similar to: [asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/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

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? Here?s a link to the notes posted on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2014

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

I see the following paragraph in the Asterisk trunk LICENSE file: "In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI). It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL or a compatible license, as we believe

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

On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > I am connecting to the ARI with subscribe all, so I can see channels being > created. I now want to extract a variety of header variables (at the > moment the from and to tag). I tried to read them from the ARI but > Asterisk refuses since the channel is not in a stasis app. > > > > Is there a way

On 6/26/23 9:00 AM, Joshua C. Colp wrote: > On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > > I am connecting to the ARI with subscribe all, so I can see > channels being created. I now want to extract a variety of header > variables (at the moment the from and to tag).  I tried to read > them from the ARI but Asterisk refuses since the

I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > On 6/26/23 9:00 AM, Joshua C. Colp wrote: >> On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: >> >> I am connecting to the ARI with subscribe all, so I can see >> channels being created.  I now want to extract a variety of >> header variables (at the moment the from and to tag).  I

Up till now, I have only used Asterisk versions 1.2, 10 and 11, on CentOS 4, 5 and 6, and have made extensive use of AMI and FastAGI connections to a multi-threaded backend written in C. For a new project, I am looking at trying Asterisk 16 with ARI, on CentOS 7. I was looking at the various ARI libraries available, particularly the ones for Python and Node.js in github. I noticed that the

It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel. I also read (on stackoverflow) that the PJSIP_HEADER function will only return the headers from the INVITE of the inbound channel. If that’s correct, how would I get the headers from the outbound channel (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact return the

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 Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

Le 2015-12-15 15:25, Joshua Colp a ?crit : > Sylvain Boily wrote: >> Hello, > > Just a note - there's an asterisk-app-dev mailing list[1] which is > better suited for these kind of posts. Ok > >> I did some tests because i'm interesting to transfer a non stasis bridge >> to a stasis bridge and i found a strange situation. > > You can't, you have

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

I am connecting to the ARI with subscribe all, so I can see channels being created. I now want to extract a variety of header variables (at the moment the from and to tag). I tried to read them from the ARI but Asterisk refuses since the channel is not in a stasis app. Is there a way to read these from either the ARI or AMI ? I'm trying not to modify the dialplan. Thanks Brian

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this beta: New

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

I'm pretty sure there isn't a way to do that currently. ?My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used.? On Sat, May