| 00:26.00 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 00:26.13 | ZX81 | nah - no response lol |
| 00:26.20 | ZX81 | would've been nice :-) |
| 00:51.19 | ZX81 | when you register an app with stasis via web sockets (I'm assuming from a command line script) are you running a server or a client? |
| 00:51.30 | ZX81 | a client yeah? |
| 00:51.41 | ZX81 | and Asterisk is the server? |
| 01:05.09 | jgowdy | http://llvm.org/bugs/show_bug.cgi?id=17482 |
| 01:05.30 | jgowdy | Trying to get some answers on how they expect useful use of the cleanup extension with no nested functions |
| 01:05.56 | jgowdy | If blocks are such a great alternative for nested functions, why don't they work with your implementation of cleanup? |
| 01:06.20 | jgowdy | This is re: RAII_VAR on clang |
| 01:06.43 | jgowdy | I tried several ways, damned thing won't even accept void** |
| 01:07.26 | ZX81 | ok, got it connected but [Oct 4 21:06:46] WARNING[32567]: res_http_websocket.c:607 __ast_websocket_uri_cb: WebSocket connection from '127.0.0.1:52420' could not be accepted - no protocols out of 'stasis' supported |
| 01:07.41 | ZX81 | should the header not be Sec-WebSocket-Protocol: stasis |
| 01:38.51 | *** join/#asterisk-dev serafie (~erin@24.96.64.240) |
| 02:07.43 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 02:09.40 | *** join/#asterisk-dev serafie1 (~erin@24.96.64.240) |
| 02:22.06 | ZX81 | when I get a response that a channel is started in a stasis app it looks weird: |
| 02:22.07 | ZX81 | b{"application":"test","type":"StasisStart","timestamp":"2013-10-04T22:20:50.673-0400","args":[],"channel":{"id":"1380939648.4","state":"Up","name":"SIP/matt-00000002","caller":{"name":"","number":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"matt","exten":"","priority":1},"creationtime":"2013-10-04T22:20:48.787-0400"}} |
| 02:22.14 | ZX81 | that's not json |
| 02:22.29 | ZX81 | is there something I should be doing? |
| 02:22.33 | ZX81 | is the protocol wrong? |
| 02:22.55 | ZX81 | where does the b come from at the start? |
| 02:26.55 | ZX81 | also the rfc says I should have a protocol |
| 02:26.56 | ZX81 | Sec-WebSocket-Protocol: chat, superchat |
| 02:27.08 | ZX81 | I guess stasis but that isn't accepted |
| 02:27.25 | ZX81 | and I can't see in /usr/src/asterisk-12/res/res_http_websocket.c where they are defined |
| 02:30.06 | ZX81 | This is way harder than just writing an app in c |
| 04:28.11 | *** join/#asterisk-dev zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 04:39.59 | *** join/#asterisk-dev otherwiseguy (~otherwise@69.23.116.90) |
| 07:00.40 | *** join/#asterisk-dev ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 12:53.15 | *** join/#asterisk-dev protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:27.58 | tzafrir_laptop | Is there a separate channel for Astricon? |
| 13:28.16 | tzafrir_laptop | Anyway, is there any reasonable way to get to the neue in public transport? |
| 13:32.33 | file | there is #astricon and #astridevcon |
| 13:33.36 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 13:33.43 | ZX81 | hi |
| 13:33.52 | WIMPy | And don't accidentailly end up in #astrocon. |
| 13:33.58 | ZX81 | still trying to get web sockets and ari going (kinda) |
| 13:34.02 | ZX81 | http://pastebin.com/pzBGJrkx |
| 13:34.11 | ZX81 | why do I get .~.c |
| 13:34.17 | ZX81 | and .~.. |
| 13:34.23 | ZX81 | in-between packets? |
| 13:34.35 | ZX81 | that pastebin is an ngrep output |
| 13:36.12 | ZX81 | they're weird characters or something because they cause subsequent characters to disappear if i print them without ngrep |
| 13:37.03 | *** join/#asterisk-dev serafie (~erin@24.96.64.240) |
| 13:37.39 | file | erm... how are you connecting? |
| 13:37.57 | ZX81 | php fsockopen |
| 13:38.13 | ZX81 | but that paste bin shows the same thing |
| 13:38.17 | ZX81 | just with ngrep |
| 13:38.23 | file | you should use an actual websocket client, it's an actual protocol and not just text going to you |
| 13:38.38 | file | there's a header and framing |
| 13:38.55 | ZX81 | there isn't a web socket client library for php |
| 13:38.59 | ZX81 | same for most languages |
| 13:39.04 | ZX81 | there are server libraries |
| 13:39.09 | ZX81 | but not client libraries |
| 13:39.27 | file | https://github.com/nicokaiser/php-websocket |
| 13:40.11 | ZX81 | most people expect the client to be a browser and the server to be perl/python etc |
| 13:40.17 | ZX81 | that doesn't support client anymore |
| 13:40.25 | ZX81 | I've been through like 15 projects |
| 13:40.37 | ZX81 | "The client-side libraries are no longer supported: some libraries are included but are packaged only as examples" |
| 13:41.09 | file | doesn't mean it won't work, but ultimately that's why you are seeing what you are seeing |
| 13:41.35 | ZX81 | it kinda does because Asterisk doesn't support version 76 |
| 13:41.41 | ZX81 | which is where that one was left |
| 13:41.42 | ZX81 | :-) |
| 13:41.43 | ZX81 | ok |
| 13:41.51 | ZX81 | so it's back to the rfc then I guess |
| 13:42.31 | ZX81 | it's only because of the rfc I've got any communication going - but I didn't do framing :-) |
| 13:42.32 | ZX81 | thanks |
| 13:43.08 | ZX81 | it would be much easier if a stasis application was a server |
| 13:43.19 | ZX81 | and Asterisk connected to it |
| 13:43.35 | ZX81 | then there are like 1000 libraries etc :-) |
| 13:44.05 | file | there's other libraries for other languages, PHP may just be the outlier |
| 13:45.22 | file | I use python myself |
| 13:45.36 | ZX81 | mostly server libraries |
| 13:45.39 | ZX81 | rather than client |
| 13:45.44 | ZX81 | you use a library? |
| 13:45.49 | file | yes. |
| 13:45.55 | ZX81 | or have a hello world type example |
| 13:46.04 | ZX81 | I've already done pip install web socket etc |
| 13:46.05 | file | I use a library for the websocket part |
| 13:46.18 | file | and I believe Matt gave an example yesterday |
| 13:46.23 | ZX81 | didn't run lol |
| 13:50.14 | ZX81 | it didn't have any authentication in it |
| 13:50.32 | ZX81 | <PROTECTED> |
| 13:50.33 | ZX81 | <PROTECTED> |
| 13:50.33 | ZX81 | <PROTECTED> |
| 13:50.40 | ZX81 | also, I didn't know at the time |
| 13:50.53 | ZX81 | but after browsing source found that stasis should have been ari |
| 13:51.02 | ZX81 | so it threw an error for that too |
| 13:51.41 | ZX81 | yeah websocket.WebSocketException: Handshake Status 401 |
| 13:53.34 | file | http://pastebin.com/E5Lh0Atj |
| 13:53.34 | file | fill in IP address, username, and password |
| 13:53.55 | ZX81 | &api_key |
| 13:53.58 | ZX81 | heh |
| 13:54.07 | ZX81 | awesome thanks man!!! |
| 13:54.11 | file | yes, mine was an updated copy |
| 13:54.48 | ZX81 | :-)))))))) |
| 13:54.49 | ZX81 | sweeet |
| 13:54.55 | ZX81 | thanks heaps! |
| 13:56.13 | ZX81 | awesome, works perfectly! |
| 14:37.16 | *** join/#asterisk-dev [TK]D-Fender (~chatzilla@64.235.216.2) |
| 16:17.09 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 16:17.29 | ZX81 | is there something I need to do to keep a channel in a stasis app? It seems to disconnect after 30 seconds |
| 16:18.25 | ZX81 | http://pastebin.com/Q8geKn74 |
| 16:18.53 | file | answer it. |
| 16:19.14 | ZX81 | inside stasis? |
| 16:19.16 | ZX81 | <PROTECTED> |
| 16:19.16 | file | ie: call Answer() before Stasis() |
| 16:19.36 | ZX81 | it's a ARI call sending the call to a stasis app |
| 16:19.47 | ZX81 | the python one you sent |
| 16:20.18 | ZX81 | do I need to do ws.send("Answer") or something? |
| 16:20.42 | file | no, you can't send stuff over the websocket - it's for receiving events |
| 16:20.52 | ZX81 | ok lol |
| 16:21.11 | ZX81 | so if I originate a call to a stasis app how does it get answered? |
| 16:21.17 | ZX81 | better to use dialplan? |
| 16:21.28 | file | I can't remember if there is an answer command or not, I know it'll work for sure with dialplan |
| 16:21.38 | ZX81 | ok, I'll just use dialplan |
| 16:21.39 | ZX81 | ty |
| 16:21.42 | file | that's why the other side is hanging up... it's not answered and they gave up after 30 seconds |
| 16:21.47 | ZX81 | right cool |
| 16:22.20 | ZX81 | so stasis is purely for receiving events |
| 16:22.28 | ZX81 | and use ARI for the control? |
| 16:22.59 | file | yes |
| 16:23.02 | ZX81 | ok |
| 16:23.52 | ZX81 | so would I be better off creating a stasis app from the web site for each client and just using web sockets to receive info? |
| 16:24.00 | ZX81 | one for each client? |
| 16:24.06 | file | I don't understand |
| 16:24.12 | ZX81 | so I have a website |
| 16:24.24 | file | you wouldn't connect a website directly |
| 16:24.29 | ZX81 | it sends a /ari/channels?endpoint=SIP%2Fmatt&app=hello |
| 16:24.39 | file | well, I mean, you COULD if you wanted |
| 16:25.30 | ZX81 | maybe a one or two sentence description of the idea of ARI and stasis? |
| 16:25.50 | file | it's for writing applications outside of Asterisk |
| 16:26.00 | file | a queueing application, potentially in the future a voicemail application |
| 16:26.10 | file | you could write an entire IVR in it if you really wanted |
| 16:26.23 | ZX81 | so you hold two connections? |
| 16:26.30 | ZX81 | one with rest and one with web sockets? |
| 16:26.46 | file | you hold a websocket connection which allows you to receive events (channel comes into your application, they hit DTMF, etc) |
| 16:26.59 | file | and then the HTTP connection which invokes actions may or may not be persistent (depends on the HTTP client implementaiton) |
| 16:27.05 | ZX81 | ok cool |
| 16:27.22 | file | so for example you could write a small app that sees a channel comes into your application, and then invokes an action to tell Asterisk to play a specific file |
| 16:27.37 | file | then if they hit DTMF you receive that on the websocket connection as an event and you can then invoke a different action |
| 16:27.55 | file | or you can create an outgoing channel and a bridge, and connect them together |
| 16:27.59 | file | or connect multiple channels in your application together |
| 16:28.11 | ZX81 | ok starting to gel :-) |
| 16:28.37 | file | so you could write your own ConfBridge if you wanted, and have completely control over stuff |
| 16:28.48 | ZX81 | yep |
| 16:29.23 | file | and then in your application, outside the scope of Asterisk, you could have a further mechanism to allow something else (like a web portal) to control it |
| 16:29.51 | file | ie: a portal for controlling the conference bridge (muting people, kicking people out) |
| 16:30.25 | file | but do you need to know about internal Asterisk bridges or how everything is accomplished? nope |
| 16:30.40 | file | add channel X to bridge Y, mute channel X, etc |
| 16:31.23 | ZX81 | ok cool |
| 17:03.53 | ZX81 | it still ended after 30 seconds |
| 17:03.55 | ZX81 | lol |
| 17:04.04 | ZX81 | maybe bridging attempt or something |
| 17:04.11 | ZX81 | or rtptimeout |
| 17:05.01 | ZX81 | trying now manual dial instead of ari |
| 17:05.49 | ZX81 | same thing |
| 17:05.49 | ZX81 | http://pastebin.com/B96AmSWy |
| 17:06.24 | ZX81 | looks like bria requested a bye |
| 17:06.25 | ZX81 | [Oct 5 13:05:12] DEBUG[18691][C-00000001]: chan_sip.c:28050 handle_incoming: **** Received BYE (8) - Command in SIP BYE |
| 17:06.39 | ZX81 | I answered the call first |
| 17:07.06 | ZX81 | maybe because there is no rtp or something? |
| 17:37.57 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 18:09.51 | ZX81 | hmmm looks like no audio passing on 127.0.0.1 (obviously no nat) that's causing it to disconnect |
| 18:10.00 | ZX81 | even backgrounding a file from dialplan doesn't play |
| 18:10.17 | ZX81 | tried a couple of different clients |
| 18:12.10 | ZX81 | registration, calls etc work fine just not rtp |
| 18:14.08 | ZX81 | anyone got a sec to look at a sip debug? |
| 18:14.09 | ZX81 | http://pastebin.com/4fu7EZRj |
| 18:14.33 | ZX81 | I tried binding to 0.0.0.0 and 127.0.0.1 |
| 18:14.49 | ZX81 | and tried setting server addy to 127 or 10.x |
| 18:15.03 | ZX81 | the sdp etc all looks fine |
| 18:17.40 | ZX81 | Got RTP packet from 127.0.0.1:56929 (type 00, seq 024785, ts 000160, len 000160) |
| 18:17.41 | ZX81 | and |
| 18:17.46 | ZX81 | Sent RTP packet to 10.0.1.29:56929 (type 00, seq 027191, ts 000160, len 000160) |
| 18:17.57 | ZX81 | but no sound |
| 18:23.10 | ZX81 | maybe just an osx'ism |
| 18:24.07 | pabelanger | how is 10.0.1.29 will route to 127.0.0.1 |
| 18:24.25 | pabelanger | s/will/going to |
| 18:56.19 | *** join/#asterisk-dev serafie (~erin@24.96.64.240) |
| 19:14.52 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 19:15.26 | ZX81 | tried with both 127 and 10.0 for bind and server addy |
| 19:17.16 | ZX81 | Got RTP packet from 10.0.1.29:4002 (type 00, seq 014722, ts 000160, len 000160) |
| 19:17.22 | ZX81 | Sent RTP packet to 10.0.1.29:4002 (type 00, seq 016960, ts 000160, len 000160) |
| 19:17.28 | ZX81 | same result |
| 19:21.30 | ZX81 | it only seems to send/receive one rtp packet |
| 19:21.39 | ZX81 | or does it not show more than one in debug? |
| 19:22.26 | ZX81 | its like when there is no timer working |
| 19:23.41 | ZX81 | how do I change timers? |
| 19:23.52 | ZX81 | wasn't there like a pthread timers option somewhere? |
| 19:25.04 | ZX81 | XXX res_timing_dahdi, XXX res_timing_timerfd |
| 19:25.16 | ZX81 | I don't have to have one of those in menuselect do i? |
| 19:26.14 | ZX81 | res_timing_kqueue doesn't show up in menuselect options |
| 19:26.23 | ZX81 | although it looks like it should according to docs |
| 19:27.07 | ZX81 | ah it's under extended |
| 19:27.11 | ZX81 | and enabled |
| 19:27.19 | ZX81 | so that's not it |
| 19:28.52 | ZX81 | fml |
| 19:33.52 | ZX81 | by the way |
| 19:33.54 | ZX81 | <PROTECTED> |
| 19:34.00 | ZX81 | strings.h:1006:10: warning: 'toupper' is static but used in inline function 'ast_tech_to_upper' which is not static [enabled by default] |
| 19:34.01 | ZX81 | <PROTECTED> |
| 19:34.12 | ZX81 | comes up like 100000 times during make lol |
| 19:39.18 | ZX81 | YESSSSSSSSS |
| 19:39.22 | ZX81 | found it!!!!!! |
| 19:39.32 | ZX81 | res_timing_kqueue doesn't work!!!! |
| 19:39.35 | ZX81 | saweeeeeeeeeet |
| 19:39.39 | ZX81 | :-) |
| 19:43.20 | ZX81 | lol |
| 19:43.20 | ZX81 | [Oct 5 15:43:03] ERROR[23811][C-00000000]: res_stasis_playback.c:295 play_on_channel: Unimplemented |
| 19:43.22 | ZX81 | great |
| 19:43.23 | ZX81 | fml |
| 19:44.54 | ZX81 | ah |
| 19:44.56 | ZX81 | sound: |
| 19:48.38 | ZX81 | example of a play sound uri? |
| 19:48.40 | ZX81 | for rest? |
| 19:48.58 | ZX81 | not working: /ari/channels/1381002091.0/play?media=sound%3Ademo-echotest.gsm&lang=en |
| 19:49.13 | ZX81 | I mean it finds the chan and app etc - no errors |
| 19:49.16 | ZX81 | just no sound |
| 19:49.26 | ZX81 | and doesn't say anything in debug about playing |
| 19:49.43 | ZX81 | I tried without the .gsm too |
| 19:51.48 | ZX81 | tried this too |
| 19:51.50 | ZX81 | HTTP Request URI is /ari/channels/1381002091.0/play?media=http%3A%2F%2Fwww.nch.com.au%2Facm%2F8k16bitpcm.wav |
| 19:54.04 | ZX81 | hmmm mohstart doesn't work either |
| 19:54.12 | ZX81 | maybe the channel has to be bridged to something |
| 19:57.42 | ZX81 | ok weird |
| 19:57.51 | ZX81 | playing an audio file before stasis seems to have helped |
| 19:59.08 | *** join/#asterisk-dev serafie (~erin@24.96.64.240) |
| 19:59.35 | ZX81 | this is awesome!!! |
| 20:02.04 | ZX81 | omg |
| 20:02.08 | ZX81 | that is so so cool |
| 20:02.18 | ZX81 | finally! |
| 20:02.19 | ZX81 | :-) |
| 20:26.49 | *** join/#asterisk-dev protocoldoug (~doug@unaffiliated/protocoldoug) |
| 21:40.06 | *** join/#asterisk-dev protocoldoug (~doug@unaffiliated/protocoldoug) |
| 21:45.03 | *** join/#asterisk-dev ZX81 (~ZX81@184.91.71.197) |
| 21:45.10 | ZX81 | ok, so things are going pretty good |
| 21:45.16 | ZX81 | but if I try to use POST /bridges/{bridgeId}/play |
| 21:45.21 | ZX81 | I get the 409 |
| 21:45.27 | ZX81 | 409 - Bridge not in a Stasis application |
| 21:45.42 | ZX81 | but there doesn't seem to be any way to put a bridge into a stasis application |
| 21:45.51 | ZX81 | the channel was originally in stasis |
| 21:46.11 | ZX81 | and I joined a new endpoint to it |
| 21:46.16 | ZX81 | with /ari/channels |
| 21:46.44 | ZX81 | hmmm |
| 21:46.52 | ZX81 | maybe I have to send it to stasis |
| 21:47.01 | ZX81 | then bridge them once that happens? |
| 22:05.44 | ZX81 | arghhh |
| 22:05.51 | ZX81 | why can't creating a channel return the channel id |
| 22:06.01 | ZX81 | if I create a new channel I have no way to find it |
| 22:06.18 | ZX81 | if I use POST /channels/{channelId}/dial Dialed Create a new channel (originate) and bridge to this channel. |
| 22:06.28 | ZX81 | then it isn't in the stasis app |
| 22:06.33 | ZX81 | even though the original was |
| 22:06.48 | ZX81 | and if I create a new channel I have no idea of its channel id |
| 22:08.45 | ZX81 | it seems to be fine for a registered sip peer |
| 22:08.52 | ZX81 | because I can search for the name I created |
| 22:09.01 | ZX81 | but if I do a remote dial it converts it to the ip |
| 22:11.33 | pabelanger | that sounds odd. I would expect some sort of dataset to be returned when you POST info |
| 22:14.45 | ZX81 | yeah |
| 22:14.54 | ZX81 | explicitly says it returns voip |
| 22:14.56 | ZX81 | void |
| 22:14.57 | ZX81 | lol |
| 22:14.58 | ZX81 | :-) |
| 22:30.20 | file | https://issues.asterisk.org/jira/browse/ASTERISK-22485 that's the issue to improve it |