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| 13:16.13 | skrusty | danjenkins: good post to list - something i've raised a few times |
| 13:16.35 | danjenkins | ta - me and mjordan were talking about it |
| 13:18.01 | skrusty | yeah, having Play("http://....") would be of massive help, i've posted my thoughts to the list too |
| 13:18.11 | skrusty | it would really help bridge the TTS gap ARI has |
| 13:18.13 | file | retrieving files via HTTP has been brought up but has yet to be done, streaming from a live source hasn't been brought up |
| 13:18.42 | file | (before danjenkins mentioned it) |
| 13:19.12 | danjenkins | :) |
| 13:19.14 | skrusty | sorry, my comments before were playing audio via HTTP and not "streaming" as such yes |
| 13:19.42 | skrusty | however, although technically they're different, from a consumer point of view, they differ very little |
| 13:19.49 | skrusty | on the surface at least |
| 13:20.33 | skrusty | my replay related specifically to "- prompts".. :) |
| 13:20.39 | Marquis42 | Exactly. If http://myserver.com/api/text-to-speech/somephrase returns a "file" that plays, who cares what the backend does? |
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| 13:24.56 | danjenkins | ++ |
| 13:25.00 | skrusty | :) |
| 13:25.30 | file | I think everyone is in agreement that this is all good and needed |
| 13:25.41 | skrusty | yeah |
| 13:25.53 | skrusty | tts is almost impossible atm with ARI |
| 13:25.55 | Marquis42 | file: Is the code done yet? |
| 13:25.56 | Marquis42 | ;) |
| 13:26.06 | skrusty | i had to make a nasty hack over a shared network drive! :) |
| 13:26.31 | file | if I was evil I'd have it done but keep it to MYSELF |
| 13:26.38 | skrusty | haha :) |
| 13:26.39 | Marquis42 | haha |
| 13:26.42 | Marquis42 | That *would* be evil |
| 13:26.54 | skrusty | who's saying you've not? :) |
| 13:26.58 | skrusty | we wouldn't know! |
| 13:26.59 | skrusty | :) |
| 13:27.02 | Marquis42 | haha |
| 13:27.07 | file | indeed |
| 13:27.39 | danjenkins | i can confirm - file is evil |
| 13:27.51 | skrusty | lol :) |
| 13:28.24 | file | so - when should I expect a patch to add http scheme support to ARI from all of you? I'll review it as soon as I see it :P |
| 13:28.52 | file | interestingly libcurl actually has support for many backends |
| 13:29.13 | skrusty | heh |
| 13:29.28 | skrusty | * written in c# yet? ;) once it is, you'll have my patch :) |
| 13:30.22 | file | I can confirm it is not. |
| 13:31.16 | danjenkins | once ARI lets me replace voicemail in a nice uniformed way within Asterisk, i'll do that ;) |
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| 13:32.35 | pabelanger | I'm actually trying to stream audio, using RTP, with ARI |
| 13:32.45 | pabelanger | POC works right now |
| 13:32.51 | pabelanger | just need to add it into ARI directly |
| 13:32.57 | Marquis42 | danjenkins: What's missing that you can't do VM in ARI? |
| 13:34.02 | danjenkins | Marquis42 - the fact that if someone wanted to implement VM and still let people communicate with VM via AMI/API then you couldnt over the AMI and every app's API would be implemented differently |
| 13:34.18 | danjenkins | I want Asterisk to have a standard and to be a "proxy" essentially |
| 13:34.23 | Marquis42 | Gotcha |
| 13:34.34 | danjenkins | so that users of Asterisk - 5 years down the line, will still have a PBX that works |
| 13:34.37 | Marquis42 | It's the AMI aspect of it that's at issue. |
| 13:35.02 | danjenkins | even tho that PBXis made up of asterisk core in C and VM in node and conf bridge in python |
| 13:35.10 | danjenkins | the general user doesnt care |
| 13:35.18 | danjenkins | they just want a nice free phone system ;) |
| 13:35.37 | danjenkins | people like pabelanger don't need that - but thats because they're doing something different, at different scales |
| 13:35.53 | danjenkins | :) |
| 13:36.18 | Marquis42 | danjenkins: Yeah I understand your concern now. It's not a concern of mine, for the same reasons it isn't for pabelanger basically, but I get what you're saying. |
| 13:37.07 | pabelanger | ExternalIVR FTW |
| 13:37.25 | danjenkins | Marquis42 yeah, i completely get why people who are looking at ARI *right now* aren't bothered about it -0 but if ARI is enabling us to replace these things, lets do it nicely and think abut the future needs etc now :) |
| 13:37.30 | danjenkins | HA pabelanger |
| 13:47.05 | jsmith | pabelanger: How do you get Asterisk to handle the RTP without an associated SIP channel? |
| 13:48.04 | jsmith | pabelanger: And ARI really is just ExternalIVR with a RESTish interface |
| 13:49.11 | pabelanger | jsmith, that's what I am trying to figure out right now |
| 13:49.34 | pabelanger | talking with file and mjordan, I need some channel, likely a local channel |
| 13:49.49 | file | there are multiple ways to do it. |
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| 13:50.14 | file | do it as an application and wrap it with a Local channel, or write a thin channel driver which uses RTP and is controlled somehow |
| 13:50.27 | pabelanger | right, I was considering that |
| 13:51.55 | file | heck, model it loosely after snooping in ARI and you've got an RTP pipe and using ARI as a signaling mechanism |
| 13:52.49 | pabelanger | my POC is going to create app_ffplay, which support RTP streams, then pipe it into asterisk (like app_mgp1243). If I get audio working, then I am going to try hacking the res_rtp_engine to replace ffplay |
| 13:53.31 | pabelanger | file, okay, will check that out |
| 13:54.00 | pabelanger | danjenkins, ^ that's how I plan to use external audio stream. Stream using ffmpeg |
| 13:54.28 | danjenkins | cheers pabelanger |
| 13:56.07 | pabelanger | HTTP caching of audio is likely the best way to deal with static files |
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| 15:39.34 | putnopvut | I'm hitting a bit of a snag. Check out this pastebin: http://ur1.ca/gzn2e . I'm trying to make device state events to end up as AMI events. However, it ain't happening. Is there something other than defining a to_ami callback that I need to do? In my test, I have confirmed that other device state subscriptions are invoked properly (e.g. app_queue). |
| 15:41.15 | file | putnopvut, the manager code has to be subscribed to wherever it is sent |
| 15:41.38 | putnopvut | file: the manager code has a default route set up. Does that not handle the case properly? |
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| 15:42.29 | file | that's just what happens when it gets a message, doesn't control the subscription |
| 15:42.36 | putnopvut | hrm |
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| 15:46.34 | putnopvut | file: hm, I'm not following what you're saying. I'm trying to use something else as an example. I've picked the ast_channel_varset_type(). All I can see is the type being defined, initialized, cleaned up, and having a to_ami callback defined. I'm not seeing where anything else is being done with that type within manager code. |
| 15:47.18 | file | putnopvut, nothing is explicitly - manager is subscribed to the topic the message goes out on, and the generic handler simply calls code which invokes to_ami to turn it into an AMI |
| 15:47.47 | file | but the message had to go out on a topic that the manager code is subscribed to |
| 15:48.27 | putnopvut | Okay, I see. I'm looking at message types as opposed to topics in my search. |
| 15:49.31 | putnopvut | Gotcha now I see. I need to have the device state code get the manager message router and add the device state topic as a route. |
| 15:50.09 | mjordan | 0 people. Strange. |
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| 15:50.16 | putnopvut | what? |
| 15:51.07 | file | putnopvut, yessssssss |
| 15:51.08 | Qwell | I think he just called us nobodies. |
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| 17:33.18 | leifmadsen | who? |
| 17:38.21 | rustlebee | your mom |
| 17:59.38 | pabelanger | hio |
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| 18:13.30 | rmudgett | putnopvut: Do you have any opinion about the question I had in https://reviewboard.asterisk.org/r/3414/ about the API change? |
| 18:14.37 | putnopvut | looks |
| 18:15.07 | putnopvut | rmudgett: I think just trunk and up should get the API change. |
| 18:15.52 | rmudgett | k |
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| 20:25.14 | coreyf1513 | any testsuite experts have a tip for how to access global config (/test-config.yaml) from asterisk.py? the first use of this would be ASTERISK-23369 - a config option to SIGABRT asterisk when shutdown times out |
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