HiPath optiPoint 410 S / 420 S Using with Asterisk Configuration Guide
bkIVZ.fm Nur für den internen Gebrauch 1 Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 2 System Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3 Configuration of Asterisk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3.1 SIP Registration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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. 30.
c01.fm 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.
c01.fm 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.
c01.fm 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". 12 30.
c01.fm 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. 30.
c01.fm 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.
c01.fm Basic Configuration of optiPoint Telephones SIP Environment 4.4 SIP Environment Use this dialog to enter basic SIP parameters. 30.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
c01.fm 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.
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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.
c01.fm 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.
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 28 30.
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.
