Display name
Application name
HTTP Server address
HTTP Server port
Protocol
http
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
All tabs Start
0
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
318
Reset
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Menus
Add help application
Add help application
Display name
Application name
HTTP Server address
HTTP Server port
http
Protocol
Program name on server
Auto start
Use proxy
Yes
XML Trace enabled
Yes
Debug program on server
Number of tabs
All tabs Start
0
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
Reset
Submit
Bluetooth
Bluetooth
;
Enable Bluetooth interface
Reset
Submit
General IP configuration
General IP configuration
Protocol Mode
IPv4_IPv6
LLDP-MED Enabled
DHCP Enabled
DHCPv6 Enabled
VLAN discovery
;
Manual
VLAN ID
DNS domain
Primary DNS
192.168.1.105
Secondary DNS
192.168.1.2
HTTP proxy
Submit
Reset
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Menus
IPv4 configuration
IPv4 configuration
LLDP-MED Enabled
;
DHCP Enabled
DHCP lease reuse
IP address
192.168.1.235
Subnet mask
255.255.255.0
Default route
192.168.1.2
Route 1 IP address
Route 1 gateway
Route 1 mask
Route 2 IP address
Route 2 gateway
Route 2 mask
Reset
Submit
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
Reset
Submit
Update Service (DLS)
Update Service (DLS)
DLS address
DLS port
Contact gap
192.168.1.242
18443
300
Revert to default security
Mode
Security PIN
Submit
320
Default
Reset
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Menus
QoS
QoS
Service
Layer 2
Layer 2 voice
5
Layer 2 signalling
3
Layer 2 default
0
;
Layer 3
Layer 3 voice
EF
Layer 3 signalling
AF31
Layer 3 video
AF41
MLPP
EF
Priority
EF
Immediate
Flash
EF
Flash override
EF
Submit
Reset
Port configuration
Port configuration
5060
SIP Server
SIP registrar
5060
SIP gateway
5060
5060
SIP local
5060
Backup proxy
5010
RTP base
21
Download server (default)
LDAP server
389
HTTP proxy
0
LAN port speed
Automatic
PC port speed
PC port mode
PC port autoMDIX
Automatic
disabled
Reset
Submit
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
4711
openstage
4711
;
Enable ID
Web name
DNS name construction
Only number
Reset
Submit
SIP interface
SIP interface
;
Outbound proxy
Default OBP domain
SIP transport
UDP
Response timer (ms)
32000
NonCall trans. (ms)
32000
Reg. backoff (seconds)
60
Connectivity check timer (seconds)
Sequence
Media Negotiation
Single IP
Media IP Mode
Submit
322
0
Keep alive format
IPv4
Reset
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Menus
Registration
Registration
SIP Addresses
SIP server address
192.168.1.165
SIP registar address
192.168.1.165
SIP gateway address
SIP Session
Session timer enabled
Session duration (seconds)
3600
Registration timer (seconds)
3600
Server type
HiQ8000
Realm
User ID
Password
MLPP base
Local
MLPP Domain
dsn+uc
Other Domain
SIP Survivability
;
Backup registration allowed
Backup proxy address
Backup registration timer (seconds)
Backup transport
3600
UDP
Backup OBP flag
Submit
Reset
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Menus
SNMP
SNMP
Generic traps
Traping sending enabled
Trap destination
Trap destination port
162
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
12010
QCU community
*****
QoS togeneric destination
Submit
324
Reset
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Menus
Configuration
Configuration
General
3335
Emergency number
Voice Mail number
MWI LED
Key & AlertBar
Missed call LED
Key only
;
Allow refuse
No action
Hot/warm phone
Hot/warm destination
30
Initial digit timer (seconds)
;
Allow uaCSTA
Server features
5
Not used timeout (minutes)
Transfer on hangup
;
;
Bridging enabled
Dial plan enabled
FPK program timer
On
Audio
;
;
Group pickup tone allowed
Group pickup as ringer
Group pickup visual alert
BLF alerting
Prompt
Beep
MLPP ringer
Callback ringer
alert-internal
Impact level ringer
Impact-Level
Bluetooth
;
Enable Bluetooth interface
Call Recording
Recorder Address
Recording Mode
Disabled
Audible Notification
Off
Reset
Submit
DSS settings
DSS settings
Call pickup detect timer (seconds)
3
Deflect alerting call enabled
Allow pickup to be refused
Forwarding shown
Submit
Reset
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Menus
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
Line
edit
Label: Primary Line
Selected dialling
Label: Selected dialling
Hold
edit
edit
Label: Hold
Shifted
1
Clear (no feature assigned)
edit
2
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
7
Clear (no feature assigned)
edit
edit
8
Clear (no feature assigned)
edit
edit
9
Clear (no feature assigned)
Mobility
Label: Mobility
edit
edit
Clear (no feature assigned)
Shift
Label: Shift
Clear (no feature assigned)
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
0
0
Address
Realm
User Identifier
Password
Shared type
Allow Overview
Hot warm action
Hot warm destination
Submit
326
shared
No Action
Reset
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Menus
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
edit
1
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
2
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
7
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
8
Clear (no feature assigned)
edit
Clear (no feature assigned)
Clear (no feature assigned)
edit
9
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
10
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
11
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
12
Clear (no feature assigned)
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
edit
1
Clear (no feature assigned)
edit
edit
2
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
3
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
4
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
5
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
6
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
7
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
8
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
9
Clear (no feature assigned)
edit
Clear (no feature assigned)
Clear (no feature assigned)
Clear (no feature assigned)
edit
10
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
11
Clear (no feature assigned)
edit
Clear (no feature assigned)
edit
12
Clear (no feature assigned)
edit
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Menus
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
Built-in release
Edit
Forwarding key
Built-in forwarding
Edit
Send URL
Edit
Voice recognition key
Keyset operation
Keyset operation
alert beep
;
Rollover ring
LED on registration
idle line
Originating line preference
ringing line
Terminating line preference
hold
Line action mode
Show focus
;
Reservation timer (seconds)
60
Forwarding indicated
single button
Preselect mode
Preselect timer
Preview mode
8
Preview timer
Preview overwrites brid-
Bridging priority
Reset
Submit
Addressing
Addressing
MW server URI
Conference
192.168.1.2
Group pickup URI
Callback: FAC
Callback cancel all
BLF pickup code
Submit
328
*0
Reset
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Technical Reference
Menus
Call completion
Call completion
Functionnal CCSS
Callback ringer
Allow after call (s)
30
Max. callbacks
5
Submit
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
Line overview
Video calls
;
;
;
;
;
CTI control
;
Phone book lookups
DSS feature
BLF feature
CTI
Services
Bluetooth
Web based manag.
USB device access
Backup to USB
Feature toggle
Phone lock
Submit
;
;
;
;
;
;
Reset
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Menus
System
System
SIP server certificate validation
Use secure calls
SRTP type
MIKEY
Use SRTCP
Reset
Submit
SDES Config
SDES config
SDES status
SDP negotiation
Disabled
SRTP + RTP
SHA1-80 ranking
X
SHA1-32 ranking
X
Reset
Submit
Access control
Access control
CCE access
Factory reset claw
Serial port
Enable
;
No Password
Reset
Submit
Logging
Logging
Max. lines
Archive to DLS
Archive when at
Last archived
500
50%
20101105-0010
Reset
Submit
Faults
Faults
Security log entry
OCSR Failure
20111009-2206
Admin access
User access
Cancel faults
Submit
330
All
Reset
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Menus
Defaults
Defaults
FTP
Download method
FTP Server address
FTP Server port
21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Reset
Submit
Phone application
Phone application
Use defaults
FTP
Download method
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
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
Submit
do nothing
Reset
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Menus
Picture Clip
Picture Clip
Use defaults
FTP
Download method
FTP Server address
FTP Server port
21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
do nothing
After submit
Reset
Submit
LDAP
LDAP
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
Submit
332
do nothing
Reset
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Menus
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
Reset
Submit
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
Submit
do nothing
Reset
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Menus
Ringer file
Ringer file
Use defaults
FTP
Download method
FTP Server address
FTP Server port
21
FTP account
FTP username
FTP password
FTP path
HTTPS base URL
Filename
After submit
do nothing
Reset
Submit
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
Submit
do nothing
Reset
Directory settings (OpenStage 40/20/15 )
Directory settings
LDAP Server address
LDAP Server port
Authentication
389
Anonymous
User name
Password
Submit
334
Reset
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Menus
Directory settings
Directory settings
LDAP Server address
Transport
TCP
LDAP Secure port
636
389
LDAP Server port
Authentication
Anonymous
User Name
Password
Search trigger timeout
3
Reset
Submit
Messages settings
Messages settings
New items
Show
Alternative label
New urgent items
Show
Alternative label
Old Items
Show
Alternative label
Old urgent items
Show
Alternative label
Submit
Reset
Canonical dial settings
Canonical dial settings
Local country code
49
National prefix digit
0
Local national code
89
Minimum local number length
4
Local enterprise node
723
PSTN access code
0
International access code
00
Operator codes
Emergency numbers
Initial extension digits
Submit
1,2,3,4
Reset
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Menus
Canonical dial lookup
Canonical dial lockup
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:
International code 5:
Reset
Submit
Canonical dial
Canonical dial
Internal numbers
External numbers
External access code
International gateway code
Local enterprise form
Local public form
Not required
Use national code
Reset
Submit
Phone location
Phone location
Phone location
Signalled
Submit
Reset
Energy saving
Energy saving
Backlight time
2 hours
Submit
Reset
Call logging
Call logging
FAC prefixes
Submit
336
Reset
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Technical Reference
Menus
Date and time
Date and time
Time source
SNTP IP address
192.43.244.18
Timezone offset (hours)
1
Daylight saving
Daylight saving
Difference (minutes)
;
Auto time change
DST zone
60
Europe (Rest)
Reset
Submit
Codec preferences
Codec preferences
Silence suppression
;
Allow "HD" icon
Packet size
Automatic
G.711 ranking
X
G.729 ranking
X
G.722 ranking
Reset
Submit
Audio settings
Audio settings
Mute Settings
Microphone ON-Loudspeaker ON
;
DTMP playback
Reset
Submit
General information
General information
MAC address:
0001e323f9a1
Software version: 0.7.5.0004-061027
Last restart:
Backlight type
1
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Menus
Generic policy
Generic policy
99
Expires after (days)
Warn before (days)
1
Force changed
Tries allowed
5
No change for (hours)
0
Suspended for (mins)
5
History valid for (days)
0
Reset
Submit
Admin policy
Admin policy
Expiry date
2038-01-19T03:14:07+00:00
6
Minimum length
0
Password history
Active
Current status
Reset
Submit
User policy
User policy
Expiry date
2038-01-19T03:14:07+00:00
6
Minimum length
0
Password history
Active
Current status
Reset
Submit
Character set
Character set
0
Ucase chars reqd.
Lcase chars reqd.
0
Digits required
0
Special chars reqd
Bar repeat length
0
0
Min char difference
0
Submit
Reset
338
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Menus
Change Admin password
Change Admin password
Current password
New password
Confirm password
Reset
Submit
Change User password
Change User password
Admin password
New password
Confirm password
Reset
Submit
Generic
Generic
OCSP check
OCSR 1 address
OCSR 2 address
Reset
Submit
Authentication policy
Authentication policy
Secure file transfer
None
Secure send URL
None
Secure SIP server
None
Secure 802.1x
None
XML Applications
None
Submit
Reset
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Menus
Distinctive
Distinctive
This page allows you to set up interworking with other IP phone
systems that support distinctive ringing
Name
Ringer sound
Pattern melody
Pattern
sequence
Duration (sec) Audible
Bellcore-dr1
Pattern
8
1
0
Ring
Bellcore-dr2
Ringer2.mp3
3
2
60
Ring
Bellcore-dr3
Ringer2.mp3
3
2
60
Ring
alert-emerge
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Ringer2.mp3
3
2
60
Ring
Reset
Submit
Map To Specials
Map To Specials
Internal
Bellcore-dr1
External
Bellcore-dr2
Recall
Emergency
Special1
Special2
Special3
Bellcore-dr3
alert-emerge
Reset
Submit
Mobility
Mobility
;
Unauthorised Logoff Trap
Logoff Trap Delay
300
Timer Medium Priority
60
Mobility Feature
Managed Profile
Error Count Local
Submit
340
0
Reset
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Technical Reference
Menus
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:
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
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
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User acess
User access
User Access
Submit
;
Reset
LLDP-MED TLVs
LLDP-MED TLV’s
Received
Sent
Sent: Mon Oct 27 10:41:14 2013
Received: Mon Oct 27 10:41:14 2013
Channel ID TLV Data
Channel ID TLV Data
.ID = 163.165.2.105
.ID = 00:3E:37:01:20:01
TTL TLV Data
TTL TLV Data
.records = 120
.records = 120
System Caps TLV Data
System Caps TLV Data
.Supported = Bridge, Telephone
.Supported = Other, Repeater
.Enabled = Telephone
.Enabled = Other, Repeater
342
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Menus
Fault trace configuration
Fault trace configuration
File size (Max 6290000 bytes)
Trace timeout (minutes)
1048576
0
Automatic clear before start
Trace levels for components
Administration
OFF
AGP Phonelet
OFF
Application framework
OFF
Application menu
OFF
Bluetooth service
OFF
Call Log
OFF
Call View
OFF
Certificate management
OFF
Clock Service
OFF
Communications
DEBUG
Component registrar
OFF
CSTA service
OFF
Data Access service
OFF
Desktop
OFF
Digit analysis service
OFF
Directory service
OFF
DLS client management
OFF
Health service
OFF
Help
OFF
HTTP Service
OFF
Instrumentation service
OFF
Journal service
OFF
Media control service
OFF
Media processing service
OFF
Media recording service
OFF
Mobility service
OFF
OBEX service
OFF
OpenStage client management
OFF
Password management service
OFF
Phonebook
OFF
Performance Marks
OFF
Physical interface service
OFF
Security Log Service
OFF
Service framework
DEBUG
Service registry
DEBUG
Sidecar service
DEBUG
SIP call control
OFF
SIP messages
OFF
SIP signalling
OFF
Team service
OFF
Tone generation service
OFF
Transport service
OFF
USB backup service
OFF
vCard parser service
OFF
Voice engine service
OFF
Voice mail
OFF
Web server service
OFF
802.1x service
OFF
SIP messaging traces are enabled after reboot
Download trace file
Download old trace file
Download saved trace file
Download syslog file
Download old syslog file
Download saved syslog file
Download exception file
Download old exception file
Download dial plan file
Download Database file
Download upgrade trace file
Download HPT remote service
log file
Submit
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Download upgrade error file
Download security log file
Reset
343
referenz.fm
Technical Reference
Menus
Bluetooth handsfree profile
Bluetooth handsfree profile
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Data Access service
TRACE
Media control service
TRACE
OpenStage client management
LOG
Physical interface service
DEBUG
Voice engine service
TRACE
Media processing service
TRACE
Bluetooth service
Download trace file
TRACE
Download saved trace file
Reset
Submit
Bluetooth headset profile
Bluetooth headset profile
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Data Access service
TRACE
Media control service
TRACE
OpenStage client management
LOG
Voice engine service
TRACE
Media processing service
TRACE
Bluetooth service
Download trace file
Submit
344
TRACE
Download saved trace file
Reset
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Technical Reference
Menus
Call connection
Call connection
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Service registry
TRACE
SIP signalling
DEBUG
SIP call controll
DEBUG
Call View
TRACE
Communications
TRACE
CSTA service
TRACE
SIP messages
DEBUG
Download trace file
Submit
Download saved trace file
Reset
Call log problems
Call log problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Call Log
TRACE
Component registrar
TRACE
Health service
LOG
Application framework
TRACE
Desktop
TRACE
Journal service
Download trace file
Submit
TRACE
Download saved trace file
Reset
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Menus
Call Recording
Call recording
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Call View
DEBUG
Communications
DEBUG
SIP call control
DEBUG
Media recording service
Download trace file
Submit
DEBUG
Download saved trace file
Reset
DAS connection
DAS connection
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Certificate management
LOG
Component registrar
TRACE
Health service
LOG
DLS client management
TRACE
Service framework
Download trace file
Submit
TRACE
Download saved trace file
Reset
DLS data errors
DLS data errors
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Certificate management
LOG
Component registrar
TRACE
Data Access service
TRACE
Health service
LOG
DLS client management
TRACE
OpenStage client management
Service framework
Download trace file
Submit
346
LOG
TRACE
Download saved trace file
Reset
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Technical Reference
Menus
Help application problems
Help application problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Application menu
LOG
Component registrar
TRACE
Health service
LOG
Application framework
TRACE
Help
DEBUG
Web server service
Download trace file
TRACE
Download saved trace file
Reset
Submit
Key input problems
Key input problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Physical interface service
DEBUG
Download trace file
Download saved trace file
Reset
Submit
LAN connectivity problems
LAN connectivity problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Transport service
TRACE
HTTP Service
TRACE
Download trace file
Submit
Download saved trace file
Reset
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Technical Reference
Menus
Messaging application problems
Messaging application problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Application framework
TRACE
Call View
TRACE
Communications
TRACE
CSTA service
TRACE
Desktop
TRACE
SIP signalling
DEBUG
Download trace file
Download saved trace file
Reset
Submit
Mobility problems
Mobility problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Administration
TRACE
Data Access service
TRACE
DLS client management
LOG
Mobility service
TRACE
Download trace file
Submit
348
Download saved trace file
Reset
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Technical Reference
Menus
Phone administration problems
Phone administration problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Administration
Health service
OpenStage client management
TRACE
WARNING
LOG
Application framework
TRACE
Communications
TRACE
CSTA service
TRACE
Desktop
Download trace file
TRACE
Download saved trace file
Reset
Submit
Phonebook (LDAP) problems
Phonebook (LDAP) problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Application menu
TRACE
Component registrar
TRACE
Directory service
TRACE
Health service
LOG
Application framework
TRACE
Desktop
TRACE
Journal service
TRACE
Transport service
Download trace file
Submit
LOG
Download saved trace file
Reset
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Technical Reference
Menus
Phonebook (local) problems
Phonebook (local) problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Application menu
LOG
Component registrar
TRACE
Health service
LOG
Application framework
TRACE
Desktop
TRACE
Journal service
TRACE
Download trace file
Download saved trace file
Reset
Submit
Server based application problems
Server based application problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
AGP Phonelet
Download trace file
LOG
Download saved trace file
Reset
Submit
Sidecar problems
Sidecar problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
LOG
Sidecar service
TRACE
Download trace file
Submit
350
TRACE
Health service
Download saved trace file
Reset
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Technical Reference
Menus
SIP standard multiline
SIP standard multiline
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Call View
DEBUG
Communications
DEBUG
CSTA service
DEBUG
Team service
DEBUG
SIP signalling
DEBUG
SIP call control
DEBUG
SIP messages
Download trace file
DEBUG
Download saved trace file
Reset
Submit
SIP standard singleline
SIP standard singleline
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Call View
DEBUG
Communications
DEBUG
CSTA service
DEBUG
SIP signalling
DEBUG
SIP call control
DEBUG
SIP messages
Download trace file
Submit
DEBUG
Download saved trace file
Reset
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Technical Reference
Menus
Speech problems
Speech problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Voice engine service
TRACE
Media processing service
TRACE
SIP signalling
DEBUG
SIP call control
Download trace file
DEBUG
Download saved trace file
Reset
Submit
Tone problems
Tone problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Component registrar
TRACE
Health service
LOG
Tone generation service
TRACE
Media processing service
TRACE
Download trace file
Download saved trace file
Reset
Submit
USB backup/restore
USB backup/restore
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Administration
TRACE
Physical interface service
DEBUG
USB backup service
Download trace file
Submit
352
TRACE
Component registrar
DEBUG
Download saved trace file
Reset
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Technical Reference
Menus
Voice recognition problems
Voice recognition problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Media control service
TRACE
Voice engine service
TRACE
Call View
TRACE
Media processing service
TRACE
Voice recognition
TRACE
Phonebook
TRACE
Download trace file
Download saved trace file
Reset
Submit
Web based management
Web based management
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
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
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Technical Reference
Menus
802.1x problems
802.1x problems
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Certificate management
LOG
Clock service
TRACE
Component registrar
CSTA service
TRACE
Data Access service
DLS client management
Mobílity service
SIP call control
SIP messages
SIP signalling
TRACE
802.1x service
TRACE
Download trace file
Submit
354
TRACE
TRACE
TRACE
TRACE
TRACE
Download saved trace file
Reset
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Technical Reference
Menus
Clear all profiles
Clear all profiles
File size (Max 6290000 bytes)
1048576
Trace timeout (minutes)
0
Automatic clear before start
Trace levels for components
Administration
OFF
Call Log
OFF
Call View
OFF
Phonebook
OFF
Help
OFF
Application menu
OFF
Certificate management
OFF
Communications
OFF
Component registrar
OFF
CSTA service
OFF
Data Access service
OFF
Digit analysis service
OFF
Digital data service
OFF
Directory service
OFF
DLS client management
OFF
Health service
OFF
Instrumentation service
OFF
Jounal service
OFF
Media control service
OFF
Media processing service
OFF
Mobility service
OFF
OBEX service
OpenStage client management
Performance Marks
OFF
OFF
OFF
Password management service
OFF
Physical interface service
OFF
Sidecar service
OFF
Team service
OFF
Tone generation service
OFF
Transport service
OFF
Voice engine service
OFF
Web server service
OFF
SIP signalling
OFF
SIP call control
OFF
SIP messages
OFF
Application framework
OFF
Desktop
OFF
AGP Phonelet
OFF
Service framework
Service registry
OFF
Bluetooth service
OFF
OFF
vCard parser service
OFF
Voice mail
OFF
USB backup service
OFF
802.1x service
Voice recognition
OFF
OFF
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Menus
Bluetooth Advanced Traces
Bluetooth Advanced Traces
Automatic clear before start
;
File size (Max 6290000 bytes)
256000
;
;
Extended dump
Verbose decoding
Start
Tracing is stopped
Download trace file
Reset
Submit
Generation
Generation
Report mode
Report interval (seconds)
EOS Threshold exceeded
60
10
Observation interval (seconds)
Minimum session length (100 millisecond units)
20
Codec independent threshold values
Maximum jitter (milliseconds)
20
Average round trip delay (milliseconds)
100
Non-compressing codec threshold values
Lost packets (per 1000 packets)
10
Consecutive lost packets
2
Consecutive good packets
8
Compressing codec threshold values
Lost packets (per 1000 packets)
10
Consecutive lost packets
2
Consecutive good packets
8
Resend last report
Submit
356
Reset
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Technical Reference
Menus
View Session Data
View Session Data
Codec change on the fly
QoS Statistics 1
Submit
Start of report period - seconds
2011/10/16 21:51:29 UTC
End of report period - seconds
2011/10/16 21:56:36 UTC
SNMP specific trap type
2
IP address (local)
192.168.1.235
Port number (local)
5012
IP address (remote)
192.168.1.202
Port number (remote)
5010
SSRC (receiving)
1481715715
SSRC (sending)
3244864262
Codec
G.711 PCMU
Maximum packet size
20
Silence suppression
0
Count of good packets
15203
Maximum jitter
2
Maximum inter-arrival jitter
0
Periods jitter threshold exceeded
0
Round trip delay
433
Round trip delay threshold exceeded
Count of lost packets
0
Count of discarded packets
0
Periods of lost packets
0
Consecutive packet loss (CPL)
255,255,255,255,255,255,255,255,255,255,255
Periods of consecutive lost packets
255
Consecutive good packets (CGP)
255,255,255,255,255,255,255,255,255,255,255
Periods of consecutive good packets
255
Count of jitter buffer overruns
0
Count of jitter buffer under-runs
0
IP tests
IP tests
Pre Defined Ping tests
Ping DLS
Ping
Ping tests
Ping
Pre Defined Trace tests
Traceroute DLS
Traceroute
Traceroute
Traceroute
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Menus
Memory information
Memory information
Memory Monitor Configuration
Disable Reboot
High Threshold(MBs)
10
Low Threshold(MBs)
8
Working Hour Start
5
Working Hour End
24
Download memory info file
Download old memory info file
Submit
Reset
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
6098
1908 root
R
2063 1876 root
24HR 0 NO_APP_PROP
VSZ %MEM %CPU COMMAND
1420
1%
S N
34148
28%
40% /bin/busybox top -d 0 -a -n 1 -l 600 -b
10% PhoneletLauncher desktopphonelet.phd V3 R0.50.0
SIP
110924 WP4 Siemens SIP DE de DD.MM.YYYY
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
Reset
Submit
Remote trace
Remote trace
Remote Trace Status
;
User Notification
Remote Server IP
Remote Server Port
Submit
358
Reset
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Technical Reference
Menus
Restart Phone
Restart Phone
Confirm Restart
Factory reset
Factory reset
Factory reset password
Reset
Submit
HPT interface
HPT interface
Disable HPT
Secure Shell
Secure Shell
Enable access
Session password
Access minutes
1
Session minutes
1
Submit
Reset
Diagnostic call
Diagnostic call
Prefix Code
Submit
*333#
Reset
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Menus
4.1.2
Local Phone Menu
Menu
|--- Administration
| |--- Applications (OpenStage 60/80 only)
| | |--- XML applications
| |
|--- CAdd application: New phonebook application
| |
|--- AAdd application: New call log application
| |
|--- FAdd application: New messages application
| |
|--- FAdd Xpressions: Xpressions application
|--| |
? Add application: New help application
| |
| |--- 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
-> 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
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
referenz.fm
Technical Reference
Menus
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|--- 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
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
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
referenz.fm
Technical Reference
Menus
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| |
|--- 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
362
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
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
referenz.fm
Technical Reference
Menus
Menu
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Further information ...
| |--- Trap sending enabled
-> Section 3.3.9
| |--- Trap destination
-> Section 3.3.9
| |--- Trap destination port
-> Section 3.3.9
| |--- Trap community
-> Section 3.3.9
| |--- Diagnostic sending enabled
-> Section 3.3.9
| |--- Diagnostic destination
-> Section 3.3.9
| |--- Diagnostic destination port
-> Section 3.3.9
| |--- Diagnostic community
-> Section 3.3.9
| |--- QoS traps to QCU
-> Section 3.3.9
| |--- QCU address
-> Section 3.3.9
| |--- QCU port
-> Section 3.3.9
| |--- QCU community
-> Section 3.3.9
-> Section 3.3.9
| |--- QoS to generic destination
|
|--- Features
| |--- Configuration
| | |--- General
| | | |--- Emergency number
-> Section 3.5.2
| | | |--- Voicemail number
-> Section 3.5.2
| | | |--- MWI LED
-> Section 3.28
| | | |--- Missed Call LED
-> Section 3.29
| | | |--- Allow refuse
-> Section 3.7.1
| | | |--- Hot / warm phone
-> Section 3.7.2
| | | |--- Hot / warm destination
-> Section 3.7.2
| | | |--- Initial digit timer
-> Section 3.7.3
| | | |--- Allow uaCSTA
-> Section 3.7.11
| | | |--- Server features
-> Section 3.7.10
| | | |--- Transfer on hangup
-> Section 3.7.5.2
| | | |--- Not used timeout
-> Section 3.7.12
| | | |--- DSS Pickup timer
-> Section 3.11.5.1
| | | |--- Bridging enabled
-> Section 3.11.2
| | | |--- Dial plan
-> Section 3.13.4
| | | |--- FPK prog. timer
-> Section 3.8
| | |--- Audio
| | | |--- Pickup tone allowed
-> Section 3.7.4.2
| | | |--- Pickup as ringer
-> Section 3.7.4.2
| | | |--- Pickup visual alert
-> Section 3.7.4.2
| | | |--- BLF alerting
-> Section 3.7.4.2
| | | |--- MLPP ringer (not used with OpenScape Voice)
| | | |--- Callback ringer
-> Section 3.7.6.1
| | | |--- Lower IL ringer
-> Section 3.30
| | |--- Keyset Lines (OpenStage 15/40/60/80)
| | | |--- Details For Keyset Line <n>
| | |
|--- Address
-> Section 3.11.1
| | |
|--- Ring on/off
-> Section 3.11.1
| | |
|--- Selection order
-> Section 3.11.1
|
| | |
--- Hot/warm action
-> Section 3.11.1
|
| |
--- Call recording
| |
|--- Recorder number
-> Section 3.7.13
| |
|--- Recording mode
-> Section 3.7.13
|
| |
--- Audible notification
-> Section 3.7.13
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
363
referenz.fm
Technical Reference
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|--- 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
| |--- Forwarding shown
|--- Addressing
| |--- MWI server URI
| |--- Conference
| |--- Group pickup URI
| |--- Callback FAC
| |--- Callback: busy (upto V2R2)
| |--- Callback: no reply (upto V2R2)
| |--- Callback: cancel all
| |--- BLF pickup code
|--- Call completion
| |--- Functional CCSS
| |--- Callback ringer
| |--- Allow after call (s)
| |--- Max. callbacks
|
--- 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
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
referenz.fm
Technical Reference
Menus
Menu
| | |
| |--- Video calls
| | |
|--- CTI
| | |
| |--- CTI control
|--- Services
| | |
| | |
|--- Bluetooth
| | |
|--- Web based mang.
| | |
|--- USB device access
| | |
|--- Backup to USB
| | |
|--- Feature togggle
|--- Phone lock
| | |
| | |
| | |--- Security
| |
|--- System
| |
|
|--- Server certificate
| |
|
|--- Use secure calls
| |
|
|--- 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
| |
|
|--- Last archived
|--- Archive when at
| |
|
|
| |
--- Faults
| |
|--- Security log entry
| |
|--- OCSR failure
| |
|--- 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
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
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Technical Reference
Menus
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366
| |--- Port
| |--- Account
| |--- Username
| |--- Password
| |--- FTP path
| |--- HTTPS base URL
| |--- 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
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
A31003-S2030-M100-10-76A9, 01/2014
OpenStage SIP V3R3 for OpenScape Voice, Administration Manual
referenz.fm
Technical Reference
Menus
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
| | | |--- Filename
| | |--- HPT dongle
| |
|--- 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
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
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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
| | | |--- Old urgent items
| | | |--- Alternative label
| | |--- Call logging
|--- FAC prefixes
| |
| |
| |--- Date and Time
| | |--- Time source
| | | |--- SNTP IP address
| | | |--- 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
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Technical Reference
Menus
Menu
| | |--- 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
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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
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Technical Reference
Menus
Menu
| | | |--- Change User password
| | |
|--- Admin password
| | |
|--- New password
|--- Confirm password
| | |
|
| |
--- Certificates
| |
|--- Generic
| |
| |--- 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.
| |
| |
| |--- Ringer setting
| | |--- Distinctive
| | | |--- <1 .... 15>
| | |
|--- Name
| | |
|--- Ringer sound
| | |
|--- Pattern melody
| | |
|--- Pattern sequence
| | |
|--- Duration
|--- Audible
| | |
| | |--- 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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Menus
Menu
|
|
|
|
|
|
|
|
|
|
|
|
| |--- Trace status
| |--- User notification
| |--- Remote server
| |--- Remote port
|--- Memory monitor
| |--- Disable reboot
| |--- High threshold
| |--- Low threshold
| |--- Working Hour start
| |--- Working Hour end
|--- Diagnostic call
|--- Prefix code
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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
Server Default Port
Client Default
Port
Protocol Stack
Payload transport (VoIP)
5010 - 5059
5010 - 5059
RTP - RTCP
Payload transport (VoIP)
5010 - 5059
5010 - 5059
SRTP - SRTCP
SIP subscriber - TCP is used
5060
32786 - 61000
SIP / TCP
SIP subscriber - TLS is used
5061
32786 - 61000
SIP / TLS
SIP subscriber - UDP is used
5060
5060
SIP / UDP
XML applications in phone, connec- --ting to an application server
32786 - 61000
HTTP / TCP
HTTPS / TCP-TLS
XML Push service
8085
---
HTTP / TCP
XML Push service
443
---
HTTPS / TCP-TLS
Directory access via LDAP
---
32786 - 61000
TCP
Directory access via LDAP
---
32786 - 61000
TCP- SSL/TLS
DHCP Client
---
68
DHCP / UDP
DNS Client
---
1024 - 65535
DNS / TCP_UDP
DLS contact me service - workpoint 8085
side
---
HTTP / TCP
Default communication with the DLS --workpoint interface
18443
HTTPS / TCP SSL / TLS
Secure communication with the DLS --workpoint interface
18444
HTTPS / TCP SSL / TLS
Connection to the control port of FTP 21
server
32786 - 61000
FTP / TCP
FTP client; uses the FTP server in
active mode
32786 - 61000 20
FTP / TCP
HTTPS file download server
443
32786 - 61000
HTTPS / TCP SSL/TLS
Client application which sends QDC --data to the QCU
32786 - 61000
SNMP / UDP
Part of SNMP-Agent - sending Traps ---
32786 - 61000
SNMP / UDP
Part of SNMP-Agent - receive Set/
Get commands
---
SNMP / UDP
372
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Default Port List
Service
Server Default Port
Client Default
Port
Protocol Stack
SNTP client - queries time informati- --on in unicast operation
123
SNTP / UDP
SNTP client - receives time
information in broadcast operation
123
---
SNTP / UDP
Web server for WBM access
8085
---
HTTP / TCP
Secure Web Server for WBM access 443
---
HTTPS / TCP SSL / TLS
OpenStage Phone Manager
65532
---
TCP - SSL/TLS
Remote Trace
---
514
UDP
HPT- debug IF (Available only if a
dongle file is present on phone.)
65532
---
TCP - SSL/TLS
SSH (Secure Shell Remote Login)
22
---
TCP
Syslog Client (sends Traces to Sys- --log Server)
32786 - 61000
UDP
Video H.263
5050-5059
5050-5059
RTP - RTCP
Secure Video H.263
5050-5059
5050-5059
SRTP - SRTCP
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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
Description
Error code
Network Problem
No network connection
LI1
Not Initialised
Waiting for data
I1
Unable to use LAN
802.1x error
LX1
Unable to use LAN
Physical connection missing
LP1
Unable to Register
Server timeout
RT2
Unable to Register
Server failed
RF2
Unable to Register
Authentication failed
RA2
Unable to Register
No number configured
RN2
Unable to Register
No server configured
RS2
Unable to Register
No registrar configured
RG2
Unable to Register
No DNS domain configured
RD2
Unable to Register
Rejected by server
RR2
Unable to Register
No phone IP address set
RI2
Survivability
Backup route active
B8
Survivability
Backup not configured
RS8
Survivability
Backup timeout
RT8
Survivability
Backup authentication failed
RA8
Cloud-Deployment
abandoned by user
Cloud-Deployment abandoned by user
Occurs when the pin prompt is dismissed
AU
Cloud-Deployment: Cloud-Deployment: unable to get the address RS
unable to get the
for the SEN Redirect server. DNS lookup
address for the SEN failed
Redirect server
Unable to establish Cloud-Deployment: unable to establish concontact with SEN
tact with SEN Redirect server - no reply
Redirect server - no
reply
Tabelle 4-1
374
RN
Troubleshooting Error Codes
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Troubleshooting: Error Codes
Problem
Description
Unable to establish Cloud-Deployment: unable to establish concontact with SEN
tact with SEN Redirect server - refused
Redirect server - refused
Unable to establish
contact with SEN
Redirect server unauthorised
Error code
RR
Cloud-Deployment: unable to establish con- RU
tact with SEN Redirect server - unauthorised
Unable to establish Cloud-Deployment: unable to establish concontact with SEN
tact with SEN Redirect server - no or invalid
Redirect server - no OCSP response
or invalid OCSP response
RO
Unable to establish Cloud-Deployment: unable to establish con- RV
contact with SEN
tact with SEN Redirect server - certificate reRedirect server - cer- voked
tificate revoked
Unable to get the
address for the Deployment server
Cloud-Deployment: unable to get the address DS
for the Deployment server. DNS lookup failed
Unable to establish Cloud-Deployment: unable to establish concontact with Deploy- tact with Deployment server - no reply
ment server
DN
Unable to establish Cloud-Deployment: unable to establish concontact with Deploy- tact with Deployment server - refused
ment server - refused
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
Example value Explanation
Local country code
44
International country code for the UK.
National prefix digit
0
Used in front of national codes when dialled
without international prefix.
Local national code
115
Area code within the UK (here: Nottingham).
Minimum local number
length
7
Minimum number of digits in a local PSTN
number (e. g. 3335333 = 7 digits).
Local enterprise node
780
Prefix to access Nottingham numbers from
within the Siemens network.
PSTN access code
9
Prefix to make an international call in the UK.
Operator codes
0,7800
Set of numbers to access the local operators.
(No blank after comma, or else the subsequent entry is ignored.)
Emergency numbers
999,555
Set of numbers to access emergency services. (No blank after comma, or else the subsequent entry is ignored.)
Initial extension digits
2,3,4,5,6,8
1st 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
Example value Explanation
Local code <1>
780
Enterprise node prefix (here: Nottingham).
International code <1>
+44115943
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.
Local code <2>
7007
Enterprise node prefix (here: Munich).
International code <2>
+49897007
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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Examples and HowTos
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
2345
External numbers
Local public form
External access
code
Not required
International gateway code
Use national code
Number stored in the
phonebook
+441159432345
Dial string sent when Internal numbers = Local enterprise form
dialing from the
Internal numbers = Always add node
phonebook
Internal numbers = Use external numbers
1234
7802345
9432345
Example 2: Internal number, different node
User entry
70072345
External numbers
Local public form
External access
code
Not required
International gateway code
Use national code
Number stored in the
phonebook
+498970072345
Dial string sent when Internal numbers = Local enterprise form
dialing from the
Internal numbers = Always add node
phonebook
Internal numbers = Use external numbers
2345
378
7802345
9432345
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Examples and HowTos
Canonical Dialing
Example 3: External number, same local national code as the local phone
User entry
011511234567
External numbers
Local public form
External access
code
Not required
International gateway code
Use national code
Number stored in the
phonebook
+4411511234567
Dial string sent when External numbers = Local public form
dialing from the
External numbers = National public form
phonebook
External numbers = International form
234567
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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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Examples and HowTos
How to Create Logo Files for OpenStage Phones
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.
>
1.
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.
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
Color Code
OpenStage 60
Crystal Sea
#E7E7E7
OpenStage 60
Warm Grey
#4242421
OpenStage 80
Crystal Sea
#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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How to Create Logo Files for OpenStage Phones
2.
Create a New Image
Create an image with the size according to the phone type:
Phone Type
Size (px)
OpenStage 60
240 x 70
OpenStage 80
480 x 142
Adobe Photoshop:
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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How to Create Logo Files for OpenStage Phones
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".
v
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.
>
5.3.1
The Corporate Phonebook is available on OpenStage 15/20/40/60/80 phones with
firmware version V3R3 onwards.
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
LDAP Attribute
Example
Value
Last name
ATTRIB01
sn
Doe
First name
ATTRIB02
givenName
John
Business 1
ATTRIB03
telephoneNumber
9991234
Business 2
ATTRIB04
facsimileTelephoneNumber 9992345
Mobile
ATTRIB05
mobile
017711223344
Private
ATTRIB06
homePhone
441274333444
Company
ATTRIB07
o
Example Inc.
Address 1
ATTRIB08
departmentNumber
0815
Address 2
ATTRIB09
Job function
ATTRIB10
title
Product Manager
Email
ATTRIB11
mail
doe@example.com
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
LDAP Attribute
Example
Value
Last name
ATTRIB01
sn
Doe
First name
ATTRIB02
givenName
John
Business 1
ATTRIB03
ipPhone
9991234
Business 2
ATTRIB04
otherTelephone
9992345
Mobile
ATTRIB05
mobile
017711223344
Private
ATTRIB06
homePhone
441274333444
Company
ATTRIB07
company
Example Inc.
Address 1
ATTRIB08
department
Administration
Address 2
ATTRIB09
Job function
ATTRIB10
title
Product Manager
Email
ATTRIB11
mail
doe@example.com
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="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
FTP Server address
FTP Server port
21
FTP account
FTP username
phone
FTP password
FTP path
media
HTTPS base URL
Filename
After submit
Submit
ldap-template.txt
do nothing
Reset
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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
TCP
LDAP Secure port
636
389
LDAP Server port
Authentication
Anonymous
User Name
Password
Search trigger timeout
3.
3
Reset
Submit
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.
392
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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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
•
•
•
•
•
396
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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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
6.
Enter the following data:
to create the first dial plan rule.
Parameter
Value
Description/Remarks
Digit string
3
This rule matches numbers beginning with 3. For instance, theses might be internal numbers.
Action
S
When all criteria are met, the number is sent to the server.
Minimum length
4
This rule matches numbers with a length of 4 digits.
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
Value
Description/Remarks
Digit string
0
This rule matches numbers beginning with 0. In the
USA, this number calls the operator.
Action
C
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.
Minimum length
1
This rule matches numbers with a length of 1 digits.
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.
398
Click on
to create the third dial plan rule.
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Dial Plan
10. Enter the following data:
Parameter
Value
Description/Remarks
Digit string
011
This rule matches numbers beginning with 011. In the
USA, this digit string is the prefix international calls.
Action
S
When the entered digit string reaches the Minimum
length, the Timer is started. On expiry, the digit string
is sent.
Minimum length
4
When the length of the digit sequence entered by the
user reaches this value, the Timer is started.
Maximum length 13
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.
Timer
3
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.
Terminating
Character
#
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 telecommunications 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 network 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 bandwidth.
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 simultaneously. 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 conferences. 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 forwarding 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-order 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 especially in streaming media applications such as ->VoIP. Reliability and order of packet delivery 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 conversations 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
Z
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
408
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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Dongle Key (Download) 1-237
Download 1-217
DSS key settings 1-187
DST Zone (Daylight Saving Time Zone) 1104
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
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) 1194
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
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
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
N
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) 1194
Local Country Code (International Code) 1194
Local Enterprise Number 1-194
Local National Code (Local Area Code) 1194
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
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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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
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
WBM (Web Based Management) 1-20, 1-26,
1-407
Web Interface Menu 1-311
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
X
XML ApplicationsApplications
XML 1-246
Xpressions 1-246
Z
Zip Tone 1-164
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
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