voip-info.org中介绍如下：

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:

exten => 100,hint,SIP/peername

(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)

在asterisk1.8中

在dialplan中分机对应拨号计划上添加如下：

exten=>600,hint,SIP/600

exten=>600,1,Dial(SIP/600)

如遇到通话中状态不对应在sip.conf中添加callcounter=yes 即可

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asterisk1.8中设置presence或blf

voip-info.org中介绍如下：

Asterisk presence

asterisk1.8中设置presence或blf相关推荐

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