Changing The Caller Information - Pai Header - Siemens OpenStage Asterisk Installation And Maintenance Manual

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.
Open Communications Principles and Best Practices 19/03/2012, page 11