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| 00:47.14 | robink | OK, I have a working install of Asterisk 16.2.1, but I'm unable to load the res_pjsip.so module (it's there). When it attempts to load, Asterisk complains: ERROR[24550] loader.c: Error loading module 'res_pjsip.so': /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: pj_ssl_cipher_name |
| 00:47.33 | robink | This happens whether I use the bundled pjproject 2.8 library or one (of the same version) installed systemwide |
| 00:48.18 | robink | I'm thinking it might have something to do with the fact that I have OpenSSL 1.1.0j installed (instead of 1.0.x), but I'm not sure, and purportedly compatibility issues with OpenSSL 1.1.x have been ironed out. |
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| 09:36.00 | dym | Hey guys! Im running Asterisk 16.2.1 with spandsp installed and res_fax compiled in. My Dialplan Application ReceiveFax tells me "res_fax.c:1173 fax_session_reserve: Could not locate a FAX technology module with capabilities (RECEIVE)" - what am i missing? |
| 09:37.10 | dym | whoops! think i found something. app_fax is missing. |
| 09:38.06 | dym | ah no, app_fax is deprecated. |
| 09:38.28 | dym | res_fax is there, system has libspandsp |
| 09:40.11 | dym | core show applications like fax shows ReceiveFAX and SendFAX |
| 09:43.30 | dym | think i've got it. 0 fax technologies registered. rebuilding with res_fax_spandsp. |
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| 14:49.33 | domlaz11 | hello, i am having an issue with RTP delay, around 5 seconds, very random, causing 5 seconds audio silence, can see max delta up to 5000ms on wireshark |
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| 15:17.43 | rrittgarn | is there any built-in functionality of app_confbridge that allows me to do "something" when the last caller leaves a bridge? |
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| 16:30.44 | rrittgarn | If a variable is set with GLOBAL should I expect it to persist past the hangup of the channel that set it? |
| 16:30.53 | rrittgarn | or when the channel hangs up does it go out of scope? |
| 16:31.16 | file | global persists, and is across all channels |
| 16:39.40 | rrittgarn | interesting... I'll have to double check my syntax - I may have found a bug - but I am using 13.4 - so guessing that might be expected |
| 16:40.36 | file | maybe? global variable functionality hasn't been touched in a very long time |
| 16:41.37 | rrittgarn | I'm setting a global in a gosub - but I check its value before I set it... and thus far with single calls I keep showing it as an empty value. When I have multiple calls up at the same time the value is consisent. |
| 16:41.44 | Samot | Globals don't live on channels. |
| 16:41.54 | rrittgarn | which is why i was asking for clarification on scope |
| 16:42.08 | Samot | Your Global var will be available in h |
| 16:42.36 | rrittgarn | correct - i see it there at the end of the call / channel |
| 16:42.55 | Samot | So to answer your original question, yes. |
| 16:43.11 | Samot | It will persist past the hangup. |
| 16:43.40 | Samot | Are you setting this via GLOBAL()? |
| 16:43.45 | Samot | Or in [globals]? |
| 16:43.56 | rrittgarn | via GLOBAL() - working on a pb right now |
| 16:46.11 | rrittgarn | relevant dialplan: https://pastebin.com/0rQzB489 |
| 16:51.12 | Samot | Yeah, quick look... |
| 16:51.22 | Samot | You're not setting it the first time properly. |
| 16:52.13 | Samot | Still looking.. |
| 16:52.34 | rrittgarn | Sorry got a call and have to switch gears - i'll get the output in a few minutes |
| 16:53.46 | Samot | Just would like to see a a single call output in the full log. |
| 16:53.53 | Samot | The one that doesn't set for you. |
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| 17:18.32 | rrittgarn | I guess I must have typed something wrong and fixed it when trying to get everything nicely into PB - working fine now... |
| 17:18.38 | rrittgarn | appreciate the insight either way |
| 17:19.41 | rrittgarn | or because there are other active channels its reporting differently... i'll retest and report back once everyone is off the phone |
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| 20:19.55 | Pegasus_RPG | Hello. I have Fail2Ban installed and working on my * but one abuser is somehow able to try many logins to the point of DoS. |
| 20:20.48 | Pegasus_RPG | 'core show channels' shows 36 active channels, 36 active calls even though nobody is on any legitimate calls |
| 20:21.34 | Pegasus_RPG | 'sip show channels' shows 160 active dialogs even though there are only 2 legitimate ones |
| 20:21.39 | csavinovich | if the channels are not in a legit call then what are they doing? |
| 20:21.44 | [TK]D-Fender | core channels = calls, not "logins" |
| 20:22.01 | Pegasus_RPG | csavinovich: They're all in state Ring |
| 20:22.04 | [TK]D-Fender | or "just channels" actually being processed |
| 20:22.12 | [TK]D-Fender | which would imply they were acceptd |
| 20:22.21 | Pegasus_RPG | Looks like they're running through the script I set up to check for valid users |
| 20:22.22 | [TK]D-Fender | and not suject to fail2ban since clearly they didn't fail |
| 20:22.54 | csavinovich | if they are ringing then fail2ban didn't ban them |
| 20:23.20 | Pegasus_RPG | Okay, that makes sense. So does that mean one of my extensions' passwords was cracked? |
| 20:23.31 | [TK]D-Fender | no |
| 20:23.36 | [TK]D-Fender | they simply exist |
| 20:23.41 | [TK]D-Fender | we don't know anything about those call |
| 20:23.46 | [TK]D-Fender | because we're not looking at anything |
| 20:24.16 | Pegasus_RPG | Can I get more information? |
| 20:25.07 | [TK]D-Fender | You should |
| 20:26.11 | csavinovich | The call just shouldn't even get to asterisk if blocked by fail2ban. your fail2ban script is failing |
| 20:26.58 | [TK]D-Fender | not yet it isn't |
| 20:27.13 | [TK]D-Fender | and from the description it's not part of the issue |
| 20:27.32 | Pegasus_RPG | Yeah, I think fail2ban is working fine. It E-mails me every time it blocks someone. This particular person managed to make it through. |
| 20:27.44 | csavinovich | ....or is it maybe the SIP headers are the one going through, and the RTP packets are the ones being block... that's why you see the call trying to initiate... you might just be blocking rtp |
| 20:27.59 | [TK]D-Fender | RTp doesn't get blocked like that |
| 20:28.07 | [TK]D-Fender | the logic used here is flawed |
| 20:30.23 | Pegasus_RPG | How can I get more information on what those channels are doing? And what information would be useful? |
| 20:30.51 | Pegasus_RPG | (At this point, I'm just tempted to block the netblock at my firewall so legitimate calls can stop having quality problems due to system load.) |
| 20:31.34 | Pegasus_RPG | But that's only a temporary fix until the attackers move somewhere else. |
| 20:32.54 | Pegasus_RPG | Increasing console verbosity shows that they are trying extension numbers sequentially |
| 20:33.17 | csavinovich | scary |
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| 20:44.00 | [TK]D-Fender | Why are we not looking at this? |
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| 21:32.25 | rrittgarn | https://pastebin.com/hqPz3E0y - with app_confbridge am I missing something for telling it the recording file name here? |
| 21:41.04 | rrittgarn | as it sits confbridge is recording to /var/spool/asterisk/monitor with default file names |
| 21:41.20 | rrittgarn | i'd be happy if i could just specify the name of the file dynamically... like documentation says to... |
| 21:44.46 | Pegasus_RPG | [TK]D-Fender: because I don't know what information would be helpful in stopping future such abuses, other than disabling federated VoIP of course. |
| 21:45.06 | [TK]D-Fender | We don't know what's happening |
| 21:45.22 | Pegasus_RPG | And I don't know what commands to find that out |
| 21:45.46 | Pegasus_RPG | A Web search of "what's happening with asterisk channels" is not specific enough |
| 21:47.12 | rrittgarn | I believe what Fender is trying to say is to help you, we would need to see logs with verbosity up high enough to see the requests. It also sounds like your PBX is internet facing - your default context might be allowing people to make calls (or try to) and as such show you your current oddities |
| 21:53.47 | [TK]D-Fender | "show us the calls" |
| 21:54.46 | [TK]D-Fender | <[TK]D-Fender> we don't know anything about those call |
| 21:54.46 | [TK]D-Fender | <[TK]D-Fender> because we're not looking at anything |
| 21:55.37 | rrittgarn | https://pastebin.com/hqPz3E0y - with app_confbridge am I missing something for telling it the recording file name here? |
| 22:08.15 | [TK]D-Fender | rrittgarn, You're not paying attention to what bridge you are setting that for |
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| 22:16.24 | [TK]D-Fender | 2 hours later .... nothing to see and a channel quit... |
| 22:17.33 | rrittgarn | am I missing an argument on the CONFBRIDGE() call? |
| 22:18.04 | [TK]D-Fender | look at what you're feeding it |
| 22:18.36 | file | you're not using the dynamic profile support correctly, so it won't work |
| 22:19.03 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_ConfBridge |
| 22:19.09 | file | https://www.irccloud.com/pastebin/9Xq80wnj/ |
| 22:21.25 | rrittgarn | file - that should be in the documentation. works perfectly (other than the timestamp - which i can deal with) |
| 22:21.28 | rrittgarn | thank you both |
| 22:21.43 | file | it is specified in the documentation that it has to be blank to use a dynamic profile |
| 22:22.26 | rrittgarn | yep - see that now. completely glazed over it for far longer than i'd like to admit |
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| 23:51.26 | file | https://blogs.asterisk.org/2019/03/06/asterisk-usage-survey/ the survey calls to you all... |
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| 23:55.28 | Samot | 22 PAGES?! |
| 23:56.15 | file | Samot: we want actual useful information :P |