| 00:04.49 | velix | Samot: Thanks, it finally works. |
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| 02:34.51 | velix | That's my manager.conf. Module is loaded and running. Anyone with an idea, why it doesn't listen? https://bpaste.net/ASTA |
| 02:46.52 | velix | ah, found it ;) |
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| 12:20.04 | zafu | hi, what file formats does Monitor() support? |
| 12:35.18 | Cyrillax | Looking at the source code it seems wav and gsm |
| 12:35.47 | Cyrillax | You could do like I always do an execute sox on the resulting wav and transcode to mp3 or whatever you want |
| 12:42.51 | Cyrillax | zafu: ^ (ping). best of luck, have a good day :) |
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| 13:32.08 | zafu | Cyrillax: thanks ! |
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| 14:47.53 | igcewieling | Monitor should support any format Asterisk can transcode to/from. |
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| 15:40.40 | DannyA | does anyone have experience with FollowMe causing a huge spike in CPU and loud static/crackling on the line when calling multiple lines at the same time? |
| 15:41.27 | drmessano | How many lines? |
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| 15:47.18 | Samot | 11:41:28 AM <drmessano> How many lines? <-- Oddly the standard question asked at Studio 54. |
| 15:49.31 | DannyA | 6 |
| 15:49.55 | DannyA | actually 7, 7 simultaneous channels dialing out at the same time |
| 15:50.41 | drmessano | Shouldn't be an issue unless the box is under-spec or the dialplan is doing something funky |
| 15:51.28 | Samot | Does this happen when only one line is called? |
| 15:51.58 | Samot | Is there point where there is no spike or static/crackling? |
| 15:57.40 | igcewieling | DannyA: so 7 incoming calls at the same time? |
| 15:58.06 | igcewieling | That is 14 calls, 7 call in and 7 calls out. |
| 15:58.54 | Samot | At most 8 |
| 15:58.57 | igcewieling | Using ulaw, I think that is a bit over 1Mbps of audio. Most any internet connection should be able to handle that. |
| 15:59.03 | Samot | It's FollowMe dialing 7 destinations. |
| 15:59.29 | DannyA | so here's the flow: Verizon DID over SIP into asterisk -> dispatcher activates custom feature (*28) which redirects the peer to an extension which acitvates followme, which dials 7 numbers at the same time over sip |
| 15:59.44 | igcewieling | ah, 1 call in, 7 calls out. that sounds like a ring group to me, but the term isn't all that inportant. |
| 15:59.48 | Samot | ulaw is 64kbps / 80kbps roughly with the overhead. |
| 16:00.08 | DannyA | @Samot yes. at the same time. not "ring my desk phone, and if i dont answer, then ring my cell phone, blah blah". im doing it to a group of people all at once, whoever answers and presses 1 first gets the call |
| 16:00.24 | Samot | That's how FollowMe works. |
| 16:00.47 | Samot | So again, does this happen with one destination/ |
| 16:00.49 | drmessano | Sounds like the dialplan is doing something more than just dialing |
| 16:00.54 | Samot | 2 destinations? |
| 16:01.03 | drmessano | something is causing the CPU spike |
| 16:01.29 | Samot | Well I'm going to guess there are local channels involved since an IVR for call confirmation is being used. |
| 16:01.50 | drmessano | Yup |
| 16:02.19 | DannyA | how would i know if local channels are involved? |
| 16:02.20 | Samot | DannyA: What are the specs for this server? CPU/RAM? |
| 16:02.27 | drmessano | Show us the dialplan |
| 16:02.30 | DannyA | i think its 4GB of ram |
| 16:02.33 | drmessano | and the specs |
| 16:02.38 | DannyA | its a VM |
| 16:02.43 | Samot | How many CPU? |
| 16:02.43 | DannyA | what specs can i tell you about it |
| 16:02.49 | DannyA | 2 i believe |
| 16:02.52 | drmessano | Show us the dialplan involved in the call |
| 16:02.55 | DannyA | k one sec |
| 16:05.33 | DannyA | https://pastebin.com/Aw6utM5y |
| 16:06.30 | DannyA | https://pastebin.com/GHy8WV3Q |
| 16:07.12 | igcewieling | Only mask passwords. |
| 16:07.31 | DannyA | not phone numbers? |
| 16:08.55 | igcewieling | phone numbers are not secret |
| 16:09.06 | drmessano | Are you using call recording? |
| 16:09.13 | DannyA | yes |
| 16:10.07 | drmessano | Who hosts the VM? Amazon? |
| 16:10.18 | DannyA | our own internal datacenter |
| 16:13.13 | DannyA | my IT guy is checking on the specs right now, i asked him to bump up the number of cpu's and ram |
| 16:13.18 | DannyA | what's ideal? |
| 16:13.24 | DannyA | (if there is such a thing) |
| 16:14.30 | drmessano | Depends on the specs of the underlying hardware |
| 16:14.42 | drmessano | Does the server have SSDs or magnetic drives? |
| 16:16.27 | DannyA | magnetic |
| 16:18.24 | drmessano | I would throw at least 2 CPUs and 8GB RAM at it |
| 16:20.09 | sibiria | for 7 calls? :P |
| 16:21.20 | sibiria | my ridiculously low-powered Arm-based SBC can keep 100 calls running simultaneously, with lots of dial plan functions being executed intermittently during |
| 16:21.39 | igcewieling | sibiria: are you recording the calls? |
| 16:21.43 | drmessano | Call recording |
| 16:21.58 | drmessano | Doesn't ANYONE read the scrollback |
| 16:22.14 | sibiria | i do actually, but that write-out is neither CPU nor disk intensive |
| 16:22.28 | drmessano | It's absolutely disk intensive |
| 16:22.33 | drmessano | It's all disk |
| 16:22.46 | drmessano | And he's got magnetic drives on an unknown spec server |
| 16:23.35 | sibiria | 160 kbit/s isn't a biggie for 5 separate output, nor for 10, nor for 50 |
| 16:23.47 | sibiria | not on a magnetic drive either |
| 16:24.10 | drmessano | Then why don't you help him resolve his CPU spikes |
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| 16:24.48 | sibiria | start with running an strace on asterisk to see where it spends most of its time |
| 16:25.04 | DannyA | regularly or during the followme? |
| 16:25.17 | DannyA | (bumping the specs anyhow now...in process) |
| 16:25.41 | DannyA | is there a different way to do call recording in RAM or something else? |
| 16:25.44 | sibiria | during the occasions where everything spikes - i presume you've verified that it is the asterisk parent process or one of its children spiking? |
| 16:26.14 | sibiria | yes. if you're running on linux you can mount a tmpfs file system and point the monitor dir there |
| 16:26.15 | DannyA | 1 CPU and 4GB ram currently |
| 16:26.35 | DannyA | @sibiria can u give me a pointer to an article about how to do that? |
| 16:27.04 | drmessano | 1 CPU and we still don't know the underlying specs of the server |
| 16:29.45 | sibiria | DannyA: start with making the mount point to the ram drive: mkdir /ramdrive; chown asterisk:asterisk /ramdrive |
| 16:30.13 | sibiria | then add to your /etc/fstab: tmpfs /ramdrive tmpfs rw,nodev,nosuid,noauto,noatime,size=500M 0 2 |
| 16:30.22 | sibiria | then, mount /ramdrive |
| 16:30.40 | sibiria | test the directory out to make it works and that it's consuming RAM |
| 16:31.09 | sibiria | e.g.: dd if=/dev/zero of=/ramdrive/blah bs=1M count=200M |
| 16:31.25 | sibiria | check your free memory, delete /ramdrive/blah, check free memory again |
| 16:31.32 | sibiria | there should be a ~200 mb difference |
| 16:31.59 | sibiria | then, edit asterisk.conf |
| 16:32.13 | sibiria | up at the top you have a stanza called [directories]. remove the (!) part on that |
| 16:32.47 | sibiria | then uh, hold on... let me check where the monitor dir is defined |
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| 16:38.57 | sibiria | looks like it's hardcoded, so you may need to point your mount point directly there |
| 16:39.03 | sibiria | to /var/spool/asterisk/monitor that is |
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| 16:45.55 | Samot | DannyA: You have the private emergency services company, right? |
| 16:45.59 | DannyA | yes |
| 16:46.07 | Samot | OK stop jerking around. |
| 16:46.21 | DannyA | excuse me? |
| 16:46.22 | Samot | 1 CPU / 4 GB RAM for that type of system is just wrong. |
| 16:46.31 | Samot | It's not enough. |
| 16:46.42 | DannyA | ok, its now 4 CPU and 8GB |
| 16:46.43 | Samot | So before you start doing things like recording into RAM, etc.. |
| 16:46.49 | Samot | Yes, now test it |
| 16:47.00 | Samot | Also you've never answered my question from before.. |
| 16:47.09 | Samot | Does this happen when you don't use 7 destinations? |
| 16:47.14 | Samot | Is there a breaking point? |
| 16:48.21 | DannyA | @Samot it does not happen when only using one destination |
| 16:48.36 | Samot | So then add another. |
| 16:48.38 | Samot | then another |
| 16:48.44 | DannyA | ok |
| 16:48.47 | Samot | At some point it will start to do what you are saying |
| 16:48.49 | Samot | Is that 7? |
| 16:48.51 | Samot | Is that 5? |
| 16:48.55 | DannyA | will test now |
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| 17:34.29 | Kobaz | ugh |
| 17:34.35 | Kobaz | i need to get my new group stuff into mainline |
| 17:34.50 | Kobaz | there's no standard functions to get channels in a group |
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| 17:48.33 | drmessano | DannyA: been an hour. Any results? |
| 17:49.50 | igcewieling | function GROUP() won't do it for you Kobaz? |
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| 17:54.27 | DannyA | sorry, was putting my son down for a nap. testing it now. |
| 17:59.27 | DannyA | [Mar 30 13:59:09] WARNING[39529][C-0000000d]: app_dial.c:1672 wait_for_answer: Unable to write frametype: 2 |
| 17:59.31 | DannyA | what do those mean? |
| 17:59.44 | DannyA | before the specs upgrade i was getting a TON of those with every FollowMe, now, just a couple |
| 18:04.54 | igcewieling | usually it means the far end of one of the channels dropped. Usually it means "caller hung up" |
| 18:08.31 | Samot | DannyA: What comes before it? |
| 18:09.05 | Samot | I'm guessing that appears when you test the FollowMe calls? |
| 18:09.22 | DannyA | https://pastebin.com/mXVh8jdw |
| 18:09.23 | DannyA | yes |
| 18:13.07 | Samot | DannyA: Is the static/garbled sound still an issue? |
| 18:13.07 | DannyA | so whats interesting is: it sounds fine now. BUT, when i call from my verizon did => adtran => asterisk, it sounds choppy now from the get-go. but when i call in from twilio DID, it sounds crystal clear, and followme is also perfect. |
| 18:13.11 | DannyA | (cc @Kobaz) |
| 18:13.19 | DannyA | @Samot yes, when using verizon it seems |
| 18:13.36 | Samot | For both inbound and outbound? |
| 18:13.40 | DannyA | but when using twilio, i just did two test calls, i didn't hear once ounce of choppiness |
| 18:13.40 | DannyA | yes |
| 18:13.48 | DannyA | wait, no, i didn't test outbound |
| 18:13.51 | Samot | Then it's probably the Adtran. |
| 18:13.56 | Samot | Or something Verizon is doing. |
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| 18:14.11 | Samot | Which is messing with the codecs/audio. |
| 18:14.45 | DannyA | i want to use Twilio/AWS. My IT guys insists on using Veirzon/In-house :( |
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| 18:23.47 | Kobaz | igcewieling: that's current groups on my own channel... no |
| 18:34.01 | igcewieling | Maybe ${GLOBAL(GROUP(blah,blah,blan))} |
| 18:34.10 | Kobaz | meh |
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| 18:34.37 | sid32 | Hi folks, Good evening |
| 18:35.32 | sid32 | How do i debug issues with my slave servers not being registered with the master servers [iax2]? |
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| 18:36.09 | sid32 | I've added appropriate entries to /etc/asterisk/iax.conf in both master as well as slave servers. |
| 18:38.36 | igcewieling | Sorry, I've not used IAX2 since about 2003. It lost the VoIP Wars, SIP won. Accept it. |
| 18:38.37 | Kobaz | igcewieling: no idea if GROUP() would work at a global level.... based on the docs it probably wouldn't work since it's channel based |
| 18:38.46 | igcewieling | It should. |
| 18:38.59 | igcewieling | otherwise, how would you put 2 different channels in the same group? |
| 18:39.14 | Kobaz | you call GROUP() each on its own channel |
| 18:39.23 | Kobaz | but to inspect... is another story |
| 18:40.17 | Kobaz | this is why I wrote GROUP_CHANNEL_LIST() and GROUP_MATCH_LIST_START() / GROUP_MATCH_LIST_NEXT() |
| 18:40.34 | Kobaz | sooo, yeap, time to get that stuff comitted |
| 18:41.28 | Kobaz | https://reviewboard.asterisk.org/r/464/ |
| 18:41.47 | sid32 | Well, in any case we are still fighting it out with IAX2. |
| 18:41.59 | sid32 | On master my slave IP appears like this |
| 18:42.12 | sid32 | <slave IP>:4569 N iaxconf <Unregistered> 60 Request Sent |
| 18:43.21 | sid32 | <master IP>:4569 N iaxconf <perceived IP>:4569 60 Registered |
| 18:43.50 | sid32 | What is this perceived IP anyway. I don't recognize it, and it is not my master IP address either. |
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| 19:45.48 | drmessano | DannyA: What about the CPU spikes? |
| 19:49.03 | Samot | drmessano: Good followup. That slipped my mind. |
| 19:49.29 | drmessano | This problem is all over the place |
| 19:50.22 | Samot | DannyA: Do all your outbound calls go out Verizon? |
| 19:50.40 | DannyA | checking cpu spikes now |
| 19:50.45 | DannyA | yes |
| 19:51.55 | Samot | And how bout those CPU spikes? |
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| 20:26.27 | drmessano | lol |
| 20:27.35 | drmessano | DannyA: At 12:46 ET you told us the VM was given more resources. 4 hours later, did it help with the CPU spikes? |
| 20:28.44 | drmessano | That to me was the most significant detail, and we've got zero followup data |
| 20:30.04 | DannyA | sorry, here we go, doing it right now.....calling.....top |
| 20:31.56 | DannyA | ok, so it started off at like 2% before i called |
| 20:32.40 | DannyA | then when the followme was activated, it jumped to 37%, then 108% for a second, then down to 60-something%, then for the duration of my phone call with the one person that answered, it stayed in the 30's%. when i hung up, it's now back down to 1.7% |
| 20:33.08 | DannyA | there's a "regular" phone call going on now, not using the hotline, = 3.3% currently |
| 20:34.45 | Samot | DannyA: Is it better, worse, the same? |
| 20:34.59 | DannyA | the call quality is better |
| 20:35.06 | DannyA | i didn't hear the same crackles and static |
| 20:35.11 | Samot | Even over Verizon? |
| 20:35.15 | DannyA | maybe just a drop here or there, but all around better |
| 20:35.16 | DannyA | yes |
| 20:36.08 | Samot | So 1 CPU was the issue. |
| 20:40.46 | DannyA | i think so. also per Kobaz's recommendation, and Siberia, im going to change mixmonitor to sln format and use ramdisk |
| 20:40.54 | DannyA | hopefully that will help with performance as well |
| 21:08.41 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 21:30.24 | defnotarobot | Anyone have experience configuring WebRTC with FreePBX/Asterisk? I'm trying to use sipML, and seem to be successfully connecting, but getting a "Not Acceptable Here" error when I try to dial a number. |
| 21:36.21 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 21:36.21 | *** mode/#asterisk [+o gtjoseph] by ChanServ |
| 21:51.48 | Kobaz | defnotarobot: sip debugs |
| 21:52.38 | Kobaz | use dpaste.com -- tcpdump -A -s0 -i interface ...etc etc |
| 21:53.43 | Kobaz | or asterisk console sip set debug on, or pjsip set logger on |
| 21:54.02 | Kobaz | dinner time |
| 22:05.05 | *** join/#asterisk hfb (~hfb@47.139.18.84) |
| 22:06.03 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:20.08 | defnotarobot | Kobaz: This what you're looking for? http://dpaste.com/2MMRR5J |
| 22:20.25 | defnotarobot | fingers crossed that I properly sanitized ips addresses and domains in that |
| 22:20.37 | *** join/#asterisk _abc_ (~usre@unaffiliated/ccbbaa) |
| 22:20.59 | _abc_ | Hello. Any idea why android clients do not register on *? Is there a special setting for androids? |
| 22:21.36 | [TK]D-Fender | _abc_, android isn't special |
| 22:21.39 | _abc_ | The androids can call * on wifi, but they never register. Other softphones have no trouble registering on the same * and wifi. |
| 22:21.49 | [TK]D-Fender | As for why yours is failing you should be looking at tyour SIP debug |
| 22:22.06 | _abc_ | I think there is a known bug in android sip stack, found several dozen links. No solution for my kind of problem. |
| 22:22.33 | _abc_ | I even set fixed ip's on the droids and set the wifi ip as it is, qualify claims they are not reachable |
| 22:22.49 | [TK]D-Fender | having a fixed IP has nothing to do with that |
| 22:22.58 | [TK]D-Fender | the fact you think it matters is problematic |
| 22:23.07 | [TK]D-Fender | Go look at it |
| 22:23.25 | [TK]D-Fender | "I think" is vastly inferior to "I'm actually looking at it" |
| 22:23.56 | _abc_ | Do I need pjsip at all for this? |
| 22:24.18 | [TK]D-Fender | either works |
| 22:24.36 | _abc_ | either? sip OR pjsip? |
| 22:24.43 | [TK]D-Fender | chan_sip or chan_pjsip |
| 22:24.56 | _abc_ | Ah I thought pjsip has extra registrar features |
| 22:25.03 | [TK]D-Fender | based on? |
| 22:25.12 | _abc_ | hunch? |
| 22:25.24 | [TK]D-Fender | Go look at your comms |
| 22:25.35 | _abc_ | What exactly should I look at please? |
| 22:25.36 | [TK]D-Fender | You're running on "think" and "hunch" |
| 22:25.43 | [TK]D-Fender | the register obviously |
| 22:25.50 | _abc_ | register? |
| 22:26.00 | [TK]D-Fender | .... |
| 22:26.01 | [TK]D-Fender | _abc_> Hello. Any idea why android clients do not register on *? Is there a special setting for androids? |
| 22:26.09 | [TK]D-Fender | YOU SAID YOU HAVE PROBLEMS REGISTERING |
| 22:26.15 | _abc_ | do not register means no registration yes |
| 22:26.28 | [TK]D-Fender | ... |
| 22:26.31 | [TK]D-Fender | and so.... |
| 22:27.44 | _abc_ | Now I get peer is not supposed to register?! |
| 22:28.01 | [TK]D-Fender | facepalms |
| 22:28.17 | Cyrillax | _abc_, enable sip debugging and show us the SIP messages being exchanged when the android client tries to REGISTER |
| 22:28.34 | [TK]D-Fender | FUCKING LOOK AT THE THING REGISTERING |
| 22:28.35 | _abc_ | AH, friend not peer |
| 22:28.39 | [TK]D-Fender | wrong |
| 22:28.46 | Cyrillax | .. |
| 22:28.48 | [TK]D-Fender | peer vs friend means NOTHING for this |
| 22:29.01 | [TK]D-Fender | Your hunches are worthless and you have no clue |
| 22:29.10 | [TK]D-Fender | go look at what it is SENDING |
| 22:30.23 | [TK]D-Fender | "My car won't start what else can I do while refusing to look under the hood?" |
| 22:32.09 | Kobaz | defnotarobot: and what's your endpoint that's having a problem. what's the ip/client uri |
| 22:32.56 | Kobaz | _abc_: 'android sip stack' ??? what is this.. what soft phone are you using |
| 22:34.43 | Kobaz | _abc_: ie: how does this relate to what you are using... does your soft phone use the android stack? or does it use its own, ie: pjsip |
| 22:34.51 | Kobaz | _abc_: sounds like you're making some assumptions |
| 22:35.15 | _abc_ | the soft phone I tried uses it's own stack twinkle. |
| 22:35.24 | _abc_ | It's own afaik. |
| 22:36.07 | _abc_ | qualify=10 f.ex. never qualifies the phones. qualify=no has them working without qual. |
| 22:36.24 | _abc_ | Both phones are pingable |
| 22:36.37 | _abc_ | Fromt the * |
| 22:36.57 | _abc_ | What does asterisk qualify use? ping? SIP status message? |
| 22:37.38 | [TK]D-Fender | SIP OPTIONS |
| 22:38.07 | [TK]D-Fender | and "qualify=10" is ridiculously stupid |
| 22:38.14 | _abc_ | So then what are the odds phone sip stacks do not support options in the way * expects them to be? |
| 22:38.19 | defnotarobot | Kobaz: The WebRTC client is represented by <xxx>.<xxx>.<xxx>.198 |
| 22:39.38 | defnotarobot | I doubt that any of the messages there are not related to the WebRTC activity, because I am the only active user until this is working |
| 22:40.13 | Kobaz | _abc_: sip debugs? |
| 22:40.36 | [TK]D-Fender | Kobaz, He's been told to look a dozen times just today |
| 22:41.07 | _abc_ | Kobaz: apparently it is the client(s) on android doing this. |
| 22:41.25 | Kobaz | <--- Transmitting SIP response (412 bytes) to WS:<xxx>.<xxx>.<xxx>.198:49823 ---> SIP/2.0 488 Not Acceptable Here |
| 22:41.27 | _abc_ | sipdroid has a known bug, sip android stack is crippled since 6 (I use a 4.4 and an 8.1 for testing here) |
| 22:41.28 | Kobaz | and what does asterisk say |
| 22:41.30 | [TK]D-Fender | First says "do not register", then starts talking about "qualify" |
| 22:41.40 | [TK]D-Fender | Kobaz, he's not looking |
| 22:41.41 | Kobaz | defnotarobot: do you have common codecs on the asterisk side and the webrtc side? |
| 22:42.29 | Kobaz | defnotarobot: what does asterisk say |
| 22:42.43 | Kobaz | defnotarobot: any codec errors/warnings? |
| 22:42.47 | defnotarobot | Kobaz: what does Asterisk say re: codecs? |
| 22:43.21 | defnotarobot | I have not seen any error messages on the asterisk side even with like 5 v's. |
| 22:43.32 | Kobaz | time for core set debug |
| 22:43.40 | defnotarobot | ok |
| 22:43.45 | file | is "792001" configured for WebRTC? |
| 22:47.55 | defnotarobot | file: So I am using FreePBX 14 on top of Asterisk and in the configuration for that extension in FreePBX, I did set Enable AVPF, Enable ICE support, and enable rtcp Mux |
| 22:50.05 | Kobaz | defnotarobot: #freepbx |
| 22:50.41 | Kobaz | #asterisk is for generic asterisk without massive systems running on top of them, who knows what specifics freepbx is throwing into asterisk |
| 22:51.27 | defnotarobot | Kobaz: have you tried joining #freepbx lately |
| 22:51.41 | Kobaz | no, but |
| 22:51.47 | defnotarobot | I'm the only one in there |
| 22:52.16 | Kobaz | i disagree |
| 22:52.20 | file | there's... 83 people in it |
| 22:52.26 | defnotarobot | Actually hold on that might be because I'm connected to multiple IRC servers with weechat |
| 22:52.30 | defnotarobot | good call |
| 22:52.49 | Kobaz | i see 83 people |
| 22:57.51 | defnotarobot | You are correct |
| 22:58.02 | defnotarobot | Thanks anyway |
| 23:01.06 | velix | I've seen some code using cdr_custom with ${CSV_QUOTE()}. How is it different to cdr_csv ? |
| 23:01.50 | [TK]D-Fender | one obviously allows for "custom" |
| 23:02.34 | velix | Yeah, but can cdr_csv not be customied? I think, you can set the fields etc. |
| 23:27.10 | velix | eww... ${CDR(uniqueid)}) <-- is not unique! |
| 23:29.50 | velix | oh it is, but ... it's hard to interpret. |
| 23:34.52 | sibiria | cdr_custom is what you use to customize that |
| 23:55.01 | _abc_ | on debian, dahdi must be built to match the current kernel from source, right? I can't find dahdi.ko in the packages for my release. |
| 23:57.34 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:58.17 | velix | sibiria: I mean, what can I customize, what I cannot do in cdr_csv? |
| 23:58.40 | [TK]D-Fender | You can simply add extra purely manual fields |
| 23:58.48 | [TK]D-Fender | And you should go read the docs on all of this |
| 23:59.16 | velix | [TK]D-Fender: I did. But I thought I could do this in cdr_csv also. Sorry. |