| 00:02.35 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 00:31.08 | *** join/#asterisk cemotyz09 (~cemotyz09@cpe-70-121-128-59.satx.res.rr.com) |
| 00:32.56 | *** part/#asterisk dimestop (~h@2607:5300:201:3000::16d5) |
| 01:00.35 | *** join/#asterisk pruonckk (~pruonckk@4-137-11-177.raimax.com.br) |
| 01:04.06 | *** join/#asterisk overyander (~jeffspeff@209.141.208.197) |
| 01:19.39 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 01:33.22 | *** join/#asterisk pruonckk (~pruonckk@4-137-11-177.raimax.com.br) |
| 01:37.36 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 01:51.41 | Samot | What triggers the voicemail_app to rename a folder's files into a sequential order? Is the trigger only by moving, deleting, etc via the app? |
| 02:37.27 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 03:44.35 | *** join/#asterisk simbalion` (~simba@45.77.108.203) |
| 03:45.39 | simbalion` | So I dove into FreePBX and found it's really short-bus special. Most significantly the latest stable version apparently requires PHP 5.6 which has been unsupported for months now. Can anyone recommend an open source GPL/MIT GUI for Asterisk that has better development standards? |
| 03:47.08 | Samot | You may want to look at the op list |
| 03:47.15 | Samot | And also realize that Sangoma owns Digium. |
| 03:47.20 | Samot | Ie. FreePBX owns Asterisk. |
| 03:49.19 | simbalion` | Samot: That's irrelevant, FreePBX does not fit the definition of "Active development", and I would like to find some alternatives. We're talking about open source software, if you want to lose all credibility in the open source community start talking about intellectual property rights and privileges and see where that gets you. |
| 03:49.42 | Samot | Yes, it's open source. |
| 03:49.45 | Samot | So fix it. |
| 03:49.59 | Samot | Find what doesn't work in PHP 7 and fix it. |
| 03:50.16 | simbalion` | Samot: You really are a hostile person. I'm asking for alternative GUIs that folks using Asterisk can recommend, if you don't want to contribute positively then why contribute at all. |
| 03:50.26 | Samot | Again, v15 will be on PHP 7 it is _actively being developed right now_ |
| 03:50.34 | Samot | There are options. |
| 03:50.38 | Samot | Most are older than FreePBX. |
| 03:50.46 | Samot | Including their support for Asterisk. |
| 03:51.48 | simbalion` | I understand that which is why I'm asking for recommendations from other Asterisk users. |
| 03:51.57 | Samot | Right now the FreePBX Project is the only project that is current-ish. |
| 03:52.00 | Samot | Across the board. |
| 03:52.06 | Samot | That is OSS. |
| 03:52.16 | Samot | Asterisk users don't use GUI's. |
| 03:52.29 | Samot | GUI is a four letter word here. |
| 03:56.07 | Samot | Sincerely, if you need a GUI based release to run Asterisk and it must be on PHP7.2 or higher then your best option is to make FreePBX wotk with it. |
| 03:56.12 | Samot | Honestly. |
| 03:56.29 | Samot | Otherwise you are developing it yourself or you can pick another project and do the same. |
| 04:03.24 | *** kick/#asterisk [simbalion`!sid6728@gateway/web/irccloud.com/x-eqxceknhxqahjowt] by tm1000 (cool story) |
| 04:12.30 | X-Rob | Aww man I wish I had seen that earlier |
| 04:12.35 | X-Rob | I would have loved to heckle. |
| 04:13.04 | tm1000 | such a waste of a person |
| 04:13.05 | tm1000 | "FreePBX does not fit the definition of "Active development"" |
| 04:13.37 | tm1000 | https://www.openhub.net/p/freepbx |
| 04:13.38 | Samot | "VoIP PBXes are going to blow up in a couple years" |
| 04:13.44 | tm1000 | "Very High Activity" |
| 04:29.53 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 04:36.25 | *** join/#asterisk tomaluca95_ (~quassel@kde/developer/tomaluca) |
| 05:12.50 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 05:13.44 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 05:42.15 | *** join/#asterisk dogcow (~dogcow@freebsd/user/dogcow) |
| 05:48.47 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 06:08.16 | *** join/#asterisk pchero_work (~pchero@87.213.247.82) |
| 06:20.33 | *** mode/#asterisk [+b *!~simba@45.77.108.203] by tm1000 |
| 06:20.38 | X-Rob | tm1000: hah |
| 06:20.46 | X-Rob | I was just about to ping you |
| 06:21.01 | tm1000 | the guy threatened me and if you look at his own website you'd agree |
| 06:21.04 | tm1000 | :-) |
| 06:21.06 | tm1000 | I know you did... |
| 06:38.12 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 06:41.44 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 06:44.21 | *** join/#asterisk tomaluca95_ (~quassel@kde/developer/tomaluca) |
| 06:55.58 | *** join/#asterisk Downlots (~Downlots@185.73.41.1) |
| 06:59.25 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 07:34.50 | jkroon | in dialplan, is there a way to get the SIP domain? I need to add a custom SIP header to an outbound INVITE that needs to contain the same domain as for the From: header, ie, if From: "Foo" <sip:123@my.domain> then I need my.domain in the extra header. |
| 07:35.14 | jkroon | I'd prefer to not use a dialplan variable as this needs to deploy to multiple (different domain) servers. |
| 07:50.42 | pchero_work | jkroon: Have you check this? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER |
| 07:52.15 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 07:52.22 | *** join/#asterisk sa02irc (~sa02irc@212.43.79.155) |
| 08:08.33 | jkroon | Hi pchero chan_sip. |
| 08:08.53 | jkroon | pchero_work, internal project for migration to PJSIP still ongoing. |
| 08:10.27 | pchero_work | jkroon: Then, how about this? https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIP_HEADER |
| 08:10.35 | *** join/#asterisk TandyUK (~admin@TandyUK/staff/James) |
| 08:12.04 | jkroon | pchero_work, yes, that's exactly how I'm adding the extra header, but how can I find the DOMAIN that will be used for the outbound From: header? |
| 08:12.54 | jkroon | Well, SIP_HEADER gets the incoming headers on the incoming conversation, AddSipHeader adds to outgoing INVITE. |
| 08:17.03 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 08:33.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:35.00 | *** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net) |
| 08:46.51 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 09:17.40 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:18.05 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 09:40.29 | *** join/#asterisk mukeshtilwani (67e97767@gateway/web/freenode/ip.103.233.119.103) |
| 09:51.20 | *** join/#asterisk zaf (~zaf@104.254.192.70) |
| 09:55.29 | *** join/#asterisk stevedavies (~stevedavi@197.155.252.3) |
| 10:51.38 | *** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com) |
| 10:55.24 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 11:28.59 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 11:38.29 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 12:01.33 | *** join/#asterisk phodius (~quassel@host-92-4-24-93.as43234.net) |
| 12:01.51 | i_ | hello |
| 12:02.38 | i_ | i got a simple asterick question how do i make phone call as in outbound ivr call is there a free plugin? on asterisk 13 freepbx |
| 12:17.28 | *** join/#asterisk mukeshtilwani (67e97767@gateway/web/freenode/ip.103.233.119.103) |
| 12:19.55 | i_ | is there anything, i thought it was built in? |
| 12:20.26 | bhuddah | it is best to use some form of phone for that. |
| 12:20.42 | sibiria | it's a basic, native function of asterisk. nothing complicated. but since you're working with freepbx maybe you should try #freepbx to get help on how to configure it |
| 12:20.48 | pchero_work | jkroon: I'm not clear what you want. |
| 12:21.20 | pchero_work | jkroon: OOps sorry, mis typing. |
| 12:22.04 | i_ | im using a dongle for sim access |
| 12:22.46 | i_ | and i have freepbx in vmware and i access the freepbx by my host on chrome |
| 12:23.25 | i_ | i just want to beable to call and hear call using the trunk ivr |
| 12:24.10 | Samot | https://wiki.asterisk.org/wiki/display/AST/Local+Channel |
| 12:24.11 | Samot | Read up. |
| 12:26.26 | Samot | 4:12:04 AM <jkroon> pchero_work, yes, that's exactly how I'm adding the extra header, but how can I find the DOMAIN that will be used for the outbound From: header? <-- IT's set by the fromdomain= setting in Chan_SIP or it's going to use the default domain of the PBX which is generally it's IP. |
| 12:28.38 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 12:28.53 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 12:32.17 | i_ | i know there barge call , how would i be able to hear the it when it dials i was accessing the dongle and sending texts |
| 12:33.21 | i_ | i want to hear it my computer for example there is an application called Flash Operator Panel 2 |
| 12:33.49 | *** join/#asterisk ganbold (~ganbold@173.244.215.173) |
| 12:33.54 | i_ | but there is a free one called https://sourceforge.net/projects/monast/ |
| 12:34.02 | i_ | but i dont think it got live calls |
| 12:34.57 | i_ | also can i treat my dongle like a sip as i put a custom trunk as i think it not a sip |
| 12:36.04 | i_ | you say read up but this this stuff is all for sips i am using gsm |
| 12:38.50 | *** join/#asterisk jkroon (~jkroon@165.16.203.52) |
| 12:47.41 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:48.36 | i_ | you would think i asking a simple question to 159 astrisk experts |
| 12:51.40 | i_ | Flash Operator panel 2 will proabably be what i need |
| 12:53.14 | i_ | can i connect a sip gui to it |
| 13:07.05 | *** join/#asterisk as3ty (uid332392@gateway/web/irccloud.com/x-edwiowzjbilgwnyt) |
| 13:35.36 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:38.32 | jkroon | pchero, what i'm trying to achieve is to if asterisk is going to send a From: <sip:123@fneh> I want to add a header "X-Custom-Header: stuff@fneh" |
| 13:39.18 | Samot | The From Header is going to be populated by Asterisk. |
| 13:39.39 | Samot | It's going to look at global settings for the driver and then at the peer settings for these things. |
| 13:39.55 | Samot | You can set fromdomain= to whatever you'd like and that will override the Asterisk header. |
| 13:43.28 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 13:59.52 | *** join/#asterisk somepoortech (~somepoort@207.136.208.220) |
| 14:13.05 | *** join/#asterisk LoKoMurdoK (~LoKoMurdo@fedora/LoKoMurdoK) |
| 14:14.05 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-sjspebimjpfoybif) |
| 14:14.05 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:21.06 | *** join/#asterisk sekil (~sekil@87.116.191.114) |
| 14:23.53 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 14:33.47 | *** join/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 14:34.00 | jrun | trying to parse this line: exten => s,n,GotoIf($["${RESULT}" == "notfound"]?0,1) |
| 14:34.05 | jrun | where is 0,1? |
| 14:34.36 | jrun | "line 1 of extension 0", the author tells me. not quite sure what that means? |
| 14:34.46 | Samot | Did you write this? |
| 14:34.50 | jrun | no |
| 14:34.54 | Samot | Or are you looking at some other generated dialplan? |
| 14:35.04 | jrun | latter |
| 14:35.10 | Samot | And what else is there? |
| 14:35.10 | jrun | looking at some's code |
| 14:35.14 | Samot | One line isn't enough. |
| 14:35.53 | Samot | That right now means if the statement is true it will go to extension 0 priority 1 in that same context. |
| 14:36.24 | jrun | what's priority ? |
| 14:36.43 | sibiria | the sequential step of the extension |
| 14:36.47 | jrun | and by extension 0, we mean do whatever would have happened had 0 was pressed? |
| 14:37.10 | Samot | Yes. |
| 14:37.14 | sibiria | jrun: yes, unless there happens to be a label named 0 |
| 14:37.31 | Samot | Labels are for priorities |
| 14:37.34 | Samot | Not extensions. |
| 14:37.55 | Samot | 0,start would be a labeled called |
| 14:38.06 | Samot | if you had n(start) in 0 exten |
| 14:38.07 | sibiria | ah sorry, i misread that as a colon, not a comma |
| 14:38.16 | jrun | Samot: "Labels"? you mean goto labels? ...as in C's goto |
| 14:38.18 | Samot | There is no false action right now in that code. |
| 14:38.34 | jrun | Samot: yes, i made the same mistake first time reading it |
| 14:38.37 | Samot | <PROTECTED> |
| 14:38.39 | Samot | Like that |
| 14:39.19 | Samot | jrun: Are you just trying to understand dialplan? |
| 14:39.32 | jrun | Samot: yes |
| 14:39.46 | jrun | in this case there is no extension and/or label 0 |
| 14:39.58 | jrun | then we fallthrough extension 1? |
| 14:39.59 | Samot | Then it's poorly written dialplan |
| 14:40.02 | Samot | No. |
| 14:40.10 | Samot | That's not an either/or thing. |
| 14:40.35 | Samot | When you want to move around in the dialplan you have to use the format context,extension,priority(label) |
| 14:41.02 | Samot | So the ?0,1 means what I said earlier. Extension 0 in the same context at priority 1 |
| 14:41.17 | Samot | Only if the statement is true |
| 14:41.18 | jrun | but there is no 'extension 0' |
| 14:41.25 | jrun | what happens in this case? |
| 14:41.26 | Samot | Then someone hasn't written proper dialplan. |
| 14:41.41 | Samot | It errors out and ends the call because there's no more dialplan. |
| 14:41.44 | sibiria | jrun: there might be an i extension |
| 14:41.45 | file | could be a pattern match I suppose, or you aren't seeing it because it's included |
| 14:41.52 | Samot | ^^^ |
| 14:41.57 | Samot | There could be that. |
| 14:42.02 | file | without seeing everything it's hard to understand the flow or the gist |
| 14:42.11 | Samot | Dialplan is written by the author.. |
| 14:42.27 | file | you can use "dialplan show" to kind of help though, dialplan show <extension>@<context> and it'll tell you how it would play through |
| 14:42.38 | Samot | We can look at certain syntax's like what you've shown but we can't give details on the logic. |
| 14:43.22 | jrun | sorry everyone, there is 0. but in Samot's posted syntax it's sitting at place of context: |
| 14:43.28 | jrun | exten => 0,1,Verbose(1,${CHANNEL(name)} IVRRECORD *** ${IVRMSG} - Playing menu) |
| 14:43.36 | Samot | No it's not. |
| 14:43.41 | jrun | oh |
| 14:43.47 | Samot | If the context is missing it assumes the _current context_ |
| 14:43.53 | jrun | i see |
| 14:44.04 | Samot | 10:35:53 AM <Samot> That right now means if the statement is true it will go to extension 0 priority 1 in that same context. |
| 14:44.20 | jrun | Samot: correct |
| 14:45.39 | Samot | <PROTECTED> |
| 14:45.55 | Samot | That would mean if true jump to priority 20 of the extension. |
| 14:53.20 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 14:59.50 | jrun | it clear now, thanks |
| 15:04.11 | *** part/#asterisk jrun (~jrun@unaffiliated/glphvgacs) |
| 15:16.06 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-wrasfukcqfuuqpoz) |
| 15:16.06 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:24.10 | *** join/#asterisk elguero (~miguel323@74-95-21-41-Connecticut.hfc.comcastbusiness.net) |
| 15:39.41 | *** join/#asterisk andrebarbosa (uid341412@gateway/web/irccloud.com/x-vrwvmopvrwmwordv) |
| 15:42.28 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 15:44.14 | *** join/#asterisk jkister (~jkister@pool-100-11-124-171.phlapa.fios.verizon.net) |
| 15:45.24 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 16:03.53 | jkister | i have an issue with dtmf in asterisk/1.8.15-cert2. i send it SIP INFO dtmf and then it has to send out RFC2833 dtmf. it always receives sip info correctly, but it often regenerates dtmf # into dtmf 3. i looked at changelogs for newer versions of 1.8-cert but dont see anything that seems interesting for this- any ideas what i can look at? (besides the obvious upgrade-and-see-if-that-works |
| 16:03.54 | jkister | option) |
| 16:12.08 | seanbright | if upgrade-and-see-if-that-works is not an acceptable answer, then i got nothing |
| 16:12.50 | seanbright | (1.8.15-cert2 is a 6 year old piece of software) |
| 16:14.31 | jkister | 1.8 is still probably my favorite :D maybe 11. |
| 16:14.46 | jkister | (nothing to do with why im not upgrading) |
| 16:20.06 | sibiria | gosh, running 1.8 still |
| 16:20.12 | sibiria | in the words of Ali G: bitch, is you crazy |
| 16:21.09 | sibiria | the list of important bug fixes must be longer than the Brexit deal |
| 16:21.39 | Samot | There are no important bug fixes in 1.8 |
| 16:21.51 | Samot | Not for six years at least. |
| 16:21.59 | sibiria | 1.8->16, obviously |
| 16:24.15 | file | huh, mixing 1.8 and Brexit, creative |
| 16:25.47 | *** join/#asterisk narziss (~philt@unaffiliated/narziss) |
| 16:28.41 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 16:30.18 | *** join/#asterisk miralin (~Thunderbi@81.177.58.137) |
| 16:32.28 | *** join/#asterisk overyander (~jeffspeff@209.141.208.197) |
| 16:33.57 | jkister | it very clearly has to do with timing, separing the dtmf # from surrounding dtmf makes the regeneration always work. e.g., 3 <wait 1> # <wait 1> 3 <wait 1> # <wait 1> has a 100% success rate. elinimating the wait/dialing naturally exposes the problem. |
| 16:34.08 | jkister | seems similar/same as ASTERISK-23197 |
| 16:34.59 | jkister | except this is all tcp/sipinfo and rfc2833 so no tones are really used. |
| 16:36.11 | Samot | jkister: Are you hoping for a resolution? |
| 16:36.47 | jkister | just ideas, places to look- i mean a resolution sure :) |
| 16:37.15 | Samot | Unfortunately that is going to be slim. |
| 17:04.48 | *** join/#asterisk pchero_work (~pchero@dhcp-077-249-058-090.chello.nl) |
| 17:09.01 | *** join/#asterisk rpifan (~rpifan@dslb-178-000-177-094.178.000.pools.vodafone-ip.de) |
| 18:03.56 | *** join/#asterisk i_ (~quassel@host-92-4-24-93.as43234.net) |
| 18:04.44 | *** part/#asterisk jkister (~jkister@pool-100-11-124-171.phlapa.fios.verizon.net) |
| 18:13.08 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 18:16.29 | i_ | i connect zoiper to the the extension with chan_sip but when i dial numbers on zoiper it says call reject service unavailable code 503 |
| 18:17.02 | i_ | is say chat is a premium feature does it not do voice call with the free version? |
| 18:17.42 | i_ | what does the free version do then? |
| 18:23.36 | i_ | im getting error 53 |
| 18:23.44 | i_ | *error 503 |
| 18:27.22 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:1579:3650:c9a9:5403) |
| 18:46.37 | *** join/#asterisk scgm11_ (~scgm11@r186-49-12-131.dialup.adsl.anteldata.net.uy) |
| 19:02.48 | [TK]D-Fender | <i_> what does the free version do then? <- voice obviously |
| 19:09.37 | *** join/#asterisk wonderworld (~ww@ip-88-153-137-218.hsi04.unitymediagroup.de) |
| 19:10.58 | wonderworld | i am trying to use externnotify, but my script isn't run by asterisk. i can run it from the CLI just fine with !/my/script but it isn't executed after a dropped voicemail. i set verbosity and debug to 1000 but can't spot asterisk trying to run the script at all? |
| 19:17.32 | seanbright | pastebin the relevant part of your voicemail.conf |
| 19:19.02 | seanbright | if you turn on debug output you should see a message showing that it is trying to run it |
| 19:32.25 | *** join/#asterisk rpifan (~rpifan@dslb-178-000-177-094.178.000.pools.vodafone-ip.de) |
| 19:36.17 | wonderworld | seanbright: thank you, i fixed it in the meantime...permissions as always i guess |
| 19:36.37 | wonderworld | but strangely still no notice in the cli, but it works now |
| 19:38.36 | *** join/#asterisk Downlots (~Downlots@2a02:85f:534:1000:88b1:9c05:4993:9ded) |
| 19:47.45 | seanbright | you need to turn on debug (core set debug 10) to see a notice in the CLI |
| 19:47.57 | seanbright | (assuming you've configured debug output to go to the console) |
| 19:49.54 | *** join/#asterisk Dovid (~dovid@216.66.65.171) |
| 19:50.15 | Dovid | Is ther any way to have Asterisk launch an agi (say a script) and fork it so it keeps going? |
| 19:52.05 | [TK]D-Fender | Of course not |
| 19:52.15 | [TK]D-Fender | AGi is INTERACTIVE to the channel |
| 19:53.18 | *** join/#asterisk Deeewayne (~dwayne@2605:a601:a803:7800:ffaf:2fa4:84dc:10e4) |
| 20:02.21 | wonderworld | seanbright: actually i did core set debug 10000 but didn't see anything. i guess i must have configured it to *not* go to the console. where would i configure this? |
| 20:08.00 | [TK]D-Fender | not configured what to go to console? |
| 20:08.24 | [TK]D-Fender | It's not an AGI call so I'm not sure console will grab anything at all... |
| 20:10.14 | seanbright | wonderworld: logger.conf |
| 20:18.06 | wonderworld | thanks. ok it seems to be disabled on console by default. at least on debian. |
| 20:32.12 | *** join/#asterisk defsdoor (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net) |
| 20:44.25 | *** join/#asterisk azyl (~azyl@71-80-227-144.dhcp.mghl.ca.charter.com) |
| 21:21.56 | *** join/#asterisk TandyUK2 (~admin@TandyUK/staff/James) |
| 21:22.07 | *** join/#asterisk andrebarbosa (uid341412@gateway/web/irccloud.com/x-uuruhvzopcegcacq) |
| 21:37.26 | *** join/#asterisk i_ (~quassel@host-92-4-24-93.as43234.net) |
| 22:02.06 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:06.50 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 23:26.20 | *** join/#asterisk troyt (zncsrv@2601:681:4100:8981:44dd:acff:fe85:9c8e) |