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| 03:37.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0 (2018/10/09), Security Only: 15.6.1 (2018/09/20); 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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| 04:20.58 | PixelAsterisk | Hi everyone! I'm looking for a company that can provide contracted support to a cluster of Asterisk servers with customised dial plans (Digium support said they couldn't help due to the customisations) - does anyone have any recommendations? I've tried calling the businesses listed on Voip-info without responses yet. Thanks :) |
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| 09:12.40 | ep4sh | good day! i have a interesting trouble: trying to set new asterisk server, but when i do regiestartion - i see no reg on remote server and see it on new server. Look it https://paste.ubuntu.com/p/kBK7q6CBMv/ |
| 09:12.48 | ep4sh | why so? |
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| 13:36.05 | n3t | What happens in a dialplan after using Dial() app? According to https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial there is "g" option ("Proceed with dialplan execution at the next priority in the current extension if the destination channel hangs up"). |
| 13:36.28 | n3t | It seems like a common behaviour and in Dial it's an option. |
| 13:37.02 | n3t | Does it by default jump into "h" extension? After successful dial? After any? Where can I read more about Dial's behaviour? |
| 13:37.44 | xpheres | hello, can questions about security and fail2ban be done here? |
| 13:38.23 | [TK]D-Fender | n3t, the default is to end the channel after a bridged call ends. That's what it does. Everything extra is in the dial options like "g" you just found |
| 13:39.01 | xpheres | someone is trying brute force against my asterisk server and fail2ban does not ban it |
| 13:39.09 | Samot | n3t: What are you trying to do? When Dial() executes the dialplan is no longer processed until the Dial() completes and returns a status. In order for Dial() to complete, at least one side has to hangup. |
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| 13:55.02 | n3t | Samot: first of all, I'm trying to understand before building production setup. I understand that Dial() completes after hangup. But I can't answer following questions. 1. What if the connection (call?) won't be established in the first place? Does it count as a hangup? 2. Where should I process DIALSTATUS? What is the best practice? |
| 13:55.45 | n3t | > Dialplan execution will continue if no requested channels can be called, or if the timeout expires |
| 13:55.48 | n3t | Oh, I see. |
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| 13:58.43 | Samot | This is all in the wiki |
| 13:59.08 | Samot | DIALSTATUS will return a status telling you what happened... |
| 13:59.14 | Samot | A connected/bridged call is "ANSWERED" |
| 14:00.03 | n3t | I mean, where should I log DIALSTATUS? I guess it shouldn't be the next priority in the dialplan (unless using "g" option). |
| 14:01.06 | n3t | Should I just use "g" option? Should I log DIALSTATUS in "h" extension? What If requested channel can't be called? |
| 14:01.25 | Gugge | if you just want to log it, use cdr |
| 14:01.44 | Gugge | if you want to do something with it, either check it right after Dial(), or in the h extension |
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| 14:03.54 | n3t | Just to be sure. If I would like to check it right after Dial(), I should use "g" option, right? |
| 14:04.01 | Samot | No. |
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| 14:04.07 | Samot | Did you read of all the g option? |
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| 14:04.17 | Samot | "When the _destination_ channel hangsup" |
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| 14:04.26 | Samot | That happens when a certain side hangs up |
| 14:04.32 | n3t | Oh, right, my bad. |
| 14:04.46 | Gugge | you could try to set this up, and see what happens :) |
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| 14:05.31 | n3t | I'll RTFM for a bit more, I guess. Thanks for patience anyway. |
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| 15:04.39 | monsterco | hello - is there another Freepbx channel now? |
| 15:04.45 | monsterco | I see Sangoma shot it down |
| 15:09.13 | [TK]D-Fender | no they haven't |
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| 15:26.40 | igcewieling | When looking at "pjsip show channelstats" there is a TX Lost column. How does Asterisk know if the transmit failed other than getting an error back from the network stack? |
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| 15:43.45 | Samot | It's UDP. |
| 15:43.56 | Samot | There's no expectation of delivery... |
| 15:44.06 | Samot | So "transmit lost" is no reply. |
| 15:44.14 | Samot | in expected time. |
| 15:44.58 | file | RTCP |
| 15:45.09 | Samot | Or that |
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| 16:02.02 | TECFALL | can you jump major versions of sip firmware on the cisco 7961G? |
| 16:02.10 | TECFALL | or do you have to step up? |
| 16:02.44 | TECFALL | I tried jumping from 8.4.4 to 9.4 and it is stuck on registering |
| 16:08.54 | Samot | TECFALL: What does the documentation say? |
| 16:09.47 | TECFALL | the documentation on cisco's website is giving me a 403 from the download page... |
| 16:09.53 | TECFALL | Samot ^ |
| 16:12.06 | [TK]D-Fender | <TECFALL> can you jump major versions of sip firmware on the cisco 7961G? <- yes |
| 16:12.16 | [TK]D-Fender | there is no implicit concept of "jump" |
| 16:12.42 | [TK]D-Fender | it loads what you give it and if the firmware and configs match and the settings match what your environment demands then it works |
| 16:12.55 | [TK]D-Fender | I was running 9.2 on the 7941's I briefly had |
| 16:13.54 | TECFALL | [TK]D-Fender: thanks! |
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| 16:26.55 | igcewieling | thanks file. That sounds like TX loss is actually the remote side reporting RX loss |
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| 16:32.13 | igcewieling | I'm seeing around 18% packet loss on one call, as shown by pjsip show channelstats" which sounds crazy |
| 16:32.38 | igcewieling | no ping packet loss, but that isn't unexpected. |
| 16:40.19 | TECFALL | if i want to add more codecs after i have already installed asterisk... can i simply rerun makemenuselect and then reinstall without losing existing configs? |
| 16:40.53 | [TK]D-Fender | you only touch configs if you tell it to |
| 16:42.48 | TECFALL | so just to be clear: it is okay to rerun the following : ./configure..., make menueselect, make, make install. Then reload asterisk correct? |
| 16:43.42 | [TK]D-Fender | Sure |
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| 18:25.45 | igcewieling | Has anyone heard of GLO COM? If you've used it, was it any good? It looks to be some crappy "unified communications platform" from a company which sells GUIs for Asterisk (not Sangoma or Digium) |
| 18:33.28 | trmg | That is a first for me. |
| 18:33.32 | trmg | Never heard of 'em. |
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| 19:09.01 | jsmith | igcewieling: Nope, never heard of them. |
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| 19:35.36 | drmessano | igcewieling: Gorgeous Ladies of Communication? |
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| 20:16.32 | igcewieling | https://www.bicomsystems.com/ Their web designer needs to be beaten with a book about user interfaces. |
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| 20:45.01 | trmg | Woah, the initial background of the phone on a table or whatever rotating like that is very disorienting...at least for me. |
| 20:45.48 | trmg | Does anyone here play with ZoIPer on Android? |
| 20:47.51 | trmg | I'd like to deploy it, but in my limited testing I've noticed that outbound audio has an annoying amount of latency. Inbound audio is totally fine with very acceptable latency. Any tips/tricks to improve outbound (from the Android phone) audio latency? FWIW all my testing has been local and not over 4G/LTE data. |
| 20:48.06 | trmg | I don't know if there is some Android system setting and/or ZoIPer setting or something I'm just missing... |
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| 21:53.22 | PixelAsterisk | Has anyone used or can recommend an asterisk consulting/support company that will provide support and troubleshooting for customised asterisk servers? |
| 21:56.34 | Samot | PixelAsterisk: What are you looking for and how are these "customized"? |
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| 22:12.41 | PixelAsterisk | Hi Samot, the customisations are entirely dial plan/config file/agi changes, however Digium have said they can't provide support for systems with custom dial plans |
| 22:13.20 | Samot | I'm not sure what that means. |
| 22:13.30 | PixelAsterisk | I have 3 asterisk servers load balanced by opensips, I'm purely looking for a company that can provide remote support to log into the asterisk servers and troubleshoot issues when they arise and can log in from time to time to do maintenance/check for any potential problems |
| 22:13.30 | Samot | By default when you install Asterisk, there is no dialplan. |
| 22:14.39 | Samot | So basically, you need standard Asterisk support/management. |
| 22:14.49 | PixelAsterisk | I haven't changed any of asterisks base code, all the changes are in extensions.conf, sip.conf, features.conf, etc and we're running the rest of our configuration in realtime |
| 22:15.09 | Samot | Oh. |
| 22:15.25 | PixelAsterisk | yes correct, although Digium said they couldn't provide that, which is why I'm looking for "the next best thing" as it were :P |
| 22:16.20 | Samot | Good luck. Real-Time is a sparsely used thing... |
| 22:18.12 | PixelAsterisk | why is that? we've got about 60,000 lines of dial plan information across our customers so it makes the most sense for us to manage it within a database, is there a better alternative? |
| 22:18.29 | PixelAsterisk | (this is why we're looking for consulting companies to help manage/advise on this) |
| 22:19.32 | Samot | Yeah, I'm not sure why you would need 60K lines of dialplan. |
| 22:20.13 | PixelAsterisk | we have 5,500 DID's each with their own unique call flow |
| 22:20.44 | Samot | Still don't see the need. |
| 22:22.59 | PixelAsterisk | there must be some knowledge you have of asterisk that I don't have (which is likely, I'm not an asterisk tech), as far as I knew, you have to have each DID's call flow mapped out for it eg: check if its during this time range, if so dial these extensions, then play this sound, then go to this queue, then go to this voicemail box. |
| 22:23.37 | PixelAsterisk | and it seemed like the most efficient way to do that was to store it in a database where we can give clients easy interfaces to make changes to their respective dial plans without having to directly edit extensions.conf |
| 22:24.09 | PixelAsterisk | and I possibly incorrectly thought the way to read in dial plans from the database is via realtime |
| 22:24.23 | Samot | Yes. |
| 22:24.36 | Samot | But the real question is how often is that database being updated? |
| 22:24.55 | Samot | Followed by, aren't there other options that could be used. |
| 22:25.19 | PixelAsterisk | it's not updated often, roughly 300 times a day |
| 22:25.49 | Samot | Why does it have 300 updates a day? |
| 22:25.53 | Samot | New customers? |
| 22:25.55 | PixelAsterisk | I don't know what those options might be, like I said I don't really know asterisk - always keen to explore better options |
| 22:26.17 | Samot | Real-Time has always lacked in depth documentation and support. |
| 22:26.38 | Samot | It has it's limitations which require you to use both it and static anyways to get features that you want |
| 22:26.40 | PixelAsterisk | new customers, existing customers porting numbers, customers changing dial plans to reflect changes in their business etc |
| 22:26.52 | Samot | OK, so basically what I do. |
| 22:27.31 | Samot | I'll update dialplan, when needed but you can design hooks, subroutines, etc that can pulled from AstDB, other sources, etc.. |
| 22:28.39 | Samot | Even things like FreePBX rely on macros/routines that are used globally..... |
| 22:29.21 | PixelAsterisk | I think that's what we're doing? we have macro's pre-created in extensions.conf, and the data stored in the database is just "macro-x, variable1, variable2, variable3", and the dial plan is essentially created from a collection of macros being called |
| 22:29.29 | PixelAsterisk | if I've understood what you're saying |
| 22:29.48 | Samot | Right but Real-Time isn't really needed for that. |
| 22:29.49 | PixelAsterisk | and I'm guessing AstDB is just whatever database we're running (in this case mysql) |
| 22:30.01 | Samot | No, Asterisk requires AstDB |
| 22:30.06 | PixelAsterisk | oh ok, that I didn't know |
| 22:30.07 | Samot | Even in static mode. |
| 22:30.46 | PixelAsterisk | so you're saying storing in astdb avoids the need for realtime - is that right? |
| 22:32.03 | PixelAsterisk | and if I'm reading between the lines to my original topic, that might make me eligable to fit within digiums criteria for their support plans? |
| 22:32.12 | Samot | Well, in my experience I've run up to 50K+ users on Asterisk systems + OpenSER and Real-Time was never used. |
| 22:34.44 | Samot | Sadly, while what you do is in my wheelhouse (and what I do for others like you)....Real-Time... |
| 22:37.48 | PixelAsterisk | I'm understanding from what you're saying that real-time is a problem, also that there are better alternatives, which is great - that's the kind of information and support we're looking for, but hopefully from this you can see why we're looking for a company that can provide that kind of support and consultancy, otherwise we're rather in the dark here as to what's best practice, and what's problematic |
| 22:41.55 | Samot | PixelAsterisk: PM |
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