similar to: AMI can not receive events unless keyboard input

Hi. I have three servers running corosync and pacemaker, to maintain a floating address between them. This is working fine, and I can, for example, SSH to the floating address and get to whichever server has the address at the time. I am trying to connect to the same server (using the same address) for AMI, and it just isn't working, even though I can connect to the primary address of

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

I was playing with the AMI today. Trying to get a setup that only responds to connections on the SAME box. like 192.168.1.10 So .10 is my server and I only want responses originating from .10 to answer the AMI. I set bindaddr to 192.168.1.10 I left it as 0.0.0.0 I set permit to be 127.0.0.1/255.255.255.0 I tried 192.168.1.10/255.255.255.0 I tried 192.168.1.10/255.255.255.255 No matter what I

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

On Thu, Feb 6, 2020 at 12:34 PM sduthil at wazo.io <sduthil at wazo.io> wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > 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 > >

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ----------- Connection closed by foreign host. root at pbx:/tmp# telnet localhost 5038 Trying 127.0.0.1... Connected to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I made some tests with asterisk-13.2.0 and chan_pjsip this weekend myself, and came to the same conclusion: some peerstatus events are missing (eg. when contacts become unreachable / unavailable, IIRC), and I could not find a way to get contacts status through AMI. It looks a bit similar to issues 23172, 23173: PJSip missing functionalities.

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

Ok – so if I forgot to add “security” to the read= line in manager.conf for this user, will that cause the user to be unable to subscribe to the “security” events upon login? (in other words, although subscribed at login, no security events will be shown to this user) Thanks From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday,

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

Hello all, I've got some PHP code that opens an AMI socket and does a ConfBridgeList for a specific bridge (8888). This all works just fine but I need to filter the information displayed to only CallerIDName so I can see a complete list of names of participants. After days of googling and playing with it, I'm no closer than I was when I started. I'm not at all married to a table.

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, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

Hello, I use UserEvents generated by the Message/ast_message_queue channel with the UserEvent application. Regards Jean Le 29/01/2020 à 20:31, George Joseph a écrit : > 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

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

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

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 Thu, Jan 30, 2020 at 12:18 AM Jean Aunis <jean.aunis at prescom.fr> wrote: > Hello, > > I use UserEvents generated by the Message/ast_message_queue channel with > the UserEvent application. > > Regards > > Jean > Thanks Jean. We're looking at alternatives. > Le 29/01/2020 à 20:31, George Joseph a écrit : > > For those of you who actually

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 everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is