similar to: AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

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

On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> 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?

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul Albrecht

What's up with the user list archive? It hasn't been updated since the 23rd.

I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks > Dear Ben, > I've also the problems as Chris Mason, Can you send us your own AGI CDR > logging application? > Best regards, > Jian Hong Guan > France > www.directcentrex.com > > >

Hi all, please reply to this thread of you have any agenda items that you would like to add. Meeting minutes of past minutes are under * http://wiki.centos.org/SpecialInterestGroup/Virtualization "Meetings" * The TODO list is under http://wiki.centos.org/SpecialInterestGroup/Virtualization/Status (just updated it) * Draft roadmap is under "Roadmap" and probably needs

Hi all, we do have an IRC SIG meeting scheduled for 2pm UK today. George is in the US and may not be able to join as the call is early. It also looks as if I may not be able to join. Are there any items to discuss? If so, please raise anything you want to discuss Who is planning to join? Raise your hand in the next hour If I don't get items/participants, I suggest we cancel today's

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

Hi, In Europe many Telcos implement power-save mode (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information). Would you agree to have this feature added to the ones already discuused for next Asterisk release ? (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010) Regards -------------- next part -------------- An

In putting together the SoundPack code, I am looking at the various language/locale specific code, and wondering how it all really stands... So, share with me, non-English speakers, what is your experience and impression? I heard a few comments during AstriDevCon, that some of the languages are not quite right; some said their language was understandable, but... Would anyone be willing to share

Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to write out the audio frames

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

Hello, never posted to a mailing list before. I've been trying to work out this problem for quite awhile now. I have a PHP script which is run whenever an emergency situation happens. The script connects to the AMI and originates calls to previously defined "emergency" extensions. I'm looking for a way to disconnect all calls on a device if it is in use, in order to deliver the

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 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,

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

On Monday 15 June 2020 at 00:41:14, Bruce Ferrell wrote: > Way back in the mists of time, I built my asterisk installation with SNMP > support. Heh... I never even knew that was possible :) > That said, I actually prefer ARA/ARI to flat file configuration of endpoints > and dialplans. Changes are more or less instantaneous and easily shared > between instances. Agreed - ARA is

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After