Limitations of the “sip join” feature, Conference ends and transfers remaining parties, Reference – AASTRA 6700i series, 9143, 9480i, 9480i CT SIP Administrator Guide EN User Manual
Page 620
41-001343-01 Rev 03, Release 3.2.2
5-343
• 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.
Configuring the “SIP Join” Feature Using the Configuration Files
You use the following parameters to configure the “SIP Join feature in the configuration files:
sip join support
Conference Ends and Transfers Remaining Parties
On the IP Phones, a Host can drop from a 3-way conference and let the other two parties remain
connected by pressing a Xfer key.
Reference
For more information about using the 3-way conference and the Xfer key, see the IP Phone
Model-specific User Guide.
Configuration Files
For specific parameters you can set in the configuration files, see Appendix A, the section,
Notes:
1.
The 3-line LCD display phones (6730i, 6731i, 6753i, and 9143i)
and cordless handsets (CT handsets) do not support scrolling between
lines.
2.
The phones MUST have a Xfer key (hardkey, or configured
programmable or softkey) on the phone.