Do not operate the equipment in environments where there is a danger of explosions.
uebersicht.fm
Overview
Important Notes
Q
7
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.
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.
1.3Product Identification
1.4About 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 (ht-
(
tp://wiki.siemens-enterprise.com).
1.5Conventions 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 "/".
1The Handset lets you pick up and dial calls in the usual manner.
2The Display provides intuitive support for telephone operation.
3The user-friendly Application Keys provide easy access to your telephone’s
applications.
4Press the Function Keys to access frequently used telephony functions.
5With the Navigation Key, the user/administrator can navigate in the various
phone functions, applications, and configuration menus.
6The Keypad is used for entering phone numbers and text.
1.7Administration Interfaces
You can configure the OpenStage phone by using any of the following methods.
1.7.1Web-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.
To use this method, the phone must first obtain IP connectivity.
The Deployment Service (DLS) is a HiPath Management application for administering phones
and soft clients in both HiPath and non-HiPath networks. It has a Java-supported, web-based
user interface, which runs on an internet browser. For further information, please refer to the
Deployment Service Administration Guide.
1.7.3Local 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
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).
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:
ModelPower Consumption/Supply
OpenStage 20Power Class 1
OpenStage 20 GPower Class 2
OpenStage 40
OpenStage 40 + 2nd Key ModulePower Class 3
OpenStage 40 G
OpenStage 40 G + 2nd Key ModuleExternal power unit required
OpenStage 60/80
1
1
2
Power Class 2
Power Class 3
Power Class 3
OpenStage 60/80 + 2nd Key ModulePower Class 3
OpenStage 60/80 G
2
Power Class 3
OpenStage 60/80 G + 2nd Key ModuleExternal power unit required
Tabelle 2-1
1Includes 1 Key Module.
2Includes 1 Key Module + USB-Extension with Acoustic Unit.
2.Only if Power over Ethernet (PoE) is NOT supported:
Use only the plug-in power supply unit fitting the OpenStage phone:
This section describes a typical case: the setup of an OpenStage endpoint in an environment
using a DHCP server and the web interface. For different scenarios, cross-references to the
corresponding section of the administration chapter are given.
Alternatively, the DLS (Deployment Service) administration tool can be used. Its
>
>
Plug & Play functionality allows to provide the phone with configuration data by assigning an existing data profile to the phone’s MAC address or E.164 number. For
further information, see the Deployment Service Administration Manual.
Any settings made by a DHCP server are not configurable by other configuration
tools.
2.3.1Access the Web Interface (WBM)
1.Open your web browser (MS Internet Explorer or Firefox) 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.
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.
2.3.2Set the Terminal Number
If the user and administrator menus are needed in the course of setup, the terminal number,
which by default is identical with the phone number, must be configured first. The terminal num
ber input form is presented to the user/administrator right after booting, unless the Plug&Play
facility of the DLS is used. For further information about this setting, please refer to Section
3.5.1.1, “Terminal Identity”. With the WBM, the teminal number is configured as follows:
4.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.
For basic functionality, DHCP must provide the following parameters:
•IP Address: IP Address for the phone.
•Subnet Mask (option #1): Subnet mask of the phone.
•Default Route (option #3 "Router"): IP Address of the default gateway which is used for
connections beyond the subnet.
•DNS IP Addresses (option #6 "Domain Server"): IP Addresses of the primary and secondary DNS servers.
If no DHCP server is present, see Section 3.3.3, “IP Address - Manual Configuration” for IP address and subnet mask, and Section 3.3.4, “Default Route/Gateway” for default route.
2.3.4Date 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 hours in relationship to the UTC
time provided by the SNTP server.
For manual configuration of date and time see Section 3.5.4, “Date and Time”.
2.3.5SIP Server Address
The IP Address or hostname of a SIP server can be provided by DHCP.
The option’s name and code are as follows:
•option #120 "SIP Servers DHCP Option"
For manual configuration of the SIP server address see Section 3.5.5.1, “SIP Addresses”.
To have constant access to network subscribers of other domains, you can enter a total of two
more network destinations. For each further domain/subnet you wish to use, IP addresses for
the domain and gateway, and a subnet mask must be entered. 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”.
2.3.7Vendor specific: VLAN Discovery and DLS address
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 (DHCP)”).
If a DLS (Deployment Service) server is in use, its IP address must be provided. It is recommended to configure the DLS server address by DCHP, as this method enables full Plug & Play:
having received the DLS address from DHCP, the phone will contact the DLS during startup.
Provided that the DLS is configured appropriately, it will send all necessary configuration data
to the phone. Additionally, this method is relevant to security, as it ensures the authenticity of
the DLS server.
For manual configuration of the DLS server address see Section 3.3.7, “Configuration & Update
Service (DLS)”.
For the configuration of vendor-specific settings by DHCP, there are two alternative methods:
1) the use of a vendor class, or 2) the use of DHCP option 43.
It is recommended to define a vendor class on the DHCP server, thus enabling server and
phone to exchange vendor-specific data exclusively. The data is disclosed from other clients.
In the following, the configuration of vendor classes is explained both for a Windows DHCP
Server and for Unix/Linux.
Configuration of the Windows DHCP Server
For DHCP servers on a pre-SP2 Windows 2003 Server:
>
1.In the Windows Start menu, select Start > Programs > Administrative Tools > DHCP.
2.In the DHCP console menu, right-click the DHCP server in question and select Define Vendor Classes... in the context menu.
Windows 2003 Server contains a bug that prevents you from using the DHCP
console to create an option with the ID
stead, this entry must be created with the netsh tool in the command line (DOS
shell).
You can use the following command to set the required option (without error
message), so that it will appear in the DHCP console afterwards:
netsh dhcp server add optiondef 1 "Optipoint element 001"
STRING 0 vendor=OptiIpPhone comment="Tag 001 for Optipoint"
The value "Siemens" for optiPoint Element 1 can then be re-assigned using the
DHCP console.
This error was corrected in Windows 2003 Server SP2.
1 for a user-defined vendor class. In-
3.A dialog window opens with a list of the classes that are already available.
5.Enter "OptiIpPhone" as Display name and give a description of this class. Provide the
class name proper by setting the cursor underneath ASCII and typing "OptiIpPhone". The
binary value is displayed simultaneously.
Click OK to apply the changes. The new vendor class now appears in the list:
7.In the DHCP console menu, right-click the DHCP server in question and select Set Predefined Options from the context menu.
8.In the dialog, select the previously defined OptiIpPhone class and click on Add... to add
a new option. (If the workaround for a pre-SP2 Windows 2003 Server has been applied,
the first option will be there already.)
9.In the following dialog, specify the option type as follows. (If the workaround for a pre-SP2
Windows 2003 Server has been applied, the option type dialog will be skipped for the first
option.)
•Name: Free text, e. g. "OptiIpPhone element 01".
•Data type: "String".
•Code: "1".
•Description: Free text, e. g. "tag 1 for OptiIpPhone class".
Click OK to return to the previous window.
10. The newly created option is displayed now. Enter "Siemens" in the Value field.
11. If the VLAN is to be provided by DHCP: Repeat step 7 and 8, and then specify the option
type as follows. If you want to proceed to the configuration of the DLS address, continue
with step 13.
•Name: Free text, e. g. "OptiIpPhone element 02"
•Data type: "Long"
•Code: "2"
•Description: Free text, e. g. "tag 2 for OptiIpPhone class".
Click OK to return to the previous window.
12. The newly created option is displayed now. Enter the VLAN ID as a hexadecimal number
in the Value field. In the example, the VLAN ID is 10 (Hex: 2A).
If you do not intend to configure the DLS address, click OK and continue with step 15.
13. If the DLS address is to be provided by DHCP: Repeat step 7 and 8, and then specify the
option type as follows.
•Name: Free text, e. g. "OptiIpPhone element 03".
•Data type: "String".
•Code: "3".
•Description: Free text, e. g. "tag 3 for OptiIpPhone class".
Click OK to return to the previous window.
14. The newly created option is displayed now. Enter the DLS address in the Value field, using
the following format:
<PROTOCOL>:://<IP ADDRESS OF DLS SERVER>:<PORT NUMBER>
In the example, the DLS address is "sdlp://192.168.3.30:18443".
15. To define a scope, select the DHCP server in question, and then Scope, and right-click
Scope Options. Select Configure Options... in the context menu.
16. Select the Advanced tab. Under Vendor class, select the class that you previously de-
fined (OptiIpPhone) and, under User class, select Default User Class.
Activate the check boxes for the options that you want to assign to the scope (in the example, 001, 002, and 003). Click OK.
17. The DHCP console now shows the information that will be transmitted to the corresponding
workpoints. Information from the Standard vendor is transmitted to all clients, whereas information from the OptiIpPhone vendor is transmitted only to the clients (workpoints) in
this vendor class.
The following snippet from a DHCP configuration file (usually dhcpd.conf) shows how to set up
a configuration using a vendor class and the "vendor-encapsulated-options" option.
class "OptiIpPhone" {
option vendor-encapsulated-options
# The vendor encapsulated options consist of hexadecimal values for
the option number (for instance, 01), the length of the value (for instance, 07), and the value (for instance, 53:69:65:6D:65:6E:73). The
options can be written in separate lines; the last option must be followed by a ’;’ instead of a ’:’.
# Tag/Option #1: Vendor "Siemens"
#1 7 S i e m e n s
01:07:53:69:65:6D:65:6E:73:
# Tag/Option #2: VLAN ID
# 2 4 0 0 0 10
02:04:00:00:00:0A;
# Tag/Option #3: DLS IP Address (here: sdlp://192.168.3.30:18443)
# 3 25 s d l p : / / 1 9 2 . 1 6 8 . 3 . (...etc.)
03:19:73:64:6C:70:3A:2F:2F:31:39:32:2E:31:36:38:2E:33:2E:33:30:
3A:31:38:34:34:33;
match if substring (option vendor-class-identifier, 0, 11) =
"OptiIpPhone";
}
2.3.7.2Using Option #43 "Vendor Specific"
Alternatively, option #43 can be used for setting up the VLAN ID and DLS address. The following tags are used:
•Tag 1: Vendor name
•Tag 2: VLAN ID
•Tag 3: DLS address
Optionally, the DLS address can be given in an alternative way:
•Tag 4: DLS hostname
The Vendor name tag is coded as follows (the first line indicates the ASCII values, the second
line contains the hexadecimal values):
The following example shows a VLAN ID with the decimal value "10". Providing
:
CodeLengthVLAN ID
240010
02040000000A
Table 2-3
For manual configuration of the VLAN ID see Section 3.2.2.2, “Manual configuration of a VLAN
ID”.
The DLS IP address tag consists of the protocol prefix "sdlp://", the IP address of the DLS server, and the DLS port number, which is "18443" by default. The following example illustrates the
syntax:
4.If the VLAN ID is to be provided by DHCP: Enter the hexadecimal value in Data entry. Pro-
viding the length is not required here, as the VLAN ID is always 4 Bytes long. In the example, the VLAN ID is 10 (Hex: 2A).
5.If the DLS address is to be provided by DHCP: Enter the DLS address in the Value field,
using the following format:
<PROTOCOL>:://<IP ADDRESS OF DLS SERVER>:<PORT NUMBER>
For ensuring proper functionality, the port number should not be followed by any
>
character.
In the example, the DLS address is "sdlp://192.168.3.30:18443".
Note that the screenshot also shows the VLAN ID described in step 4.
For registration at the HiPath 8000 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 3.5.6, “Authenticated Registration”):
1.In the administration menu, select System > Registration. The "Registration" dialog opens.
2.In the Server type field, enter "HiQ8000".
3.In Realm, enter the SIP realm the targeted user/password combination refers to.
4.In the User ID and Password fields, enter the user name/password combination for the
phone.
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)”.
3.1Access via Local Phone
The data entered in input fields is parsed and controlled by the phone. Thus, data is
>
1.Access the Administration Menu
accepted only if it complies to the value range.
OpenStage 60/80:
Press the v key to activate the administration menu (the v key toggles between the user’s
configuration menu and the administration menu).
OpenStage 60/80 V1R3.x upwards:
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 3.14, “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.
Use the 3-way Navigator to navigate and execute administrative actions in the administration menu.
Press the mkey briefly:
- scroll up
Hold down:
- scroll to top of list
Press the
- confirm entries
- perform an action
i key:
Press the
- scroll down
Hold down:
- scroll to end of list
l key briefly:
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:
KeyFunction
*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.
If a parameter is set by entering a number or character data, the onscreen keypad is used.
Press the i key to enter the editor. Within the editor, solely use the key numbers or the
Sensor Wheel for selecting numbers, characters, or groups of characters. The h key deletes one character in the input field, and the g key moves the cursor to the OK field.
The following figure describes the elements of the onscreen keypad and their functions:
Element with focus
Letters, digits, punctuation marks or special characters
Command line
Copy contents of active field to clipboard
Insert clipboard contents at cursor position
Move cursor left/right
Shift to punctuation and special characters
Shift to numeric entry
Shift to upper/lower case
Additionally, you can use the following keys on the keypad as shortcuts for the selection of
character groups
ElementFunction
*
#
Switch to punctuation and special characters.
Toggle between lowercase characters, uppercase characters, and digits.
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.
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 60/80 G) and duplex method (full or half duplex) with whatever equipment is
connected. Optionally, the required transfer rate and duplex mode can be specified manually
using the LAN port speed parameter.
In the default configuration, the LAN port supports automatic detection of cable con-
>
figuration (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.
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".
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.
The implementation of a voice network based on VLANs requires the network infra-
>
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.
structure (the switch fabric) to support VLANs.
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 an Ethernet packet is belonging to 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.
The VLAN ID can be configured automatically by DHCP or manually.
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
Administration via Local Phone
|--- Administration
|--- Network
|--- IP Configuration
|--- VLAN discovery
To configure layer 2 VLAN and QoS manually, first make shure that QoS layer 2 and 3 are configured as described in Section 3.3.1, “Quality of Service (QoS)”, and VLAN discovery is set to
"Manual" (see Section 3.2.2.1, “Automatic VLAN discovery (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
Administration via Local Phone
|--- Administration
|--- Network
|--- IP Configuration
|--- VLAN ID
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.
3.3.1.1Layer 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.
PREAM.SFDDASA
TAG
4 Bytes
Three Bits Used for CoS
PTDATAFCS
(User Priority)
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.
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).
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", "CST".
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", "CST").
Default: "AF31".
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 change will only have effect if you restart the phone.
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The following parameters can be obtained by DHCP:
Basic informations
•IP Address
•Subnet Mask
Optional informations
•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)
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.
The change will only have effect if you restart the phone.
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Administration via WBM
Network > IP configuration
Administration via Local Phone
|--- Administration
|--- Network
|--- IP Configuration
|--- Route (default)
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.
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.
3.3.6.1DNS 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
The Deployment Service (DLS) is a HiPath Management application for administering workpoints in both HiPath and non-HiPath networks. Amongst the most important features are: security (e.g. PSS generation and distribution within an SRTP security domain), mobility for optiPoint and OpenStage SIP phones, software deployment, plug&play support, as well as error
and activity logging.
DLS address, i.e. the IP address or hostname of the DLS server, and DLS port, i.e. the port
on which the DLS server is listening, are required to enable proper communication between
phone and DLS. The Contact gap parameter controls a security function. It specifies a minimum time interval that must elapse between HTTP requests; any requests coming within that
time will be ignored. The purpose is to prevent DoS (Denial of Service) attacks on the phone.
The Security mode determines whether the communication between the phone and the DLS
is secure. A secure connection is established by exchanging credentials between the DLS and
the phone for mutual authentication. After this, the communication is encrypted, and a different
port is used.
Data required
•DLS address: IP address or hostname of the server on which the Deployment Service is
running.
•DLS port: Port on which the DLS Deployment Service is listening.
Default: 18443.
•Contact gap: Minimum time interval in seconds that must elapse between HTTP requests,
in order to prevent DoS attacks.
Default: 300.
•Security mode / Security status: Determines whether the communication between the
phone and the DLS is secure.
Value range: "Default mode", "Secure mode".
Default: "Default".
|--- Administration
|--- Network
|--- Update Service (DLS)
|--- DLS address
|--- DLS port
|--- Contact gap
|--- Security status
3.3.8SNMP
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.
The OpenStage phone supports SNMP version 1 and 2.
There are currently 4 categories of trap 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 stan
dard Network Management System) to highlight important telephony related problems.
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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.
•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: Enables or disables queries from 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".
•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".
With software version V1R4.x or higher, secure speech transmission via SRTP is possible.
If Use secure calls is activated, the encryption of outgoing calls is enabled, and the phone is
capable of receiving encrypted calls. An icon in the call view tells the user whether a call is secure or not. If an active call changes from secure to insecure, e. g. after a transfer, a popup window and an alert tone will notify the user. For enabling secure calls, a TLS connection to the
HiPath 8000 is required.
For secure calls, it is required that both endpoints support SRTP. The secure call in-
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dication tells the user that the other endpoint has acknowledged the secure connection.
In order to use SRTP, the phone must be configured for NTP (for further information
please see Section 3.5.4, “Date and Time”). The reason is that the key generation
(MIKEY) uses the system time of the particular device as a basis. Thus, encryption
will only work correctly if all devices have the same UTC time.
If SIP server certificate validation resp. Backup SIP server certificate validation is activat-
ed, the phone will validate the server certificate sent by the HiPath 8000 in order to establish a
TLS connection. The server certificate is validated against the root certificate from the trusted
certificate authority (CA), which must be stored on the phone first. For delivering the root certificate, a DLS (Deployment Software) server is required.
Administration via WBM
System > Security
Administration via Local Phone
|--- Administration
|--- System
|--- Security
|--- Server cerfificate
|--- Backup certificate
|--- Use secure calls
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
|--- Administration
|--- System
|--- Identity
|--- Display identity
|--- Enable ID
After the phone has beeen incative within a specified timespan, the display backlight is
switched off. The length of the timespan ranges from 2 hours to 8 hours.
Administration via WBM
Local functions > Pixel saver
Administration via Local Phone
|--- Administration
|--- Local Functions
|--- Pixel saver
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.
3.5.4.1SNTP 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".
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 (Hipath
8000). 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
|--- Network
|--- Port Configuration
|--- SIP server
|--- SIP registrar
|--- SIP gateway
|--- SIP local
3.5.6SIP 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.
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Unauthenticated Registration
For unauthenticated registration, the following parameters must be set on the phone: Terminal
number or Terminal name (see Section 3.5.1.1, “Terminal Identity”), SIP server and SIP registrar address (see Section 3.5.5.1, “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 usernames and passwords.
A challenge from the server for authentication information is not only restricted to the
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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 pre-
The Response timer is started whenever the phone sends a new 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 there
is no backup server configured, the phone just tidies up internally.
The data is given in milliseconds. The default value is 32 000; for the HiPath 8000, the recommended setting is 3.7 seconds (3700 ms).
A regular check ensures that the TLS link 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 searches for another links.
If this option set to "Yes", the phone routes outbond requests to the configured proxy, i. e. the
SIP server/registrar. 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.
Session timers provide a basic keep-alive mechanism between 2 user agents or phones. This
mechanism can be useful to the endpoints concerned or for stateful proxies to determine that
a session is still alive. This is achieved by the phone sending periodic re-INVITEs to keep the
session alive. If no re-INVITE is received before the interval passes, the session is considered
terminated. Both phones are supposed to terminate the call, and stateful proxies can remove
any state for the call.
This feature is sufficiently backward compatible such that only one end of a call needs to implement the SIP extension for it to work.
The parameter Session timer enabled determines whether the mechanism shall be used, and
Session duration (seconds) sets the expiration time, and thus the interval between refresh
re-INVITEs.
Some server environments support their own mechanism for auditing the health of
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a session (e.
g. Broadsoft). In these cases, the Session timer must be deactivated.
Data required
•Session timer enabled: Activates or deactivates the session timer mechanism.
Value range: "Yes", "No".
Default value: "No".
•Session duration (seconds): Sets the expiration time for a SIP session.
Default: 3600.
The survivability feature will allow the SIP User Agent to register with a backup SIP proxy which
will be used to make and receive calls when the primary SIP proxy fails or is not reachable due
to a network failure.
The prime reason for this feature is to maintain basic call functionality when network failures
occur, and it is therefore expected that some features and functionality will not be available
when working in survivability mode.
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 mes-
sages 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 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.
This feature allows a user to collect a call from any ringing phone that is in the same pickup
group. To be a member of a Call Pickup group, the phone must be configured with the corresponding URI of the Call Pickup group service provided by the server. This URI has the following form: <groupcallpickup>@<SIP Server IP> (for instance **3@172.16.127.95 or Domain
Name).
Administration via WBM
System > Features > Addressing
Administration via Local Phone
|--- Administration
|--- System
|--- Features
|--- Addressing
|--- Group pickup URI
If desired, an incoming call for the pickup group can be indicated acoustically.
The Group pickup tone allowed parameter activates or deactivates the generation of an
acoustic signal for incoming pickup group calls. The default is "Yes". If this is activated, Group pickup as ringer determines whether the current ringtone or an alert beep is used. If set to
"Yes", a pickup group call will be signaled by a short standard ringtone. If set to "No", a pickup
group call will be signaled by an alert tone. The default is "Yes".
Group pickup visual alert defines the user action required to accept a pickup call. If "Prompt"
is selected, an incoming pickup call is signaled by an alert on the phone GUI. As soon as the
user goes off-hook or presses the speaker key, the pickup call is accepted. Alternatively, the
user can press the corresponding function key, if configured. If "Notify" is selected, an incoming
pickup call is signaled by an alert on the phone GUI. To accept the call, the user must confirm
the alert or press the corresponding function key, if configured.
|--- Administration
|--- System
|--- Features
|--- Configuration
|--- Audio
|--- Group pickup tone allowed
|--- Group pickup as ringer
|--- Group pickup visual alert
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
3.6.3.2Transfer 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 Callback option allows the user to request a callback on certain conditions. The callback
request is sent to the SIP server. The Code for callback busy requests a callback if the line is
busy, i. e. if there is a conversation on the remote phone. Code for callback no reply applies
when the call is not answered, i. e. if nobody lifts the handset or accepts the call in another way.
The Code for callback cancel all all deletes all the callback requests stored previously on the
telephone system/SIP server.
Data required
•Code for callback busy / Callback: Busy: Access code that is sent to the server if the
line is busy.
•Code for callback no reply / Callback: No reply: Access code that is sent to the server
if the callee does not reply.
•Code for callback cancel all / Callback: Cancel all: Access code for canceling all call-
back requests on the server.
Administration via WBM
System > Features > Services
Administration via Local Phone
|--- Administration
|--- System
|--- Features
|--- Addressing
|--- Callback: Busy
|--- Callback: No reply
|--- Callback: Cancel all
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 HiPath 8000, this setting is not typically necessary for enabling MWI functionality.
Administration via WBM
System > Features > Services
Administration via Local Phone
|--- Administration
|--- System
|--- Features
|--- Addressing
|--- MWI server URI
The Conference URI provides the number/URI used for system based conferences, which can
involve more than three members. This feature is not available with every system.
It is recommended not to enter the full URI, but only the user part. For instance, enter
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Administration via WBM
System > Features > Services
"123", not "123@<SIP SERVER ADDRESS>". A full address in this place might cause a conflict when the HiPath 8000 uses multiple nodes.
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.