similar to: Problem with R websocket package

On 5/29/20 8:50 AM, Daniel P. Berrangé wrote: >>> (2) You need to persuade qemu's NBD client to read from a WebSocket. >>> I didn't really know anything about WebSockets until today but it >>> seems as if they are a full-duplex protocol layered on top of HTTP [a]. >>> Is there a WebSocket proxy that turns WS into plain TCP (a bit like >>>

[adding qemu list] On 5/29/20 4:37 AM, Richard W.M. Jones wrote: > Going back to the original email from 2018: > >> It might be neat to attach ISOs to KVM guests via websockets.  Basically >> the  browser would be the NBD "server" and an NBD client would run on the >> hypervisor, then use `virsh change-media vm1 hdc --insert /dev/nbd0` could >> use

On Fri, 29 May 2020, Richard W.M. Jones wrote: > On Fri, May 29, 2020 at 08:58:06AM -0500, Eric Blake wrote: > > On 5/29/20 8:50 AM, Daniel P. Berrangé wrote: > > > > >>>(2) You need to persuade qemu's NBD client to read from a WebSocket. > > >>>I didn't really know anything about WebSockets until today but it > > >>>seems as if

Hello, I run websockets 1.1.1 from R 2.15 without any problems. I would like to deploy R script running automatically for all users when R is starting from Rprofile.site like follow: .First <- function() { source("C:\\R\\orig-websockets.R") } This script is from websockets package distribution. The error is follow: Error in .parse_header(x) :

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 all, I can't find a working archive search for this list, so please forgive me if this has been discussed before. Has any thought been given to supporting websockets in the ssh client? I'm talking about solely using a websocket as the transport layer, and leaving the actual protocol intact, as opposed to the (to me, frankly terrifying) idea of allowing a web server to act as an ssh

On Thu, 28 May 2020, Richard W.M. Jones wrote: > On Thu, May 28, 2020 at 12:24:22AM +0000, Eric Wheeler wrote: > > On Mon, 15 Oct 2018, Nir Soffer wrote: > > > On Sat, Oct 13, 2018 at 9:45 PM Eric Wheeler <nbd@lists.ewheeler.net> wrote: > > > Hello all, > > > > > > It might be neat to attach ISOs to KVM guests via websockets. 

Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf file: [3002] username=3002 secret=XXXXXXXXX host=dynamic type=friend context=test disallow=all allow=g729 ;allow=all ; Allow codecs in order of preference allow=ilbc

Does Rails have anything that supports real-time events and communication with WebSockets and gracefully fallbacks to other techniques if WebSockets are not supported? I would like the possibility to pub/sub events and messages for live chats and real time notifications. -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To

I must be missing the point here somehow. From my simple mind I think that two things would be needed - first a mod, e.g., mod_sshd, or better an addition to mod_auth and mod_proxy so that a URL could be used to initiate contact to an sshd server elsewhere. The mod_auth part could/should be used to verity the credentials to used - basically setting up the VPN between ssh and httpd as ssh; the

On Tue, Oct 23, 2018 at 1:35 PM Dan Cropp <dan at amtelco.com> wrote: > The res_pjsip_transport_websocket failing to load seems to be a conflict > with the chan_sip.so loading. > > When I make the chan_sip.so not load, res_pjsip_transport_websocket.so > does load. > > We have customers who need chan_sip and chan_pjsip, so we need to load > both. Is there a way to

Pinging @tenderlove Some questions on how best to move forward with this: 1.) AC::Live is making use of the stream object on ActionDispatch::Response. Should the rack.hijack_io object be used instead of this in AC::Live or should ActionDispatch::Response be changed to use rack.hijack_io? 2.) What do you think about changing AC::Live to work with adapters? I suspect the primary use case for

Hi everyone, I'm currently trying to play a little with WebRTC using sipml5 client and Asterisk trunk (370821) It seems the the WebRTC implementation for Asterisk 11 is already available in the trunk? Am I right? http://lists.digium.com/pipermail/asterisk-dev/2012-July/055940.html I'm having trouble to even register to my Asterisk server using sipml5 client. The only reference to

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

On 29/01/15 21:15, Alex Bligh wrote: > Be frightened: > https://chrome.google.com/webstore/detail/secure-shell/pnhechapfaindjhompbnflcldabbghjo?hl=en > That's a ssh client implemented in chromium, not a web server acting as sshd. However... ?Secure Shell also knows how to connect to an HTTP-to-ssh relay that was built inside Google. Unfortunately that relay isn't open source,

On Fri, May 29, 2020 at 08:58:06AM -0500, Eric Blake wrote: > On 5/29/20 8:50 AM, Daniel P. Berrangé wrote: > > >>>(2) You need to persuade qemu's NBD client to read from a WebSocket. > >>>I didn't really know anything about WebSockets until today but it > >>>seems as if they are a full-duplex protocol layered on top of HTTP [a]. >

I'm learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side. I noticed that the Asterisk wiki mentions the need for res_pjsip_transport_websocket, so does that mean Asterisk requires the signaling to occur over a websocket? If I used

On Fri, May 29, 2020 at 07:50:14AM -0500, Eric Blake wrote: > [adding qemu list] > > On 5/29/20 4:37 AM, Richard W.M. Jones wrote: > > Going back to the original email from 2018: > > > > > It might be neat to attach ISOs to KVM guests via websockets.  Basically > > > the  browser would be the NBD "server" and an NBD client would run on the

Asterisk Project Security Advisory - AST-2018-009 Product Asterisk Summary Remote crash vulnerability in HTTP websocket upgrade Nature of Advisory Denial Of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate

Asterisk Project Security Advisory - AST-2014-019 Product Asterisk Summary Remote Crash Vulnerability in WebSocket Server Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate