| 00:13.28 | *** join/#asterisk j4m3s_ (~j4m3s_@cus-38-127-221-218.ip.hiqdata.net) |
| 00:14.09 | j4m3s_ | hello, anyone using switchvox on a xen guest with a digium card and pci passthru? |
| 00:19.25 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:19.48 | ruben23 | hi guys how to add custom dial plan to do recordings on asterisk on every call..? |
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| 00:39.47 | *** join/#asterisk ledoktre (~chatzilla@omni-02-181.cable.netins.net) |
| 00:42.30 | ledoktre | hey, can someone show me whats wrong with this little dialplan snippet : http://pastebin.com/C3W6C5QU Trying to have a 'toggle switch' type extension |
| 00:59.30 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 01:10.00 | sarthor | Hi, i have a rate plan given by provider, size is 1.6 mb, while my voip-server accept 1000 kb, solution please? |
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| 01:32.29 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:36.41 | WIMPy | ledoktre: incorrect parameter for GotoIf. You can only go to priorities. |
| 01:57.54 | leifmadsen | WIMPy: huh? you can go to context,extension,priority |
| 01:58.25 | leifmadsen | ledoktre: I suspect the round braces in your GotoIf() are a problem) |
| 01:59.14 | leifmadsen | GotoIf($[ "${GLOBAL(${ARG1}-switch)}" = "0"]?true_statement:false_statement) |
| 01:59.19 | leifmadsen | ledoktre: ^^^ |
| 02:23.03 | WIMPy | 'core show application GotoIf' says lable, nothing else. |
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| 03:12.08 | *** join/#asterisk prjktdtnt (~asterisk@c-71-195-202-114.hsd1.ut.comcast.net) |
| 03:13.29 | prjktdtnt | I'm having an interesting issue that I can't seem to find the solution for as much as I try. The asterisk server is setup with two extensions, no outside phone calls just an internal network. When I call one phone, wait for it to begin to ring, then hang up from the originating phone before answering the other extension continues to ring until it is picked up and hung up. |
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| 03:24.32 | ledoktre | leifmadsen - going to try your syntax now... |
| 03:24.53 | *** part/#asterisk prjktdtnt (~asterisk@c-71-195-202-114.hsd1.ut.comcast.net) |
| 03:27.48 | ledoktre | leifmadsen - having similar results as i did before. it just goes one way. wont enable, just says disabled. says this either way : Executing [s@macro-mobileswitch:1] GotoIf("SIP/EXT21-000001f5", "0?switchon,1:switchoff,1") in new stack |
| 03:34.41 | ledoktre | leifmadsen - i think part of my problem is that its not setting the variable properly.... |
| 03:46.14 | p3nguin | wimpy: But context,extension,priority still works. |
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| 04:13.28 | jmordica | hello all. well i tried the RV082 out for qos today and didn't have much success.. |
| 04:13.37 | jmordica | what do you guys use for qos? any suggestions? |
| 04:19.15 | *** join/#asterisk reber (~reber@tsm83-4-78-232-65-13.fbx.proxad.net) |
| 04:51.09 | sarthor | How can i get voip dialer on my own name if I want to start voip calling business? |
| 04:54.08 | [TK]D-Fender | What is "voip dialer" and what is this about your name? |
| 04:54.40 | sarthor | [TK]D-Fender: here is one http://downloads.itellvoip.com/softphones/iTellVoip.msi |
| 04:56.58 | [TK]D-Fender | satGo get a programmer to write one for you or keep seaching for one that offers you rebrandings and reskinning options |
| 04:57.22 | [TK]D-Fender | Or live with having the software not have your name on it |
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| 06:10.35 | ziz212 | Dear all, Can we register external service providers sip account with asterisk and route all system created sip accounts through this connection?. I have searched this in google and didnt get proper information. Pls help me. |
| 06:11.34 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:11.37 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 06:13.55 | WIMPy | ziz212: Off course you can. But if you've got questions at this level, I'd recommend the |
| 06:13.58 | WIMPy | ~book |
| 06:13.58 | 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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| 06:17.04 | ziz212 | ok thanks. Cant you give any hint on this. Do I need to use sip.conf to register external sip account? Or any other way? |
| 06:18.30 | WIMPy | Sure. Everything sip goes in to sip.conf. |
| 06:18.55 | WIMPy | The rest is your dialplan. |
| 06:19.50 | ziz212 | ok thanks thats good. |
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| 08:55.31 | bulkorok | hi |
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| 09:11.16 | FLeiXiuS | SIP/flowroute-00000003 is circuit-busy |
| 09:11.20 | FLeiXiuS | I think that's a lie. |
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| 10:10.21 | jacobw | hi, i'm testing how my pbx puts calls on hold, my sip phone issues a reinvite to the pbx with sdp a=sendonly |
| 10:11.50 | jacobw | my pbx trunk to another sip pbx and doesn't use issue reinvites, it just puts music into the rtp stream without informing the remote pbx |
| 10:12.19 | *** join/#asterisk oej (~olle@edifici-pub-67.adam.es) |
| 10:12.46 | jacobw | i want my pbx to send a=sendonly to the remote pbx as it receives a=sendonly from the sip hone |
| 10:16.49 | kaldemar | well, asterisk is not a proxy. |
| 10:18.38 | *** join/#asterisk Ahmad_Khokhar (~chatzilla@116.71.179.59) |
| 10:19.31 | *** join/#asterisk iulhk (~iulhk@116.71.183.190) |
| 10:19.54 | Ahmad_Khokhar | Hi everyone, I just purchased a TMD1600P with 8FXO ports, I have AsteirskNow running with FreePBX, Having problems with incoming or outgoing |
| 10:20.25 | Ahmad_Khokhar | I have created basic trunk with g0 as well as inbound and outbound |
| 10:20.40 | Ahmad_Khokhar | Can not connect to the PSTN it seems, because no incoming or outgoing |
| 10:21.02 | kaldemar | ~freepbx |
| 10:21.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 10:21.16 | iulhk | is there any option of PTT (PUSH TO TALK) in asterisk ? |
| 10:21.46 | Ahmad_Khokhar | Kaldemar >> how about the basic asterisknow installation with a different GUI other than freepbx |
| 10:22.00 | kaldemar | Ahmad_Khokhar: #asterisk-gui |
| 10:22.27 | Ahmad_Khokhar | ok , does the Asterisknow DVD.iso have that builtin? |
| 10:22.28 | kaldemar | Ahmad_Khokhar: or perhaps #asterisknow would be a more active place, not quite sure. |
| 10:22.36 | Ahmad_Khokhar | oh great |
| 10:22.38 | Ahmad_Khokhar | thanks |
| 10:28.06 | jacobw | kaldemar: yes, but shouldn't it be able to do rfc3264 hold methods? |
| 10:28.24 | *** part/#asterisk Ahmad_Khokhar (~chatzilla@116.71.179.59) |
| 10:33.59 | kaldemar | jacobw: and that's what it does. between the phone and asterisk. |
| 10:35.07 | kaldemar | jacobw: but since asterisk is not a proxy, it won't deliver the a=sendonly to the other endpoint (the other pbx), but sends it hold music. |
| 10:36.58 | jacobw | kaldemar: i see |
| 10:52.20 | jacobw | kaldemar: i'm going to try using 'outboundproxy' to produce this behavoir |
| 11:08.57 | *** join/#asterisk rjvvliet (~rjvvliet@2001:610:600:86a5:1540:2a24:c9bf:d586) |
| 11:09.58 | *** join/#asterisk Rolandow (~rolandow@92.68.81.83) |
| 11:10.31 | Rolandow | hi .. could somebody tell me where the number of digits for attendend transfer is defined? |
| 11:10.42 | Rolandow | when i do an attended transfer through asterisk, only the first three digits are read |
| 11:10.54 | Rolandow | even if i set the timeout to 3 seconds (which should be default anyways) |
| 11:11.08 | rjvvliet | Rolandow: features.conf |
| 11:11.33 | Rolandow | rjvvliet: do you know what config option? i checked features.conf already |
| 11:12.19 | rjvvliet | Rolandow: oeps, sorry though of the timeout option.... number of digits is only DTMF (and enough speed while typing) |
| 11:13.15 | rjvvliet | Rolandow: i though you meant transferdigittimeout= |
| 11:13.39 | Rolandow | rjvvliet: no .. i tried setting that.. but when I do *2, and then type my extension .. it just doesn't ready beyond 3 digits |
| 11:14.26 | rjvvliet | Rolandow: take a look at a verbose log end see what happens, i have the feeling that its the detection. |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF begin '9' received on SIP/Roland_od_Akker-000034b9 |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF begin ignored '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF end '9' received on SIP/Roland_od_Akker-000034b9, duration 150 ms |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF end passthrough '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF begin '9' received on SIP/Roland_od_Akker-000034b9 |
| 11:16.42 | Rolandow | [Jun 15 13:15:35] DTMF[20085] channel.c: DTMF begin ignored '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.43 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF end '9' received on SIP/Roland_od_Akker-000034b9, duration 150 ms |
| 11:16.43 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF end passthrough '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.44 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF begin '9' received on SIP/Roland_od_Akker-000034b9 |
| 11:16.44 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF begin ignored '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.45 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF end '9' received on SIP/Roland_od_Akker-000034b9, duration 150 ms |
| 11:16.45 | Rolandow | [Jun 15 13:15:36] DTMF[20085] channel.c: DTMF end passthrough '9' on SIP/Roland_od_Akker-000034b9 |
| 11:16.46 | Rolandow | [Jun 15 13:15:36] VERBOSE[20086] pbx.c: [Jun 15 13:15:36] -- Executing [999@StumpelZwaag:1] Verbose("Local/999@StumpelZwaag-989b;2", "3,Incoming call for ext 999@StumpelZwaag from start") in new stack |
| 11:16.55 | rjvvliet | Rolandow: eeks... |
| 11:16.55 | Rolandow | only three are passing through |
| 11:16.58 | Rolandow | but why? |
| 11:17.02 | rjvvliet | ~PB |
| 11:17.03 | 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. |
| 11:17.17 | Rolandow | ah sorry |
| 11:17.22 | Rolandow | are you dutch btw? |
| 11:17.30 | rjvvliet | Rolandow: Yep.... |
| 11:17.37 | Rolandow | do you mind if i msg you? :) |
| 11:18.09 | kaldemar | Rolandow: what do you have in [StumpelZwaag] in extensions.conf? |
| 11:19.01 | Rolandow | hehe.. well .. kinda complicated maybe.. i have includes |
| 11:19.05 | Rolandow | StumpelLocal for all extensions |
| 11:19.16 | Rolandow | StumpelExternal for global externals (to route our 'external' numbers internally) |
| 11:19.21 | Rolandow | and StumpelZwaagExternal |
| 11:19.24 | rjvvliet | Rolandow: you mean private chat (sorry still new to IRC) , yea sure |
| 11:19.52 | Rolandow | now StumpelLocal has a template for extensions |
| 11:19.58 | Rolandow | pattern i mean |
| 11:19.59 | kaldemar | your dialplan probably matches 3 digits immediately. |
| 11:20.23 | Rolandow | no because 999 doesn't exist |
| 11:21.49 | *** join/#asterisk aberrios (~aberrios@host81-136-157-198.in-addr.btopenworld.com) |
| 11:22.32 | Rolandow | maybe it's because i am using GoSub .. but that should return to my originating context right? |
| 11:22.33 | kaldemar | 999 is not the only thing that would match 999. |
| 11:22.42 | Rolandow | hm.. true |
| 11:23.01 | Rolandow | but when i dial this extension without transfering, but directly, then it works. |
| 11:23.26 | kaldemar | Verbose("3,Incoming call for ext 999@StumpelZwaag from start") is what gets executed. |
| 11:23.45 | Rolandow | yes, exactly |
| 11:24.00 | Rolandow | that is my dialExten subroutine |
| 11:24.40 | Rolandow | exten => _ZXX!,hint,${SIP_BYEXT(${EXTEN})} |
| 11:24.48 | Rolandow | that is my pattern .. that normally matches 9999 too |
| 11:25.06 | Rolandow | but now when i enter 9999 after the transfer button, it just immediately starts transferring after reading first three digits |
| 11:26.28 | kaldemar | because of ! |
| 11:26.55 | Rolandow | yes.. the ! says there could be more digits |
| 11:27.00 | Rolandow | but not necessarily .. right? |
| 11:27.08 | Rolandow | so that would match 999 or 9999 |
| 11:27.31 | Rolandow | if i dial an extension directly, this pattern works fine. |
| 11:27.51 | kaldemar | when you're dialing overlap, it matches 999 immediately. |
| 11:28.18 | kaldemar | when you're not dialing via transfer, the number gets sent all at once. |
| 11:28.48 | Rolandow | so the ! overlaps? |
| 11:28.49 | kaldemar | maybe you'd want . instead of ! and tune digit timeout. |
| 11:29.20 | kaldemar | when you dial a number and the number gets sent one digit at a time, it's called "overlap dialing". |
| 11:29.31 | Rolandow | but . matches one or more right? |
| 11:29.31 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.202) |
| 11:29.38 | kaldemar | yes |
| 11:29.49 | Rolandow | so that wouldn't match .. for example 201 |
| 11:30.24 | Rolandow | hm .. i think you have a good point though .. |
| 11:30.39 | Rolandow | we use 9<xxx> for connection to other locations .. (stores, actually) |
| 11:30.55 | Rolandow | maybe i should create a pattern for XXX and 9XXX |
| 11:31.16 | Rolandow | so i guess that would be |
| 11:31.23 | Rolandow | _[1-8]XX |
| 11:31.29 | Rolandow | and _9XXXX right? |
| 11:31.32 | Rolandow | then i won't have overlap |
| 11:33.34 | Dovid | hi all |
| 11:34.23 | Dovid | instead of doing Exten => 1800XXXXXXX and then 1888XXXXXXX is there any way to do 18[00,88,77,66,55]XXXXXXX ? |
| 11:35.54 | Rolandow | Dovid: just guessing, but have you tried | yet? |
| 11:36.40 | Rolandow | [00|88|77|66|55] .. don't know if it works but maybe you can try :) |
| 11:38.47 | kaldemar | does not work. |
| 11:39.09 | Rolandow | no OR stuff in patterns? :) |
| 11:39.36 | kaldemar | only for single digits or characters. [12] would mean 1 or 2. |
| 11:39.44 | aberrios | anyone able to help out with a Sangoma problem? Support over there is currently shut. Trying to get a port to send the clock for PRI by setting it to MASTER but it doesnt seem to be doing this and the port remains in RED alarm. |
| 11:39.52 | aberrios | anyone able to help out with a Sangoma problem? Support over there is currently shut. Trying to get a port to send the clock for PRI by setting it to MASTER but it doesnt seem to be doing this and the port remains in RED alarm. |
| 11:40.00 | kaldemar | whatever is inside square brackets matches a single digit only. |
| 11:40.04 | aberrios | oops |
| 11:40.13 | aberrios | stupid window. |
| 11:45.35 | Rolandow | kaldemar: you were right.. i changed my pattern, now it works |
| 11:48.57 | Dovid | aberrios: Where did you set it to master? |
| 11:49.31 | Dovid | aberrios: I assume you want asterisk to be the master. maybe you also need a cross over cable? what kind of system are you connecting to? |
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| 11:52.56 | aberrios | Dovid an NEC. Cross over eh... didnt think of that? Set it to master in wanpipe confs |
| 11:53.23 | aberrios | nah cant be a cross over, the ISDN tester works fine with the NEC without using a cross over cable |
| 11:54.09 | aberrios | unless the tester changes its pin outs |
| 11:54.14 | aberrios | worth a try i guess |
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| 11:55.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 12:07.36 | aberrios | Dovid i could kiss you..... :S |
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| 12:11.07 | Rolandow | nice vibes in this channel :-) like it |
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| 12:16.33 | Rolandow | any comments on hylafax or hylafax+ ?? i installed hylafax, and then i found that hylafax+ is supposed to be better? |
| 12:17.02 | Rolandow | would the experts here recommend switching to hylafax+ ? |
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| 12:17.17 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:19.28 | aberrios | Rolandow I know a teleco provider in Denmark is using plain Hylafax |
| 12:19.32 | aberrios | they loved it |
| 12:23.50 | Rolandow | so no need to swap then |
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| 12:33.54 | CommaCrazy | hey all, in extensions.conf when I wanna dial for instance 88 and any other number as in as many other dialed number, how do I write that |
| 12:33.57 | CommaCrazy | ? |
| 12:34.11 | CommaCrazy | can it be doen with * |
| 12:34.37 | CommaCrazy | cause I know X is any 1 number but how do I tell it any number and as many typed |
| 12:34.44 | CommaCrazy | entered* |
| 12:39.13 | [TK]D-Fender | CommaCrazy, "_88." |
| 12:39.27 | [TK]D-Fender | CommaCrazy, You should reread the chapter on dialplan basics |
| 12:40.21 | CommaCrazy | yes, I should and I will, sorry to ask a newb q I'm in a hurry, really sorry |
| 12:52.42 | leifmadsen | WIMPy: https://issues.asterisk.org/jira/browse/ASTERISK-20007 |
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| 13:16.32 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hbqmswiekmfsyijd) |
| 13:16.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:18.08 | jacobw | what options do have for sip proxying with asterisk? |
| 13:18.46 | [TK]D-Fender | You don't. Asterisk is NOT a SIP proxy. |
| 13:18.49 | [TK]D-Fender | ~b2bua |
| 13:18.49 | infobot | methinks b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 13:19.49 | jacobw | all i'm trying to do is use the a=sendonly method of holding calls over a trunk |
| 13:20.21 | jacobw | where the other end of a trunk is also sip |
| 13:22.08 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 13:22.12 | nickfennell | hihi |
| 13:22.24 | nickfennell | can someone tell me how manually clear a hung sip call |
| 13:22.54 | [TK]D-Fender | jacobw, * doesn't initiate any kind of "hold". * receives it however |
| 13:23.08 | [TK]D-Fender | nickfennell, channel request hangup |
| 13:23.19 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 13:24.59 | nickfennell | [TK]D-Fender: I don't think that works because it's a hung call |
| 13:26.31 | jacobw | [TK]D-Fender: i see what you mean, my requirements are kind of arbitrary anyway |
| 13:26.43 | nickfennell | Phones get stuck in 'Rx: BYE' |
| 13:27.18 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 13:27.22 | jacc0 | hi all! |
| 13:27.26 | jacc0 | almost weekend :) |
| 13:27.28 | jacobw | the sender of BYE responds 487 busy to new INVITES ? |
| 13:27.45 | nickfennell | jacobw: yes. |
| 13:30.28 | jacc0 | anyone from the netherlands in here? and looking for an (asterisk) job? |
| 13:30.57 | jacc0 | I've just told my boss I'm quiting |
| 13:31.02 | leifmadsen | jacc0: oh snap |
| 13:31.03 | jacc0 | :P |
| 13:32.02 | jacc0 | I'm leaving to india next month :) |
| 13:32.07 | jacc0 | for a month |
| 13:32.14 | *** join/#asterisk serafie (~erin@nat/digium/x-bvudtbvmdwlnsqqy) |
| 13:32.32 | jacc0 | after that I'll be starting my new job as hacker/security specialist |
| 13:32.40 | [TK]D-Fender | nickfennell, that isn't a "call"... |
| 13:33.52 | nickfennell | Now it's a hung state |
| 13:33.58 | nickfennell | but it shows as a channel in use |
| 13:34.02 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:35.08 | [TK]D-Fender | Where? |
| 13:35.14 | [TK]D-Fender | "core show channels" <- ? |
| 13:35.18 | [TK]D-Fender | if not, it's not a call... |
| 13:35.27 | [TK]D-Fender | Where is the real impact? |
| 13:36.03 | nickfennell | yes |
| 13:36.08 | nickfennell | core show channels shows as in use |
| 13:36.13 | nickfennell | also sip show channels |
| 13:36.15 | nickfennell | sip show inuse |
| 13:36.28 | nickfennell | eg, SIP/412-000009bc 200@from-internal:1 Up AppDial((Outgoing Line)) |
| 13:37.12 | nickfennell | The "real impact" is the phone is in use and returns 487. To clear it I have to restart Asterisk |
| 13:38.56 | [TK]D-Fender | nickfennell, use AMI to dump them off a cliff |
| 13:39.22 | nickfennell | I can't find a tool to push them |
| 13:39.54 | nickfennell | I'm on the CLI but are you saying there's more barbaric methods via AMI ? |
| 13:40.37 | [TK]D-Fender | yes. AMI Redirect |
| 13:42.39 | nickfennell | so where would you recommend to redirect them |
| 13:42.44 | nickfennell | Not used this command before |
| 13:47.05 | *** join/#asterisk oej (~olle@86.183.216.87.static.jazztel.es) |
| 13:47.25 | nickfennell | oh wait |
| 13:47.27 | nickfennell | sussed it |
| 13:47.40 | nickfennell | that works nicely, channel redirect channel extension |
| 13:47.45 | nickfennell | channel now cleared. |
| 13:48.32 | nickfennell | but now I have AsyncGoto/SIP/412-00 200@from-internal:1 Up AppDial((Outgoing Line)) |
| 13:53.54 | *** join/#asterisk dandate2 (~dan@180.190.195.83) |
| 13:54.11 | dandate2 | why does sip show peers show a higher ping time than a cli ping test? |
| 13:55.09 | dandate2 | about 150ms more |
| 13:55.18 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 14:02.57 | Rolandow | jacc0: where do you work? |
| 14:03.03 | Rolandow | jacc0: what area? i'm from NL |
| 14:03.03 | [TK]D-Fender | because that has nothing to do with PING times |
| 14:03.27 | [TK]D-Fender | that is based on client priority and can be as stupidly off as the designers made it |
| 14:03.36 | dandate2 | got it |
| 14:03.44 | [TK]D-Fender | totally different layers and levels of prioritization |
| 14:04.55 | Rolandow | is there a way i can force asterisk to complete the dialplan when with ReceiveFax the other party hangs up early |
| 14:05.06 | Rolandow | now the fax won't send as email |
| 14:06.31 | [TK]D-Fender | Rolandow, "h" <---- |
| 14:07.02 | Rolandow | [TK]D-Fender: ok, but are the variables still set then? |
| 14:07.17 | [TK]D-Fender | go look |
| 14:07.25 | Rolandow | [TK]D-Fender: i mean, if two faxes come in, that will conflict |
| 14:07.42 | [TK]D-Fender | One channels variable s have NOTHING to do with anothers |
| 14:07.43 | Rolandow | or is it thread safe? |
| 14:07.46 | Rolandow | ok |
| 14:08.44 | Rolandow | so i could configure the extension i need to receive faxes, to goto fax-in context |
| 14:08.46 | Rolandow | and handle it there |
| 14:08.58 | Rolandow | and add the h exten |
| 14:10.53 | Rolandow | thanks .. i will figure it out :) |
| 14:10.59 | Rolandow | have a nice weekend! |
| 14:11.03 | Rolandow | chao |
| 14:14.19 | rjvvliet | Can anyone suggest an reason why asterisk.ctl is not created when started from init script?, works when staring in forground but, cannot see an error in the log. |
| 14:14.46 | rjvvliet | OS is CentOS with init script from source tar. |
| 14:14.49 | WIMPy | Not or not where you expect it? |
| 14:16.13 | rjvvliet | asterisk -r is not working. build from source sould be in /var/run/asterisk/ but contains only .PID file. asterisk is listed with ps -aef |
| 14:26.26 | rjvvliet | Sorry guys my bad. its the second time i forget to check if kernel sources and running kernel are in sync.... Fixed.... |
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| 14:52.46 | tommy87it | which is the best way to get the caller id of an incoming call to an extension? |
| 14:52.54 | tommy87it | with AMI |
| 14:54.24 | rjvvliet | tomaw: in the dialplan of external application? |
| 14:54.34 | rjvvliet | of = or |
| 14:54.46 | [TK]D-Fender | AMI <--------- |
| 14:54.59 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:56.37 | WIMPy | tommy87it: You fiond it in the Newchannel event. |
| 14:58.49 | tommy87it | ok, and if I don't want to parse events? I want to check if the extension of a users is ringing, and if it's ringing, I want to know the caller id of the incoming call |
| 14:59.55 | WIMPy | Then you can use getvar once you have found out what channel it is about. |
| 15:00.04 | WIMPy | So parsing events should be a lot easier. |
| 15:00.49 | tommy87it | yes, but if the phone is ringing, there is no channel with the extension |
| 15:01.24 | WIMPy | The phone won't be ringing without a channel. |
| 15:06.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:06.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 15:26.30 | tommy87it | WIMPy, ok, you're right… I have the channel… Which is the var to get? |
| 15:26.58 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:33.05 | WIMPy | core show function CALLERID |
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| 15:54.10 | jmordica_mbp | which QOS routers have you guys had good luck with? i've tried rv082 and rv042 with no luck really.. |
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| 16:05.14 | carrar | Cisco RVxxx are rebranded routers |
| 16:05.38 | carrar | Cisco 2800/2900 are great small office routers |
| 16:06.57 | carrar | RVxxxx are really "Linksys" |
| 16:12.34 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
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| 16:32.07 | RoyK | coppice: long time :) |
| 16:33.34 | RoyK | anyone that knows how I can find a description of analogue signalling as to design a somewhat hackish ATA? |
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| 17:02.01 | *** join/#asterisk DerkKo (~afernande@12.237.164.151) |
| 17:02.34 | DerkKo | Hey guys, I'm having a little bit of trouble with meet me.. Im trying to set a moderator pin rot a meet me room but no pin for users… |
| 17:06.42 | *** part/#asterisk RoyK (~roy@213.236.233.237) |
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| 17:21.41 | phix | hey, sip instance message in asterisk 1.8, possible? |
| 17:21.49 | phix | instant even |
| 17:22.22 | ibercom | Hello, what means this "app_dial.c: Unable to write frametype: 2" ? |
| 17:22.25 | ibercom | Why does this happen? |
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| 17:29.48 | *** join/#asterisk fenrus (fenrus@drickheroin.se) |
| 17:30.34 | fenrus | evening lads |
| 17:30.37 | fenrus | and girls |
| 17:30.59 | *** join/#asterisk MichaelB8225 (46441f88@gateway/web/freenode/ip.70.68.31.136) |
| 17:31.36 | MichaelB8225 | good morning. Anyone online able to help with a MeetMe issue? |
| 17:31.49 | Qwell | ~ask |
| 17:31.49 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:32.57 | MichaelB8225 | so I've setup a MeetMe conferecing room in the Meetme.conf and in Extensions.conf based on this Asterisk The Definitive Guide book. I've given it extension 6000, reloaded SIP, the DIalplan and even rebooted for kicks... |
| 17:33.16 | MichaelB8225 | but when I try to call with my softphone (Zoiper), it just says No Route to Destination |
| 17:33.34 | MichaelB8225 | softphone to softphone calling works, verified that last night. |
| 17:34.31 | Qwell | So that's not a meetme question |
| 17:35.06 | DerkKo | i have a meet me question to: How can i have a moderator with a pin, without the user having to key the conference PIN |
| 17:35.21 | Qwell | DerkKo: use 2 extensions. |
| 17:35.37 | p3nguin | Or one extension and conditional branching. |
| 17:36.02 | ibercom | What means this error "app_dial.c: Unable to write frametype: 2" ? |
| 17:36.15 | ibercom | Why does this happen ? |
| 17:36.32 | MichaelB8225 | @Qwell - so if its not a MeetMe question, what kind of question is it? :-) |
| 17:37.06 | p3nguin | networking and phone configuration |
| 17:37.32 | MichaelB8225 | hmmmm. Well, I can call another softphone from my softphone. But I can't call into the MeetMe extension |
| 17:37.54 | p3nguin | I'm not seeing the evidence. |
| 17:38.02 | ibercom | Context. |
| 17:38.13 | fenrus | i'd like a queue to playback a message, cant seem to get anything playing at all.. but the queue works for incoming calls.. |
| 17:38.46 | p3nguin | Look at the announcements for the queue. |
| 17:38.49 | MichaelB8225 | I had a 15 minute conversation with my co-worker last evening, he and I are both connected to Asterisk via Zoiper... |
| 17:39.06 | p3nguin | I'm still not seeing the evidence. |
| 17:39.09 | p3nguin | ~pb |
| 17:39.10 | 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. |
| 17:39.19 | fenrus | i tried queue-callswaiting=some message and the option "announce-frequency=5" |
| 17:41.13 | DerkKo | Qwell how would one look VS the other in the dial plan ? |
| 17:41.27 | Qwell | one would require an admin pin |
| 17:43.21 | DerkKo | exten => _X.,3,MeetMe(,Msw) <- is what i have now |
| 17:48.52 | MichaelB8225 | p3nguin, what would you like to see in the pastebin? |
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| 17:55.06 | *** join/#asterisk sofh (~sofh@92.96.151.112) |
| 17:55.26 | sofh | hello all |
| 17:56.20 | DerkKo | Qwell http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#Options is not clear to me how to specify to ask for pin or not from dialplan |
| 17:56.32 | Qwell | core show application meetme |
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| 17:57.35 | sofh | if callerid of incoming call is alphanumeric or digits only , how can i change its callerid into something random numeric ? |
| 17:59.30 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-38-83.user.veloxzone.com.br) |
| 17:59.55 | sofh | some one please help |
| 18:00.17 | Kalidarn | oh Qwell migrated to Asterisk 1.8, chan-sccp-B trunk now works nicely with it. |
| 18:00.27 | *** part/#asterisk iEatChildren (~iEC@mail.redglaze.com) |
| 18:02.09 | anonymouz666 | very nice that sox 14 can generate spectogram from recorded calls |
| 18:02.22 | anonymouz666 | a ready .png file |
| 18:07.58 | fenrus | win 14 |
| 18:08.01 | fenrus | fail! :D |
| 18:09.10 | p3nguin | michaelb8225: I'd like to see that certain calls work and others do not work. core set verbose 3 |
| 18:09.48 | p3nguin | sofh: How many digits of randomness do you want? |
| 18:10.01 | sofh | 6-7 may be |
| 18:11.08 | p3nguin | Set(CALLERID(num)=${RAND(100000-9999999)}) |
| 18:11.11 | p3nguin | oops |
| 18:11.15 | p3nguin | Set(CALLERID(num)=${RAND(100000,9999999)}) |
| 18:11.22 | p3nguin | sorry about the typo |
| 18:11.40 | sofh | ok , but tihng is i want to change Caller ID only when its NOT numeric , if its already numeric then it should pass as it is |
| 18:12.41 | p3nguin | Okay, so we just use a condition. You can do it with ExecIf, GotoIf, GosubIf, or anything else that does something based on the result of something else. |
| 18:13.14 | sofh | oh its a new thing for me |
| 18:13.29 | sofh | do i need to use some AGI script |
| 18:13.42 | sofh | or can be done directly in extensions.conf ? |
| 18:14.34 | MichaelB8225 | p3nguin: here you go. this gives me the Hello World prompt when I dial extn 200 - http://pastebin.ca/2161731 |
| 18:14.38 | p3nguin | ExecIf($[${REGEX("[A-Za-z]" ${CALLERID(num)})}]?Set(CALLERID(num)=${RAND(100000,9999999)})) |
| 18:14.44 | p3nguin | Something like that. |
| 18:14.49 | MichaelB8225 | when I dial extn 60000 there's nothing there at all |
| 18:14.52 | p3nguin | directly in extensions.conf |
| 18:14.56 | MichaelB8225 | sorry, extn 6000 |
| 18:15.51 | p3nguin | What does extension 6000 do? |
| 18:15.59 | sofh | ok let me check that ... |
| 18:16.17 | MichaelB8225 | exten => 6000,1,MeetMe(600,i,75391) |
| 18:17.12 | MichaelB8225 | sorry should be exten => 6000,1,MeetMe(6000,i,75391) |
| 18:17.57 | WIMPy | MichaelB8225: If you don't see anything on the console, your call never gets there. |
| 18:18.17 | fenrus | p3nguin, could you throw a quick look at my queue announcement issue at http://paste.debian.net/174695/ |
| 18:18.49 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 18:19.28 | WIMPy | Is chan_sip able to receive AOC in some way? |
| 18:19.34 | eXcAliBuR | can asterisk do video conferencing? |
| 18:20.14 | WIMPy | eXcAliBuR: Not really. You can get switching on talker detection. |
| 18:21.12 | mjordan | WIMPy: which conferencing app are you referring to? |
| 18:21.13 | p3nguin | fenrus: What is or isn't happening when in that queue? |
| 18:21.37 | fenrus | p3nguin, it does not playback any message, and it seems that it does not log anny error |
| 18:21.55 | WIMPy | mjordan: Doesn't ConfBridge offer it as standard? |
| 18:22.07 | WIMPy | Or was that some add-on? |
| 18:22.17 | mjordan | WIMPy: yes - its got multiple ways of setting talk detection |
| 18:22.29 | mjordan | WIMPy: you can lock the 'talker' on a particular user |
| 18:24.06 | WIMPy | So follow who's talking is not a standard feature? |
| 18:24.41 | p3nguin | fenrus: When watching the CLI, do you see the playback of the sound file? |
| 18:25.01 | mjordan | WIMPy: no, it can do that too :-) The way you phrased your answer made it sound like doing talk detection based on 'who is talking', with no control, was a detriment to using ConfBridge |
| 18:25.12 | mjordan | I was just pointing out that in ConfBridge, you can control that |
| 18:25.15 | mjordan | that's all :-) |
| 18:25.28 | WIMPy | Oh, ok. |
| 18:25.42 | MichaelB8225 | ok, so I can call extension 200 (hello world) and extension 101 (co-workers softphone), but 6000 didn't work (when set to meetme), and I changed that to 130 for fun and it still didn't work. Clearly I'm missing something... |
| 18:26.18 | p3nguin | It sounds like your extension 6000 is not in a context that your phone has access to. |
| 18:29.27 | fenrus | nope - cant see anything, but i did a playback of the file to verify that it exist. |
| 18:29.54 | MichaelB8225 | thats the ticket. Thank you. |
| 18:30.09 | p3nguin | To be honest, all the queue settings confuse the hell out of me. |
| 18:30.17 | MichaelB8225 | now... how do I give a device in iax.conf access to more than one context? |
| 18:30.28 | p3nguin | include => other-context |
| 18:30.36 | WIMPy | MichaelB8225: Not |
| 18:30.37 | p3nguin | (extensions.conf) |
| 18:30.40 | MichaelB8225 | ahhh. ok. thank you. |
| 18:30.41 | WIMPy | What p3nguin said |
| 18:31.00 | WIMPy | Well, you can give access to all contexts. |
| 18:31.18 | MichaelB8225 | I'm a noob at this, obviously. Asterisk is a pretty steep learning curve. |
| 18:31.40 | p3nguin | I assign a context of 'phones' for my phones. The phones context contains only includes for contexts that the phone should be able to reach. |
| 18:31.43 | WIMPy | MichaelB8225: It won;t stop any time soon :-) |
| 18:32.49 | WIMPy | changed the includes to loopback switches to work around some evil bug. |
| 18:32.57 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:33.09 | p3nguin | I wish those were documented better. |
| 18:33.14 | MichaelB8225 | ok, so I've got a context of [LocalSets] I guess trying to separate 'meeting rooms' broke it... but if I wanted to set up a template for meeting rooms then I'd want a different context, no? |
| 18:33.47 | p3nguin | I don't understand your concept. |
| 18:34.03 | p3nguin | LocalSets is the equivalent of my 'phones' context. |
| 18:34.09 | MichaelB8225 | right. |
| 18:34.11 | WIMPy | I wish a lot of things were documented better. Like the seemingly partially duplicated functionality of CALLERID and REDIRECTING, none of which are really documented at all. |
| 18:34.22 | p3nguin | Within the LocalSets context, you'd just include any other context that you want your phones to be able to reach. |
| 18:34.29 | MichaelB8225 | so I went to try to put the MeetMe "rooms" in a different context. |
| 18:34.59 | p3nguin | I have a context calls 'conference' for all my ConfBridge and MeetMe extensions. |
| 18:35.15 | p3nguin | Then I just include the conference context within the LocalSets context. |
| 18:35.30 | WIMPy | Plus the misnomer in CALLERID with *-plan probably meaning type, not plan. |
| 18:35.32 | MichaelB8225 | with that other-context bit? |
| 18:35.36 | p3nguin | [LocalSets] |
| 18:35.43 | p3nguin | include => conference |
| 18:35.43 | WIMPy | MichaelB8225: exactely |
| 18:36.02 | sofh | p3nguin, it worked nice |
| 18:36.08 | sofh | thanks a ot |
| 18:36.11 | sofh | a lot* |
| 18:36.16 | p3nguin | Then put your meetme extensions under [conference]. |
| 18:36.59 | MichaelB8225 | and that all goes in the extensions.conf, not iax.conf |
| 18:37.03 | p3nguin | Correct. |
| 18:37.15 | p3nguin | Extensions are configured in extensions.conf. |
| 18:37.28 | p3nguin | IAX2 devices are configured in iax.conf. |
| 18:37.56 | p3nguin | Similarly, SIP devices in sip.conf. |
| 18:39.34 | MichaelB8225 | yes, that I know |
| 18:39.40 | MichaelB8225 | :-) |
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| 18:48.00 | p3nguin | fenrus: I'm testing your queue, and it keeps playing the following messages while I wait: your call is now first in line... and all of our representatives are currently busy... |
| 18:51.01 | fenrus | p3nguin, thats interesting. my phone is available, perhaps thats some issue |
| 18:53.00 | fenrus | say there is a couple of phones in this office, and the people working there dont sit at their desks all the time - all phones are in the same queue.. so when it rings, it will ring all phones, but if nobody answers in 10-15 seconds i'd like to playback a message that says "hold or send us an email" |
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| 19:05.25 | p3nguin | The periodic-announce is working for me. I would set that to my custom file. |
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| 19:11.12 | fenrus | p3nguin, interesting, how minimalistic can you make the configuration for it ? |
| 19:11.17 | fenrus | i cant seem to get it working |
| 19:12.35 | p3nguin | I copied yours from the pastebin, then altered the member (since SIP/314 is non-existent). |
| 19:13.13 | fenrus | have any global queue -settings other than persistentmembers = yes |
| 19:14.00 | p3nguin | No. |
| 19:15.06 | fenrus | okay, this feels fishy |
| 19:18.56 | fenrus | will sleep on it. thanks for your help! |
| 19:20.43 | p3nguin | Additional testing reveals a problem. |
| 19:21.36 | *** join/#asterisk gusto (~gusto@ip-109-84-0-6.web.vodafone.de) |
| 19:21.36 | gusto | hey |
| 19:21.43 | p3nguin | I added the min announce setting and tried again, and it caused the queue to run a full cycle before ever playing any sounds. I removed that setting and tried again, and it still behaves this way. |
| 19:21.51 | gusto | is there somewhere a list of modules i need at least? |
| 19:22.07 | gusto | because i have now only a few modules loaded but it still is quite big |
| 19:22.15 | Qwell | big? |
| 19:22.27 | p3nguin | So my device has a timeout of 60, and the queue has to go 60 seconds before it tries the device again. On the second attempt, then it plays the announcements. |
| 19:22.34 | WIMPy | We don;t know what YOU need. |
| 19:22.45 | gusto | SIP ;-) |
| 19:23.00 | p3nguin | If you need sip, load chan_sip. |
| 19:23.02 | gusto | would it work with only chan_sip.so loaded? |
| 19:23.07 | gusto | ha? |
| 19:23.12 | p3nguin | Well, that was easy. NEXT! |
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| 19:23.37 | gusto | will RTP, digest auth and everything work with only chan_sip.so loaded? |
| 19:23.38 | p3nguin | chan_sip will work with only chan_sip loaded, but you're going to want other things, I'm quite sure. |
| 19:24.05 | gusto | yes, i am now trying to make it as small as possible, i can load other modules as needed |
| 19:24.40 | gusto | but i disabled autoload because on some installations there are a lot of modules loaded then |
| 19:24.58 | p3nguin | Maybe you should be using autoload, and only configure what you need to use. |
| 19:25.09 | gusto | yes |
| 19:25.34 | gusto | but i wanted to go one step forward |
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| 20:13.06 | DerkKo | can i collect the conference room and pin before going into meet me? |
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| 20:22.06 | Ryushin | In the US. Going to convert a company from using a PRI to VoIP solution using a VoIP trunking provider. However, I don't know who is reliable and who is not. Has anyone had any good success with a voip provider in the US that they could recommend? |
| 20:23.18 | WIMPy | ~itsplist-us |
| 20:23.18 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:23.29 | Ryushin | I would prefer them to provide service to us using IAX2. Have about 110 DID's that have to port as well. |
| 20:23.57 | Ryushin | WIMPy: Thank you. |
| 20:25.43 | *** join/#asterisk metabsd (454672ae@gateway/web/freenode/ip.69.70.114.174) |
| 20:25.49 | metabsd | hi all |
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| 20:26.50 | luteriano | hi |
| 20:27.18 | metabsd | i need to know if i use asterisk 1.8.11 before the Certified Asterisk 1.8.11-cert2 Released. i have good reason to upgrade ? |
| 20:27.29 | metabsd | for an example meet-me bug |
| 20:28.07 | *** join/#asterisk diatonic (~diatonic_@70.98.114.228) |
| 20:28.28 | metabsd | right now i have problem with paging with meet-me |
| 20:28.44 | metabsd | i have aastra phone and i use paging feature by meet-me |
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| 20:50.28 | ruunix | Hi there .. |
| 20:50.52 | ruunix | somebody can helpme with this ? exten=>_8X.,1,DIAL(DAHDI/4/${EXTEN:1}) |
| 20:51.06 | gusto | hey p3nguin |
| 20:51.10 | gusto | p3nguin: what are you doing now? |
| 20:51.19 | ruunix | how can i make if the dahdi 4 is used |
| 20:51.24 | ruunix | to jump to dahdi 5 .. |
| 20:51.28 | ruunix | something like that ? |
| 20:51.54 | WIMPy | ruunix: Define a group in chan_dahdi.conf and dial that. |
| 20:52.03 | ruunix | thanks. . |
| 20:52.53 | p3nguin | gusto: pooping |
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| 20:57.23 | gusto | p3nguin: what is poooping? fearting? |
| 20:58.59 | shadowapex | Hey everyone, a provider of ours requires that we have ":5060" is at the end of the request URI of our REGISTER request. When I put a non-default port, such as 5064, in the registration string, then it will show up correctly. If I put in :5060 in the registration string, however, then it excludes it from the request URI in the REGISTER packet. Anyone know of a way to force Asterisk to include ":5060" in the request URI of t |
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| 21:01.29 | shadowapex | My current register string is this: user:pw@some.provider.net:5060/user. But the request URI in the REGISTER packet itself is just: "sip:some.provider.net SIP/2.0" instead of "sip:some.provider.net:5060 SIP/2.0" |
| 21:08.49 | p3nguin | If you do not specify port 5060, port 5060 is used. |
| 21:09.19 | p3nguin | No port means port 5060. |
| 21:15.30 | shadowapex | p3nguin: I know that, but with this particular provider, they NEED to have port 5060 in the request URI of the REGISTER |
| 21:15.47 | Qwell | Sounds like you NEED to find a new provider. :) |
| 21:15.56 | shadowapex | Qwell: Yeah, I know :P |
| 21:16.11 | Qwell | If you can't even register... what hope do you have of actually making a call? |
| 21:17.19 | shadowapex | Qwell: Yeah, it's pretty bad. I don't know any other providers that require you to have the default port in the REGISTER uri |
| 21:18.17 | shadowapex | So if they don't get the REGISTER packet in this format: "sip:crappy.provider.net:5060 SIP/2.0" then they don't respond to the request. |
| 21:19.35 | shadowapex | If I put in :5064 or something similar in the register string, then it will show up as: "sip:crappy.provider.net:5064 SIP/2.0", but not if I put :5060. |
| 21:19.50 | Qwell | yes, we get the problem |
| 21:20.32 | Qwell | You're likely going to have to patch chan_sip. |
| 21:20.49 | shadowapex | That's what I was hoping wasn't the case >.<; |
| 21:21.20 | WIMPy | You could also try another provider. |
| 21:21.21 | Qwell | well, I gave you your better option already |
| 21:21.25 | WIMPy | Or another software. |
| 21:22.21 | shadowapex | Yeah, I was hoping there was some magical configuration option somewhere that I missed. |
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| 21:36.35 | DerkKo | how can i invoke a custom greeting from meet me ? |
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| 22:27.20 | fubada | hey, does "make install" create the asterisk user? |
| 22:27.55 | WIMPy | no |
| 22:30.09 | carrar | PEOPLE craete the asterisk user!! |
| 22:30.16 | carrar | create even |
| 22:30.26 | carrar | the circle of life |
| 22:31.18 | [TK]D-Fender | Kinda like : http://img2.moonbuggy.org/imgstore/om-nom-nom-nom-snake.jpg |
| 22:35.30 | carrar | hah |
| 22:36.02 | carrar | exactly!! |
| 22:45.25 | jpsharp | Assuming you even use an asterisk user. |
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