Re: [asterisk-users] Wanted: WebRTC tutorial
On Tue, Apr 24, 2018 at 9:56 AM, Bruce Ferrell wrote: I'd REALLY like to get it working. And for the record, I REALLY HATE pjsip. > > I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux > since '93 and telecom since 1980. The config is so opaque, poorly > documented and error prone I, to this day, use the legacy sip config > wherever I can. No one has ever been able to show me an advantage for it > and it doesn't seem to use realtime configuration (even more of a > drawback). I much prefer realtime for my configuration on Asterisk; > Having configuration picked up from a DB is far preferable to reloading > flat files. > I've configured PJSIP to use realtime as a test, I don't use it in production but I feel I could. Just set it up in extconfig and create the necessary tables. ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_registrations => odbc,asterisk ps_phoneprov => odbc,asterisk There's a tutorial in the Wiki for it https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime The major advantage of PJSIP over chan_sip to me is PJSIP is being developed where chan_sip isn't. It also introduces things like resource lists, and parallel forking of contacts which are both nice features. -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: WebRTC tutorial
On 4/25/18 6:05 AM, Joshua Colp wrote: On Wed, Apr 25, 2018, at 12:40 AM, Bruce Ferrell wrote: OK, I've gone back and refreshed myself; When I try to access cybermega in /var/lib/asterisk/static-http at port 8088 the asterisk debug shows: [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session. Top level [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/ index.html [Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] has no handler [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. status_code:404 When using static-http you have to have /static at the front so the path would be: /static/cyber/index.html When I serve it from apache, the web ui appears, but never connects. Using the firefox dev tools/console I see firefox can't establish a connection the server at wss://:8089/ws The asterisk debug log shows: [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session. Top level [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws [Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no handler [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. status_code:404 Suggestions? Is there anything in the console at startup stating that stuff didn't load? The module which does websockets is res_http_websocket, and you can see if all that is needed is loaded using: "module show like websocket" on the CLI. That's a step closer! The URI has to be /asterisk/static/cyber/index.html. I should make the asterisk part go away, but for now... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wanted: WebRTC tutorial
On Wed, Apr 25, 2018, at 12:40 AM, Bruce Ferrell wrote: > OK, I've gone back and refreshed myself; When I try to access cybermega > in /var/lib/asterisk/static-http at port 8088 the asterisk debug shows: > > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session. Top level > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/ > index.html > [Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] > has no handler > [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. > status_code:404 When using static-http you have to have /static at the front so the path would be: /static/cyber/index.html > > When I serve it from apache, the web ui appears, but never connects. > > Using the firefox dev tools/console I see firefox can't establish a > connection the server at wss://:8089/ws > > The asterisk debug log shows: > > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session. Top level > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws > [Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no > handler > [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. > status_code:404 > > Suggestions? Is there anything in the console at startup stating that stuff didn't load? The module which does websockets is res_http_websocket, and you can see if all that is needed is loaded using: "module show like websocket" on the CLI. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
