similar to: ARI 0.06

Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/a8d923ae/attachment.htm

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Seems like other postings tend to think that saving recordings as files and not as blobs in the database are a more reliable way to go. Opinions on this? Looking at supporting it for ARI and judging interest. Dan 512.791.0137 www.littlejohnconsulting.com

Hi, Just wanted to pass on a fix that I found with the ARI recordings interface (www.littlejohnconsulting.com) for using a browser to access voice mail. It turned out to be a rights issue and group membership issue. I was planning on moving Asterisk to a non-root (http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25) user but I had not done this prior to

Anybody knows why ARI gives this error message when I enter extension number and password. *Warning*: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg0000.txt): failed to open stream: Permission denied in * /var/www/html/recordings/modules/voicemail.module* on line *525* It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A

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

Support Alternative Routing-ID Interpretation (ARI), which increases the number of functions that can be supported by a PCIe endpoint. ARI is required by SR-IOV. PCI-SIG ARI specification can be found at http://www.pcisig.com/specifications/pciexpress/specifications/ECN-alt-rid-interpretation-070604.pdf Signed-off-by: Yu Zhao <yu.zhao at intel.com> Signed-off-by: Eddie Dong <eddie.dong

Support Alternative Routing-ID Interpretation (ARI), which increases the number of functions that can be supported by a PCIe endpoint. ARI is required by SR-IOV. PCI-SIG ARI specification can be found at http://www.pcisig.com/specifications/pciexpress/specifications/ECN-alt-rid-interpretation-070604.pdf Signed-off-by: Yu Zhao <yu.zhao at intel.com> Signed-off-by: Eddie Dong <eddie.dong

Support Alternative Routing-ID Interpretation (ARI), which increases the number of functions that can be supported by a PCIe endpoint. ARI is required by SR-IOV. PCI-SIG ARI specification can be found at http://www.pcisig.com/specifications/pciexpress/specifications/ECN-alt-rid-interpretation-070604.pdf Signed-off-by: Yu Zhao <yu.zhao at intel.com> Signed-off-by: Eddie Dong <eddie.dong

Hello, I am experimenting with Asterisk 12.0.0 alpha1. I have a couple of SIP phones working. Good. I can retrieve data using curl to interact with the new Asterisk REST API (ARI). Good. Now I want to use the new ARI events API, which requires a WebSocket connection. I am using Node.js for the client, and have a stable connection to ARI events on the Asterisk 12 server. What I hope for is

Hi there! Is there any way I can turn on debug for ARI and sending the output to a separate log file? So far I have only been able to turn on ARI debugging in the console which results in the debug output being logged in /var/log/asterisk/messages I would love to have ARI debug log messages in /var/log/asterisk/debug or even better in it's own ari-debug file. With best regards Florian

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

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 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 attempting to run a test of the ARI recording of audio from the channel. When I send the record command, it's failing. curl -v -u asterisk:asterisk -X POST "http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest&format=WAV&maxDurationSeconds=300&maxSilenceSeconds=3" [08/11 09:14:13.290] WARNING[23806]: ari/resource_channels.c:812

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

hi, do you have someone example of http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/ in node.js asterisk-ari ? thanks Marek

Hi, I have '*1' in my features.conf file and I'm facing with a serious problem: - A and B are engaged in a call - C and D are engaged in a different call and decide to record their conversation hitting *1 - at the end, A and B are able to see C/D call recording using ARI with their user/pwd!!! Where is the problem? Asterisk or ARI? Thanks in advance -- Domenico Viggiani

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and