similar to: AMI version to Asterisk version mapping

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMITEL123456 1296197222 at entryhome<mailto:1296197222 at entryhome> Ringing

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 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

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

Is there a mapping of AMI versions to Asterisk versions somewhere? For example, Asterisk 1.4 includes AMI version 1.0 (at least that's what I see when I connect to Ast 1.4 via telnet to the AMI port) Also, doe the AMI version changes reflect changes to the AMI commands? If so, is there also a list of what commands changed by AMI version? Thanks -------------- next part -------------- An

?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:

On Fri, Jan 9, 2015 at 5:24 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote: > I'd suggest taking a look at the free edition of SecAst ( > www.generationd.com). It handles these messages perfectly (and can also > use AMI security events) - so you don't need to constantly be updating > fail2ban rules. It's a drop in replacement for fail2ban. > > -M- > >

Asterisk Project Security Advisory - AST-2019-007 Product Asterisk Summary AMI user could execute system commands. Nature of Advisory Remote Code Execution Susceptibility Remote Authenticated Sessions Severity Minor

Asterisk Project Security Advisory - AST-2014-018 Product Asterisk Summary AMI permission escalation through DB dialplan function Nature of Advisory Permission Escalation Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-018 Product Asterisk Summary AMI permission escalation through DB dialplan function Nature of Advisory Permission Escalation Susceptibility Remote

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>

On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI

I've searched through the wiki but I can't find what I need...I'm trying to figure out what the max call duation is. I found references to "show application AbsoluteTimeout" but that isn't in 1.6 (not even prepending "core" to the front). A core help show didn't help... -------------- next part -------------- An HTML attachment was scrubbed... URL:

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:

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:

I'm looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only use type=peer Is that still correct? Will type=user be phased out, and should even new installs of older asterisk versions (eg: 1.6) use

I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating, and places the number dialed into the "accountid" field. I'm trying to distinguish between failed attempts to register and attempts to dial without registering, but the security log treats them identically (using the accountid field for either the username or number

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session