Sip features, Sip join” feature (not applicable to the 6751i), Limitations of the “sip join” feature – Aastra Telecom 675xi Series User Manual

Page 13: Sip join” feature (not applicable to the, 6751i)

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New Features in Release 2.4

RN-001029-03, Release 2.4, Rev 00

9

SIP IP Phone Models 9143i, 9480i, 9480i CT, and 675xi Series Phones Release Note 2.4

“SIP Join” Feature (not applicable to the 6751i)

Release 2.4 includes support for 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.

SIP Features

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