I have followed the instructions in this thread: Asterisk AMI - pickup call. However, I am still unable to answer calls via AMI. I can make the call to the extension, but corresponding phone for that extension doesn't ring. I can then run the AMI command to answer that call, it does answer, but obviously there isn't any actual response.
Dialplan (testing with extension 116):
exten => 116,1,AGI(agi:async)
Any ideas what I am doing wrong here?
Use
exten => 116,1,Answer
exten => 116,2,AGI(agi:async)
or use Answer action via ami.
http://www.voip-info.org/wiki/view/Asterisk+manager+API
You need listen event, when see agi-async event issue Answer on same channel. For example you can do playback command with answer.
Very likly you need start with AGI interface, which is much more simpler for understanding. Not use agi:async, it require understanding of asterisk internals.
Found the Answer. For those wanting to answer with API Manager you can use the following -
http://ip-address:port/asterisk/rawman?action=Originate&Channel=Local/(exten you want to answer with)#(context)&Application=Exec&Data=Pickup((exten you want to answer with)#PICKUPMARK)
I am of course using http to send my requests, if you are using something else you will need to change the format.
Or you can use the bridge command. Two channels will be created when you make a call from one extension to the other, bridge those channels and you will have better overall functionality than pickup.
Related
Running Asterisk 13.12.1, FreePBX 13.0.192.19.
We had to install new server and since we previously used much older asterisk, there were some fixes applied. We DIDN'T update previous, but we made clean install, just copied dialplans, sip config etc.
The problem is that we are now having multiple CDR records per call. We previously had NOCDR lines for local contexts, and we tried I have tried to change those to exten => _X!,1,Set(CDR_PROP(disable)=1) but that didn't work at all.
Here is the example:
[main context]
exten => remote-mon-1,1,Dial(SIP/lokal300&SIP/lokal301&Local/06xxxxxx#shift-remote-1&Local/06xxxxxx#shift-remote-2&Local/06xxxxxx#shift-remote-3&Local/06xxxxxx#shift-remote-4&Local/06xxxxxx#shift-remote-5&Local/06xxxxxx#shift-remote-6,,m(remote)M(whoanswered,remote))
[shift-remote-1]
exten => _X!,1,Set(CDR_PROP(disable)=1)
exten => _X!,n,Dial(SIP/gsm10/${EXTEN},540)
Basically what the above does is calling two local phones (300 and 301) as well as multiple (6) remote mobile phones via gsm gateway.
1) So CDR PROP is completely ignored (I think someone said how its not working with Local context but I need confirmation). How can I fix it?
2) Any other ideas how to avoid creating multiple CDR record for each call?
Thank you!
Update: As this was flagged as a duplicate of Asterisk 13.4 cdr engine is creating 2 records per call , I need to explain that In that question the solution is applying unofficial patch, which is not something we want to do. I was looking for official approved way on why CDR_PROP is not working correctly. Furthermore (I just checked) the link to patch in that post is not working, as site is unreachable. One more reason to not flag this as duplicate.
1) use NoCDR, not forget add '/n' to local channels
pro-sip*CLI> core show application NOCDR
-= Info about application 'NoCDR' =-
[Synopsis]
Tell Asterisk to not maintain a CDR for this channel.
[Description]
This application will tell Asterisk not to maintain a CDR for the current
channel. This does *NOT* mean that information is not tracked; rather, if the
channel is hung up no CDRs will be created for that channel.
If a subsequent call to ResetCDR occurs, all non-finalized CDRs created for the
channel will be enabled.
NOTE: This application is deprecated. Please use the CDR_PROP function to
disable CDRs on a channel.
[Syntax]
NoCDR()
[Arguments]
Not available
[See Also]
ResetCDR(), CDR_PROP
2) Read /etc/asterisk/cdr.conf params.
I'm trying to find answer how to make Asterisk execute some command (my script) after confbridge's recording is finished
There is the next info in confbridge.conf:
record_conference=yes
Records the conference call starting when the first user enters the
room, and ending when the last user exits the room.
It records file well but I want it sending wav file via email.
Could anybody help me?
My config now looks like this (if it's necessary):
exten => 333,1,ConfBridge(100010,100010_bridge_profile,100010_user_profile)
Dialplan scripting is limited to events relating to each call channel. To get event info for other parts of asterisk (such as the ConfBridge application) you should hook into the Asterisk Manager Interface (AMI).
There are many libraries already created to make working with the AMI easier. (That site may be outdated. Refer to the official Asterisk Wiki whenever possible.)
The AMI event you're interested in is "ConfBridgeEnd". Docs here.
You can use h-extension after confbridge, in which you have check if confbridge still active(last user).
If yes, run your script via System call.
Scenario:
Asterisk receives incoming call to a DID number
Asterisk forwards incoming call to a mobile number using PSTN termination.
Call is answered on a mobile
Question:
Is there a possibility to transfer call to a different extension during the call ?
yes,but require guru level.
it can be done by writing special application or using conference or using AMI.
probably easy way is conference
Yes. This is very straight forward.
Presuming your DID is 1-234-567-8900 and your moble is 1-987-654-3210:
exten => _12345678900,1,NoOp(Call for DID to Mobile)
same => n, Dial(DAHDI/g1/19876543210,30,t)
same => n, HangUp()
The dial option "t" allows "...the called user to transfer the call by hitting the blind xfer keys (features.conf)"
Further reading: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
I want to pickup call in Asterisk using AMI. I can originate call, but totally don't know, how to answer the phone...
Script for calling:
#login
sock = socket.socket(af, socktype, proto)
sock.connect(sockaddr)
sock.send('Action: login\r\n')
sock.send('Events: off\r\n')
sock.send('Username: '+str(ast_server.login)+'\r\n')
sock.send('Secret: '+str(ast_server.password)+'\r\n\r\n')
#originate call
sock.send('Action: originate\r\n')
sock.send('Channel: ' + str(user.asterisk_chan_type) + '/' + str(user.internal_number)+'\r\n')
sock.send('Timeout: '+str(ast_server.wait_time*1000)+'\r\n')
sock.send('CallerId: '+str(user.callerid)+'\r\n')
sock.send('Exten: '+str(ast_number)+'\r\n')
sock.send('Context: '+str(ast_server.context)+'\r\n')
if ast_server.alert_info and user.asterisk_chan_type == 'SIP':
sock.send('Variable: SIPAddHeader=Alert-Info: '+str(ast_server.alert_info)+'\r\n')
sock.send('Priority: '+str(ast_server.extension_priority)+'\r\n\r\n')
#logout
sock.send('Action: Logoff\r\n\r\n')
time.sleep(1)
sock.close()
I need something similar, but for answering calls.
Can't find any useful command in *CLI> manager show command
Halp me, plox
You can't answer a call directly via AMI. This is because a new call will "arrive" at the given context/priority/extension configured in the dialplan (or it will be rejected if cant find one that applies). So whatever happens with that call will start at the given context/priority/extension in the dialplan.
If you want to handle calls via AMI, try using asynchronous AGI, like this:
exten => _X.,1,AGI(agi:async)
This will handle all calls to any extension that has at least 1 digit, by issuing an event (AsyncAGI) that you can handle with your AMI client.
Then, from your AMI client, you can send AGIAction's, like:
Action: AGI
Channel: SIP/adevice
Command: ANSWER
CommandID: MyCommandID
This will effectively allow you to run AGI commands (and handle a call like you would normally do in any AGI script) from your AMI client.
Hope it helps!
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)