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| 00:14.51 | [TK]D-Fender | Depends what you mean |
| 00:16.05 | bbt | incoming, did to ext routing |
| 00:16.20 | [TK]D-Fender | Those terms are still rather vague |
| 00:16.25 | [TK]D-Fender | "routing" isn't a thing. |
| 00:16.28 | bbt | outgoing, dialplans to make use of specific trunks |
| 00:16.35 | [TK]D-Fender | dialplan = extensions.conf |
| 00:16.41 | [TK]D-Fender | which is all call processing |
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| 01:44.06 | bbt | trying to figure out how to make a did ring an ext. i dont see the syntax in extensions.conf ? |
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| 02:09.36 | [TK]D-Fender | It's your job to make whatever you want it to do |
| 02:16.20 | bbt | im confused, coming from a pretty gui (freepbx) to trying to work this out in cli |
| 02:16.43 | bbt | examples arent forcoming with google fu |
| 02:16.50 | bbt | trunk to ext, that is. |
| 02:17.43 | [TK]D-Fender | That's because dialplan is functionally a programming language. You don't get samples for writing exactly the thing you want, because what you want it 100% up to you. There is no such thing as "defaults' or "standard" really. |
| 02:18.46 | [TK]D-Fender | Trunk & ext mean nothing here. There is only "call coming in from device D landing in context X, asking for extension Y with callerid of Z" |
| 02:18.54 | [TK]D-Fender | you need to learn the basics |
| 02:18.56 | [TK]D-Fender | ~book |
| 02:18.56 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:18.58 | [TK]D-Fender | ^ |
| 02:19.12 | [TK]D-Fender | Learning the dialplan is 90% of Asterisk |
| 02:19.45 | [TK]D-Fender | Every step it takes, every kind of thing you let any device call in dialing is up to you |
| 02:21.29 | bbt | ok thanks |
| 02:24.58 | [TK]D-Fender | bbt> trunk to ext, that is. <- calls from a provider should match a device definition that includes the context to process the call in. When it comes in they are sending something as what they are calling (typically a DID you pay them for in the case of ITSP's). You make extens' in there to match what was sent in and the take whatever steps you want. |
| 02:26.06 | [TK]D-Fender | "Trunk" in FreePBX terms is really only the thing that defines the communications with the other end. It doesn't define when you'll want another call to use it to go out, nor does it define what to do when they send you a call. |
| 02:42.17 | bbt | thing is i know what the trunk is, where i want it to go, what extension i need it to ring, etc. just so trvivial to do with freepbx.. this is turning into a saga via cli |
| 02:45.42 | bbt | got nothing against learning it, just feel it would be wasteful as it isnt a profession i want to take up :) |
| 02:46.42 | [TK]D-Fender | It's not my profession even though I use a FreePBX system at my day-job |
| 02:47.05 | [TK]D-Fender | If you're taking on doing this yourself go learn your basics |
| 02:48.40 | bbt | what about this.. the box its replacing has asterisk14+freepbx everything is setup, would i be able to migrate the extension stuff to this asterisk 13? |
| 02:49.21 | [TK]D-Fender | There is no back & forth |
| 02:49.32 | [TK]D-Fender | FreePBX generates a shit-ton of interdependent junk |
| 02:49.38 | [TK]D-Fender | And you're not learning anything at that point |
| 02:50.02 | [TK]D-Fender | You don't start with a fully-built LEGO castle to try to build something else |
| 02:50.09 | [TK]D-Fender | and you still need to understand the basics |
| 02:50.18 | [TK]D-Fender | You're in the dilaplan. welcome to programming |
| 02:50.22 | [TK]D-Fender | dialplan* |
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| 02:50.32 | [TK]D-Fender | it's hardly complicated |
| 02:50.48 | [TK]D-Fender | Go learn how pattern matching works and some basic dialplan apps. |
| 02:51.01 | [TK]D-Fender | Learn what functions are for. Then pick what you need and make it do what you want |
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| 05:22.02 | FarhaadN | hi everyone, i have issue with asterisk cli, when i enter some command suck az "core show channesl" or somthing else, after that randomly cli show wrong character when i typing , such az "\U+515B\U+5141" ... |
| 05:22.05 | FarhaadN | also in cli show error : utils.c:1446 ast_careful_fwrite: fwrite() returned error: |
| 05:22.08 | FarhaadN | can anyone help me? |
| 05:22.39 | [TK]D-Fender | You keep showing that same pbit over and over |
| 05:22.48 | [TK]D-Fender | don't just paste those individual lines. Show what is AROUND them |
| 05:23.01 | [TK]D-Fender | And provide proper information on your system |
| 05:23.08 | [TK]D-Fender | You've told us nothing for days on this |
| 05:23.54 | Laptop01 | Hello, my auto assistant is working well for the most part. But occasionally, when a human is talking, the auto assistant won't pick up what they're saying. Which of these AMD values would I need to adjust to fix this? https://pastebin.com/UdQ0aDEz Thank you for your help |
| 05:24.05 | FarhaadN | [TK]D-Fender: what information do you need? |
| 05:24.46 | [TK]D-Fender | Laptop01, "core show application amd" |
| 05:25.14 | [TK]D-Fender | Laptop01, It already shows you what test it failed so adjust accordingly |
| 05:28.23 | FarhaadN | [TK]D-Fender: i exacly told you what happend for me, and i dont know what must i tell |
| 05:28.46 | Laptop01 | Thanks. Any suggestions on what value I should change? It's very strange, 90% of the time it will work perfectly. The 10% of the time it doesn't work, the human will have to repeat themselves so that it will pick it up. Very strange. |
| 05:28.46 | [TK]D-Fender | the full output all around those messages . |
| 05:29.04 | [TK]D-Fender | You also haven't bothered to tell us what version you're running, or anything else about your operating environment |
| 05:30.50 | FarhaadN | [TK]D-Fender: OS: VERSION="9 (stretch)" , asterisk version: 11.25.3 |
| 05:31.01 | FarhaadN | and error around those, is nothing |
| 05:31.08 | FarhaadN | AzarngMehr*CLI> core show channesl |
| 05:31.08 | FarhaadN | No such command 'core show channesl' (type 'core show help core show channesl' for other possible commands) |
| 05:31.08 | FarhaadN | AzarngMehr*CLI> \U+A15B\U+A141\U+A173 |
| 05:31.18 | [TK]D-Fender | Your version isn't supported |
| 05:32.19 | FarhaadN | ok |
| 05:32.30 | [TK]D-Fender | And those are very obviously unicode charaters |
| 05:32.38 | [TK]D-Fender | charactors |
| 05:33.43 | [TK]D-Fender | http://www.mkgmap.org.uk/websvn/filedetails.php?repname=mkgmap&path=%2Fu%2Fmaxc%2Fmain%2Fresources%2Fchars%2Fascii%2Frowa1.trans |
| 05:35.00 | FarhaadN | when i typing these charactors show on cli randomly |
| 05:35.19 | FarhaadN | with every character i click |
| 05:35.37 | FarhaadN | one of those unicode charactors show up |
| 05:35.42 | [TK]D-Fender | Go check your terminal settings |
| 05:38.09 | FarhaadN | If the problem was from the settings, should not these be displayed every time? |
| 05:38.24 | [TK]D-Fender | Laptop01, There is nothing strange. It tells you what test it didn't pass. You're not going to get 100% |
| 05:38.46 | [TK]D-Fender | tweak the one it tripped on. Retest. keep testing. |
| 05:39.29 | FarhaadN | https://pastebin.com/P2ryA0hB |
| 05:39.51 | FarhaadN | can you please check this link , every time i enter correct command these show up |
| 05:41.24 | [TK]D-Fender | Go upgrade and see if they continue |
| 05:42.06 | FarhaadN | update to 11 last version? or 13? |
| 05:42.19 | [TK]D-Fender | https://superuser.com/questions/1101668/unicode-characters-suddenly-appearing-in-asterisk-cli-ssh-console |
| 05:42.32 | [TK]D-Fender | This is already documented |
| 05:42.57 | [TK]D-Fender | GOOGLE <- your friend |
| 05:43.21 | Laptop01 | Ok, I ran that command and I'm not aware of any "test" but I'll keep researching. Thanks. |
| 05:43.32 | [TK]D-Fender | FarhaadN> update to 11 last version? or 13? <- unless there's a special reason you can't go right to 16 then that is where you should be going |
| 05:43.45 | [TK]D-Fender | Laptop01, test = USE IT |
| 05:44.50 | FarhaadN | [TK]D-Fender: i try it , thank you again |
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| 10:07.52 | Zhadnost | I've got a strange problem, When someone makes an outgoing call, asterisk doesn't seem to be aware that the call is answered and will automatically cut off after the clients ring timeout. (has anyone seen this before?) |
| 10:08.07 | Zhadnost | SIP clients, IAX termination. |
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| 10:08.25 | Zhadnost | in cdr the calldisposition is NOANSWER |
| 10:09.00 | Zhadnost | (still using chan_sip). |
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| 11:06.11 | file | what is the console output of a call? |
| 11:11.57 | Zhadnost | To be honest, I can't remember (It's been happening for a long time, I've been gertting round it by setting long timeouts) |
| 11:12.06 | Zhadnost | 1 mo. |
| 11:13.12 | Zhadnost | unfortunately there was nothing logged (I don't have debug logging on atm). |
| 11:13.38 | file | then I can do nothing except say that it's up to the remote side to say that it is answered, if it doesn't... then Asterisk won't know |
| 11:14.09 | Zhadnost | So I should file a ticket with the outgoing ITSP? |
| 11:14.25 | file | it's most likely them, yes |
| 11:14.32 | Zhadnost | Thanks. |
| 11:15.46 | Zhadnost | Looking throuh cdr, it's not only happening on a single provider |
| 11:16.13 | Zhadnost | or even a single channel driver (IAX and DAHDI {FXO}) |
| 11:16.43 | file | without console output showing the problem and ideally protocol level in the case of IAX, then we can only give guesses |
| 11:17.59 | Zhadnost | How do I know what the protocol level is? |
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| 11:18.16 | file | "iax2 set debug on" spits it into the console |
| 11:19.33 | Zhadnost | well, fwiw I'm currently on a call to Sage Support, so there's a fair chance that I'll hit the 1hour timeout before anyone answers the call and I'll get a debug then |
| 11:20.26 | file | it would have needed to be seen from start to end |
| 11:20.29 | file | not from in the middle |
| 11:22.30 | Zhadnost | Ah, well, when I'm off the phone, I'll set a short timeout on this phone and get a log then. Thanks for your help. |
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| 16:29.19 | zx81 | Hey, trying to use ARI with NodeJS - this doesn't work: |
| 16:29.20 | zx81 | play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav'); |
| 16:29.20 | zx81 | should it? |
| 16:29.21 | zx81 | https://wiki.asterisk.org/wiki/display/AST/ARI+and+Channels%3A+Simple+Media+Manipulation says: |
| 16:29.23 | zx81 | A sound file located on the Asterisk system. You can use the /sounds resource to query for available sounds on the system. You can also use specify a media file which is consumed via HTTP (e.g sound:http://foo.com/sound.wav) |
| 16:29.26 | zx81 | But I get |
| 16:29.27 | zx81 | [2019-02-18 11:18:29] WARNING[23860][C-0002f9e3]: file.c:774 ast_openstream_full: File http://www.nch.com.au/acm/8k16bitpcm.wav does not exist in any format |
| 16:37.02 | MLC | it is looking for a file on the asterisk server, not on the network |
| 16:39.35 | Samot | zx81: The file doesn't exist at that location. |
| 16:39.55 | Samot | I click on your like for that wav file and it just redirects me to the main page of the site. |
| 16:40.16 | Samot | Are you sure the site is allowing that file to be pulled directly? Have you checked what the logs say where the file exists? |
| 16:41.53 | Samot | 11:37:03 AM <MLC> it is looking for a file on the asterisk server, not on the network <-- with v14+ Asterisk Playback() now supports loading sound files via HTTP |
| 16:42.11 | MLC | yeah, I put my fingers in front of my brain again |
| 16:42.16 | Samot | As does Background(), I believe now. |
| 16:42.46 | zx81 | ok so I need to upgrade to 14 |
| 16:42.49 | zx81 | cool thanks |
| 16:42.53 | Samot | So when I click on the link in their ARI code, it doesn't load that wav file. |
| 16:44.00 | zx81 | I'm on Asterisk 13 |
| 16:44.10 | Samot | Well that's a problem. |
| 16:44.16 | zx81 | yep :-) |
| 16:44.34 | Samot | So there are two. |
| 16:44.43 | [TK]D-Fender | no 14. No 15. Move right along to 16 |
| 16:44.44 | Samot | 1) You don't have the right version to support what you are trying to do. |
| 16:45.13 | Samot | 2) The file you are attempting to use doesn't seem to allow direct HTTP connections as the site redirects attempts to load it. |
| 16:45.19 | Samot | So Asterisk will have the same issue. |
| 16:45.24 | Samot | Since it's making an HTTP request. |
| 16:45.25 | zx81 | http://www.nch.com.au/acm/8k16bitpcm.wav |
| 16:45.29 | zx81 | I just clicked it |
| 16:45.34 | zx81 | and it plays in my browser |
| 16:45.36 | zx81 | (chrome) |
| 16:45.37 | Samot | Sigh |
| 16:45.50 | Samot | 11:29:21 AM <zx81> play(channel, 'sound:http://www.nch.com.au/acm/8k16bitpcm.wav'); <-- Note the difference. |
| 16:46.05 | Samot | Yes, the HTTPS link lets me load it. |
| 16:46.15 | Samot | The HTTP link you provided in your code, doesn't. |
| 16:46.34 | Samot | So my statement still stands. |
| 16:46.38 | zx81 | ah |
| 16:46.45 | zx81 | it converts to https in the browser |
| 16:46.52 | zx81 | cool ty |
| 16:48.06 | zx81 | will try 16 |
| 16:57.53 | zx81 | cool that worked |
| 16:57.59 | zx81 | actually without changing to https too |
| 16:58.07 | zx81 | ah no |
| 16:58.10 | zx81 | I changed the code |
| 16:58.11 | zx81 | lol |
| 16:58.14 | zx81 | without me knowing |
| 16:58.17 | zx81 | :-) |
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| 17:58.38 | Janos | hey there, question, is there any way to know which asterisk module export what symbol ? |
| 17:59.09 | Janos | i'm getting a warning: Error loading module 'chan_sip.so': undefined symbol: ast_websocket_write |
| 17:59.20 | Janos | and I was wondering about that |
| 17:59.39 | Janos | I guess it can be ignored but still wondering :P |
| 18:00.07 | Janos | the main motivation was to actually try to strip down and noload all unused modules from my asterisk |
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| 18:01.19 | Janos | hence the need to actually be able to figure out which module provides what symbol so i can noload and if something fails figure out which module to load back in |
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| 19:03.45 | igcewieling | Janos: doing that is more trouble than it is worth. Here is my modules.conf, it removes the most common stuff I don't use. Use it as an example. |
| 19:04.15 | igcewieling | https://pastebin.com/U4M74P2F |
| 19:31.42 | Janos | igcewieling well that's pretty much what i wanted to achieve, remove everything I was not using |
| 19:31.49 | Janos | thanks |
| 19:35.30 | MLC | Anyone know if it is possible to log button presses from a Digium phone to somewhere? |
| 19:38.56 | [TK]D-Fender | just saying "button presses" isn't very specific. |
| 19:40.22 | [TK]D-Fender | You could possibly set up a call where you set dynamic features on all DTMF keys and log them in your own scripting that way. I think I saw (need to verify) DTMF as an AMI event notice (which might work), or run DTMF inband-only and try to decode it from recordings |
| 19:41.12 | MLC | Any button on the phone, not necessarily DTMF. Problem is that some of our queue members are hitting "Ignore" when a call comes in, but I need to be able to prove it. |
| 19:44.27 | [TK]D-Fender | You don't prove that from the phone |
| 19:44.46 | [TK]D-Fender | And no there is no complete button-pushing history on any kind of phone |
| 19:45.04 | [TK]D-Fender | What you can do is look at the ringing time to the agent in the queue log |
| 19:45.19 | MLC | I didn't think there would be. Grasping at straws. |
| 19:45.24 | [TK]D-Fender | If it didn't reach the timeout, you know why |
| 19:45.51 | [TK]D-Fender | RINGNOANSWER <-------------- |
| 19:46.08 | [TK]D-Fender | at least in the case where "ignore" = reject" |
| 19:46.15 | [TK]D-Fender | verify if your phone acts this way |
| 19:46.22 | MLC | Did that and it definitely shows that they full timeout is not being reached. Am seeing the RINGNOANSWER. |
| 19:46.24 | [TK]D-Fender | check to see if you can force it to. |
| 19:46.36 | MLC | I verified this in my dev and ignore triggers that exact log entry |
| 19:46.52 | [TK]D-Fender | you get that regardless |
| 19:46.58 | MLC | I just need to be fairly certain that there could not be another cause |
| 19:47.09 | [TK]D-Fender | it's a question of whether it actually SIP rejects the call or just silences it and rings until * gives up |
| 19:47.33 | [TK]D-Fender | go verify what it's doing now, and if it just goes silent, then you're going to have to change that |
| 19:48.02 | MLC | doesn't just go silent - it triggers the "did not answer" and goes to the next queue memmber |
| 19:48.48 | [TK]D-Fender | that happens either way |
| 19:48.58 | [TK]D-Fender | do you see it ACT on that press immediately? |
| 19:49.03 | MLC | yes |
| 19:49.06 | [TK]D-Fender | it = asterisk |
| 19:49.08 | [TK]D-Fender | not the phone |
| 19:49.24 | [TK]D-Fender | if so then you'll see how long it rang for before the rejection |
| 19:49.56 | MLC | yes - in my dev I have both ends of the call, as soon as I "Ignore" button I get a response on the callers end. In my dev it is "no operators available" because I'm the only one in the queue. |
| 19:50.37 | [TK]D-Fender | "on the callers" end is nothing like what I asked |
| 19:51.22 | [TK]D-Fender | I asked if you are sitting in * CLI staring at the call processing scrolling by do you see the call to the phone terminate instantly or not. |
| 19:51.41 | [TK]D-Fender | Go check your queue log for the ringtime |
| 19:51.44 | MLC | Sorry - I guess I misunderstood the question then. |
| 19:51.52 | MLC | Yes - the call terminates immediately on the ignore button |
| 19:52.01 | MLC | I've wtached the log and verbose_log time stamps |
| 19:52.12 | [TK]D-Fender | Then go check your queue log |
| 19:52.15 | [TK]D-Fender | it's all there |
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| 19:54.05 | Samot | MLC: If the call is ending once that button is pushed and it is being logged as RINGNOANSWER despite the timeout not being reached. That means the Ignore button is triggering a 408 message. |
| 19:54.39 | Samot | Which is what reaching the timeout would do. |
| 19:54.48 | MLC | I assume that I could see that by turning on sip debug? |
| 19:55.00 | [TK]D-Fender | Just go look at your queue logs |
| 19:55.14 | [TK]D-Fender | you've confirmed it aborts. |
| 19:55.20 | Samot | 2:46:23 PM <MLC> Did that and it definitely shows that they full timeout is not being reached. Am seeing the RINGNOANSWER. |
| 19:55.24 | [TK]D-Fender | So go get your logs and do what you have to do |
| 19:55.41 | Samot | I think his original question was pretty clear. |
| 19:55.49 | Samot | "How can I log key presses on the phone" |
| 19:56.03 | [TK]D-Fender | The answer was "no" |
| 19:56.07 | Samot | OK. |
| 19:56.09 | [TK]D-Fender | and that wasn't the reason for that question |
| 19:56.16 | [TK]D-Fender | that's jsut the means he thought he'd accomplish it |
| 19:56.23 | MLC | I've seen it in the queue logs. I'm getting RINGNOANSWER in the queue log. I just have to be able to prove beyond reasonable doubt that the human is doing it by pressing the "ignore" button. |
| 19:56.30 | Samot | 2:41:13 PM <MLC> Any button on the phone, not necessarily DTMF. Problem is that some of our queue members are hitting "Ignore" when a call comes in, but I need to be able to prove it. |
| 19:56.32 | Samot | ^^^^ |
| 19:56.35 | Samot | Sure it was. |
| 19:56.56 | Samot | An agent can press Ignore and the call will be treated like a RINGNOANSWER. |
| 19:57.02 | [TK]D-Fender | I already said that |
| 19:57.07 | Samot | OK. |
| 19:57.14 | [TK]D-Fender | and that it will get logged in a way he could tell |
| 19:57.33 | [TK]D-Fender | He doesn't have to track the PHONE |
| 19:57.48 | [TK]D-Fender | the QUEUE does it already based on how the phone acts |
| 19:57.59 | [TK]D-Fender | So he's got his answer |
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| 21:21.46 | igcewieling | leaves a plate of muffins for file |
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