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| 15:00.24 | overyander | does * still have issues when using pjstip transports from db or is that working fine now? the last time i tried was a few years ago and it caused some unexpected behaviors. |
| 15:01.35 | file | I can't remember if they load from the database these days or not |
| 15:03.55 | file | static realtime would be equivalent, regardless |
| 15:07.41 | overyander | remind me, is there a way to differentiate between realtime and static db? |
| 15:08.27 | file | I don't understand the question |
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| 15:13.13 | overyander | you mentioned 'static realtime'. that's opposed to dynamic realtime right? |
| 15:13.27 | overyander | if so, how do you specify the different types? |
| 15:13.47 | overyander | forgive me, i'm still drinking coffee and waking up. |
| 15:15.40 | Samot | What do you mean between static realtime and dynamic realtime? |
| 15:15.48 | Samot | A static realtime would be .conf files. |
| 15:16.26 | file | yes, static realtime effectively maps .conf files into a database column and it acts as if it is a file |
| 15:16.29 | file | the wiki probably has info |
| 15:17.20 | overyander | my understanding is that * has static realtime which is where it loads config from db instead of a file when the module starts and it has dynamic realtime where it references the db on-demand like sippeers and voicemail. |
| 15:17.49 | igcewieling | I found it is usually far easier to write a script to generate actual config files from the database rather than messing with realtime static. |
| 15:18.00 | seanbright | this ^^ |
| 15:18.06 | igcewieling | Makes debugging a hell of a lot easier, that is for sure. |
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| 15:19.18 | igcewieling | That is the way FreePBX does handles configs |
| 15:26.47 | Samot | I guess the real question would be, how often are you updating things? |
| 15:27.10 | Samot | Are things changing so much and so fast that you need to read the changes in real time vs rewriting and reloading configs into memory |
| 15:29.48 | igcewieling | static realtime still requires reloading a module to update the realtime data |
| 15:29.55 | seanbright | static realtime is garbage |
| 15:29.57 | seanbright | don't use it |
| 15:30.05 | seanbright | dynamic realtime isn't much better |
| 15:30.20 | igcewieling | seanbright: dynamic realtime is worse. |
| 15:30.29 | seanbright | these opinions are my own and do not reflect the opinions of anyone else |
| 15:32.55 | Samot | I've never seen a need for it. |
| 15:33.24 | Samot | Gotten by over a dozen years without using it. |
| 15:34.14 | igcewieling | I tried chan_sip realtime, but since none of my peer register getting Asterisk to update from realtime was...complicated. |
| 15:37.11 | seanbright | i do use realtime for pjsip authentication because i generate ephemeral credentials for each session |
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| 15:45.01 | file | with everything there are pros and cons, but you have to truly think about the implications of it all before jumping in |
| 15:47.35 | Samot | True. There are plenty that have realtime working as they needed it and it suites their needs. |
| 15:47.50 | Samot | For me, personally, never saw the overall benefits of it. |
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| 16:27.09 | FuriousGeorge | hey all |
| 16:27.46 | FuriousGeorge | anyone know how i can open a ulaw file recorded by asterisk in windows? surprisingly vlc can't. ive also installed cccp |
| 16:28.31 | Samot | Well if the Russian's can't do it.... |
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| 16:29.23 | FuriousGeorge | back |
| 16:30.46 | igcewieling | Are you SURE it is an ulaw file and not a GSM with a .WAV extension? |
| 16:31.42 | igcewieling | Audacity can open just about anything. |
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| 16:35.40 | sibiria | convert with sox, or import it as raw data in audacity |
| 16:36.04 | sibiria | exit |
| 16:36.07 | sibiria | oops |
| 16:37.19 | FuriousGeorge | igcewieling: exten => 123,3,Record(asterisk-recording%d:ulaw) will try audactiy, thanks for the tip |
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| 16:39.07 | igcewieling | tell it to use .wav .wav is raw ulaw wrapped in a header players to read. |
| 16:40.30 | FuriousGeorge | igcewieling: im trying to avoid asking the lady to re-record, but i that's what im planning to do in the worst case |
| 16:41.08 | igcewieling | It would be easy to convert from ulaw to wav. Asterisk could do it in the CLI, but I can't find the command anymore. |
| 16:41.24 | file | file convert |
| 16:41.32 | sibiria | asterisk -x 'file convert <path> <path>' etc. |
| 16:41.33 | overyander | when loading app_queues.so asterisk throws a warning "Realtime table queue_members@myvoip requires column 'paused', but that column does not exist!" and also the same for column 'uniqueid'. I have double checked that those columns do exist. here's the full output https://paste.centos.org/view/ebf334ee |
| 16:44.09 | FuriousGeorge | ty |
| 16:44.40 | FuriousGeorge | audacity makes sound if i import raw data, but i can't find the settings to decode correctly |
| 16:44.49 | FuriousGeorge | so ill convert per the suggestion |
| 16:44.58 | igcewieling | Try this: https://www.sangoma.com/audio-converter/ |
| 16:45.31 | sibiria | FuriousGeorge: audacity has an actual "u-law" encoding type to ease things up |
| 16:45.54 | igcewieling | the digium website and "file convert" only converts between Asterisk formats, so it is less useful than you might expect |
| 16:46.51 | igcewieling | FuriousGeorge: Better to record to something like g722 or sln16 then you'll have high quality originals to convert from. |
| 16:47.33 | FuriousGeorge | sibiria: i had found that, but it didn't help |
| 16:47.55 | sibiria | FuriousGeorge: u-law, no endianess, 8000 hz |
| 16:48.13 | sibiria | also 1 channel mono, of course |
| 16:48.56 | FuriousGeorge | sibria: i'd tried that, and it just makes a 1s noise |
| 16:50.12 | FuriousGeorge | sibria: my mistake i had the wrong sampling rate in there |
| 16:50.33 | FuriousGeorge | definitely gonna use wav or gsm next time though |
| 16:56.58 | igcewieling | FuriousGeorge: only use GSM if you hate your callers. |
| 16:57.27 | FuriousGeorge | right cuz its compressed a bunch, no? wav would probably be a better choice |
| 16:58.21 | igcewieling | Think of GSM as "cell phone quality" and ulaw/wav as "PSTN quality", though that is a major oversimplification |
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| 17:04.20 | FuriousGeorge | if i clean up these files and export them as wavs, is a 16bit PCU wav likely to be fine? |
| 17:05.15 | FuriousGeorge | or maybe unsigned 8bit PCU is better |
| 17:05.35 | FuriousGeorge | or maybe unsigned 8bit PCM*** is better |
| 17:06.11 | igcewieling | um, asterisk doesn't support 16-bit wav |
| 17:07.08 | igcewieling | Stop making it so complicated. Use sln16 when possible, use wav (not WAV or WAV49) when sln16 is not possible. Transcoding to ulaw/alaw from snl16 doen't take up much resources. |
| 17:07.37 | FuriousGeorge | im selecting export format as WAV (Microsoft) and encoding as 8-bit unsigned PCM |
| 17:07.51 | igcewieling | don't forget the 8khz too |
| 17:08.32 | FuriousGeorge | igcewieling: im not getting an option for sampling. let's see if it keeps the same. as to the complication: i have these ulaw files now, and i'm just trying to avoid asking to re-record |
| 17:08.47 | FuriousGeorge | even if i convert to wav, i may still want to clean them up, so ill need to know how to export |
| 17:09.19 | igcewieling | clean them up how? |
| 17:09.42 | FuriousGeorge | for instance, there's a pause at the beginning and end of the files that is not necessary |
| 17:10.10 | FuriousGeorge | actually, it looks like exporting as WAV (Microsoft), and encoding as ULAW may be the best bet |
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| 18:06.41 | FuriousGeorge | i have wav files that im editing for use with asterisk. I'm always using the format WAV (Microsoft) |
| 18:06.53 | FuriousGeorge | when i use encoding is ulaw i get the message: Not a supported wav file format (7). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are supported with a lowercase '.wav' extension. |
| 18:07.38 | FuriousGeorge | when i use 16-bit signed PCM, i get no errors, but i also hear nothing |
| 18:08.15 | FuriousGeorge | s/format/header... the HEADER is WAV |
| 18:08.39 | TandyUK2 | you were told earlier asterisk doesnt support 16-bit wav |
| 18:09.23 | TandyUK2 | 8-bit unsigned 8khz mono |
| 18:09.31 | FuriousGeorge | TandyUK2: asterisk is telling me PCM encoded 16 bit. If I use Unsigned 8-bit, i get an error |
| 18:10.03 | TandyUK2 | sorry what do you mean "asterisk is telling you".. how? where? |
| 18:10.03 | FuriousGeorge | it's no longer in my buffer, but i think the error explicitly tells me to use 16 |
| 18:10.41 | seanbright | asterisk supports pcm encoded 8/16Hz .wav files |
| 18:10.43 | FuriousGeorge | console |
| 18:10.51 | seanbright | mono |
| 18:10.52 | FuriousGeorge | similar to the error i posted above |
| 18:15.27 | FuriousGeorge | nope, it's mono. 8000 sample, and 16-bit |
| 18:15.28 | FuriousGeorge | https://pasteboard.co/IImaZsU.png |
| 18:15.46 | FuriousGeorge | maybe it's the Microsoft Header? I could try a different type |
| 18:16.20 | seanbright | you're saying that file does _not_ work? |
| 18:17.06 | file | I'm on vacation this week, so I'm at least trying not to |
| 18:17.14 | seanbright | hilarious |
| 18:18.48 | igcewieling | compare the output of "file file /var/www/html/admin/modules/ivr/sounds/en/no-valid-responce-pls-try-again.wav" with the output of your own sound file. |
| 18:19.13 | igcewieling | It should say "RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz" |
| 18:19.35 | igcewieling | If it doesn't, then the file is not in the require format. |
| 18:20.00 | igcewieling | s/file file/file |
| 18:22.00 | FuriousGeorge | res-open.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 18:22.18 | FuriousGeorge | thats my file. let me call the person who took the test call, as im not on site. maybe we are miscommunicating |
| 18:31.24 | FuriousGeorge | yeah, so the test worked. the person onsite forgot to mention that she had heard the recording |
| 18:31.38 | seanbright | simple mistake |
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| 20:19.55 | FuriousGeorge | i want to run asterisk -rvvvv, dump the console log into a file, then run some tests using call files, and finally kill the console |
| 20:20.29 | igcewieling | FuriousGeorge: that won't do what you want it to do. |
| 20:21.15 | FuriousGeorge | however, i want to do this within php code. everything is working except the kill the console part, because i'm using asterisk -rvvvvv > my_log.txt& to start asterisk. since the process never terminates, i never get the PID with which to kill the session |
| 20:21.31 | igcewieling | If you MUST do that, use the program "script" program to capture the console output. |
| 20:21.46 | igcewieling | FuriousGeorge: all the same info is in the asterisk log file. |
| 20:22.07 | FuriousGeorge | igcewieling: that's what i'm doing now, but i want to automate it. |
| 20:22.51 | FuriousGeorge | im assuming script won't work from within php code but maybe it will |
| 20:23.31 | igcewieling | Are you doing this inside a standalone php script or as a web page script? |
| 20:23.41 | FuriousGeorge | standalone |
| 20:24.23 | FuriousGeorge | i dont see how this will work. i can run script& followed by asterisk -rvvv&, but script won't get anything. if idont use & then the php script will stop until the process is killed |
| 20:24.43 | igcewieling | script takes all output and logs it to a file until the process terminates. |
| 20:24.49 | igcewieling | isn't that what you wanted? |
| 20:25.00 | FuriousGeorge | igcewieling: yeah, but within a php script? ill try it |
| 20:25.35 | igcewieling | I suspect what you want to is not possible without spending a lot of time on it. |
| 20:26.26 | FuriousGeorge | what's frustrating is that it almost works, except i cant get the pid to kill asterisk |
| 20:26.33 | igcewieling | Hell, it would be easier to edit /etc/asterisk/logger.conf and add a file destination, asterisk -rx "logger" reload, then remove the entry when done and reissue the logger reload. |
| 20:27.01 | FuriousGeorge | i could just parse the output of ps at that point |
| 20:27.53 | igcewieling | I was referring to getting a set of logs during a specific timeframe. |
| 20:30.34 | igcewieling | FuriousGeorge: you are aware that you will still see info from all the other active calls and dialplan right? |
| 20:30.48 | FuriousGeorge | yes, im aware |
| 20:31.21 | FuriousGeorge | i know what you mean, but i was saying that i could parse ps to get the pid, and use that to kill asterisk |
| 20:31.51 | FuriousGeorge | so i wouldn't have to temporarily move the log output, |
| 20:31.57 | FuriousGeorge | or restart asterisk |
| 20:32.11 | FuriousGeorge | i could just reload the logger |
| 20:32.31 | FuriousGeorge | but that still has the issue of also disappearing those active calls you mention from the real log |
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| 22:11.46 | Slade | latest in Slade's interviewing tips.. in the first 5 minutes of the interview, dont list all the diseases you have |
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| 22:55.56 | bkervaski | Asterisk 16.latest ... incoming PJSIP immediately forwarding to another server ... directmedia=no ... destination server's audio won't play *unless* I have the middle server Playback(something) .. odd .. channel answered or not answered .. any ideas? |
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| 23:02.00 | bkervaski | (re-post for TK if you're feeling generous) Asterisk 16.latest ... incoming PJSIP immediately forwarding to another server ... directmedia=no ... destination server's audio won't play *unless* I have the middle server Playback(something) .. odd .. channel answered or not answered .. any ideas? |
| 23:06.51 | bkervaski | 175 .. so sad .. IRC is dead :( |
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| 23:10.48 | boch | hi all, anyone knows why im getting duplicated DialEnd AMI event for a cancelled call? the only difference between events is the SequenceNumber header |
| 23:13.49 | bkervaski | boch one event for both channels |
| 23:22.08 | boch | bkervaski, but both events are for the same channel |
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