Set up Asterisk 13.1 on Ubuntu 16.04,Calls are Working fine as well as conference calls.However I have two Question regarding the recording of calls in asterisk.
This is the extensions.conf.
[default]
exten => 1000,1,MixMonitor(${CALLERID(num)}-${STRFTIME(|EST5EDT|%m%d%Y-%H%M%S)}.gsm)
same => n,Dial(SIP/1000)
exten => 1001,1,MixMonitor(${CALLERID(num)}-${STRFTIME(|EST5EDT|%m%d%Y-%H%M%S)}.gsm)
same => n,Dial(SIP/1001)
exten => 100,1,Answer()
exten => 100,n,Authenticate(234)
same => n,MixMonitor(${CALLERID(num)}-${STRFTIME(|EST5EDT|%m%d%Y-%H%M%S)}.gsm)
same => n,ConfBridge(1234)
And this is confbridge.conf
conf => 1234
[1234]
record_conference = yes
1)Is it necessary to define MixMonitor for every call(As I have done above for user 1000 and 1001) ,or is there a possibility where you can define it once and all calls in your asterisk server get recorded.
2)When the Conference call is recorded ,suppose if there are two users in the conf room,there are two audio files generated instead of one,What changes do I make so theres is only one recorded audio file of the full conference?
In asterisk it is nessesary do any action you want DO.
In your case you can use mixmonitor command in every dialplan, use pattern to do that command for large list, use external macro or gosub function and call it for calls you want be recorded.
For conference you can start enother call with mixmon(third call) using Originate or you can use internal recording.
For ANY record you can have Monitor(two files for each side of call) or mixmonitor(same, but 1st party in left channel, 2nd party in right channel, so file is single).
So if you have 2 files, you just have non-mixed variant. Consult conference room options for mixed version.
Related
As a beginner that just installed his first asterix server I came across a small issue. i want to check if a number that i dial is busy or not before actually dialing it.
So for example I would call 0904 => number busy => redirect to 0905
However I don't want it to dial 0904 first if its busy but just instantly redirect it to 0905 instead.
I have a very basic setup with just 3 users. 2 of them are in a queue "support" one of them is in the queue "admin"
I have seen and read a little about checking if a channel is available but i didn't fully understand that and I'm not sure if it could be done easier.
Here is my simple dialplan, if i call *12 my simple python script will just execute the Dial command to the 0904.
exten => *12,1,Answer()
same => n,set(PHONE_EXTEN=0904)
same => n,AGI(test-agi.py,${PHONE_EXTEN})
same => n,Hangup()
There is no way know if external number is busy.
If number is your extension, you can count calls using GROUP/GROUP_COUNT or you can check extensions states and hints, like described in docs
https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints
https://wiki.asterisk.org/wiki/display/AST/Device+State
I'm trying to make my Asterisk run a script whenever a certain phone (Sip Phone) answers or makes an outgoing call and when it ends that call. The purpose of that is to automatically mute my TV when a call is done with the phone in the same room.
I've tried using the System command but that doesn't get information about the phone that answered. As I am having multiple phones ringing, I can not distinguish if it was indeed the phone next to the TV or a completely different phone. Same problem I am having for outgoing calls.
Is there a different approach to this? I also tried using the AMI but I haven't found a way to get the status of a specific peer (Offline, Online, In Call, lagged).
Lets say you have ext sip/111
Support extension
[from-internal-very-special]
exten => 111,1,UserEvent(TVRingStarted)
same => n,Set(CHANNEL(hangup_handler_push)=from-internal-very-special,ends,1)
same => n,Dial(SIP/111,,U(from-internal-very-special,answer,1))
exten =>answer,1,UserEvent(TVRingAnswer)
same => n,Set(GOSUB_RESULT=);we accept call
same => n,Return;continue
exten => ends,1,Noop(end of call)
same => n,UserEvent(TVRingEnds)
same => n,Return;
After that call extension as Local/111#from-internal-very-special/n instead of SIP/111
You will have user events you can parse via AMI. Or you can replace that by System calls as you wish.
I have set up basic call monitoring for individual extensions in my Asterisk setup. This is what I have done for recording individual calls:
[macro-automon]
exten => s,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
same => n,MixMonitor(${MONITOR_FILENAME}.wav,b)
[LocalSets]
exten => 101,1,noop(dialing 101)
same => n,Macro(automon)
same => n,Dial(${EXT_TESTONE},20,m) ; Replace 0000FFFF0001 with your device name
same => n,Playback(vm-nobodyavail) ; Play "no one's available"
same => n,Hangup()
This works and a call which is picked up by extension 101 is saved under /var/spool/asterisk/monitor
Now, I have setup a couple of test queues called sales and support, like so:
[Queues]
exten => 7001,1,Verbose(2,${CALLERID(all)} entering the support queue)
same => n,Queue(support)
same => n,Hangup()
exten => 7002,1,Verbose(2,${CALLERID(all)} entering the sales queue)
same => n,Queue(sales)
same => n,Hangup()
I have also added the user called 0000FFFF0001 under the sales queue. Thus, when I dial 7002, it rings at ext.101 and the call can be picked up. However, the call monitoring doesn't take place.
How could I enable monitoring for calls which come to 101 via the queue?
Call queue have own monitoring flags.
However you always can do it via local channel. Instead of do queue do
Dial(Local/s#toqueuesales/n)
and put in extensions.conf
[toqueuesales]
exten => s,1,Answer
exten => s,2,Queue(sales)
That way it have work anyway(but will show 2 channels and 2 cdrs)
You can also add local channels to the Queue instead of extensions. Technically when the queue is ringing your agent, it's ringing SIP/101 (or whatever the agent extension is). If you added Local/101#LocalSets as an agent in your queue config, the dialplan you provided would work. You'd need to tweak the default log in/out macros, but this is the easiest way to get your dialplan to work.
See the docs for a comprehensive guide to adding queue members.
Couldn't find a specific answer for this. I'm a newbie to asterisk and AMI. I need to auto generate calls using asterisk and pass parameters to an AGI program. Using a call file seems to generate the call first which is not wanted. So, how do I use asterisk AMI API (PHP) to execute a dialplan with AGI in it, by passing all parameters to it? So, the AGI will take over and make the call.
I think in your case, using call files would actually be simpler. Here's why:
The AMI requires you to write networked code, which (if you're a beginner) will be a lot more tricky the building simple text files (Call Files).
Call Files allow you to pass variables to Asterisk that you can use in your dialplan code.
Call Files are extremely simple.
Below is a full example of a simple way to do it using call files. I've tried my best to explain it in the associated comments.
Firstly, let's assume you have some dialplan code that uses variables, and calls an AGI script (which is what I assume you're doing based on your question). That means you'll have code in your extensions.conf file that looks something like:
[test_stuff]
exten => s,1,NoOp(starting test!)
exten => s,n,NoOp(my variable's value is: ${somevar})
exten => s,n,NoOp(my other variable's value is: ${some_other_var})
exten => s,n,AGI(/path/to/my/script.sh,${somevar})
exten => s,n,NoOp(i just ran an AGI script and passed it a command line argument!)
exten => s,n,Hangup()
Below is a call file that will:
Need to be created in some temporary directory (maybe /tmp/).
Once it has been saved, you can run it by moving it to /var/spool/asterisk/outgoing/ (eg: mv /tmp/blah.call /var/spool/asterisk/outgoing/).
The call file will the immediately dial outbound to the phone number 818-222-3333.
Once the person at 818-222-3333 picks up the call, Asterisk will immediately start executing your [test_stuff] dialplan code, and will have the variables set in your call file available to it.
Call file:
Channel: SIP/trunkname/18182223333
Context: test_stuff
Extension: s
Priority: 1
Set: somevar=hithere
Set: some_other_var=woot
Hope that helps!
Your problem can be solved with the help of local channel for example
In call file use Local/1812222222#test_stuff as channels while using following dialplan
[test_stuff]
exten => _X.,1,NoOp(starting test!)
exten => _X.,n,Set(phone=${EXTEN})
exten => _X.,n,AGI(/path/to/my/billing,${phone},${other_variables_account_etc..})
exten => _X.,n,Dial(SIP/trunkName/${phone})
You can use this method with both interfaces AMI or .call file
I have an asterisk server. I use the server to connect an incoming call to another extension based on a few key presses. There is a certain time lag (after the key/extension press and before the call connects). How can I play a small music file (of my choice) in this period? There are some constraints that come to my mind:
The music should play only as long as the call does not connect. So, the method used should be a non-locking one.
Any help on this is most welcome.
Thanks,
Sriram
Use the m flag to the Dial application, to play music on hold while the call is connecting.
exten => 9000,1,Noop
exten => 9000,n,Answer
exten => 9000,n,Dial(SIP/device,0,m)
I think Background is your friend (http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGround)