IRC log for #asterisk on 20191223

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06:00.12*** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.2 (2019/11/21) 16.6.2 (2019/11/21) Standard: 17.0.1 (2019/11/21); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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12:51.23KioubitHi, I have been trying for a few hours now to setup TLS1.2 signaling encryption and media encryption but I haven't gotten very far.. Could somebody take a quick look at my config?
12:51.23Kioubithttps://pastebin.com/V6Rprd9R
13:01.25TandyUKhttps://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport
13:01.32TandyUKsome of your settings dont look valid
13:03.02sibiriaalso avoid ssl. it's no longer secure
13:05.17sibiriaor, uh, maybe chan_sip uses the old openssl methodology, in which case "sslv23" should be able to negotiate for tls 1.x
13:06.20sibiriaeither way, i don't think there is a "method" option for chan_sip. it's named tlssomething
13:06.57KioubitYeah I would like to have TLS1.3 for both media and signaling if possible
13:07.14KioubitSo this is only possible with pjsip?
13:08.13sibiria"sslv23" in old openssl methodology implies TLS, all supported versions
13:08.23sibiriai just don't know what chan_sip does, exactly
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13:08.45sibiriai've only used SIPS/SRTP with pjsip
13:09.17sibiriabut your configuration options are still not correct. "method" is for pjsip, not chan_sip
13:09.27TandyUKafaik, chan_sip only does up to tls1.2
13:11.15KioubitI mean i can try switching to pjsip then with the conversion script and try to configure encryption there
13:13.54sibiriapjsip appears a bit daunting at first, but using the wizard to set up endpoints/contacts helps a lot
13:15.10KioubitIs there a wizard?
13:18.12sibiriapjsip_wizard.conf
13:32.02KioubitHmm the conversion script gives me this: https://pastebin.com/mhb5iz82 which doesnt work either
13:32.09KioubitI think I'll give pjsip_wizard.conf a try then and see if that helps me understand pjsip better
13:32.10KioubitThanks anyway for the suggestions
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13:59.45engine20191hello with what tool can I test how many simultaneous calls you can have, I'm trying a banana pi m64 = 4 core arm64 bits and 2 GB ram
14:07.47SamotHow many do you need?
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14:27.32engine20191I'm trying with this sipp tool
14:27.40engine20191https://wiki.asterisk.org/wiki/display/AST/Measuring+SIP+Channel+Performance
14:31.14SamotThat still wasn't an answer to my question.
14:33.49sibiriaengine20191: you can have 150 calls ongoing on that computer without any problems
14:34.07sibiriaassuming you are not transcoding between something more costly than slin<->g.711
14:34.35engine20191sibiria: thanks
14:34.49engine20191codec ulaw
14:35.04engine20191the machine is for small bussiness
14:35.17engine2019120 extension
14:35.28sibiriayeah that's peanuts. no problem
14:37.42Samotsibiria: It's a RasPI
14:37.49SamotSo let's not get crazy.
14:39.42engine20191I downloaded ubuntu bionic arm64 with kernel sinovo 4.4 in this asterisk was installed from the apt with freepbx so that the client can easily manage it
14:41.11sibiriaSamot: yes but it's capable of it
14:41.19sibiriamy test rig is a pine64 with 1gb of ram
14:41.24sibiriasimilar performance Arm CPU
14:41.42SamotAnd what is the life span of the card?
14:41.45sibiriawhen i run 100 calls on it during tests, it sits at around 30% total CPU usage across all cores
14:41.54sibiria150 calls and it goes up to about 50% total CPU
14:42.11SamotAnd what is the life span of the card?
14:42.14sibiriawith a mix of no transcoding and transcoding from 8khz wav -> ulaw/alaw
14:42.52sibiriacard? of the small-board computer?
14:43.19SamotThey  use SD cards, yes?
14:43.25SamotFor the HDD.
14:44.00sibiriayes they do. the lifespan of that depends entirely on how much writing you do and obviously the brand and size of SD card you use
14:44.09SamotCorrect.
14:44.17SamotSo at some point 150 calls will be a lot of I/O
14:44.19sibiriai recommend logging to RAM and regularly flushing, isntead of directly to the storage
14:44.29sibiriait depends entirely on how oyu set up logging and CDRs
14:44.36SamotOK.
14:44.38sibiriai do batched CDRs and so
14:44.44sibiriait's a requirement, reall
14:44.47SamotI would never trust a system that had to do a 150 calls to a RasPI
14:44.55SamotThat's just me.
14:44.57sibiriaimpossible to do any heavy writing directly to SD. everything comes to a complete crawl
14:45.06sibiriaso that particular setup does need some effort put into it first
14:45.47sibiriai'm merely underlining that the RPi (and similar) are efficient enough due to how well-written and high-performing Asterisk is
14:45.59sibiriabut, again, the setup needs specific planning for this to work
14:46.37sibiriaan SBC with eMMC or regular SATA is really preferrable
14:49.47SamotYes.
14:50.07SamotJust because something can do something doesn't mean it should be doing it.
14:51.16SamotLike right now in another room, a guy wants to use his router as a CA and CRL server.
14:51.38SamotWhich, sure it can do that but that's more for internal stuff not being a public facing CA/CRL.
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15:32.53sibiriai personally think an SBC like an RPi or Pine64 is perfectly fine for a SoHo setup, even with just SD storage. i wouldn't go close to the idea for anything "Big Corporate"
15:33.31SamotSure but a SOHO setup isn't a 150/calls.
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15:51.18electronic_eelI think the sd storage is a real problem when using the raspi for some infrastructure thing
15:51.51electronic_eelI use a odroid-xu4 for some years now as monitoring system at home (Icinga 2)
15:52.55electronic_eelIt has a pluggable emmc storage. That works really well, even if monitoring is a write-heavy workload for such a system (about 100 metrics measured every few minutes, graphing and so on)
15:53.19electronic_eelI would suggest to use something similar for asterisk too
15:54.50sibiriaSD (just like USB flash storage) is a problem for anything doing small frequent writes
15:55.44sibiriait's just not a suitable medium for that scenario, no matter how fast the storage is for large sequential writes
15:56.45*** join/#asterisk zBeeble (~zBeeble@2001:1928:1::35)
15:57.38zBeeblewhy-oh-why can't we control the codec negotiation in the dialplan? _or_ why-oh-why is codec negotiation so dumb?
15:58.25drmessanoDumb how?
15:58.27zBeebleI have traffic where some comes in asking for g.729 and some doesn't.  I want to offer g.729 to the outbound side IFF it was offered on the in.
15:58.48zBeebleit would be sufficient to offer the codecs I get on the incoming.
15:59.48SamotDoes the outbound side support it?
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16:00.24zBeebleyes.  The problem is I _don't_ support g729 ... but I'm happy to take advantage of the passthru.
16:00.37zBeebleso I want the in and out codecs to match.
16:00.41drmessanozBeeble: Do you have g729 at the top of the allow list?
16:00.47igcewielingNo, you don't want to use passthru.
16:00.55electronic_eelyeah, more influence on the codec selection would be really nice. I have some problems with g.711 - calls from mobile phones sometimes claim to support it, but the calls then have a 20% chance of breaking. So I'd like to disable g.711 for calls from mobile phones (detection based on remote phone number)
16:01.18SamotThat's not how codecs work.
16:01.23zBeebleI use passthru all the time.  The goal is to make the incoming unaware of the identity of the terminating leg.
16:01.26SamotWhen the INVITE comes in, they offer what they want to use.
16:01.36SamotSo if it's using g711 they are offering g711
16:01.56SamotBoth sides must offer the codec.
16:02.03electronic_eelyes, they are offering g711, but I know that they have a shitty implementation of it
16:02.10SamotIt's the standard.
16:02.15SamotIt is literally the PSTN standard.
16:02.19electronic_eelso I want to disable it for them
16:02.19zBeeblesure... but the problem is: I offer to the terminating side and if I offer g.729, they'll always take it.
16:02.30electronic_eeloh, sorry mixed it up, g722 I meant
16:02.42zBeeble... whereas if I don't offer g.729, they'll take g.711.  so I need to only offer it when it's offered to me.
16:02.53Samotelectronic_eel: So don't offer g722
16:02.58SamotSimple. Done.
16:03.07filePJSIP provides https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER for control, and there is also a PJSIP_SEND_SESSION_REFRESH to renegotiate as well
16:03.17Samot^^^
16:03.18zBeebleproblem is ... some incoming traffic is g.729 only.
16:03.25SamotThat. I was looking for that in the wiki
16:03.29electronic_eelno, I want to use g722 for all calls to other peers
16:03.45igcewielingI found all my G729 problems went away when I got a transcoder card.
16:03.55zBeebleyeah... I've seen that.  Last time I tried pjsip, everything went to heck in a handbasket.
16:03.57SamotThen you're transcoding.
16:04.10zBeeble... and I'm not sure I have time to implement that before lossing the traffic.
16:04.14electronic_eelfile: but PJSIP_MEDIA_OFFER seems to be only for Dial, not for incoming calls
16:04.23SamotCorrect.
16:04.31SamotBecause your offer goes back to them in a reply
16:04.36fileit may work on incoming too, I forget how it works
16:04.36SamotNot a new request
16:04.59zBeebledoes the dialplan have access to the incoming offer in pjsip?
16:05.04SamotYou maybe able offer a re-invite
16:05.09SamotMaybe.
16:05.15SamotTo change the codec.
16:05.29fileWhen read, returns the codecs offered based upon the media choice.
16:06.39SamotYeah, I think those two are it.
16:07.15SamotPJSIP_MEDIA_OFFER() and then PJSIP_SEND_SESSION_REFRESH() to issue an UPDATE/re-INVITE with the new codec offer.
16:07.56Samotelectronic_eel: That might be the answer to your mobile g722.
16:08.01fileI vaguely recall chan_sip has some special dialplan variables for stuff
16:08.46SamotzBeeble: You can just use PJSIP_MEDIA_OFFER() on the incoming channel to see if it's g729 and then use it to set g729 on the ountbound channel.
16:08.48zBeebleAFAICT, only SIP_CODEC* ... and they don't seem to be populated with anything on the inbound call.
16:08.55SamotBut I'm not sur ehow that would work with passthru
16:09.10electronic_eelwhere would I do the PJSIP_SEND_SESSION_REFRESH in the dialplan? right before doing the dial to my local endpoint? or after the connection to the local endpoint is established?
16:09.17zBeeblepassthru kicks in if the codec are equal.
16:09.21Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_SEND_SESSION_REFRESH
16:09.33fileyou don't need to call PJSIP_SEND_SESSION_REFRESH unless you want to initiate a re-negotiate after the session is established and answered
16:09.44SamotThat's what he wants to do.
16:09.49fileah, then yeah
16:10.03SamotNot use g722 based on the from user
16:11.21Samotelectronic_eel: It's probably going to be the first thing you do on a matching for an incoming call.
16:11.45zBeeblenothing in the pjsip doc there say that media_offer is populated by the inbound call.
16:11.50SamotCheck the from user to see if its mobile and then do the session refresh to force it to g711 if it's using g722
16:13.21electronic_eelso I do the PJSIP_SEND_SESSION_REFRESH before the Dial() to my local endpoint?
16:13.49SamotOK
16:13.50filezBeeble: on 16 it would be it looks like
16:13.56electronic_eelI'm asking because the docs say " on an established media session", which sounds like after the dial
16:16.39sibiriais the session established before you Answer() the call? the channel isn't, but maybe these two are not equivalents here
16:16.44Samotelectronic_eel: On the incoming channel you can do Set(IN_CODEC=CHANNEL(rtp,media_type))
16:16.57SamotIf you're using pjsip.
16:17.07SamotThat will get the codec on the incoming channel that is being used.
16:17.23SamotYou can then check the from user/callerid for the source to see if it is mobile.
16:17.50SamotYou then set PJSIP_MEDIA_OFFER() to g711 if it is mobile _and_ using g722
16:18.12SamotThen you do PJSIP_SEND_SESSION_REFRESH to do a re-invite on the incoming channel to make it use g711
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16:18.48SamotThen you can use whatever you want with the other endpoint.
16:19.37SamotBut you'll be transcoding if you want g722 on the endpoint.
16:19.38SamotSo if you don't want that before you Dial(), you use PJSIP_MEDIA_OFFER() again to force g711
16:19.39SamotSo that the call is g711 on both channels.
16:26.33Samothttps://www.irccloud.com/pastebin/iXn14WFJ/
16:27.02Samotfile or someone might want to confirm but that should be what can cause you're mobile to not use 722.
16:27.27SamotSorry, change that GoSubIf to GoToIf
16:28.07SamotNM, fixed it.
16:30.41filetbh, it's easiest to just experiment and play with it
16:32.52drmessanoor only offer the codecs you want to support
16:33.43SamotI see the issue.
16:33.48SamotHe wants to support g722.
16:34.01SamotThe provider doesn't handle mobile calls over g722 very well.
16:34.10SamotSo only on those calls does he not want to use g722.
16:34.21drmessanoshrug
16:34.30drmessanoSwitch to OPUS then
16:34.40SamotThe g729 one though...
16:35.00SamotI agree the the original assessment. Don't use passthrough.
16:35.04drmessanoSeems like codec order would fix that one
16:35.47zBeeblecodec order doesn't fix it.
16:36.24zBeebleasterisk uses pass-thru all the time.  You have natted phones chatting to carriers... even if the codecs are the same, you're using passthru.
16:36.56SamotNot if you have direct media disabled.
16:38.52fileif circumstances allow then the media stream contents will be passed through the core, thus passthru
16:38.59filewithout being touching
16:39.01fileer touched
16:39.16SamotOK.
16:39.33SamotSo as long and both channels are g729, it's pass thru.
16:39.44Samotas*
16:39.55fileand you aren't doing recording, or spying, or other things
16:40.09SamotRight, DTMF, etc.
16:40.57zBeebledtmf is ok as long as it's RFC.
16:41.03SamotOK.
16:42.45SamotzBeeble: Why doesn't codec order fix things?
16:43.08SamotzBeeble: Because direct media/pass thru would state it would.
16:44.06SamotIf phone A only does g729 and only offers g729 then if phone B offers g711 and g729, they are going to use g729.
16:44.16drmessano^^^
16:44.31SamotIf phone B only offers g711 then nothing will happen.
16:44.36SamotAsterisk will get involved.
16:44.54drmessanoThat was my thought from the original description
16:46.45SamotIf 1 phone of 20 only does g729 and you want pass thru, all 20 phones must support and offer g729
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17:27.29electronic_eelSamot: thanks for the snippet, will give it a try
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18:48.07irrgitHello, is it possible with asterisk, we use digium as our provider, to use multiple long distance providers?
18:48.35[TK]D-FenderAsterisk can use whatever service you tell it to
18:48.47[TK]D-FenderIt's your dialplan... do whatever you want
18:49.15irrgitExample: Say we have ATT and Spectrum, we want to use ATT for the first 10,000/inbound minutes since we have a deal with them, but after that the rate goes to $0.03/m , whereas with Spectrum we have a base rate of $0.02
18:49.45irrgitCan we configure it so after the 10,000 minutes we use spectrum for the remainder of the month?
18:50.21sibiriathere's no specific configuration for that exact scenario, so you will have to accomplish it yourself manually or programmatically
18:50.35[TK]D-FenderAgain, it's YOUR dialplan
18:50.47[TK]D-FenderYou are in charge of making it do what you want
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18:59.52irrgit[TK]D-Fender, as long as you are saying that its possible, then thats all I care about.
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20:30.32MICROburstI grabbed the config for WebRTC from https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients -- but for some reason only 8088 is shown when doing 'http show status' - how do I get more information?
20:38.26MICROburstWhen I connect to port 8088 via browser, I get a Message 'Upgrade'.
20:42.19seanbrightwebrtc in asterisk uses sip over websocket
20:42.38seanbrightUpgrade means you need to connect with a websocket client, not just typing it in the browser
20:45.00MICROburstseanbright: Shouldn't firefox support webrtc? Why does the server not start on 8089?
20:45.24seanbrightfirefox does support it, but you can't just go to a URL in the browser and expect something to happen
20:45.36seanbrightyou need to have an application that speaks SIP over WebSockets like jssip or sip.js
20:46.11seanbrightthe 8089 thing is something to do with your configuration. you should check your asterisk logs for error messages related to HTTP/HTTPS
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20:53.15MICROburstIndeed: it complains about problems when loading the private key file. However the file exists and the permissions are 640 owner:group asterisk:asterisk
21:02.51seanbrighti believe in you, i know you can figure it out
21:03.55seanbrightMICROburst: what is the exact error message you are seeing?
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21:15.37*** join/#asterisk classsic (ba897291@186.137.114.145)
21:16.26classsicHi, somebody can recommend any freepbx service?
21:17.38fileyou probably want #freepbx and you'd need to define what that means
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21:18.48classsicany SaaS service provider for management my sip phones
21:21.53SamotYou need to be a little more detailed.
21:24.18classsicI want to login, add,remove and manage my sip phones, I don´t want deplay my own server
21:24.25classsic*deploy
21:24.57SamotManaging SIP phones does not mean having a PBX
21:25.10SamotSo you want something more like RingCentral.
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21:38.18igcewielingclasssic: the term you are looking for is "hosted pbx", or "hosted voip".  If you want to find providers where the marketing department runs things, replace "hosted" with "cloud"
21:40.01classsicok, I´m very new, I have some doorbells with sip support
21:40.56classsicand want to get a service for connect them
21:43.26MICROburstseanbright: ERROR[3172] tcptls.c: TLS/SSL error loading private key file. </etc/asterisk/keys/asterisk.key>
21:56.49MICROburstseanbright: any idea?
22:01.01seanbrightmust not be a valid PEM
22:01.13seanbrightor it doesn't exist or is not readable by asterisk
22:01.16seanbrightthose are the only options
22:10.04MICROburstfor the last test I used the ast_tls_cert script. No matter whether .pem or .crt ist doesn't work
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22:14.49seanbrightok
22:14.53seanbrightit's not asterisk, it's you
22:15.15seanbrightwithout access to your machine, i would just be guessing
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22:22.29seanbrightBUT, i would bet $1000 it was:
22:22.35seanbright1) invalid PEM
22:22.38seanbright2) file does not exist
22:22.43seanbright3) file is not readable by asterisk
22:23.11seanbrightonce you check all 3 of those things, and you are running on a supported asterisk version, you can open an issue at https://issues.asterisk.org
22:23.31seanbrightbut there are plenty of people (myself included) that load the private key just fine
22:29.56MICROburstseanbright: false. It is asterisk: the docs and the implementation. this posting here was helpful https://community.asterisk.org/t/websocket-tls-certificate-gives-error/81015/6 - but it seems you can't start the http server as tls only - very ugly!
22:38.26MICROburstso you just lost $1000 :)
22:48.31*** topic/#asterisk by kharwell -> AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.30.0 (2019/12/23) 16.7.0 (2019/12/23) Standard: 17.1.0 (2019/12/23); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
23:14.48seanbrightshucks
23:15.03seanbrighti knew you could do it on your own
23:15.30*** join/#asterisk jeffspeff (~overyande@216.163.24.235)
23:37.23*** join/#asterisk AsteriskRoss_ (~AsteriskR@r01.nt-r1.nor.gb.voicehost.co.uk)

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