similar to: Registration failure event from AMI

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? For anyone else looking...is there a table somewhere online that maps events to their eventmask categories? I checked the asterisk wiki and voip-info but can't find this... -------------- next part

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

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

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

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

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

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, Jan 30, 2020 at 3:18 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. > Do you use any aspects of the channel itself in the user events, or merely the contents of the user event and what you've placed in it? -- Joshua C. Colp Asterisk Technical Lead

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD

After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

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:

Hello, I'm using an AMI Originate command to send a fax. The fax is sent by a script, and I'd like my script to send the fax, wait until it has succeeded or failed, then exit with an appropriate error code (it is driven by a mail system, so the exit code will tell the mail system whether to retry the fax later). The script works great if the fax succeeds, or if the line is busy or

We have an application that plays a variety of sound files on one leg of a call (generated by a call file). We've been told that the party listening to the audio files intermittantly hears "robotic" sounding audio (on/off during the same call). Anyone have ideas on cause? These calls are on an internal network (lots of network bandwidth), and from a server running 99% idle.