| 00:00.32 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 00:24.49 | *** join/#asterisk t4nk350 (1896f506@gateway/web/freenode/ip.24.150.245.6) |
| 00:26.19 | t4nk350 | Was wondering if anyone knows what version of asterisk will work with a Digium TDM400P analog card? the current one will not work properly |
| 00:27.53 | WIMPy | Asterisk doesn't care it's DAHDI, but I don't know of anything having been dropped. |
| 00:31.18 | t4nk350 | hmm, no matter what i config, nothing wants to reconize this damn card. |
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| 01:02.59 | ChannelZ | you likely have some other config issue |
| 01:03.05 | ChannelZ | Does the machine see the hardware? |
| 01:03.49 | ChannelZ | IE does dahdi_hardware or dahdi_scan see anything? |
| 01:10.17 | *** join/#asterisk cyford33 (support@c-73-137-1-6.hsd1.ga.comcast.net) |
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| 01:21.06 | tjhowse | Has anyone here connected sipML5 to asterisk? I'm investigating frameworks for a browser-to-browser video intercom system for a hotel or residential tower. |
| 01:21.28 | tjhowse | I'm wondering if I need asterisk, it seems pretty heavy. |
| 01:27.27 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 01:35.48 | esaym153 | I am having issues figuring out how to connect my asterisk box to a PBX that uses a 3rd party provider and a different sip domain. This is what I have: http://pastie.org/private/t0zqs0ysburyj4jsv96jq |
| 01:35.58 | esaym153 | what I can't figure out is how to change "To: <sip:139@sip.sip-provider.co>" to "To: <sip:139@5159306467.com>" |
| 01:36.39 | esaym153 | I've never really had to deal with domains before, or at least not mis-matched ones |
| 01:37.01 | WIMPy | outboundproxy |
| 01:37.40 | esaym153 | WIMPy: set it under [work-phone]? and set it to what? |
| 01:38.10 | esaym153 | to the domain or provider host? |
| 01:38.30 | WIMPy | You set the host to what you need in the request and the outboundproxy to the host to send it to. |
| 01:39.30 | esaym153 | WIMPy: but the domain is not a real thing, it doesn't resolve, so I can't really have as a host then? |
| 01:39.59 | WIMPy | That's the idea of setting the proxy. |
| 01:40.45 | esaym153 | asterisk threw a fit, but let me try again.... |
| 01:42.11 | WIMPy | Oh, you probably need to add a ",force" to the proxy. |
| 01:43.23 | esaym153 | I'll try that. Right now it won't leave my local PBX and I get: WARNING[25148]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 01:43.34 | esaym153 | for the outbound call |
| 01:44.29 | esaym153 | same thing, with outboundproxy=my.proxy.com,force |
| 01:45.35 | esaym153 | dailplan has: Dial(SIP/work-phone/${EXTEN}) (among other things) |
| 01:45.35 | WIMPy | where my.proxy.com is the hostname of the server? |
| 01:45.50 | esaym153 | WIMPy: hostname of the provider server |
| 01:46.22 | WIMPy | Why do you call a provider something-phone, BTW? |
| 01:46.22 | esaym153 | but in sip reload, asterisk complains about the host=domain.com not resolving, so I don't think the peer is even added |
| 01:46.39 | esaym153 | yes, peer has unknown state |
| 01:47.16 | WIMPy | Yes, I think there was (is?) a bug where it tries to resolve the host even if it's not needed. |
| 01:47.28 | WIMPy | Check the sip debug. |
| 01:47.41 | esaym153 | WIMPy: I don't, I have another name. But the point is the same, this is just a PBX that an office uses and I am a remote employee |
| 01:47.43 | [TK]D-Fender | I'm sure it is needed which is why it's trying... |
| 01:47.54 | WIMPy | Also note that 'sip reload' doesn't alway do what you'd hope. |
| 01:48.12 | esaym153 | yea, I've had issues with reload, let me restart the whole thing |
| 01:49.07 | esaym153 | same thing, fail to get out of dailplan, peer is unknown state |
| 01:49.59 | esaym153 | maybe I should try in the dailplan to have the whole sip uri and ip/hostname at the end? |
| 01:50.07 | esaym153 | in extensions.conf |
| 01:50.39 | [TK]D-Fender | no |
| 01:50.42 | WIMPy | I doubt that makes it any easier (or more readable). |
| 01:50.45 | [TK]D-Fender | fix your peer |
| 01:51.18 | esaym153 | [TK]D-Fender: would really like to fix the peer, yes.... |
| 01:51.34 | esaym153 | but been at this for a couple of days now... |
| 01:52.24 | [TK]D-Fender | do it now. Show us now. |
| 01:52.36 | [TK]D-Fender | I'm not seeing actual configs and debug |
| 01:52.37 | *** join/#asterisk robink_ (~quassel@unaffilated/robink) |
| 01:52.39 | esaym153 | [TK]D-Fender: did you see my paste bin link? |
| 01:52.39 | [TK]D-Fender | that is not "trying" |
| 01:52.59 | [TK]D-Fender | full debug |
| 01:53.09 | [TK]D-Fender | I saw one packet..... |
| 01:53.18 | [TK]D-Fender | And no status dump |
| 01:53.23 | [TK]D-Fender | and no configs |
| 01:53.49 | [TK]D-Fender | <esaym153> I'll try that. Right now it won't leave my local PBX and I get: WARNING[25148]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) <- this says there's nowhere * can call and isn't going to even try |
| 01:54.05 | [TK]D-Fender | Show us the proper attempt and peer status dump |
| 01:56.12 | esaym153 | [TK]D-Fender: Will do... |
| 01:56.47 | [TK]D-Fender | is here for the next 5 minutes. |
| 02:06.00 | [TK]D-Fender | Time's up on my side. I'm out for a while |
| 02:13.26 | esaym153 | [TK]D-Fender: ok... |
| 02:13.34 | esaym153 | stuff coming up |
| 02:14.20 | esaym153 | [TK]D-Fender: still here |
| 02:14.26 | esaym153 | or anyone else wan to take a look? |
| 02:31.12 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-bpurnxwvaibvnrta) |
| 03:13.10 | esaym153 | [TK]D-Fender: ugh, this is what fixed it: Dial(SIP/workphone/${EXTEN}!${EXTEN}@4159206637.com) |
| 03:13.35 | esaym153 | the '!' at the end is what is shown in the To: field |
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| 03:14.12 | esaym153 | so after two days of messing with the, is this the right way? seems like if it was, it would not be such a hidden feature.. |
| 03:16.57 | igcewieling1 | . |
| 03:19.07 | igcewieling1 | esaym153: you can assume anything after the 2nd / is passed to the endpoint. |
| 03:19.36 | igcewieling1 | however none of that will fix a cause 20 |
| 03:21.31 | esaym153 | igcewieling1: 'cause 20' was caused by the help I go on here... |
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| 04:21.55 | babak | Hi, How to hangup an agent with AMI ? |
| 04:22.09 | babak | or cmd ? |
| 04:22.25 | babak | cli |
| 04:24.33 | WIMPy | 'action: hangup' or 'channel request hangup' |
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| 04:58.21 | esaym153 | igcewieling1: true anything after the 2nd/ will go to the endpoint, but the sip host will be appended to it. I need "To: 139@@4159206637.com" in the sip header, and that needs to go to some.proxy.com. Just sticking stuff at the end of the extension makes "To: 139@@4159206637.com@some.proxy.com", which will not work... |
| 04:58.37 | esaym153 | the '!' trick seems to work. Just don't know if that is the right way,... seems like a hack |
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| 05:15.33 | esaym153 | how to ignore the "Outbound Registration: Expiry for sip.provider.com is 60 sec (Scheduling reregistration in 45 s)" log message all up in my log files? |
| 05:16.03 | esaym153 | I noramally only use iax, just moved some stuff over to SIP and now I have all that in my logs and don't like it... |
| 05:17.24 | [TK]D-Fender | Disable logging |
| 05:17.34 | [TK]D-Fender | Because you're not going to get rid of just this one message alone |
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| 05:38.57 | esaym153 | [TK]D-Fender: have to just disable all logging ? probably shouldn't do that... |
| 05:39.18 | [TK]D-Fender | So live with the message |
| 05:39.23 | [TK]D-Fender | it isn't worth caring about |
| 05:39.51 | esaym153 | mainly want to know when I loose reg, or when people are trying to brute force (that is the only time I look at logs), so having a zillion re-reg in 45 seconds messages sux... |
| 05:40.45 | [TK]D-Fender | How is that a zillion in 45 seconds? |
| 05:40.56 | [TK]D-Fender | that looks like it should be ONE every 45 or so |
| 05:41.21 | [TK]D-Fender | And that's their defined interval. Go set a higher minimum and see if it sticks |
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| 05:48.34 | babak | WIMPy: Hi,thx for answers... befor disconnecting agent with AMI hangup I want to setVar channel variable for caller, how I find his channel? |
| 05:49.04 | babak | =caller |
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| 05:56.01 | esaym153 | [TK]D-Fender: yes, a message every minute is a lot :) |
| 05:57.00 | esaym153 | [TK]D-Fender: you watch asterisk-users@lists.digium.com? is my message: "SIP domain different than provider's", complete with giant log! |
| 05:57.10 | [TK]D-Fender | babak, There is a variable set that shows the channel it is bridged to |
| 05:57.13 | esaym153 | see my message* |
| 05:57.26 | [TK]D-Fender | I don't bother with mailing lists |
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| 05:57.40 | esaym153 | heh |
| 05:58.37 | esaym153 | [TK]D-Fender: link to post: http://tinyurl.com/pphc5zb |
| 05:58.57 | esaym153 | of my "now fixed in a hackish way" issues.... |
| 05:59.21 | esaym153 | got to eat, brb |
| 06:00.14 | [TK]D-Fender | Should have just fixed your peer... |
| 06:12.21 | tompaw | Guys, would this be considered an SDP mismatch for a proxy media? http://hastebin.com/nisupaqaji.hs |
| 06:12.54 | tompaw | There's Asterisk on one end, Flowroute on another, freeswitch in the middle. And no luck. |
| 06:13.01 | esaym153 | [TK]D-Fender: you keep saying 'fix peer', I am not following.... how do you recommend I fix it? |
| 06:13.55 | [TK]D-Fender | There is no parameter you can set on a dial you can't set in the peer itself |
| 06:15.21 | esaym153 | I would love to know which parameter |
| 06:16.20 | [TK]D-Fender | That's why there is a sample config... |
| 06:16.22 | [TK]D-Fender | read it |
| 06:16.28 | [TK]D-Fender | And on that note I'm off to bed... |
| 06:16.36 | esaym153 | ok :/ |
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| 06:26.04 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 06:57.42 | juned | Hi all |
| 07:00.05 | juned | I want add custom log file for my dialplan |
| 07:00.16 | juned | How do i do that ? |
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| 10:57.29 | Demon_VoIP | Good day. Please tell me were there any problems with AMI, which periodically breaks established connections (RST Flag in tcpdump packet). While in debug level 9 no logs about the disconnection or any problems. |
| 10:57.46 | Demon_VoIP | asterisk 11.7.0 (FreePBX) |
| 11:01.26 | Demon_VoIP | used socket connection with keep_alive and without it. No difference. In case of without keep_alive RESET of connection happens when the random next attempt to send the command. |
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| 12:23.40 | robmal | Hi, what happened to svnview.digium.com ? |
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| 12:29.22 | file | Asterisk was moved to git |
| 12:29.47 | tompaw | Morning. |
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| 12:38.49 | robmal | file: And what happened to the files there? Specifically i'm looking for adaptive odbc backport for 1.4 from http://svncommunity.digium.com/view/tilghman/branches/1.4/ |
| 12:39.08 | file | the SVN server remains up |
| 12:39.34 | file | although I have no idea about community... |
| 12:39.37 | file | that was so rarely used |
| 12:39.46 | robmal | :-/ |
| 12:42.23 | robmal | file: Maybe by chance you've got tilghmans e-mail? |
| 12:42.32 | file | I do not |
| 12:42.54 | robmal | That would be too easy ;-) Thanks for the clarification. |
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| 12:52.21 | dan_j | It's been a while since I last install asterisk. Where are the jansson devel packages? |
| 12:53.48 | tompaw | I usually build it from sources, as per tutorial. |
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| 12:53.55 | tompaw | le me check my ansible files |
| 12:54.17 | tompaw | git: repo=https://github.com/akheron/jansson.git |
| 12:54.33 | dan_j | Ok. I'll do that. Thought it might be on a yum repo |
| 12:54.57 | dan_j | Thanks |
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| 12:55.56 | tompaw | np |
| 12:56.21 | tompaw | Dear SDP experts, would this be considered a codec mismatch? http://hastebin.com/nisupaqaji.hs |
| 12:56.54 | tompaw | I have Asterisk talking to Flowroute via Freeswitch with media proxy. Or rather - trying to talk. |
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| 13:14.34 | pjensen00 | clear |
| 13:14.41 | pjensen00 | oops, this is not a console |
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| 13:15.55 | tompaw | I mean, we can all leave if you want ;> |
| 13:16.34 | pjensen00 | rm -rf tompaw |
| 13:16.38 | pjensen00 | hrm..... |
| 13:16.55 | tompaw | one day perhaps |
| 13:17.08 | pjensen00 | One day... all humanity will be rm -rf'd |
| 13:17.14 | pjensen00 | DOOMSAYER |
| 13:17.25 | tompaw | DOOM SLAYER sounds way more metal. |
| 13:17.34 | pjensen00 | \m/ |
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| 13:42.08 | dan_j | Does anyone have a script that can take mysql voicemail data and transfer them to imap storage? |
| 13:44.12 | phpboy | dan_j: please explain what you're trying to achieve |
| 13:44.44 | [TK]D-Fender | Pretty self-explanatory |
| 13:44.59 | [TK]D-Fender | He has his VM in MySQL ... he want to port it all over to IMAP. |
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| 13:45.34 | [TK]D-Fender | Now I'm trying to imagine who would have actually written a script for that, let alone actually be in-channel. |
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| 13:45.45 | [TK]D-Fender | I'd the odds are incredibly low. |
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| 13:48.36 | dan_j | Thought it would be worth a try. |
| 13:48.43 | dan_j | Thanks anyway |
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| 14:03.35 | [TK]D-Fender | I'd say* |
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| 14:25.56 | dan_j | [TK]D-Fender: In your opinion, whats the best method for sharing VM data between load balanced asterisk servers? I'm not happy doing it with mysql. |
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| 14:26.35 | dan_j | Oh, the asterisk servers could be relatively far from each other. |
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| 15:18.30 | robmal | file: Any chances you've got backups of svncommunity in some underground vault? Tilghman wrote in an e-mail he doesn't have any, since it was backuped on svn better than he could. Or maybe there is some other way to use custom fields in cdr mysql on 1.4.25.1 ? |
| 15:18.47 | file | I do not know |
| 15:18.50 | file | mjordan may |
| 15:18.57 | robmal | mjordan: Ping. |
| 15:19.14 | file | well, he may know |
| 15:19.37 | robmal | I'll keep my fingers crossed. |
| 15:35.08 | *** join/#asterisk asteriskATmarmuD (~mu@193.158.65.23) |
| 15:40.33 | asteriskATmarmuD | we get a lot of dialstatus congestion now, even before the call is placed on the carrier. so asterisk seems to block and return "congestion" because of some limitation. still on "Asterisk 1.4.39.2" here. will be updated, but we need a solution for now. any hints? |
| 15:41.14 | [TK]D-Fender | "look at the actual call |
| 15:41.19 | WIMPy | Maximum system load exceeded? |
| 15:41.22 | [TK]D-Fender | "some limitation" tells us nothing |
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| 15:45.58 | mjordan | robmal: Hm. |
| 15:46.32 | mjordan | I'm trying to recall where that SVN server was |
| 15:47.22 | asteriskATmarmuD | [TK]D-Fender: I know, I me neither :) I need to get to know the possible limitations |
| 15:48.13 | igcewieling1 | I didnt follow the whole conversation, but http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-1.4.39.2.tar.gz |
| 15:48.18 | mjordan | robmal: are you looking for something other than http://svncommunity.digium.com/svn/tilghman/branches/1.4/ |
| 15:50.05 | asteriskATmarmuD | [TK]D-Fender: that could cause such a behavior. or such a load. we added some IAX channels and more encoding has to take place. but this should not be a problem for the hardware |
| 15:50.54 | igcewieling1 | asteriskATmarmuD: a pastebin of the asterisk CLI showing the issue might be a good place to start. |
| 15:51.05 | igcewieling1 | If it is freepbx I can't help, but maybe someone else can. |
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| 15:53.13 | [TK]D-Fender | [11:47]asteriskATmarmuD[TK]D-Fender: I know, I me neither :) I need to get to know the possible limitations <- this is USELESSLY vague. "Could there be a thing doing stuff?!?!!" |
| 16:03.41 | asteriskATmarmuD | [TK]D-Fender: will hoepfully be back after further investigation |
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| 17:30.19 | robmal | mjordan: Only that, yes, i need to add custom fields to my CDRs in mysql on 1.4.25 and afaik cdr_mysql didn't support that until 1.6 so Tilghmans backport of cdr_adaptive is a game saver for me. |
| 17:30.38 | *** part/#asterisk asteriskATmarmuD (~mu@193.158.65.23) |
| 17:36.54 | babak | Hi, I want to play announcement you are connecting to number(say real number..). and Dial at same time, is it possible ? |
| 17:37.22 | babak | simultaneously |
| 17:43.22 | babak | I can make a wav file for my custom announcement, so the question is , is it possible to play custome wav file and Dial simultaneously ? |
| 17:44.47 | robmal | The easiest way i can think of is dialing SIP/1&SIP/2 where SIP/1 is 1,playback(tt-monkeys) and SIP/2 is queue(infinite-wait) |
| 17:48.01 | igcewieling1 | babak: you could set custom hold music if you want something played back DURING the dial. It is trivial to announce the number BEFORE the dial. |
| 17:49.25 | igcewieling1 | We use the same method to play international ringback when customers call internationally. |
| 17:51.21 | babak | igcewieling1: thx , I want more dynamic messages every call voice messages is different |
| 17:53.38 | babak | robmal : thx ,you mean SIP/1 plays my custom voice message without answer (early media) and SIP/2 is real destination? why queue ? |
| 17:55.06 | robmal | Because that's how i have it set up, my brain ignored your suggestion ;-) |
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| 18:14.25 | *** join/#asterisk jayka (6c1db782@gateway/web/freenode/ip.108.29.183.130) |
| 18:16.01 | jayka | Hello everyone! Does anyone have any experience tuning the linux kernel to optimize the Asterisk throughput? I am interested in interrupt frequencies, RTC, preemption... |
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| 18:34.51 | igcewieling1 | jayka: I've never needed to. |
| 18:35.39 | WIMPy | That should be part of the basic installation. |
| 18:35.40 | jayka | igcewieling1: Initially the kernel was compiled with the default 250 HZ and Voluntary preemption, and the cyclictest is currently showing a maximum latency of 5859, which is fine, but not optimal. Ideally, I'd like to have it under 3000, and that's why I decided to dig deeper. |
| 18:36.03 | igcewieling1 | I find the best solution to spending hours and hours optimizing a few percent more performance is to buy a better server and go so something useful 8-) |
| 18:36.25 | jayka | Shouldn't it be 1000 HZ with CONFIG_PREEMPT=y ? |
| 18:36.47 | igcewieling1 | jayka: I suspect most people use the kernel from their distro (in my case CentOS) |
| 18:37.32 | igcewieling1 | jayka: what problem are you trying to solve? |
| 18:38.58 | jayka | igcewieling1: we have a pretty big server farm, with many Asterisk boxes working as backends for our Kamailio cluster. I am just trying to cut the fat where possible, and make sure that our latencies don't degrade as we expand further |
| 18:39.20 | WIMPy | It's always a good idea to remove stuff you don't need. Size does matter! |
| 18:40.20 | igcewieling1 | what exactly is "cyclictest"? |
| 18:40.49 | jayka | http://kb.digium.com/articles/Configuration/How-to-perform-a-system-latency-test https://rt.wiki.kernel.org/index.php/Cyclictest |
| 18:41.02 | igcewieling1 | Ah, OK. |
| 18:41.08 | igcewieling1 | I thought you were referring to |
| 18:41.35 | igcewieling1 | I thought you were referring to 'timing test' or 'core show translation' |
| 18:42.14 | igcewieling1 | I any case, I wish you the best of luck. |
| 19:05.16 | SFJulie1 | jayka, have you tried playing with syscontrol? |
| 19:05.40 | SFJulie1 | stuff life tcp behaviour, memory_overcommit, swappiness? |
| 19:06.15 | SFJulie1 | would profile asterisk before finding to optimize, but /me is not a sip person |
| 19:06.20 | SFJulie1 | *trying |
| 19:11.38 | igcewieling1 | What system load do you usually see on the asterisk boxes? |
| 19:14.19 | igcewieling1 | system load of around .25 with 50 active calls, though they are all sip and we use hardware for transcoding. |
| 19:17.57 | jayka | SFJulie1: no, haven't tried those. any particular suggestions that have proven beneficial for throughput? |
| 19:19.47 | jayka | SFJulie1: AFAIK 1000 hz + preemption is a pretty common setting for telephony apps, so decided to start with the basics |
| 19:22.05 | SFJulie1 | jayka, I honestly don't know the profile of your application (I am much more a backend dev of SIP) there are sysctl that can decrease latency by changing the behaviour of network stack (scaling, buffer cache, slab allocations), of memory management (favouring swappiness) ... |
| 19:22.53 | SFJulie1 | I use a lot of measure for our asterisk and I have custom probes (open fd, interrupt, ctx swithc) per class of process) for in case we need more measures |
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| 19:37.03 | kplong | Is it possible to enable SIP debugging on 1 or 2 extensions only, logging these messages to a unique log file |
| 19:43.08 | [TK]D-Fender | 1 yes, 2 no |
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| 19:43.22 | [TK]D-Fender | and not to "unique" log |
| 19:45.49 | robmal | Why not? If you sip set debug peer 123 and in logger.conf set debug to some file you'll only get that peers sip trace. |
| 19:46.39 | [TK]D-Fender | That's not a "unique" log", and that is only ONE |
| 19:48.10 | robmal | I love how you cling to choice of words. |
| 19:50.25 | [TK]D-Fender | Someone asks for A you don't give them B instead and try to pawn it off. |
| 19:51.07 | robmal | Sure, maybe my not-so-perfect answer will lead him somewhere. |
| 19:51.23 | robmal | Maybe his question was not-so-perfect ;-) |
| 19:51.44 | [TK]D-Fender | Or maybe your suggestion doesn't give him what he wants. |
| 19:52.09 | robmal | That's an option too. |
| 19:52.25 | [TK]D-Fender | So far it's what we've got with his words and your words. |
| 19:54.00 | robmal | Well, at least i hope he's shifting from 'not possible' to 'maybe i want too much at once' |
| 19:56.15 | [TK]D-Fender | how does that even work? |
| 19:56.41 | [TK]D-Fender | Being possible and "too much" do not intersect beyond the precise point of being "not possible". |
| 19:56.53 | [TK]D-Fender | Which makes the other redundant. |
| 19:58.40 | robmal | Sure, but 'too much' with clarification where 'this isn't too much, just debug one peer at a time' is on the line between 'not possible' and 'sure' is usually enough for someone to start asking the right question. |
| 19:59.32 | [TK]D-Fender | I fail to see how his question is wrong. |
| 19:59.47 | robmal | And again. |
| 19:59.53 | robmal | Well played ;-) |
| 20:00.01 | [TK]D-Fender | Who is to judge that his expectation is "too much". |
| 20:01.08 | robmal | Well, the question is not wrong, the interpretation is subjective, so for you it's 'not possible', for me it's 'try making smaller steps' |
| 20:01.27 | robmal | And nobody is wrong. |
| 20:02.18 | [TK]D-Fender | He asked for multiple specific peers. that is not happening with Asterisk. The feature list is documented. It's not there. That's pretty much final. The code we have does not do it. |
| 20:02.50 | [TK]D-Fender | There are no smaller steps. The pieces don't combine for this. |
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| 20:06.27 | robmal | Well, there are options you don't want to acknowledge. He can set sip debug on, then remove debugs from extensions he doesn't want to debug. He can tcpdump with specific hosts. If it's not a production site he can just force unregister the peers he doesn't want debugged. |