search for: asyncagi

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten => _1.,n,Set(MEETME_RECORDINGFILE...

...eject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17, 2016 at 6:26 AM, Amit Patkar <amit at avhan.com> wrote: >> Hi >> >> Is there any way to detect inactivity on channel when AsyncAGI is >used? >> I want to detect whether application handling calls using AMI & AGI >has >> stopped responding. > >What do you mean by "stopped responding"? > >> Alternatively, how can dialplan check if there is any AMI user >connected and >> dec...

Hi Is there any way to detect inactivity on channel when AsyncAGI is used? I want to detect whether application handling calls using AMI & AGI has stopped responding. Alternatively, how can dialplan check if there is any AMI user connected and decide dial plan execution? Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML at...

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the call, everything works fine. I am notified the agiexec completed successfully and gi...

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug log (logging set t...

...'t been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The command "asyncagi break" does stop ongoing playing but also breaks the async agi control. I only want the first. Thanks in advance, /Henrik

We have a customer who wants us to record anywhere from 2-4 participants on a call in stereo (as opposed to mono) quality audio. Some background.. We are using asterisk 16.6.1 We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties together. I believe recording in the various file formats (based on extension), it's always recording in mono quality. My one thought is to transition to using ARI Bridge (instead of ConfBridge) and streamed audio using ExternalMedia. Then have a media serv...

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new offer by Adhearsion, which then drops the call on trying to re-bridge the two...

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the Redirect while a playback, the Redirect fails disconnecting the call: http://docs.google.com/Doc?id=ahfnfrcrh3rr_31g...

Some background... We use AMI and AsyncAGI to be able to receive events about calls (and other Asterisk details) and control it from our application. Works great and have about 100 sites (some newer, some older) without issues. I was notified this morning about a customer who had something strange happen and I can't explain it. Aster...

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications to execute, like Playback, Read, etc. This only applies to originating a call from an external application by using the AMI Manager and the Originate action. I need to know the following: 1) Does the Originate action support multiple

...k agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the originate, redirect and hangup AMI commands plus the stream file AGI command (AsyncAGI patch will be required). The application will make outbound calls, then they will be redirected on the fly to dynamically defined meetme rooms, then the application will call extensions (registered endpoints) where it will know there are available agents in order to redirect them to the previous m...

...response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338 CommandID: C44415 Command: asyncagi break Asterisk processes it and the Asterisk log shows it sends the following for the AsyncAGIEXEC event... This is coming directly from the Asterisk log so it's clearly Asterisk filling the Result with what appears to be random characters. [03/21 15:44:27.793] DEBUG[1213] manager.c: Examinin...

Now that everything seems to rock I've hit the next hurdle. In my extensions.conf I have the extension: [agi-async] exten => _01XXXX,1,Agi(agi:async) and I can see that the context is "hit" when dialing into *. However my java app that's supposed to receive async agi events get no such events at all, but it does receive other manager API events. * version is 1.6.1.4

...itoring Resource) == WebSocket registered sub-protocol 'echo' res_http_websocket.so => (HTTP WebSocket Support) res_crypto.so => (Cryptographic Digital Signatures) res_speech.so => (Generic Speech Recognition API) == AGI Command 'answer' registered == AGI Command 'asyncagi break' registered == AGI Command 'channel status' registered == AGI Command 'database del' registered == AGI Command 'database deltree' registered == AGI Command 'database get' registered == AGI Command 'database put' registered == AGI Command...

Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone using the desk phone that sends a REFER request. When leg B is hung up as part of t...