| 00:00.35 | FuriousGeorge | but now it evaluates to false, where it should be true |
| 00:04.36 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 00:05.53 | FuriousGeorge | that was just a clierical error on my part... arg:1 instead of arg1:1 |
| 00:29.16 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 00:50.30 | *** join/#asterisk gentoorax (~gentoorax@gateway/tor-sasl/gentoorax) |
| 00:55.44 | mct | FuriousGeorge: Can you paste your config somewhere? |
| 01:01.02 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 01:03.28 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 01:22.53 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 01:46.35 | *** join/#asterisk LiuYan (~NiHola@unaffiliated/liuyan) |
| 01:51.30 | *** join/#asterisk deavmi (~quassel@165.255.253.162) |
| 01:53.47 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 02:01.11 | *** join/#asterisk dougquaid (dougquaid@gateway/vpn/privateinternetaccess/dougquaid) |
| 02:03.30 | dougquaid | I'm looking into setting up an Asterisk server for a small company. They currently use freeconferencecall.com with 5-6 people on the same call. Does Asterisk have a similar feature? |
| 02:05.13 | Samot | Yes, Asterisk can be used as a conference server. |
| 02:07.06 | dougquaid | Awesome, I figured it could. Do you know if SwitchVox Cloud or any other hosting providers off that feature? |
| 02:27.51 | *** join/#asterisk gams (~user@cm245-139.liwest.at) |
| 02:36.20 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 02:47.38 | *** join/#asterisk sahmed (~sahmed@cpe-70-114-236-63.austin.res.rr.com) |
| 02:48.43 | drmessano | dougquaid: Any Asterisk/FreePBX/SwitchVOX instance offers basic conferencing |
| 02:49.14 | drmessano | That's pretty standard |
| 02:51.00 | dougquaid | That's good to know. I just installed FreePBX in a VM to test it out |
| 02:51.08 | dougquaid | What's a good Android SIP client? |
| 02:54.15 | drmessano | Bria is the only one worth a crap |
| 02:56.23 | dougquaid | Bria looks like it's $.99/month. Is that just for the SIP client or do they provide the server too? |
| 02:57.07 | drmessano | You won't get a full functioning PBX for $1 a month |
| 02:57.13 | drmessano | That's for the client |
| 02:57.32 | drmessano | You're paying for a SIP client that doesn't suck |
| 02:57.39 | dougquaid | Makes sense |
| 02:58.19 | dougquaid | It's just kind of strange for a client side app to charge monthly/yearly rather than a one time charge |
| 03:04.12 | drmessano | No, it's not |
| 03:04.26 | drmessano | Subscription model is pretty popular in quite a few industries |
| 03:05.08 | drmessano | Ever heard of Office365? Adobe? |
| 03:05.28 | dougquaid | Yeah I guess you have a point |
| 03:05.28 | drmessano | I pay for many subscriptions on my IOS apps |
| 03:05.43 | drmessano | So not odd in the slightest |
| 03:06.20 | dougquaid | Is there a guide on how to setup end-to-end encryption in asterisk? |
| 03:08.23 | drmessano | Google for Asterisk SIP TLS SRTP |
| 03:08.26 | drmessano | Lots of hits |
| 03:08.40 | drmessano | One of them should take you to the wiki |
| 03:10.02 | dougquaid | Yup, I found it. Thanks again |
| 03:55.01 | *** join/#asterisk Chotizei (chotaire@unaffiliated/chotaire) |
| 04:07.01 | *** join/#asterisk Chotizei (chotaire@unaffiliated/chotaire) |
| 05:27.17 | *** join/#asterisk gentoorax (~gentoorax@gateway/tor-sasl/gentoorax) |
| 05:35.42 | *** join/#asterisk joepublic (~joepublic@fsf/member/joepublic) |
| 05:54.40 | *** join/#asterisk jkroon (~jkroon@165.16.203.126) |
| 05:55.32 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 06:39.21 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 07:09.33 | *** join/#asterisk guerby (~guerby@april/board/guerby) |
| 07:10.20 | *** join/#asterisk pchero_work (~pchero@87.213.247.82) |
| 07:11.17 | *** join/#asterisk tsal (~tsal@i59F4A777.versanet.de) |
| 07:54.00 | *** join/#asterisk BakaKuna (~user@2a04:9a00:1002:701:81ea:b8cd:e034:5c21) |
| 07:56.15 | *** join/#asterisk lankanmon (~LKNnet@CPEb4fbe4e331bd-CM64777d632380.cpe.net.cable.rogers.com) |
| 08:20.07 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 08:29.13 | *** join/#asterisk sahmed (~sahmed@cpe-70-114-236-63.austin.res.rr.com) |
| 08:36.44 | *** join/#asterisk lankanmon (~LKNnet@CPEb4fbe4e331bd-CM64777d632380.cpe.net.cable.rogers.com) |
| 09:53.38 | jkroon | i've got an odd situation where I'm receiving an INVITE via PJSIP (WebRTC). The moment this call completes, the recipient channel is placed on hold, and there is no audio between Chrome and asterisk (13.29.1). From the logs I can see that the moment the call is answered the answering channel is placed on hold. I just can't figure out *why*. |
| 09:54.16 | jkroon | the SDP from Chrome does state a=sendrecv, and the last dialplan action before the call is placed on hold is the Dial() that results in an answer. |
| 09:54.42 | jkroon | I'm personally unable to reproduce using chromium on Linux. |
| 10:00.53 | *** join/#asterisk DodgeThis (~DodgeThis@246.102.90.149.rev.vodafone.pt) |
| 10:32.24 | *** join/#asterisk ganbold (~ganbold@66.85.186.234) |
| 11:10.25 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 12:40.39 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 13:33.06 | *** join/#asterisk dougquaid (dougquaid@gateway/vpn/privateinternetaccess/dougquaid) |
| 13:35.28 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:50.46 | *** join/#asterisk dakudos (~dakudos@c-73-229-175-50.hsd1.co.comcast.net) |
| 13:58.00 | *** join/#asterisk FH_thecat (~FH_thecat@75.11.25.212.ftth.as8758.net) |
| 13:58.41 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 13:58.41 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 14:24.29 | *** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net) |
| 14:28.38 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:35.50 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 14:39.08 | *** join/#asterisk FH_thecat (~FH_thecat@75.11.25.212.ftth.as8758.net) |
| 14:55.27 | dougquaid | Anyone know of a good US based hosted asterisk/freepbx provider? |
| 14:55.36 | *** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be) |
| 14:57.29 | *** join/#asterisk rShadowhand (~Shadowhan@secretalgorithm.com) |
| 14:58.48 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 15:08.14 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 15:11.07 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-utxwvkjwhwjkmaha) |
| 15:11.07 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:17.42 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-kfxohaqafzlhcfdy) |
| 15:17.42 | *** mode/#asterisk [+o bford] by ChanServ |
| 15:18.34 | Samot | For that specifically? I think the only option is FreePBXHosting.com |
| 15:21.10 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-numxmklnecktmnpc) |
| 15:23.28 | dougquaid | The more I read/learn about asterisk the more questions I have. Are there other web UIs for asterisk besides FreePBX? |
| 15:23.48 | dougquaid | Also, is there a benifit to using IAX instead of SIP? Or should I use them both? |
| 15:24.03 | igcewieling | dougquaid: All PBX GUIs suck. FreePBX sucks far less that all others. |
| 15:24.15 | [TK]D-Fender | IAX is only meaningful if you're really tight on bandwidth and need it |
| 15:24.21 | igcewieling | dougquaid: using IAX2 means you'll be on your own. Almost nobody uses it. |
| 15:24.31 | dougquaid | SIP it is then |
| 15:24.31 | [TK]D-Fender | And yes there are other GUI's |
| 15:24.38 | [TK]D-Fender | Most others are commercial |
| 15:24.45 | Reinhilde | IAX2 is considered deprecated, but I still think it has a place. All sane Asterisk setups use SIP. Some ALSO use IAX2. |
| 15:24.46 | [TK]D-Fender | Thirdlane, ScopServ, etc |
| 15:25.17 | igcewieling | PBXWare is another -- I used it for our hosted service, but that isn't a "cloud" thing. |
| 15:25.21 | dougquaid | I'm not opposed to a commercial GUI if it allows a non-techie to easily add extensions and whatnot |
| 15:25.22 | [TK]D-Fender | Whatever that PIAF or fork or whatever is called now.... |
| 15:27.57 | dougquaid | I'm looking at SwitchVox now. Their GUI looks pretty nice |
| 15:28.26 | Reinhilde | I don't use a web UI at all. |
| 15:28.51 | dougquaid | Do you just manually edit the config files? |
| 15:28.52 | Reinhilde | I also use the deprecated Asterisk SIP channel instead of PJSIP because I could never work out how PJ works. |
| 15:28.54 | Reinhilde | Yes. |
| 15:29.24 | igcewieling | Reinhilde: pjsip is actually easier to setup since it doesn't have all that user/peer/friend sillyness. |
| 15:29.47 | Reinhilde | igcewieling: it's actually not |
| 15:30.02 | Reinhilde | I've never really had any issues knowing what user, peer and friend mean |
| 15:30.50 | file | dougquaid: Switchvox is an appliance (all in one solution), not a GUI for Asterisk just so you are aware |
| 15:30.53 | Reinhilde | by all counts chan_sip's configuration file is more sensible than the setup PJ uses. |
| 15:31.10 | Reinhilde | file: is their GUI available separately? |
| 15:31.15 | file | no. |
| 15:31.22 | dougquaid | file: I wasn't aware. I thought they were a hosted asterisk provider with their own GUI |
| 15:31.49 | file | dougquaid: Switchvox Cloud is a solution, but it's also available in physical appliances and virtualized |
| 15:32.13 | file | the point I'm trying to make is that Asterisk is an implementation detail that isn't exposed/that you don't care about |
| 15:34.37 | dougquaid | So let me ask this. I'm consulting for a company who needs the ability to have up to simultaneous 6 conference calls with about 5 people on each call. I'd also like a GUI so the average user can create extensions. What would you guys do in this case? |
| 15:34.50 | [TK]D-Fender | Or don't get to considering that solution is designed to keep you out from it. |
| 15:44.56 | drmessano | 10:30:54 <Reinhilde> by all counts chan_sip's configuration file is more sensible than the setup PJ uses. <--- False, because templates make it a lot less work |
| 15:45.25 | *** join/#asterisk dacod (~dacod@201.47.74.146) |
| 15:47.20 | drmessano | Also, the phrase is "By all accounts" and I don't see all accounts that chan_sip's config is easier. Most would say it is not |
| 15:47.46 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 15:52.42 | [TK]D-Fender | I see chan_sip as "easier", but that's because it is so much less flexible. Multiple transports, multiple contact, etc... PJSIP, while more broken down, is totally worth it. |
| 16:02.03 | Samot | Chan_SIP, has always been a lacking SIP driver. |
| 16:11.35 | dougquaid | I've got FreePBX running in a VM and 2 Androids with SIP clients. The call works and I get audio for about a minute and then the call drops. I tried different SIP clients but it happens on all of them so far. How can I troubleshoot this problem? |
| 16:11.55 | *** join/#asterisk jkroon (~jkroon@165.16.204.108) |
| 16:12.04 | igcewieling | disable direct media |
| 16:12.26 | dougquaid | In FreePBX or the SIP clients? |
| 16:12.41 | [TK]D-Fender | FreePBX |
| 16:12.48 | [TK]D-Fender | PBX = king |
| 16:12.51 | igcewieling | FreePBX |
| 16:16.24 | dougquaid | Is DEVICE_SIP_CANREINVITE the correct option to disable? |
| 16:18.25 | *** join/#asterisk cybrNaut (~cybrNaut@unaffiliated/cybrnaut) |
| 16:18.38 | Samot | What? |
| 16:19.36 | igcewieling | I think he is looking in the general settings instead of the settings for the specific extensions. |
| 16:20.06 | dougquaid | Yeah I am looking at general settings and I saw that one about direct media |
| 16:20.18 | igcewieling | Look in the EXTENSION setup. |
| 16:21.50 | dougquaid | Found it. So there's no server wide direct media setting and I need to disable it for each extension? |
| 16:23.59 | igcewieling | dougquaid: you can disable it globally if you want, but that won't reset the settings on the existing extensions. |
| 16:28.15 | Samot | Wait. |
| 16:28.24 | Samot | If you are using Chan_SIP you can disable it. |
| 16:28.40 | Samot | Globally that is. PJSIP doesn't have global settings like that. |
| 16:28.50 | Samot | There are no general/global endpoint settings. |
| 16:28.58 | dougquaid | Ah ok. FreePBX defaulted to PJSIP |
| 16:29.28 | Samot | Then you will need to look at each extension. |
| 16:36.15 | dougquaid | Disabling direct media seemed to do the trick, thanks guys! Now to figure out TLS/SRTP |
| 16:46.24 | cybrNaut | these instructions to install asterisk on debian look bad => https://blog.here-host.com/install-asterisk-debian-9/ |
| 16:46.53 | cybrNaut | why would it make sense to tell ppl generally to unpack and compile a tarball on debian? |
| 16:47.25 | igcewieling | How else would you install it? |
| 16:47.30 | cybrNaut | even if a debian user wants to build from source, I don't think they need to deal with a tarball |
| 16:47.45 | cybrNaut | igcewieling: "aptitude install asterisk" should work, in theory |
| 16:48.12 | igcewieling | cybrNaut: sounds like you should be complaining to Debian about lack of Asterisk in whatever they call their repos. |
| 16:48.44 | cybrNaut | igcewieling: not at all. It's precisely because I see "asterisk" in the repos that those instructions look lousy |
| 16:49.00 | joepublic | I am running an asterisk server that began its life with "apt install asterisk" and it works perfectly |
| 16:49.18 | cybrNaut | joepublic: thanks. in that case i'll take that approach |
| 16:51.03 | igcewieling | Remember, you won't be able to install commercial modules or get support if you don't use the distro. |
| 16:57.28 | dougquaid | I still seem to be having trouble with dropped conference calls, but I think it's a NAT problem with my phone's 4G network. One phone on wifi works and does not drop, but the phone on the 4G network drops after a few seconds and I saw a packet that said "udp port 26275 unreachable" on my device's public 4G IP address. Is there a way to fix that? |
| 16:58.37 | igcewieling | *shrug* Cellular SIP never worked for me. |
| 16:58.47 | igcewieling | Are you on....Verizon? |
| 16:58.57 | dougquaid | AT&T |
| 16:59.02 | Reinhilde | And that's why I think IAX still has a place. |
| 16:59.04 | Reinhilde | Because cell. |
| 16:59.39 | Samot | There are like three platforms at best that support IAX. |
| 16:59.44 | Samot | It's not a standard solution. |
| 17:01.37 | igcewieling | Digium/Sangoma really should deprecate IAX and IAX2 |
| 17:01.49 | Samot | Technically it is. |
| 17:01.57 | Samot | No one at Digium touches it. |
| 17:02.06 | dougquaid | Would a STUN server help resolve NAT issues on 4G networks? |
| 17:02.09 | Samot | It's community driven like Chan_SIP now is. |
| 17:03.11 | Samot | IAX has seen very few updates in the last couple years and most that were done to to move around things in Asterisk so they wouldn't get deleted/removed because nothing else uses them anymore but IAX. |
| 17:03.15 | igcewieling | dougquaid: I doubt it. Are you not on NAT when using WiFi? |
| 17:03.45 | cybrNaut | i've seen a voip provider who caters to pros (e.g. pbx users) reduce their SIP support in favor of IAX. I got the impression IAX had more capability. |
| 17:03.52 | dougquaid | My router does nat and the wifi phone has a 192.168.0.0/24 address |
| 17:04.11 | igcewieling | dougquaid: Obviously your problem is not JUST a NAT issue then. |
| 17:04.48 | dougquaid | You're probably right. I think I'll save a pcap and try to see what's going on in wireshark |
| 17:08.51 | Reinhilde | Samot: You should just /nick Karen and be done with it |
| 17:09.15 | Samot | I have no idea what that comment means. |
| 17:15.58 | drmessano | Samot: Being down on the unsupported IAX2 makes you a Karen |
| 17:16.05 | drmessano | Oh and chan_sip |
| 17:16.26 | Samot | I went to Urban Dictionary. |
| 17:16.45 | Samot | I didn't see where stating facts made you a Karen. |
| 17:17.24 | Samot | Would I be fine with IAX being removed? Sure. Am I being a Greta about it? No. |
| 17:17.49 | igcewieling | Urban Dictionary: letting old people understand young people. |
| 17:17.58 | Samot | Well |
| 17:18.05 | igcewieling | I have to use it all the time. 8-| |
| 17:18.14 | Samot | That and learning how young people have misused older words. |
| 17:18.24 | Samot | The best is "poppers" |
| 17:18.30 | *** join/#asterisk miralin (~Thunderbi@94.233.240.236) |
| 17:18.44 | Samot | Because whoa man, that can give someone a real wrong idea about you. |
| 17:19.41 | Reinhilde | Removing IAX would turn me into a full hermit |
| 17:20.01 | salviadud | IAX is getting removed? |
| 17:20.08 | Reinhilde | salviadud: not to my knowledge |
| 17:20.10 | Samot | No. |
| 17:20.12 | Samot | It's not. |
| 17:20.18 | *** join/#asterisk rodolfojcj (~rodolfojc@190-36-187-205.dyn.dsl.cantv.net) |
| 17:20.28 | salviadud | You're doing some crazy hipotheticals |
| 17:20.33 | drmessano | IAX2 was fine like 10 years ago |
| 17:20.43 | drmessano | Sorry, 15 |
| 17:21.04 | igcewieling | Even 20 years ago, SIP was wining. |
| 17:21.05 | Samot | IAX had an RFC draft. |
| 17:21.12 | Samot | It was never adopted. |
| 17:21.20 | Reinhilde | igcewieling: SIP is the thing everyone blocks |
| 17:21.27 | drmessano | lol |
| 17:21.28 | Samot | Because SIP is the boss. |
| 17:21.33 | Reinhilde | people care less about IAX |
| 17:21.40 | Reinhilde | so IAX gets through |
| 17:21.41 | Samot | If IAX had won 20 years ago, it would be what is blocked. |
| 17:21.49 | Samot | If MGCP won, that would be blocked. |
| 17:21.57 | Samot | If H.323 won, same thing. |
| 17:22.04 | Samot | SIP won the VoIP wars. |
| 17:22.06 | igcewieling | Might as well use H323 is you want to use a zombie protocol. |
| 17:22.13 | drmessano | PJSIP is so much more robust than IAX2 |
| 17:22.16 | Reinhilde | Removing any of them seems irresponsible, especially if there are not fundamental flaws. |
| 17:22.22 | rmudgett | If SS7 won nobody would know |
| 17:22.26 | dougquaid | Is there a setting in FreePBX or Asterisk that tells clients which ports to use for RTP? |
| 17:22.28 | Reinhilde | rmudgett: it has |
| 17:22.35 | igcewieling | dougquaid: no. |
| 17:22.40 | salviadud | rtp.conf dougquaid |
| 17:22.43 | Reinhilde | SIP is clearly the most flexible of the protocols |
| 17:22.44 | igcewieling | because that is up to the client. |
| 17:22.50 | Samot | There is no need for two SIP drivers. |
| 17:22.50 | salviadud | Oh, my bad |
| 17:23.15 | Reinhilde | someday someone will write an alternative chan_iax |
| 17:23.24 | igcewieling | SIP is a crappy protocol built in the style of H323. It sucks. It also won the protocol wars, so get over it. |
| 17:24.37 | igcewieling | Personally, I wish MGCP won the protocol wars. |
| 17:24.48 | Reinhilde | neurodegenerates |
| 17:25.09 | Reinhilde | The protocol wars are something that are fought every time you install a PBX. |
| 17:25.23 | Reinhilde | Because most of the useful clients are SIP, most of the time, SIP wins. |
| 17:25.32 | dougquaid | So what range of ports should I open on my FreePBX server? Right now I have ports 10000 - 20000 forwarded, but I think my dropped call problem was that a client tried using port 50000 |
| 17:25.40 | Reinhilde | dougquaid: 0-65535 |
| 17:25.42 | Samot | Asterisk is not a PBX. |
| 17:25.46 | Samot | It's a toolkit. |
| 17:25.55 | Reinhilde | Samot: you're one of /those/ |
| 17:26.02 | igcewieling | dougquaid: don't need to forward ports. The outgoing audio will open the ports and incoming audio can use that NAT translation. |
| 17:26.10 | Samot | I guess. |
| 17:26.22 | Samot | I'm one of those that agrees with the developers standpoint. |
| 17:26.24 | Reinhilde | 09:26:09 --- Samot is now dimmed on fn |
| 17:26.25 | igcewieling | dougquaid: your best bet is leave it as the default 10000 to 20000. |
| 17:26.32 | Samot | Asterisk is a toolkit. |
| 17:26.58 | salviadud | and that toolkit sometimes does PBX stuff, that's what you mean? |
| 17:27.13 | salviadud | Asterisk: it does pbx stuff |
| 17:27.18 | Samot | Correct, it is a toolkit that can be made into a PBX. |
| 17:27.26 | igcewieling | dougquaid: if Asterisk is behind NAT AND you have off-site phones registering to Asterisk, then you'll need to forward port 5060/UDP. |
| 17:27.32 | Samot | But it can also only be a voicemail server. |
| 17:27.36 | Samot | A conference bridge server |
| 17:27.45 | Samot | It could just pass calls through it and transcode. |
| 17:27.59 | Samot | It will only do an IVR if it is told to do an IVR |
| 17:28.06 | Samot | It doesn't have Ring Groups, that a method |
| 17:28.06 | dougquaid | igcewieling: Yup, I have 5060/UDP and 10000-20000/UDP forwarded |
| 17:28.17 | Samot | So you would write you own Ring Group methods. |
| 17:28.27 | igcewieling | dougquaid: the 10000 - 20000 is not needed, but I doubt it will hurt either. |
| 17:29.08 | salviadud | It can be as complex, or as simple as you want it to be. |
| 17:29.18 | Samot | Exactly. |
| 17:29.54 | Samot | FreePBX, Switchvox, Thirdlane, ScopeServ. Those are vendors that use Asterisk to make a PBX system. |
| 17:32.10 | salviadud | I wonder if it will ever come out for ARM. |
| 17:32.21 | salviadud | It would be pretty cool to squeeze it in a raspberry pi |
| 17:32.39 | Samot | People use FreePBX on RasPi all the time. |
| 17:32.58 | Samot | There's a distro just for RasPi. |
| 17:33.16 | salviadud | Why not mainline asterisk? |
| 17:33.25 | Samot | I'm sure you can install it. |
| 17:33.35 | Samot | But RasPBX is meant to be a PBX |
| 17:33.41 | salviadud | Well, freepbx should be easier. |
| 17:33.41 | Samot | Those generally have GUIs. |
| 17:36.05 | salviadud | http://www.raspberry-asterisk.org/downloads/ <--- found it |
| 17:36.26 | salviadud | It does say it has both asterisk and freepbx |
| 17:36.41 | Samot | Well Asterisk is kind of require to have FreePBX. |
| 18:36.53 | *** join/#asterisk hfb (~hfb@47.139.16.213) |
| 18:57.44 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-ysbzkbuxynkimbmj) |
| 18:57.44 | *** mode/#asterisk [+o tm1000] by ChanServ |
| 18:58.05 | *** part/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-ysbzkbuxynkimbmj) |
| 19:47.54 | *** join/#asterisk hfb (~hfb@47.139.16.213) |
| 19:52.54 | *** join/#asterisk Ai9zO5AP (BQcdf9eiZ8@gateway/vpn/protonvpn/ai9zo5ap) |
| 20:44.59 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 21:54.26 | *** join/#asterisk defsdoor (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net) |
| 22:13.47 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 22:28.41 | *** join/#asterisk sh_smith (~sh_smith@cpe-172-88-21-24.socal.res.rr.com) |
| 23:09.21 | *** join/#asterisk aa1001 (~aa1001@office.ptera.net) |
| 23:59.10 | *** join/#asterisk paulgrmn (~paulgrmn@c-68-34-113-42.hsd1.mi.comcast.net) |