| 00:00.53 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
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| 13:44.14 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 13:44.14 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.0 (2013/07/15), 10.12.2 (2013/03/27), 1.8.23.0 (2013/07/15), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 13:44.17 | Qwell | WIMPy: no, hold on, go back a second |
| 13:44.28 | Qwell | WIMPy: Why are there plants in the shower? |
| 13:44.57 | Greenlight | I did wonder that as well |
| 13:44.57 | Qwell | Greenlight: we all were... |
| 13:45.09 | Greenlight | A watering can just didn't cut it / |
| 13:46.16 | WIMPy | Because I have an issue with spider mites. :-( So I tried to get the worst of the issue off mechanically. |
| 13:46.44 | igcewieling | jmetro: mu ascii over dtmf experiment |
| 13:47.14 | WIMPy | These things are extremely annoying. They re-appear over and over again. Even if you thought you defeated them and didn;t see anythign for months. |
| 13:47.26 | WIMPy | So if anyone knows a cure... |
| 13:53.48 | WIMPy | Hmm. 2nd phone doesn't ring at all. Well, CUL |
| 13:58.34 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 14:04.35 | igcewieling | "Thank you for calling. If you are calling before 9:30am, please hangup and call back at a decent hour." |
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| 14:09.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 15:01.57 | human39 | Hello all! I'm need a recommendation on hardware. We have a need to send SMS's directly from asterisk. We decided that we want the hardware to do it ourselves. I assume that we would need a GSM modem and a carrier that supports this type of a system. Any recommendations? |
| 15:02.35 | wasanzy | if I want to start asterisk and see any error on startup i use something like this: asterisk -rvvvvvvvv? |
| 15:02.50 | human39 | It won't be high traffic, so single or dual channel would be fine. |
| 15:02.52 | [TK]D-Fender | wasanzy: No, that connects to a RUNNING asterisk |
| 15:03.27 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 15:03.34 | wasanzy | ok |
| 15:03.40 | [TK]D-Fender | wasanzy: asterisk -gvvvvvvvvvvvc |
| 15:03.51 | wasanzy | ah right |
| 15:03.53 | wasanzy | thank you |
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| 15:10.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:11.52 | mic_ | WIMPy: I cannot remember if I discussed the delay issue with you |
| 15:12.07 | mic_ | WIMPy: but a malfunctioning DNS was the problem. |
| 15:17.51 | Greenlight | Hmm am I missing something or is my telco being stupid. I'm tracking down a problem with some ISDN PRI lines. They're asking for the DDI I'm using when calling OUT for the failed calls. I've offered the CLI I'm presenting but they're saying, no, they need a DDI number. They're not wanting the "main bearer" number either. |
| 15:18.37 | igcewieling | Greenlight: they want the CID you are sending when you call out |
| 15:19.48 | lodac | I am able to 4 digit call from any of the building to the other building. Asterisk routes all of the calls via the digium trunks. I have incoming calls routing to the correct trunks and the phones are picking it up correctly. However when an inside phone dials out (91NXXNXXXXXX), It doesn't go out the AT&T PRI. But if I dial out with a softphone, it works. |
| 15:19.58 | lodac | http://i.imgur.com/uDztR5f.png (Topo), http://pastebin.com/E8my9Qga (asterisk debug), http://pastebin.com/j2B3ZtgY (Digium G100 debug) |
| 15:20.08 | lodac | What am I missing? |
| 15:21.04 | Greenlight | igcewieling: Yes, that's what I thought, and offered to them the "calling party" or "presentation CLI" or "caller id" |
| 15:21.22 | Greenlight | But, not they are certain they don't want that they want the specific DDI |
| 15:21.44 | Greenlight | Which has me going "wtf" |
| 15:22.43 | Greenlight | Just wanted some reassurance that I'm not crazy really |
| 15:23.08 | Greenlight | Before I start telling them they are |
| 15:23.39 | mic_ | Greenlight: don't get me wrong |
| 15:24.02 | mic_ | Greenlight: but your worries kind of are comforting to me |
| 15:24.09 | mic_ | Greenlight: because I was going through similar thing in july |
| 15:24.10 | igcewieling | huh? the specific DDI is the callerid number |
| 15:24.26 | mic_ | Greenlight: it seems it's just the ISDN itself that makes people go nuts. |
| 15:24.34 | mic_ | (sorry for a unrelated note) |
| 15:24.38 | Greenlight | Yea, it seems that way ;) |
| 15:24.47 | Greenlight | Or maybe telcos are just PITA to deal with |
| 15:24.48 | igcewieling | apparently y'all don't use SIP. |
| 15:24.57 | Greenlight | I try to when I can |
| 15:25.07 | Greenlight | BUt this particular customer clings to their ISDN |
| 15:25.39 | igcewieling | I meant, if you think PRI is confusing, try SIP, you'll realize ISDN is simple. |
| 15:25.59 | igcewieling | I love ISDN, so much easier to troubleshoot than SIP |
| 15:26.04 | Greenlight | Oh, I thought you mean SIP ITSP's were easier to work with |
| 15:26.53 | Greenlight | SIP I find that generally I have more control over fixing issues, with ISDN it's going to involve the telco who'll swear everything is working okay for as long as they can |
| 15:26.59 | mic_ | Greenlight: we service some health-related things from the town hall - out there they haven't heard of SIP - all ISDN & bazillions of cables |
| 15:27.01 | igcewieling | Greenlight: if you can reproduce the issue on demand, send them a PRI debug capture of a failed call |
| 15:27.25 | Greenlight | Sent them that THREE weeks ago ;/ |
| 15:27.31 | *** join/#asterisk navaismo (~navaismo@189.241.28.242) |
| 15:27.41 | igcewieling | send more. |
| 15:27.56 | Greenlight | Can't reproduce on demand, have to trawl the logs for it |
| 15:28.27 | mic_ | if you can borrow from someone a pro ISDN tester |
| 15:28.33 | mic_ | this is how we shut the telco |
| 15:28.45 | Greenlight | Ahh it's an odd issue that cuts calls off mid call though |
| 15:28.45 | mic_ | or at least convinced them that it's their fault. |
| 15:28.57 | mic_ | aaa, ok. |
| 15:29.00 | Greenlight | Perhaps one in every 10,000 calls ;/ |
| 15:29.09 | Greenlight | Maybe 1 in 1000 |
| 15:29.11 | mic_ | Love bugs like that :D |
| 15:29.15 | igcewieling | Greenlight: we get that with SIP. |
| 15:29.37 | Greenlight | Calls just cutting off ? |
| 15:29.55 | igcewieling | yup |
| 15:30.02 | Greenlight | It shouldn't happen. |
| 15:30.09 | igcewieling | random, very small percentage |
| 15:30.14 | Greenlight | A call 8 minutes into a conversation, suddenly failing for example |
| 15:30.18 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 15:30.31 | igcewieling | seems to mostly happen with High Drama customers. |
| 15:31.04 | Greenlight | Yes, I know those sort |
| 15:31.25 | Greenlight | This we get back in the PRI Trace: Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Network beyond the interworking point (10) |
| 15:31.39 | Greenlight | Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) |
| 15:31.59 | Greenlight | I don't see how I can get "congested" mid call. Very odd. |
| 15:32.04 | igcewieling | are you sure you are RECEIVING that message and not SENDING that message? |
| 15:32.11 | hjf | I was wondering, could I use an external device, like an SPA3102 for FXO for asterisk? |
| 15:32.22 | Greenlight | 100% :) |
| 15:32.28 | igcewieling | hjf: yes. |
| 15:32.41 | igcewieling | assuming the spa3102 has an FXO port, of course. |
| 15:32.45 | hjf | igcewieling: cool. does it work well? or is a dedicated card prefered? |
| 15:32.58 | hjf | yes the 2102 is dual fxs, the 3102 is 1fxo/1fxs |
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| 15:33.18 | Greenlight | If I'd pasted the whole line you'd see the arrows "<" |
| 15:33.43 | hjf | I want to try voip pbx for home but dedicated cards are a bit expensive. also my server is freebsd and i'd rather keep asterisk in a VM |
| 15:33.45 | igcewieling | hjf: I *might* use that setup for a personal line, but never for a customer. Adds far too much additional complexity and makes troubleshooting tough. |
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| 17:38.29 | *** join/#asterisk TheGuy- (d1cc9202@gateway/web/cgi-irc/kiwiirc.com/ip.209.204.146.2) |
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| 17:51.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:53.19 | igcewieling | has anyone see this warning? WARNING[12707]: channel.c:3622 ast_waitfordigit_full: The FD we were waiting for has something waiting. Waitfordigit returning numeric 1 |
| 17:57.09 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
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| 18:05.40 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 18:06.55 | igcewieling | heh, I broke the pbx |
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| 18:23.54 | WIMPy | Greenlight: It says temporary failure. Not congested. |
| 18:30.16 | TheGuy- | where can I begin to troubleshoot random users not being able to access meetme, according to log files the users are entering the wrong pin but when I stand next to the user and watch them dial they enter the correct digits |
| 18:31.32 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-flhrlvpxexhlcoso) |
| 18:31.55 | [TK]D-Fender | TheGuy-: go make a test extensions for them to run through while you watch |
| 18:32.20 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:32.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:34.03 | jmetro | [TK]D-Fender: go make a test sandwich for me to nom while i wait. |
| 18:34.35 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:34.46 | [TK]D-Fender | jmetro: http://xkcd.com/149/ |
| 18:35.20 | igcewieling | TheGuy-: sounds like a dtmf issue to me |
| 18:37.55 | TheGuy- | [TK]D-Fender: I have added a few test meetme rooms as one of our users thought the number 2 was the problem. I looked at the logs and asterisk says they are dialing 31146 while the user is dialing 31174 and I have verified with my own eyes they are dialing the proper sequence. They do seem to have more luck if they dial slowly though, relaxdtmf=off |
| 18:37.57 | TheGuy- | <PROTECTED> |
| 18:42.52 | igcewieling | TheGuy-: missing digts are common, duplicated diigits are common, dtmf being detected as the wrong digit is very very rare. |
| 18:43.25 | igcewieling | what brand of SIP phone are you using? is the phone set for rfc2833? |
| 18:44.13 | TheGuy- | Polycom 501 and yes rfc2833 and in some cases users are on a cell phone |
| 18:45.23 | igcewieling | set your dtmf length to 70ms in the polycom configs |
| 18:46.00 | igcewieling | as for the cell issues, you should verify with your carrier they are using rfc2833 on your sip account |
| 18:47.34 | TheGuy- | What can I do for users that are using the PRI to access meetme? Does AT&T offer any info regarding this type of issue? |
| 18:48.03 | WIMPy | No. It's always inband, so your telco is out. |
| 18:48.46 | WIMPy | You could try to either use hardware or software DTMF detection. |
| 18:49.29 | *** join/#asterisk sidus (~abracadab@37-5-74-93-dynip.superkabel.de) |
| 18:50.32 | Katty | hi kids |
| 18:50.39 | igcewieling | or enable long tones on the cell, like everyone should so they can reliabily use IVRs |
| 18:53.43 | jmetro | Katty: Aye aye captain. |
| 18:55.28 | Katty | hugs jmetro |
| 18:57.26 | jmetro | :< -> :> |
| 19:00.06 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 19:01.23 | Vann | Hello. Does anyone know if it's possible to play a beep (or any sound) when someone connects to a conference call? |
| 19:01.41 | drmessano | yes |
| 19:01.49 | jmetro | you mean a join/leave notification in confbridge? |
| 19:03.19 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:04.06 | igcewieling | Vann: "conference call" doesn't mean anything. there are at least 4 ways to do "conference calls" with Asterisk |
| 19:04.45 | Vann | er sorry, yeah jmetro that sounds about right |
| 19:04.46 | Vann | I believe we do it via meetme |
| 19:04.59 | jmetro | Vann: i would switch to confbridge =) |
| 19:05.34 | jmetro | otherwise http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe tells you what to do for meetme. |
| 19:05.55 | TheGuy- | WIMPy:Do I do that in /etc/dahdi/system.conf with cidsignaling=dtmf? How do I change it from software? |
| 19:05.59 | jmetro | or more accurately, core show application meetme |
| 19:06.31 | Vann | Awesome. I appreciate it jmetro. |
| 19:06.58 | WIMPy | TheGuy-: You change it when loading the card driver via module parameter. |
| 19:07.02 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:07.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:08.23 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:09.47 | *** join/#asterisk boom^time (~boomtime@c-68-42-110-49.hsd1.mi.comcast.net) |
| 19:09.59 | boom^time | Hey guys, any good application for checking for file existance? |
| 19:10.13 | jmetro | stat. |
| 19:10.15 | boom^time | From my research it looks like STAT used to exist, but doesn't seem to on 11.5 |
| 19:10.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:10.20 | WIMPy | test |
| 19:10.30 | jmetro | stat is part of linux. |
| 19:10.36 | jmetro | you can call it from dialplan |
| 19:11.22 | boom^time | [Aug 23 15:02:30] WARNING[5184][C-000000f9]: pbx.c:4621 pbx_extension_helper: No application 'STAT' for extension (record-ivr-s1-s1, s, 2) |
| 19:11.32 | jmetro | lemme show you |
| 19:11.45 | [TK]D-Fender | boom^time: Show your research |
| 19:11.46 | Qwell | Application? |
| 19:11.54 | [TK]D-Fender | Qwell: SHHH!!! |
| 19:12.08 | boom^time | research: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_STAT |
| 19:12.13 | skrusty | evening all |
| 19:12.55 | jmetro | huh wait a second.. |
| 19:13.00 | skrusty | wondering if i can get a bit of community feedback on a website i've started to put together to index AGI scripts: http://theagigallery.azurewebsites.net |
| 19:13.15 | [TK]D-Fender | boom^time: .......... FUNCTION |
| 19:13.21 | boom^time | gotcha. |
| 19:13.26 | skrusty | the idea is to create a resource for people to find AGI scripts open source or otherwise |
| 19:13.35 | boom^time | my mistake |
| 19:13.44 | jmetro | http://pastebin.com/pKnqN6CL |
| 19:14.13 | jmetro | 0 means no exist |
| 19:14.27 | boom^time | jeez jmetro didn't you see [TK]D-Fender telling everyone to "shh"?!! now |
| 19:14.34 | boom^time | I'll never learn for myself. J/k thank you very much. |
| 19:16.09 | [TK]D-Fender | boom^time: It was to see what you were using as a reference so we could point out what you missed in it. Or to out a bad reference |
| 19:16.19 | [TK]D-Fender | boom^time: Devil's in the details |
| 19:16.35 | igcewieling | boom^time: remember every dumb question reduces your karma. n00bs get some amount of karma when they start. If you use it all up people stop helping you. |
| 19:16.49 | boom^time | [TK]D-Fender: agreed. |
| 19:17.31 | jmetro | igcewieling: Psh, i'll always help people, but usually force them to not be annoying. |
| 19:17.34 | boom^time | igcewieling, I suppose I'll just have to make a new nick whenever I run out then. |
| 19:17.42 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 19:17.55 | navaismo | they check your IP |
| 19:17.59 | navaismo | and block you |
| 19:18.03 | boom^time | navaismo, tor |
| 19:18.08 | navaismo | people here are evil |
| 19:18.11 | boom^time | and it was just a joke anyhow. |
| 19:18.39 | igcewieling | jmetro: before you know it you will be as bitter and cynical as the rest of us. |
| 19:19.17 | igcewieling | My karma comment was only partially a joke. People WILL stop helping if you ask too many really dumb questions. |
| 19:19.34 | boom^time | So far I've found the jadedness quite entertaining. Maybe someday I'll be fortunate enough of an expert to partake. |
| 19:19.45 | boom^time | Yeah, I wouldn't blame them. |
| 19:19.46 | [TK]D-Fender | Which you're nowhere near a reasonable threshold of... |
| 19:20.44 | boom^time | Well that's good to know :) |
| 19:22.05 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:22.05 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:22.20 | [TK]D-Fender | heads out for a while |
| 19:24.20 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:26.21 | igcewieling | Customer is complaining about their internet being slow and voice issues. Turns out they are maxing out their circuit -- I even sent them graphs showing this, what is the reply "Knowing why the internet is slow does not help make it faster. MAKE IT FASTER." |
| 19:27.36 | navaismo | yes dude, they now its slow you need to make it faster, what your problem? |
| 19:33.42 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 19:36.12 | TheGuy- | WIMPy: I've been browsing sangoma's documentation and they recommend trying to turn relaxdtmf on, everything I've read says don't turn relaxdtmf on, whaddya think? |
| 19:37.55 | WIMPy | I haven't played with it. I don't use much dahdi. |
| 19:53.21 | WIMPy | Ok, so now I made the test of calling two sip devices. |
| 19:53.52 | WIMPy | Creating teh 1st destination channel took 385 ms, creating the 2nd one "only" 84 ms. |
| 20:05.49 | WIMPy | Adding both the current local IP and that of the server to /etc/hosts does not change timing. |
| 20:08.18 | navaismo | 385ms is really a bad time? |
| 20:09.08 | WIMPy | Don't you think so? |
| 20:09.08 | jmetro | depends on the destination |
| 20:09.21 | jmetro | china, not too bad. across the hallway, awful |
| 20:09.40 | WIMPy | That is the time before trying to do anything. Only for creating the channel. |
| 20:10.02 | jmetro | like a performance issue? |
| 20:10.18 | WIMPy | Yes |
| 20:10.23 | navaismo | 385milliseconds?? aint nobody notice that |
| 20:11.06 | WIMPy | I notice it every call as I send a nootifiert to my desktop right befor doing the Dial. |
| 20:11.11 | jmetro | its a system operation not a network op |
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| 20:11.30 | WIMPy | When I get the banner I have to wait until a phone rings. |
| 20:12.13 | WIMPy | Like: He, I want to answer that call. Where is it? |
| 20:12.54 | WIMPy | Times vary, I've also had 460ms. |
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| 20:14.33 | navaismo | top report peaks on cpu or memory when yo make the call? |
| 20:14.59 | WIMPy | Not that I'd need it at home, but at the current performance, it probably couldn't handle more than 20 calls/minute. |
| 20:15.26 | WIMPy | I'm not sure I'd be able to see it at that speed. |
| 20:17.38 | WIMPy | When using vmstat I see a peak in context switches. |
| 20:19.16 | WIMPy | And a little peak at waiting time. |
| 20:21.09 | WIMPy | And an even smaller one for user time, reaching like 12%. |
| 20:22.52 | WIMPy | Thinking of which... 12% in one second would translate to 100% for half a second. |
| 20:23.07 | WIMPy | So I guess that makes a yes after all. |
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| 20:33.00 | navaismo | maybe the harddisk? waht if you test that on SSD |
| 20:33.36 | WIMPy | Do you think there are disk accesses at that time at all? |
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| 20:39.55 | WIMPy | I get like 130 to 640 blocks out in that second. So unless something is switing with fsync that would be nothing. |
| 20:40.27 | WIMPy | Does creating a channel write anything to AstDB or something? |
| 20:40.48 | WIMPy | *writing |
| 20:40.54 | igcewieling | doesn't astdb (now sqlite) get flushed/sync'd frequently |
| 20:41.22 | igcewieling | WIMPy: registrations do, I assume qualifies too, but I've not checked. |
| 20:42.15 | WIMPy | NFI, how that works. As you might have read, jkroon found out that AstDB seriousely slows down sip registrations, while I found out that writing to it from Dialplan doesn't take time. |
| 20:42.34 | WIMPy | Yes, but this only happen on sip channel creation. |
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| 20:44.24 | WIMPy | Oh, and only if the source channel is sip as well. |
| 20:45.27 | navaismo | any developer have responded to that? |
| 20:46.15 | WIMPy | No. Wrong if the source channel is iax, it happens as well. But not if the source channel is lcr. |
| 20:46.15 | navaismo | or is considered as -how they called that- |
| 20:46.51 | WIMPy | I did discuss it when it was even more extreme due to thread debugging enabled. |
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| 20:48.27 | navaismo | triage! |
| 20:49.03 | WIMPy | But what? |
| 20:49.55 | navaismo | i mean the status for a ticket in jira, i undesratnd that as "in the limbo" |
| 20:50.04 | navaismo | bbl going to eat |
| 20:51.03 | WIMPy | Maybe I should take it to -dev again. |
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| 22:05.15 | file | For those not keeping track libuuid will be an optional dependency in the next release of Asterisk 11 |
| 22:06.14 | WIMPy | And what do(n't) you get without it? |
| 22:06.37 | file | TURN and ICE support |
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| 22:13.08 | navaismo | hmm |
| 22:13.17 | navaismo | why that change |
| 22:14.04 | navaismo | Im feel now like a dog that first was trained to poop outside, then punished to poop outside and now dont care |
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| 23:46.57 | igcewieling | file, you are indeed awesome. |
| 23:51.46 | *** join/#asterisk trapito (~trapito@11.43.80.200.host.ifxnw.com.ar) |
| 23:51.55 | trapito | hi |
| 23:52.10 | WIMPy | lo |
| 23:52.21 | trapito | I'm trying to build dahdi but i'm getting an error and i wasn't able to find an answer in google |
| 23:53.04 | trapito | i'm getting "sed: can't read conftest.c: No such file or directory" after executing the first "make" |
| 23:53.53 | WIMPy | Sounds rather strange. |
| 23:54.05 | WIMPy | But before we go deeper: Do you need dahdi at all? |
| 23:54.32 | trapito | i'm trying to use a digium T410 |
| 23:54.45 | trapito | for my PSTN lines |
| 23:55.50 | trapito | i'm working on Centos 6.4 x64 |
| 23:56.29 | trapito | actually i'll upload the log |
| 23:56.38 | WIMPy | I wonder what wants that file. I don't have it. Not even on the box that uses dahdi. |
| 23:57.34 | trapito | well i think it's a compile time file that's used to test the config (something autotools related) |
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| 23:58.27 | trapito | http://pastie.org/pastes/8264225/text?key=4umghagmu8o68dtuxzbmw |
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| 23:58.55 | trapito | i'm executing make from "/usr/src/asterisk/dahdi-linux-complete-2.7.0.1+2.7.0.1" |
| 23:59.13 | trapito | should i ask in asterisk-dev ? |
| 23:59.56 | WIMPy | First of all it would help if that paste was multiple lines. |
