IRC log for #asterisk on 20210316

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05:57.21*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
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14:32.44igcewielingAmazing how fast I get IP info from our people when I tell them I'll have the contractor call them for the info if they don't send it to me.
14:33.12igcewielingThe older I get the more I think being an asshole is the only way to get anything done.
14:37.21SamotI figured that out in my 20s
14:42.46seanbrightyeah ya did
14:44.50SamotNo regrets
14:46.24SamotBecause as much as people complain I am an asshole, they seem to get over it when they need my help.
14:47.08seanbright👍
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14:49.25SamotEither I am an asshole when I help or I am an asshole for not helping. Either I am an asshole but you end up with a 50/50 chance of your shit not working. Im good with that.
14:49.37Samot+way
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14:55.52igcewielingCustomer is complaining of inconsistent internet.   They have a T-1 frame relay connection.   *sigh*
14:55.53kerouac[m]Hi. I have two linphone clients, and an instance of asterisk. When I call from one to another client, the sound goes only one way, from caller to handling client. And there is no sound from answering side. I use codec opus, and NAT is opened for sip and rtc protos. Even when i'm truing to send text message, I see in log something like this on receiving side:
14:55.53kerouac[m]https://termbin.com/csyw
14:56.47igcewielingkerouac[m]: sorry, too complicated.    Stick with a simple phone call using ulaw.
14:56.57igcewielingonce that works, then try messaging.
14:57.12kerouac[m]Ulaw behaves the same way
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14:59.55SamotAnd where are these linphone clients at?
15:00.37kerouac[m]to be true, when I do allow=ulaw the codec used is PCMA, is it correct?
15:00.53kerouac[m]PCMU*
15:01.37kerouac[m]<Samot "And where are these linphone cli"> They are in local network both, seeing each other on different local ip's
15:02.38kerouac[m]But they are as you could see in that log https://termbin.com/csyw doesn't see each other correctly (anonymous.invalid and so on). And they also doesn't handle online status of each other correctly. But I still can call and see incoming calls.
15:03.30igcewielingneither phone is running on the same server as Asterisk, correct?
15:03.51igcewielingincoming calls have nothing to do with outgoing calls.
15:05.08kerouac[m]yes, three machines , one for asterisk and two for linphones
15:06.27SamotWe need to see actual debugs from Asterisk
15:07.47kerouac[m]Now I'll record that
15:10.03kerouac[m]what debugs should I rutn on?
15:13.36SamotIs this pjsip?
15:13.41Samotpjsip set logger on
15:13.46SamotFrom the Asterisk console.
15:14.54Kobazthe loooooogs
15:15.02KobazLogs, logs, they're big, heavy and wood
15:21.03kerouac[m]https://termbin.com/4326
15:21.55kerouac[m]Here it is. There is log of coccention of one of the clients, ans two calls, from 6001 to 6003 and then backward. Also a couple of tries to send text message.
15:24.53igcewielingGood luck with that Samot and Kobaz!
15:26.34SamotWhy in the fuq is DTLS being used?
15:26.37SamotTurn that shit off.
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15:29.06kerouac[m]what is dtls? how to turn it off?
15:30.45SamotIt's a encryption method for this. Check the linphone configs.
15:32.42kerouac[m]Could it be the reason of the troule?
15:34.28SamotWell probably
15:35.41igcewielingDoes it break the rule of "keep it simple when testing"?   Yes, it does.
15:40.28kerouac[m]both Linphones' encryption is turned off, is it enought to remove dtls_auto_generate_cert=yes from pjsip.conf?
15:44.45kerouac[m]removed that line, the same result
15:46.41electronic_eelSamot: you were asking about BLF and pickup yesterday
15:47.02electronic_eelmy users are using it regularly
15:47.04SamotNot really.
15:47.13SamotWell see Kobaz is a C programmer
15:47.18SamotLikes to mess with Asterisk code.
15:47.31electronic_eelah, that was i was just going to mention
15:47.32SamotI am not a C programmer. There is something that needs an improvement.
15:47.48electronic_eelto make it work, i always have to apply a patch from https://issues.asterisk.org/jira/browse/ASTERISK-24601
15:48.11electronic_eelwith this patch it works well for me on polycom phones
15:48.16SamotI haven't applied it yet..
15:48.22SamotDoes it fully work?
15:48.30electronic_eelfor me, yes
15:48.31SamotTell you not only who is calling but who the call was ment for?
15:48.54SamotBecause I had a feeling that patch was incomplete on one of those.
15:48.59electronic_eelhmm, ment for?
15:49.02SamotMeant
15:49.14SamotIf I call 2100 and you're a BLFing 2100
15:49.25SamotYou should see that I'm the caller and the call was meant for 2100
15:49.29SamotOr perhaps a ring group
15:49.43SamotIt has both the remote and the target?
15:49.45electronic_eeli see a blf line blinking and press on it. then i see the caller id that is ringing on that line
15:50.01SamotCan you show a notify?
15:50.07SamotThat will answer my question
15:50.38electronic_eellet me just check if i see the target
15:51.45kerouac[m]maybe I should use something like  localnet=192.168.2.0/255.255.255.0 ?
15:51.51kerouac[m]in pjsip.conf?
16:00.00igcewielingperhaps remove all the extra stuff from pjsip.conf?
16:00.28igcewielingIf you had dtls_auto_generate_cert set, who know what other crap you have in pjsip.conf
16:02.30electronic_eelSamot: ok, here is the full dialog-info from the NOTIFY: https://paste.debian.net/1189616/
16:02.46electronic_eelas you can see the full caller id and target are included
16:03.08electronic_eelso for me it works well, i get everything i want from it
16:03.38SamotWhat version of asterisk
16:04.07electronic_eel16.16.1
16:04.21SamotSo the 16.8 patch is working?
16:05.03electronic_eelyes, seems like
16:05.21SamotYeah looks like it covers it mostly.
16:05.37SamotUnless that was truly an anonymous call
16:05.43electronic_eeliirc, i asked file sometime ago why the patch wasn't merged yet and i got the answer that it wasn't clean enough
16:05.49seanbrightthat is correct
16:05.56seanbrightjust based on a glance at it
16:06.03SamotWhat needs cleaning?
16:06.34electronic_eelSamot: the call wasn't anonymous, but a call from external with a regular numeric caller id
16:06.40seanbrightcoding style guidelines would be a good first pass
16:06.45seanbrightlet me do it, hold on
16:07.21electronic_eelthe "@anonymous.invalid" seems to be added by default if there is no proper domain or something like that
16:07.37SamotThat's what I was referring to
16:07.50SamotThat's an anonymous call format
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16:08.26seanbrightpatch: **** malformed patch at line 77: +
16:08.27igcewielingthat often means "incoming call didn't match anything"
16:08.29seanbrightwomp womp
16:08.52SamotOr it was anonymous
16:09.20Samotseanbright: so line 77 has something malformed?
16:09.23electronic_eelso with "anonymous call format" you mean you have a regular numeric caller id, but no sip domain?
16:09.52seanbrightSamot: what would make you think that?
16:10.09SamotDont know.
16:10.22SamotA hunch
16:10.40SamotI heard it from a guy
16:10.51SamotSome people were saying..
16:12.27Kobazunfortunately I don't have any experience in the text messaging part of the code
16:13.16SamotKobaz could you clean up the current patch?
16:13.23seanbrighti am working on it
16:13.24seanbrightjeeez
16:13.39kerouac[m]<igcewieling "If you had dtls_auto_generate_ce"> https://termbin.com/l3ub
16:13.46KobazHow do you make a thumbs up in ascii-emoji
16:13.50SamotOh i just thought you were seeing what was messed up
16:13.58Kobazseanbright: <3
16:14.00SamotMy bad
16:14.02KobazThat's more like a heart
16:14.44electronic_eeloh, iirc i had to adapt the patch some asterisk releases ago since it wasn't applying cleanly anymore
16:14.56electronic_eelhere is the version i'm using now: https://paste.debian.net/1189620/
16:16.23igcewielingkerouac[m]: looks like you have webrtc stuff too.   that breaks the "keep it as simple as possible when testing" rule.
16:20.39igcewieling*sigh* We want to move one of our "border" servers to another subnet and Level3 wants to do full interop testing first.
16:24.06seanbrightpatch updated
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16:25.32seanbrightsome of the code in the `if (callee) {` block is questionable and would not pass a review
16:26.18seanbrightbut it applies cleanly now at least
16:27.57kerouac[m]<igcewieling "kerouac: looks like you have web"> that helped to hear in both directions!
16:28.53igcewielingkerouac[m]: not if you were using SIP.
16:29.24kerouac[m]why
16:29.34igcewielingwebrtc is a different protocol.
16:29.57kerouac[m] < https://matrix.org/_matrix/media/r0/download/matrix.org/brNYQFNaNSOLvmeopbMuydXF/message.txt >
16:30.20kerouac[m]<igcewieling "webrtc is a different protocol."> afaik webrtc=yes puts a lot of parameters to endpoint, including dtls
16:30.30kerouac[m]I think that was the reason
16:30.43igcewielingthe first three lines are junk unless you are dialing 200
16:31.00igcewielingactually 2-4 is junk
16:31.16kerouac[m]of course, but the last, is it enough? Or should contain comething else?
16:32.44igcewielingkerouac[m]: it should dial a basic call with no handling of errors.
16:41.27kerouac[m]It does. But linphones still doesn't see online status of each other. What to do to avoid that @anonymous.invalid stuff?
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17:23.18seanbrightthe 'gross' part about that patch is the "find ringing channel" part
17:27.57SamotOh?
17:29.35SamotSo that entire function is gross?
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17:34.55seanbrightno, it's just kinda arbitrary
17:35.18seanbrighti dunno, it's fine
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