similar to: Event showing who called who

Ok that worked. Since I have not declared a statis app called “test”, does that mean any non-existent app name on the URL will subscribe to all system events? (Or is test a built-in app name) Brian From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday, June 7, 2023 9:56 AM To: Asterisk Users Mailing List - Non-Commercial

On Wed, Jun 7, 2023 at 10:46 AM TTT <lists at telium.io> wrote: > I’ve reread the documentation a few times, and what isn’t clear is whether > I need an app=X parameter in the url. In other words, can I only get > events for a single named statis app? Or can I get events for the entire > Asterisk server? > > > > The command below (without app= parameter) results in

I’ve reread the documentation a few times, and what isn’t clear is whether I need an app=X parameter in the url. In other words, can I only get events for a single named statis app? Or can I get events for the entire Asterisk server? The command below (without app= parameter) results in no events being shown, but no error either. Thanks Brian (Ast newbie) From: asterisk-users

I tried the command below (with subscribeAll=yes). I made a couple of calls but didn’t see any events. Should I see events? From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Tuesday, June 6, 2023 6:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users]

I’ve been looking through the docs (near https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API) and am searching for a list of events I can subscribe to. Is this list published? Or can I query the ARI for a list of available events? Thanks Brian From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday,

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

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

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

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

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll

I have the ARI enabled on my Asterisk test box, and want to listen to all events. I can't find the syntax to do that. Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk " I know how to do it form the AMI.looking for

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. >

I think that’s getting me close. I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP. I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like: FROM=<URI>;tag=TAG And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like: TO=<URI>;tag=TAG Would it be correct to assume

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

On Wed, Jun 7, 2023 at 12:04 PM TTT <lists at telium.io> wrote: > Ok that worked. > > > > Since I have not declared a statis app called “test”, does that mean any > non-existent app name on the URL will subscribe to all system events? (Or > is test a built-in app name) > Applications are not declared or configured anywhere. The act of connecting a websocket with an

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > 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,

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io> wrote: > 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