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00:20.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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15:35.54 | neo2k | hello, i have a very strange behaviour of asterisk here. pjsip shows a warning, that it gets no response from the sip server but when I call my number the calls come in anyway. what could be the issue here? |
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15:42.46 | file | You'd need to provide more information like console output and SIP trace |
15:42.58 | [TK]D-Fender | (which I responded in #freepbx) |
15:43.22 | [TK]D-Fender | Which given that's what you need to use to fix anything that might be misconfigured is where this should probably be.... |
15:48.08 | neo2k | @file, [TK]D-Fender i'll make a gist then |
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15:50.55 | clack | I am working on an issue, or several issues, that appear to be dahdi related. In short, I have a newish Asterisk system that has both SIP and landline trunks, of which the landline side is repeatedly having intermittent issues. I am looking for some guidance on where next to focus the troubleshooting effort. Much of the effort so far has been documented here: https://community.freepbx.org/t/reoccurring-issues-multiple-problems-with-dahdi-line/49036. Any |
15:50.55 | clack | help from the community here would be appreciated. |
15:53.02 | neo2k | @file, [TK]D-Fender i get this response https://gist.github.com/tschreiner/cc6c6c0855ed72c3acec7d41843c696f but it looks like asterisk somehow doesnt acknowledge it |
15:54.13 | neo2k | is there maybe a setting to permit the ip range 217.*/16 or so? |
15:54.39 | file | That response isn't for that REGISTER |
15:55.22 | [TK]D-Fender | FROM/TO mis-match |
15:56.32 | neo2k | [TK]D-Fender how can you tell that? |
15:57.56 | file | The callid doesn't match and the cseq doesn't match |
15:59.53 | [TK]D-Fender | look at the FROM and TO between the REGISTER and the OK |
16:00.02 | [TK]D-Fender | can you see the numbers don't match? |
16:01.08 | neo2k | this is my trunk config https://imgur.com/a/6M22ea1 |
16:02.12 | neo2k | but i see now, from, to, call-id and cseq don't match |
16:02.38 | neo2k | is there anything i can do about? that looks actually like a misconfiguration of the remote server to me |
16:05.14 | Samot | Well, start with going to #freepbx |
16:05.22 | Samot | There's no GUI in Asterisk. |
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16:54.17 | clack | I suspect it is not a GUI issue or even related to FreePBX, which is why I am seeking the assistance of the experts here. |
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17:10.00 | Free99 | for some reason my android system is saying I'm registered, but tcpdump on my server is saying 401 unauthorized. Anyone seen that before? |
17:10.07 | Free99 | I've verified the credentials and such |
17:12.13 | [TK]D-Fender | Who said getting a 401 is bad? |
17:19.33 | Free99 | [TK]D-Fender, specifically, 401 unauthorized |
17:19.40 | Free99 | but ok, a phone call went through |
17:19.57 | Free99 | weird that android allows you to have a SIP account but optional registration |
17:21.41 | [TK]D-Fender | "android system is saying I'm registered" <- nothing in there says "optional" to me yet |
17:21.58 | [TK]D-Fender | Though technically SIP specs alone don't necessarily mandate it |
17:22.38 | [TK]D-Fender | but none of the circumstances or your current line of reasoning actually lead to any real conclusion yet |
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17:52.24 | Free99 | what was strange was silence in the calls etc until I actually registered the extension, testing against the echo test extension |
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18:59.56 | Free99 | [TK]D-Fender, getting this as a broadcast message from root on the TTY: Loop detected in monitoring script. Issue with Asterisk? Restarting! |
19:00.01 | Free99 | seen that before? |
19:01.26 | Free99 | looks like systemd is complaining about voipfirewalld |
19:03.00 | Free99 | https://issues.freepbx.org/browse/FREEPBX-17396 <- ticket about it created yesterday. Sigh. |
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