voip-info.org中介紹如下:
Asterisk presence
Current CVS -HEAD provides some support of SIP presence as defined in RFC 3856 - A Presence Event Package for the Session Initiation Protocol (SIP). The release version (1.0.9) has some support too.
To support this, you have to somehow tell which SIP user is mapped to the extension mentioned. A "hint" priority is used to describe this, as in this example:
(it says, that for presence purposes, extension 100's presence should be mapped the SIP peer)
- IMPORTANT: In Asterisk 1.4 the inner workings of the hint functionality have slightly changed. It is now imperative that you set a call-limit (even if it's an arbitrarily-high value like 100) and/or the new limitonpeers value in sip.conf. Presence will not work without this. If you are using friend instead of peer, you will need limitonpeers = yes as well as a call-limit statement (new in Asterisk 1.4) for each SIP device
- If you do not add a hint, the extension will be free ("open") always, more or less.
- If you add incominglimit=1 to your peer (in Asterisk 1.2 only) in sip.conf, the SIP channel will notify you when that extension is busy.
- IMPORTANT: In Asterisk 1.6.2.X in sip.conf you need callcounter=yes for presence to work (http://lists.digium.com/pipermail/asterisk-users/2010-May/248222.html)