Siemens HiPath optiPoint 410 S, HiPath optiPoint 420 S User Manual
HiPath
optiPoint 410 S / 420 S
Using with Asterisk
Configuration Guide
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1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2 System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3 Configuration of Asterisk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1 SIP Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3.2 Dial Plan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
4 Basic Configuration of optiPoint Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
4.1 Web Interface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
4.2 Network IP Address and Routing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
4.3 Quality of Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.4 SIP Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
5 Multi Line . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
6 Voicemail . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
6.1 Settings in voicemail.conf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
6.2 Settings in extensions.conf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
6.3 Settings in the file sip.conf . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
6.4 Setting up the Message Key on the optiPoint Telephone . . . . . . . . . . . . . . . . . . . . . . . 21
7 Call Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
7.1 Group Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
7.2 Execution of Call Pickup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
8 Music on Hold (MOH). . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
9 Reference List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
10 Sample Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
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Overview
1Overview
Using the products from the Siemens HiPath portfolio you can design communication solutions
for a wide variety of requirements (i.e. scalability and features). Apart from IP communication
systems, optiPoint telephones with SIP protocol are an integral part of these solutions. These
telephones can - due to the standards used - easily be integrated into open SIP-based environments.
This documentation helps you to start up a Siemens optiPoint SIP telephone with only a few
steps. It also shows the preparation of the SIP system, using an Asterisk open source PBX environment as an example.
2 System Requirements
Asterisk runs on the following operating systems: Linux, BSD, Mac OS X.
As long as you use Asterisk for VoIP only, no specific hardware requirements have to be met.
If a larger number of codec transcodings is necessary, you should use a more powerful PC
hardware.
In case of compatibility questions regarding protocols or hardware please refer to
http://www.asterisk.org (see Section 9, "Reference List").
3 Configuration of Asterisk
We assume that you are familiar with the installation and basic usage of an Asterisk environment. If you need further help, there is a wide variety of literature available as well as comprehensive information in the world wide web (see Section 9, "Reference List").
We will explain the basic statements in the configuration files sip.conf, extensions.conf and
voicemail.conf using an running sample configuration for two optiPoint telephones.
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Configuration of Asterisk
SIP Registration
3.1 SIP Registration
The file sip.conf has to have an entry for each user agent, i.e. for each telephone.
3.1.1 Syntax
The syntax of the configuration file must correspond to the following pattern:
[general]
parameter1=value
parameter2=value
[User Agent 1]
parameter1=value
parameter2=value
3.1.2 Sample configuration
The global section [general] contains statements that are valid for all user agents.
Example:
[general]
context=default
realm=usergruppe@asterisk
bindport=5060
bindaddr=0.0.0.0
allow=alaw
qualify=yes
dtmfmode=info
Explanations:
• context
The context groups certain extensions that are defined in extensions.conf.
• realm
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Configuration of Asterisk
SIP Registration
Name of the realm. Within a realm certain combinations of user & password are valid.
During registration the Asterisk server assigns the realm name to the telephone. The presetting is asterisk.
• bindport
UDP port the asterisk server uses for receiving data. Pressetting: 5060.
• bindaddr
IP address used to receive incoming SIP connections. Using the pressetting 0.0.0.0 Asterisk receives SIP connections on all network interfaces and aliases.
• allow
Permitted codecs. Only the codecs listed here are permitted. The order of the allow entries determines the preference of the codecs.
• qualify
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If the value yes is used, you can monitor latency and availability of the end point. As per
presetting, an end point is considered to be available if the latency is below 2,000 ms. Instead of yes/no you can also specify a maximum latency in milliseconds.
• dtmfmode
Using DTMF (Dual Tone Multiple Frequency) tone signals to control the connection are
sent via the keypad. These are used in menu-driven info services (Attendant Control), e.g.
the voicemail features of Asterisk. There are 3 options to transmit the signals1) inband: the
tone signal is faded into the voice signal; 2) outband: another data stream is started for thetransmission of the tone signal; 3) instead of a tone signal an INFO field is sent. optiPoint
telephones support both inband as well as INFO fields. We recommend to use INFO fields
as in case of using inband and a high degree of compression there might be errors in the
data interpretation. Set dtmfmode to info.
The following section explains the entries for the individual user agents.
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Configuration of Asterisk
SIP Registration
Example First User/Telephone:
[10]
type=friend
context=myphones
username=10
secret=123456
host=dynamic
Example Second User/Telephone:
[11]
type=friend
context=myphones
username=11
secret=123456
host=dynamic
Explanations:
• type
Three types are available: peer, user and friend. For telephones (i.e. user agents) you
have to enter friend.
• context
The section in extensions.conf that is valid for the particular user, has to be labelled with
the value used here; in the example: [myphones].
• username
This parameter is only necessary if the user name contacted by Asterisk in case of an incoming call is different from the registered user name or if the user is not registered. The
user is registered under the name or the number heading the section.
• secret
This is the password the user needs to register at the SIP server. If you do not want the
save the password in plain text, you can use md5secret instead of secret and use the
MD5 hash of a string consisting of the user name, realm and password for the value. In this
case, the following steps have to be completed:
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Configuration of Asterisk
1. To generate the MD5 hash for the password, enter the following command into the
Linux command line:
echo -n "username:realm:secret" | md5sum
(in the example: echo -n "10:usergruppe@asterix:123456" | md5sum)
2. Enter the MD5 hash. In the example:
md5secret=9ef347178a279fffea649ac16c6b7510
• host
Possible values are dynamic, an IP address or a host name. If the telephone is to register
itself, enter dynamic.
3.2 Dial Plan
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Dial Plan
Syntax:
exten => name, priority, application()
Example:
[myphones]
exten => 10,1,Dial(SIP/10,5,r)
exten => 11,1,Dial(SIP/11,5,r)
Explanations (for lines 1 and 2 in the example):
• [myphones]
Context grouping several extensions. Other sections refer to this context, e.g. in the file
sip.conf.
• exten =>
This indicates that one step of an extension is defined. Extensions define how Asterisk is
to act in case of an incoming call or if a certain number is entered in a channel.
• 10
Extension number. If this number is called, the actions defined in the corresponding extension are executed.
• 1
Priority of this step of the extension. An extension may consist of several steps executed
in the order of their priorities.
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Basic Configuration of optiPoint Telephones
Web Interface
• Dial(SIP/10,5,r)
The application defined in this step is Dial(). In the example, the system tries to call the
SIP participant number 10 for a duration of 5 seconds. The option r specifies that the caller
hears a ring tone while the telephone called is ringing.
4 Basic Configuration of optiPoint Telephones
Currently, the portfolio of the SIP-based optiPoint devices 410 S and 420 S comprises 9 models. To a large extent, the instructions and steps in the following section are independent of the
model. For additional information about the individual optiPoint models, please refer to the corresponding data sheets, user guides and administrator guides, see Section 9, "Reference
List").
There are 4 options to configure an optiPoint telephone:
• Web Interface: The telephone can be configured and controlled via a standard web browser (recommended).
• Database-driven configuration tool DLS (Deployment Service): This Siemens IP telephone
manager (available at no charge) is a powerful tool for configuring larger telephone systems.
• optiGuide: User interface of the telephone. For additional information, please refer to the
corresponding User Guide or Administration Guide, see Section 9, "Reference List".
• XML interface of the telephone. For more information, please refer to the XML Administrator Guide, see Section 9, "Reference List".
The following section describes the configuration via the web interface as this is the most convenient and viable method for most installations.
4.1 Web Interface
Open your browser and enter "https://" followed by the IP address of the optiPoint telephone,
e.g.:
"https://192.168.1.5".
Prior to loading the page the browser displays a warning indicating that the certifi-
>
The website of the telephone is opened. If it is already configured, the terminal number (i.e. the
phone number) of the device is displayed in the upper right corner.
8 optiPoint 410/420 - Using with Asterisk, Configuration Guide
cate sent by the optiPoint device is not verified or installed. In this context, you can
ignore this warning.
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Basic Configuration of optiPoint Telephones
Web Interface
Select the Administration section and enter the administrator password. The initial password
is "123456". You should change the password after the first login.
The user of the telephone can also modify device settings. To do so, he has to log in
>
The Administrator Menu is displayed. In the following screenshot all sub-menus that are or may
be relevant for operation with Asterisk are highlighted.
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into the User section.
9
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