| 00:00.22 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:00.53 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 00:03.13 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
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| 00:15.08 | perece | hello everyone |
| 00:15.20 | WIMPy | hello perece |
| 00:15.50 | perece | is there any way to use dyna hints when SIP username != extension number? |
| 00:19.14 | WIMPy | Only if they share the dynamic part. |
| 00:20.54 | perece | that is, only ext number is different, right? |
| 00:27.42 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 00:32.25 | WIMPy | Or part of it. |
| 00:34.45 | perece | ahh, nevermind. solved it with ${PREFIX${EXTEN}}. was just wild try, never actually expected _nested_ variavle expansion (w/o EVAL) but it works.. |
| 00:35.38 | WIMPy | Some nice litte magic. Yes. |
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| 00:57.01 | bibz2 | anyone a idea on my anonymous caller topic? I want to display the number even if someone is calling in anonymously |
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| 01:05.11 | *** join/#asterisk bibz2 (bibz2@193-81-137-62.adsl.highway.telekom.at) |
| 01:05.17 | bibz2 | re |
| 01:21.47 | WIMPy | bibz2: From where? |
| 01:44.20 | WIMPy | <pzn> [TK]D-Fender, great! AMI originate solves it. tks |
| 01:44.27 | WIMPy | [15:01] calum_ has joined #asterisk |
| 01:44.30 | WIMPy | (~calum_@host81-154-109-145.range81-154.btcentralplus.com) |
| 01:44.37 | WIMPy | [15:02] calum_ has left IRC (Remote host closed the connection) |
| 01:44.39 | WIMPy | <mjt> hello. I'm trying out asterisk, started with a zoiper sip client on my |
| 01:44.42 | WIMPy | mobile phone, using a wifi network (the acces point is the same machine |
| 01:44.47 | WIMPy | Argh |
| 01:53.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
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| 03:42.22 | *** join/#asterisk al_nz1 (~bigal.nz@118-93-220-69.dsl.dyn.ihug.co.nz) |
| 03:42.33 | al_nz1 | Can someone hack me if I give you my ip please? |
| 03:54.34 | [TK]D-Fender | WIMPy: Accidental copy/paste? |
| 04:00.46 | *** join/#asterisk sizeof9 (~sizeof9@ip98-165-137-211.ph.ph.cox.net) |
| 04:11.14 | al_nz1 | Hey [TK]D-Fender |
| 04:13.02 | *** join/#asterisk babak_ (02b2b806@gateway/web/cgi-irc/kiwiirc.com/ip.2.178.184.6) |
| 04:27.13 | ChannelZ | is it 118.93.220.69? |
| 04:29.17 | al_nz1 | ChannelZ: no |
| 04:29.39 | al_nz1 | but I can send you it |
| 04:29.54 | al_nz1 | using notice |
| 04:32.29 | ChannelZ | What is it you would like me to do? |
| 04:32.43 | al_nz1 | attack it with sip |
| 04:32.46 | al_nz1 | try to register |
| 04:36.36 | ChannelZ | ok |
| 04:38.18 | al_nz1 | let me know when you have had 1 attempt, I may make some changes |
| 04:41.38 | ChannelZ | No response. |
| 04:42.49 | ChannelZ | no ping either |
| 04:43.20 | al_nz1 | perfect - let me make my iptables a little more permissive |
| 04:44.23 | al_nz1 | try now |
| 04:44.34 | al_nz1 | it might ban after 3 attempts |
| 04:44.47 | al_nz1 | but I dont think it working |
| 04:45.54 | al_nz1 | I am tail -f /var/log/messages now so I hope i see your attempts |
| 04:46.08 | ChannelZ | you should be seeing them. Though maybe not in messages. |
| 04:46.28 | ChannelZ | I'm trying to reg as poopyface |
| 04:46.29 | al_nz1 | whats your ip? |
| 04:47.21 | ChannelZ | 173..35.173 |
| 04:47.26 | al_nz1 | yip I see yoru attempts |
| 04:47.29 | al_nz1 | grrr |
| 04:47.37 | al_nz1 | fail2ban not picking them up |
| 04:47.59 | al_nz1 | you 173.160 something? |
| 04:48.13 | ChannelZ | yes |
| 04:48.19 | al_nz1 | k |
| 04:48.26 | al_nz1 | let me do some tweaking :-) |
| 04:49.06 | ChannelZ | what version of asterisk? Looks like FreePBX.. is it 11.2.1? |
| 04:49.13 | al_nz1 | piaf |
| 04:49.15 | al_nz1 | yes |
| 04:49.26 | ChannelZ | So are you using the security log? |
| 04:49.54 | al_nz1 | <PROTECTED> |
| 04:50.01 | al_nz1 | with security turned on |
| 04:50.07 | al_nz1 | I can see your attempts on the log |
| 04:50.16 | al_nz1 | but I need to probably check the regex? |
| 04:50.54 | ChannelZ | It's easier to turn on security as a separate log and use that |
| 04:51.41 | ChannelZ | And here's the fail2ban filter I use - http://pastebin.com/7Zc04eRX |
| 04:52.04 | ChannelZ | How you hook it up is up to you (the action you've made) |
| 04:52.32 | *** join/#asterisk zerous (~naveens@88.201.43.222) |
| 04:53.25 | zerous | i compiled and installed asterisk from source but i don't have any files in /etc/asterisk/ |
| 04:53.47 | zerous | could anyone tell me what i should do.. or is this the way it should be ? |
| 04:54.17 | ChannelZ | you need to "make samples" if you want it to install all of the default/sample configs. Which is probably not what you want, but.. |
| 04:54.18 | zerous | i mean when i install the binary, there are configuration files in the asterisk directory but not when in build it. |
| 04:54.25 | snadge | can SIP ALG interfere with BLF? .. the reason im asking, just turned SIP ALG off on a Netgear Nighthawk R8000 x6 router.. and all of a sudden, blf starts working |
| 04:54.38 | snadge | and im failing to understand if thats even possible |
| 04:54.45 | zerous | ChannelZ: oh.. |
| 04:54.46 | ChannelZ | http://burner.com/asterisk-primer/getting-started/ |
| 04:55.13 | snadge | im also failing to understand how its an asterisk related question.. but you guys know everything there is to know about SIP, voip etc.. ;) |
| 04:55.40 | ChannelZ | snadge: well if it's sending your SIP off into space.. or fudging with the headers such that asterisk no longer finds the peer authorized to do things |
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| 04:56.06 | snadge | ChannelZ, thats what i was thinking.. the other sysadmin basically said.. if you can register.. then its not ALG related |
| 04:56.21 | snadge | and that disabling ALG shouldn't have fixed it.. but it clearly did.. i tried lots of things |
| 04:56.27 | Penguin | When it comes to Asterisk, always disable ALG. |
| 04:56.49 | snadge | the nighthawk router also has a firmware update.. i wouldn't be suprised if it has fixes for ALG in it |
| 04:56.54 | snadge | without even bothering to look at the changelog |
| 04:56.56 | zerous | ChannelZ: i did the hello program but i am not getting any console messages. |
| 04:57.10 | ChannelZ | "the hello program" ? |
| 04:57.29 | zerous | ChannelZ: https://wiki.asterisk.org/wiki/display/AST/Hello+World |
| 04:57.33 | zerous | that one. |
| 04:58.12 | ChannelZ | Are you able to connect to the console or are you saying you can't even get asterisk to run? |
| 04:58.40 | zerous | yes, i am able to connect to asterisk |
| 04:59.01 | zerous | and when i make the call, i can hear hello world. |
| 04:59.19 | ChannelZ | Doesn't sound like you have a problem them. |
| 04:59.21 | zerous | but i don't get any console messages when the call is in progress. |
| 04:59.29 | ChannelZ | If it's just the console, you probably didn't turn on verbose |
| 04:59.57 | zerous | i used asterisk -cvvvvv |
| 04:59.59 | ChannelZ | type core set verbose 3 |
| 05:00.16 | zerous | oh.. |
| 05:00.16 | al_nz1 | ChannelZ: it banned you :-) for asterisk |
| 05:00.23 | al_nz1 | but how do I make security seperate log? |
| 05:00.25 | ChannelZ | hurray |
| 05:00.39 | ChannelZ | Well, you edit logger.conf. But you're using a GUI so who knows. |
| 05:00.47 | ChannelZ | /etc/asterisk/logger.conf that is |
| 05:01.23 | zerous | ChannelZ: should i make any changes to the logger.conf to set the verbosity level |
| 05:01.56 | ChannelZ | No |
| 05:02.25 | ChannelZ | logger.conf is just for turning logging on and off. Sorry we're having multiple conversations here, I should be addressing people |
| 05:03.40 | zerous | ChannelZ: thanks a lot :-) |
| 05:03.42 | [TK]D-Fender | zerous: "core set verbose 10", "sip set debug on" |
| 05:03.56 | [TK]D-Fender | zerous: PASTEBIN the complete attempt |
| 05:03.59 | [TK]D-Fender | ~pb |
| 05:03.59 | 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. |
| 05:04.03 | [TK]D-Fender | ^^^ |
| 05:04.22 | [TK]D-Fender | zeroAnd you have told us nothing about the networking involved and what you're calling with. |
| 05:04.45 | zerous | [TK]D-Fender: i used a softphone on a windows machine which is on the same lan. |
| 05:04.58 | zerous | i made the call using xlite. |
| 05:05.07 | [TK]D-Fender | zerous: Enable the debugs I have specified and show us |
| 05:05.14 | zerous | ok.. |
| 05:06.56 | [TK]D-Fender | [00:59]zerousi used asterisk -cvvvvv <- you should also not generally be runnign * direct like this but instead via a standard init script or direct call to "safe_asterisk &" to start it as a daemon, and the "asterisk -r" to CONNECT to it as a running process |
| 05:07.51 | zerous | oh. |
| 05:08.21 | zerous | i would try that too. i am sorry i am really new to asterisk. |
| 05:08.36 | [TK]D-Fender | just start with what I first gave you to prove what's coming in. |
| 05:09.14 | [TK]D-Fender | X-lite also should have given you a status code to say what happened to its attempt |
| 05:09.25 | ChannelZ | You said you heard 'hello world' - you seem to have it working, just not showing in the console. |
| 05:12.15 | [TK]D-Fender | Scrolling back, yes.... |
| 05:12.25 | [TK]D-Fender | well lets see if he does now |
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| 05:50.51 | [TK]D-Fender | heads to bed |
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| 08:42.33 | shadej | hi all |
| 08:43.12 | shadej | is another version of chan_mobile that uses wifi instead of bluetooth? |
| 08:43.33 | shadej | or how much does it take to develop this version? |
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| 09:48.15 | mjt | chan_mobile? What's that? |
| 09:49.27 | mjt | teresting |
| 09:50.49 | mjt | shadej: what do you mean 'using wifi' ? For wifi you have a voip client in a cell phone. |
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| 11:39.16 | shadej | mjt: sorry I was away from my keyboard. do you think voip client can replace the chan_mobile module? |
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| 12:09.42 | mjt | shadej: I don't understand your question |
| 12:10.03 | shadej | do you know chan_mobile? |
| 12:10.29 | mjt | I've read what it is for, yes |
| 12:11.18 | shadej | mjt: okay, so tell me what it is for? |
| 12:11.46 | mjt | you can read about it yourself |
| 12:12.25 | shadej | I know what it is for. am asking to know how much of it you take. |
| 12:12.31 | mjt | but I assumed you know what it is if you asking for a replacement |
| 12:13.18 | shadej | anyways it uses bluetooth to do its purpose, so my question is can use wifi connection to do same thing? |
| 12:14.01 | mjt | can USB be used to do the same thing a COM port was used for? |
| 12:14.13 | mjt | that's what you're asking |
| 12:14.40 | [TK]D-Fender | [08:13]shadejanyways it uses bluetooth to do its purpose, so my question is can use wifi connection to do same thing? <- NO, that would be a very different thing |
| 12:15.23 | mjt | well, the end result will be similar -- ability to place and receive calls |
| 12:15.29 | shadej | [TK]D-Fender: yeah, I first asked if there is another module that uses WIFI to do same thing |
| 12:15.31 | [TK]D-Fender | shadej: And would require an app on the phone to offer that. |
| 12:15.32 | mjt | but done very differently indeed |
| 12:15.43 | *** join/#asterisk blee (~blee@50-88-4-82.res.bhn.net) |
| 12:15.49 | [TK]D-Fender | shadej: Don't expect it to have WiFi. |
| 12:16.14 | [TK]D-Fender | shadej: There as SIP > GSM gateways. So go should a WiFi AP in FRONT of it. |
| 12:16.29 | [TK]D-Fender | shadej: There is just about no market for an all-in-one for that |
| 12:17.14 | shadej | [TK]D-Fender: sorry, are you talking about the phone? |
| 12:17.23 | [TK]D-Fender | shadej: And you wouldn't need a "module" for it.... if it's going over WiFi you'd be using a standard VoIP protocol to talk to it. |
| 12:17.46 | [TK]D-Fender | shadej: If you're looking to still use a CELL PHONE to do the actual call, then you need an APP on it that offfers this |
| 12:17.59 | [TK]D-Fender | shadej: Not some "module" for *. |
| 12:18.07 | [TK]D-Fender | shadej: So go look for an app that offers it |
| 12:18.29 | shadej | [TK]D-Fender: there are VOIP client apps but that is not what I want |
| 12:18.46 | mjt | if the phone don't have a built-in service already, like eg most samsung galaxy phones offers today |
| 12:18.48 | [TK]D-Fender | shadej: Get one that lets you use your cell function as a gateway |
| 12:18.56 | shadej | I want an app that can make my phone(android phone) be used as SIP trunk, that is what chan_mobile does |
| 12:19.21 | eirirs | hum |
| 12:20.07 | shadej | but instead of bluetooth I want to use WIFI. |
| 12:20.08 | mjt | btw, why chan_mobile isn't a standard module? That smells very useful... |
| 12:20.52 | file | chan_mobile is in mainline but it has no maintainer and how well it works is highly dependent on the phone and bluetooth device |
| 12:20.56 | [TK]D-Fender | [08:20]shadejbut instead of bluetooth I want to use WIFI. <- go look for a SIP gateway app then |
| 12:22.05 | shadej | [TK]D-Fender: okay looking.. |
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| 12:31.26 | shadej | [TK]D-Fender: I found this https://play.google.com/store/apps/details?id=com.bonrix.voip.auto.dialer&hl=en but I am not sure if it's voip client or gateway |
| 12:33.21 | [TK]D-Fender | shadej: I recommend reading. |
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| 12:45.47 | shadej | [TK]D-Fender: yeah, but there is no sip gateway app that works as I intend |
| 12:46.01 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-awsexzqxzjenwcsk) |
| 12:46.17 | [TK]D-Fender | shadej: Then there is your answer. |
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| 12:49.06 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 13:03.44 | cervajs2 | hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1&sip/phone2). when i cancel call on phone1, the call is still ringing on phone2 . can i cancel call on both phones? thanks |
| 13:04.10 | newtonr | cervajs2, where are you "cancel"ing the call from? |
| 13:04.49 | cervajs2 | from phone1 (i push button "reject call") |
| 13:04.57 | newtonr | it is two separate outbound legs, so canceling one from the phone that is ringing will not hang up both legs |
| 13:05.09 | newtonr | there may be an option for the Dial app, did you check there? |
| 13:05.51 | cervajs2 | checking now... |
| 13:07.33 | newtonr | I don't see one |
| 13:09.23 | cervajs2 | i'm probably blind but i dont see anything usefull |
| 13:10.52 | newtonr | It may be possible, but I can't remember a way to do it right now. |
| 13:11.36 | cervajs2 | i'll try asterisk-users |
| 13:11.47 | newtonr | Alrighty |
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| 13:14.26 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 13:14.29 | [sr] | hi |
| 13:14.55 | [sr] | got a Wildcard TE131/TE133, but dahdi 2.10x doesn't recognize it |
| 13:14.56 | [sr] | :( |
| 13:15.14 | [sr] | pci:0000:01:00.0 wcte13xp- d161:800a Wildcard TE131/TE133 |
| 13:17.46 | [TK]D-Fender | [09:03]cervajs2hi. i have dialplan with 2 simultaneous calls - dial(sip/phone1&sip/phone2). when i cancel call on phone1, the call is still ringing on phone2 . can i cancel call on both phones? thanks <- the best you could do is dial these as local channels, look at the response from the call that chooses to abort, and then use some dialplan trickery to abort the call. |
| 13:18.39 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:18.47 | Chainsaw | [sr]: TE131 is supported starting 2.9.0 |
| 13:19.16 | [sr] | ChanServ: ok reboot solved it |
| 13:19.17 | [sr] | ops |
| 13:19.19 | [sr] | Chainsaw: |
| 13:19.40 | [sr] | Chainsaw: this card to change to E1 instead of T1, where can i find info? |
| 13:20.13 | Chainsaw | [sr]: I do not have that information for you, hopefully others do. |
| 13:21.05 | cervajs2 | [TK]D-Fender: thanks. example? ;) |
| 13:21.15 | [TK]D-Fender | cervajs2: there is no example |
| 13:22.01 | [TK]D-Fender | cervajs2: Go check the bridgechan for the one that aborts, and track that to the other channels dialaing out and then kill them |
| 13:33.38 | WIMPy | newtonr: That's the feature I have been looking for for a long time, but Dial doesn't have. |
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| 13:37.04 | newtonr | WIMPy, :( |
| 13:37.40 | WIMPy | <standard PBX features blah blah> |
| 13:38.19 | WIMPy | Although I think it should be a rather easy one to add. |
| 13:40.56 | newtonr | If it is a really common feature, then I'm surprised no one has added it yet. |
| 13:41.59 | [sr] | guys, is there an separated oslec for the main dahdi? |
| 13:42.24 | WIMPy | Well, you can the standard BOB via dialplan, but not while ringing. |
| 13:43.48 | WIMPy | +get |
| 13:44.36 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-coemtccbskwulyua) |
| 13:44.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:53.46 | *** join/#asterisk z3bra (z3bra@gateway/shell/blinkenshell.org/x-rhlymokaipyofgtr) |
| 13:53.50 | z3bra | Hi |
| 13:56.17 | z3bra | I'm configuring asterisk to use dundi, but there is somthing I don't get. I have two servers A and B. extension from 10 to 100 are mixed between the two servers (so 23 can be on A, and 24 on B). Am I supposed to specify ALL the local extensions to dundi ? |
| 14:09.59 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 14:12.48 | [sr] | is there a separated package for oslec? |
| 14:13.34 | *** join/#asterisk madax (~joshua@200.174.18.34) |
| 14:17.00 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
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| 14:46.55 | [sr] | guys, simple way to enable the oslec shipped on dahdi package? |
| 14:47.22 | *** part/#asterisk z3bra (z3bra@gateway/shell/blinkenshell.org/x-rhlymokaipyofgtr) |
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| 15:09.17 | Lex2 | Hey, is there a guide for newbies on how to install and set up Asterisk on a droplet? |
| 15:10.44 | [TK]D-Fender | Installing * is the same process.... |
| 15:10.53 | [TK]D-Fender | instructions come bundled with the tarball... |
| 15:11.00 | [TK]D-Fender | And there is that wonderful book... |
| 15:11.02 | [TK]D-Fender | ~book |
| 15:11.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:11.04 | [TK]D-Fender | ^^^ |
| 15:11.06 | [TK]D-Fender | And the wiki... |
| 15:11.42 | Lex2 | Yep |
| 15:11.52 | Lex2 | I have no idea on how to use Linux well |
| 15:12.17 | Lex2 | Should probably learn commands first |
| 15:12.19 | *** join/#asterisk net_ (79f28b33@gateway/web/freenode/ip.121.242.139.51) |
| 15:12.28 | [TK]D-Fender | probably. |
| 15:12.39 | [TK]D-Fender | It is an expected prerequisite... |
| 15:12.57 | net_ | can I use READ application continously two or three times? |
| 15:13.18 | [TK]D-Fender | You can call it as many times as you want |
| 15:13.26 | [TK]D-Fender | Each call to Read will do whatever you tell it to do... |
| 15:17.28 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:29.23 | net_ | [TK]D-Fender: I am using two READ application and gathering dtmf value in two different variable, since I have two different messages to play, but the dtmf value is not capturing... |
| 15:30.07 | [TK]D-Fender | net_: then perhaps you have the wrong mode and * isn't seeing it |
| 15:30.16 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 15:31.23 | net_ | this is my dialplan http://pastebin.com/LVgaJtcT |
| 15:32.24 | [TK]D-Fender | Show us the call... |
| 15:32.26 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:32.26 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:38.41 | net_ | http://pastebin.com/0wZCwTMy |
| 15:39.02 | net_ | I think if i use the gotoif statment between the read application it will work fine |
| 15:39.28 | [TK]D-Fender | - User entered '1' |
| 15:39.33 | [TK]D-Fender | So it's getting input. |
| 15:39.42 | [TK]D-Fender | And we didn't move on to see what happened for the 2nd one |
| 15:41.57 | net_ | I think I got it... its my mistake... again getting the input its going to next READ statement... so i will place the gotoif condition between the read... so upon first input it will go the desired context |
| 15:46.15 | *** join/#asterisk Pernat (~Pernat@host15-89-206-24.limes.com.pl) |
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| 17:03.19 | Pernat | hi |
| 17:03.59 | Pernat | i just installed debian 7 and asterisk 11.13 and compiled chan_dongle to it and i have a problem |
| 17:05.01 | Pernat | Asterisk CLI shows me: Channel 'Dongle/dongle0-0100000000' sent to invalid extension but no invalid handler: context,exten,priority=default,+XXXXXXXXXXX,1 when i try to call GSM number |
| 17:12.17 | Pernat | okay. i solved this... |
| 17:25.17 | Pernat | now i have to change permissions to ttyUSB after any reboot of system. |
| 17:25.22 | Pernat | some ideas? |
| 17:27.37 | [TK]D-Fender | Go find a spot to do it. modify your * startup script maybe. Or use something like rc.local |
| 17:27.48 | [TK]D-Fender | It's your systsme... do it wherever you feel is appropriate |
| 17:28.22 | WIMPy | That's what udev rules are there for. |
| 17:29.10 | WIMPy | Although I fully understand if you don't want udev. |
| 17:35.42 | Pernat | it will work if i add chmod 777 /dev/ttyUSB1 to rc.local? |
| 17:41.50 | *** join/#asterisk madax (~joshua@191.181.94.168) |
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| 17:51.01 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 17:51.45 | mjt | Pernat: it'll restore the perms once you re-plug it |
| 17:52.31 | Pernat | so how to set permissions for dongle? |
| 17:52.48 | WIMPy | udev |
| 17:53.38 | *** join/#asterisk linuxfool (~james@DHCP-149-228.resnet.ua.edu) |
| 17:56.32 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 18:00.23 | Pernat | if i create file in /etc/udev/rules.d/ and write in it ATTR{idVendor}=="12d1" ATTR{idProduct}=="0001", MODE="777" it shall work? |
| 18:05.36 | mjt | something like that, yes. maybe you want group=something mode=0660. mode=777 makes no sense, as it is not an executable, but that's minor. |
| 18:05.49 | mjt | (or just user=you) |
| 18:06.25 | mjt | or maybe you need to make it to follow permissions of a logged-in user, some other devices are (eg /dev/kvm) |
| 18:06.40 | mjt | at any rate this is not an asterisk question :) |
| 18:07.31 | Pernat | without permissions asterisk don't use dongle and i have calling signal, when i manually change permissions it do dialplan |
| 18:10.00 | Pernat | that rule didn't work |
| 18:17.30 | *** join/#asterisk zerick (~eocrospom@190.118.16.131) |
| 18:18.52 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 18:34.01 | Penguin | Why would /dev/ttyUSB1 need to be executable? |
| 18:34.45 | Penguin | Ah, mjt already questioned that. |
| 18:37.44 | ipengineer | When I do a core show channels verbose I have a device with several hung channels. Is there a way to hang them up? The channels are all PJSIP channels: PJSIP/201 |
| 18:38.10 | Penguin | Did you try a soft hangup? |
| 18:38.14 | Penguin | channel request hangup ... |
| 18:38.38 | Pernat | i need change permissions to let asterisk use usb modem |
| 18:38.45 | Pernat | and 777 was example |
| 18:38.57 | ipengineer | I didnt think soft hangup was an option anymore in v12 |
| 18:39.24 | Penguin | If they took out the ability to hangup a channel, I'll never upgrade to 12. |
| 18:39.25 | ipengineer | I did try channel request hangup and that is wqhen I got 201 is not a known channel even though I put ârequest hangup PJSIP/200â |
| 18:39.42 | ipengineer | âchannel request hangup PJSIP/201â sorry |
| 18:39.52 | Penguin | Is the channel name really PJSIP/201? |
| 18:39.57 | Penguin | It isn't likely that. |
| 18:40.20 | ipengineer | When I do core show channels verbose that is all it shows.. I wouldnt think so either.. |
| 18:40.40 | Penguin | What you should be doing is using Tab-key completion. |
| 18:40.55 | Penguin | channel request hangup PJSIP/201<Tab> |
| 18:41.54 | ipengineer | There you go.. It said requested hangup but didnt hang them up |
| 18:42.15 | Penguin | I encounter that a lot. |
| 18:42.22 | Penguin | Something is blocking that channel. |
| 18:43.08 | ipengineer | I guess the only option is to restart asterisk |
| 18:43.15 | Penguin | That's what I end up doing. |
| 18:43.46 | [TK]D-Fender | Or toss them off a cliff |
| 18:43.47 | Penguin | You could just leave it. |
| 18:44.34 | ipengineer | Thats what I was afraid of.. Ok Thanks.. Not good for a production environment for sure. That user was trying to transfer a call and the extension didnt exist or the res_parking had not been reloaded. The problem is it messes their hints and everything else up. Can always restart after hours |
| 18:44.59 | Penguin | core restart when convenient |
| 18:45.20 | WIMPy | If it restarts. |
| 18:45.49 | WIMPy | So the conclusion is to not transfer calls when using Asterisk. |
| 18:45.49 | Penguin | Ah, yeah... that's a good point. It probably won't actually restart that way because of the channels existing. |
| 18:45.53 | *** join/#asterisk jpsharp (~jsharp@c-76-101-138-247.hsd1.fl.comcast.net) |
| 18:46.20 | WIMPy | For me that usually ends with Asterisk locking up since a few versions. |
| 18:46.30 | *** join/#asterisk babak_ (02b2b806@gateway/web/cgi-irc/kiwiirc.com/ip.2.178.184.6) |
| 18:46.33 | Penguin | seemingly |
| 18:46.37 | Penguin | It doesn't really lock up. |
| 18:46.47 | Penguin | You just don't have the patience to wait for it to restart. |
| 18:46.49 | WIMPy | Mine does lock up. |
| 18:46.59 | Penguin | I've thought that about mine as well. |
| 18:47.06 | WIMPy | It won't do anything sensible untill I kill -9 it. |
| 18:47.10 | Penguin | It just hangs the current CLI. |
| 18:47.14 | jpsharp | I'm getting Asterisk internal timing slips: "Expected to acknowledge 1 ticks but got 3 instead" using timerfd. Is that just a matter of my system being overloaded? Or would shifting to differeing timing source help. |
| 18:47.19 | ipengineer | WIMPy: I have printed a new label: NO TRANSFERS now taped to every phone |
| 18:47.20 | ipengineer | ok maybe not |
| 18:47.32 | WIMPy | Ah, well, yes, but the thing is that you have to wait forever. |
| 18:48.06 | WIMPy | But recently I also had the issue that core restart actually just stopped Asterisk without restarting it. |
| 18:48.24 | Penguin | And, as you mentioned, it may never restart. But that's because channels exist that will not go away, therefore asterisk doesn't find the convenient time that it is waiting for. |
| 18:50.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:51.12 | ipengineer | Dont you have to do âcore restart nowâ to force it |
| 18:51.59 | WIMPy | Might work. |
| 18:52.11 | WIMPy | But will also kill calls, if present. |
| 18:52.43 | Penguin | I have a script that runs from cron, so I just core stop now and wait on the next cron execution to restart asterisk. |
| 18:55.57 | *** join/#asterisk al_nz1 (~bigal.nz@118-93-220-69.dsl.dyn.ihug.co.nz) |
| 18:56.10 | al_nz1 | join #fail2ban |
| 18:56.27 | al_nz1 | woops |
| 19:07.33 | *** join/#asterisk linuxfool (~james@DHCP-149-228.resnet.ua.edu) |
| 19:16.09 | cmewr1 | this is soooo complicated |
| 19:22.18 | *** join/#asterisk calum_ (~calum_@host86-176-63-147.range86-176.btcentralplus.com) |
| 19:25.58 | mjordan | Penguin: no, we did not take out the ability to hang up a channel |
| 19:26.11 | mjordan | ipengineer: if you can figure out how those channels were getting stuck, please, please, please file an issue |
| 19:26.32 | cmewr1 | so I think I've installed Asterisk - it's running |
| 19:26.38 | cmewr1 | Now what should I do |
| 19:26.39 | jpsharp | That's a good start |
| 19:26.58 | Penguin | I was pretty sure it wasn't taken out. I'm certain he was just confused about how soft hangups work in modern asterisk. |
| 19:27.02 | cmewr1 | My end goal is to connect it to an outside line and setup rules |
| 19:27.19 | cmewr1 | like connect it to flowroute |
| 19:27.19 | jpsharp | Then you need a connection hardware of some sort. |
| 19:27.31 | jpsharp | Or an account with flowroute |
| 19:27.39 | cmewr1 | jpsharp: I have an account with them |
| 19:27.49 | *** join/#asterisk babak_ (413144c5@gateway/web/cgi-irc/kiwiirc.com/ip.65.49.68.197) |
| 19:28.26 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 19:28.29 | cmewr1 | What are the next steps? |
| 19:28.32 | ipengineer | mjordan: Will do. Right now I am trying to figure out why Asterisk is not consistently sending a BYE for certain contacts.. I cannot find out what exactly causes it but when I do will let yall know⦠Trying to figure out if I can reproduce it in 13b2 or if its just an issue in 12.5. |
| 19:28.50 | jpsharp | Configure asterisk to talk to flowroute in the SIP configurations, then build your dialplans to talk to your other devices. |
| 19:29.17 | mjordan | ipengineer: I know we just fixed one bug related to that. You may want to try the latest from the 12 or 13 branch |
| 19:29.17 | ipengineer | mjordan: I am pretty sure the hung channels were being caused by a user trying to transfer a call to a parkinglot and the res_parking module had not been reloaded to include that parking lot so it didnt exist. |
| 19:29.26 | mjordan | interesting |
| 19:29.45 | cmewr1 | jpsharp: where can I find an easy to understand guide? |
| 19:29.54 | Penguin | ~primer |
| 19:29.54 | infobot | somebody said primer was http://burner.com/asterisk-primer |
| 19:29.59 | jpsharp | ~book |
| 19:29.59 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:30.07 | jpsharp | There you go. |
| 19:30.16 | ipengineer | mjordan: Ok that may explain why I am having a hard time reproducing in 13. I will keep hamering it and see what I come up with.. I just didnt see anything in the release notes about it (prob overlooked it). |
| 19:30.23 | cmewr1 | I have the book in front of me |
| 19:30.30 | cmewr1 | It's all command based |
| 19:30.35 | cmewr1 | Which is confusing |
| 19:30.37 | jpsharp | Yes. |
| 19:31.02 | jpsharp | Native asterisk has no GUI. You'll need a 3rd party system for that. |
| 19:31.22 | cmewr1 | OK |
| 19:31.33 | Penguin | Or just read the book and the primer. You don't NEED a GUI to write text in files. |
| 19:32.09 | cmewr1 | Sure. |
| 19:33.26 | cmewr1 | So my /etc/asterisk directory is empty by default |
| 19:33.33 | cmewr1 | Is this normal? |
| 19:33.39 | jpsharp | ONly if you haven't run "make samples" |
| 19:33.49 | cmewr1 | Gave me an error |
| 19:33.52 | Penguin | You'll need the sample files to help you along. |
| 19:34.01 | mjordan | ipengineer: the patch for that in 13 didn't hit beta2, it's still in the branch - so it wouldn't be in release notes yet |
| 19:34.32 | cmewr1 | no rule to make target samples |
| 19:34.49 | jpsharp | you have to be in the Asterisk source directory. |
| 19:35.10 | Penguin | (the same one where you make and make install) |
| 19:35.20 | jpsharp | What he said |
| 19:36.47 | ipengineer | mjordan: Ok. Yea I am on b2 I didnt pull in branch. |
| 19:37.52 | cmewr1 | I have no idea what I'm doing |
| 19:38.04 | Penguin | Did you install asterisk from source? |
| 19:38.08 | cmewr1 | Yep |
| 19:38.12 | cmewr1 | It's running |
| 19:38.13 | Penguin | Do you remember ./configure? |
| 19:38.20 | Penguin | and make? and make install? |
| 19:38.23 | cmewr1 | It's in the main directory /root] |
| 19:38.42 | Penguin | Go to the same place where you did ./configre and make and make install. |
| 19:38.50 | Penguin | Then make samples |
| 19:40.16 | cmewr1 | 'No rule to make `samples |
| 19:40.39 | Penguin | Also, if you're in /root, that means you are ./configure-ing and make-ing as root, which isn't a very great idea. |
| 19:40.50 | cmewr1 | Sorry I was wrong |
| 19:40.59 | cmewr1 | It's in /src/asterisk/asterisk |
| 19:41.10 | cmewr1 | asterisk-complete/asterisk |
| 19:41.18 | Penguin | Go there and make samples |
| 19:41.26 | cmewr1 | It's giving me that error |
| 19:41.35 | Penguin | Then it's probably the wrong place. |
| 19:42.05 | cmewr1 | I don't think it is |
| 19:42.12 | cmewr1 | It's got all the files there |
| 19:43.03 | cmewr1 | ah there we go |
| 19:44.32 | cmewr1 | so in the /etc directory, are these where all the main files are, or these are just samples? |
| 19:46.04 | Penguin | The typical directory for all asterisk config files is /etc/asterisk, but you can change that if you know what you're doing. |
| 19:46.14 | mjordan | cmewr1: have you read Asterisk: The Definitive Guide, or read any of the content on the Asterisk wiki? |
| 19:46.23 | Penguin | You ordinarily have the samples in there, too. |
| 19:46.33 | cmewr1 | mjordan: yes |
| 19:46.41 | cmewr1 | I'm finding it way too complex |
| 19:46.48 | mjordan | you probably should go back and re-read the setting up Asterisk chapters in the book, and follow it carefully |
| 19:47.00 | cmewr1 | Sure. |
| 19:47.15 | Penguin | Some of the samples don't need very much work to be able to use them. Others are not to be used -- they are only samples. |
| 19:47.24 | Penguin | And read the primer again. |
| 19:47.27 | Penguin | ~primer |
| 19:47.27 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 19:47.44 | cmewr1 | By the word Samples, do you mean the settings .conf files? |
| 19:47.51 | cmewr1 | in /etc/asterisk |
| 19:47.53 | Penguin | Yes. Sample files. |
| 19:48.12 | cmewr1 | So they'er not the original settings files, or they are? |
| 19:48.23 | *** join/#asterisk echelon (~echelon@gateway/tor-sasl/harel) |
| 19:48.26 | Penguin | The sample files contain examples and commentation of most of the settings used within the respective file. |
| 19:48.39 | Penguin | <Penguin> Some of the samples don't need very much work to be able to use them. Others are not to be used -- they are only samples. |
| 19:49.03 | cmewr1 | So where are the originals stored? |
| 19:49.12 | Penguin | That doesn't make sense to me. |
| 19:49.21 | Penguin | What are "originals?" |
| 19:49.32 | cmewr1 | You say Samples, so I'm thinking they are copys |
| 19:49.42 | cmewr1 | or are they files which make the system work if replaced? |
| 19:49.43 | Penguin | They are EXAMPLES. |
| 19:49.55 | Penguin | You've got to read the words I am typing. |
| 19:50.28 | Penguin | I'm beginning to see why the instruction manual is hard for you. |
| 19:50.28 | cmewr1 | I'm trying to understand. What I understand is that if you replace these files with whatever you customised, it'll take effect in Asterisk at reboot |
| 19:51.05 | cmewr1 | What am I missing? |
| 19:51.19 | Penguin | Take sip.conf for example. |
| 19:51.26 | cmewr1 | Yep |
| 19:51.43 | Penguin | The one that you get when you make samples is just an EXAMPLE of what your actual sip.conf will look like. |
| 19:51.59 | cmewr1 | But what do you mean by Make Samples? |
| 19:52.06 | cmewr1 | You mean the originals? |
| 19:52.38 | Penguin | Good luck with your project. I see I'm not going to make any more progress here. |
| 19:52.54 | cmewr1 | Penguin: I'm trying to understand what you're telling me |
| 19:53.49 | cmewr1 | Is a sample file like an untouched settings file which if you edit, will change Asterisk settings if replaced? |
| 19:54.43 | newtonr | cmewr1, the sample file naming is sort of misnomer. The sample files are pretty much documentation. They are not intended to be "defaults" and most them are not even examples. |
| 19:54.59 | cmewr1 | aha |
| 19:55.00 | newtonr | They do contain a lot of example settings |
| 19:55.04 | cmewr1 | That makes sense |
| 19:55.21 | cmewr1 | So where are the main config files, or should I use it from the commandline? |
| 19:55.49 | mjordan | cmewr1: Have you considered FreePBX? |
| 19:55.57 | Penguin | bellows |
| 19:56.27 | mjordan | I'm not trying to discourage you, but I'm not sure Asterisk's mechanism of configuration is for you. You may find that FreePBX - which is a far more end-user friendly system - is more to your liking |
| 19:56.56 | cmewr1 | OK |
| 19:57.13 | *** join/#asterisk linuxfool (~james@DHCP-149-228.resnet.ua.edu) |
| 19:57.15 | cmewr1 | I'll try again with the primer |
| 19:57.16 | mjordan | It's still an Asterisk based system, and is very powerful, but you can use a UI to configure it. |
| 19:57.37 | cmewr1 | ok |
| 19:58.15 | newtonr | cmewr1, If you continue to experiment with directly configuring Asterisk without a GUI - know that there are only the samples. There are no "main" config files as you asked about. Most users would create a blank .conf file for whatever module they are configuring and then write it from scratch based on their knowledge of the configuration (or whilst referencing the sample files). |
| 19:59.16 | cmewr1 | newtonr: sure. So where would the edited sample file be saved? Say if I'm adding my flowroute account info |
| 20:00.02 | newtonr | cmewr1, Typically /etc/asterisk/ - See https://wiki.asterisk.org/wiki/display/AST/Directory+and+File+Structure for more about the various directories used. |
| 20:00.18 | cmewr1 | So they are actual system files |
| 20:00.39 | cmewr1 | Config files which make Asterisk work |
| 20:01.47 | newtonr | cmewr1, Glancing over https://wiki.asterisk.org/wiki/display/AST/Hello+World will quickly give you an idea for very very basic configuration. |
| 20:02.07 | cmewr1 | ok |
| 20:02.11 | cmewr1 | I'll take another look |
| 20:11.20 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 20:34.31 | *** join/#asterisk tenspeed705 (~keiths@goatfarm.vianet.ca) |
| 20:35.15 | tenspeed705 | Hey guys, im trying to use aAUDIOHOOK_INHERIT(MixMonitor)=yes when I transfer the call, the audio is still stopping. any ideas? |
| 20:36.39 | tenspeed705 | this is 11.13.0 |
| 20:38.13 | *** join/#asterisk ipengineer_ (~zconkle@static-72-64-118-10.dllstx.fios.verizon.net) |
| 20:55.07 | *** join/#asterisk theron_ (~theron@199.201.65.135) |
| 20:57.15 | Penguin | tenspeed705: Can you show me the dialplan where you are setting that? |
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| 21:21.00 | Kobaz | i love how audiocodes gateways randomly become unregistered for no reason |
| 21:21.19 | Kobaz | and you reset them and then it works fine, no configuration changes |
| 21:33.56 | ChannelZ-Wk | Siesta! |
| 21:38.35 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
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| 22:14.28 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-coemtccbskwulyua) |
| 22:22.26 | snadge | Kobaz, randomly and for no reason? ;) .. im guessing that it reaches the registration expiry time.. then fails to re-register |
| 22:22.50 | snadge | and then you have to consider which of the 50,000 different routers you are using.. and how its configured |
| 22:23.05 | snadge | i work for a voip provider, who supports random configurations |
| 22:23.12 | snadge | and when i say random.. i really do mean it |
| 22:24.48 | snadge | every day.. some random combination finds a new inventive way to fail |
| 22:25.56 | snadge | ... and then you have mobile broadband providers .. using double, and apparently sometimes triple layers of NAT.. randomly disabling inbound calls, and then denying it.. etc |
| 22:27.15 | snadge | i would wager most asterisk devs are sheltered from ALL of that.. and with good reason |
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| 23:08.36 | Kobaz | snadge: it doesn't even attempt to re-register |
| 23:08.41 | Kobaz | it just dies |
| 23:09.07 | Kobaz | snadge: i have one audiocodes that drops the lan link after 2 hours |
| 23:10.14 | snadge | registration expiry on a cisco ata is 1 hour.. from memory |
| 23:10.44 | snadge | that's generally the problem though.. the server and the client keep track of whether they are registered, and need to re-register.. seperately |
| 23:11.10 | *** join/#asterisk wrxed (~rrittgarn@72.252.11.170) |
| 23:11.39 | snadge | check for updates ? bin the audiocodes maybe |
| 23:12.48 | wrxed | Good evening. Working with a new client in a remote location (non us) trying to get DAHDI channels working. Specifically when i have an unanswered call hang up, asterisk doesn't hang it up when the caller hangs up. Not sure on any of the settings as I've only worked with SIP up till today... Digium 8 port FXO card. |
| 23:13.15 | wrxed | Also, does dahdi restart reload most of the chan_dahdi settings? |
| 23:35.55 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 23:41.54 | Kobaz | snadge: i've had many audiocodes randomly fail... i blame the audiocodes mostly |
| 23:42.18 | Kobaz | i've had some that just refuse to boot properly |
| 23:42.22 | Kobaz | plug them in and the fail light is on |
| 23:42.35 | snadge | at least it has a fail light |
| 23:44.06 | Kobaz | and then there's the aforementioned one that just randomly drops lan |
| 23:44.19 | Kobaz | and it's not the cable or the network switch port |
| 23:44.49 | Kobaz | as crappy as grandstream boxes are, at least they never drop off the network for no reason |