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| 00:19.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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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| 01:36.14 | Slade | Does anyone have any recommendations for a good sms out provider? |
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| 02:07.05 | nickaugust | hey after upgrading to v13 I can't get my extensions.conf to load... it's in the directory with the correct permissions but it doesnt show up in the startup log, and looking at strace it seems asterisk doesnt even try to load it |
| 02:07.42 | nickaugust | do i need some new settings somewhere? I have a fairly minimal setup, not loading a lot of default modules, maybe im missing something? |
| 02:15.40 | DanFromUK | Hi, in my dialplan, I am answering a call, playing a welcome message, and then I Dial a local channel like this LOCAL/feature_5359@features/n. |
| 02:16.13 | DanFromUK | feature_5359@features then dials a PJSIP endpoint. |
| 02:16.49 | DanFromUK | When there is no answer and the PJSIP dial timesout, feature_5359@features executes a Busy command, but the original dialling channel doesn't appear to respond to it and the caller continues to hear ringing. |
| 02:17.02 | DanFromUK | Firstly, is this expected? |
| 02:17.11 | DanFromUK | and secondly, should I be doing something different? |
| 02:17.17 | DanFromUK | differently |
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| 02:21.39 | DanFromUK | Also, just before the Busy command is executed, the feature dialplan executes an external script via the System command and it passes ${CDR(duration)}, but it appears to always be zero. What could be causing that? |
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| 08:08.44 | Ast001 | Hello is there any limit in Asterisk 13 regarding number of sip or iax accounts which can I make on Asterisk server ? |
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| 08:24.13 | Ast001 | If CPU is Intel® Xeon® Processor E5-2660 v4, RAM 256 GB DDR4 ECC and bandwith 750 Mbps. Can this server hold 1000 concurent calls and 10,000 sip accounts on itself ? |
| 08:24.40 | Ast001 | if calls are gsm codec. |
| 08:25.29 | Ast001 | and if I record every call. |
| 08:29.17 | Ast001 | Server will not do anything else. Another question is there a way in Asterisk to find out how many MB particular call used (in sum). Like I can see seconds of calls I need to see MB used for every call. |
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| 08:33.58 | stefan27 | You're asking for a thumb-rule for a performance related question... But the typical general recommendation to evaluate performance is to make measurements... Reasoning about bottlenecks purely from a specification (your specification is not enough) is awfully complex |
| 08:34.16 | stefan27 | Maybe someone can help you with good guesses |
| 08:35.16 | stefan27 | as for resources go, bottleneck usually belongs to one of the categories Disk IO, CPU, Memory, Network or some application specific restriction |
| 08:39.07 | stefan27 | I know too little of asterisk and hardware ... but it's also gonna be relevant what channel driver you use, I've heard chan_sip doesn't scale well with cores... Imagine if your 10000 sip devices send a lot of sip requests for subscribe and stuff |
| 08:39.52 | Ast001 | chan_sip and chan_iax will be in use. |
| 08:40.25 | Ast001 | I think I am ok with RAM and bandwith, but not sure about CPU, hard disk will be SSD so i/o should be faster. |
| 08:41.31 | Ast001 | About CPU I read that 1 alaw/ulaw call requires 40 Mhz, g729 x5 of that, for gsm I am not sure but probably x3 (120 Mhz) |
| 08:42.11 | stefan27 | sounds like you have a fun project... If you have good load generating tools and OS and application monitoring tools, you can analyze better |
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| 08:44.31 | stefan27 | your system is gonna have a lot of workflows... You can always try guess which work-flow that will dominate the "load"... Maybe in your case it would be just sustaining the calls (shuffling rtp, writing audio to disk,...) |
| 08:45.26 | stefan27 | are you creating a system from nothing, that will instantly get the volume you're describing? |
| 08:46.28 | stefan27 | Our asterisk 13s sit in pretty weak hardware, but we never had more than 1000 sip devices and 100 simultaneous calls per machine |
| 08:48.35 | Ast001 | I need to create system from zero and system needs to handle 10,000 accounts and 1000 concurrent calls and I am trying to figure out if the strongest online.net dedicated server will handle it. What hardware do you have for 1000 sip devices and 100 simultaneous calls ? |
| 08:54.17 | stefan27 | I can't give cpu/hardware-details, but it's typically 2-6 cores x86_64 on some linux OS and the average load is like 1/10th of what i described... I cannot help you with provisioning recommendations because our scale is much lower, and our system do different things |
| 08:59.47 | stefan27 | I just wanted to tell you our story as an anecdote, because it shows you'll only be working with guesses... I collected statistics across our 100 servers/asterisk-instances and it wasn't even the case that "average calls" or "average sip user count" was a particularly good predictor of average CPU usage (but for the normal asterisk installation it probably is). Our machine that worked the |
| 08:59.47 | stefan27 | hardest turned out to be one that had few calls, but busy SIP-traffic for other reasons. |
| 09:05.27 | stefan27 | That said, making guesses is still important... Maybe what you read about alaw/ulaw is accurate, and maybe someone else here will give better pointers if you stick around :) |
| 09:07.09 | Ast001 | I had asterisk server with around 100-200 sip accounts which did up to 400-500 concurent calls (dialer calling) and since I put ssd disk in I never saw load greater than 10% and server is not so strong as one I am looking now, but codec was ulaw/alaw, it did recordings, too. Before ssd disk it was always disk i/o who caused problems. I have some experience with Asterisk, but never implemented so big installation wtih 10k accounts. So you |
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| 09:08.15 | stefan27 | your last message was cut off at " wtih 10k accounts. So you<end>" |
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| 09:09.45 | Ast001 | So you recommend to choose server and to fill it up slowly and monitor loads and when I notice degradation on calls or to big load to split installation between multiple call servers ? |
| 09:10.11 | stefan27 | There's two different problem statements "How much load can I handle granted my whole system is an asterisk X on hardware Y" or "I want to support the following load: Z... How should I provision/architecture my system?" |
| 09:11.51 | stefan27 | With the latter statement, it's possible you should consider inserting non-asterisk components in your-system to handle load balancing and stuff |
| 09:13.03 | stefan27 | that of which I have very little experience... My recommendation is what I would do, but I would not trust my judgement |
| 09:15.21 | Ast001 | Ok I will think about how to implement that project with multiple weaker call servers. |
| 09:15.38 | Ast001 | Thanks |
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| 19:20.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); 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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| 19:49.37 | P-NuT | It looks like freebsd ports does the latest version all much easier also |
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| 20:12.51 | jeffspeff | I noticed the tracker for Asterisk SCF got some updates recently. Is that project coming back to life? https://issues.asterisk.org/jira/browse/ASTSCF/?selectedTab=com.atlassian.jira.jira-projects-plugin:summary-panel |
| 20:15.21 | file | nope |
| 20:15.30 | file | I had to change things in order to disable sub-task support |
| 20:20.11 | jeffspeff | ah |
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| 20:31.43 | NirS_ | @file - any reason why Asterisk 15.5.0 won't load the Digium Opus codec properly? |
| 20:31.48 | NirS_ | we're doing a manual install |
| 20:34.10 | [TK]D-Fender | I recommend showing the actual attempt & failure... |
| 20:34.27 | [TK]D-Fender | Right now you're asking why it failed without showing anything. |
| 20:35.06 | [TK]D-Fender | heads home ... |
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| 20:35.29 | NirS_ | [Aug 20 20:34:43] WARNING[14820]: config_options.c:1058 xmldoc_update_config_type: Cannot update type 'opus' in module 'codec_opus' because it has no existing documentation! |
| 20:35.45 | NirS_ | but the .xml file is in the right place, /var/lib/asterisk/documentation/thirdparty/ |
| 20:36.27 | file | that would be the only reason it wouldn't load with that warning |
| 20:36.53 | NirS_ | well, the file is surely in there, but it doesn't seem to like it |
| 20:36.57 | NirS_ | for some odd reason |
| 20:37.10 | file | have you checked permissions? ownership? have you looked at the console at startup to see if it says anything early? |
| 20:37.28 | file | catted the file just to make sure? |
| 20:40.37 | NirS_ | fu** |
| 20:40.45 | NirS_ | the directory was missing the +x permission |
| 20:40.48 | NirS_ | stupid ! |
| 20:41.03 | NirS_ | it's 23:40, and I'm starting to make stupid mistakes |
| 20:44.22 | saint_ | Anyone tried 3CX by anychance, and know if it is possible to script (as in the extension.conf file) incoming calls depending on their caller ID ? I'm trying to persuade someone who has 3CX to switch , and want to sell this point, if they (3cx) can't do it ... i can t find anything about this in their doc. |
| 21:10.47 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:25.04 | jpsharp | yes, you can do that. |
| 21:26.00 | jpsharp | exten => _X./2108675309,1,Playback(hi-jenny) |
| 21:26.19 | jpsharp | that will match any incoming number with the caller id of 2108675309 and play hi-jenny. |
| 21:28.03 | jpsharp | Or you can do IF. exten => _X.,1,ExecIf("$[$CALLERID(num)" = "2108676309"]?Playback(hi-jenny)) |
| 21:34.45 | [TK]D-Fender | He didn't actually ask that |
| 21:34.51 | [TK]D-Fender | He asked if 3CX could |
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| 22:04.51 | jpsharp | Oh. |
| 22:04.52 | jpsharp | Doh. |
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| 22:21.43 | Samot | saint_: Yes, it can. |
| 22:22.05 | Samot | https://www.3cx.com/docs/manual/inbound-rules/#h.9dvorfngfl2b |
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| 22:40.20 | saint_ | Samot just found that too. right now, i use a script in asterisk that lets the end user black list incoming did on the fly. trying to figure out if i can do that with 3CX, which it looks i can't (i can direct an incoming did somewhere , but i don t see more flexibility than that).. so that's a point for * .. |
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