SIP Devices and Asterisk
The Wiki of Unify contains information on clients and devices, communications systems and unified communications. - Unify GmbH & Co. KG is a Trademark Licensee of Siemens AG.
This article describes the setup, operation, and operation of OpenStage SIP phones in an Asterisk telephony environment. For a detailed HowTo, please see HowTo_OpenStage_Asterisk.
All phones of the OpenStage SIP family with SIP firmware version ≥ V1 R5.6.0 are interoperable with Asterisk:
- OpenStage 15 (Digium certified, see Asterisk Exchange)
- OpenStage 20
- OpenStage 40
- OpenStage 60 (Digium certified, see Asterisk Exchange)
- OpenStage 80
Common telephony features are supported out of the box, such as call transfer, call forwarding, consultation, voicemail, and many more. To add more functionality in order to build a feature-rich communication system, two additional interfaces can be used by developers:
- The OpenStage_WPI (WorkPoint Interface) provides an open, XML-based provisioning interface to support mass deployment and to enable automated updates and configuration.
- OpenStage 60 and 80 provide an XML-based application interface which allows for developing graphical applications hosted on a remote server. Besides displaying information, sending data, and controlling all sorts of remote processes, these applications have the capability of controlling calls.
| The direct contact for OpenStage and Asterisk related questions from the community:|
- 1 Preparation
- 2 Using OpenStage Phone with Asterisk
- 3 Service and Troubleshooting
- 4 Documentation
- 5 References
- 6 See Also
Power over Ethernet
The following phone configurations can be operated with PoE, provided that the switch has the appropriate power class:
|Model / Configuration||Power Class|
|OpenStage 15 (includes 1 Key Module 15)||1|
|OpenStage 20 E||1|
|OpenStage 20 G||2|
|OpenStage 40 (includes 1 Key Module)||2|
|OpenStage 40 + 2nd Key Module||2|
|OpenStage 40 G||3|
|OpenStage 40 G + 2nd Key Module||3|
|OpenStage 60/80 (includes 1 Key Module + USB-Extension with Acoustic Unit)||3|
|OpenStage 60/80 + 2nd Key Module||3|
|OpenStage 60/80 G (includes 1 Key Module + USB-Extension with Acoustic Unit)||3|
External Power Supply
For an OpenStage 60/80 G with a 2nd Key Module, an external power unit is required. The order no. for the plug-in power supply is region specific:
- EU: C39280-Z4-C510
- UK: C39280-Z4-C512
- USA: C39280-Z4-C511
OpenStage phones offer an energy saving mode. The display backlight is switched off after a configurable timeout. With OpenStage 40, the main display and key module backlight will be switched off after 90 seconds of inactivity (firmware version V2R0 onwards). Readability even without backlight is ensured by the transflective display. With OpenStage 60 and 80, the timer is configurable by the administrator (Local Functions > Energy saving); the timeout ranges between 2 and 8 hours.
Connecting to an IP Network
OpenStage phones support 802.1x EAP-TLS. Certificates for authentication can be downloaded via the WPI.
OpenStage SIP phones support the layer 2 protocol LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery). When an OpenStage phone is connected to a switch with LLDP-MED capabilities, the phone is able to
- advertise and receive a VLAN ID,
- advertise and receive QoS parameters,
- advertise the power requirements to the LAN access switch by means of an "Extended Power via MDI" TLV.
LLDEP-MED usage is configured in the administratio menu under Network > IP configuration.
The following parameters can be obtained by DHCP:
- IP Address
- Subnet Mask (option 1)
- Default route (option 3)
- Static IP routing (option 33)
- SNTP server (option 42)
- Timezone offset (option 2)
- Primary/secondary DNS server (option 6)
- DNS domain name (option 15)
- SIP Addresses / SIP Server & Registrar (SIP Server option 120)
- Vendor unique (option 43)
The vendor specific option (code 43), or alternatively, a vendor class, is used to provide the phone with the location of an optional configuration/provisioning service. By this means, full Plug&Play is possible (see the Plug&Play) section. For further information, including an example configuration for dhcp, please refer to the Administration Manual OpenStage Asterisk.
A fully automated mass rollout of OpenStage phones can be realized by combining a DHCP server and a provisioning service which uses the WPI. On startup, the phone receives the IP address of the provisioning server from the DHCP server. After that, it contacts the provisioning service. The provisioning service may then request all settings from the phone in order to decide which parameters must be set or updated. When all these parameters have been sent to the phone, it is ready for operation. For further information, please see the WPI article; for a deeper understanding, refer to the OpenStage Provisioning Interface Developer's Guide.
Using OpenStage Phone with Asterisk
For an overview of the features introduced with firmware version V2R1, please refer to ReadMe V2 R1 100907
In this table, you find information on all features which are supported by OpenStage phones connected to an Asterisk PBX.
|Feature||Phone with firmware version ≥||Short Description||Further Information|
|Local Phone Book|| OpenStage 60/80 ≥ V1 R5.6.0
OpenStage 40 ≥ V2R1
|Stores names and call numbers locally on the phone.|| OpenStage_Phone_Book_Application|
Copy LDAP results and Call Logging records into Local Address book
|LDAP|| OpenStage 60/80 ≥ V1 R5.6.0
OpenStage 40 ≥ V2R1
|The phone can be used to retrieve call numbers or other address data from an LDAP directory.|| LDAP on OpenStage|
Support of LDAP on OpenStage 40
|Call Log||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Stores a list of missed, dialed, received, and forwarded calls. With V2R1 onwards, the call log can be cleared via the WPI.||Configurable flag to delete call log contents|
|MWI||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Message Waiting Indication. The user is notified of new and old voicemail messages. With V2 onwards, the message indication is configurable.|| MWI Subscription|
Configuration of MWI count names
|Help||OpenStage 60/80 ≥ V1 R5.6.0||Quick reference which is shown on the phone display when the Help mode key is pressed.||-|
|XML Application Platform||OpenStage 60/80 ≥ V1 R5.6.0||XML-based interface to the phone which allows for developing interactive applications. The phone acts as a front-end for a server-side program.||XML Application Platform at the SEN Community Portal|
|CTI Applications||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Allows a computer to interact with the phone, e.g. in setting up and terminating calls via a PC application. OpenStage phones support 3rd party call control via SIP and uaCSTA. By the use of uaCSTA, operations like call answering, putting a call on hold, making a call, setting microphone and speaker settings, and many more, can be performed.|| OpenStage_CTI_Applications|
|FPK (Free Programmable Keys)||OpenStage 15/40/60/80 ≥ V1 R5.6.0||These keys can be associated with special phone functions.||OpenStage_Free_Programmable_Keys|
|Hotdesking||OpenStage 15/40/60/80 ≥ V1 R5.6.0||A user can move from one phone to another one, taking along his profile data, for instance, call number, call list, programmed keys, ring tones. The profile data is transferred using the WPI.|| Hotdesking for OpenStage phones and other phone models is implemented in the AMOOMA Gemeinschaft PBX.|
For information about this solution, please see this screencast ( in german).
|Call Related Features|
|Busy Lamp Field (BLF)||OpenStage 15/40/60/80 ≥ V1 R5.6.0||This function offers the possibility to monitor another extension, and to pick up calls for the monitored extension. The LED of the key will indicate the state of the extension monitored.||Asterisk Feature Busy Lamp Field (BLF)|
|Phone Based Conference||OpenStage 15/40/60/80 ≥ V1 R5.6.0||The phone establishes and controls a 3-way conference (3pcc, third party call control).||-|
|Voice Mail||OpenStage 15/40/60/80 ≥ V1 R5.6.0||The user can contact the Asterisk voice mail service by means of the messages key.||-|
|Call Group||OpenStage 15/40/60/80 ≥ V1 R5.6.0||If a phone belongs in a pickup group that matches one of the caller's call groups, that phone may pickup the incoming call by sending the appropriate feature code.||-|
|Call Pickup||OpenStage 15/40/60/80 ≥ V1 R5.6.0||The BLF function offers the possibility to pick up calls for a monitored extension. The LED of the key will indicate the state of the extension monitored.||Asterisk Feature Busy Lamp Field (BLF)|
|Call Forward||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||The user can configure local or server based call forwarding to specified destinations.||For information about server based call forwarding, see Fix Forwarding Key to send URL to Server.|
|Call Waiting||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||If allowed, a call from a third party will be indicated acoustically during an active call.||-|
|Do Not Disturb (DND)||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||If this feature is activated, incoming calls will not be indicated to the user.||-|
|Auto Answer||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Automatic call answering can be requested from within the incoming call via the SIP Alert-Info header.||Auto Answer Configuration|
|Call completion (CFBS, CFNR)||- TBD -||Call completion is a telephony feature which takes action on a failure to complete a call. It allows for notifying the calling user when the called user is available again. CCBS (Call Completion Busy Subscriber) will take effect when the called party is busy; CCNR (Call Completion No Reply) will take effect when the called party does not respond.||White_Paper_CC|
|Call Transfer||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Blind call transfer and call transfer with consultation are supported. In a blind transfer scenario, user A selects the blind transfer option during a conversation with user B and enters the number of user C. After that, user B is disconnected from user A and rings at user C's phone. In a consultation scenario, user A initiates a consultation call to user C during a conversation with user B. After haveing returned to the conversation with B, he selects the transfer option. User A is disconnected from user B, and B is connected to C.||-|
|Executive/Assistant configuration||OpenStage 60/80 ≥ V1 R5.6.0||Complex configurations with multiple executive and assistents which indicate the current status of the relevant persons can be realized using the OpenStage XML application platform.||XML Application Platform at the SEN Community Portal|
|DTMF||OpenStage 60/80 ≥ V1 R5.6.0||If control codes are to be sent to the PBX during a call, DTMF (Dual Tone Multi Frequency) tones can be used.||-|
|Alternate Call||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||The user can alternate between the currently active call and another call that is on hold.||-|
|Call Hold||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||The user can put a call on hold in order to switch over to another connected call or to call another party.||-|
|Consultation||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||During an active call, the user can initiate a consultation call to a third party. After that, he can alternate between the two parties.||-|
|CLIP||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||When the call number resp. caller ID is transmitted within an incoming call, it is displayed in ringing state.||-|
|CLIR||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||Caller ID transmission is suppressed.||-|
|Local Music on Hold||OpenStage 15/20/40/60/80 ≥ V1 R5.6.0||If desired, OpenStage phones can be configured to play custom hold music to the user when put on hold. The audio or mp3 file can be uploaded via FTP; the download can be initated via local menu, WBM or WPI.||FAQ - Music On Hold for OpenStage|
|MAA (Multiple Address Appearance)||-TBD -||The Multiple Address Appearance feature provides the served user with multiple addresses appearing on a single telephone. The served user has the ability to originate, receive and otherwise control calls on each of these address appearances. These address appearances behave independently of each other.||White_Paper_MAA|
Service and Troubleshooting
OpenStage phones provide plenty of tools and options to find the cause of a problem quickly, even if it is not located at the phone.
For a guide to error tracing with OpenStage phones, please refer to Service_Info_How_to_trace_OST_SIP.
LAN Port Mirroring
Every OpenStage phone has a built-in Ethernet switch with a LAN port and a PC port. For development and error tracing, the PC port enables network monitoring when configured as a mirror for the LAN port. For this purpose, PC port mode must be set to "mirror". If configured this way, the complete traffic of the LAN port will be passed through to the PC port, just like with a simple network hub. Now, a network tracing tool on the PC can trace all IP traffic, like SIP over UDP, or XML over HTTP, for instance.
Tracing Capabilities within the Phone
For tracking network issues, the phone can execute ping and traceroute tests; these can be controlled and viewed online using the WBM.
For elementary troubleshooting, the phone provides an overview about basic issues in the user menu. The admin can ask the user to read that basic information to get a first hint about the possible causes of an issue. For a table which contains the possible error codes and their causes, please see the Error Codes section of the OpenStage SIP FAQ.
Local and Remote Tracing
The phone is able to write internal trace files, and to send the trace data to a remote syslog server. The tracing can be configured in a differentiated way by setting discrete trace levels for each service. Please note that, order to preserver phone ressources, it is not recommended to enable all traces to the deepest level.
QoS Data Collection
OpenStage phones generate QoS reports using a HiPath specific format, QDC (QoS Data Collection). The reports created for the last 6 sessions, i. e. conversations, can be viewed on the WBM or are reported to the QCU (QoS data Collection Unit). SEN provides a server application to collect the data. The collected data is sent via SNMP. If an SNMP server is available, the QDC MIBS can be downloaded from our software supply server (SWS). Meanwhile, third party solutions are available which can also deal with the OpenStage QDC data.
HUSIM Phone Tester
This tool enables the service staff to access a defined group of phones remotely.
For each phone, a PC application window shows the current status. Every OpenStage phone model is represented with its complete key layout and display content. The remote visitor can see all user interactions on the phone. Moreover, he can access the phone keys actively and in this way operate the phone by remote control. Please note that, for privacy protection, the user is always informed about the remote interaction.
To get the phone tester up and running, a special dongle key must be uploaded to the phone. The dongle key and the HUSIM software can be downloaded without additional charge from SWS/Partner Portal. The key can be distributed to the phone using the SEN DLS (Deployment Service) or the phone’s WPI (WorkPoint Interface).
- Administration Manual OpenStage Asterisk
- HowTo OpenStage Asterisk (Umbrella document how to install, administrate and use OpenStage@Asterisk.)
- OpenStage Provisioning Interface Developer's Guide
- ReadMe V2 R1 100907 (List of all new features contained in software version V2 R1.)
- Service Info How to trace OST SIP (A guide for getting needed trace information from the phone.)
- White Paper CC (How to use the OpenStage built in call completion support.)
- White Paper MAA (Multiple Address Appearance on OpenStage SIP.)
- White_Paper_uaCSTA_for_OpenStage_SIP (Using uaCSTA to control the phone from the server and vice versa.)