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| 00:29.21 | boch | hi all, could anyone give me a hand debugging why Asterisk 15.2.1 takes over 150% CPU ? I have 20 chan_sip extensions logged in a queue, answering calls made by a daemon coded by myself using AMI. I have already disabled cdr and cel, even converted MOH files to ulaw, same codec extensions and provider use |
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| 01:23.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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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| 03:05.03 | aissar | hi guys from Asterisk Community |
| 03:06.38 | aissar | I have a problem with Local Channels |
| 03:06.47 | aissar | If anyone can help me |
| 03:07.10 | aissar | I create a gist to explain my problem |
| 03:07.11 | aissar | https://gist.github.com/aissarmurad/b8f1c6bc773f1148cfb0c43191cec35a |
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| 03:13.12 | kunwon1 | aissar: maybe this will help you https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance |
| 03:14.47 | aissar | Thank you kunwon1... but It's don't works |
| 03:15.10 | aissar | and I don't know why |
| 03:15.37 | kunwon1 | on incoming: Set(__Foo=test) |
| 03:15.48 | kunwon1 | on outbound: Verbose(${Foo}) |
| 03:15.53 | kunwon1 | does that work? |
| 03:15.58 | aissar | No |
| 03:16.09 | kunwon1 | hrm :/ |
| 03:16.09 | aissar | Only works in the same leg |
| 03:16.26 | aissar | :( |
| 03:16.37 | kunwon1 | how are you creating the outbound leg |
| 03:17.22 | aissar | foo.bar (Local/5213@from-queue/n from hint:5213@ext-local) |
| 03:17.54 | kunwon1 | I don't understand what that means |
| 03:18.01 | kunwon1 | are you using Dial() ? |
| 03:18.27 | aissar | I'm using Queue... |
| 03:18.41 | aissar | I log a Local Channel in the queue |
| 03:18.50 | aissar | *log in |
| 03:19.28 | aissar | That's associated with a SIP channel |
| 03:19.58 | aissar | Like in Figure 3 - https://wiki.asterisk.org/wiki/display/AST/Channels |
| 03:21.02 | kunwon1 | the Queue application accepts a gosub or macro argument to run on the called party's channel after the call is connected |
| 03:21.12 | kunwon1 | that might work as a workaround |
| 03:21.19 | aissar | Variable Inheritance only works in the same leg |
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| 03:24.44 | aissar | I've tried that's approach too |
| 03:25.10 | aissar | Maybe I missed something... I'll see again it |
| 03:25.10 | kunwon1 | Queue(testqueue,t,,,3000,,setthings^${Foo},,,) |
| 03:25.34 | kunwon1 | then [macro-setthings] with Set(Foo=${ARG1}) |
| 03:26.02 | aissar | Wow... I'll try this |
| 03:26.09 | aissar | thanks :D |
| 03:26.14 | kunwon1 | np, i hope it works |
| 03:26.31 | aissar | me too |
| 03:29.59 | kunwon1 | could try a gosub instead of a macro too, queue allows for that |
| 03:30.18 | kunwon1 | i'm going to bed, g'luck |
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| 06:35.40 | aissar | I found the solution kunwon1... thank you very much |
| 06:35.43 | aissar | https://gist.github.com/aissarmurad/b8f1c6bc773f1148cfb0c43191cec35a |
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| 09:17.34 | elcontrastador | hi all, I'm trying to cutover my server from chan_sip to pjsip. My pjsip ITSP config is working fine. My legacy Cisco router has PRI connections (which have some DIDs coming in on it). I struggling to produce a working pjsip config for that. My chan_sip config is here: https://gist.github.com/elcontrastador/9d4178d07812376213bed362af92f847 |
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| 11:00.03 | Worldexe | it looks like while im in lua extension, (for ex, executing Wait()), commands from manager like SerVar are not being processed |
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| 11:00.07 | Worldexe | am i right? |
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| 11:15.57 | tzafrir | Worldexe, I'm not sure what you mean. Manager commands don't run in a channel context. They run in their own separate context. |
| 11:16.08 | tzafrir | Somewhat like CLI commands. |
| 11:17.13 | Worldexe | well, i clarified it ia bit in dev channel ) |
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| 11:18.28 | tzafrir | <Worldexe> well, the situation is: im waiting inside lua context like this: ' ... while true do app.Wait(0.1) ... end' and it seems SetVar commands from AMI are not getting through |
| 11:20.01 | tzafrir | What do you mean by "not getting through"? Do you check the value of the variable on the channel? |
| 11:21.35 | Worldexe | yep |
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| 11:37.48 | bittis | guys i have a question regarding queues and penalties, in a ring all strategy, is it possible to have a set of extensions ring first and if no answer for a set of second extensions to ring after? |
| 11:38.05 | bittis | i tried this using penalties, played around with queue timeout, but it doesn't seem to have any effect |
| 11:38.10 | bittis | using the ringall strategy |
| 11:40.23 | Samot | You have to do multiple queues. |
| 11:41.13 | bittis | no other way right? |
| 11:41.24 | tzafrir | Worldexe, I suspect that the AMI action wasn't properly done. Wrong channel name? |
| 11:42.24 | Worldexe | well, all this setup works OK on another machine with plain old extensions.conf; the only thing I changed - I rewrited that part into Lua and it stopped working |
| 11:43.11 | Worldexe | are you sure there are no locks preventing that SetVars from being executed before I leave lua context? |
| 11:44.13 | Worldexe | maybe there are some error in my code but i triple checked everything |
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| 11:47.02 | bittis | one other question, using realtime, if i set the queue timeout there to a specific value, it seems that once it reaches it it dials back into the queue again, is this expected behaviour? |
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| 12:03.30 | bittis | queuerules seems to be a viable option of implementing this, but it doesn't seem to allow you to specify qhich queue they apply to |
| 12:05.35 | *** join/#asterisk boch (~fseratti@host222.190-139-97.telecom.net.ar) |
| 12:06.12 | boch | morning all, could anyone give me a hand debugging why Asterisk 15.2.1 takes over 150% CPU ? I have 20 chan_sip extensions logged in a queue, answering calls made by a daemon coded by myself using AMI. I have already disabled cdr and cel, even converted MOH files to ulaw, same codec extensions and provider use |
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| 12:40.12 | Samot | boch: What are the server resources: CPU/RAM? |
| 12:40.56 | Samot | Oh, they left. Must not be that big of a deal. 10 minutes is all they waited. |
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| 12:55.59 | jkroon | @Samot, @boch is back :p |
| 12:56.07 | jkroon | <Samot> boch: What are the server resources: CPU/RAM? |
| 13:01.11 | *** join/#asterisk flujan (~flujan@189.39.53.65) |
| 13:03.21 | flujan | Hi guys. I have two asterisk boxes and one sip provider. Box 1 SIP calls box 2. Box 2 dial throught the TELCOâs sip. The telcoâs sip return cause 504 and box 1 receives cause 603. Any ideas? |
| 13:08.11 | Samot | 504 is a server timeout error |
| 13:08.39 | Samot | If the Telco is returning a 504 you need to call them. |
| 13:09.03 | flujan | Samot: shouldnât asterisk send back to machine 1 the 504 code instead of 603 |
| 13:09.55 | Samot | No. |
| 13:09.59 | Samot | It's not a pass through |
| 13:10.10 | flujan | Samot: I donât get it. |
| 13:10.14 | Samot | Box 1 calls Box 2, that's one channel.. |
| 13:10.19 | Samot | That's it.. |
| 13:10.25 | Samot | Box 1 is calling Box 2. |
| 13:10.32 | Samot | It's up to Box 2 to deal with that call. |
| 13:10.37 | Samot | It doesn't even need to leave Box 2. |
| 13:10.42 | Samot | But you have it doing that.. |
| 13:10.53 | Samot | So it has to do a Dial() and open a second channel. |
| 13:11.28 | Samot | So the 504 is between Box 2 and the Telco |
| 13:11.42 | Samot | But when it can't bridge the call from Box 1 |
| 13:11.51 | Samot | Box 2 is telling Box 1 "I am declining your call" |
| 13:12.35 | Samot | 603 is "Call accepted but could not be processed for X reason" |
| 13:12.43 | Samot | Or would not be processed. |
| 13:12.57 | Samot | 603 is the right response from Box 2 to Box 1 |
| 13:16.14 | flujan | is there a way to fix it? bridge the call ? |
| 13:17.07 | Samot | Sure. |
| 13:17.31 | Samot | You can handle the 504 response from the Telco and send the call to a recording that says there's an error. |
| 13:17.40 | Samot | You're telco is not completely the call.. |
| 13:17.47 | Samot | That is the bigger issue here. |
| 13:18.01 | Samot | Not that Box 2 isn't passing back the response. |
| 13:18.12 | Samot | completing* |
| 13:18.40 | Samot | That 504 is a server error from the Telco, you need to figure out why they aren't processing your calls prperly. |
| 13:32.56 | flujan | Samot: thanks. I would be great to return back the call. Cause 27 just works⦠|
| 13:39.00 | Samot | jkroon; I don't think boch is paying attention. |
| 13:39.18 | jkroon | Samot, apparently not :). |
| 13:39.29 | Samot | So must not be that big of an issue |
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| 13:40.43 | jkroon | glad for him ... my CPU woes are only just getting started ... currently peaking at 3 core maxed out and the number of SIP trunks is set to double in the next three months. |
| 13:43.00 | Samot | And what are you running? |
| 13:50.51 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
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| 13:55.15 | bittis | so queuerules.conf works for changing the QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY and allow ringing to extensions of different penalty categories, for some reason there is a few seconds of delay, around 5 or so, before the second group rings |
| 13:57.32 | [TK]D-Fender | you still have agent ringtime & timeout to respect |
| 13:58.09 | [TK]D-Fender | much like periodic announcements only happening after X amount of time but between agent calls. |
| 14:00.09 | jkroon | Samot, i've got enough RAM and CPU ... my concern is there is seeminly an individual asterisk thread that is maxing a single core, but until I can pinpoint exactly what that is and what's causing it ... |
| 14:00.38 | jkroon | that said, i suspect it's asterisk -rx "file convert"'s that's running. so busy looking to offload that to a different setup to confirm. |
| 14:01.17 | file | that would be limited to a single thread, and transcodes as fast as it can so it can peg a core |
| 14:01.26 | jkroon | normally in my experience disk io is the first thing to choke on asterisk servers with loads of SIP clients. |
| 14:01.35 | jkroon | file, exactly why I suspect the transcode process. |
| 14:06.01 | *** join/#asterisk pruonckk (~pruonckk@135-143-11-177.raimax.com.br) |
| 14:06.16 | Samot | How many SIP clients? |
| 14:07.10 | jkroon | disk becomes an issue in my experience around 400-500 SIP clients. this is due to astdb. once you put that onto ramdisk my estimates are that chan_sip should on my hardware at least scale to ~10k clients. |
| 14:07.23 | Samot | And what is the CPU/RAM? |
| 14:07.26 | Samot | Humor me. |
| 14:07.38 | Samot | Because you just claimed 3 cores at max. |
| 14:08.04 | jkroon | RAM is ~1MB/client and CPU is the next bottleneck (processing incoming sip frames in a single thread). |
| 14:08.12 | Samot | What is it? |
| 14:08.14 | jkroon | rtp switching is multi-threaded, so that's not a problem. |
| 14:08.29 | *** part/#asterisk flujan (~flujan@189.39.53.65) |
| 14:08.45 | jkroon | CPU on that box right at the moment is running at 280%. |
| 14:08.59 | jkroon | according to top. |
| 14:09.28 | Samot | WHAT IS IT? |
| 14:09.34 | Samot | How many CORE how much RAM |
| 14:09.36 | jkroon | asterisk is consuming around 1G of RAM at the moment with 700 odd peers. |
| 14:09.43 | jkroon | oh, 64GB RAM, and 16 cores. |
| 14:09.48 | Samot | Thank you. |
| 14:09.55 | jkroon | so as i say, i'm not too worried about that just yet. |
| 14:10.07 | Samot | "3 core max" |
| 14:10.15 | Samot | With no context. |
| 14:10.25 | Samot | 3 out of 64 is not as bad as 3 out of 6 |
| 14:10.42 | jkroon | well, it goes to around 300% CPU as per the graph. so sorry if i was unclear about what i meant. i meant it consumes up to three cores worth of CPU. does that make more sense? |
| 14:10.51 | Samot | Sure |
| 14:10.58 | Samot | But again 3 out of ? |
| 14:11.03 | Samot | That's what I was getting it. |
| 14:11.13 | Samot | If you are at 300% and have 5 cores, you're in a pickle. |
| 14:11.27 | Samot | If you're at 300% and have 64 cores, you're not so much in a pickle. |
| 14:11.34 | jkroon | :). indeed. |
| 14:12.21 | file | mmm pickles |
| 14:12.35 | Samot | On a stick. |
| 14:12.45 | Samot | Because it's Merica. |
| 14:12.50 | Samot | And everything is better on a stick. |
| 14:13.20 | Samot | Oh and deep fried. |
| 14:13.30 | Samot | Deep fried pickles on a stick. |
| 14:13.34 | Samot | It's a thing somewhere, I know it |
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| 14:17.39 | tzafrir | Press '1' in top to see per-core display (and press 'H' (shift-h) for per-tHread display to make sure display is completely useless) |
| 14:36.33 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 14:37.15 | bittis | [TK]D-Fender: respected, 15 second queue timeout (realtime), 30 seconds in Queue(), and the rule is set for 10 seconds (assumed better have it run before the timeout) |
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| 15:11.55 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:19.26 | boch | Samot, jkroon sorry guys im using a new irc client not used to. The server has 16 cores and 24 Gb of RAM |
| 15:20.11 | boch | its running Centos 6.9 and Asterisk 15.2.1 |
| 15:20.42 | Samot | OK. |
| 15:20.46 | Samot | 100% = 1 Core |
| 15:21.13 | Samot | So you are running 1.5 cores out of 16. |
| 15:21.16 | Samot | That's not bad. |
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| 15:24.07 | jkroon | Samot, for 20 SIP peers it's horrible. We do 700 (with up to 400 active calls) with 300%. Granted there may be variance in CPU spec, but still. |
| 15:25.08 | jkroon | boch, that's 150% *asterisk* - not 150% overall on the system? what kernel version? |
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| 18:21.04 | igcewieling | my "stop users from setting dumb passwords" script is at https://pastebin.com/A3jQfAZ5 in case anyone finds it helpful. use like this in voicemail.conf: externpasscheck=/usr/local/bin/sm_voicemail_password_check |
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| 19:40.18 | kfife | anybody know where I can do a progrommatic lookup of NPA-NXX to get a primary city (e.g. as a supplement to CNAM)? |
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| 19:42.48 | [TK]D-Fender | https://www.google.ca/search?q=npa-nxx+city+list&oq=npa-nxx+city+list |
| 19:43.01 | [TK]D-Fender | Pretty sure you'll find a downloadable list... |
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| 19:46.17 | kfife | [TK]D-Fender: thanks. I've just stumbled onto Vitelity, who appears to have an API that will return RATECENTER:STATE. I'm not sure that ratecenter will be human-friendly, but it will mean that never have to re-import the newest list to get new allocations, overlays and splits. |
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| 20:14.20 | Samot | Rate Center isn't going to give you a city. |
| 20:15.03 | Samot | Multiple cities can be in the same Rate Center. |
| 20:15.48 | Samot | Also, Vitelity's API is for *their* Rate Center presence, not all the Rate Centers. |
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| 20:17.52 | igcewieling | I can export and upload my list, it also covers some areas other than NANPA too. I' |
| 20:17.55 | igcewieling | ll look into it. |
| 20:18.13 | Samot | localcallingguide.com |
| 20:18.28 | Samot | They have an API, free, that will give you a bunch of XML data on numbers. |
| 20:18.35 | Samot | Like what Rate Center it's out ot |
| 20:18.37 | Samot | Like what Rate Center it's out of |
| 20:18.39 | Samot | Or state |
| 20:18.53 | Samot | The RBOC, etc. |
| 20:21.27 | igcewieling | try this: http://help.nyigc.net/tmp/nanpa+.mysql.gz |
| 20:22.54 | igcewieling | The original database came from NANPA, but I've cleaned up a lot of the location names. |
| 20:24.08 | igcewieling | To clarify: NONE of the data came from the LERG |
| 20:25.51 | Samot | That's why I like LCG. It's funding and gets LERG updates. |
| 20:26.06 | Samot | Well sponsorship more than funding. |
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| 20:38.45 | igcewieling | I don't like the LERG because it costs so much. |
| 20:43.51 | Samot | Oh yeah, I know. |
| 20:44.07 | Samot | Hence me just using LCG and dealing with their lag.. |
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| 20:44.20 | Samot | With the LERG, they aren't instant.. |
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| 21:28.18 | kfife | Samot: this is a good resource |
| 21:28.23 | kfife | LCG |
| 21:32.12 | kfife | So LCG is laggy, eh? |
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| 21:58.09 | Samot | Laggy in terms of LERG updates |
| 21:58.14 | Samot | Not in terms of API requests. |
| 21:58.59 | Samot | The LERG DB is $$$$$$ |
| 21:59.08 | Samot | They get it via one of their sponsors. |
| 21:59.22 | Samot | So there is a lag in LERG updates |
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