Download  Print this page

Siemens OpenStage Asterisk Installation And Maintenance Manual

Hide thumbs

Advertisement

HowTo
OpenStage@Asterisk
Installation and Maintenance
Guide
Issue 1.0
Siemens Enterprise Communications GmbH & Co KG
Munich, 09/07/2010
Germany
Siemens Enterprise Communications
www.siemens-enterprise.com

Advertisement

loading

  Summary of Contents for Siemens OpenStage Asterisk

  • Page 1 HowTo OpenStage@Asterisk Installation and Maintenance Guide Issue 1.0 Siemens Enterprise Communications GmbH & Co KG Munich, 09/07/2010 Germany Siemens Enterprise Communications www.siemens-enterprise.com...
  • Page 2: Scope

    Scope This document provides a best practice guide on setting up, operating, servicing, and troubleshooting OpenStage phone in an Asterisk environment. Open Communications Principles and Best Practices 19/03/2012, page 2...
  • Page 3: Table Of Contents

    Contents Scope Contents Preparation Supplying Power for the Phones Connecting OpenStage Phones to the IP Network 802.1x LLDP-MED Configuration Options DHCP Options Plug & Play – One Step Provisioning and Configuration Single Phone Configuration (Local Menu, WBM) Using OpenStage@Asterisk Busy Lamp Function (BLF) XML Applications Send URL / Remote Server Control Call Completion (CCBS/CCNR)
  • Page 4: Preparation

    Preparation This chapter contains all information that is necessary to connect an OpenStage phone to an Asterisk based communication system. This includes the power supply options (PoE or external supply) for each OpenStage model and its possible sidecar combinations. To enable a secure environment, 802.1x support for OpenStage is specified.
  • Page 5 Energy saving mode To reduce the energy consumption to a minimum, OpenStage phones offer an energy saving mode. The display backlight (phone and Key Module, if attached) 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).
  • Page 6: Connecting Openstage Phones To The Ip Network

    Power consumption [W] - Gigabit Ethernet variants Energy saving Idle state During call Ringing (max. mode (handset) vol.) OpenStage 20 G OpenStage 40 G OpenStage 60 G 10,0 10,3 Please see note *) 10,1 4,2 *) 8,4 *) 10,0 9,0 *) 10,7 11,0 OpenStage 80 G...
  • Page 7: Dhcp Options

    By this means, full Plug & Play is possible (see the following section). For further information, including an example configuration for dhcp, please refer to the OpenStage Asterisk Admin Guide [3]. Plug & Play – One Step Provisioning and Configuration...
  • Page 8: Single Phone Configuration (Local Menu, Wbm)

    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 refer to the WPI Developer’s Guide [8]. If a Firewall or NAT get in the Way In case the phones and the provisioning service reside in different networks or subnets that are separated by a firewall and/or NAT, it may be impossible for the provisioning service to contact the phones.
  • Page 9: Using Openstage@Asterisk

    Possible uses for OpenStage XML Applications might be • Integration with groupware (e.g. Microsoft Exchange Server) or Unified Messaging systems (e.g. Siemens Enterprise Communications OpenScape) • phonebooks with access to address databases •...
  • Page 10: Call Completion (Ccbs/Ccnr)

    An implementation according to the IETF BLISS draft is planned, but currently not available. The OpenStage call completion implementation is purely stimulus based and can be found http://wiki.siemens-enterprise.com/images/6/65/White_Paper_CC_10090.pdf CTI for OpenStage - UACSTA There are several use cases where remote control of a VoIP phone is required. Among these are server based features like ‘Call Forwarding’...
  • Page 11: Changing The Caller Information - Pai Header

    OpenStage does not generate CSTA Events. With these services a SIP server can easily control basic OpenStage functions. For futher information please have a look at: http://wiki.siemens- enterprise.com/images/e/e7/white_paper_uaCSTA_Public_version_2010803.pdf Changing the Caller Information – PAI Header SIP is a great protocol for call processing. However, in some use cases, additional and up-to- date information about a caller might prove to be very useful.
  • Page 12: Multi Address Appearance (Maa)

    Multi Address Appearance (MAA) A telephone is normally associated with a directory number, or, generally, with a SIP AoR. This number is used for placing calls to the associated telephone and for displaying the telephone's (user's) identity when placing calls to another party. The number is also used when more than one call appearance is supported due to additional features like call waiting.
  • Page 13: Automatic Call Pickup - Using Alert Info Header

    Example: OpenStage operates in MAA mode. Further information can be found at: http://wiki.siemens-enterprise.com/images/a/a3/White_Paper_MAA.pdf Automatic Call Answering Using Alert-Info Header Besides using uaCSTA, the phone can be set to automatically answer a call by adding an alert info header to the call. Thus, the SIP server is enabled to control whether a call is to be answered immediately and without user interaction.
  • Page 14: Logging And Tracing

    Logging and Tracing OpenStage phones are perfect. But if something should go wrong anyhow, the customer service needs tools to focus on the problem. Service effort is needed, but should be minimized. Therefore, 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.
  • Page 15: Local And Remote Tracing

    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 it is not recommended to enable all traces to the deepest level.
  • Page 16 Example Screen: OpenStage 80 represented in the HUSIM Phone Tester Open Communications Principles and Best Practices 19/03/2012, page 16...
  • Page 17: Limitations

    [1] TIA-811-A: Performance and Interoperability Requirements for Voice-over-IP (VoIP) Feature Telephones (http://www.tiaonline.org/standards/technology/voip/documents/TIA-811-A-final-for- global.pdf) [2] Session Initiation Protocol (SIP)-Specific Event Notification: (RFC 3261) [3] OpenStage Asterisk Admin Guide: (http://wiki.siemens- enterprise.com/images/e/e1/Administration_Manual_OpenStage_Asterisk.pdf) [4] WPI Guide: (http://wiki.siemens- enterprise.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s_Guide.p [5] Trace Guide Openstage SIP: http://wiki.siemens- enterprise.com/images/1/1b/Service_Info_How_to_trace_OST_SIP.pdf...
  • Page 18 Copyright © Siemens Enterprise Communications GmbH & Co.KG Hofmannstr. 51, D-80200 München Siemens Enterprise Communications GmbH & Co. KG is a Trademark Licensee of Siemens AG The information provided in this document contains merely 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 products.