Siemens Asterisk OpenStage 20, Asterisk OpenStage 60, Asterisk OpenStage 80, Asterisk OpenStage 40 Owner's Manual

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Communication for the open minded

Siemens Enterprise Communications www.siemens.com/open

Documentation

Asterisk OpenStage 20, OpenStage 40, OpenStage 60, OpenStage 80

Administration Manual

A31003-S2000-M102-1-76A9

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Siemens Enterprise Communications GmbH & Co. KG 2007 Hofmannstr. 51, D-80200 München

Siemens Enterprise Communications GmbH & Co. KG is a Trademark Licensee of Siemens AG

Reference No.: A31003-S2000-M102-1-76A9

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. OpenScape, OpenStage and HiPath are registered trademarks of Siemens Enterprise Communications GmbH & Co. KG. All other company, brand, product and service names are trademarks or registered trademarks of their respective holders.

Siemens Enterprise Communications www.siemens.com/open

Communication for the open minded

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1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1

1.1 Important Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-1

1.2 Maintenance Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2

1.3 About the Manual. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2

1.4 Conventions for this Document . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-2

1.5 The OpenStage Family . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-3

1.5.1 OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-3

1.5.2 OpenStage 40 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-4

1.5.3 OpenStage 20 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-5

1.6 Administration Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-5

1.6.1 Web-based Management (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-5

1.6.2 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1-6

2 Startup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1

2.1 Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-1

2.2 Assembling and Installing the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2

2.2.1 Shipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2

2.2.2 Connectors at the bottom side . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2

2.2.3 Assembly. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-4

2.2.4 Connecting the Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-5

2.3 Quick Start . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-7

2.3.1 Access the Web Interface (WBM) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-7

2.3.2 Set the Terminal Number . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-8

2.3.3 Basic Network Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-9

2.3.4 Date and Time / SNTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-9

2.3.5 SIP Server Address. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-9

2.3.6 Extended Network Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-10

2.3.7 Vendor specific: VLAN Discovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-10

2.3.8 Example DHCP server configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-11

2.3.9 Registering at Asterisk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-13

3 Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1

3.1 Access via Local Phone. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-1

3.2 LAN Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-5

3.2.1 LAN Port Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-5

3.2.2 VLAN. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-7

3.2.2.1 Automatic VLAN discovery using DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-8

3.2.2.2 Automatic VLAN discovery using LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . 3-9

3.2.2.3 Manual configuration of a VLAN ID . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-11

3.2.3 LLDP-MED Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-12

3.3 IP Network Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13

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3.3.1 Quality of Service (QoS). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13

3.3.1.1 Layer 2 / 802.1p. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-13

3.3.1.2 Layer 3 / Diffserv . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-14

3.3.2 Use DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-16

3.3.3 IP Address - Manual Configuration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-18

3.3.4 Default Route/Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-19

3.3.5 Specific IP Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-20

3.3.6 DNS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-21

3.3.6.1 DNS Domain Name . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-21

3.3.6.2 DNS Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-22

3.3.7 SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-23

3.4 System Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27

3.4.1 Terminal and User Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27

3.4.1.1 Terminal Identity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-27

3.4.1.2 Display Identity. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-28

3.4.2 Emergency and Voice Mail. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-29

3.4.3 Energy Saving (OpenStage 40/60/80) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-30

3.4.4 Date and Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-31

3.4.4.1 SNTP is available, but no automatic configuration by DHCP server . . . . . . . 3-31

3.4.4.2 No SNTP server available . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-33

3.4.5 SIP Addresses and Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-34

3.4.5.1 SIP Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-34

3.4.5.2 SIP Ports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-35

3.4.6 SIP Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-36

3.4.7 SIP Communication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-39

3.4.7.1 Outbound Proxy. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-39

3.4.7.2 SIP Transport Protocol. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-40

3.4.8 SIP Session Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-41

3.4.9 Resilience and Survivability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43

3.4.9.1 TLS Connectivity Check. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-43

3.4.9.2 Response Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-44

3.4.9.3 Non-INVITE Transaction Timer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-45

3.4.9.4 Maximum Registration Backoff Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-46

3.4.9.5 Backup SIP Server. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-47

3.5 Features - Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-50

3.5.1 Allow Refuse. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-50

3.5.2 Call Transfer. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-51

3.5.2.1 Transfer on Ring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-51

3.5.2.2 Transfer on Hangup. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-52

3.5.3 Message Waiting Address . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-53

3.5.4 Server Based Features. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-54

3.5.5 uaCSTA Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-55

3.5.6 Local Menu Timeout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-56

3.6 Features - Program Keys . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57

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3.6.1 Selected dialing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57

3.6.2 Repeat dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-57

3.6.3 Forwarding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-58

3.6.4 Ringer off. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-58

3.6.5 Hold. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59

3.6.6 Alternate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59

3.6.7 Move blind. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-59

3.6.8 Join . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-60

3.6.9 Deflect . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-60

3.6.10 Shift . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-60

3.6.11 Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61

3.6.12 Headset. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61

3.6.13 Do Not Disturb. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-61

3.6.14 Repertory dial . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-62

3.6.15 Consult and transfer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-62

3.6.16 Call waiting toggle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-63

3.6.17 Server feature . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-63

3.6.18 BLF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-64

3.6.19 Key Modules . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-65

3.7 Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-66

3.7.1 Canonical Dialing Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-66

3.7.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-70

3.8 Transferring Phone Software, Application and Media Files . . . . . . . . . . . . . . . . . . . . 3-72

3.8.1 FTP/HTTPS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-72

3.8.2 Common FTP/HTTPS Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-72

3.8.3 Phone Software. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-74

3.8.3.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-74

3.8.3.2 Download/Update Phone Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-76

3.8.4 Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-77

3.8.4.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-77

3.8.4.2 Download Music on Hold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-79

3.8.5 Picture Clips . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-80

3.8.5.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-80

3.8.5.2 Download Picture Clip . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-82

3.8.6 LDAP Template. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-83

3.8.6.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-83

3.8.6.2 Download LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-85

3.8.7 Logo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-86

3.8.7.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-86

3.8.7.2 Download Logo. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-88

3.8.8 Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-89

3.8.8.1 FTP/HTTPS Access Data. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-89

3.8.8.2 Download Screensaver . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-91

3.8.9 Ringer File. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-92

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3.8.9.1 FTP/HTTPS Access Data . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-93

3.8.9.2 Download Ringer File. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-95

3.9 Corporate Phonebook: Directory Settings. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-96

3.9.1 LDAP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-96

3.10 Speech . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-98

3.10.1 RTP Base Port . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-98

3.10.2 Codec Preferences. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-99

3.10.3 Audio Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-101

3.11 Applications. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-102

3.11.1 XML Applications/Xpressions (OpenStage 60/80) . . . . . . . . . . . . . . . . . . . . . . 3-102

3.11.1.1 Setup/Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-102

3.11.1.2 HTTP Proxy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-106

3.11.1.3 Modify an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-108

3.11.1.4 Remove an Existing Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-109

3.11.1.5 Application Start by Program Key . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-110

3.12 Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-111

3.13 Troubleshooting: Lost Password. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-112

3.14 Restart Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-113

3.15 Factory Reset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-114

3.16 Display License Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-115

3.17 Diagnostics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116

3.17.1 Display General Phone Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-116

3.17.2 LAN Monitoring. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-117

3.17.3 LLDP-MED . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-118

3.17.4 IP Tests . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-120

3.17.5 Process and Memory Information. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-121

3.17.6 Fault Trace Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-122

3.17.7 Easy Trace Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-128

3.17.7.1 Bluetooth Handsfree . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-128

3.17.7.2 Bluetooth Headset . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-128

3.17.7.3 Call Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-129

3.17.7.4 Call Log . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-129

3.17.7.5 LDAP Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-130

3.17.7.6 DAS Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-130

3.17.7.7 802.1x . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-131

3.17.7.8 Help Application. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-131

3.17.7.9 Sidecar. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-131

3.17.7.10 Key Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-132

3.17.7.11 LAN Connectivity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-132

3.17.7.12 Local Phonebook . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-132

3.17.7.13 Messaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-133

3.17.7.14 Mobility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-133

3.17.7.15 Phone administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-133

3.17.7.16 Server based applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-134

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3.17.7.17 Speech . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-134

3.17.7.18 Tone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-134

3.17.7.19 USB Backup/Restore . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-134

3.17.7.20 Voice Dialling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135

3.17.7.21 Web Based Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-135

3.17.7.22 No Tracing for All Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-136

3.17.8 QoS Reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-137

3.17.8.1 Conditions and Thresholds for Report Generation . . . . . . . . . . . . . . . . . . 3-137

3.17.8.2 View Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-140

3.17.9 Core dump . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144

3.17.10 Remote Tracing - Syslog. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-144

3.18 Bluetooth . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3-145

4 Examples and HowTos. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1

4.1 Canonical Dialing. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1

4.1.1 Canonical Dialing Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-1

4.1.2 Canonical Dial Lookup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-2

4.1.2.1 Conversion examples . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-3

4.2 How to Create Logo Files for OpenStage Phones. . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-5

4.2.1 For OpenStage 40. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-5

4.2.2 For OpenStage 60/80 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-6

4.3 How to Set Up the Corporate Phonebook (LDAP). . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-9

4.3.1 Prerequisites: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-9

4.3.2 Create an LDAP Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-10

4.3.3 Load the LDAP Template into the Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-13

4.3.4 Configure LDAP Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-14

4.3.5 Test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4-14

5 Technical Reference. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1

5.1 Menus . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1

5.1.1 Web Interface Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1

5.1.1.1 Menu Structure . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-1

5.1.1.2 Web Pages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-4

5.1.2 Local Phone Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-31

5.2 Troubleshooting: Error Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5-39

Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6-1

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7-1

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Overview

Important Notes

1Overview

1.1 Important Notes

Do not operate the equipment in environments where there is a danger of explosions.

For safety reasons the phone should only be operating using the supplied plug in power unit.

Use only original Siemens accessories!

Using other accessories may be dangerous, and will invalidate the warranty, extended manufacturer’s liability and the CE mark.

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.

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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Maintenance Notes

1.2 Maintenance Notes

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 SIP. 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 Siemens Enterprise Communications website (

http://www.enterprise-communications.siemens.com) and on the Siemens Enterprise Wiki

(http://wiki.siemens-enterprise.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 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 "/".

Do not operate the telephone in environments where there is a danger of explosions.

Use only original Siemens accessories. 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.

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The OpenStage Family

1.5 The OpenStage Family

1.5.1 OpenStage 60/80

1 The Handset lets you pick up and dial calls in the usual manner.

2 The Graphics Display provides intuitive support for telephone operation.

3 The user-friendly Application Keys provide easy access to your telephone’s

applications.

4 With the TouchGuide, the user/administrator can navigate in the various pho-

ne functions, applications, and configuration menus.

5 You can customize your telephone in line with your personal needs by assig-

ning individual phone numbers and functions to the Program Keys.

6 Press the Function Keys to access frequently used telephony functions.

7 The Audio Keys let you optimize the audio settings on your telephone.

8 With the TouchSlider, the user can adjust the volume, e.g. of ringtones.

9 Inbound calls are visually signaled on the Call Display.

10 The Keypad is used for entering phone numbers and text.

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The OpenStage Family

1.5.2 OpenStage 40

1 The Handset lets you pick up and dial calls in the usual manner.

2 The Graphics Display provides intuitive support for telephone operation.

3 The user-friendly Application Keys provide easy access to your telephone’s

applications.

4 With the Navigation Key, the user/administrator can navigate in the various

phone functions, applications, and configuration menus.

5 You can customize your telephone in line with your personal needs by assig-

ning individual phone numbers and functions to the Program Keys.

6 Press the Function Keys to access frequently used telephony functions.

7 The Audio Keys let you optimize the audio settings on your telephone.

8 Inbound calls are visually signaled on the Call Display.

9 The Keypad is used for entering phone numbers and text.

6 7

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Administration Interfaces

1.5.3 OpenStage 20

1.6 Administration Interfaces

You can configure the OpenStage phone by using any of the following methods.

1.6.1 Web-based Management (WBM)

This method employs a web browser for communication with the phone via HTTP or HTTPS. It is applicable for remote configuration of individual IP phones in your network. Direct access to the phone is not required.

1 The Handset lets you pick up and dial calls in the usual manner.

2 The Display provides intuitive support for telephone operation.

3 The user-friendly Application Keys provide easy access to your telephone’s

applications.

4 Press the Function Keys to access frequently used telephony functions.

5 With the Navigation Key, 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.

To use this method, the phone must first obtain IP connectivity.

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Administration Interfaces

1.6.2 Local Phone Menu

This method provides direct configuration of an the OpenStage phone. Direct access to the phone is required.

As long as the IP connection is not properly configured, you have to use this method to set up the phone.

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Startup

Prerequisites

2Startup

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.

• Asterisk SIP server.

• An FTP Server for file transfer, e. g. firmware, configuration data, application software.

• A DHCP (Dynamic Host Configuration Protocol) server (recommended).

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.

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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

• Emergency Number Sticker

2.2.2 Connectors at the bottom side

OpenStage 60

Switch

Headset

Handset

Keyboard

Power supply

Key Module

USB Extension

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Startup

Assembling and Installing the Phone

OpenStage 40 (similar to OpenStage 20, except 1)

Switch

Headset

[1]

Handset

Power supply

Key Module

[1]

[1] OpenStage 40 only

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Assembling and Installing the Phone

2.2.3 Assembly

1. Handset

Insert the plug on the long end of the handset cable into the jack on the base of the telephone and press the cable into the groove provided for it. Next, insert the plug on the short end of the handset cable into the jack on the handset.

2. Emergency Number Sticker

Write your telephone number and those for the fire and police departments on the included label and attach it to the telephone housing underneath the handset (see arrow).

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Assembling and Installing the Phone

2.2.4 Connecting the Phone

1. Plug the LAN cable into the connector at the bottom of the telephone and connect the cable to the LAN resp. switch. If PoE (Power over Ethernet) is to be used, the PSE (Power Sourcing Equipment) must meet the IEEE 802.3af specification.

For details about the required power supply, see the following table:

2. Only if Power over Ethernet (PoE) is NOT supported:

Plug the power supply unit into the mains. Connect the plug-in power supply unit to the jack at the bottom of the phone.

Model Power Consumption/Supply

OpenStage 20 E Power Class 1

OpenStage 20 Power Class 1

OpenStage 20 G Power Class 2

OpenStage 40

1 Includes 1 Key Module.

Power Class 2

OpenStage 40 + 2nd Key Module Power Class 3

OpenStage 40 G

Power Class 3

OpenStage 40 G + 2nd Key Module External power unit required

OpenStage 60/80

2 Includes 1 Key Module + USB-Extension with Acoustic Unit.

Power Class 3

OpenStage 60/80 + 2nd Key Module Power Class 3

OpenStage 60/80 G

Power Class 3

OpenStage 60/80 G + 2nd Key Module External power unit required

Table 2-1

Use only the plug-in power supply unit fitting the OpenStage phone: EU: C39280-Z4-C510 UK: C39280-Z4-C512 USA: C39280-Z4-C511

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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)

To prevent damage on the OpenStage phone, connect an USB stick using the adapter cable C39195-Z7704-A5.

Do not connect a USB hub to the phone’s USB port, as this may lead to stability problems.

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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.

2.3.1 Access the Web Interface (WBM)

1. Open your web browser and enter the transfer protocol, IP address and port number of your phone. If HTTP is used, port 8085 must be added, for example http://192.168.1.15:8085. For HTTPS, the phone uses the standard port 443.

After entering the URL, the browser might display a certificate notification. 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.

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.

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Quick Start

2. Click on the tab "Administrator Pages". In the dialog box, enter the admin password:

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 extension 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. For further information about this setting, please refer to Section 3.4.1.1, “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.

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2.3.3 Basic Network Configuration

For basic functionality, the following parameters must be provided, preferably by DHCP:

• 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 3.3.3, “IP Address - Manual Configuration” for IP address and subnet mask, and Section 3.3.4, “Default Route/Gateway” for default route.

2.3.4 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 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 3.4.4, “Date and Time”.

2.3.5 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 3.4.5.1, “SIP Addresses”.

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2.3.6 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 3.3.5, “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 3.3.6.1, “DNS Domain Name”.

2.3.7 Vendor specific: VLAN Discovery

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

3.2.2.1, “Automatic VLAN discovery using DHCP”).

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.

The VLAN ID can also be configured by LLDP-MED (see Section 3.2.2.2, “Automatic VLAN discovery using LLDP-MED”).

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2.3.8 Example DHCP server configuration

The following description demonstrates by example how a DHCP server can be configured in order to build up a working infrastructure for OpenStage phones.

The configuration example is tested with ISC DHCP server version 3 running on Debian Linux (package: dhcp3-server).

On startup, the OpenStage phone sends a DHCP discovery message which contains the vendor-class-identifier option. The fixed value is OptiIpPhone. This identifier is used by the DHCP server to send a specified set of sub-options to each OpenStage phone by means of vendor encapsulated options. The vendor-encapsulated-options option allows for defining custom-made options in either a single series of bytes or a sequence of parameters which is structured as follows: 1) single-byte vendor-specific option code; 2) single-byte length indication for the option data (the length does not include itself or the option code); 3) the data itself in as many bytes of data as are specified in the length indication.

The value of the vendor-encapsulated-options option can either be set directly by coding the bytes in hexadecimal notation, or, preferably, by using the option space statement. This statement is available with ISC DHCP 3.0.

To assign the specific vendor-encapsulated-options to the appropriate devices, the use of classes is recommended. In the example, the class vendor-classes is divided into subclasses automatically for each vendor-class-identifier option that is received from the clients. As OpenStage phones send the value OptiIpPhone, the specifications for theses phones are defined in the correpsonding sub-class.

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Example dhcpd.conf

# General configuration for all clients in the subnet subnet 192.168.2.0 netmask 255.255.255.0 { option domain-name-servers 192.168.2.2; option broadcast-address 192.168.2.255; option routers 192.168.2.251; option subnet-mask 255.255.255.0; option domain-name "phones.intranet"; option ntp-servers 192.43.244.18; option time-offset 3600; default-lease-time 864000; max-lease-time 8640000; }

# Set up an option space to contain the vendor-encapsulated options option space OpenStage; option OpenStage.vendor code 1 = string; option OpenStage.vlanid code 2 = unsigned integer 32; option OpenStage.dls code 3 = string;

# Each vendor-class-identifier will constitute a class class "vendor-classes" { match option vendor-class-identifier; }

# Specifications for OpenStage phones subclass "vendor-classes" "OptiIpPhone" { vendor-option-space OpenStage; option OpenStage.vendor "Siemens"; option OpenStage.vlanid 4; option OpenStage.dls "sdlp://192.168.2.103:18443"; option sip-server 192.168.2.117; }

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2.3.9 Registering at Asterisk

For registration at the Asterisk SIP server, a SIP user ID and passwort must be provided by the phone. The following procedure describes the configuration using the web interface (see Section 2.3.1, “Access the Web Interface (WBM)”; if the web interface is not applicable, please refer to Section 3.4.6, “Authenticated Registration”) for configuration via the local menu.

1. In the administration menu, select System > Registration. The Registration dialog opens.

2. Make sure that SIP server address and SIP registrar address contain the IP address of your Asterisk server. If not provided by DHCP or a provisioning service, enter the appropriate values. If necessary, enter the 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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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, “Access the Web Interface (WBM)”.

3.1 Access via Local Phone

1. Access the Administration Menu OpenStage 60/80:

The v key toggles between the Settings menu, the Applications menu, and the applications currently running. Press the v key repeatedly until the "Settings" tab is active. (The

v key toggles between the Settings menu, the Applications menu, and the applications

currently running.)

OpenStage 20/40:

Press the keys D, l, and i consecutively to select 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 recommended to change the password (see Section 3.12, “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) -> (ABC) -> back to start.

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.

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Access via Local Phone

3. Navigate within the Administration Menu

OpenStage 60/80

Use the TouchGuide to navigate and execute administrative actions in the administration menu.

For using the TouchGuide, see the following figure:

OpenStage 40

Use the 5-way Navigator to navigate and execute administrative actions in the administration menu.

Press the mkey briefly:

- scroll up

Press the l key briefly:

- scroll down

Press the g key:

- open a context menu

Press the h key:

- cancel a function

Press the i key:

- confirm entries

Run your finger around the sensor ring W:

- browse lists and menus

- set up volume

Press the mkey briefly:

- scroll up Hold down:

- scroll to top of list

Press the

l key briefly:

- scroll down Hold down:

- scroll to end of list

Press the

g key:

- open a context menu

- down one level

Press the

h key:

- cancel a function

- delete character left of cursor

- up one level

Press the

i key:

- confirm entries

- perform an action

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OpenStage 20

Use the 3-way Navigator to navigate and execute administrative actions in the administration menu.

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. Press the key to enter the selective list. Use the Sensor Wheel resp. the m and l key to scroll up and down in the selective list. To select a list entry, press the i key.

5. Enter the parameter value

For selecting numbers and characters, you can use special keys. See the following table:

Key Function

* Switch to punctuation and special characters. # Toggle between lowercase characters, uppercase characters, and digits in

the following order: (Abc) -> (abc) -> (123) -> (ABC) -> back to start.

Table 3-1

Press the mkey briefly:

- scroll up Hold down:

- scroll to top of list

Press the l key briefly:

- scroll down Hold down:

- scroll to end of list

Press the i key:

- confirm entries

- perform an action

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Access via Local Phone

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:

Additionally, you can use the following keys on the keypad as shortcuts for the selection of character groups

OpenStage 20/40

With the OpenStage 20/40, use the keypad for entering parameters. With the 3 way/5 wayNavigator, 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 .

Element Function

Switch to punctuation and special characters.

Toggle between lowercase characters, uppercase characters, and digits.

Element with focus

Letters, digits, punctuation marks or special characters

Command line

Shift to numeric entry

Shift to upper/lower case

Copy contents of active field to clipboard

Move cursor left/right Shift to punctuation and special characters

Insert clipboard contents at cursor position

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3.2 LAN Settings

3.2.1 LAN Port Settings

The OpenStage phone provides an integrated switch which connects the LAN, the phone itself and a PC port. By default, the switch will auto negotiate transfer rate (10/100 Mb/s, 1000 Mb/s with OpenStage 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.

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.

When PC port autoMDIX is enabled, the switch determines automatically whether a regular MDI connector or a MDI-X (crossover) connector is needed, and configures the connector accordingly.

Data required

• LAN port speed / LAN port type: Settings for the ethernet port connected to a LAN switch. Value range: "Automatic," "10 Mbps half duplex", "10 Mbps full duplex", "100 Mbps half duplex", "100 Mbps full duplex", "1 Gbps half duplex" (OpenStage 60/80 G), "1 Gbps full duplex" (OpenStage 60/80 G). 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", "1 Gbps half duplex" (OpenStage 60/80 G), "1 Gbps full duplex" (OpenStage 60/80 G). Default: "Automatic".

• PC port mode / PC port status: Controls the PC port. Value range: "disabled", "enabled", "mirror". Default: "disabled".

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. If the phone is set up to manually configure the switch port settings, the cable detection mechanism is disabled. In this case, care must be taken to use the correct cable type.

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• PC port autoMDIX: Switches between MDI and MDI-X automatically. Value range: "On", "Off". Default: "Off".

Administration via WBM

Network > Port configuration

Administration via Local Phone

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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).

Physically partitioning the LAN 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.

In a layer 1 VLAN, the ports of 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 possible to assign one device, i.e. one MAC address, to different VLANs.

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

The implementation of a voice network based on VLANs requires the network infrastructure (the switch fabric) to support VLANs.

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3.2.2.1 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". The DHCP server must be configured to supply the Vendor Unique Option in the correct Siemens 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. In these circumstances network routing will probably not be correct.

Administration via WBM

Network > IP configuration

First, click on change mode. Afterwards, the IP configuration mode dialog opens.

Network > IP configuration > change mode

To enable VLAN discovery by DHCP, select DHCP used in the Discovery mode menu. Afterwards, click Submit.

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To enable VLAN discovery by DHCP, select DHCP used in the Discovery mode menu.

3.2.2.2 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 > IP configuration

First, click on change mode. Afterwards, the IP configuration mode dialog opens.

|--- Administration |--- Network |--- IP Configuration |--- Discovery mode

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Network > IP configuration > change mode

To enable VLAN discovery by LLDP-MED, select LLDP-MED with DHCP in the Discovery

mode menu. Afterwards, click Submit.

Administration via Local Phone

To enable VLAN discovery by DHCP, select LLDP-MED with DHCP in the Discovery mode menu.

|--- Administration |--- Network |--- IP Configuration |--- Discovery mode

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3.2.2.3 Manual configuration of a VLAN ID

To configure layer 2 VLAN manually, first make shure that VLAN discovery is set to "Manual" (see Section 3.2.2.1, “Automatic VLAN discovery using DHCP”). 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 > IP configuration

First, click on change mode. Afterwards, the IP configuration mode dialog opens.

Network > IP configuration > change mode

To enable manual VLAN configuration, select Manual settings in the Discovery mode menu. Afterwards, click Submit.

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3.2.3 LLDP-MED Operation

OpenStage phones support LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery) for auto-configuration and network management. The auto-configurable parameters are VLAN ID (see Section 3.2.2, “VLAN”) and Quality of Service parameters (see Section 3.3.1, “Quality of Service (QoS)”).

The data sent by a network device is stored in neighboring network devices in MIB (Manegement 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.

Administration via WBM

Administration via Local Phone

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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 / 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.

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 default: Sets the default CoS (Class of Service) value. Value range: 0-7. Default: 0.

Layer 2 and 3 QoS for voice transmission can be set via LLDP-MED (see Section

3.17.3, “LLDP-MED”). If so, the value can not be changed by any other interface.

Three Bits Used for CoS

(User Priority)

PREAM. SFD DA SA

TAG

4 Bytes

PT DATA FCS

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Administration via WBM

Network > QoS

Administration via Local Phone

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) 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 (high priority), AF2Y, AF3Y and AF4Y (low priority).

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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: "AF11", "AF12", "AF13", "AF21", "AF22", "AF23", "AF31", "AF32", "AF33", "AF41", "AF42", "AF43", "EF", "CS7". Default: "EF".

• Layer 3 signalling: Sets the CoS (Class of Service) value for signaling. Value range: "AF11", "AF12", "AF13", "AF21", "AF22", "AF23", "AF31", "AF32", "AF33", "AF41", "AF42", "AF43", "EF", "CS7"). Default: "AF31".

Administration via WBM

Network > QoS

Administration via Local Phone

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IP Network Parameters

3.3.2 Use DHCP

For an example configuration, see Section 2.3.8, “Example DHCP server configuration”.

If this parameter is set to "Yes", 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 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)

• SNTP IP Address (NTP Server option 42)

• Timezone offset (Time Server Offset option 2)

• Primary/Secondary IP Addresses (DNS Server option 6)

• DNS Domain Name (DNS Domain option 15)

• SIP Addresses / SIP Server & Registrar (SIP Server option 120)

• Vendor Unique (option 43)

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Administration via WBM

Network > IP configuration

Administration via Local Phone

|--- Administration |--- Network |--- IP Configuration |--- Use DHCP

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IP Network Parameters

3.3.3 IP Address - Manual Configuration

If not provided by DHCP dynamically, the phone’s IP address and subnet mask must be specified manually.

Data required

• IP address: used for addressing the phone.

• Subnet mask: subnet mask that is needed for the subnet in use.

Administration via WBM

Network > IP configuration

Administration via Local Phone

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3.3.4 Default Route/Gateway

If not provided by DHCP dynamically (see Section 3.3.2, “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 > IP configuration

Administration via Local Phone

|--- Administration |--- Network |--- IP Configuration |--- Route (default)

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IP Network Parameters

3.3.5 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 or if the LAN is divided into subnets.

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 > IP configuration

Administration via Local Phone

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IP Network Parameters

3.3.6 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 nameservers needed for DNS resolving.

3.3.6.1 DNS Domain Name

This is the name of the phone’s local domain.

Administration via WBM

Network > IP configuration

Administration via Local Phone

|--- Administration |--- Network |--- IP Configuration |--- DNS domain

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IP Network Parameters

3.3.6.2 DNS Servers

If not provided by DHCP automatically, a primary and a secondary DNS server can be configured.

Data required

• Primary DNS: IP address of the primary DNS server.

• Secondary DNS: IP address of the secondary DNS server.

Administration via WBM

Network > IP configuration

Administration via Local Phone

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3.3.7 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.

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IP Network Parameters

Traps specific to OpenStage phones

Currently, the following traps are defined:

TraceEventFatal: sent if severe trace events occur; aimed at expert users.

TraceEventError: sent if severe trace events occur; aimed at expert users.

Data required

• Trap sending enabled: Enables or disables the sending of a TRAP message to the SNMP manager. Value range: "Yes", "No". Default: "No".

• Trap destination: IP address or hostname of the SNMP manager that receives traps.

• Trap destination port: Port on which the SNMP manager is receiving TRAP messages.

• Default: 162.

• Trap community: SNMP community string for the SNMP manager receiving TRAP mes- sages. 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.

• 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 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".

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IP Network Parameters

• QoS to generic destination / QoS to generic device: Enables or disables the sending of QoS traps to a generic destination. Value range: "Yes", "No". Default: "No".

Administration via WBM

System > SNMP

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IP Network Parameters

Administration via Local Phone

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System Settings

3.4 System Settings

3.4.1 Terminal and User Identity

3.4.1.1 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

Administration via Local Phone

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3.4.1.2 Display Identity

If an individual name oder 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 or Terminal name.

Administration via WBM

System > System Identity

Administration via Local Phone

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3.4.2 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 a mailbox located at a remote server shall be used, its Voice mail number must be entered.

Administration via WBM

System > Features > Configuration

Administration via Local Phone

If more than one emergency number is needed, additional numbers can be configured in the canonical dial settings (Section 3.7.1, “Canonical Dialing Configuration”).

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System Settings

3.4.3 Energy Saving (OpenStage 40/60/80)

After the phone has been inactive within the timespan specified here, the display backlight is switched off. The length of this timespan ranges from 2 hours to 8 hours. The default value is 3 hours.

Administration via WBM

Local functions > Energy saving

Administration via Local Phone

|--- Administration |--- Local Functions |--- Energy saving |--- Backlight timeout

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System Settings

3.4.4 Date and Time

If the DHCP server in your network provides information about the SNTP server access, the correct date and time is automatically shown on the phone. If the DHCP server in your network does not provide an SNTP address, you have to set the SNTP address manually, using the SNTP IP address parameter. If no SNTP server is available, you have to configure the date and time manually.

For correct display of the current time, the Timezone offset must be set appropriately. This is the time offset from UTC (Coordinated Universal Time). 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 daylight saving, the administrator can choose whether daylight saving time is activated manually or automatically. If Daylight saving is enabled, and

Auto time change is disabled, daylight saving time (DST) is in effect immediately. If Auto time change is enabled, daylight saving is controlled by the Time zone parameter. This selects the

daylight saving time zone which is characterized by the start and end date for daylight saving time.

The Difference (minutes) provides the time difference for daylight saving time in minutes. This parameter is required also when Auto time change is enabled. In Germany, for instance, as in most countries, this is +60.

3.4.4.1 SNTP is available, but no automatic configuration by DHCP server

Data required

• SNTP IP address: IP address or hostname of the SNTP server.

• Timezone offset (hours): Shift in hours corresponding to UTC.

• Daylight saving: Enables or disables daylight saving time in conjunction with Auto time change. Value range: "Yes", "No".

• Difference (minutes): Time difference when daylight saving time is in effect.

• Auto time change / Auto DST: Enables or disables automatic control of daylight saving time according to the Time zone. Value range: "Yes", "No".

• 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".

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System Settings

Administration via WBM

Date and Time

Administration via Local Phone

|--- Administration |--- Date and Time |--- SNTP IP address |--- Timezone offset

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3.4.4.2 No SNTP server available

If no SNTP server is available, date and time must be set manually.

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.

Administration via WBM

(User pages >) Date and time

Administration via Local Phone

The manual setting of time and tate is located in the user menu, not in the administrator menu.

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System Settings

3.4.5 SIP Addresses and Ports

3.4.5.1 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. 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. The SIP gateway performs a conversion of SIP to TDM, which enables to phone directly into the public network.Data required

• SIP server address: IP address or host name of the SIP proxy server.

• SIP registrar address: IP address or host name of the registration server.

• SIP gateway address: IP address or host name of the SIP gateway.

Administration via WBM

System > Registration

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Administration via Local Phone

3.4.5.2 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 3.4.5.1, “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.

Administration via WBM

Network > Port configuration

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System Settings

Administration via Local Phone

3.4.6 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 num

ber of configuration parameters to allow this to happen. Registration can be authenticated or un-authenticated depending on how the server and phone is configured.

Unauthenticated Registration

For unauthenticated registration, the following parameters must be set on the phone: Terminal number or Terminal name (see Section 3.4.1.1, “Terminal Identity”), SIP server and SIP registrar address (see Section 3.4.5.1, “SIP Addresses”). Moreover, the correct Server type must be set. Additionally, the expiry time of a registration can be specified by Registration timer.

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 authen

ticates 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 user names 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.

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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 ". Default value: "OS Voic e".

• Realm: Protection domain for authentication.

• User ID: user name required for an authenticated registration.

• Password: Password required for an authenticated registration.

Administration via WBM

System > Registration

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Administration via Local Phone

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3.4.7 SIP Communication

3.4.7.1 Outbound Proxy

If this option 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: "No".

• Default OBP domain: Alternative value for the domain that is given in the outbound request.

Administration via WBM

System > SIP interface

Administration via Local Phone

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System Settings

3.4.7.2 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".

Administration via WBM

System > SIP interface

Administration via Local Phone

If TLS is to be used, please ensure that the SIP over TLS connection between phones and sever work properly.

|--- Administration |--- System |--- SIP Interface |--- SIP transport

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System Settings

3.4.8 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.

Data required

• Session timer enabled: Activates or deactivates the session timer mechanism. Value range: "Yes", "No". Default value: "No".

• Session duration (seconds): Sets the expiration time for a SIP session. Default: 3600.

Some server environments support their own mechanism for auditing the health of a session. In these cases, the Session timer must be deactivated.

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System Settings

Administration via WBM

System > Registration

Administration via Local Phone

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System Settings

3.4.9 Resilience and Survivability

To allow for a stable operation even in the presence of network or server failure, OpenStage phones have the capability to switch to a fallback system. The switchover is controlled by various configurable check and timeout intervals.

In survivability mode, it is expected that some features and functionality will not be available. The user will be informed by a message in the Call View display.

3.4.9.1 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.

Administration via WBM

System > SIP interface

If TLS is to be used, please ensure that the SIP over TLS connection between phones and sever work properly.

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System Settings

3.4.9.2 Response Timer

The 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.

Administration via WBM

System > SIP interface

Administration via Local Phone

|--- Administration |--- System |--- SIP Interface |--- Response timer (ms)

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System Settings

3.4.9.3 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 just tidies up internally.

The data is given in milliseconds. The default value is 32 000.

Administration via WBM

System > SIP interface

Administration via Local Phone

|--- Administration |--- System |--- SIP Interface |--- NonCall transactions (ms)

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System Settings

3.4.9.4 Maximum Registration Backoff Time

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.

System > SIP interface

Administration via Local Phone

|--- Administration |--- System |--- SIP Interface |--- Reg. backoff

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3.4.9.5 Backup SIP Server

The Backup registration flag indicates whether or not the phone treats the backup proxy serv- er 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.

The Backup registration timer determines the duration of a registration with the 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", "TLS". 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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System Settings

Administration via WBM

System > Registration

Network > Port configuration

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Administration via Local Phone

|--- Administration |--- Network |--- Port Configuration |--- Backup proxy

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Features - Configuration

3.5 Features - Configuration

3.5.1 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".

Administration via WBM

System > Features > Configuration

Administration via Local Phone

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Features - Configuration

3.5.2 Call Transfer

3.5.2.1 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) Configuration > Outgoing calls

Administration via Local Phone

|--- User |--- Configuration |--- Outgoing calls |--- Transfer on ring

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Features - Configuration

3.5.2.2 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 Transfer on hangup is disabled, C will be released when A hangs up, and A has the possibility to reconnect to B.

The default is "No".

Administration via WBM

System > Features > Configuration

Administration via Local Phone

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Features - Configuration

3.5.3 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 Asterisk, this setting is not typically necessary for enabling MWI functionality.

Administration via WBM

System > Features > Services

Administration via Local Phone

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Features - Configuration

3.5.4 Server Based Features

The use of server based DND (Do Not Disturb) and server based call forwarding is enabled or disabled here.

Administration via WBM

System > Features > Configuration

Administration via Local Phone

When phone based DND and phone based call forwarding are to be used, Server features must be switched off.

To enable server based features, uaCSTA must be allowed (see Section 3.5.5, “uaCSTA Interface”).

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3.5.5 uaCSTA Interface

User Agent CSTA (uaCSTA) is a limited subset of the CSTA protocol, which allows external CTI applications to interact with the phone.

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

Administration via Local Phone

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Features - Configuration

3.5.6 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 five minutes. The default value is 2.

Administration via WBM

System > Features > Configuration

Administration via Local Phone

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Features - Program Keys

3.6 Features - Program Keys

OpenStage 40/60/80 phones feature freely programmable keys (FPKs) which can be associated with special phone functions.

In the Administrator pages of the WBM, the program 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.

The functions available and their parameters are described in the following sub-sections.

3.6.1 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

3.6.2 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>.

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System > Features > Program keys

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3.6.3 Forwarding

If forwarding is enabled, the phone will forward incoming call to the predefined call number, depending on the current situation.

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 "All calls" is selected, any incoming call will be forwarded. If "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 Configuration > Incoming calls > Forwarding > No replay delay (seconds). If "On busy" is selected, incoming calls will be forwarded when the phone is busy.

Administration via WBM

System > Features > Program keys

3.6.4 Ringer off

Turns off the ringtone. 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>.

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System > Features > Program keys

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3.6.5 Hold

The call currently selected or active 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

3.6.6 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

3.6.7 Move blind

A call is transferred without consultation, as soon as the phone goes on-hook or the target phone goes off-hook.

The label displayed to the left of the key is defined in Key label <key number>.

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System > Features > Program keys

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3.6.8 Join

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

3.6.9 Deflect

On key press, an incoming call is deflected to the specified destination.

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

3.6.10 Shift

Shift the level for the program keys. When activated, the functions assigned to the shifted level are available on the program keys.

The label displayed to the left of the key is defined in Key label <key number>.

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System > Features > Program keys

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3.6.11 Conference

Establishes a three-party conference from an active call and a held call.

The label displayed to the left of the key is defined in Key label <key number>.

Administration via WBM

System > Features > Program keys

3.6.12 Headset

On key press, 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

3.6.13 Do Not Disturb

If this feature is activated, incoming calls will not be indicated to the user.

The label displayed to the left of the key is defined in Key label <key number>.

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System > Features > Program keys

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3.6.14 Repertory dial

Special calling functions. The desired function is selected via the Dial string parameter. The "<" character disconnects a call; "~" puts a call on hold for consultation; ">" starts a call; "-" enters a pause, e. g. for international dialing.

The label displayed to the left of the key is defined in Key label <key number>.

Administration via WBM

System > Features > Program keys

3.6.15 Consult and transfer

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>.

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System > Features > Program keys

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3.6.16 Call waiting toggle

Enables or disables the call waiting feature. If enabled, calls from a third party are allowed during an active call.

The label displayed to the left of the key is defined in Key label <key number>.

Administration via WBM

System > Features > Program keys

3.6.17 Server feature

Invokes a feature on the SIP server. The status of the feature can be monitored via the LED associated to the key.

The label displayed to the left of the key is defined in Key label <key number>.

The code entered in Feature code will be sent in a SIP INVITE to the SIP server.

The code entered in DTMF digits will be sent to the SIP server as a DTMF signal. As a precondition, an audio path routed via the SIP server must exist.

The SIP URI configured in LED control will receive the status notifications (SIP NOTIFY) from the SIP server. The LED is switched on or off accordingly.

Administration via WBM

System > Features > Program keys

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3.6.18 BLF

This function offers the possibility to monitor another phone, and to pick up calls for the monitored phone.

The label displayed to the left of the key is defined in Key label <key number>.

The Monitored phone parameter stores the call number (SIP URI) of the monitored phone.

If Audible alert is enabled, an alert beep or ringtone will indicate an incoming call for the monitored phone.

If Popup alert is enabled, a popup window will indicate an incoming call for the monitored phone.

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3.6.19 Key Modules

A Key module provides 12 additional program keys. It is available for the OpenStage 40, 60 and

80. A maximum of 2 key modules can be connected to one phone. The configuration of a key

on the key module is just the same as the configuration of a phone key.

Administration via WBM

System > Features > Key module 1/2

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Dialing

3.7 Dialing

3.7.1 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 phone book are automatically converted and stored in canonical format, thereby adding "+", Local country code, Local na- tional code, and Local enterprise number as prefixes. If, for instance, the user enters the ex- tension "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.

Data required

• Local country code: E.164 Country code, e.g. "49" for Germany, "44" for United Kingdom. Maximum length: 5.

• National prefix digit: Prefix for national connections, e.g. "0" in Germany and United Kingdom. Maximum length: 5.

• Local national code: Local area code or city code, e.g. "89" for Munich, "20" for London. Maximum length: 6.

• Minimal local number length: Minimum number of digits in a local PSTN number, e.g. 3335333 = 7 digits.

• Local enterprise number: Number of the company/PBX wherein the phone is residing. Maximum length: 10. (Optional)

• PSTN access code: Access code used for dialing out from a PBX to a PSTN. Maximum length: 10. (Optional)

• International access code: International prefix used to dial to another country, e.g. "00" in Germany and United Kingdom. Maximum length: 5.

• Operator codes: List of extension numbers for a connection to the operator. The numbers entered here are not converted to canonical format. Maximum length: 50. (Optional)

• Emergency number: List of emergency numbers to be used for the phone. If there are more than one numbers, they must be separated by commas. The numbers entered here are not converted to canonical format. Maximum length: 50. (Optional)

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

3.7.2, “Canonical Dial Lookup”).

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Tthese emergency numbers can also be dialed when the phone is locked, in line with the emergency number configured in Features > Configuration (see Section 3.4.2, “Emer- gency 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 the PBX, each number will start with 3, 4, or 5. Therefore, the digits to be entered are 3, 4, 5.

• Internal numbers

• "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 ex

tension 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.

To enable the phone to discern internal numbers from external numbers, it is crucial that a canonical lookup table is provided (

Section 3.7.2, “Canonical Dial

Lookup”).

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• "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:

• "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

Local functions > Locality > Canonical dial

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3.7.2 Canonical Dial Lookup

The parameters given here are important for establishing outgoing calls and for recognizing incoming calls.

In the local phonebook, and, mostly, in LDAP directories, numbers are stored in canonical format. In order to generate an appropriate dial string according to the settings in Internal num- bers and External numbers (-> Section 3.7.1), internal numbers must be discerned from ex- ternal numbers. The canonical lookup table provides patterns which allow for operation.

Furthermore, these patterns enable the phone to identify callers from different local or international telephone networks by looking up the caller’s number in the phone book. 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 phone book.

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 phone book entries. Example: "+4989722" for Siemens Munich.

Administration via WBM

Locality > Canonical dial lookup

To make sure that canonical dial lookup works properly, at least the following parameters of the phone must be provided:

• Local country code (-> Section 3.7.1)

• Local area code (-> Section 3.7.1)

• Local enterprise code (-> Section 3.7.1)

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Transferring Phone Software, Application and Media Files

3.8 Transferring Phone Software, Application and Media Files

New software images, hold music, picture clips for phonebook entries, LDAP templates, company logos, screensaver images, and ringtones can be uploaded to the phone via WBM (Web Based Management).

3.8.1 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.

3.8.2 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, FTP Server, FTP Server port, FTP account, FTP 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".

Data required

• Download method: Selects the protocol to be used. Value range: "FTP", "HTTPS". Default: "FTP".

• FTP Server address: IP address or hostname of the FTP server in use.

• 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.

For all user data, which includes files as well as phonebook content, the following amounts of storage place are available:

• OpenStage 20/40: 4 MB

• OpenStage 60/80: 8 MB

•

If Use defaults is activated for a specific file type, any specific settings for this file type are overridden by the defaults.

Siemens Asterisk OpenStage 20, Asterisk OpenStage 60, Asterisk OpenStage 80, Asterisk OpenStage 40 Owner's Manual

Specifications and Main Features

Frequently Asked Questions

User Manual