similar to: Do AMI Events have timestamps?

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will

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

Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is

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

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event 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

Hi all. I'm trying to write an AMI handler to take care of cases where a user uses the feature code to start recording a call. When a user starts a recording, my AMI listener receives this event: $VAR1 = { 'Server' => 'example.com', 'Event' => 'MonitorStart', 'Uniqueid' => 'server-1374906101.132305',

HiCould someone please help me how to set Custom generated UUID in Originate action in AMI ? RegardsBabak -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150508/528d5ff1/attachment.html>

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:

Hi, I'm stucked in situation, and look for a work around if possible in Asterisk. I have a dialplan, [default] exten => 111222,n,Set(fu_callerid=141688xyxzz) exten => _X.,n,NoOp(Callerid ${fu_callerid}) exten => _X.,n,wait(2) exten => _X.,n,Answer() ? When, ?Answer Application is called AMI Event is triggered like this.. ? ? ? ? ? 'Event' => 'Newexten', ? ? ? ?

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using

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

Hi all, I'm required to send a dialplan variable with every AMI event triggered for the duration of the call. For example; ... exten => s,n,Set(MyVar=${ODBC_GetSomething(${EXTEN})) ... so can I have this Variable MyVar attached in all AMI events for this call ? I can understand that untill this variable has not been set some value it may even be empty but as soon as its set I expect some

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There

On Tue, Aug 8, 2023 at 12:44 PM TTT <lists at telium.io> wrote: > I'm looking at an old app I wrote that upon AMI login would subscribe to > events as follows: > > > > Action: Login > > ActionID: myid > > Username: myun > > Secret: mypw > > Events: call, system, security > > > > I noticed this old code isn't working, and I

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/9999-00000029 Channel2:

Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context? Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

For those of you who actually process SIP MESSAGE requests... Do you use any of the AMI events generated by the "Message/ast_msg_queue" channel? We want to change that channel to an "internal" channel that doesn't generate AMI events (for performance reasons) but we need to know if anyone's using them first. Thanks! -- George Joseph Asterisk Software Developer

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event: