| 00:03.49 | mazpe | carrar: http://pastebin.centos.org/38831 ntpq -c works from my computer... |
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| 00:19.11 | mazpe | ok, to the next cdr_odbc |
| 00:24.40 | mazpe | dammit.. i guess i didn't have unixODBC installed when i installed asterisk.. now i have to recomplile. |
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| 00:31.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 03:33.24 | sixohquad | hey guys, i thought i saw around somewhere that Digium had a voice actor that would do music on hold stuff, but i cant seem to find the info now |
| 03:33.31 | sixohquad | can anybody point me in the right direction? |
| 03:35.21 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 03:36.06 | sixohquad | quit |
| 03:36.08 | sixohquad | oops |
| 03:41.34 | *** join/#asterisk sixohquad (~cody@184.65.142.249) |
| 03:41.39 | sixohquad | sorry had to reboot |
| 03:41.44 | sixohquad | still looking for voice actor info |
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| 03:51.32 | sixohquad | sits on the couch |
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| 04:23.34 | sixohquad | quit |
| 04:23.37 | sixohquad | \quit |
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| 04:49.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 05:21.16 | *** join/#asterisk BeeBuu (0ed6f6a5@gateway/web/freenode/ip.14.214.246.165) |
| 05:21.43 | BeeBuu | I got a big problem,any one help me? |
| 05:22.14 | ectospasm | BeeBuu: don't ask to ask, just ask |
| 05:22.16 | jpsharp | Throw out your problem and see if it sticks. |
| 05:22.19 | BeeBuu | The asterisk disconnect my connection |
| 05:22.34 | ectospasm | elaborate |
| 05:22.43 | BeeBuu | i ran 'nc 127.0.0.1 5038' |
| 05:22.52 | BeeBuu | and login |
| 05:23.27 | ectospasm | at what point does the manager interface disconnect you? |
| 05:23.42 | BeeBuu | and get messages,but when i make call over 20,asterisk disconnect me! |
| 05:24.56 | BeeBuu | i had no idea this, very down now~~~ |
| 05:24.57 | ectospasm | what's the last message you get before the disconnect? Is it the same one every time? |
| 05:25.12 | BeeBuu | ectospasm: nothing,it just disconnected |
| 05:25.21 | ectospasm | ...asterisk isn't running? So Asterisk is crashing? Or is it just your manager connection? |
| 05:25.35 | BeeBuu | and asterisk is running~~~ |
| 05:25.53 | BeeBuu | and asterisk just disconnect my connectino |
| 05:26.31 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-12-114.bois.qwest.net) |
| 05:26.35 | ectospasm | do you get the same behavior if you use telnet rather than nc? |
| 05:26.35 | Nugget | telnet is eeeeeeevil! |
| 05:26.49 | ectospasm | ...but for these purposes it should be OK |
| 05:27.01 | ectospasm | ...unless BeeBuu is not on a secure network. |
| 05:27.31 | BeeBuu | telnet? are you sure? |
| 05:28.22 | ectospasm | BeeBuu: is the Asterisk system not on your local network? |
| 05:28.47 | BeeBuu | sure |
| 05:29.28 | ectospasm | so telnet is probably a bad idea |
| 05:29.54 | ectospasm | ...what is the last message you received from the manager interface before it disconnected you? Is it the same every time you get disconnected? |
| 05:29.54 | BeeBuu | when the asterisk disconnect my telnet |
| 05:30.00 | BeeBuu | it get "Connection closed by foreign host. " |
| 05:30.22 | ectospasm | are you also looking at the Asterisk CLI (asterisk -r) at the same time? |
| 05:30.50 | ectospasm | we don't have much details on what is causing the disconnect. You should watch both in separate windows, see if there any clues in the CLI. |
| 05:31.14 | BeeBuu | ectospasm: Oh? where kind INFOs i will get? |
| 05:32.11 | ectospasm | BeeBuu: ...if logger.conf is appropriately configured, you should see Asterisk execute dialplan as you make test calls. |
| 05:33.19 | BeeBuu | and i can get ERROR or WARNING INFO,am i? |
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| 05:36.37 | ectospasm | BeeBuu: possibly |
| 05:37.00 | ectospasm | BeeBuu: you aren't familiar with the Asterisk CLI, and you're trying to use the manager interface? |
| 05:37.08 | ectospasm | ...bizarre. |
| 05:37.41 | BeeBuu | i can't get any reson that disconnected my connection |
| 05:38.19 | ectospasm | do you see anything like manager interface disconnected? What is your verbosity set to? |
| 05:38.30 | ectospasm | try "core set verbose 10" and try again. |
| 05:38.36 | ectospasm | ..in the Asterisk CLI |
| 05:39.36 | BeeBuu | let me try that. |
| 05:40.31 | BeeBuu | i want to know, when set verbose to 10, and the log file can record more infomation ,does it? |
| 05:40.56 | ectospasm | no. You'll need to enable the full log in logger.conf, then "logger reload" |
| 05:42.30 | BeeBuu | full => notice,warning,error,debug,verbose |
| 05:42.38 | BeeBuu | is it correct? |
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| 05:49.40 | raden | if i have 2 phones that support g.729 will it use a license when i call between them > |
| 05:53.10 | ectospasm | raden: http://kb.digium.com/articles/FAQ/G-729-explanation-of-encoders-decoders?retURL=%2Fapex%2FknowledgeHome&popup=false |
| 05:53.37 | ectospasm | raden: it will only use an encoder/decoder if there is any transcoding |
| 05:53.44 | ectospasm | if you record the call, it will be transcoded |
| 05:54.12 | ectospasm | if you bridge another channel (call 'leg') that doesn't use G.729, you'll use one encoder/decoder. |
| 05:54.29 | ectospasm | BeeBuu: sorry, stepped away... that is correct |
| 05:54.41 | coppice | if you record the call you will use 2 licences |
| 05:54.59 | ectospasm | coppice: no, just one. |
| 05:55.17 | ectospasm | ...if both endpoints use G.729, there's no transcoding for that channel. |
| 05:55.25 | ectospasm | ...called "passthrough" |
| 05:55.35 | coppice | only if you just record one side of the call |
| 05:57.05 | ectospasm | hmmm...that's not how I understand it |
| 05:59.06 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.214) |
| 05:59.14 | ectospasm | ...since the RTP stream has both directions in it. Also this assumes no directmedia between the endpoints. |
| 06:00.21 | ectospasm | ...but then again, it's rather late, and I've had a fair amount to drink. I'll have to think about this later. |
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| 06:19.23 | BeeBuu | ectospasm: no problem. |
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| 06:19.56 | BeeBuu | ectospasm: but it still disconnected my connection |
| 06:21.24 | ectospasm | BeeBuu: you still haven't said what the final AMI message is |
| 06:21.32 | ectospasm | ...if it's the same, or different |
| 06:22.36 | BeeBuu | same |
| 06:22.54 | ectospasm | ...so what is the message? |
| 06:23.00 | ectospasm | pastebin it if you would. |
| 06:23.07 | BeeBuu | and i get some msg like this: "[2012-05-12 14:22:52] NOTICE[3124]: chan_sip.c:26193 sip_poke_noanswer: Peer '868' is now UNREACHABLE! Last qualify: 0 " |
| 06:23.39 | ectospasm | BeeBuu: that just means peer 868 has become unreachable for whatever reason, that shouldn't kick you off of AMI |
| 06:24.49 | BeeBuu | ectospasm: i don't know what's the thing i'm looking for that |
| 06:25.27 | ectospasm | BeeBuu: are you capturing the AMI log to a file? |
| 06:26.16 | BeeBuu | ectospasm: how? and i just know the only way is nc connect |
| 06:28.51 | ectospasm | BeeBuu: try "nc localhost 5038 | tee ami.log" |
| 06:29.11 | BeeBuu | ok,let me try it |
| 06:33.58 | BeeBuu | disconnect again |
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| 06:34.24 | BeeBuu | what can i do now? |
| 06:38.01 | ectospasm | Go to pastebin.com, and post the ami.log file |
| 06:41.02 | BeeBuu | and i think the ami is there any connection limited? |
| 06:41.35 | BeeBuu | my another programm connect to ami for make call |
| 06:41.38 | ectospasm | not that I know of... but there is a limit to the number of messages in the buffer. |
| 06:41.57 | ectospasm | ...you have to periodically clear the buffer or else there won't be any more messages. |
| 06:41.58 | BeeBuu | oh? what's it? |
| 06:42.09 | ectospasm | I don't know, I don't deal with AMI very much. |
| 06:42.21 | ectospasm | probably look in the AMI section of the book |
| 06:42.24 | ectospasm | ~thebook |
| 06:42.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 06:51.46 | BeeBuu | ~pastbin |
| 06:52.49 | BeeBuu | ectospasm: the ami.log file size is 931367 |
| 06:54.14 | ectospasm | ~pastebin |
| 06:54.14 | 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. |
| 06:55.14 | ectospasm | BeeBuu: that's OK, that's only roughly 900K. Should be able to pastebin that |
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| 07:16.34 | BeeBuu | ectospasm: pastebin can't over 500 kilobytes |
| 07:17.48 | BeeBuu | and i paste my manager setting in http://pastebin.com/n0ttgdfS |
| 07:18.14 | BeeBuu | and what's the "block sockets" mean? |
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| 07:27.36 | BeeBuu | anyone help me please. |
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| 07:52.10 | ectospasm | BeeBuu: so paste the last 100 lines of the ami.log. |
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| 11:06.55 | joshauh | hello, a problem i am having is, i have registered sip and two extensions.. problem is, when i call the DID it comes up with busy tone |
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| 11:29.27 | mazpe | anything in particular causes: cdr_odbc.c:148 odbc_log: Unable to retrieve database handle. CDR failed." trying to get cdr_odbc to work |
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| 15:35.06 | saxa | hi, is there some how-to about BLF configurations in * ? |
| 15:35.27 | saxa | I have a grandstream gxp2120 but it seems I cant make it work |
| 15:35.45 | saxa | i see the fast dial buttons green all the time |
| 15:36.06 | saxa | I have put the hint option in extensions.conf |
| 15:36.41 | saxa | but do I need another context for this or can I put this hint option directly in the sip phone extension ? |
| 15:37.42 | saxa | http://pastebin.com/W88xh0GJ |
| 15:37.52 | saxa | is this ok ? |
| 15:38.36 | [TK]D-Fender | call-counter=yes |
| 15:38.41 | [TK]D-Fender | sfor your phones |
| 15:42.00 | saxa | hi [TK]D-Fender ,i have callcounter=yes in general |
| 15:42.06 | saxa | but is without a - |
| 15:43.59 | saxa | i found that guide http://www.grandstream.com/products/gxp_series/gxp2000/documents/gxp2000_interop_asterisk_blf.pdf |
| 15:44.08 | saxa | but here it sets up some groups |
| 15:44.47 | saxa | and is not clear if its needed to add this sip extensions with hint option in extensions.conf |
| 15:47.16 | [TK]D-Fender | "core show hints" |
| 15:47.22 | [TK]D-Fender | See if it even subscribes |
| 15:49.27 | saxa | http://pastebin.com/v7E0msBX |
| 15:49.31 | saxa | seems it doesnt |
| 15:49.47 | saxa | but I had before the watchers numbers not 0 |
| 15:49.56 | saxa | but stil unsubscribed |
| 15:50.18 | saxa | sorry, state:Unavailable |
| 15:50.25 | saxa | not unsubscribed |
| 15:51.10 | saxa | like here http://pastebin.com/RLpEuQZW |
| 15:54.40 | saxa | I used this lik to configure |
| 15:54.44 | saxa | http://www.expressinterconnect.com/497/small-business-voip-enable-asterisk-blf-busy-lamp-field-in-yealink-and-grandstream-ip-phones/ |
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| 16:48.36 | mazpe | any ideas on how to create a dial plan that will force only a certain group of extensions to Authenticate before they dial out. |
| 16:49.38 | [TK]D-Fender | mazpe: you can check if they've registered first |
| 16:49.47 | [TK]D-Fender | Technically ALL calls "auth" |
| 16:49.50 | [TK]D-Fender | by default |
| 16:50.03 | mazpe | I mean force them to use Authenticate() |
| 16:50.22 | mazpe | right now i have exten => _NXXNXXXXXX,n,Authenticate(/etc/asterisk/conf.d/codes,a) |
| 16:50.25 | [TK]D-Fender | Well.. it's YOUR dialplan.. ysure |
| 16:50.43 | mazpe | but i want ONLY like 10 extensions. |
| 16:51.04 | [TK]D-Fender | Again, it's yur dialplan... go for it... |
| 16:51.10 | mazpe | something like 1001 => _NXXNXXXXXX,n,Authenticate() |
| 16:51.12 | [TK]D-Fender | 10 "devices" <- |
| 16:51.27 | [TK]D-Fender | And that last sample line isn't right |
| 16:51.49 | mazpe | i know… i was wishing i could do something like that. |
| 16:52.03 | [TK]D-Fender | exten => _NXXNXXXXXX/1001,n,Authenticate() <-- you don't shove 1001 there.... |
| 16:52.10 | [TK]D-Fender | like so |
| 16:52.19 | [TK]D-Fender | actually.. you'd do this on the FIRST pririty... |
| 16:52.23 | [TK]D-Fender | something like that. |
| 16:53.11 | mazpe | reaaaaaallly… <evil grin> |
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| 17:04.21 | itbroke | I mentioned previously I was having trouble with inbound calls on a cisco 7960. p3n suggested checking the dial plan. I made a simple test and I'm still having problems with inbound calls on the phone. i'm seeing the invite but i'm getting a 401 unauth in response. On other occasions the phone acks the invite but the call fails with the same message. Asterisk console reports congested busy cause 20 unknown. The phone is register fine and can make outbound |
| 17:04.55 | itbroke | i'm seeing the variedresponse when modifying some settings related to nat on the phone and sip.conf |
| 17:06.50 | [TK]D-Fender | back shortly... |
| 17:08.15 | carrar | OK!!! |
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| 19:02.44 | qakhan | i m using agent callback login on 1.4.38 |
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| 20:06.09 | Ice_Strike | hi |
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| 20:32.31 | *** join/#asterisk denaltroy (~denaltroy@bas18-montreal02-1279348813.dsl.bell.ca) |
| 20:37.36 | denaltroy | I feel bad |
| 20:38.17 | denaltroy | I don't know much (read: shit) about VoIP, I have not RTFM but still I would like to ask a question... :) |
| 20:41.54 | [TK]D-Fender | then ask |
| 20:43.38 | denaltroy | First scenario: I have a POTS line. I would like to have Asterisk act as an IVR that gives the following choice: 1- leave a message, 2- forward call to cellphone, with each option giving the choice of who you which to reach (me or my girl) |
| 20:43.53 | denaltroy | Second scenario: same thing, but with TechSavvy VoIP line... |
| 20:44.46 | [TK]D-Fender | Either is fine |
| 20:44.56 | denaltroy | oh and the "leave a message" would actually send an email with an attached mp3 of the voice message |
| 20:45.03 | [TK]D-Fender | However on POTS... that explains how the call comes IN... doesn't assume what youll send it OUT on... |
| 20:45.35 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 20:46.04 | denaltroy | I remember doing something very similar in 2000-2001 with some kind of weird linux scripting that I had just customised to my likings |
| 20:46.21 | Ice_Strike | why this dont work |
| 20:46.22 | Ice_Strike | exten => 123,1,ExtenSpy(_07XXXXXXXXX@test,w) |
| 20:46.36 | [TK]D-Fender | Well you've got 1 POTS line.... that still only says how it comes in in that first scenario. How you go out is another matter. |
| 20:46.43 | Ice_Strike | I want to spy exten => _07XXXXXXXXX |
| 20:46.46 | [TK]D-Fender | Ice_Strike: _ = no good |
| 20:46.53 | Ice_Strike | ohh yea |
| 20:46.55 | Ice_Strike | thx |
| 20:47.00 | [TK]D-Fender | AND you can't put a PATTERN there |
| 20:47.06 | Ice_Strike | oh :/ |
| 20:47.22 | Ice_Strike | so what is the solution? |
| 20:47.32 | [TK]D-Fender | depends on the situation. |
| 20:48.20 | Ice_Strike | Well when _07XXXXXXXXX is on call |
| 20:48.21 | denaltroy | [TK]D-Fender: Either is fine meaning, I could use Asterisk on both scenarios? |
| 20:48.28 | Ice_Strike | I want to spy that exten |
| 20:48.56 | [TK]D-Fender | delOf course... but 1 analog line typically doesn't let you throw that call back out.. |
| 20:49.29 | denaltroy | [TK]D-Fender: I'm still not ready to actually do (configure) anything as Asterisk is not even installed but just wanted to know if it was at all possible to do and if Asterisk was the right tool |
| 20:49.50 | [TK]D-Fender | denaltroy: * can do whatver you want based on the things you have connect it with |
| 20:50.13 | [TK]D-Fender | denaltroy: Analog lines have their capabilities, VoIP-based providers offer their own services, etc |
| 20:55.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:56.16 | denaltroy | Just to make sure I understand I'll break it down... |
| 20:57.54 | denaltroy | Phone rings 3 times. Nobody home. Asterisk take the call and say, with a "Cave Johnson" like voice that nobody's home |
| 20:58.44 | WIMPy | The hard part will be finding the voice. |
| 20:59.03 | denaltroy | Offers to press one to leave a message. When the persons press one, its makes a proverbial "beep" and then Asterisk records the message in MP3 format and mail it to me (using my DSL internet connection) |
| 20:59.39 | denaltroy | WIMPy: I can be a good voice actor.. ;) |
| 21:00.18 | denaltroy | So, so far so good, Asterisk is the tool for the job...? |
| 21:00.39 | [TK]D-Fender | sure |
| 21:00.41 | denaltroy | ok |
| 21:01.24 | [TK]D-Fender | I phone in and make COFFEE.... your request looks kinda banal in comparison ;) |
| 21:02.58 | denaltroy | Then if Asterisk offer to press two to transfer the phone call to a cellphone, that's where I'm not sure... I'd need a second POTS line? |
| 21:03.20 | [TK]D-Fender | "or something" |
| 21:03.33 | [TK]D-Fender | your telco MIGHT offer a transfer feature. |
| 21:03.38 | [TK]D-Fender | See if they do. |
| 21:03.56 | [TK]D-Fender | Remember that you line has the features... * doesn't invent new ones on it that aren't offered |
| 21:03.58 | denaltroy | I know I sound like a complete idiot but I just want to be sure before I start setting up a server, digging the doc to only find out it just won't work... ;) |
| 21:04.49 | WIMPy | That's the doc of your line, not the doc of Asterisk. |
| 21:05.11 | denaltroy | yeah ok I get it... |
| 21:05.42 | denaltroy | so Asterisk would simply "trigger" the call forward feature of the POTS line and use it with my provided cell #... |
| 21:06.08 | [TK]D-Fender | If you have such a feature |
| 21:06.34 | [TK]D-Fender | I have seen this once before. Rare but possible |
| 21:06.34 | WIMPy | Or you use something else |
| 21:06.39 | WIMPy | . |
| 21:07.10 | denaltroy | and if I don't could * use some kind of Google Talk/Voice service? |
| 21:07.54 | [TK]D-Fender | or whatevr |
| 21:07.58 | denaltroy | I know it probably would not be so simple but would it be just possible? |
| 21:08.07 | [TK]D-Fender | *'s support of GV is spotty. |
| 21:08.16 | [TK]D-Fender | I'd receommend a more traditional supplier |
| 21:10.02 | denaltroy | a traditional supplier of... Voip? |
| 21:10.14 | WIMPy | ~itsp |
| 21:10.14 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:11.00 | denaltroy | ~itsplist-ca |
| 21:11.01 | infobot | it has been said that itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 21:14.06 | denaltroy | is off reading a bit |
| 21:43.59 | *** join/#asterisk qakhan (~qakhan@202.142.158.93) |
| 21:44.33 | qakhan | i have TDM800p 8 port card, it doesnt hangup calls. |
| 21:45.01 | qakhan | i enabled busydetecta and busycount but issue didnt resolve |
| 21:45.25 | Ice_Strike | Anyone know good VOIP provider that allow any callerID without validation? |
| 21:45.35 | Ice_Strike | Apart from voip.ms |
| 21:53.03 | WIMPy | qakhan: Are you still on that one? |
| 22:02.46 | *** join/#asterisk qakhan (~qakhan@202.142.158.93) |
| 22:02.51 | qakhan | WIMPy yes |
| 22:03.02 | qakhan | i m stuck on call hangup |
| 22:04.24 | WIMPy | So you still have hope? |
| 22:12.54 | *** join/#asterisk qakhan (~qakhan@WimaxUser36150-61.wateen.net) |
| 22:13.59 | qakhan | WIMPy i m stuck with call hangup |
| 22:14.16 | WIMPy | So you still have hope? |
| 22:14.56 | denaltroy | bangs his head on the wall until it bleeds |
| 22:15.31 | denaltroy | Thanks WIMPy for the infobot introduction, thanks [TK]D-Fender for your answers |
| 22:19.13 | denaltroy | I'll go and try to sort everything out and come back when I understand a bit better |
| 22:19.23 | denaltroy | thanks again and see you soon! |
| 22:19.33 | *** part/#asterisk denaltroy (~denaltroy@bas18-montreal02-1279348813.dsl.bell.ca) |
| 22:28.33 | *** join/#asterisk qakhan (~qakhan@202.142.158.93) |
| 22:28.43 | qakhan | WIMPy any idea? |
| 22:29.17 | WIMPy | You know what they say about hope: |
| 22:29.23 | WIMPy | Hope encourages patience, but impairs the ability to judge. |
| 22:29.30 | WIMPy | And you need internet, BTW. |
| 22:32.44 | qakhan | sir i did alot of search on internet |
| 22:32.58 | qakhan | what is disconnect supervision |
| 22:34.21 | WIMPy | That's from the telco to you. |
| 22:39.19 | WIMPy | Or at least I have only seen it mentioned in that direction. But I'm not much into analog stuff. |
| 22:42.11 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 22:45.26 | qakhan | WIMPy my telco said they dont have any option name disconnect supervision |
| 22:51.55 | *** join/#asterisk mazpe (~lesterm@c-174-61-72-8.hsd1.fl.comcast.net) |
| 22:56.14 | WIMPy | Wasn't your issus that you couldn;t hangup? |
| 22:56.40 | WIMPy | Or is it only about detecting hangup now? |
| 23:17.58 | *** join/#asterisk wswartz (~wswartzen@2001:470:1f05:1698::7000) |
| 23:18.53 | wswartz | Whenever I dial an outside address (wbdemo@conf.zipdx.com) Asterisk says the call is rejected because the extension isn't found in the default context. |
| 23:19.26 | *** part/#asterisk mazpe (~lesterm@c-174-61-72-8.hsd1.fl.comcast.net) |
| 23:20.35 | qakhan | WIMPy when i hanup call. my dahdi channel get hangup but caller phone shows call in active and my line get busy till caller hanup from his phone |
| 23:21.47 | WIMPy | I'd say your card is either faulty or not compatible to your line. |
| 23:22.59 | qakhan | but it works with 1 operator out of 5. i changed the card but same issue |
| 23:57.12 | wswartz | I completely hate NAT and want it to die. |
| 23:58.20 | WIMPy | IPv6 |
| 23:58.49 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:59.54 | wswartz | Oh sure, my server has IPv6 all setup and functioning. |