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
1
Overview
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.
•
4
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
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
Dial Plan
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
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 certificate sent by the optiPoint device is not verified or installed. In this context, you can
ignore this warning.
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.
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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
into the User section.
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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Basic Configuration of optiPoint Telephones
Web Interface
General Information
This section contains general information about the telephone, e.g. software release or MAC
address.
Network IP and routing
In this section the IP settings are entered. If DHCP is supported within the network, no changes
should be necessary here. During the first setup of the optiPoint telephone without DHCP these
settings have to be entered via the local Administrator Menu in the telephone (optiGuide) as
follows:
1.
Press the right arrow key and enter the numbers 1,3 in idle mode.
2.
Enter the administrator password.
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Basic Configuration of optiPoint Telephones
Web Interface
3.
Select the menu options using the arrow keys. The following options are available: DHCP
IP assign (activate/deactivate DHCP usage), Terminal IP address (IP address of the optiPoint), Terminal mask and Default route (IP address of the router).
SIP environment
This section is used to enter the SIP-related settings.
Quality of Service
The QoS (Quality of Service) settings have to be synchronized with the network.
Lan Port setting
You can specify the transfer rates of the two LAN ports here.
For other settings, please refer to the Administrator Guide, see Section 9, "Reference List".
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Basic Configuration of optiPoint Telephones
Network IP Address and Routing
4.2
Network IP Address and Routing
Use this section to configura network settings such as DHCP usage, IP address of the telephone, terminal mask, DNS server and default route.
For additional information about the configuration options, please refer to the Administrator
Guide, see Section 9, "Reference List".
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Basic Configuration of optiPoint Telephones
Network IP Address and Routing
DHCP
If DHCP is to be used, mark the checkbox DHCP (default setting).
Terminal hostname
The contents of the field Terminal hostname is sent to the DNS server. Thus, the web interface
is also available for users know knowing the IP address of the telephone.
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Basic Configuration of optiPoint Telephones
Quality of Service
4.3
Quality of Service
The QoS settings ensure packet-optimized transmissions within the network.
Usually, QoS is only supported in large networks. Please deaktivate Layer 2 und Layer 3 in
low-end network that do not support QoS / Vlan.
>
Older network components may not be able to read packets with Vlan tag. In these
cases, Layer 2 und Layer 3 have to be set to "Off" using the user interface of the
telephone.
The QoS settings have to be synchronized with the network administrator.
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Basic Configuration of optiPoint Telephones
SIP Environment
4.4
SIP Environment
Use this dialog to enter basic SIP parameters.
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Basic Configuration of optiPoint Telephones
SIP Environment
Phone number
Phone number of the user agent or telephone. This number must refer to the corresponding
user data section in the file sip.conf (see Section 3.1, "SIP Registration"). Alternatively, you can
enter a name under Phone Name. To ensure that the telephone is registered with this name,
mark the checkbox Register by name.
Registrar IP address or DNS name / Server IP address or DNS name
Here you have to enter one of the IP addresses or hostnames that entered as bindaddr in the
section [general] of the file sip.conf.
SIP Port
Specifies the destination port of the server, i.e. the port the server uses to receive the telephone’s SIP packets. The default value is 5060.
SIP server type
Enter the type of the SIP server. If you use Asterisk, enter "Other".
SIP realm
If the parameter realm is set in the section [general] of the file sip.conf, Asterisk transmits
this parameter to the telephone during the first registration. Therefore, no value has to be entered here.
SIP user ID
This value complies with the term in the corresponding user-specific section in the file sip.conf
(see Section 3.1, "SIP Registration").
New SIP password / Confirm SIP password
The SIP password has to comply with the one set using the parameter secret in the corresponding user-specific section (client definition) in the file der sip.conf.
>
Both optiPoint telephones as well as Asterisk support the MD5 password encryption.
The password can also be saved on the Asterisk server as a MD5 hash (see Section
3.1.2, "secret").
Message Waiting IP address or DNS name
IP address or hostname of the server sending the MWI notifications (Message Waiting Indicator). The presetting 0.0.0.0 should work in most cases.
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Multi Line
SIP Environment
Voicemail number
Specifies the number to reach the voicemail feature of Asterisk. This complies with the extension specified for the feature VoiceMailMain() in the file extensions.conf.
5
Multi Line
During setup, a phone number - or alternatively: a name - is assigned to the telephone; this
number or name can be used to call this telephone and is used at the called party’s side to identify the caller.
Very often this is referred to as a (telephone) "line" the telephone is assigned to - in analogy to
conventional (TDM) telephones using phone numbers.
In addition, powerful telephones support the so-called "Multi Line" feature. This feature enables
one end device to have access to several lines. Apart from that, one line can be assigned to
several end devices. If a call comes in on such a shared line, this triggers a ringing tone on all
end devices assigned to that line. Depending on the model, optiPoint 410 S / 420 S can attend
to up to 10 lines.
If you want to use the Multi Line feature with Asterisk, the parameter pedantic=yes has to be
set in the file sip.conf; this ensures that Asterisk can identify the Multi Line Header in the SIP
messages (presetting is no).
The lines are not configured in the SIP environment section but using the key programming of
the optiPoint telephone. For each line one function key has to be configured as "line" key. User
data such as E.164 number, user name and password have to be entered. The LED belonging
to the line key indicates the status of that line. For more information about functionality and setup, please refer to the User Guide and the Administrator Guide (see Section 9, "Reference
List").
For each line on the end device there has to be a corresponding entry in the file sip.conf. The
server settings have to be entered in the section SIP environment.
>
As Asterisk Version 1.2.7 has some problems with forking, Multi Line with several
user agents (shared line) are not yet possible. So far, only one user agent can be
connected per line. However, Digium is already working on improving forking support.
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Voicemail
Settings in voicemail.conf
6
Voicemail
Voicemail can be considered an answering machine or voice-based info service integrated into
Asterisk. There are two ways to notify the user of a new voice message: he receives an email
with the voice message as an attachment, and the VoiceMail Message LED on his optiPoint
starts blinking. The following section briefly describes the voicemail configuration of Asterisk.
6.1
Settings in voicemail.conf
The global section [general] contains - amongst other values - the audio formats for recording voice messages.
Example:
[general]
format=wav49|gsm|wav
Explanation:
•
format
The parameter can contain a number of audio formats (of course you can also select only
one file format). The voice messages are recorded in all formats listed.
To give the user access to his mailbox, his access data have to be entered into the file voicemail.conf.
Example:
[myphones]
10 => 123456,10,user1@mail.com
11 => 123456,11,user2@mail.com
Erläuterungen:
•
10
Number of the mailbox. In the example, this number is the same as the phone number of
the user.
•
123456
The user password for access to the mailbox.
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Voicemail
Settings in extensions.conf
●
10
SIP user name or phone number.
•
user1@mail.com
Email address of the user. If the Asterisk server is set up accordingly, the voice message
can be sent to the user as an email attachment.
6.2
Settings in extensions.conf
In the user-specific section of the dial plan, an overflow option for voicemail has to be set up.
You can differentiate between two available options: a) the called party is not available, and
b) the called party’s line is busy. The following example shows the configuraton for the user with
phone number 10; the lines have to be entered into the corresponding section of the file extension.conf.
Example:
exten => 10,2,VoiceMail(u10@myphones)
exten => 10,102,VoiceMail(b10@myphones)
Explanations:
•
10
SIP user name or phone number.
•
2 / 102
Priority for forwarding to the mailbox. The value 2 in the example determines that - if the
called party is unavailable - the caller is forwarded after the attempted call (priority 1 in the
example) is forwarded the the mailbox 10@myphones and hears a corresponding message. If the line is busy, Asterisk jumps to a priority number that is 101 higher than the priority of the attempted call, in the example: 1 + 101 = 102.
•
u10@myphones / b10@myphones
Further specification of the mailbox. The prefix u makes sure that the message designated
for unavailability is played. If you use the prefix b, the message for "line busy" is played. 10
is the numbe of the mailbox. @myphones specifies the context for this entry. This is necessary if the user entry in the file voicemail.conf was entered in a specific context instead of
the context [default].
Weiterhin muss für den jeweiligen Kontext (im Beispiel: myphones) die zu wählende Nummer
der Voicemail-Funktion eingetragen werden, damit der Benutzer seine Sprachnachrichten abhören kann. Auch dies geschieht in Form einer Extension, also eines Eintrags im Wählplan.
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Voicemail
Settings in the file sip.conf
Example:
exten => 99,1,VoiceMailMain()
Explanations:
•
99
Phone number for the mailbox.
•
1
Priority of the VoiceMailMain feature in the dial plan.
•
VoiceMailMain()
The user is forwarded to the Voicemail main menu.
6.3
Settings in the file sip.conf
As soon as a new voice message is available, a MWI signal (Message Waiting Indicator) is sent
to the telephone which displays a corresponding message. On the optiPoint 410 S / 420 S you
can easily retrieve messages using a previously configured key. A blinking LED indicates that
new messages are available (see Section 6.4, "Setting up the Message Key on the optiPoint
Telephone"). In the section [general] the MIME type has to be set as follows in order to ensure the MWI signal is transmitted correctly to the telephone:
notifymimetype=text/plain
Finally, the mailbox to be used has to be specified in the user-specific section of the file sip.conf.
Example:
mailbox=10@myphones
Explanation:
The number of the mailbox together with the context is added to the key word mailbox.
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Voicemail
Setting up the Message Key on the optiPoint Telephone
6.4
Setting up the Message Key on the optiPoint Telephone
In the web interface, go to Administration > Function keys... Phone to get to the function keys
assignment feature.
Click the Edit button of the key you want to assign the message function to.
On the right side of the web interface another input mask is displayed. Select "Voice Message"
from the pull-down menu Select a function. Using the option Locked by administrator you
can define whether this key is to be locked. In case of an optiPoint 420 S telephone you can
enter the key labeling into the field Key Label. Once you have entered all the data you are
asked to save them. As soon as you have saved the data, the new function key is available.
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Call Pickup
Group Definition
7
Call Pickup
Asterisk provices the possibility of creating Pickup Groups. Calls to a member of such a group
can be picked up by all group members.
7.1
Group Definition
Groups are defined in the file sip.conf. Using callgroup=X within the section for user A, this
user A is assigned to a group called X. This group pools all call targets that can be picked up
by a pre-defined pickup group. To make sure a user B can pick up a call for user A, you have to
enter pickupgroup=X into the section for user B. X can be set to values from 0 through 63.
Example:
[10]
callgroup=1
pickupgroup=1
Explanations:
•
[10]
Indicates the user for which the following parameters are valid.
•
callgroup
Using this parameter the user is assigned to a group within which calls can be picked up
by members of a pickup group with the same number (in the example: 1).
•
pickupgroup
Using this parameter the user is assigned to a pickup group; the members of this pickup
group can pick up calls to any telephone within the group of call targets that have the same
number (in the example: 1).
7.2
Execution of Call Pickup
To execute a call pickup, i.e. to pick up a call to the group, the user has to select the sequence
"*8".
The Pickup() command has to be defined in the file extension.conf.
Example:
exten => _*8.,1,Pickup(10&11)
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Call Pickup
Execution of Call Pickup
Explanations:
•
_*8.
Notation for the dial sequence "*8".
•
1
Priority of this step in the extension.
•
Pickup(10&11)
If a call arrives for one of the telephones specified in the parameters, this call is picked up
as soon as the sequence detailed above is dialed. The telephones listed are checked for
their status in their order of appearance in the parameter list.
>
The optiPoint 410 S / optiPoint 420 S telephones feature a key function Call Pickup
signalling a call within a pickup group via an LED. However, in an Asterisk environment this feature is inoperable as Asterisk does not send the necessary SIP notification.
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Music on Hold (MOH)
Execution of Call Pickup
8
Music on Hold (MOH)
The RFC 3261 offers several options for indicating " On Hold". In case of an optiPoint telephone, a method "INACTIVE" is added to the session description if the other party is put on
hold. Asterisk, however, only recognizes "SENDONLY".
There is a workaround to make sure that Asterisk plays the MOH (Music on Hold) in spite of
that. You have to modify the file chan_sip.c which by default is stored in the folder /usr/src/asterisk/channels.
In line 3552 the RTP MAP is assigned.
This is the original source code:
sdpLineNum_iterator_init(&iterator);
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough
space */
if (!strcasecmp(a, "sendonly")) {
sendonly=1;
continue;
}
if (!strcasecmp(a, "sendrecv")) {
sendonly=0;
}
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
if (debug)
ast_verbose("Found description format %s\n",
mimeSubtype);
/* Note: should really look at the 'freq' and '#chans' params
too */
ast_rtp_set_rtpmap_type(p->rtp, codec, "audio",
mimeSubtype);
if (p->vrtp)
ast_rtp_set_rtpmap_type(p->vrtp, codec, "video",
mimeSubtype);
}
For evaluating the INACTIVE of the optiPoint telephones, the If loop has to be adjusted in a certain way so that the INACTIVE is interpreted as SENDONLY.
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c01.fm
Music on Hold (MOH)
Execution of Call Pickup
This is the modified source code:
sdpLineNum_iterator_init(&iterator);
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough
space */
if (!strcasecmp(a, "sendonly")) {
sendonly=1;
continue;
}
/* condition added by VGarroy to test MOH */
if (!strcasecmp(a, "inactive")) {
sendonly=1;
continue;
}
if (!strcasecmp(a, "sendrecv")) {
sendonly=0;
}
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
if (debug)
ast_verbose("Found description format %s\n",
mimeSubtype);
/* Note: should really look at the 'freq' and '#chans' params
too */
ast_rtp_set_rtpmap_type(p->rtp, codec, "audio",
mimeSubtype);
if (p->vrtp)
ast_rtp_set_rtpmap_type(p->vrtp, codec, "video",
mimeSubtype);
}
To compile the changes into the system, complete the following steps:
1.
Go to the folder containing the source files (default: /usr/src/asterisk/).
2.
Execute the commands make and make install.
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25
c01.fm
Reference List
Execution of Call Pickup
9
Reference List
This list comprises references to documentation or other sources for additional information.
This list is not exhaustive.
Official Asterisk Website
News, general information, supported hardware, support and forums, bug tracking.
URL: http://www.asterisk.org/
Asterisk Wiki on voip-info.org
Up-to-date and comprehensive collection of information about Asterisk, e.g. articles, how-to’s
references.
URL: http://www.voip-info.org/wiki/view/Asterisk
Digium
Digium Inc., co-founded by Mark Spencer, the creator of Asterisk, is the most important developer of Asterisk software.
URL: http://www.digium.com
User Guides for the optiPoint 410 / 420 family of products
Description of local and system-based operating processes, explanation of user interface, etc.
URLs:
http://www.siemens.de/hipath/ > Products, Solutions & Services > Clients & Devices > SIP
Telephones > User Guides
http://www.siemens.de/hipath/ > Products, Solutions & Services > Clients & Devices > SIP
Telephones > User Guides
Administrator Guides for the optiPoint 410 / 420 family of products
Details on administrative tasks for setup and configuration.
URL: http://www.siemens.de/hipath/ > Downloads > Administration Guides
Select "Englisch" and the product category "Phones".
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Sample Configuration
Execution of Call Pickup
10
Sample Configuration
The following section is a summary of all the sample configurations used in this Configuration
Guide.
sip.conf
[general]
context=default
realm=usergruppe@asterisk
bindport=5060
bindaddr=0.0.0.0
allow=alaw
qualify=yes
dtmfmode=info
pedantic=yes ; fuer Multi Line
notifymimetype=text/plain
[10]
type=friend
context=myphones
username=10
secret=123456
host=dynamic
mailbox=10@myphones
callgroup=1
pickupgroup=1
[11]
type=friend
context=myphones
username=11
secret=123456
host=dynamic
mailbox=11@myphones
callgroup=1
pickupgroup=1
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27
c01.fm
Sample Configuration
Execution of Call Pickup
extensions.conf
[myphones]
exten => 10,1,Dial(SIP/10,5,r)
exten => 10,2,VoiceMail(u10@myphones)
exten => 10,102,VoiceMail(b10@myphones)
exten => 11,1,Dial(SIP/11,5,r)
exten => 11,2,VoiceMail(u11@myphones)
exten => 11,102,VoiceMail(b11@myphones)
exten => 99,1,VoiceMailMain()
exten => _*8.,1,Pickup(10&11)
voicemail.conf
[general]
format=wav49|gsm|wav
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www.siemens.de/hipath
The information provided in this document contains mereley 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 product.
An obligation to provide the respective characteristics shall only exist if
expressly agreed in the terms of contract.
The trademarks used are owned by Siemens AG or their respective
owners.
© Siemens AG 2006 • Siemens Communications •
Hofmannstraße 51 • D-81359 München
Subject to availability. Right of modification reserved.