similar to: Asterisk & 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

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100

Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

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

I am trying to run the CyberMegaPhone demo to see the WebRTC Video Conference demonstration from AstriDevCon 2017 I have been able to make WebRTC work on this same box with SIPML5 demo but not the CMP2K. When I attempt to access the https://myip:8089/cmp2k I am prompted for the unsecure web. I enable unsecure web. (Using the asterisk local certificate generation from the SIPML5 demo). After

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support . I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I

Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws <wss://54.254.228.251:8080/ws>' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event =

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5851 dialog_initialize_dtls_srtp: Attempted to set an invalid DTLS-SRTP configuration on RTP

Hi all, I am using the following set up: SIPML5 --> Nginx --> Asterisk, where NGINX as a reverse proxy, main purpose is to take in wss and route to Asterisk's ws. However, I am facing this issue recently where Nginx will return 502 gateway error of 8018#0: *24183 upstream prematurely closed connection while reading response header from upstream, client: 116.15.31.xxx, server:

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

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

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

tengo la siguiente pagina pero no se como seguir despues del punto 22 http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html gracias! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141112/59751a87/attachment.html>

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from the wiki [1]. > So I'm using [2] live demo from

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