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作者: ckleea 時間: 2011-6-27 18:38 標題: 3 人會議

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[asterisk-users] Three-way conference in Asterisk
Satish Barot satish4asterisk at gmail.com
Thu Jun 2 05:50:12 CDT 2011

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Nikhil,

This is how I would implement '3 way conference' in Asterisk with the help
of dynamic features.
Assume 3 SIP friends 1110,1111 and 1112 in sip.conf. For 1110 in sip.conf,
context=test3way

Add following in applicationmap section of features.conf....

[applicationmap]

3way-start => **0,caller,Macro,3way-start
3way-conf => **1,caller,Macro,3way-conf
3way-noconf => **2,caller,Macro,3way-noconf

My dialplan would be....

[test3way]
exten => 1212,1,Noop(########## TLC Check ##########)
same => n,set(DYNAMIC_FEATURES=3way-start)
same => n,Dial(SIP/1111,30,m)

[dynamic-3way]
exten => _XXX.,1,Answer
exten => _XXX.,n,Set(CONFNO=1212)
exten => _XXX.,n,Set(DYNAMIC_FEATURES=)
exten => _XXX.,n,ConfBridge(${CONFNO},M)
exten => _XXX.,n,Hangup

[macro-3way-start]
exten => s,1,Set(CONFNO=1212)
exten => s,n,ChannelRedirect(${BRIDGEPEER},dynamic-3way,${CONFNO},1)
exten => s,n,wait(1)
exten => s,n,Set(DYNAMIC_FEATURES=3way-conf#3way-noconf)
exten => s,n,Dial(SIP/1112,,g)
exten => s,n,Set(DYNAMIC_FEATURES=)
exten => s,n,ConfBridge(${CONFNO},M)

[macro-3way-conf]
exten => s,1,ChannelRedirect(${BRIDGEPEER},dynamic-3way,${CONFNO},1)

[macro-3way-noconf]
exten => s,1,SoftHangup(${BRIDGEPEER})

You can dial 1212 from SIP Extension 1110 which will connect 1110 to 1111.
No while talking to 1111, 1110 can press **0 to invoke '3way-start' feature
which in turn call 1112.

Now while talking to 1112, 1110 can press **1 to start the conference.

I suggest you should go through features.conf for more information. This is
very basic dialplan for 3 way conference. You will have to add some more
stuffs to make it work in the way you want.

Hope this helps.
[SATISH]

作者: 亞星 時間: 2011-6-27 21:02

IP01 用咗好耐啦, 真係唔明點解 asterisk 之前只係 analog 先可以 3 ways.

http://www.telecom-cafe.com/foru ... &extra=page%3D1

作者: ckleea 時間: 2011-6-27 21:19

回復 2# 亞星

有時設計上的問題。IPPBX 通常都會貼心D，不過這不是絕對。

作者: Qnewbie 時間: 2011-6-28 01:14

回復 2# 亞星

I think IP01 firmware from ATCOM does not apply patch to Zaptel/Dahdi timer, while the switchfin does.

作者: alang 時間: 2011-6-28 14:19

突然有個想法：
既然這裡高手如雲，大家何不一起來針對這個項目 3(n)-way Conference 一起來合作開發，使這一個功能更適合每個人用。
有些人就貢獻 code, 有些人就貢獻 idea, 有些人就幫忙測試,有些人也可以在旁邊鼓勵。
只是，究竟會有多少人對這項目有興趣呢？

作者: ckleea 時間: 2011-6-28 14:35

回復 5# alang

好建議，雖然使用3 人會議的機會很少，但其他方面會提出。

如最近的想法是整合唔同的sip / iax trunks into a common dialplan macro，然後透過speed dial 完成。即是 user 完全唔知asterisk server用那條 trunk 打出，背後目的是將可以在GUI 見到的減到最少。

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