similar to: ARI with * 1.4.2 won't display recordings

Evnin' (o; As chan_sccp is pretty much dead, doesn't compile on FBSD anyway and isn't supported on * 1.4.x I tried going with chan_skinny... The Cisco 7970 registers and is being acknowledged by * but that's it... I see no lines on the 7970 display configured and it is not reachable or it can't make any outboudn calls... The docs are pretty non-existent for skinny and the

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

ARI (Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have. This release supports: call monitor page ? new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor

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

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

Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '0354c42214142e5d6cb8e05568c59837@10.0.2.2'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick

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

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

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

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

recreate Echo, if that is possible. trying to recode all dialplan to stasis application > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > Can anyone tell me how can I create echo test using ARI stasis application? > I'm not sure an 'echo' test really makes much sense with ARI, but we do have some nice documentation on getting started with ARI on the wiki. The basic tutorial example should give you an ARI event over a WebSocket

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

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

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

I am trying to use ARI for call monitoring. Recording conversations and such. The problem is that I don't use AMP, and don't have any sort of a database for CDR setup. It is all stored in the CSV file by default. When I setup ARI I tell it to go into standalone mode, and I set the asterisk manager username and password that was defined in manager.conf, but it also wants a cdr username and