similar to: Hanging up all call on a device via CLI/AMI/AGI

Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText" Channel: SIP/600" Body: This is a test. Message: This is a test. Extension: 600";

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is

Hello, I'm pleased to announce the first release of Obelus, a MIT-licensed Python library to interact with Asterisk using the AMI and AGI protocols. Compared to existing libraries, Obelus is framework- and programming-style-agnostic, and compatible with Python 3 as well as Python 2. It also has an integrated test suite. This is version 0.1, and as such some APIs are a bit draftish and not

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

When using call files I see Dial command when on the CLI however when using the AMI I dont see the originate Dial. Is there a logging level that will show the originate dial on the CLI? If there isnt currently a log state for that shouldnt it do that just like the call files do? My logger.conf has: ; debug => debug console => notice,warning,error,dtmf ;console =>

Just run ‘core show calls’ as a command from the AMI, and parse the results. I don’t think there is an equivalent pure AMI command. From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent: Sunday, June 14, 2020 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re:

Hi, I've defined a feature using a macro in features.conf : special => #2,peer,Macro,special Everything is working if the user use the phone key. But i would like to call the feature (or the Macro on the peer channel) from AMI or CLI. First i thought i would be simple, but i did not find any solution. Does someone has an idea ? Thank you very much. Matthieu NICAISE Responsable

Hi, is there a way to put a conference participant in talk only mode (not listening) using CLI or AMI like mute/unmute ? MeetMe in Asterisk 1.8 Thanks for any hint. -- Daniel

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

Hi, Can anyone explain the difference between Asterisk Gateway Interface and Asterisk Management Interface ? Is it correct to consider AGI scope to focus on call handling and AMI scope to anything which can be done with Asterisk froma loading new modules to originating calls ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone

I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me this. Any ideas?!? In case it helps and you're wondering why... I need to ensure

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I

On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: > Paul Albrecht wrote: >> Really? Shouldn?t something this major affecting the entire Asterisk >> community get discussed on the lists? Any idea what Leif is talking >> about when he says the community is in transition, moving from dial >> plan model to external control. > > It was

Way back in the mists of time, I built my asterisk installation with SNMP support. It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that moment in time. That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.

Dear all i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this: 1) receive a call and put it on-hold in a queue (OK) 2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK) 3) when the remote number answer, trigger an

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir