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00:18.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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12:33.56 | Alblasco1702 | Sombody in here knows how to load function write? |
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12:41.09 | Samot | What? |
12:43.33 | Alblasco1702 | Samot, i try to set the last caller id in the adterisk database ... |
12:43.44 | Alblasco1702 | Samot, and i get: |
12:43.46 | Alblasco1702 | (DB(lastcallerid/102)=0610957588)") |
12:43.46 | Alblasco1702 | [Apr 24 14:42:31] ERROR[12814][C-00000002]: pbx.c:4390 ast_func_write: Function not registered |
12:45.28 | Alblasco1702 | Samot, i have loaded on modules.conf pbx_functions as found on the web and restarted asterisk. |
12:46.13 | Samot | Do: module show |
12:46.18 | Samot | ~pb |
12:46.18 | infobot | from memory, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
12:46.24 | Samot | ^^^ put the results there. |
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12:51.21 | Alblasco1702 | Samot, |
12:51.23 | Alblasco1702 | https://pastebin.com/wdf0cmPq |
12:53.01 | Samot | func_db.so <-- Is that running? |
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12:55.46 | Alblasco1702 | Samot, how do i find that out? |
12:55.55 | Samot | It would say |
12:55.59 | Samot | In that output |
12:56.14 | Samot | func_db.so Database (astdb) related dialplan functi 0 Running core |
12:58.05 | Alblasco1702 | Samot, i see nothing with running and core . |
12:58.24 | Samot | on your screen? |
12:58.31 | Samot | Or just that it's running? |
12:59.39 | Alblasco1702 | on the screen |
13:00.39 | Samot | Are you looking at a console screen? |
13:00.45 | Samot | Or are you SSH'd into the box? |
13:01.01 | Alblasco1702 | ssh into the box |
13:01.09 | Samot | Fix your screen to show the details. |
13:01.23 | Samot | You have to see what modules are loaded and running.. |
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13:55.13 | ghoti | So... 'core show channels concise' is apparently deprecated according to help in asterisk 1.8. What should I be using instead of 'concise'? 'verbose' doesn't give me full channel names. |
13:56.12 | ghoti | (What I'm looking for is the names of "stuck" channels that I can 'channel request hangup'.) |
13:59.17 | Samot | The command still exists |
13:59.40 | Samot | Though core show channels should give you want you are looking for. |
14:02.15 | ghoti | Samot: `core show channels` also abridges the names... I get "SIP/staticbean-00006" instead of "SIP/staticbean-00006826". |
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14:02.59 | ghoti | Is there another way to adjust column width, or some other command that provides machine-readable output? |
14:03.55 | ghoti | I'm trying to write a Nagios check plugin to watch for stuck channels, and I'd prefer not to use a command that is slated for demolition. But I need the full channel name. |
14:06.41 | Samot | What is the status of the channels in the list? |
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14:15.32 | ghoti | Samot: do you mean, the 'State' column from `core show channels`? That would be "Up". However: |
14:15.35 | ghoti | [root@pbx monitor]# find 2* -type f -size +10G -print0 | xargs -0 ls -lth |
14:15.38 | ghoti | -rw-rw---- 1 asterisk asterisk 125G Apr 24 10:15 2017/01/17/conf-832-832-20170117-155916-1484686751.63669-1484686767.wav |
14:15.41 | ghoti | -rw-rw---- 1 asterisk asterisk 108G Apr 24 10:15 2017/01/30/conf-829-829-20170130-143032-1485804625.65964-1485804641.wav |
14:15.45 | ghoti | That's "not right". :) |
14:16.04 | Samot | The recordings? |
14:16.09 | Samot | How is that not right? |
14:16.27 | ghoti | 125GB and still growing, for a call that was initiated in January? |
14:16.33 | ghoti | Even *our* sales folks are not that dedicated. |
14:17.01 | ghoti | So .. I try to find the channel associated with the recording and request a soft shutdown. |
14:17.03 | Samot | Has Asterisk been restarted since then? |
14:17.19 | ghoti | No. We run 24/7, so I try really really hard not to restart. |
14:17.43 | Samot | So you've made no configuration changes in 3 months? |
14:18.07 | ghoti | This is FreePBX, so it does a `sip reload` fairly frequently, but the main asterisk process hasn't been restarted. |
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14:20.02 | MacroMan | Sometimes when I'm on a call I get these errors on the cli: https://paste.ngx.cc/c895508ab1f4316c |
14:20.30 | MacroMan | Nothing seems to go wrong with the call, but the nature of the error makes me want to fix what ever is causing it |
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14:26.33 | ghoti | Samot: and ... |
14:26.33 | ghoti | # ps opid,lstart,cmd -uasterisk | grep /asterisk |
14:26.34 | ghoti | <PROTECTED> |
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14:27.34 | Samot | How many active channels do you have now? |
14:28.14 | ghoti | "5 active calls" according to 'core show channels count'. |
14:28.31 | Samot | So look at them. |
14:28.34 | ghoti | 28 active channels. |
14:29.25 | Samot | Look at their location. |
14:29.26 | ghoti | Look at them ... how? If `core show channels concise` is deprecated, what's the undeprecated version that is machine readable? |
14:29.37 | Samot | core show channels |
14:29.39 | ghoti | s/version/alternative/ |
14:29.58 | ghoti | Samot, as I mentioned before, `core show channels` cuts off the channel name at 19 characters. |
14:30.40 | Samot | I also said that core show channels concise still works. |
14:31.26 | ghoti | Yes, but you didn't say that it is no longer deprecated. Are you saying that in newer versions of Asterisk it has become undeprecated? |
14:31.33 | Samot | Unless it really really has to be removed, deprecated commands/settings never are fully removed in Asterisk. |
14:31.55 | Samot | i.e. they still work |
14:32.00 | ghoti | Why does something become deprecated? Is it not because there's a preferred alternative? |
14:32.20 | Samot | Because things are changed over the years.. |
14:32.38 | ghoti | And if `core show channels` is "the thing to use", then is there a place where I can configure the format string it uses? |
14:32.43 | Samot | They just don't remove stuff they don't really need to so people can move forward without issues. |
14:32.58 | Samot | No. |
14:33.03 | Samot | It's just core show channels |
14:33.34 | Samot | You realize these calls exist in the CDRs right? |
14:33.36 | ghoti | No configuration parameter, environment variable, etc, which can affect the output? No other mechanism to get machine readable output from Asterisk, to help with integration with external tools? We're missing something here. |
14:34.08 | Samot | The Asterisk console is for humans |
14:34.33 | Samot | Using programs to get and parse the data is not a consideration anymore. |
14:34.51 | Samot | Hence things like 'core show channels concise' becoming deprecated. |
14:35.12 | Samot | 10:15:36 AM G<ghoti> -rw-rw---- 1 asterisk asterisk 125G Apr 24 10:15 2017/01/17/conf-832-832-20170117-155916-1484686751.63669-1484686767.wav |
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14:36.06 | Samot | ^^^ The last two parts of that file name are the timestamp and channel's unique id |
14:38.48 | ghoti | Samot: the CDR in FreePBX is available throught he webgui, but I believe it's stored in a database file rather than a text file. Makes it a little harder to query from the command line. |
14:39.47 | ghoti | Yes, I can see that `date -d @1484686751` tells ms the start time of the call. |
14:40.22 | ghoti | But my two goals are (1) detect these "stuck" channels in our monitoring system, and (2) be able to manually issue a command that hangs them up. |
14:40.54 | ghoti | Are you suggesting that integration with outside software is not a consideration at all, or that there are other mechanisms by which I should be doing this? |
14:41.22 | ghoti | An API for example, that does not consist of things like parsing `concise` output? |
14:41.27 | igcewieling1 | ghoti: The Asterisk manager interface |
14:42.14 | ghoti | igcewieling1: Ah, I have seen AMI mentioned in a few places, but haven't explored it. I'll do some reading, thanks. |
14:42.45 | igcewieling1 | ghoti: AMI allows you to do stuff outside of calls. |
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14:50.49 | ghoti | igcewieling1: Okay, I've set up a profile and I can log in to the AMI interface. Now I just need to figure out how to pull out the data provided by `core show channels concise`, as well as issue the `channel request hangup` command on things that exceed acceptable limits. |
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14:56.00 | ghoti | Looks like `Action: Status` would be the AMI command to give me channel details. I can see one call that appears to have a duration of 14334260 seconds. I can definitely use this.. No call should be more than 8 hours long, ever. |
14:57.58 | igcewieling1 | I think most people solve the stuck call issue rather than try to run search and destroy scripts on stuck channels. |
14:59.06 | ghoti | I'd love to do that. We get perhaps five or six stuck conference lines per year, and they do eat up disk space. |
14:59.20 | ghoti | What would you suggest I look for? |
15:00.52 | ghoti | (That said, I do have two stuck channels at the moment, and they aren't responding to `channel request hangup`. Is there a way to "force" that hangup rather than simply requesting it?) |
15:01.40 | igcewieling1 | Setting a maximum call length sounds like a good start. |
15:05.09 | igcewieling1 | Trying to use rtptimeout and rtpkeepalive are other things to try. |
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15:29.31 | Samot | Well this is a almost a 3 month old stuck call... |
15:29.35 | ghoti | igcewieling1: while that would cut off orphaned calls at some point, it seems like just a passive "search and destroy" rather than a solution to the underlying problem. If a timeout cuts something off at 8 hours that should have disappeared at 90 minutes, how do I figure out why the call didn't end at 90 minutes? |
15:29.48 | Samot | And the only proof there is a "call" is that the wav file increases in size. |
15:31.09 | ghoti | Samot: these were indeed valid conference calls -- they just never "ended" from Asterisk's perspective. I'm having trouble figuring out how to list participants in a conference call though, to determine whether they failed to end because one participant's hangup was never registered. |
15:32.28 | voipmonk | sounds like an old issue we had at work a few lifetimes ago. Users would leave the polycom conference room phones in a conference room for hours. |
15:32.29 | Samot | Look at the location of the active channels |
15:32.44 | Samot | How many are in conf rooms? |
15:33.30 | Samot | Or the Application they are in.. |
15:33.31 | Samot | SIP/800-00017bcf s@macro-dial:1 Up AppDial((Outgoing Line)) |
15:33.39 | voipmonk | does TX and RX increase during silence? |
15:33.41 | Samot | ^^^ I have a channel that is using Dial() right now. |
15:34.12 | Samot | There should be no reason this is taking that long to find two "hung" channels. |
15:34.32 | Samot | The call recording files have the unique channel id in them |
15:34.57 | Samot | "core show channels" is going to tell you WHERE the channel is in the dialplan and what app it's using.. |
15:35.35 | Samot | You have two hung calls in two conf rooms that are also listed in the recording name. |
15:35.43 | Samot | On top of that |
15:35.57 | Samot | fwconsole restart <-- would fix the problem in 10 seconds |
15:38.44 | MacroMan | I'm still getting this "Serious Network Trouble" error. Any ideas what it could be? https://paste.ngx.cc/c895508ab1f4316c |
15:43.08 | Samot | MacroMan: Show your peer settings |
15:43.12 | Samot | ~pb |
15:43.13 | infobot | from memory, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
15:46.04 | MacroMan | https://paste.ngx.cc/a666f5f053bb1bd1 |
15:47.50 | MacroMan | Samot, is that what you wanted? |
15:48.31 | MacroMan | As I said earlier, it doesn't seem to be causing any noticeable problems. |
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16:19.35 | Samot | Yeah, sorry been on the phone. |
16:21.14 | Samot | Is that the only peer? |
16:21.39 | Samot | The error you're getting is generally due to a bad sip setting somehwere.. |
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17:46.14 | Kunsi | hi, are there any recent package builds for raspbian? sid only offers 13.14.1 |
17:47.55 | Kunsi | ⦠doesn't seem so, atleast i can't find any |
17:48.05 | Kunsi | builing from scratch it is, i think |
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