Im looking at an old app I wrote that upon AMI login would subscribe to events as follows: Action: LoginActionID: myidUsername: myunSecret: mypwEvents: call, system, security I noticed this old code isnt working, and I *think* that the events parame..
Author : TTT
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Is there an AMI command/action which reports back the username used to authenticate (to the AMI), and the permissions in effect for ..
I want to use CoreSettings via the AMI.I checked the documentation for the action (command) and it doesn’t list any required permissions:https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_CoreSettings I tried using the CLI “mana..
Is there a web page that lists the AMI versions mapped to Asterisk versions? I noticed that the AMI version increased quickly to 9.0.0.Will the AMIversion increase with each Asterisk version increase in the future? T..
The following AMI command works well for all of the information I want:action: Getvar actionid: act1channel: PJSIP/Twilio-NA-W-3-In-00000028Variable: CHANNEL(pjsip,XXXX)Where XXXX can be one of the many available items.However, when I create a call f..
The AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13).I thou..
I have an AGI script written in PHP that worked great with Asterisk 13.Im porting it to an Asterisk 20 site and have a strange problem.I tried running the script from the command line and it works fine; I see the script commands written to stdout likeVERB..
I am connecting to the ARI with subscribe all, so I can see channels being created.I now want to extract a variety of header variables (at the moment the from and to tag).I tried to read them from the ARI but Asterisk refuses since the channel is ..
Im learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side.I noticed that the Asterisk wiki menti..
Im looking at using Asterisk 20 with WebRTC clients (sipjs).I know the media runs over TCP, but what about the signaling? I read something about signaling over UDP was proposed as part of a webrtc standard, but cant find if that was ever ratified..