All channels on Asterisk configured as DAHDI channels.After customer make payment I want to transfer the customer to the representative who interact customer before.
I try to make it by Dial() command. This is the dialplan
exten => s,1,Set(TRFNUM=${CALLERID(num)})
exten => s,2,Set(TRFNAME=${CALLERID(name)})
exten => s,3,AGI(agi://192.168.7.20/customivr)
exten => s,4,Dial(DAHDI/1/${TRFNUM}&DAHDI/2/${TRFNUM}&DAHDI/3/${TRFNUM}&DAHDI/4/${TRFNUM}&DAHDI/5/${TRFNUM}&DAHDI/6/${TRFNUM}&DAHDI/7/${TRFNUM}&DAHDI/8/${TRFNUM},30)
exten => s,5,Hangup
For example: Call comes to DAHDI/1 after the payment DAHDI/1 dial all channels one them Answer the others Hangup. DAHDI/1 bridge call by with DAHDI/2. However, although Customer and representative close phones, Channels do not Hangup. They stay Busy.
Where do i make mistake. I should hangup call channels or find another way to transfer.
It seems to be configured correctly,
I think your AGI script hangup the call when he finishes his work,
It can happen if you have $agi>hangup in the end,
or if you make any outputs in the scripts (echo, print_r, etc...),
even empty spaces output can cause this behavior,
another thing you can try is make the Dial command from the agi itself using:
agi->exec("Dial","options");
Related
[play-context]
exten => _X.,1,Answer()
exten => _X.,n,NoOp('Exten:')
exten => _X.,n,NoOp(${EXTEN})
exten => _X.,n,ConfBridge(dconf-${EXTEN}-${STRFTIME(${EPOCH},,%d.%m.%Y-%H:%M:%S)})
exten => _X.,n,Originate(SIP/5678,exten,conferences,100,1)
[conferences]
exten => _X.,1,NoOp(${EXTEN})
exten => _X.,n,ConfBridge(dconf-${EXTEN}-${STRFTIME(${EPOCH},,%d.%m.%Y-%H:%M:%S)})
when i create a new confBridge conference, i want to join some users in it. How i can do it automaticaly? I try this config, but it dont worked. Before this line
exten => _X.,n,Originate(SIP/5678,exten,conferences,100,1)
just does not reach. I do not understand why.
Please, help.
Your channel (your audio when you call the number of the conference) will enter in the conference when it comes to the Confbridge statement in your dialplan. The problem with Confbridge is that once the dialplan has arrived to the Confbridge statement, you will remain there until you exit from it. If you wanted to find there some friends you should have made some previous tasks
Check if the conference is stablished (if there are already other parties in them, see function CONFBRIDGE_INFO) and, if it is, simply add yourself to it
If not, stablish the conference inviting the guys you want to and add yourself to the conference
So, dialplan for conferences is usually not trivial. Let us assume that you do not want to make the first check. So, you are including a guy as you stablish the conference. In order to simplify and to avoid some headaches I am going to use a name for the conference that do not depend on the second at which the dialplan is read (depending on how do you mantain the dialplan your conference could be different from the conference at which you invite your mate), let us suppose that the conference has a name that only depends on the EXTEN dialed. You should do something similar to:
exten => _X,1,Answer()
...
same => n,Originate(SIP/5678,app,ConfBridge,"${EXTEN},rest-of-parameters-for-5678")
... (Check ORIGINATE_STATUS if you do really need your friend)
same => n,ConfBridge(${EXTEN},rest-of-parameters-for-youself)
Here I have used originate with the parameter app. You can, of course, follow the hint of #arheops and use exten with a Local channel. But if you do so, be carefull with the seconds ;).
If you want to stablish automatically the conferences as asterisk starts, consider including some originate statements in cli.conf.
When you do
Confbridge
it now in bridge and not go next extension until you exit bridge.
So you have do Originate BEFORE that or do it in other thread(via Local channel) in parallel.
I'm using a cloud-based Asterisk server as my PBX. At my current location, the Internet is rather shaky, but cell phones are reliable and commonplace. However, international cell calls are expensive, VOIP calls are much cheaper.
So, I came up with a script in Asterisk which dials my local cell phone:
exten => _abcd.,1,NoOp(-- Making outbound call to number ${EXTEN:4} --)
same => n,Answer()
same => n,Wait(1)
same => n,Originate(SIP/+86[my_cell_no]#[voip_provider],exten,incoming_remote,##${EXTEN:4})
same => n,Hangup()
Let's say I want to call a UK mobile number, +4477something. I would use my softphone to dial abcd+4477something. The script above runs, makes a call into my local cell phone. As soon as I answer, it jumps into another extension _##. which dials the outbound number, and connects the two together.
It works perfectly. However, whilst I'm waiting for the local cell to connect, I've got silence on the line. I'd quite like to play music... but I can't use the MusicOnHold() application, because it just sits there & does nothing until I hang up!
I can't add any "DIAL" style commands (i.e. "m") to the Originate command because it doesn't support them.
Is there any known way of playing (one of) the MusicOnHold channels asynchronously whilst the rest of my dialplan gets on with it?
Would the AGI command SET MUSIC do what I wanted?
e.g.
exten => _abcd.,1,NoOp(-- Making outbound call to number ${EXTEN:4} --)
same => n,AGI(turn_music_on.sh)
same => n,Answer()
.....etc.
I'm using Asterisk 1.8, if a newer version fixes/changes the MusicOnHold behaviour, then that will be the accepted answer (but the documentation seems to suggest it's the same).
You can call to Local channel(dialplan). After that in dialplan you can use m of dial command.
https://www.voip-info.org/wiki/view/Asterisk+local+channels
same => n,Originate(Local/[my_cell_no]#out/n,exten,incoming_remote,##${EXTEN:4})
[out]
exten => _X.,1,Dial(SIP/+86${EXTEN}#[voip_provider],,m)
I'm trying to execute an extension from the command line (via asterisk -rx "command") on a context that makes a AGI based query to determine which extension needs to be dialed (these extensions are updated on the DB).
It's something like this:
[autodialer]
exten => 2,1,Answer()
exten => 2,n,AGI(database_query.php); Makes a database query and generates vars
exten => 2,n,Set(CALLERID(name)=${db_customer_name}); Sets callerid from DB data
exten => 2,n,Dial(SIP/${db_customer_extension}); Also, extensions are stored on DB
exten => 2,n,Playback(custom/important_message)
exten => 2,n,SayDigits(${important_numbers}); The message, stored on DB too.
exten => h,1,Hangup()
Here, I need that context executed from command line, without having to dial it from any extension (it is supposed to be executed with a crontab every X time).
I tried with originate command, but I think I misunderstood the command syntax and didn't work.
I think that it should be something like: asterisk -rx "channel originate 2#autodialer" and then Asterisk executes that context and we're all happy with our important numbers.
I know that's not the right syntax, just trying to explain how I imagine it could work.
Thanks for your help.
There are no way do originate only one leg. You have supply second argument(other channel dest)
if you not need other channel, create context like this
[wait]
exten =>s,1,Wait(10000)
and use
asterisk -rx "channel originate 2#autodialer s#wait"
Read this article:
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
NOTE: it is not recommended do diallout apps for people with less then 5 years dedicated asterisk experience. If you want one, use vicidial.org or other dialler.
I am setting up a new Asterisk system at a hosted Asterisk provider using 1.4.38, moving from a self-hosted version 1.2. Whereas before I could do:
[incoming]
exten => _1NXXNXXXXXX,1,DoSomething
Now, it appears the incoming context will only get properly called if it is:
[incoming]
exten => s,1,DoSomething
How do I determine what number was dialed in this scenario? We have dozens of numbers and I need to be able to know which was called to route correctly...
Thanks,
Ben
You can use a pattern. I have this line in a running asterisk instance:
exten => 5858876463,1,Set(__FROM_DID=${EXTEN})
However, to answer your question, you can use the ${EXTEN} variable to decide what number was called. E.g.
exten => s,1,SomeScript(${EXTEN})
We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100).
Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside).
Is there a simple way to do this?
Are you running plain Asterisk? If so you can modify your dial plan to start 'monitoring' the channel, which will record the call.
The monitor command's documentation: http://www.voip-info.org/wiki/view/Asterisk+cmd+monitor
Just for the sake of completion, here's the documentation:
[root#localhost ~]# asterisk -rx 'core show application monitor'
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a channel
[Description]
Monitor([file_format[:urlbase],[fname_base],[options]]):
Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until the channel hangs up or
monitoring is stopped by the StopMonitor application.
file_format optional, if not set, defaults to "wav"
fname_base if set, changes the filename used to the one specified.
options:
m - when the recording ends mix the two leg files into one and
delete the two leg files. If the variable MONITOR_EXEC is set, the
application referenced in it will be executed instead of
soxmix and the raw leg files will NOT be deleted automatically.
soxmix or MONITOR_EXEC is handed 3 arguments, the two leg files
and a target mixed file name which is the same as the leg file names
only without the in/out designator.
If MONITOR_EXEC_ARGS is set, the contents will be passed on as
additional arguments to MONITOR_EXEC
Both MONITOR_EXEC and the Mix flag can be set from the
administrator interface
b - Don't begin recording unless a call is bridged to another channel
i - Skip recording of input stream (disables m option)
o - Skip recording of output stream (disables m option)
By default, files are stored to /var/spool/asterisk/monitor/.
Returns -1 if monitor files can't be opened or if the channel is already
monitored, otherwise 0.
And here's a sample way you can use it:
; This fake context records all outgoing calls to /var/spool/asterisk/monitor in wav format.
[fake-outgoing-context]
exten => s,1,Answer()
exten => s,n,Monitor(wav,,b)
exten => s,n,Dial(DAHDI/g0/${EXTEN})
exten => s,n,Hangup()
Obviously you'd have to make changes to my code, but hopefully that gives you a good idea.
A real life example is
exten => _87X,1,NoOp()
exten => _87X,n,MixMonitor(${UNIQUEID}.wav,ab)
exten => _87X,n,Dial(SIP/${EXTEN},45)
exten => _87X,n,StopMixMonitor()
exten => _87X,n,Hangup()
It's good practise to always have NoOp - the first rule must start with 1, this way you can interchange the rules with the n step any way you want.
It's always best to use MixMonitor as opposed to Monitor - Monitor only records inbound or outbound audio - MixMonitor uses both.
Also wav is quite a good choice as a format - I also use a script to transform the wav files to OGG at the end of the day - the best compromise between size / quality and licensing issues.
With regards to the arguments
a is append
b is bridge (good for production - it will only record when the call is answered - not good for debugging)
With regards to StopMixMonitor(), I'm just being thorough, but for examples there are cases in which you would like to stop the recording, for example:
...
exten => _39[5-9],n,Dial(SIP/${EXTEN},45)
exten => _39[5-9],n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavailable)
exten => _39[5-9],n(busy),NoOp()
exten => _39[5-9],n,StopMixMonitor()
exten => _39[5-9],n,Voicemail(${EXTEN},u)
exten => _39[5-9],n,Hangup()
exten => _39[5-9],n(unavailble),NoOp()
exten => _39[5-9],n,StopMixMonitor()
exten => _39[5-9],n,Hangup()
...
In this example, you would stop the recording of the voice mail interaction.
Hope this will bring some light on the matter.
Depending on the specifications of your Asterisk box you might find this hack useful too. Create a rather large ramdisk and mount /var/spool/asterisk/monitor to it. That way Asterisk records to memory not disk. Then write a script under cron to move the recordings to permanent storage every 15-30 minutes or so.