PSTN incoming call to IVR

角色

If a person calls the PSTN-1, extension 6001 will ring. If PSTN-2, the system will enter an ivr-1 routine.

YH

dahdi-channels.conf

1. ; Autogenerated by /usr/sbin/dahdi_genconf on Sun Aug 29 15:00:16 2010
   
2. ; If you edit this file and execute /usr/sbin/dahdi_genconf again,
   
3. ; your manual changes will be LOST.
   
4. ; Dahdi Channels Configurations (chan_dahdi.conf)
   
5. ;
   
6. ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
   
7. ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
   
8. ;
   
9. 
   
10. ; Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
    
11. ;;; line="1 WCTDM/4/0 FXSKS  (In use) (SWEC: MG2)"
    
12. signalling=fxs_ks
    
13. callerid=asreceived
    
14. group=0
    
15. context=from-pstn-1
    
16. channel => 1
    
17. callerid=
    
18. group=
    
19. context=default
    
20. 
    
21. ;;; line="2 WCTDM/4/1 FXSKS  (In use) (SWEC: MG2)"
    
22. signalling=fxs_ks
    
23. callerid=asreceived
    
24. group=0
    
25. context=from-pstn-2
    
26. channel => 2
    
27. callerid=
    
28. group=
    
29. context=default

複製代碼

extensions.conf

1. [ivr-1]
   
2. ;ivr-1
   
3. exten => 1130,1,Answer
   
4. exten => 1130,n,Wait(2)
   
5. exten => 1130,n,Background(vm-enter-num-to-call)
   
6. exten => 1130,n,WaitExten
   
7. exten => 1130,n,Playback(vm-goodbye)
   
8. exten => 1130,n,Hangup
   
9. exten => i,1,Playback(pbx-invalid)
   
10. exten => i,2,Goto(1130,1)
    
11. exten => t,1,Playback(vm-goodbye)
    
12. exten => t,2,n,Hangup
    
13. 
    
14. 
    
15. [internal]
    
16. 
    
17. include => ivr-1
    
18. 
    
19. exten => 6001,1,Dial(SIP/6001,,r)
    
20. exten => 6002,1,Dial(SIP/6002,,r)
    
21. 
    
22. [from-pstn-1]
    
23. exten => s,1,Dial(SIP/6001,,)
    
24. 
    
25. [from-pstn-2]
    
26. exten => s,1,Goto(internal,1130,1)

複製代碼

ckleea

回復 1# 角色


    Can you give me an example to go directly into voicemail for unanswer after 30 sec?

角色

This is not belong to IVR system. I believe it is belong to the general voicemail setting for some extensions.

YH

ckleea

I mean to implement in 2 ways,
1. without going into IVR, just answering machine, go into voicemail after non-attendance for 30 sec
2. in the IVR, branch to voicemail after pressing certain keys to save the incoming message of caller.

角色

For the first question, you may use the following codes

1. #
   
2. [from-pstn-1]
   
3. exten => s,1,Dial(SIP/6001,30,)
   
4. exten => s,2,Voicemail (........) since I am not familiar with voicemail, I am not sorry that I am not able to provide the codes

複製代碼

For the second second, you make take a look at the Future of Telephony: Asterisk, you may find the answer.


YH

bubblestar

You may try this

[from-pstn-1]
exten => s,1,Dial(SIP/6001,30)       
exten => s,2,VoiceMail(6001,u)


u denotes unavailable

bubblestar

如果唔想為每一條內線都設定相同的CODE，可以試下咁樣。

sip.conf

[globals]
RINGTIME=30

[internal]
; If nobody picks up within 30 seconds, the call is sent to voicemail
; If the extension is busy, the call is sent to voicemail
exten => _6[0-3]XX,1,Set(TARGETNO=${EXTEN})
exten => _6[0-3]XX,n,Dial(SIP/${EXTEN},${RINGTIME})
; routes the call to the status priority (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => _6[0-3]XX,n,Goto(s-${DIALSTATUS},1)
; person at extension "is unavailable" message        
exten => s-NOANSWER,1,VoiceMail(${TARGETNO},u)
; person at extension "is busy" message       
exten => s-BUSY,1,VoiceMail(${TARGETNO},b)
; to be safe, clean up the call after an answer by hanging up           
exten => s-ANSWER,1,Hangup()
; handle any unhandled status the same way we handle NOANSWER               
exten => _s-.,1,Goto(s-NOANSWER,1)                  

我在QNAP NAS 都有用呢套用法。

_6[0-3]XX 代表打入到 這些內線，才會用以上的處理方法去做。如果上面最後一句不加入，那麼不是給_6[0-3]XX接的話，應該會沒有回應便收線 (當處理垃圾電話都好)

ckleea

Thanks, very useful

lttliang

你地用APL  时间长左  唔会忘记以前set左D咩吗？  我用GUI 有半个月或者一个月唔去 睇  都会忘记自己set左D咩系asterisk里

角色

用GUI，根本做不到你特别想要的功能。GUI给你有什么，你只能只能用什么。

角色

ckleea

簡單來說，GUI介面方便管理，但有需要的話，就要用 ssh 修改個別 config files 去完成所需的工作。

現時我用GUI and astreisk 1.6.2.x，基本上可以改個別 config files，但係會加入不少空白的 space，如果用SSH，就可以做得整齊和順暢D。

lttliang

用GUI，根本做不到你特别想要的功能。GUI给你有什么，你只能只能用什么。

角色 ...
角色 發表於 2010-9-8 13:35

    我唔系话GUI可唔可以加自己想要的功能，我只系话如果 用APL加的野，过段时间要改时，唔会记不起一些野吗？因为我用GUI改的野 时间耐 左  都会有D唔记得  都要慢慢 睇先记返起来    而APL冇一个界面  全是字符  点样可以很清晰的知道之前做左咩功能  要修必时  可以一眼就知改边处

角色

这个你要自己做log book，去记录你以前做的的事。

角色

bubblestar

如果驚唔記得，可以係每段設定前後加上一些敍述，方便記憶。

例如:


[from-pstn-1]
exten => s,1,Dial(SIP/6001,30)        ;  從外面接入的電話，轉往內線6001，會晌30秒。
exten => s,2,VoiceMail(6001,u)        ;  30秒沒有人接聽的話，便自動接往內線6001的留言信箱。

如此類推。但係在圖形介面便比較難做到相類似的效果了。

lttliang

如果驚唔記得，可以係每段設定前後加上一些敍述，方便記憶。

例如:


[from-pstn-1]
exten => s,1,Dial(SI ...
bubblestar 發表於 2010-9-8 14:53

    这是个不错的方法  
    退休之后  学APL