| 00:04.08 | file | it's a sub-second unique identifier, not part of the timestamp |
| 00:04.13 | file | it's an atomically incremented integer. |
| 00:04.45 | degenerate | is there some reason it might be having an off by one issue? and how can i fix that? |
| 00:05.06 | file | are you sure there's not two channels? |
| 00:07.51 | degenerate | @file: last ten records in my crddb: https://pastebin.com/eQtVcTCz event reported from ami: https://pastebin.com/2zeeKjmS |
| 00:08.02 | degenerate | its always off by 0.00001 |
| 00:08.23 | file | because there's two channels? |
| 00:08.45 | file | that AMI event shows that SIP/flowroute-0000ab8b is linked to 1542931120.50375 |
| 00:08.54 | file | 1542931120.50375 would be another channel, and would be the one with the CDR as the caller |
| 00:09.19 | degenerate | ok. i see. i must be getting the wrong event in my logic. thank you for clarification. |
| 00:12.17 | degenerate | yes, you were right @file. thank you. |
| 00:13.05 | file | always have to look at the total picture :D |
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| 02:29.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.1 (2018/11/14), Security Only: 15.6.2 (2018/11/14); 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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| 11:04.05 | pchero_work | I have a question about CDR.. In a normal call flow, the duration should be longer than billsec. Is it right? |
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| 11:31.31 | pchero_work | I have a strange CDR, in my CDR, the call duration is bigger than billsec. It's just normal call... |
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| 14:51.00 | russoisraeli | Hello folks. Good time of day. Anyone here manage to successfully configure phone calls from Twilio with chan_sip? |
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| 16:07.17 | Ryushin | So I had the issue with the dialplan stopping after hangup. I've tried change the dialplan to have 'h' but I'm getting an error. http://ix.io/1ufr |
| 16:07.32 | Ryushin | Invalid priority/label 'h' at line 488 of extensions.conf |
| 16:07.50 | Ryushin | I've been googling quite a bit to try and figure out what is wrong with my syntax. |
| 16:12.23 | file | your first priority for 'h' is same instead of exten |
| 16:12.45 | file | and your remaining 'same' is incorrect for it |
| 16:19.32 | Ryushin | file: So this finally showed no errors and worked: http://ix.io/1ufu |
| 16:20.15 | Ryushin | So the extension is 2797 is no longer valid because it is hung up, and I just use exten => h,n,System( |
| 16:20.50 | Ryushin | It worked. I was worried that it would act more as a global |
| 16:20.55 | Ryushin | I'll have to test some more. |
| 16:22.08 | Ryushin | file: Thank you for your insight yesterday. It put me on the right track. |
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| 16:57.11 | Ryushin | file: So yep, adding it to the other faxes extensions made it act as a global. I'll google some more to see if I can tie it just to the extension. |
| 16:57.29 | file | it's context specific. |
| 16:57.54 | file | if you want greater control of when, then https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 16:59.39 | Ryushin | thanks. Time to read more. :) |
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| 17:22.54 | Ryushin | file: hangup handler seem to work. Thanks much. |
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| 17:42.12 | csguth | Hi all |
| 17:42.26 | csguth | I'm struggling with setting up my asterisk as SIPS trunk. |
| 17:43.17 | csguth | Just managed to use pjsua app as sip client. I'm able to register to an external SIP server. |
| 17:43.38 | csguth | But I'd like to configure Asterisk's pjsip module make the registration |
| 17:44.15 | csguth | but i keep receiving |
| 17:44.15 | csguth | *CLI> pjsip show [Nov 23 17:24:42] ERROR[3436]: res_pjsip.c:3931 endpt_send_request: Error 120099 'Cannot assign requested address' sending OPTIONS request to endpoint my-itsp |
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| 18:33.31 | sibiria | if i understand the documentation correctly, Goto is supposed to jump to the 'i' extension in the same context if the context you specify for Goto does not exist |
| 18:33.55 | sibiria | but that's not happening. the channel just hangs up, complaining about no invalid handler being declared |
| 18:34.30 | sibiria | is this a known bug or is the documentation just incorrect? |
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| 18:40.17 | igcewieling | No. |
| 18:40.39 | igcewieling | 'i' is mainly run when an IVR response is invalid |
| 18:42.09 | sibiria | "If the location that is put into the channel information is bogus, and asterisk cannot find that location in the dialplan, then the execution engine will try to find and execute the code in the i (invalid) extension in the current context. If that does not exist, it will try to execute the h extension." |
| 18:42.16 | igcewieling | try reading this: https://www.voip-info.org/asterisk-i-extension/ |
| 18:42.20 | sibiria | documentation for Goto |
| 18:42.55 | sibiria | i'm Goto'ing to a new context, not to another extension within the same context |
| 18:43.45 | sibiria | so i suppose Goto itself can't handle bogus contexts |
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| 18:43.51 | igcewieling | if it isn't doing what is document then report a bug. |
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| 19:16.18 | [TK]D-Fender | sibiria, Show us the dialplan and the failure |
| 19:25.19 | sibiria | https://pastebin.com/raw/Erqpg6kz |
| 19:28.05 | sibiria | the exception extension (e) doesn't catch that either |
| 19:28.21 | sibiria | so i have to conclude that Goto() will only go there when you jump within the same context, not outside |
| 19:29.58 | [TK]D-Fender | Looks like the way you described. Investigating further |
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| 19:56.18 | joe_nux | Hi there! |
| 19:57.12 | joe_nux | Anyone knows a business that would allow me to use pretty much the same features of Asterisk but without having to deal with configuring everything myself? We currently use FluentStream but we find that they miss some features, so similar thing but more advanced. |
| 19:57.42 | sibiria | freepbx, possibly |
| 19:58.31 | joe_nux | sibiria: hmm thanks, so you're not talking about a business but a front end that makes Asterisk config easy? |
| 19:58.45 | sibiria | that's what freepbx is |
| 20:00.33 | joe_nux | thanks. So from there I could very easily route phone numbers from a SIP trunk to auto-attendants, etc? And place outbound calls as well? |
| 20:00.48 | sibiria | yes |
| 20:02.04 | joe_nux | like e.g. a calls gets to 310-xxx-xxxx, it's a SIP trunk that gets to my Asterisk server (configured by FreePBX). Then it goes to an auto-attendant and e.g. if people press 2 it gets secretely routed to another 10 digit phone ouside, which might be my cell phone e.g.? |
| 20:02.29 | joe_nux | (sorry if this is basic, I'm not too familiar with everything yet) |
| 20:03.59 | sibiria | i've never used freebpx so i can't answer for your rather specific requirement there, but i know it offers quite advanced exchange features |
| 20:04.02 | sibiria | it's what it's designed for |
| 20:04.31 | sibiria | and your example is entirely doable with asterisk itself |
| 20:04.38 | sibiria | nothing strange about it |
| 20:04.39 | joe_nux | ok, so the scenario I mentioned seems like a normal use case? |
| 20:04.44 | sibiria | to me, yes |
| 20:04.47 | joe_nux | ah :) ok thanks |
| 20:04.56 | joe_nux | awesome |
| 20:05.28 | joe_nux | I wonder why so many businesses make everything so complex... probably too many sales people out there who forget that some people just want access to the features, that's it :) |
| 20:05.40 | sibiria | K.I.S.S has been forgotten |
| 20:07.04 | joe_nux | :) |
| 20:15.50 | joe_nux | Wah so there's a Distro as well fully ready? That's totally awesome! Talk about easy user experience! |
| 20:17.24 | joe_nux | If it's really that awesome the only thing it's missing is a Donate button! |
| 20:17.47 | [TK]D-Fender | They sell commercial modules you may want |
| 20:17.55 | [TK]D-Fender | consider those your "donation" |
| 20:18.05 | [TK]D-Fender | After validating that it suits your needs first of course |
| 20:18.25 | joe_nux | I see :) |
| 20:18.59 | file | there's also a sale this weekend, https://www.sangoma.com/cyber-weekend/ |
| 20:19.25 | joe_nux | https://www.limestonenetworks.com/dedicated-servers/clearance.html Would the server at $71 be suffisant for a small business that will be unlikely to have more than 2 simultaneous calls? |
| 20:20.28 | sibiria | if you're concerned about asterisk's resourcefulness, don't be |
| 20:20.59 | sibiria | the specs you show there will allow asterisk to keep hundreds of simultaneous calls going |
| 20:21.46 | sibiria | i can't say how resourceful (or not) the rest of freepbx is, but asterisk itself is incredibly efficient in resource use |
| 20:21.50 | joe_nux | ok, that's what I was thinking :) Performance is cheap these days :) We live in a great area for geeks :) |
| 20:24.22 | sibiria | we have a box with circa year 2010 low-end tier hardware, running a virtual machine configured with 2gb RAM and a single virtual cpu, that handles incoming calls, and it can keep over 100 simultaneous channels going through asterisk without problems |
| 20:25.13 | sibiria | we have a box with circa year 2010 low-end tier hardware, running a virtual machine configured with 2gb RAM and a single virtual cpu, that handles incoming calls, and it can keep over 100 simultaneous channels going through asterisk without problems |
| 20:25.17 | sibiria | oops sorry for repaste |
| 20:25.22 | sibiria | stupid keyboards and their arrow keys |
| 20:27.52 | joe_nux | stupid keyboards and their arrow keys |
| 20:28.10 | sibiria | ;P |
| 20:28.41 | joe_nux | well, what takes ressources should be mostly encoding/streaming, so it should be mostly Asterisk right? |
| 20:29.10 | joe_nux | FreeBPX is really just some front end web server, there's no reason that it takes anything |
| 20:29.31 | sibiria | if you're transcoding audio that will cost some CPU resources, but you don't necessarily need to |
| 20:29.56 | sibiria | if you're working with the PSTN it's beneficial to keep everything ulaw/alaw |
| 20:30.35 | joe_nux | ok |
| 20:31.56 | sibiria | (my example above, with our box handling DIDs, is without any transcoding. only alaw/ulaw passthrough) |
| 20:33.08 | joe_nux | I see. I'm wondering if it's reasonnable these days to use only VOIP via LTE network and no phone network at all |
| 20:35.46 | joe_nux | My goal is to make a PBX system that filters all calls from many numbers I have, then forwards that to voicemail or to my cellphone in certain conditions, but then I can see a lot of details before accepting the call. |
| 20:36.13 | joe_nux | So if I forward to an actual cellphone number I will need to put all that in the Caller ID string |
| 20:37.11 | joe_nux | For ex, what number people dialed, what auto-attendant menu keys they pressed, etc. |
| 20:37.57 | joe_nux | But since ALL my calls will go through the PBX, I wonder why use the cell network in the first place, instead of connecting an app to that PBX and use the LTE data network only |
| 21:07.37 | joe_nux | Is "Zulu UC" Android app an upcoming app for controlling FreeBPX? :) |
| 21:11.51 | [TK]D-Fender | nope |
| 21:12.14 | [TK]D-Fender | Some of what you need might end up requiring some custom dialplan. |
| 21:12.52 | [TK]D-Fender | just FYI. Not everything can be done in the GUI. GUI makes boring business stuff simple. complex filtering and manipulation will be a custom job on top. |
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