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| 00:23.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.1 (2018/09/20) 16.0.0-rc3 (2018/09/20), Standard: 15.6.1 (2018/09/20); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:00.44 | jamesaxl | hi |
| 08:01.05 | jamesaxl | have you ever hidden asterisk via Tor? |
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| 11:24.14 | Guest9103 | Hi, our asterisk 13.16 stopped receiving SIP traffic. I did a tcpdump and it was indeed reaching the server. When doing a netstat -peanut the udp value (I think it is the buffer?) was stuck on this value https://i.imgur.com/3R8qGJL.png |
| 11:24.24 | Guest9103 | What exactly is that value? Is it indeed the UDP buffer queue? |
| 11:26.18 | file | yes. |
| 11:31.53 | Guest9103 | I've seen this behaviour before, but I do not recall what actions I did. Any reason why this would occur? |
| 11:33.46 | file | deadlock? you haven't specified which channel driver is in use, so it's hard to answer with any specifics and your version is old enough that I do not recall such things from so far back |
| 11:37.09 | Guest9103 | chan_sip, I apologize. We are indeed using Asterisk Realtime will check for deadlocks. |
| 11:47.27 | Guest9103 | So something caused asterisk to stop receiving SIP, meaning it never "talked" to the UDP buffer and it filled up? Is this correct or am I out in the blue? I imagine that asterisk reads from the buffer and pulls packages from it, and if it stops it will fill up and start dropping udp traffic? |
| 11:47.55 | file | yes |
| 11:48.30 | Guest9103 | so it wouldn't really be a sysctl setting that caused this, but sometihng within asterisk |
| 11:49.13 | file | yes |
| 11:50.09 | Guest9103 | fantastic |
| 11:50.26 | Samot | You should update |
| 11:51.11 | Samot | Considering 13.16 has bugs |
| 11:55.24 | Guest9103 | Yes, I agree. Could this be cpu related? I see the server had quite a high load |
| 11:55.40 | Guest9103 | Rephrased: can big load cause this? |
| 11:55.52 | file | if the database stops responding or goes into a funky state then chan_sip can and will lock up |
| 11:59.23 | Guest9103 | What type of tests/investigations can I do when it occurs next time? |
| 12:00.54 | file | you can grab a backtrace, it'll tell you if it is stuck and where |
| 12:01.24 | Guest9103 | is that something you can enable on the fly or is a flag you have to start asterisk with? |
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| 12:03.55 | file | there are instructions on the wiki |
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| 12:25.20 | jkroon | has anyone else observe inexplicable load average spikes with asterisk on a host? Whereby it'll just jump to like 150+ in a second and then drop down over time ... I suspect something is causing all threads to wake up (almost) simultaneously and then the kernel just happens to sample at the right "wrong" time ... just want to be sure if this is normal and whether this is something I should be worried about. |
| 12:26.20 | Samot | You mean when you do the load average? |
| 12:26.55 | jkroon | as in /proc/loadavg values. |
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| 12:27.06 | jkroon | or the "uptime" command |
| 12:27.14 | Samot | The 5,10,15? |
| 12:27.46 | Samot | How many cores in the CPU? |
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| 14:39.42 | rjvvliet | Hi all, has anyone have the problem that the P-asserted-identity <USER> part becomes s@SIPDOMAIN whe using a Gosub. When the Dial progresses i get a 183 progress with the <USER> part (number) of the pai header changed to s which matches the diaplan extension wher the dial is executed. |
| 14:40.48 | rjvvliet | Result is that the Phone displays s on the display instead of the dialed number. |
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| 14:41.08 | rjvvliet | asterisk version 13.17.2 |
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| 15:07.40 | jameswf | that would be awful IMHO |
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| 16:05.57 | pa | what i don't fully understand is how it is possible that sip clients that are not even listed in my sip.conf can connect to my asterisk and place calls. Even i have troubles connecting using my accounts, when, fex, i have a wrong pwd, or using the wrong protocol, etc etc |
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| 16:08.41 | pa | f.ex. a client like: sip:209@<my ip> |
| 16:08.48 | pa | asterisk even spits a: |
| 16:09.15 | pa | Failed to authenticate device 209 |
| 16:09.48 | pa | obviously i misconfigured something |
| 16:09.59 | pa | or am using a vulnerable asterisk |
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| 16:27.54 | [TK]D-Fender | <pa> or am using a vulnerable asterisk <- not really a valid extrapolation |
| 16:27.59 | [TK]D-Fender | Show us the actual debug |
| 16:28.19 | *** join/#asterisk elyob (~elyob@92.91.9.51.dyn.plus.net) |
| 16:28.22 | pa | sure |
| 16:28.28 | pa | gimme a sec |
| 16:28.35 | pa | i have it very verbose tho |
| 16:34.24 | pa | [TK]D-Fender: http://paste.debian.net/hidden/0c800ed1/ |
| 16:34.31 | pa | but i guess you need my sip.conf too? |
| 16:38.51 | [TK]D-Fender | You haven't enabled SIP debug and you need to disable core debug |
| 16:38.55 | [TK]D-Fender | fix both and show a new call |
| 16:45.35 | pa | a call from a valid SIP client? because i am not able to place calls like this attacker did, from invalid SIP accounts |
| 16:46.04 | [TK]D-Fender | the same call with the proper level of debug |
| 16:46.15 | pa | i cannot reproduce that |
| 16:46.25 | pa | i donno how that was made |
| 16:46.33 | [TK]D-Fender | Ok, well we haven't seen anything useful to work with for this yet |
| 16:46.34 | pa | this is just the log i saved |
| 16:47.01 | pa | i can paste my sip.conf |
| 16:47.05 | pa | or |
| 16:47.09 | pa | in effect that's hard |
| 16:47.25 | *** join/#asterisk degenerate (~degenerat@S0106cc2de0099182.no.shawcable.net) |
| 16:47.26 | [TK]D-Fender | So is inventing evidence from past events. |
| 16:47.34 | pa | ? |
| 16:47.40 | pa | it's past from yesterday |
| 16:47.55 | pa | after which i fixed the firewall, that patched the problem |
| 16:47.58 | [TK]D-Fender | BTW, the Time Traveler's convention is last Thursday, don't miss it! |
| 16:48.17 | [TK]D-Fender | It will be/is awesome! |
| 16:48.19 | pa | but the question remained how could this guy place calls using non-existing sip accounts |
| 16:48.40 | pa | i guess i could re-open the firewall, and change the extension |
| 16:48.41 | [TK]D-Fender | you allowing unauthed calls would be a prime example |
| 16:48.44 | pa | to not get out |
| 16:48.57 | [TK]D-Fender | <PROTECTED> |
| 16:48.57 | pa | hm |
| 16:49.04 | [TK]D-Fender | You made a context labeled [default] |
| 16:49.09 | jameswf | Asterisk has no concept of trunks/extensions etc. Everything is simply an exte(sion) so people can make it do stuff if they are clever |
| 16:49.15 | [TK]D-Fender | that is psychotically dangerous |
| 16:49.34 | [TK]D-Fender | and you are allowing unathed calls... and they fall in the same place as your authed ones |
| 16:49.42 | jameswf | I thought [default] was _. hangup |
| 16:49.42 | pa | hm |
| 16:49.44 | [TK]D-Fender | this is pretty much assured from what you've shown |
| 16:49.57 | pa | where does one allow unauthed calls? |
| 16:50.10 | [TK]D-Fender | jameswf, no such thing as "default dialplan" :) |
| 16:50.14 | [TK]D-Fender | sip.conf <- |
| 16:51.10 | jameswf | https://www.irccloud.com/pastebin/JORuabpu/ |
| 16:51.26 | jameswf | damn why do we add ext-local |
| 16:51.44 | jameswf | will file a bug so someone can explain |
| 16:51.53 | pa | http://paste.debian.net/hidden/622984c8/ <-- this is my [general] section there |
| 16:52.36 | pa | then i have sections for each sip user i define |
| 16:52.55 | [TK]D-Fender | jameswf, ... we're in #asterisk .... not #happyguiland :p |
| 16:53.10 | [TK]D-Fender | context=default <- BAD |
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| 16:53.22 | [TK]D-Fender | you let every single call land int he place that lets them do EVERYTHING |
| 16:53.26 | [TK]D-Fender | That's crazy |
| 16:53.34 | [TK]D-Fender | allowguest=no |
| 16:53.40 | [TK]D-Fender | context=dieinafire |
| 16:54.20 | pa | so i would have to specifically set "allowguest=no"? 1) why isnt that default? 2) why am i myself unable to place calls as guest? have i tried the wrong way? what client could i use to test that? |
| 16:54.23 | [TK]D-Fender | and rename your [default] to something meaningful and point all your peers that need to to it |
| 16:54.53 | [TK]D-Fender | your test probably were invalid |
| 16:55.22 | [TK]D-Fender | and you can use any client that fills in blanks in the way you need for testing and with your server configured in the way you want to test the reactions from |
| 16:55.33 | pa | most likely, i would say. but what softphone could i try to place such a unauth call? |
| 16:55.44 | file | fun fact: it's not the default in chan_pjsip |
| 16:55.47 | jameswf | oh crap.... its monday right.... somewhere? |
| 16:55.49 | [TK]D-Fender | but what you've got here is a dangerous design with a series of consequences |
| 16:56.18 | [TK]D-Fender | <pa> most likely, i would say. but what softphone could i try to place such a unauth call? <- probably a bunch. not sure offhand which is easier than others |
| 16:56.30 | jameswf | in freepbx there was a guy who was doing something in the neighborhood |
| 16:56.35 | [TK]D-Fender | <jameswf> oh crap.... its monday right.... somewhere? <- SomeWHEN :p |
| 16:56.50 | file | jameswf: hello comrade |
| 16:57.00 | jameswf | time traveler |
| 16:57.02 | [TK]D-Fender | jameswf, probably just shooting some b-ball outside the school... |
| 16:57.16 | jameswf | ~jameswf |
| 16:57.24 | jameswf | damn bot |
| 16:57.32 | *** mode/#asterisk [+o tm1000] by ChanServ |
| 16:57.43 | *** mode/#asterisk [+o jameswf] by ChanServ |
| 16:57.48 | jameswf | uh oh |
| 16:57.54 | *** mode/#asterisk [+o GameGamer43] by ChanServ |
| 16:59.12 | file | one of us... one of us... |
| 17:15.27 | pa | meh.. can't find a mobile client that lets me set up an anon account |
| 17:16.20 | *** join/#asterisk Typhon (~Typhon@dslb-084-056-165-191.084.056.pools.vodafone-ip.de) |
| 17:17.10 | [TK]D-Fender | You've also just thrown the restriction of "mobile" into that... |
| 17:17.12 | [TK]D-Fender | but ok |
| 17:17.34 | [TK]D-Fender | Anyway, fix your config, it was clearly faulty and allowed what you saw happening |
| 17:19.58 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 17:19.59 | pa | can try desktop too |
| 17:20.13 | pa | started with the mobile clients coz it was easier to install :-) |
| 17:21.29 | [TK]D-Fender | And you're working from the shittiest and smallest of pools on the least configurable options :p |
| 17:21.45 | [TK]D-Fender | bett choices would be the most stripped down linux clients probably |
| 17:21.48 | [TK]D-Fender | better* |
| 17:23.15 | pa | hm true |
| 17:23.36 | pa | well i'm surprised you guys do not know one such tool |
| 17:24.13 | pa | :-) |
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| 17:28.30 | pa | [TK]D-Fender: so i tried "Twinkle", that apparently let me create a passwordless account. Yet, after i finish, i get a prompt saying "Login required for realm "asterisk"" |
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| 17:29.04 | pa | i used 1001 as user, like one of those calls in the log |
| 17:29.45 | pa | hm |
| 17:29.46 | pa | no |
| 17:29.49 | pa | actually you are right |
| 17:29.56 | pa | i could bypass it and place the call! |
| 17:34.16 | pa | so allowguests=no isn't enough.. |
| 17:34.19 | pa | wow |
| 17:34.57 | file | it's allowguest, not allowguests |
| 17:35.50 | pa | file: right, thanks. that seems enough indeed :) |
| 17:36.06 | pa | interesting default, though |
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| 18:25.04 | [TK]D-Fender | <[TK]D-Fender> allowguest=no |
| 18:25.04 | [TK]D-Fender | <[TK]D-Fender> context=dieinafire |
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| 19:33.41 | mahafyi_ | hello, I am thinking of hypermedia gsm gateway. Is this is agood brand? Which GSM gateways are most recommended for use with Asterisk? |
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| 21:08.45 | *** join/#asterisk Typhon (~Typhon@dslb-084-056-186-125.084.056.pools.vodafone-ip.de) |
| 21:08.58 | *** join/#asterisk rwb (~Thunderbi@65-183-138-202-dhcp.burlingtontelecom.net) |
| 21:52.04 | *** join/#asterisk n8n8 (~n8@208.69.84.142) |
| 22:06.36 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 22:12.25 | *** join/#asterisk Typhon (~Typhon@dslb-084-056-162-028.084.056.pools.vodafone-ip.de) |
| 22:18.55 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:25.07 | *** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net) |
| 22:53.24 | *** join/#asterisk elyob (~elyob@92.91.9.51.dyn.plus.net) |
| 23:00.34 | *** part/#asterisk kharwell (kharwell@nat/digium/x-mijeoyeeogalxkzx) |
| 23:17.41 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 23:26.01 | *** join/#asterisk giesen (~ggiesen@ego.giesen.me) |
| 23:26.21 | *** join/#asterisk Typhon (~Typhon@ipservice-092-218-198-130.092.218.pools.vodafone-ip.de) |
