| 00:00.19 | robmal | has once again achieved full zen mode thanks to IRC. |
| 00:01.46 | robmal | Voyage: http://89.174.180.22:7080/ |
| 00:06.28 | boritos | dial_exec_full: Unable to create channel of type 'dongle' (cause 44 - Requested channel not available.. any tips how to fix it ? |
| 00:07.05 | WIMPy | Enable debugging and see what the dongle sent. |
| 00:11.31 | talntid | robmal, what do i use for creds for that? :) |
| 00:11.37 | talntid | do i gotta make my own? |
| 00:12.41 | robmal | talntid: It wasn't for you, you'll get the creds once it's ready, right now i just need Voyage to click ;-) |
| 00:13.10 | talntid | :-) |
| 00:19.57 | robmal | Voyage: You there? |
| 00:32.23 | *** join/#asterisk cesurasean (~cesurasea@c-68-35-152-116.hsd1.al.comcast.net) |
| 00:32.26 | cesurasean | yo yo yo |
| 00:33.02 | cesurasean | damn chromebox won't stay on. built like shit. |
| 00:33.12 | cesurasean | few months after warranty expire, too. sucks. |
| 00:33.38 | cesurasean | i am however back though, to learn more from yall. :) |
| 00:33.56 | cesurasean | lost convo and don't know yall's nicknames on here. |
| 00:37.05 | robmal | cesurasean: Rule no.1 for now is: be nice to Voyage |
| 00:37.36 | cesurasean | who is Voyage? |
| 00:37.40 | robmal | Nobody knows. |
| 00:37.53 | robmal | Rule 2 is: You might be nice to everyone else. |
| 00:38.30 | talntid | Rule 3 is: you must sacrifice a FreePBX installation to the Asterisk gods... |
| 00:38.56 | cesurasean | talntid, im an old warez leech. :D |
| 00:39.06 | talntid | hah |
| 00:39.14 | talntid | irc warez. i wonder if those still exist much |
| 00:39.17 | talntid | back in the days over DCC |
| 00:39.22 | cesurasean | yep |
| 00:39.23 | talntid | have not thought about that in a long time |
| 00:39.38 | talntid | used to write eggdrop bot scripts to do that stuff |
| 00:39.39 | cesurasean | dcc me 323 zips |
| 00:40.36 | cesurasean | everyone grew up, apparently. |
| 00:41.04 | talntid | yupyup |
| 00:41.18 | cesurasean | how does a person get DID blocks? |
| 00:41.26 | cesurasean | and how exactly does all DID talk to each other? |
| 00:41.57 | cesurasean | some questions i have been wondering lately...... |
| 00:42.18 | cesurasean | surely there is some centralized system somewhere the Government maintains of DID type stuff? |
| 00:43.06 | boritos | talntid: #123warez #oldwarez at undernet :) |
| 00:43.12 | cesurasean | still a lot about telephony i don't yet understand. |
| 00:43.29 | cesurasean | where do these phone numbers come from? surely a database of sorts? |
| 00:43.32 | cesurasean | like internic? |
| 00:54.16 | robmal | Polen has a government institution to keep a hand on this. Telephony providers can get 1k numbers from them and they keep a db who has which number. |
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| 01:03.11 | cesurasean | Polen? |
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| 01:03.26 | WIMPy | Poland |
| 01:03.32 | cesurasean | sorry, i was watching them mine gold on television. |
| 01:03.48 | cesurasean | does USA have a database? |
| 01:04.02 | robmal | Even better than polen. |
| 01:04.07 | WIMPy | And yes. They will come from some government agency. How depend on which country. |
| 01:04.26 | WIMPy | That should be the FCC. |
| 01:04.34 | cesurasean | oh i see. |
| 01:08.35 | cesurasean | WIMPy, do you know how to get my asterisk to not go straight to voicemail? |
| 01:08.43 | cesurasean | im using voip.ms |
| 01:08.55 | WIMPy | Their VM or your VM? |
| 01:09.15 | cesurasean | on my vm |
| 01:09.23 | cesurasean | i tried to email them but they want to see my configs. lol |
| 01:09.31 | cesurasean | is there an easy way to show someone asterisk configs? |
| 01:09.37 | robmal | ~pb |
| 01:09.40 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:09.46 | WIMPy | Well, then it's your dilplan. It will do whatever you write there. |
| 01:10.11 | cesurasean | but its hitting the voip.ms voicemail |
| 01:10.17 | cesurasean | how do i know its routed to my box correctly? |
| 01:10.34 | WIMPy | That was what I asked for. |
| 01:10.44 | cesurasean | oh yeah the debug |
| 01:10.50 | WIMPy | Well, then they probably can't reach you. |
| 01:10.51 | cesurasean | so it was getting calls? |
| 01:10.59 | cesurasean | the calls are making it to my box? |
| 01:11.09 | cesurasean | shows how much i know. :D |
| 01:11.14 | WIMPy | Probably not. |
| 01:11.26 | WIMPy | Enable sip debug and have a look. |
| 01:11.30 | cesurasean | how can i test to see if a call hits my box? |
| 01:11.39 | cesurasean | oh so the debug is the way to find out? |
| 01:11.50 | WIMPy | yes |
| 01:12.00 | cesurasean | i see. |
| 01:12.38 | robmal | I'm pretty sure you go to VM when your extension is UNREACHABLE in sip show peer. |
| 01:13.03 | robmal | So try figuring out why. |
| 01:16.11 | cesurasean | sip show peers says my extension is online |
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| 01:17.59 | cesurasean | oh yeah you guys said freepbx is newer than the gui |
| 01:18.02 | cesurasean | let me try to use that first |
| 01:18.22 | WIMPy | Good luck. |
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| 01:27.23 | Janos | hey there, got a TE4XXP card I'm trying to setup, the problem is, when I only use the first span everything works. That is only first span configured in system.conf, but if I try to use a second span, everything works according to dahdi, but asterisk `pri show spans` says down,active. Any idea what could be going on ? |
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| 03:35.05 | igcewieling | I'm just about ready to disable ssh on a router to test something and I say to myself "I should tell the router to reboot in 30 mins on the off chance I lock myself out of the router". So I disabled ssh and sure enough I locked myself out of the router. 8-| |
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| 05:33.45 | blooregard | Good evening all |
| 05:34.01 | blooregard | has anyone here used an auto dialer before? |
| 05:37.20 | drmessano | Only if they want to be stabbed and ridiculed |
| 05:37.29 | drmessano | Robodialers BAD |
| 05:39.35 | blooregard | on the receiving end yeah, they're usually bad. But they can serve a purpose when they're not being used to sell aluminum siding at 5am on a sunday morning. |
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| 12:06.53 | boritos | do we have a export in asterisk dongle here ? |
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| 12:14.27 | boritos | working on asterisk raspberry and 3g dongle. but i dont get any mode or submode |
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| 12:32.12 | lvlinux | igcewieling: but DID you tell it to reboot? |
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| 16:08.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.6.0 (2015/10/09), 11.20.0 (2015/10/09), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 21:08.28 | Encrypt | Hello o/ |
| 21:08.39 | Encrypt | boritos, Hey o/ |
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| 21:40.34 | wyoung | hihi |
| 21:40.52 | wyoung | Let's Encrypt! |
| 21:41.16 | wyoung | (I hope that comes out soon) |
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| 23:23.23 | lvlinux | wyoung: me too! |
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| 23:39.51 | rubio | hi there.... i'm loosing my head here fighting WebRTC sound issue... is this the right channel to ask about this? (running asterisk 11.13 from debian 8 with sipml5 over chrome) thanks. |
| 23:44.07 | lvlinux | rubio: Depends on if you think the issue is related to your Asterisk config. If so, then yes. |
| 23:48.39 | rubio | lvlinux: i think so, looks to me that asterisk is always sending the rtp via udp and not using ICE (as i never see a "(via ICE)" legend on the rtp debug, and i have seen this legend on most pages on internet)... |
| 23:49.01 | rubio | i think i missing something on configs... but i can't find it. |
| 23:50.03 | lvlinux | uh, ICE isn't a protocol. ICE figures out your addressing and port settings by trying to negotiate with the remote endpoint. |
| 23:50.22 | lvlinux | your rtp should be over UDP, regardless of whether ICE is being used. |
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| 23:51.12 | lvlinux | Let me rephrase that: ICE isn't a media transfer protocol, it's a NAT punch-through protocol. |
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| 23:52.00 | lvlinux | ICE is similar to STUN, but doesn't require a STUN server. |
| 23:52.57 | rubio | i know, but when i inspect trafic, udp it going directly to client (remember client is web and under a nat)... in most pages i find on google the rtp debug say: "to <ip> (via ICE)"... |
| 23:53.19 | rubio | my point is.. the client registers ok... but there is no sound at all... :( |
| 23:53.54 | rubio | i'm using wss with the asterisk http module... using an ssl cert for the site... |
| 23:54.01 | lvlinux | ok so I guess ICE just isn't working for some reason, so thus you get no RTP delivery. |
| 23:54.24 | lvlinux | Not really sure where to go from here since I haven't used the webrtc capability. |
| 23:54.49 | rubio | ok... |
| 23:55.32 | rubio | will wait for someone who has work with webrtc :D |
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| 23:55.54 | rubio | thanks lvlinux :D |
| 23:55.58 | lvlinux | np |
| 23:56.25 | lvlinux | sincerely hopes someone can help rubio out, because no audio problems are SO irritating... |
| 23:58.16 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |