Siemens OpenStage Asterisk User Manual

Open Communications Principles and Best Practices 19/03/2012, page 11
 Do Not Disturb Event
 Diverted Event
 Services on the OpenStage device:
 Make Call
 Answer Call
 Hold Call
 Retrieve Call
 Clear Connection
 Consultation Call
 Generate Digits
 Get Volume
 Set Volume
 Get Mute
 Set Mute
 Events Generated by OpenStage:
 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. Among the possibilities are:
 Add caller information from an external database
 Update caller information during call transfer
 Add hunt group information for incoming calls
 Enhance Executive/Assistant features with additional information
OpenStage supports RFC 3325 [7]. A SIP server can change the OpenStage display
information using SIP INVITE or UPDATE requests or any SIP response code.
Especially the P-Asserted-Identity Header can be used to carry additional information to the
phone user.

Siemens OpenStage Asterisk Specifications

General

Brand	Siemens
Model	OpenStage Asterisk
Category	Telephone
Language	English