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| 09:38.28 | BERKAU | Hello! |
| 09:40.09 | BERKAU | is anybody online? |
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| 09:49.50 | dadrc | someone probably is |
| 09:51.30 | BERKAU | yes, but nobody writes anything. How does it work here? |
| 09:51.37 | jaflong | Hi guys, Cany anyone suggest an company that provides bulk ddi numbers for uk. |
| 09:51.51 | dadrc | BERKAU: if you've got a question, just ask |
| 09:52.25 | BERKAU | I'm using Asterisk 13, and am attacked by brute force attacks. What can I do against it? |
| 09:53.26 | dadrc | there's always fail2ban |
| 09:53.32 | dadrc | I like fail2ban |
| 09:54.05 | dadrc | (there's documentation on f2b and asterisk here: http://www.fail2ban.org/wiki/index.php/Asterisk) |
| 09:54.27 | BERKAU | yes, but I read the following: |
| 09:54.28 | BERKAU | SECURITY NOTE: fail2ban is rather limited in its ability to detect attacks against asterisk. |
| 09:54.53 | BERKAU | so there could be other attakc that fail2ban can't detect? |
| 09:55.07 | dadrc | of course. there's always another attack. |
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| 09:55.56 | BERKAU | I've read about SecAst, but I'm afraid it isn't compatible with asterisk 13 is it? |
| 09:56.03 | dadrc | but unless they⢠are targeting you specifically, it's probably some run-of-the-mill dime-a-dozen standard attack script. |
| 09:57.42 | dadrc | The SecAst dudes claim they only support Asterisk 1.4 to 12, I'm not in a position to claim otherwise :) |
| 09:59.01 | BERKAU | so for security reasons it'll be better to use an older asterisk version? But I thought, Asterisk is very concerned about security. So what is the alternative to SecAst in Asterisk 13? |
| 10:01.09 | dadrc | Check fail2ban, the new log in asterisk 10+ makes it way more effective |
| 10:04.01 | BERKAU | ok :) But I've already installed fail2ban, and it didn't work. There was a massive brute-force attack this morning (and earlyer) with several connectiontries per second. |
| 10:08.12 | BERKAU | I'd like to reconfigure fail2ban, so that it stops these brute-force attacks. |
| 10:09.03 | BERKAU | But how can I try if this reconfiguration has worked properly, I don't want to wait until the next attack. |
| 10:16.41 | dadrc | You can run `fail2ban-regex` with logs and configs |
| 10:19.46 | BERKAU | oh, I'll try that. :) |
| 10:23.35 | BERKAU | how does fail2ban work? it takes a long time to execute this command. I think that's because the logfile is really huge. |
| 10:24.36 | BERKAU | does it read the logfiles every x minutes like fail2ban-regex does, or does it only wait for new loglines in the file? |
| 10:28.37 | dadrc | I believe f2b uses (can use) inotify to watch for changes in files |
| 10:31.30 | dadrc | In any case, f2b does not parse the entire log file every time a line gets added. That'd be insane. |
| 10:51.27 | BERKAU | when i use fail2ban-regex with my old config file I get the following error:ConfigParser.InterpolationMissingOptionError: Bad value substitution: |
| 10:51.27 | BERKAU | <PROTECTED> |
| 10:51.27 | BERKAU | <PROTECTED> |
| 10:51.27 | BERKAU | <PROTECTED> |
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| 10:52.22 | BERKAU | so that would propably have been the reason why fail2ban didn't work. |
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| 11:10.39 | WIMPy | Adds another meaning to "fail2ban"... |
| 11:19.16 | BERKAU | how do I have to configure the mailserver so that I get informed about attacks by fail2ban? |
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| 11:32.47 | jvhester | BERKAU: you may have better luck in the #fail2ban channel for some of those more f2b specific questions |
| 11:38.40 | jvhester | That being said you are correct to "not want to wait for the next attack" to test out your security tools. |
| 11:38.53 | jvhester | Have you considered simulating an attack on a very small scale to verify? |
| 11:51.37 | BERKAU | yes, but I don't know how to do that. |
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| 12:04.22 | BERKAU | could you please tell me how I can do a test like that? |
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| 14:29.11 | marceloamorim | guys, I think I have a stupid question, this is the first time I register my sip phone with MAC as default user like [00AABB_CCDDEE] everything is fine until I need to use the SIPPEER(00AABB_CCDDEE,status) |
| 14:30.17 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-fqhzrqozbxfkhuxu) |
| 14:31.04 | [TK]D-Fender | Show us |
| 14:33.04 | marceloamorim | http://pastebin.com/r1yaPhkL |
| 14:35.31 | marceloamorim | in internal:8 and :9 I use to debug and I couldn't get the return, one my extension line 23 and 24 . the variable ${ramal} is fine but the SIPPEER isn't =( |
| 14:35.46 | *** join/#asterisk mondkalbantrieb (~quassel@sama32.de) |
| 14:37.07 | [TK]D-Fender | Amd the peer now... |
| 14:37.09 | [TK]D-Fender | and* |
| 14:37.27 | marceloamorim | ? |
| 14:37.34 | marceloamorim | users.conf? |
| 14:38.00 | [TK]D-Fender | .... you're not seriously using that file are you? |
| 14:38.17 | marceloamorim | why? its so cool |
| 14:38.21 | [TK]D-Fender | it's GARBAGE |
| 14:38.29 | [TK]D-Fender | ~users.conf |
| 14:38.29 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 14:39.19 | [TK]D-Fender | but wsince you're there already, show us the config and the peer dump |
| 14:39.28 | [TK]D-Fender | And get off of that trash ASAP |
| 14:40.40 | marceloamorim | damn =( I will fix this then, I used this on too many systems o.O |
| 14:41.11 | marceloamorim | but do you think this is the problem? should I try the sip.conf first? |
| 14:41.33 | [TK]D-Fender | let's LOOK first |
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| 14:44.39 | marceloamorim | http://pastebin.com/9ti54zmt |
| 14:47.17 | [TK]D-Fender | 23 same => n,Verbose(0,verb2 -> (${SIPPEER(001A3F04_F6F80,status)})) |
| 14:47.22 | [TK]D-Fender | 186 [001A3F04_F6F801](default_user) |
| 14:47.27 | [TK]D-Fender | Missing a DIGIT there |
| 14:47.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:47.53 | [TK]D-Fender | 001A3F04_F6F80 != 001A3F04_F6F801 |
| 14:48.25 | marceloamorim | oh, damn, my sql there is a limit varchar |
| 14:48.30 | marceloamorim | this is why =( |
| 14:48.49 | marceloamorim | I missed that, ty sir |
| 14:49.37 | [TK]D-Fender | ALWAYS prove taht you are actually looking up something valid. |
| 14:49.57 | marceloamorim | I thought it was a kind of bug os something using the underscore |
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| 16:13.41 | Encrypt | Hi! o/ |
| 16:13.55 | Encrypt | I'm forwarding ports on my router for asterisk |
| 16:14.06 | Encrypt | Apparently, I need to forward RTP ports |
| 16:14.34 | Encrypt | Right now, rtp.conf show 10000 to 20000 |
| 16:14.38 | Encrypt | shows* |
| 16:14.49 | Encrypt | Can I remove ports? |
| 16:14.57 | Encrypt | I will be the only one to use asterisk outside |
| 16:15.12 | Encrypt | So, if I have understood, 4 ports are needed for audio |
| 16:15.28 | igcewieling1 | Encrypt: yes. however, you don't normally need to port forward RTP. The outgoing RTP from the server will create the forwarding for replies |
| 16:15.33 | Encrypt | Can I set rtp ports from 10000 to 10100? |
| 16:15.50 | igcewieling1 | Encrypt: you can set them to anything you want. each call will take 2 - 4 ports. |
| 16:15.59 | Encrypt | igcewieling1, I have asked my neighbors for their WPA WiFi key |
| 16:16.16 | Encrypt | And I have no sound right now, without rtp ports forwarded |
| 16:16.34 | igcewieling1 | Encrypt: maybe you won't have audio with rtp ports forwarded too. |
| 16:16.36 | Encrypt | Asterisk shows: [Sep 14 18:12:42] WARNING[2772]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 1952769866 for seqno 21 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 16:16.36 | Encrypt | Packet timed out after 32000ms with no response |
| 16:16.53 | igcewieling1 | That's NAT issue and also SIP not RTP |
| 16:16.59 | Encrypt | Hum |
| 16:17.18 | Encrypt | igcewieling1, What can I do then? :S |
| 16:17.34 | igcewieling1 | maybe a localnet / externip issue? |
| 16:19.03 | Encrypt | igcewieling1, Would the nat=yes option on the peer solve my problem? |
| 16:19.21 | igcewieling1 | Encrypt: nat is complicated. it might solve your issue, it might not. |
| 16:19.33 | igcewieling1 | thankfully I live in a world where NAT is rare. 8-) |
| 16:20.29 | Encrypt | :D |
| 16:20.29 | igcewieling1 | chances are you also need localnet and externip. go read up on NAT and Asterisk |
| 16:20.35 | Encrypt | Ok |
| 16:20.44 | Encrypt | igcewieling1, And just to be sure, if I have understood... |
| 16:20.52 | Encrypt | According to you, no need to forward RTP ports? |
| 16:21.13 | igcewieling1 | Encrypt: usually no need to forward ports. Are the phone local to your network or external to your network? |
| 16:22.29 | *** join/#asterisk karelk (~karel@31.10.145.202) |
| 16:23.08 | igcewieling1 | if you have an asterisk server behind nat and it registers to an outside provider and you have no outside phones, qualify=yes, and qualifyfreq=30 and then you often don't need to create a port 5060/udp forward either, the outgoing registration creates the NAT reply setup. |
| 16:23.38 | Encrypt | igcewieling1, Here is my set up: |
| 16:24.29 | Encrypt | My raspberry pi --> My internet "box" with NAT --> internet --> Another internet "box" with NAT --> My laptpn with asterisk |
| 16:24.33 | Encrypt | laptop* |
| 16:24.47 | igcewieling1 | Encrypt: good luck with that. |
| 16:25.01 | Encrypt | igcewieling1, That's really that difficult? :/ |
| 16:25.46 | igcewieling1 | Encrypt: far more trouble than I'm going to help troubleshoot. |
| 16:25.48 | *** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net) |
| 16:25.58 | Encrypt | Ok... :/ |
| 16:26.02 | Encrypt | Thanks anyway :) |
| 16:26.33 | igcewieling1 | double nat is a pain in the ass no matter what you are doing. |
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| 16:38.24 | igcewieling1 | Encrypt: you might consider: softphone -> local asterisk box -> nat router -> VoIP provider like Vitelity |
| 16:38.35 | igcewieling1 | that is a very common setup |
| 16:38.50 | Encrypt | Ok |
| 16:38.54 | igcewieling1 | You could also put a VPN between the two sites and avoid NAT |
| 16:38.59 | Encrypt | The thing is, I have a GSM gateway at home |
| 16:39.11 | Encrypt | igcewieling1, That's what I'll end up doing I think... |
| 16:39.27 | *** join/#asterisk airjump (~Thunderbi@p20030070CE32517F44EEFA5CC5E652E8.dip0.t-ipconnect.de) |
| 16:39.53 | igcewieling1 | Encrypt: VoIP calling is often 1/cent/min or lower. |
| 16:40.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:51.41 | Encrypt | igcewieling1, Thanks for your help, I'm going offline :) |
| 16:52.08 | Encrypt | I'll probably connect directly inside my network with a VPN tool |
| 16:52.11 | Encrypt | That will be easier |
| 16:52.21 | Encrypt | Bye! o/ |
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| 18:00.30 | HereIamAgain | Hello, Im trying SIP over TLS. Clients are connecting without problems, yet when placing a call I get 2 errors from debug: |
| 18:00.38 | HereIamAgain | 1. sip_endpoint.c Error processing packet from :58780: Rx buffer overflow (PJSIP_ERXOVERFLOW) [code 171062] |
| 18:01.08 | HereIamAgain | ERROR[32064]: pjsip:0 <?>: sip_transport. Error processing 2705 bytes packet from TLS : PJSIP syntax error exception when parsing 'Request Line' header on line 1 col 2: |
| 18:01.08 | HereIamAgain | x=[0-1280],y=[0-800]] |
| 18:41.27 | newtonr | Oh, he quit |
| 18:41.31 | newtonr | Haven't seen that error |
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| 19:08.50 | igcewieling1 | it has now been 1 week since I turned off my cellphone. yay! |
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| 19:18.16 | igcewieling1 | <-- paranoid, not retro. |
| 19:19.13 | mjordan | It isn't paranoia if they really are out to get you. |
| 19:19.30 | WIMPy | Did you use a car? |
| 19:21.13 | [TK]D-Fender | mjordanIt isn't paranoia if they really are out to get you. <- it is if you don't actually have the supporting evidence ;) |
| 19:21.48 | [TK]D-Fender | is meta FO DAYS YO |
| 19:24.02 | mjordan | ;p; |
| 19:24.04 | mjordan | er |
| 19:24.05 | mjordan | lol |
| 19:24.12 | mjordan | hand on keyboard position was off by 1 |
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| 19:34.15 | igcewieling1 | I consider snowden enough evidence |
| 19:34.44 | igcewieling1 | I miss the timer and the calculator most 8-| |
| 19:39.10 | *** part/#asterisk igcewieling1 (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 19:41.51 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 20:40.30 | *** join/#asterisk mhjkl (~mh100@host-92-11-160-72.as43234.net) |
| 20:42.16 | mhjkl | Hi, I am trying to work out how to dial 3 external telephone numbers (via SIP trunk) simultaneously and the first receving phone to pick up to be connected and the other 2 calls to be dropped. Is this possible? |
| 20:42.57 | WIMPy | That's what happens if you Dial multiple destination. |
| 20:43.00 | robmal | Like dial(sip/1&sip/2&sip/3) ? |
| 20:43.19 | WIMPy | yes |
| 20:43.50 | mhjkl | yes I think that is what I am looking for robmal |
| 20:44.02 | robmal | :-) |
| 20:44.22 | mhjkl | where can I find out more about the syntax for directing towards a SIP trunk |
| 20:44.55 | mhjkl | and how do I set this up in FreePBX |
| 20:45.04 | robmal | Aaaaand you blew it ;-) |
| 20:45.10 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 20:45.29 | robmal | In about 30s someone will point to you that freepbx is not supported here. |
| 20:45.54 | robmal | I'll use that time to point you to wikipedia for asterisk, everything's there. |
| 20:46.14 | robmal | Almost everything. No porn. |
| 20:47.20 | aandrew | ... asterisk porn? I know about rule 34, but still... |
| 20:48.01 | robmal | I'm pretty sure they have porn hidden somewhere but i couldn't find it yet, so, for now, no porn. |
| 20:48.33 | WIMPy | I'm sure it's often used for porn. |
| 20:49.23 | aandrew | IIRC normast was using it for ... lavalife? |
| 20:51.44 | *** join/#asterisk bmg505 (~leon@196-215-18-102.dynamic.isadsl.co.za) |
| 21:04.20 | mhjkl | @robmal, In Asterisk where would I enter dial(sip/1&sip/2&sip/3) |
| 21:04.48 | mhjkl | Is sip in the above command the name of the outbound trunk? |
| 21:06.47 | mjordan | mhjkl: Asterisk doesn't have a concept of a trunk. |
| 21:06.57 | robmal | It has begun. |
| 21:07.01 | mjordan | mhjkl: You would pass that as the dial string to app_dial. |
| 21:07.05 | mjordan | or rather, Dial. |
| 21:07.06 | *** join/#asterisk pgrace (~pgrace@108.161.133.150) |
| 21:07.27 | pgrace | Has anyone run into a situation with 1.8.32 where calls to MixMonitor() silently fail, never actually creating a file? |
| 21:11.07 | mjordan | mhjkl: https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 21:19.19 | *** join/#asterisk adeeln (~adeeln@2602:ffc1:1:face:8dcb:2afd:a078:403) |
| 21:26.32 | mhjkl | Can I achieve what I am looking to do by creating an appropriate .call file and placing it into /var/spool/asterisk/outgoing |
| 21:26.53 | adeeln | i'm encountering an issue with * 11.14.2 and odbc where it's randomly segfaulting....i've got the core dump and a backtrace, and it seems to be odbc related; but when i manually run the query, i dont' have any issues..... |
| 21:27.30 | robmal | mhjkl: Yes. |
| 21:27.43 | adeeln | http://pastie.org/10420014 is the backtrace |
| 21:28.03 | robmal | You can achieve almost anything when placing .call in /var/spool/asterisk/outgoing |
| 21:28.11 | robmal | Still - no porn |
| 21:28.22 | mhjkl | lol :-) |
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| 22:15.37 | snadge | hmm... polycom vvx 500 |
| 22:16.05 | snadge | played around with its nat settings.. but cant get it to stay registered.. just goes "unreachable" |
| 22:16.15 | snadge | after about a minute or two.. which just screams nat |
| 22:17.44 | snadge | going through a cisco router that i know nothing about of course.. the yealinks and ciscos going through it work fine |
| 22:19.04 | snadge | polycom.. straight in the bin ;) |
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| 23:05.13 | qdk | Is whitespaces allowed in an exten => line? like exten => _XX. ,1,noop()? |
| 23:06.06 | WIMPy | Before the _ it's ok, but not thereafter. Spaces should generelly be avoided, such as quotes. |
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