OpenStage and Asterisk



This article describes the setup, operation, and operation of OpenStage SIP phones in an Asterisk telephony environment. For a detailed HowTo, please see.

All phones of the OpenStage SIP family with SIP firmware version ≥ V1 R5.6.0 are interoperable with Asterisk: 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:
 * OpenStage 15 (Digium certified, see Asterisk Exchange)
 * OpenStage 20
 * OpenStage 40
 * OpenStage 60 (Digium certified, see Asterisk Exchange)
 * OpenStage 80


 * 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.

Power over Ethernet
The following phone configurations can be operated with PoE, provided that the switch has the appropriate power class:

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

Energy Saving
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.

802.1x
OpenStage phones support 802.1x EAP-TLS. Certificates for authentication can be downloaded via the WPI.

LLDP-MED
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 LLDEP-MED usage is configured in the administratio menu under Network > IP configuration.
 * 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.

DHCP
The following parameters can be obtained by DHCP:

Basic Configuration

 * IP Address
 * Subnet Mask (option 1)

Extended Configuration

 * 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 .

Plug&Play
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.

Using OpenStage Phone with Asterisk
For an overview of the features introduced with firmware version V2R1, please refer to

Feature Table
In this table, you find information on all features which are supported by OpenStage phones connected to an Asterisk PBX.

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.

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.

Basic Troubleshooting
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/SEBA. The key can be distributed to the phone using the SEN DLS (Deployment Service) or the phone’s WPI (WorkPoint Interface).

Documentation

 * (Umbrella document how to install, administrate and use OpenStage@Asterisk.)
 * (List of all new features contained in software version V2 R1.)
 * (A guide for getting needed trace information from the phone.)
 * (How to use the OpenStage built in call completion support.)
 * (Multiple Address Appearance on OpenStage SIP.)
 * (Using uaCSTA to control the phone from the server and vice versa.)
 * (Multiple Address Appearance on OpenStage SIP.)
 * (Using uaCSTA to control the phone from the server and vice versa.)