I have an Asterisk server that I need to reconfigure. It was built by an outside contractor, and I need to make some changes to it. Right now, all it does is answer a call, accept a 7-digit code, and hang up. On the back end, it records the timestamp of the call, the caller id, and the 7-digit code.
What we are running into is that some people do not enter in the 7-digit password, or they take too long. Then the system just restarts, and will continue in an endless loop, until they enter in 7-digits. The callers are, as of late, thinking that the system is broken, when they do not enter in a 7-digit code.
What I'm trying to figure out how to do is that when the system has to return to the beginning, it might say something like "you only entered 6 digits. pleas try again." Or something to that effect. I'm not 100% sure how to add this into the current configuration. Below is what we currently have:
[inbound]
exten => 1234567890,1,Answer
exten => 1234567890,2,Set(COUNTER=4)
exten => 1234567890,3,Set(COUNTER=$[${COUNTER} -1 ])
exten => 1234567890,4,NoOp(${COUNTER})
exten => 1234567890,5,GotoIf($[${COUNTER} > 0 ]?10:122)
exten => 1234567890,10,Wait(1)
exten => 1234567890,11,read(SCODE,EnterCode,7,)
exten => 1234567890,12,GotoIf($[${LEN(${SCODE})}=7]?13:3)
exten => 1234567890,13,Playback(YouEntered)
exten => 1234567890,n,SayDigits(${SCODE})
exten => 1234567890,n,read(SCHOICE,correctpressone,1,)
exten => 1234567890,n,Gotoif($[ ${SCHOICE} = 1 ]?20:1)
exten => 1234567890,20,NoOp(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} ${SCODE} ${CALLERID(num)})
exten => 1234567890,n,TrySystem(/bin/echo ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}, ${SCODE}, ${CALLERID(num)} >> /opt/codes.log)
exten => 1234567890,n,Playback(SuccessfullyActivate)
exten => 1234567890,n,Hangup()
exten => 1234567890,122,Playback(tt-somethingwrong)
exten => 1234567890,n,Hangup()
Thanks for any help with this issue...
Something like this:
exten => 1234567890,12,GotoIf($[${LEN(${SCODE})}=7]?13:200)
exten => 1234567890,200,Playback(your_inpout_too_short)
exten => 1234567890,201,Goto(3)
Btw, you dialplan is poor quality,looks like person who did it also have no experience.
Related
In my asterisk extension I had wrote like this
[callback]
exten => Set(FROM=${CALLERID(num)})
exten => 2020,1,Answer()
exten => 2020,n,GotoIf(callback)
exten => 2020,n(callback),System(/etc/asterisk/scripts/callback)
exten => 2020,n,Hangup()
exten => 1111,1,Answer()
exten => 1222,1,Dial(SIP/2000) ;here instead of 2000 I want to bring callerid number FROM
exten => 1222,n,Hangup()
When I give SIP/2000 everything is working fine after I give miss call to 2020 I am able to get call.
But when I give like this SIP/$FROM its not working. CallerID number is coming blank.
Can anyone help me to solve this problem
On Asterisk 1.2 or higher, replace the 2nd line with:
exten => Set(FROM=${CALLERID(num)},g)
On Asterisk 1.0 or 1.1, replace the 2nd line with:
exten => SetGlobalVar(FROM=${CALLERID(num)})
How to remove first 3 digits/letters from CALLED NUMBER. Let's say number 123456789 calls to abc987654321. I want to remove abc because in the context I have only 987654321. I know how to cut from CALLER but don't know how to cut from CALLED(Destination) number. This is what I tried so far but nothing happen:
exten => _[a-z]XXXXXXXXXXXX,1,Set({CALLEDID}=${CALLEDID:3})
exten => _[a-z]XXXXXXXXXXXX,2,Dial(SIP/${CALLEDID},1)
exten => _[a-z]XXXXXXXXXXXX,3,Voicemail(${CALLEDID}#VoiceMail)
exten => _[a-z]XXXXXXXXXXXX,4,Playback(Goodbye)
exten => _[a-z]XXXXXXXXXXXX,5,Hangup
You can use FILTER function or just do goto. You not need cut from destination,you need cut from extension.
exten => _[a-z]XXXXXXXXXXXX,1,Goto(${EXTEN:3},1)
exten => _XXXXXXXXXXXX,1,Dial(SIP/${EXTEN},1)
exten => _XXXXXXXXXXXX,n,Voicemail(${EXTEN}#VoiceMail)
exten => _XXXXXXXXXXXX,n,Playback(Goodbye)
exten => _XXXXXXXXXXXX,n,Hangup
Please note, your dialplan still do voicemail if called part hanguped after call. Read default extensions.conf.sample to see how to deal with voicemail
I want to trigger an AGI script (to activate a door opener) while calling. Preferably the script executes on pressing the #-key.
How to embed this behavior in a dialplan? All examples I have found are not dependent on a key press.
I am using FreePBX 2.8.1.4. This is what I have tried:
exten => s,1,Wait(1)
exten => s,n,AGI(test.py)
exten => s,n,Dial(SIP/mk55/203,20,tr)
exten => #,n,AGI(/home/pi/.scripts/dooropen.py)
exten => s,n,Hangup()
and this:
exten => s,n,Read(inPut,,1)
exten => s,n,Dial(console/dsp)
exten => s,n,GotoIf($["${inPut}" = "#"]?keypressed,1)
exten => keypressed,1,AGI(/home/pi/.scripts/dooropen.py)
exten => s,4,Hangup
In deed AGI(script_name, args...) is the right application to use https://wiki.asterisk.org/wiki/display/AST/Application_AGI
To retrieve a DTMF press you can use WaitExten https://wiki.asterisk.org/wiki/display/AST/Application_WaitExten
You can find here a sample of WaitExten usage
Seek Help concerning IVR Menu in Asterisk
Regards
I am using the "monitor" command to record full calls. This works well, but only when the user goes through the entire callflow. I tried monitoring the recorded file size as the call progresses. Once the call starts, the file sizes start increasing (of both the "in" and "out" sides of the call). However, if the user hangs up prematurely in the middle of the call, whatever has been recorded thus far is inexplicably dropped and stubs (44 bytes) are left in its place. Any insight into why this behavior occurs will be appreciated.
I am reproducing a snippet from the dialplan I used in my extensions.conf file below:
exten => 7611,1,Answer()
exten => 7611,n,Playback(/var/lib/asterisk/sounds/custom/transferring_with_record_wa rning)
exten => 7611,n,Set(GROUP()=outgoing)
exten => 7611,n,NoOp(The current group count : ${GROUP_COUNT(outgoing)})
exten => 7611,n,GotoIf($[${GROUP_COUNT(outgoing)}>1]?15)
exten => 7611,n,Set(GLOBAL(current_timestamp_7611)=${STRFTIME(${EPOCH},GMT+1,%s)})
exten => 7611,n,Set(GLOBAL(current_full_format_timestamp_7611)=${STRFTIME(${EPOCH},G MT-8,%d%m%Y_%H%M%S)})
exten => 7611,n,NoOp(The current timestamp : ${current_timestamp_7611})
exten => 7611,n,NoOp(The last timestamp : ${last_timestamp_7611})
exten => 7611,n,GotoIf($[(${last_timestamp_7611}+20>${current_timestamp_7611})]?15)
exten => 7611,n,NoOp(All cases passed)
exten => 7611,n,Ringing()
exten => 7611,n,Wait(2)
exten => 7611,n,Monitor(wav,HALEF_audio_ext_7611_${current_full_format_timestamp_761 1})
exten => 7611,n,Dial(SIP/1200#JVXML97,,XgF(default^7611^14))
exten => 7611,n,Set(GLOBAL(last_timestamp_7611)=${STRFTIME(${EPOCH},GMT+1,%s)})
exten => 7611,n,Hangup()
exten => 7611,n,Ringing()
exten => 7611,n,Wait(2)
exten => 7611,n,Playback(/var/lib/asterisk/sounds/custom/busy_later)
exten => 7611,n,Wait(1)
exten => 7611,n,Hangup()
I understand that the "record" command has a "k" parameter which keeps the recorded file upon hangup, but I'm not able to find any similar functionality with the Monitor command. (I'd use "record", but I'd like to record the full call (duplex) and do it automatically, without any user input requirement).
Thanks!
Use MixMonitor command.
Check you have permissions needed for write files.
Check debug if you unsure.
PS. using diaplan WITH n, without labels and goto to EXACT priority is very bad practice, can result hard-catchable bugs. Using GLOBAL variables without need also not so nice idea.
I'm a beginner on Asterisk, but already have my PBX working connected to the PSTN. The issue I'm having is that I have this rule
exten => _*X.*,1,Log(DEBUG, Calling through provider one to ${EXTEN:1:-1})
same => n,Dial(SIP/${EXTEN:1:-1}#oneProvider,60)
There are no other extensions that start with *. What I want to achieve is to dial out as soon as the second * is pressed (and there's nothing the user can press to go to a valid extension), right now it waits a few seconds and dial. I also tried adding ! at the end of the extension, but nothing changed.
Am I missing something? Is this feasible?
Thanks!
This task is not doable in current asterisk.
It will not work beacuase * matched .(dot) in your dialplan.
Except dialplan like this(very ugly one beacuase it will go dialplan for every new digit)
[originalcontext]
exten => *,1,Goto(collect_number,s,1)
[collect_number]
exten => s,1,WaitExten(); wait for single digit
exten => *,1,Set(stars=${stars}*);save stars
exten => *,2,GotoIF($[ "${stars}" == "**" ]?dial,1); if 2 star already,go dial.
exten =>_X,1,Set(digits=${digits}${EXTEN});save digits
exten => _.,3,WaitExten(); wait enother input;
exten => _.,4,Goto(dial,1); go dial if no new digits
exten => dial,1,Dial(SIP/${digits}#oneProvider,60)
Correct solution - use Read application and ask user use # to end number instead of *.
You also can try dialplan like this:
exten => _*X*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXX*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXXX*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXXXX*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXXXXX*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXXXXXX*!,1,Goto(dial,${EXTEN:1:-1},1)
exten => _*XXXXXXXX*!,1,Goto(dial,${EXTEN:1:-1},1); continue upto max number length
[dial]
exten =>_.,1,Dial(SIP/${EXTEN}#oneProvider,,);
But i not fully sure that will work. If works, will be less load(but more lines)