Limitations of the “sip join” feature, Draft 1 – Aastra Telecom 9480i Series User Manual
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Operational Features
41-001160-03, Rev 00, Releaes 2.4
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Configuring Operational Features
“SIP Join” Feature for 3-Way Conference (not applicable to the 6751i)
The IP Phones support RFC 3911 which allows an additional caller to join an
active call between two parties if the caller knows the dialog information. This
feature begins a conference using a join header as described in RFC 3911.
The “SIP Join“ feature provides the following:
•
Security via the whitelist (which is a feature that already exists on the phone).
•
Initiates an offhook action uri when it is answered.
•
Initiates an onhook action uri at call termination.
•
Creates a caller list entry.
This feature is disabled by default. You can enable the “SIP Join” feature by
setting the “sip join support” parameter in the configuration files.
Limitations of the “SIP Join” Feature
The following are limitations of the “SIP Join” feature:
•
Not applicable to a conference call already in progress.
•
Not applicable to a CT handset that has two active calls.
•
Not applicable to a phone mixing RTP.
•
Allows secondary parties to join calls if they can determine the dialog
parameters. In order to provide security, it is recommended that the
Administrator configure the SIP whitelist.
•
Not applicable while the active call between two parties is in the early dialog
state.
Draft 1