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| 03:52.36 | mbello | Hello. I am experiencing the PJSIP bug described here https://community.asterisk.org/t/13-10-0-pjsip-deadlock/67746 |
| 03:52.47 | mbello | It also exists in 13.11.2 |
| 03:53.42 | mbello | I just posted a backtrace there, should I also file a bug report? |
| 03:54.30 | mbello | Backtrace is here: http://pastebin.com/raw/wYDUkWHV |
| 03:55.21 | mbello | I have the machine on right now, if there is any further debugging I could do I am here to help getting the data to help debug it. |
| 04:26.26 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 04:26.58 | moke | Quick question. How do i prevent outbound calls (to a DID on my system) from routing out to the SIP trunk, and then back in? |
| 04:27.30 | moke | from the CDR it looks like the outbound calls to our primary number result in the call leaving to flowroute, and then come back in flowroute, billing us twice for the call. |
| 04:27.55 | moke | is there a way i can make inbound rules work on outbound calls? |
| 04:30.03 | Samot | You need to setup a custom trunk. |
| 04:31.03 | Samot | You need the custom trunk and then you match on the DIDs you want to handle the route for. |
| 04:32.00 | Samot | Route those DIDs to the custom trunk and when calls go out the custom trunk it treats it like an inbound call and routes that way. |
| 04:34.23 | Penguin | moke: Fix your extensions. |
| 04:36.30 | Samot | Why would he need to fix his extensions? |
| 04:37.04 | Penguin | Because he's complaining about something that is directly controlled by the extensions. |
| 04:37.56 | Samot | You mean when the extension dials 1NXXNXXXXXX and it matches and routes out the trunk like a regular call would? |
| 04:37.59 | moke | Samot, so im thinking the custom dial string would be PJSIP/$OUTNUM$ ? |
| 04:38.00 | Penguin | "routing" is determined by a Dial() and a SIP peer. It's controlled by the extension that has matched the called number. |
| 04:38.27 | Samot | moke: No. |
| 04:38.49 | Samot | Local/$OUTNUM$@<inbound-context> |
| 04:38.52 | Penguin | Your question doesn't make any sense to me. |
| 04:39.20 | Samot | He's saying that when a user on the PBX calls a DID that the PBX handles the call for... |
| 04:39.22 | Penguin | "extension dials 1NXXNXXXXXX and it matches and routes out the trunk" |
| 04:39.27 | Penguin | That part, not what he said. |
| 04:39.40 | Samot | He wants the call not to leave the PBX |
| 04:39.51 | Penguin | So he needs to fix the extension that is sending it out. |
| 04:40.03 | Samot | moke: Are you providing VoIP services? |
| 04:40.07 | Penguin | The call comes in, matches an extension, the extension does something with it... |
| 04:40.19 | Penguin | Clearly it's doing the wrong thing. So fix the extension. |
| 04:40.20 | moke | Samot, no, my users are just stupid. |
| 04:40.33 | Samot | moke: Oh, then train your users. |
| 04:40.46 | Samot | There is no reason they need to dial a DID to reach a local user. |
| 04:40.58 | moke | my users have ext matching 3xx, our DID is lets say 14055864125. my users try to call 14055864125 from their desk. |
| 04:41.01 | Penguin | But if you fix the extensions, it won't matter. |
| 04:41.06 | moke | the call CDR reports the call leaves our pbx, to flowroute, and then back in. |
| 04:41.17 | moke | thus, we're getting billed twice for the internal user dialing our DID |
| 04:41.26 | Samot | You shouldn't be dialing your own DID |
| 04:41.33 | Samot | Why wold you? |
| 04:41.39 | Samot | Why would you? |
| 04:41.47 | moke | because sales people like to transfer people to support, and dial the number on their mousepad. |
| 04:42.01 | Samot | Doesnt support have an extension? |
| 04:42.10 | Samot | Wouldn't dialing the extension be faster and easier?! |
| 04:42.23 | moke | look, i fix servers, not people |
| 04:42.36 | Samot | Yeah but the server doesn't need to be fixed. |
| 04:42.41 | Samot | It's performing as it should. |
| 04:42.47 | Penguin | If I wanted to solve the problem, I would create a 'locals' context with the DID numbers in it and the extensions would Dial() the local phones instead of sending out to the ITSP. The 'locals' context would be listed before the outbound dialing context. Problem solved. |
| 04:42.51 | Penguin | Very easy fix. |
| 04:42.51 | moke | i understand that, but i would like to make it function diffrently. |
| 04:42.57 | Samot | You're getting doubled build for a user issue. |
| 04:43.08 | Samot | OK, so if you know that all the users are going to dial that number... |
| 04:43.37 | Samot | Make it part of the outbound context and when that DID is matched, make it dial the support extension. |
| 04:44.00 | Penguin | I've already given you the solution. Now you just have to stop interacting and get to work. |
| 04:44.15 | Samot | Either solutin works. |
| 04:44.22 | Samot | Either solution works. |
| 04:44.56 | Samot | But the latter is the better option in this case. |
| 04:47.29 | Penguin | You probably already have a context for local devices anyway. Just list the DIDs there with the short numbers. |
| 04:48.23 | Penguin | exten => 3149691077,1,Goto(1077,1); |
| 04:48.43 | Penguin | exten => 3149699898,1,Goto(9898,1); |
| 04:49.21 | Samot | You might want to CYA even more and do the 11-digit version as well. |
| 04:49.21 | Penguin | Where extensions 1077 and 9898 dial the local devices. |
| 04:49.47 | Penguin | Sure, that's also a good idea. |
| 04:51.50 | Penguin | Now you can understand why the first thing I said was "fix your extensions." |
| 04:54.24 | moke | hmuerhm... |
| 04:54.38 | moke | sooo im actually a massive noob and im using freepbx. |
| 04:54.45 | Penguin | #freepbx |
| 04:54.51 | moke | heh. |
| 04:54.59 | moke | but the men in #asterisk are smarter |
| 04:55.00 | Penguin | If you're using FreePBX, this isn't the channel to ask for help. |
| 04:55.05 | Penguin | Everything I just told you is for asterisk. |
| 04:55.24 | moke | doesnt freepbx use asterisk? |
| 04:55.32 | Penguin | Yes, but you're not using asterisk. |
| 04:55.49 | Penguin | You're using freepbx, freepbx is using asterisk. You need to work on freepbx. |
| 04:56.10 | Samot | Custom Trunk |
| 04:56.20 | Penguin | So your question was misplaced and the advice I gave you is not relevant. This is the wrong channel. |
| 04:56.27 | Samot | Local/$OUTNUM$@from-trunk |
| 04:56.43 | Samot | Create outbound route with the DIDs that you wish to keep "local" |
| 04:56.59 | Samot | Point OB Route to use the new custom trunk. |
| 04:57.26 | Samot | Make sure all the extensions are using that route. |
| 04:57.41 | Samot | And make sure it's above the regular PSTN routing. |
| 04:59.33 | Samot | Penguin: The advice you gave can be done in FreePBX. |
| 04:59.44 | Penguin | I wouldn't know. |
| 04:59.50 | Penguin | doesn't use FreePBX. |
| 05:00.22 | Samot | It's still Asterisk. |
| 05:00.28 | Samot | It's still writing out conf files. |
| 05:06.36 | moke | Samot, thanks that worked perfectly. |
| 05:06.53 | Samot | np |
| 05:07.33 | moke | i set it to eat our entire DID block and strip away the first bits of the numbers, so all our external DIDs if called internally appear to call just the local ext. |
| 05:07.48 | moke | without hitting flowroute. |
| 05:08.33 | drmessano | Youre welcome |
| 05:09.03 | moke | if you're in Los Angeles, i will buy you beer, kind sir. |
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| 05:17.06 | *** join/#asterisk init_0 (~init_0@unaffiliated/init-0/x-7965738) |
| 05:18.28 | FarhaadN | where asterisk keep information about callcomplation ?? |
| 05:20.37 | FarhaadN | for example , how to know one extention use callcomplation and ,other active CC?? |
| 05:26.12 | Samot | Well there's only two |
| 05:26.19 | Samot | Busy and No Response. |
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| 05:29.18 | Samot | Based on what I'm reading it all looks to be in the logs. |
| 05:32.00 | Samot | FarhaadN: What's the goal? |
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| 05:38.48 | FarhaadN | Samot: i want to know what extension have a active CC |
| 05:39.29 | FarhaadN | asterisk dont keep any information about this information? |
| 05:39.46 | Samot | Well there's monitoring that happens. |
| 05:40.04 | Samot | Is this for extension to extension calling? |
| 05:42.10 | FarhaadN | yes |
| 05:42.49 | FarhaadN | i need know all information |
| 05:43.28 | FarhaadN | for example know about how secounds left for CC request |
| 05:43.33 | FarhaadN | and other things |
| 05:49.28 | Samot | Why aren't you using BLF? |
| 05:49.43 | Samot | How many extensions? |
| 05:50.17 | FarhaadN | BLF feature enable |
| 05:50.19 | FarhaadN | and use it |
| 05:50.32 | FarhaadN | 500 ext |
| 05:50.52 | FarhaadN | on every ext set 16 BLF key |
| 05:51.15 | FarhaadN | but for other calling whose ext dont monitor ,i need call back |
| 05:51.23 | FarhaadN | sry callcompletion |
| 05:52.11 | FarhaadN | and for some reason ,CC not work anytime |
| 05:52.35 | FarhaadN | i want to know about CC information that kept by asterisk |
| 05:52.50 | FarhaadN | for example when i cancele CC request (*41) |
| 05:52.59 | FarhaadN | CC work for that ext |
| 05:54.53 | Samot | I'm going to venture a guess as say it's a subscription. |
| 05:57.26 | FarhaadN | i not underestand meaning |
| 05:58.16 | FarhaadN | when i set 20s for cancel CC call request |
| 05:58.34 | FarhaadN | so asterisk kept this time |
| 05:58.49 | FarhaadN | and when CC request start, this timer goes down |
| 05:58.51 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 05:58.59 | Samot | I'm saying that when it's monitoring the state of the extension, it's using subscriptions |
| 05:59.10 | Samot | So "sip show subscriptions" |
| 05:59.24 | FarhaadN | ok |
| 05:59.26 | FarhaadN | let me check |
| 06:02.41 | Samot | I could be wrong about that |
| 06:07.32 | Samot | Did you find it? |
| 06:07.54 | FarhaadN | no |
| 06:08.09 | FarhaadN | there is no information about that in sip show subscriptions |
| 06:08.19 | FarhaadN | this is for BLF information |
| 06:08.36 | FarhaadN | and status of ext ( busy , idle , ringing ) |
| 06:19.44 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 06:21.04 | Samot | It's there. |
| 06:21.09 | Samot | *845150@ext-local : ccss:SIP/5150 State:InUse Presence:not_set Watchers 0 |
| 06:21.15 | Samot | That's how it looks. |
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| 06:23.28 | Samot | Sorry |
| 06:23.31 | Samot | core show hints |
| 06:23.35 | Samot | Not subscriptions |
| 06:24.11 | Samot | It doesn't show the expiry time left. |
| 06:24.24 | Samot | But is does show who has CCSS in user. |
| 06:24.28 | Samot | But is does show who has CCSS in use. |
| 06:24.47 | FarhaadN | no |
| 06:24.54 | FarhaadN | this was for hints |
| 06:25.09 | FarhaadN | hint for monitoring status of ext |
| 06:29.23 | Samot | *845150@ext-local : ccss:SIP/5150 State:InUse Presence:not_set Watchers 0 |
| 06:29.34 | Samot | For me *84 is the CCSS toggle. |
| 06:29.50 | Samot | As you can see the ccss:SIP/5150 |
| 06:30.24 | Samot | I called an extension I have, let it ring and did a CC request. |
| 06:30.46 | Samot | The state of *845150@ext-local when from Idle to InUse. |
| 06:31.23 | Samot | When I made a call on the extension 5150 called, the State when from Idle to InUse to Idle on that extension. |
| 06:31.35 | Samot | When it went from InUse to Idle, 5150 got a call back. |
| 06:32.11 | Samot | And the *845150@ext-local went from InUse to Idle once the call back was completed. |
| 06:32.18 | Samot | So it uses hints. |
| 06:32.30 | Samot | Perhaps [TK] can shed some light. |
| 06:33.59 | Samot | But I don't think you can see what the expiry time is. |
| 06:34.30 | Samot | You can see what extensions are using CCSS but not who they are waiting for the callback from, what time they requested and when it expires. |
| 06:36.36 | Samot | I take it not all the users are in the same location? |
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| 06:46.11 | FarhaadN | Samot: when i request the CC |
| 06:46.32 | FarhaadN | in "core show hint" nothing to show for CCSS |
| 06:46.39 | FarhaadN | but CC work |
| 06:47.35 | Samot | OK |
| 06:47.41 | Samot | core show hints |
| 06:47.43 | Samot | With an s |
| 06:48.01 | Samot | Otherwise it's core show hint <exten> |
| 06:48.58 | Samot | Well that gets you everything but the expiry time at least. |
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| 06:53.56 | FarhaadN | core show hint 100 |
| 06:53.56 | FarhaadN | <PROTECTED> |
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| 08:03.14 | FarhaadN | Samot: are u there? |
| 08:03.22 | FarhaadN | i find command for cc |
| 08:03.30 | FarhaadN | cc report status |
| 08:04.23 | FarhaadN | that's my i need |
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| 11:52.23 | goeij | accensyscYWDqVBh2R |
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| 13:00.59 | sikun | can a single IAX2 trunk be used as the destination for multiple incoming DAHDI DIDs? |
| 13:01.49 | [TK]D-Fender | Those 2 tthings have nothing to do witth each otther |
| 13:02.04 | [TK]D-Fender | You can treat any call arriving at your server however you wantt |
| 13:02.27 | sikun | I'm using an incoming route to route a DAHDI/POTS line over an IAX2 trunk to another PBX server |
| 13:02.41 | sikun | I'm asking if I can use the trunk for multiple incoming lines |
| 13:02.47 | Samot | Yes. |
| 13:02.51 | sikun | ok |
| 13:03.19 | sikun | Samot, thanks |
| 13:17.22 | [TK]D-Fender | What you do with your incoming calls is up to you. If you want to dial out to another peer then you can do whattever you want with any call to your system |
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| 13:18.15 | sikun | [TK]D-Fender, I just want to make sure I can utilize a single IAX2 trunk to handle multiple lines simultaneously or if I needed to make a trunk for each |
| 13:18.30 | [TK]D-Fender | sikun, Depends how your provider works |
| 13:18.45 | sikun | both PBX's are hosted in house |
| 13:18.55 | [TK]D-Fender | Actually since you said itès to another PBX (you control I assume), then it countts as one destination. |
| 13:19.03 | sikun | the one I'm working on is only dealing with the DAHDIs for incoming and routing them to the otuer |
| 13:19.06 | sikun | other&* |
| 13:19.14 | [TK]D-Fender | Tthere is only mulitple accounts if they tell you you have split calls for some reason and you donètt have a say |
| 13:20.02 | sikun | one our main sever, I have it set to look for the incoming DID and route it to the correct IVR just as if it was coming in from the SIP trunk |
| 13:20.15 | sikun | on our* |
| 13:20.18 | sikun | ugh, cannot type today |
| 13:20.42 | [TK]D-Fender | So if it's your server then you can do whatever you wantt |
| 13:20.55 | sikun | I know, I just wanted to make sure it was even possible to do |
| 13:21.23 | [TK]D-Fender | Why wouldn't it? It's a Dial() just like any other. |
| 13:22.11 | sikun | I don't know PBX administration very well and I just wanted to check, I wasn't sure. |
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| 13:53.01 | defswork | https://gist.github.com/anonymous/5d3ee32d64fe669324241803f0e2ea78 < anyone know what I am doing wrong here (just testing a fresh blank setup) |
| 13:53.41 | defswork | * version is 13.11.2 |
| 13:56.21 | Samot | Because 's' is not meant for a catch all. |
| 13:56.34 | Samot | You still need to match your digits. |
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| 13:59.02 | defswork | Samot, thats the first example dial plan in the * book :Oo |
| 13:59.17 | defswork | (btw I've tried it with explicit digits also |
| 14:00.24 | Samot | Show a call. |
| 14:00.28 | Samot | asterisk -rvvvvvvvvvv |
| 14:01.47 | defswork | first line in the gist is the only output |
| 14:01.54 | defswork | with verb 9 |
| 14:02.18 | [TK]D-Fender | PB the whole thing |
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| 14:47.51 | igcewieling | Does anyone know of Digium phones have the same brain dead utterly idiotic refusal to download externally modified contact directories as Polycoms? |
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| 14:48.25 | igcewieling | s/of/if/ |
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| 14:54.08 | zaf | infobot, you can build contact xmls for digiums if that's what you're asking |
| 14:54.21 | zaf | er, igcewieling |
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| 14:58.58 | igcewieling | zaf: I can do that with polycoms, but the phones are documented to not use externally generated contact directories. ever. 8-| |
| 14:59.43 | zaf | i can say that the digiums work |
| 15:18.39 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0-rc1 (2016/09/19); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 15:28.17 | file | go go rc1, power of... 14! |
| 15:30.05 | Nivex | any chance we'll see 13 in jessie-backports someday? |
| 15:31.54 | scv | thats up to debian really |
| 15:32.12 | scv | file: 3 days in and looks like it really was just pjproject being out of date |
| 15:32.14 | scv | very relieved |
| 15:32.21 | file | scv, oic |
| 15:32.43 | file | Nivex, we (the Asterisk project) don't control that - it would be up to the maintainer there |
| 15:33.36 | Nivex | ah ok. Their maintainers must be behind / burnt out. They used to have it for wheezy. |
| 15:39.37 | Nivex | ouch: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=816042 |
| 15:43.36 | igcewieling | hugs his compiled from source Asterisk |
| 15:44.15 | Nivex | I used to do that years ago. Somewhere along the line I got lazy :) |
| 15:44.45 | igcewieling | Nivex: Asterisk is the only thing I normally compile from source |
| 15:45.08 | igcewieling | I built a script to do it all automatically 8-) |
| 15:45.40 | Nivex | yeah, been there, done that |
| 15:45.55 | Nivex | looks like I may need to go back to it :/ |
| 15:47.11 | earlybird | Nivex: In July I attempted to work towards creating a new package for upstream but the time investment required for reviewing and maintaining all the custom Debian patches exceeded what was worthwhile for our local deployment so I dropped it. The situation is the same on Ubuntu (as they just pull from Debian): https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1596405. |
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| 16:51.16 | *** join/#asterisk anastmag (~anastmag@ppp-94-64-109-20.home.otenet.gr) |
| 16:51.38 | anastmag | Hi, i'm really new to asterisk and elastix. |
| 16:51.40 | *** join/#asterisk freebs (~freebs@unaffiliated/freebs) |
| 16:52.12 | anastmag | i want to implement a feature in my company's call center |
| 16:53.09 | anastmag | When an employee arrives in office he calls a number *94*XXX# in order to record that time |
| 16:53.31 | anastmag | and when he leaves, he do the same |
| 16:54.07 | anastmag | My question is how can i run a python script with a call and pass this called number ? |
| 16:54.12 | anastmag | ass an argument |
| 16:54.23 | anastmag | as an argument* |
| 17:05.40 | igcewieling | anastmag: contact Elastix. |
| 17:05.43 | igcewieling | ~distro |
| 17:05.43 | infobot | extra, extra, read all about it, distro is significa distribution (english) |
| 17:05.53 | igcewieling | ~freepbx |
| 17:05.54 | 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 |
| 17:06.08 | igcewieling | Elastix is even tougher to support. |
| 17:06.59 | anastmag | thank you |
| 17:07.05 | igcewieling | the issue is less the script and more how to integrate the script into the Elastix system. |
| 17:07.50 | igcewieling | you want to investigate Asterisk AGI |
| 17:08.41 | anastmag | I think that this is the trick (asterisk agi) |
| 17:08.54 | anastmag | thats why i joined #asterisk to ask |
| 17:19.41 | karver | anastmag, if you read-up on AGI it would be really simple to call an AGI script from your extensions.conf -- that script could be in Python, Perl, PHP, shell script -- doesn't matter |
| 17:20.05 | karver | that script then logs the argument you passed and a timestamp -- in a database or file or whatever you like |
| 17:20.55 | karver | good place to start: http://www.voip-info.org/wiki/view/Asterisk+AGI |
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| 17:51.58 | anastmag | is it possible to define a dynamic extension like \*94\*[0-9]*# ? |
| 18:12.42 | voipmonk | yep - get the book while you're at it - |
| 18:13.14 | voipmonk | ?book |
| 18:13.24 | voipmonk | !book |
| 18:16.10 | voipmonk | ~book |
| 18:16.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:24.27 | Nivex | So is PJSIP "The Way" going forward, or is it like AEL where it's kinda there and people are free to use it (and I've seen very few who do) ? |
| 18:25.02 | [TK]D-Fender | Nivex, Invalid comparison |
| 18:25.38 | [TK]D-Fender | Nivex, PJSIP is a whole separate stack with actually different auth and usability differences vs chan_sip |
| 18:26.05 | [TK]D-Fender | Chan SIP is also in "ignore mode" for new dev |
| 18:26.22 | Nivex | so chan_sip is essentially deprecated |
| 18:26.35 | file | as of Asterisk 13 it is community suppored |
| 18:26.38 | file | erm supported |
| 18:26.47 | [TK]D-Fender | AEL is a sad attempt to make a high-level syntax that still get's parsed back to standard dialplan logic so it can only do w lesser and worse job in the end |
| 18:27.14 | voipmonk | I've never heard it explained like that - but thats right on... |
| 18:27.19 | voipmonk | lol |
| 18:28.02 | Nivex | ok. I'm still running 11 because of the Debian packaging issue I mentioned earlier. I'm getting my head around what things I'll need to focus on refining once I make the jump to 13. |
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| 18:31.46 | file | chan_sip can still be used in 13 and 14 so you can migrate over later if you wish, just as [TK]D-Fender mentioned it doesn't see new development and except for community changes doesn't see much change |
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| 19:58.48 | jjrh | as someone who migrated to pjsip recently it's a fairly easy and comfortable transition |
| 19:59.07 | jjrh | I believe there is a script to convert chan_sip to pjsip too |
| 20:00.31 | igcewieling | file: do you expect chan_sip to be removed from Asterisk 15? |
| 20:00.37 | file | removed? no |
| 20:01.00 | igcewieling | file: thanks. |
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| 20:41.56 | t4nk608 | Hello. Trying to set up Asterisk with Twilio Trunk using PJSIP using the guide at https://www.twilio.com/docs/documents/14/AsteriskTwilioSIPTrunkingv2_0.pdf I can get inbound calling working, but outbound attempts result in [Sep 19 19:59:13] ERROR[6677]: chan_pjsip.c:1788 request: Failed to create outgoing session to endpoint 'twilio0' [Sep 19 19:59:13] WARNING[6729][C-00000000]: app_dial.c:2431 dial_exec_full: Unable to create ch |
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| 23:15.08 | pramsky | hello, how can I force a trunk to register ? I have the register , line in sip.conf and the peers defined . However I see 0 registrations and no attempts being made to register |
| 23:17.00 | [TK]D-Fender | The you have done it incorrectly somehow |
| 23:17.10 | [TK]D-Fender | Show us your actual configs masking only the secrets |
| 23:17.13 | [TK]D-Fender | ~pb |
| 23:17.14 | infobot | hmm... 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 23:17.15 | [TK]D-Fender | ^^^ |
| 23:19.13 | pramsky | got it |
| 23:23.04 | pramsky | http://pastebin.com/4KzGkrpr |
| 23:23.07 | pramsky | asterisk 13 |
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