similar to: How run AsyncAGI commands in background

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

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven'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

Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk 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

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject 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,

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 attachment was scrubbed... URL:

I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The

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

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Hi i am experiencing this problem: MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE exten => 9999,1,MeetMe(666|1Arxq) exten => 9998,1,MeetMe(666|1Axq) exten => 9997,1,MeetMe(666|1xq) I make a conference between 3 person dialing A dials 9999 B dials 9998 C dials 9997 all works fine but the datetime won't be updated, it still remain for example 13:40 until i do a

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

An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call:

In Asterisk 1.2.4 is love being able to recording conferences. However, using the default variables, the files are being written to /var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme. If I change MEETME_RECORDINGFILE variable to something different in works, bit I lose the ability to define CONFNO as part of the file name, which is handy when sorting for users to review. I call meetme

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

I am looking to see if anyone has seen this problem before. I am setting the MEETME_RECORDINGFILE variable in a macro, then using the r option with the Page application to record the page. But the page is only recorded to the file specified in MEETME_RECORDINGFILE sometimes... Sometimes it works and sometimes it doesn't. When it doesn't work it places the recorded file in

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

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

Hi, Now and then our SIP phones ring with "asterisk" showing as the caller-ID. Upon picking up the receiver, there is about five seconds of silence and then the channel is closed (hangup). Can anyone offer some insight? Here's relevant snippets from my extensions.conf and Master.csv log: This line shows up in Master.csv:

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. Asterisk 16.23.0 and PJSIP. Call comes into Asterisk. Asterisk

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