| 00:54.17 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:54.19 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1.1 (2010/12/15), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 01:01.09 | *** join/#asterisk geneg1 (~gene@173-230-163-176.cable.teksavvy.com) |
| 01:15.02 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 01:20.56 | ChannelZ | infobot! |
| 01:21.03 | ChannelZ | fondles infobot |
| 01:21.49 | xSmurf | how would one be able to continously pipe the output of espeak or text2wave into a MoH fifo? |
| 01:23.12 | ChannelZ | I assume either of those tools are only generating chunks of audio.. IE you run it, it says something, and then quits |
| 01:24.34 | ChannelZ | in which case that'd be a trick to inject into a stream, besides doing it ghetto-style feeding the actual audio output of a soundcard into the input with a tool that is recording the input doing the streaming. |
| 01:24.57 | ChannelZ | Maybe you could kludge together something with an shoutcast server or such |
| 01:28.55 | xSmurf | I was thinking about that |
| 01:29.06 | xSmurf | but it's getting complicated |
| 01:29.43 | xSmurf | and yeah espeak text2wave only seem to do chunks :/ |
| 01:32.34 | ChannelZ | hire an illegal to just stand by and read whatever you want into a microphone on demand |
| 01:34.43 | ChannelZ | oh! no! |
| 01:34.49 | ChannelZ | You need that homeless guy! |
| 01:35.06 | *** join/#asterisk slaney (~slaney@li102-143.members.linode.com) |
| 01:35.15 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 01:35.31 | xSmurf | ;P |
| 01:35.45 | ChannelZ | http://www.youtube.com/watch?v=hE0LYI84aw8 |
| 01:36.17 | xSmurf | fuck man I have no idea how to do this :( |
| 01:36.39 | ChannelZ | Well then maybe it's not worth doing |
| 01:36.51 | xSmurf | no, it is |
| 01:37.17 | ChannelZ | ok don't click that youtube link, it features an annoying douchebag commenting |
| 01:37.25 | xSmurf | ok |
| 01:38.29 | ChannelZ | http://www.youtube.com/watch?v=ou_XuEEGQA8 there just the original |
| 01:49.08 | xSmurf | cat moh application be "cat /path/to/wav.raw"? |
| 01:49.20 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 01:53.59 | ChannelZ | well, assuming the data was in the right format I guess it would work |
| 01:55.23 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.6) |
| 02:01.08 | xSmurf | basically what I'm looking for is the opposite of streamplayer |
| 02:05.01 | Freeaqingme | that would be a reyalpmaerts? |
| 02:07.18 | *** join/#asterisk slobberknocker (~ckwall@c-76-27-9-24.hsd1.ut.comcast.net) |
| 02:08.52 | slobberknocker | having something quirky going on... wondering if anyone could make a suggestion. I have two phones set up at my house and they connect to my server which is remote. One is a linksys phone and the other is an ATA connected phone. I see periodically one or the other extension will go unavailable from the sip show peers. Once i even set up a second extension on the lynksis phone and only one of the two extensions went unavailable. An |
| 02:10.56 | Freeaqingme | slobberknocker, your question got cut off at "went unavailable. An" |
| 02:11.46 | slobberknocker | having something quirky going on... wondering if anyone could make a suggestion. I have two phones set up at my house and they connect to my server which is remote. One is a linksys phone and the other is an ATA connected phone. I see periodically one or the other extension will go unavailable from the sip show peers. Once i even set up a second extension on the lynksis phone and only one of the two extensions went unavailable. An |
| 02:11.54 | slobberknocker | did that one come through 100%? |
| 02:14.03 | Freeaqingme | no |
| 02:14.11 | Freeaqingme | just begin your next sentence at 'An' ;) |
| 02:14.17 | Freeaqingme | (irc has got a maxlength for lines) |
| 02:15.05 | slobberknocker | Any suggestions on where to begin looking? |
| 02:15.17 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:15.30 | Freeaqingme | slobberknocker, you didn't even ask your question in its entirety. |
| 02:15.40 | Freeaqingme | just finish your question, by beginning your next sentence at 'An' |
| 02:17.36 | slobberknocker | that was it.. the missing part of what i typed was "Any suggestions on where to begin looking?" |
| 02:17.43 | Freeaqingme | ah, hehe ;) |
| 02:19.25 | slobberknocker | any ideas? |
| 02:20.23 | slobberknocker | scratches head |
| 02:20.59 | Freeaqingme | slobberknocker, what's the nat ttl? |
| 02:21.09 | slobberknocker | ttl? |
| 02:21.17 | Freeaqingme | the keepalive thingy? |
| 02:21.37 | slobberknocker | all i have is nat=yes |
| 02:21.42 | slobberknocker | and qualify=1000 |
| 02:21.56 | slobberknocker | ive tried 1000, 2000, 6000. makes no diff |
| 02:22.20 | Freeaqingme | uhm, qualify is a boolean? |
| 02:22.24 | Freeaqingme | oh, no |
| 02:22.27 | Freeaqingme | it can be |
| 02:22.30 | slobberknocker | in sip.conf |
| 02:22.32 | Freeaqingme | slobberknocker, try 500 |
| 02:22.37 | jaytee | qualify=yes is the same as qualify=2000 |
| 02:22.55 | Freeaqingme | yes, I missed the first line of the doc page |
| 02:23.30 | slobberknocker | is there a way to send a command to a device to have it re-register |
| 02:24.10 | slobberknocker | kind of like me resetting the device if i were actually at home? |
| 02:35.05 | xSmurf | in the cli this seems to work tail -n0 -f /var/spool/asterisk/tmp/live.100.wav | sox -r 16000 -t wav - -r 8000 -c 1 -t raw - vol 1.00 |
| 02:35.34 | xSmurf | but it doesn't seem to work as moh application |
| 02:36.20 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 02:36.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:38.12 | ChannelZ | might be because of the pipes, write it as a script |
| 02:41.34 | xSmurf | seems like a no go |
| 02:42.32 | *** join/#asterisk cnu (cnu@2001:470:28:1fe::10) |
| 02:42.44 | xSmurf | I don't think it even fires up |
| 02:43.42 | xSmurf | oh yeah it does |
| 02:44.22 | xSmurf | oh |
| 02:44.23 | xSmurf | ooooh |
| 02:44.31 | xSmurf | oooooooh |
| 02:44.54 | xSmurf | except it doesn't want to decode or something and the output it slowed down |
| 02:47.53 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.6) |
| 02:47.54 | xSmurf | it's like it wants to work, right... ;p |
| 02:56.44 | xSmurf | ooooooooooh |
| 02:58.26 | xSmurf | it works!!! |
| 02:58.37 | xSmurf | need to run sox before tail |
| 02:58.51 | xSmurf | though it doesn't work if I write remotely, gets slowed down again :/ |
| 03:01.31 | xSmurf | sweeeeet |
| 03:01.33 | xSmurf | hahaa |
| 03:01.36 | xSmurf | ah crap |
| 03:01.44 | xSmurf | well close |
| 03:03.54 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 03:13.44 | xSmurf | buuuut nope |
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| 03:31.42 | xSmurf | I don't get this, I have a script that checks a text file and passes it to text2wave and sox |
| 03:31.49 | xSmurf | and the moh app is just a cat on a fifo |
| 03:31.52 | xSmurf | it works just fine |
| 03:31.53 | xSmurf | but |
| 03:32.05 | xSmurf | I cannot start the listener script from the same script as the cat |
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| 04:08.03 | xSmurf | YES \o/ |
| 04:13.41 | ChannelZ | Or possibly NO /o\ |
| 04:22.38 | xSmurf | And they said it couldn't be done!! |
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| 04:41.34 | Dovid | morning ev1 |
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| 04:58.57 | mick_laptop | anyone here have the dCAA or dCAP? |
| 04:59.09 | mick_laptop | where can you order the test to take it? |
| 04:59.25 | mick_laptop | or do you need to do the training first to be able to take the test? |
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| 06:14.09 | Supari | hey everyone |
| 06:14.26 | Supari | ihave the same issue as https://issues.asterisk.org/view.php?id=18299 |
| 06:14.52 | Supari | iam running 1.8.1.1 and the bug says issue is fixed but i still get theissue anyone know a fix ? |
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| 07:31.01 | ChannelZ | Supari: get 1.8.2-rc2 |
| 07:31.14 | ChannelZ | that issue fix isn't in 1.8.1.1 |
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| 07:57.32 | shinao1 | Hi! I'm looking for a cost effective Asterisk based call center solution with ACD, Skill-based routing, and the like? Any ideas? And pricing? |
| 07:58.22 | antiwire | oh my |
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| 08:25.36 | Dovid | try queuemetrics |
| 08:25.39 | Dovid | ev1 seems to love em |
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| 09:02.36 | shinao1 | how much does it cost, Dovid ? |
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| 09:17.55 | Dovid | no idea |
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| 09:41.56 | shinao1 | queuemetrics is a reporting tool.. looking for a full-blown call center |
| 09:56.44 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 10:07.40 | *** join/#asterisk asterisknooby (~dimko@ip-78-45-38-173.net.upcbroadband.cz) |
| 10:07.42 | asterisknooby | hello |
| 10:08.09 | asterisknooby | first of all, why u guys decided to create community based on shitty service like freenode? |
| 10:08.36 | asterisknooby | it sucks arse badly, i was lucky i have experience with IRC, otherwise i wouldnt be able to log in here... |
| 10:08.48 | asterisknooby | but i am here not for bitching |
| 10:10.48 | asterisknooby | the question is, dos anyone have working sip instalation? |
| 10:11.08 | asterisknooby | i want to test me little aste-risky... |
| 10:12.43 | asterisknooby | i tried some online links, and exho tests, but they only test my client i guess |
| 10:12.51 | asterisknooby | they dont test incoming calls |
| 10:13.32 | asterisknooby | can anyone call me please on sip:203@dimko.eu please? |
| 10:13.36 | asterisknooby | pretty pleasE? |
| 10:17.06 | asterisknooby | so anyone please? |
| 10:18.35 | ChannelZ | can I at least get a reacharound? |
| 10:19.46 | ChannelZ | SIP/dimko.eu-00000028 is circuit-busy |
| 10:20.11 | asterisknooby | 8-/ |
| 10:20.17 | asterisknooby | nothing in -vvvv |
| 10:20.28 | ChannelZ | maybe it's not freenode who is shitty service |
| 10:22.46 | ChannelZ | <--- SIP read from UDP:78.45.38.173:5060 ---> |
| 10:22.46 | ChannelZ | SIP/2.0 404 Not Found |
| 10:23.09 | asterisknooby | my ip behid nat |
| 10:23.22 | asterisknooby | asterisk - is nat, machine behind nat |
| 10:23.48 | asterisknooby | but if i understood it correctly, asterisk supposed to handle this, like connection supposed to go over asterisk, right? |
| 10:23.55 | asterisknooby | or is it still p2p connection? |
| 10:24.12 | ChannelZ | it will if properly configured |
| 10:24.33 | ChannelZ | seems you're already port forwarding, *something* sent back a response to me. |
| 10:26.09 | asterisknooby | http://pastebin.com/cFQPxM6q |
| 10:26.40 | asterisknooby | canreinvite=no this option should do the trick, right? |
| 10:26.45 | *** join/#asterisk dr_ (~duckz@78.96.111.117) |
| 10:28.18 | ChannelZ | not really no |
| 10:28.20 | asterisknooby | thisoption i was talking about, is it sip proxy by any chance? |
| 10:28.29 | ChannelZ | ~sipnat |
| 10:28.29 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 10:29.35 | ChannelZ | also judging by your config me calling the URI you gave me is not going to work since apparently 203 is a SIP device, not an extension, and you're not setup for 'anonymous' sip |
| 10:30.21 | ChannelZ | or actually, allowguest might default to on now that I think about it... |
| 10:31.15 | asterisknooby | do i have to mention "externip" option if sip is on router? |
| 10:31.48 | ChannelZ | if it's behind a firewall, yes. And localnet |
| 10:32.33 | asterisknooby | well, no, gateway = asterisk server in my case |
| 10:34.06 | ChannelZ | if it has a real IP then no |
| 10:34.25 | ChannelZ | as we established your system is responding. It's just not configged right all the way. |
| 10:34.29 | ChannelZ | There is no extension 203 |
| 10:34.38 | ChannelZ | (apparently in the context 'home') |
| 10:35.42 | asterisknooby | from sip confic under 203: "context=users" |
| 10:36.22 | ChannelZ | 203 is a device, not an extension |
| 10:36.50 | asterisknooby | http://pastebin.com/2P9Yh9YW |
| 10:37.05 | asterisknooby | this is extentions.conf |
| 10:37.38 | WIMPy | noone at home |
| 10:37.39 | ChannelZ | I am not authenticating as a user. So 'anonymous' calls will wind up in the context 'home' based on your sip.conf |
| 10:37.47 | ChannelZ | and there are no extensions in home apparently |
| 10:38.09 | asterisknooby | but why would u want ananymous caller to make calls to you in the first place? |
| 10:38.54 | ChannelZ | ??? you asked |
| 10:38.58 | WIMPy | IIRC you came here, because you wanted that. |
| 10:39.10 | asterisknooby | wouldnt you look at it as spam hole?(providing there are sip spammers already) |
| 10:39.23 | ChannelZ | No more so than your telephone is I guess |
| 10:39.39 | ChannelZ | I have no idea what the hell you're trying to do |
| 10:39.51 | asterisknooby | correct me if i am wrong |
| 10:40.11 | ChannelZ | about |
| 10:40.24 | asterisknooby | if you have asterisk server, you get some sort of authentication from it, this is why you put username and password |
| 10:40.33 | WIMPy | I have never had a gues sip call, even though I'm on E164.org. |
| 10:40.43 | asterisknooby | right? so noone else can impersonate you |
| 10:40.58 | WIMPy | s/gues/guest/ |
| 10:41.01 | ChannelZ | It depends what you're trying to accomplish |
| 10:41.41 | asterisknooby | i am just trying out SIP technology, more like, trying out asterisk |
| 10:41.56 | asterisknooby | ideally i want to make and recieve calls from other sip servers |
| 10:42.04 | ChannelZ | if you want random people to be able to call you via SIP, you set a context in your [default] section of sip.conf and then put whatever extensions in that context to do something when people call |
| 10:42.07 | asterisknooby | and i want to make sure that noone can impersonate me |
| 10:42.24 | ChannelZ | s/[default]/[general]/ |
| 10:42.51 | ChannelZ | Who are "the other SIP servers"? |
| 10:43.04 | asterisknooby | other sip proxy servers/users |
| 10:43.16 | ChannelZ | in other words "unknown" |
| 10:43.21 | asterisknooby | no |
| 10:43.41 | asterisknooby | there is big difference, between unknown and unknown authenticated |
| 10:44.02 | ChannelZ | Not really. Either you've setup a connection between you and someone on purpose, or you didn't. |
| 10:44.12 | asterisknooby | its like email, you have domain name records set up correctly - u are more less safe |
| 10:44.27 | ChannelZ | what? |
| 10:44.37 | ChannelZ | nevermind. I'm going to bed |
| 10:45.01 | asterisknooby | i am not talking about joe jobs and similar attacks |
| 10:45.30 | ChannelZ | Which has nothing to do with the fact nobody here can call you as you asked. |
| 10:45.44 | asterisknooby | but with correct DNS settings with email you can always distinguish attacks(apart from man in the middle attacks) or trace back |
| 10:46.07 | asterisknooby | if someone calls me, they have to be authenticated, right? |
| 10:46.12 | asterisknooby | not anonymous |
| 10:46.25 | asterisknooby | authenticated by their own sip server |
| 10:46.25 | ChannelZ | Only if you enforce that |
| 10:46.29 | ChannelZ | no |
| 10:46.38 | asterisknooby | its like hidden phone numbers |
| 10:46.49 | asterisknooby | i dont want "hidden numbers" |
| 10:46.51 | ChannelZ | You have no control over what someone else does |
| 10:47.02 | asterisknooby | i do, i dont let them abuse me |
| 10:47.06 | ChannelZ | besides to reject calls if you don't like their caller ID or whatnot |
| 10:47.48 | ChannelZ | I can't call you because you have not setup your asterisk to know who I am and accept calls from me. In that sense I am anonymous. |
| 10:48.30 | asterisknooby | "you have not setup your asterisk to know who I am" - now this is a problem... |
| 10:48.55 | ChannelZ | You have 1 sip peer defined, "203", with a password I don't know. So I'm not sure what you expect to happen. |
| 10:49.34 | ChannelZ | You have "guest" (what I am calling 'anonymous') SIP turned on, set to a context with no extensions in it. So nobody can call you. |
| 10:49.59 | asterisknooby | you want me to create user for you on my sip server? |
| 10:50.08 | ChannelZ | Not really |
| 10:50.08 | asterisknooby | this is what you wanted to do, right? |
| 10:50.48 | ChannelZ | You asked for someone to call you. I tried. It didn't work. I'm just telling you why. |
| 10:51.38 | asterisknooby | did you call me from where? using what service? my own? did you try to get autherisaiton from my own asterisk or you were getting authorised by some other asterisk? |
| 10:52.31 | ChannelZ | I called from my softphone through my asterisk, which really makes no difference |
| 10:53.48 | ChannelZ | You seem to like email, so lets use that analogy again. I tried to send you a piece of mail. My server connected to yours. Yours rejected it. |
| 10:54.13 | asterisknooby | is it because my server declined authentication from yours? |
| 10:54.29 | ChannelZ | no |
| 10:54.51 | asterisknooby | it just rejected for no reason? |
| 10:55.03 | ChannelZ | and if I am Joe Schmoe sending you a piece of email, there is no "authorization" on my part anyway |
| 10:55.11 | asterisknooby | in email -yes |
| 10:55.25 | ChannelZ | It's no different for a SIP call, unless you ONLY want to accept calls from people you've setup the ability to do so with in advance. |
| 10:55.47 | asterisknooby | no i dont |
| 10:55.57 | asterisknooby | i just dont want ot make my sip server "open relay" |
| 10:56.08 | ChannelZ | That's defined by your dialplan |
| 10:56.13 | asterisknooby | so i want to recieve calls from external sip servers |
| 10:56.25 | ChannelZ | You're currently doing that already. |
| 10:56.29 | ChannelZ | Except that the calls have nowhere to go. |
| 10:56.47 | ChannelZ | Because they are going into a context called "home" and there are no extensions in that context to handle the calls. |
| 10:57.27 | asterisknooby | disallow=all - is that it? |
| 10:57.37 | asterisknooby | ou hold on |
| 10:57.39 | ChannelZ | no, that's for codecs |
| 10:58.05 | asterisknooby | ok, i go back and RTFM |
| 10:58.07 | asterisknooby | thanks a lot |
| 10:58.26 | ChannelZ | the fact that you have "allowguest" not specifically set to 'no' means you can accept SIP calls from anyone. |
| 10:59.15 | ChannelZ | In which case they will go into whatever context is set in the [general] section of sip.conf. Any call coming from someone who is not matched by any other peer in your config (in your case, only '203' whatever that is) will wind up there. |
| 11:00.16 | asterisknooby | ok... |
| 11:00.26 | asterisknooby | i think i start to understand it a bit better now |
| 11:00.35 | asterisknooby | thanks |
| 11:00.49 | ChannelZ | good luck |
| 11:11.52 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-135-173.pools.arcor-ip.net) |
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| 11:56.45 | asterisknooby | ChannelZ: are you still awake? |
| 11:58.50 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 11:59.10 | Dovid | asterisknooby: still need a test ? |
| 12:00.34 | asterisknooby | thanks, no, i wanted to ask him a question, since he understood my problem well |
| 12:03.56 | asterisknooby | in sip.conf file, can 2 sections of same name exist with different context's? |
| 12:04.07 | asterisknooby | probably not... |
| 12:14.14 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 12:17.44 | *** part/#asterisk ruben23 (~Innermax@119.92.78.124) |
| 12:20.27 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 12:20.40 | asterisknooby | word dialplan for me is just confusing :( |
| 12:20.51 | asterisknooby | fundamental flows of knowledge... |
| 12:21.41 | asterisknooby | http://www.asterisk.org/glossary and explanation form there isnt helping |
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| 16:00.20 | DelphiWorld | hey all |
| 16:00.22 | DelphiWorld | pabelanger: hello |
| 16:00.28 | IsUp | hey DelphiWorld |
| 16:00.41 | DelphiWorld | IsUp: isn't down? :P |
| 16:00.56 | IsUp | chan_ss7.so: SS7 link down! |
| 16:01.12 | DelphiWorld | hehe |
| 16:04.20 | IsUp | :) |
| 16:06.04 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 16:11.36 | DelphiWorld | anyone have a video phone? |
| 16:12.22 | IsUp | eyeBeam :p |
| 16:13.04 | DelphiWorld | IsUp: :D |
| 16:15.13 | IsUp | it supports video :D |
| 16:15.59 | IsUp | anyways, i gotta go |
| 16:16.03 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k1108.webspeed.dk) |
| 16:16.05 | IsUp | have a great day |
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| 18:59.08 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 19:00.51 | antiwire | hey. is there a method to log registration attempts in asterisk? |
| 19:04.40 | fenrus | isnt that on by default ? |
| 19:04.58 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3967775.cpe.net.cable.rogers.com) |
| 19:05.25 | antiwire | where would it log it to? |
| 19:05.40 | antiwire | I'm talking about producing a log file |
| 19:06.57 | carrar | <PROTECTED> |
| 19:07.15 | fenrus | i have that in /var/log/asterisk/messages too |
| 19:07.17 | *** join/#asterisk sol326 (~studyofli@c-71-56-135-99.hsd1.wa.comcast.net) |
| 19:07.20 | carrar | <PROTECTED> |
| 19:08.44 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 19:09.05 | antiwire | carrar: ah thanks |
| 19:12.49 | carrar | typical registration scans (from China) try approx 21,262 extensions |
| 19:13.30 | antiwire | how did you guess that's why I was asking? ;) |
| 19:13.44 | carrar | Cause I log |
| 19:13.50 | antiwire | hehe |
| 19:14.05 | carrar | Yesterdays Total Hits from China: 30995 |
| 19:14.25 | antiwire | what log do you produce those numbers form? |
| 19:14.27 | antiwire | from |
| 19:14.45 | brainiac | That can't be all Chinese... ppl around the world are using proxy servers there |
| 19:14.47 | carrar | combinations of router data and asterisk logs |
| 19:15.28 | carrar | Thats ALL just IP's registered to China companies |
| 19:16.04 | carrar | top 3 hit yesterday from China Hits/Protocol |
| 19:16.04 | carrar | <PROTECTED> |
| 19:16.05 | carrar | <PROTECTED> |
| 19:16.05 | carrar | <PROTECTED> |
| 19:16.15 | carrar | Hits/IP |
| 19:16.15 | carrar | <PROTECTED> |
| 19:16.15 | carrar | <PROTECTED> |
| 19:16.15 | carrar | <PROTECTED> |
| 19:16.33 | sol326 | can anyone explain if tiff2pdf can be automated to perform when a fax is receved |
| 19:16.39 | sol326 | ?patebin |
| 19:16.44 | sol326 | ?pastebin |
| 19:16.52 | sol326 | ? pastebin |
| 19:16.58 | sol326 | arghh |
| 19:17.35 | carrar | brainiac, keep in mind thats not hits towards just a single IP |
| 19:19.57 | brainiac | I've seen Europeans using Chinese servers (be they hacked, or proxy, or ?) and have used those to port scan and attack American Asterisk and Apache servers |
| 19:20.32 | carrar | more reason to watch out for China IP's :) |
| 19:21.00 | brainiac | I think 111.* may be ripe for a blacklist |
| 19:22.33 | carrar | 111 is assigned all over |
| 19:23.07 | carrar | All over Asia Pacific region |
| 19:23.22 | carrar | but it's very simeple to break it all out |
| 19:23.37 | carrar | and find out what parts of that 111/8 is assigned to what |
| 19:24.27 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:26.26 | carrar | top assignments in that 111/8 are |
| 19:26.27 | carrar | <PROTECTED> |
| 19:26.27 | carrar | <PROTECTED> |
| 19:26.28 | carrar | <PROTECTED> |
| 19:26.28 | carrar | <PROTECTED> |
| 19:26.30 | carrar | <PROTECTED> |
| 19:26.32 | carrar | <PROTECTED> |
| 19:26.52 | carrar | number of assignments that is |
| 19:27.43 | *** join/#asterisk WonTu (~WonTu@p57B569BF.dip.t-dialin.net) |
| 19:27.57 | *** part/#asterisk WonTu (~WonTu@p57B569BF.dip.t-dialin.net) |
| 19:31.41 | sol326 | carrar: you got a problem with pastebin?? |
| 19:32.08 | sol326 | http://pastebin.com/ |
| 19:32.41 | carrar | no one is talking and it's only a couple of lines |
| 19:32.48 | sol326 | no really |
| 19:32.55 | sol326 | please use pastebin |
| 19:32.57 | carrar | Fall off the wrong side of he bed? |
| 19:33.19 | sol326 | others are asking questions as well |
| 19:33.22 | carrar | You paying for IRC lines? |
| 19:33.42 | sol326 | by the time someone sees it's off the screeen because of your 'few' lines |
| 19:33.46 | carrar | Thats funny, not sure what channel you are in |
| 19:33.58 | sol326 | just respect the channel the rest of us do |
| 19:34.03 | sol326 | 10 |
| 19:34.04 | carrar | you too |
| 19:34.07 | sol326 | channels |
| 19:34.11 | carrar | you are wasting irc screen space |
| 19:34.18 | sol326 | LMAO |
| 19:34.24 | sol326 | because you're arguing |
| 19:34.33 | carrar | paste all that in pastebin |
| 19:34.38 | carrar | then send it to me |
| 19:34.54 | sol326 | Riiiiight |
| 19:34.56 | sol326 | smartass |
| 19:35.01 | antiwire | lol |
| 19:35.03 | E-bola | lol |
| 19:35.52 | E-bola | carrar: why would you borther to figure out where botnets are hammering ur sip server from? |
| 19:36.12 | carrar | I'm a stats whore I guess |
| 19:36.22 | carrar | Like to know where things are coming from |
| 19:36.24 | E-bola | its pretty useless |
| 19:36.26 | fenrus | isnt "because you can" the correct answer ? |
| 19:36.30 | carrar | and if when I need to block them |
| 19:36.31 | E-bola | unless u wanna make a banllist based on origin |
| 19:36.43 | carrar | yup |
| 19:36.45 | carrar | that too |
| 19:36.51 | antiwire | I want to make a banlist based on CP intelligence |
| 19:36.58 | fenrus | i usually ban the middle east, most of asia, china from my servers ;) |
| 19:37.15 | E-bola | ICP? |
| 19:37.18 | E-bola | -I |
| 19:37.19 | carrar | here is a exmaple of a scan of asterisk |
| 19:37.20 | carrar | http://www.osburn.com/failed-logins-example.txt |
| 19:37.23 | antiwire | calling party |
| 19:37.40 | E-bola | antiwire: something like if they fail a bunch of times ban them? |
| 19:37.48 | antiwire | haha\ |
| 19:38.00 | antiwire | carrar: which log would contain those fails? |
| 19:38.10 | carrar | There are scripted out there that will add iptables rule based on number of filed attempts |
| 19:38.14 | carrar | scripts |
| 19:38.20 | fenrus | didnt someone just paste a fail2ban link here? |
| 19:38.33 | carrar | antiwire, thats /var/log/asterisk/messages |
| 19:38.44 | fenrus | http://www.voiptoday.org/index.php?option=com_content&view=article&id=555:how-to-protect-your-asterisk-system-from-unwanted-sip-registration-attempts |
| 19:38.47 | antiwire | awesome |
| 19:38.50 | fenrus | could be one way of doing it |
| 19:38.50 | E-bola | considers bringing up the old bruteforce protection integrated in asterisk debate up again |
| 19:38.51 | E-bola | hehe |
| 19:39.11 | fenrus | now diner. ;) |
| 19:43.33 | carrar | I also use all our ISP's unused IP addresses as a honey pot |
| 19:43.41 | sol326 | anyone use tiff2pdf?? |
| 19:46.07 | wdoekes2 | no, but imagemagick convert(1) |
| 19:47.04 | sol326 | can you automate it? |
| 19:47.25 | carrar | crontab! |
| 19:47.45 | sol326 | so anything new coming in database gets converted and sent to email or another folder even... |
| 19:48.11 | carrar | hylafax does all that for you |
| 19:51.14 | sol326 | any of them compatibile with freepbx? |
| 19:51.26 | carrar | hylafax works great with Asterisk |
| 19:51.30 | carrar | so I don't see why not |
| 19:51.44 | carrar | but freepbx is who other beast |
| 19:51.48 | carrar | whole |
| 19:51.59 | sol326 | right |
| 19:52.11 | carrar | might ask in #freepbx |
| 19:52.52 | carrar | http://www.hylafax.org/archive/2004-07/msg00096.php |
| 19:53.04 | carrar | I bet you could make it work in freepbx |
| 19:53.18 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 19:54.07 | markit | hi, I'm updating the italian sound set from 1.4.15 to 1.4.20. There is a new prompt: "conf-roll-callcomplete: Roll call complete.". I don't understand it's meaning... what is a "roll call"? |
| 19:54.52 | carrar | roll call is taking a poll of who is here |
| 19:55.00 | carrar | like in school |
| 19:55.05 | carrar | teachers do a roll call |
| 19:55.13 | carrar | kids hear their name they say "Here" |
| 19:55.20 | markit | carrar: oh, I see, the dictionary was right... how can be used in asterisk? |
| 19:55.32 | *** join/#asterisk Tim_Toady (~moi@62.1.174.14.dsl.dyn.forthnet.gr) |
| 19:55.52 | markit | well, sure in conference |
| 19:55.58 | markit | forgive my stupid question :) |
| 19:56.01 | markit | carrar: thanks a lot |
| 19:56.08 | carrar | I don't off hand where that is used but I would guess in the queues or something |
| 19:56.41 | markit | "Roll call complete" and not "Roll call completed"? |
| 19:58.15 | markit | uh, listening at the sound, is very different than what is in core-sounds file |
| 19:58.16 | carrar | completed is saying it just finished, complete is not time specific as to when it compelted it |
| 19:59.23 | carrar | could be used to mean same thing |
| 19:59.51 | carrar | "Roll call complete" could be used with other stuff |
| 19:59.57 | carrar | "Roll call complete in 5 mins" |
| 19:59.59 | markit | carrar: are you a english speaker? could you please write down the content of that file? |
| 20:00.18 | carrar | hahah can I quote you on that :) |
| 20:00.33 | carrar | I am a horride at spelling |
| 20:00.46 | markit | I mean, understand what she says |
| 20:00.50 | markit | and write it for me |
| 20:00.58 | markit | so I can translate in italian and re-record |
| 20:01.07 | markit | I don't get all what she says |
| 20:01.23 | markit | and is sure different than what is reported in the .txt |
| 20:01.39 | carrar | Email me, my wife is callign me asterisk@osburn.com |
| 20:01.47 | carrar | I can do in later today |
| 20:02.12 | markit | carrar: thanks a lot |
| 20:03.18 | markit | or anyone else can do it now? Just because I would love to record it tonight, that is more silent |
| 20:03.36 | markit | and my wife in the right mood (she is the speaker) |
| 20:11.03 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
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| 21:34.33 | Supari | idont see asterisk 1.8.1 rc2 on the site anyone have a link |
| 21:34.36 | Supari | i only see rc1 |
| 21:36.25 | ChannelZ | 1.8.2-rc2 |
| 21:36.38 | ChannelZ | 1.8.1 was already released |
| 21:36.59 | ChannelZ | http://www.asterisk.org/node/51538 |
| 21:38.13 | Supari | ChannelZ thats rc1 |
| 21:39.12 | Supari | should i use that or do svn check out? |
| 21:39.15 | ChannelZ | sorry my fingers are all over the place |
| 21:39.29 | ChannelZ | it is rc1.. but 1.8.2 is the important bit |
| 21:39.37 | Supari | cool |
| 21:39.47 | Supari | yeah i was looking at summery and they did fix the fax issue |
| 21:40.11 | Supari | would you also know if they fixed this issue ? |
| 21:40.11 | Supari | FD 19239 exceeds the maximum size of ast_fdset! |
| 21:40.27 | Supari | i cant find any fix for this, it happened after upgrading from 1.4 |
| 21:40.41 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 21:41.33 | ChannelZ | no idea |
| 21:42.01 | Supari | strange issue i google it and many ppl have the issue but no oneposted a fix lol |
| 21:42.52 | ChannelZ | www.spinics.net/lists/asterisk/msg134854.html |
| 21:50.00 | Supari | ahh |
| 21:50.01 | Supari | i see |
| 21:50.01 | Supari | coo |
| 21:50.06 | Supari | thanks ChannelZ |
| 21:51.18 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 21:51.45 | citywok | can anybody copy/paste the user agent sent to a web server from a polycom phone other than an IP650? |
| 21:55.47 | Supari | if you have a weserver just look at logs |
| 21:56.40 | *** join/#asterisk boboc (~boboc@86.126.106.121) |
| 22:01.23 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 22:01.34 | z4nD4R | hi |
| 22:01.49 | ChannelZ | hi |
| 22:02.07 | z4nD4R | Q: if i use SRTP and MixMonitor, the call ale recorded as encrypted? |
| 22:03.40 | z4nD4R | because https://asterisk.4safety.cz/1234/MichalSrnecVlavo.1000-1234-20110109-220725.wav.ogg :D |
| 22:12.12 | *** join/#asterisk wesphillips (~wphillips@72-254-162-235.client.stsn.net) |
| 22:13.17 | *** part/#asterisk wesphillips (~wphillips@72-254-162-235.client.stsn.net) |
| 22:15.02 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 22:25.19 | *** join/#asterisk m_tadeu (~quassel@89-180-156-159.net.novis.pt) |
| 22:26.20 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 22:26.43 | niekvlessert | hi! when is a peer 'unreachable' and when is it 'lagged'? |
| 22:27.25 | WIMPy | When the answer didn't come at all and when it came too late. |
| 22:28.49 | niekvlessert | great answer, why didn't google tell me this. Probably because it's logical ;) |
| 22:29.29 | WIMPy | possible |
| 22:30.22 | antiwire | Sometimes I get massive lag when I register over 3G |
| 22:30.26 | antiwire | like 3000ms+ |
| 22:31.54 | niekvlessert | yeah, 3g is not very reliable bandwidth wise |
| 22:33.01 | antiwire | well it's not bandwidth in this case |
| 22:33.04 | antiwire | it's the latency |
| 22:33.51 | antiwire | it's capable of handling the traffic, just slowly |
| 22:34.32 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 22:34.58 | niekvlessert | does it work out for you in terms of good enough speech? |
| 22:35.08 | antiwire | yeah |
| 22:35.10 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.5.201) |
| 22:35.12 | DelphiWorld | hey |
| 22:35.15 | ChannelZ | if you like it sounding like you're talking to someone on the moon |
| 22:35.19 | DelphiWorld | anyone do ratlimit using iptables? |
| 22:35.45 | antiwire | I use linphone on my iPhone to test over wifi and 3G and when I have decent 3G coverage it works great |
| 22:36.18 | ChannelZ | with 3-second latency? I guess 'great' is subjective |
| 22:36.42 | antiwire | Notice I did say "sometimes" |
| 22:36.54 | DelphiWorld | ChannelZ: any iptables rul please? |
| 22:37.08 | ChannelZ | Sorry I don't even know what ratlimit is |
| 22:37.56 | DelphiWorld | ChannelZ: like if someone registered one time bad or something... thay can register only one time in 60 sec or something, just for security |
| 22:38.47 | ChannelZ | that's probably not the way to solve that problem |
| 22:39.26 | DelphiWorld | :P |
| 22:39.28 | ChannelZ | What happens when a legitimate client sends you 5 SIP packets in a short amount of time for a legitimate reason? iptables doesn't know the difference |
| 22:40.48 | ChannelZ | You're probably better off setting up something like fail2ban to block people who fail many registrations |
| 22:41.01 | neurosys_ | Fail2ban rox ;) |
| 22:41.05 | antiwire | DelphiWorld: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk |
| 22:41.13 | *** join/#asterisk wesphillips (~wphillips@72-254-162-235.client.stsn.net) |
| 22:41.33 | DelphiWorld | hehe |
| 22:41.35 | DelphiWorld | thanks |
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| 22:43.34 | *** join/#asterisk trapa (~trapa@76-10-190-121.dsl.teksavvy.com) |
| 22:44.24 | trapa | I have a problem with my conference calls. I'm using Confbridge(2292,csM); and it seems to work, but if more than two people join the conference all audio is lost |
| 22:45.19 | antiwire | Does your system have the resources to deal with more than 3 parties? |
| 22:45.52 | trapa | I believe so ... i'm basing this on the fact that it can easily have 5 or so simultanious calls at once. |
| 22:46.20 | antiwire | ok |
| 22:46.50 | antiwire | can you view verbose logging for rtp and sip? any clues there? |
| 22:47.02 | trapa | How would i do that? |
| 22:47.10 | trapa | has been doing asterisk -rvvvvvvvvvvvvvvvvv |
| 22:47.15 | trapa | but no real clues in that output |
| 22:47.28 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.5.201) |
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| 22:48.08 | antiwire | try entering : sip set debug on |
| 22:48.19 | antiwire | and build up a conference |
| 22:48.34 | antiwire | see if anything obvious shows up |
| 22:49.33 | trapa | okie |
| 22:52.00 | antiwire | also, can you make it happen with any phone that ends up being the third party to join or is it only happening with a specific phone? |
| 22:56.56 | *** join/#asterisk errr_ (~errr@fedora/errr) |
| 22:57.33 | trapa | No it's any phone. |
| 22:57.52 | trapa | I'm just provisioning another phone here at my desk in order to be able to test this with three without bothering other people |
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| 23:05.58 | trapa | I have a problem with my conference calls. I'm using Confbridge(2292,csM); and it seems to work, but if more than two people join the conference all audio is lost |
| 23:06.10 | trapa | Erf .. sorry |
| 23:06.19 | trapa | So yeah .. now i have two desk-phones set up and just tried it again. |
| 23:06.38 | trapa | and the twooriginal people who were in the conference we still connected but the third party coudln't hear anything or say anything |
| 23:22.52 | *** join/#asterisk karim786 (karim@96.51.135.33) |
| 23:23.20 | karim786 | Hi everyone |
| 23:23.55 | karim786 | Has any been able to get Fax T.38 working with Asterisk 1.6.2 and Iristel trunks |
| 23:23.57 | karim786 | ? |
| 23:25.27 | citywok | Supari: thanks, but i only have an IP650 handy. I'm curious about other phones :) |
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| 23:38.04 | a1fa | is there something smaller than freepbx thats potentially run out of sqlite? |
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