IRC log for #asterisk on 20200514

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01:20.39*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); 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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04:46.57s98259My SMS storage is full, I looked at the documentation and it says my SMS messages are stored in /var/spool/asterisk/sms but there is no sms directory however in the past my console was showing SMS messages.
04:47.05s98259So I know I am receiving and storing them
04:47.50s98259I've even tried locate command to find "sms" but no directories
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04:57.38drmessanoUm
04:57.51drmessanoYour SMS storage?
04:58.31s98259Yes
05:01.14drmessanoWhat process is telling you that your SMS storage is "full"?
05:03.56s98259The Asterisk process, or is that not what you mean?
05:06.49drmessanoWhat is the actual issue?
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05:15.54drmessanoWelp, i've not seen a message warning that sms storage is full.  smsq dumps into /var/spool/asterisk/outgoing but I would imagine you would have other disk space issues before an Asterisk warning
05:16.14drmessanoSo if you don't want to provide more detail, then good luck
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09:06.50ZombieHello.
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09:59.30ZombieI've estup a second Asterisk server running my build of Asterisk 17.4
09:59.56ZombieI can get my devices to register to it, but, you can't hear any calls.
10:00.18ZombieThe Configs are notably different
10:11.04ZombieBut right now, the server can call other devices but the calls get immediately hubg up.
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11:26.20SamotWhy did you build another server?
11:58.30ZombieBecause I didn't want to endanger the setup I have.
11:58.48ZombieOn the second system, I get RTP errors.
12:00.33SamotEndanger for what?
12:01.23ZombieI currently have a working system where Phones can internally call each other.
12:01.35ZombieI don't yet have a way to call the outside world.
12:01.58SamotOK so you need a provider.
12:02.17*** join/#asterisk verzo (~verzo@gateway/tor-sasl/verzo)
12:02.56SamotI'm not sure how that endangers the system.
12:03.38ZombieThe second system running 17.4 cannot call node to node.
12:03.41Zombieat all.
12:03.57Zombiethe phones ring. when someone picks up, they hang up.
12:04.37Samotagain, why do you need a second system?
12:04.55SamotWhy do we need to repeat the process of building a new system and troubleshooting problems so you can add a trunk?
12:05.18ZombieMy current working system is on a very old version.
12:05.47ZombieI have an upgrade path to the new version.
12:06.05Zombiebut the new version doesn't work
12:06.27SamotOK, so now we did all this with the old system..
12:06.36SamotThe one we kept telling you to update, you didn't want to.
12:06.49SamotSo now you're looking for help to do that exact process over again
12:07.02ZombieWe couldnm't update it before. Now we can.
12:07.12SamotYou need to show things then.
12:15.23SamotWell?
12:17.01ZombieLet me fire up the secondary, okay?
12:17.13ZombieI'll get you all my stuff
12:24.11ZombieSecondary system is online.
12:24.20ZombieJust let me console into it.
12:37.54Zombiehttps://pastebin.com/Q3gBCnBC
12:38.08ZombieI have many errors to chew on.
12:41.28SamotAre you using Chan_PJSIP?
12:41.45ZombieYes, on my old version I was using chan_sip.
12:42.01Zombiebut this new one requires pjsip
12:42.03SamotYou haven't even configured most this crap
12:42.25ZombieIts a relatively new setup compared to the old one.
12:42.58SamotYou've got duplicate pjsip objects. You have conf files missing or not configured with information
12:43.15SamotYou've installed a bunch of crap you don't need like chan_skinny
12:55.36*** join/#asterisk velix (~velix@unaffiliated/velix)
12:55.44velixWhat the heck is she doing? https://bpa.st/EVSA
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13:02.38SamotIs that a mobile device?
13:05.57velixSamot: No, but I bet she has broken WLAN.
13:06.22SamotSo it's not a mobile device but it's on wifi?
13:06.38velixSamot: Is a laptop a mobile device?
13:06.44velixIt might be a laptop on Wifi.
13:06.54velixlaptop = notebook
13:07.00SamotI know.
13:07.06SamotThanks for clearing that up.
13:07.23SamotIf it was a mobile device, it could be related to backgrounding issues.
13:07.34velixSorry, I thought it's a German thing Laptop / Notebook.
13:07.35SamotAnd not being able to respond to keepalives, etc.
13:07.50SamotIf it's wifi, it could be a poor signal.
13:07.53velixSamot: I'll ask her. She's the only one having problems.
13:08.01SamotIt could be something in the laptop's firewall
13:08.06SamotWho knows.
13:09.38velixSamot: Yeah, I also think so. Crazy. She's filling my logs :D
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13:31.11Zombiehttps://pastebin.com/Z5D1VK47
13:31.34ZombieTried to fix some of this.
13:38.55ZombieAllocating new SIP dialog for 05d7e945acb56a4cb51f1b163313fc6f@192.168.0.21 - OPTIONS (No RTP)
13:39.04ZombieWhat does no RTP mean?
13:39.15SamotIt's an options message, they generally don't have RTP
13:52.18Zombiehold on.
13:52.26ZombieI've messed up somewhere.
13:52.38ZombieLet me paste you an output of an attempted call
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13:56.42ZombieNow stuff doesn't even register
14:07.56SamotOK so what is the current box again?
14:08.03SamotWhat version of Asterisk is it running?
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14:21.39*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
14:21.40ZombieUgh.
14:21.45*** join/#asterisk wolrah (~wolrah@24.239.210.140)
14:23.06SamotNo, I was asking what version of Asterisk the _current_ box is running
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14:23.11*** mode/#asterisk [+o bford] by ChanServ
14:23.51wolrahcan anyone comment on the reason why the default endpoint identifier in pjsip is the IP address and not the username?  I just finished helping someone figure out a problem caused by that setting and we were discussing it but could not come up with a good reason for that behavior to be the default.
14:24.51wolrahit seems to be a recurring source of problems for people upgrading from older installs that used chan_sip, especially if they use the "match" field as a whitelist in combination with phones behind NAT
14:25.15ZombieSamot, 17.4
14:25.20Zombiehttps://pastebin.com/iP7Kw7Ut
14:25.29SamotOK..
14:25.30SamotStop.
14:26.03Samot8:05:19 AM <Zombie> My current working system is on a very old version. <- WHAT VERSION
14:26.16filewolrah: I don't remember any specific reason, it was done long ago
14:26.43ZombieSamot, 13.7.2
14:27.23SamotSo have you moved configs over from the other box?
14:27.32SamotBecause it really doesn't change that much.
14:27.38ZombieYes I have.
14:28.49Zombiebut the old box used chan_sip
14:29.01Zombiethis one, I wan to do correctly and use jjsip
14:29.03SamotOK.
14:29.05Zombiethis one, I wan to do correctly and use pjsip
14:29.12SamotSo why did you install all the other crap?
14:29.26SamotLike chan_skinny, chan_mobile, chan_sip
14:29.32ZombieInstalled by Distro.
14:29.37ZombieCan be disabled.
14:29.39SamotWhat distro?
14:29.43ZombieMageia.
14:29.54SamotSo you installed a Asterisk 17 distro from them.
14:30.00ZombieYes.
14:30.10ZombieBut this is a config issue.
14:30.10SamotOK so what is the issue now?
14:30.28Zombieon the new system, virtually nothign works now.
14:30.47Zombienodes will register to it and not much else.
14:31.03SamotSo they can't even call into the system for voicemail?
14:31.11ZombieCorrect.
14:32.07SamotThen we need to look at your configs.
14:32.15SamotAnd what is happening when a call is made
14:32.29ZombieOn the Phone? Line busy
14:33.40Zombieand on the server
14:33.45Zombie[2020-05-14 10:32:54] WARNING[18210][C-00000004]: chan_oss.c:498 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
14:35.18SamotWhy the hell is chan_oss involved in this?
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14:46.12igcewielingI assume he never went into modules.conf and removed unused modules.
14:48.34igcewielingZombie: did you read the UPGRADE.txt files, which lists important changes between versions?
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15:20.15ZombieIt shouldn't be
15:27.54ZombieI changed it to ALSA
15:28.30igcewielingwhy are you trying to use a console phone?
15:30.28ZombieI really don't know what I am doing completely and poking at this with a stick. No one has offered to completely explain things in a way I can fully understand.
15:34.07igcewielingyou might consider sticking with the old system.
15:38.40SamotCompletely explain things?
15:38.45SamotWhich things?
15:42.02ZombieThats not a viable solution.
15:42.19ZombieFirstly, the old system doesn't really do anything useful.
15:43.19ZombieSecondly, my Distribution already has the very upgrade I am using in its testing repos and is waiting for my feedback.
15:44.13SamotPardon?
15:44.22ZombieA faulty build was produced earlier that got pulled.
15:44.41ZombieIt set the BUILD Native flag to yes, and broke portability.
15:44.51ZombieThat's been fixed.
15:47.09ZombieThis project has a few goals.
15:47.50Zombie1. Devices registered should be able to call each other's extensions.
15:48.43Zombie2. Users Voicemails need storage in the local DoveCot Directory.
15:48.52Zombie(IMAP)
15:49.16ZombieI have a version of DoveCot running on these severs that acts as a storage medium
15:49.47SamotWell good luck with that.
15:50.10SamotNot sure how much voicemail in IMAP is being used by people.
15:50.15Zombie3. once the first two are done, find a SIP Provider.
15:50.29ZombieI just want to work on number one right now.
15:52.18ZombieI will fork over whatever configs are needed to see what's gone wrong here
15:52.26ZombieAre you willing to read them?
15:53.57SamotI asked for them over an hour ago
15:56.45ZombieAlright.
15:56.53ZombieLet me post them again.
15:58.30Zombiehttps://pastebin.com/QHqhrJBm
16:03.05Zombiehttps://pastebin.com/gT3PgjqR
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16:03.29SamotOK so show me where a call fails
16:06.09Zombiehttps://pastebin.com/NWdCWmJ9
16:06.41igcewielingWhich paste contains the log of the ACTUAL dial?
16:08.31igcewielinglines like this: [2020-05-14 12:08:11] VERBOSE[26300][C-000006bc] pbx.c: Executing [~~s~~@sm_dial:32] Dial("PJSIP/beaky-01-00000d7d", "PJSIP/+17187023148@vzb-regular,,gb(sm_pre_dial^s^1)U(sm_answer)F(sm_intercept_to_called^s^1)") in new stack
16:09.17SamotOK
16:09.37SamotZombie: We've done this dance recently. Let's not repeat the same wrong moves.
16:09.56SamotZombie: I want to see a call with the pjsip logger on not core debug
16:10.06SamotI want to see a failed call.
16:10.37Zombiehow do I activate that?
16:12.19Samotpjsip set logger on
16:12.23Samotcore set debug 0
16:12.51igcewielingand core set verbose 3
16:14.12ZombieCore debug was 1 and is now OFF.
16:14.12Zombie[2020-05-14 12:13:31] ERROR[20371][C-0000000a]: chan_console.c:477 console_request: Console device 'dsp' not found
16:14.12Zombie[2020-05-14 12:13:31] WARNING[20371][C-0000000a]: app_dial.c:2576 dial_exec_full: Unable to create channel of type 'Console' (cause 0 - Unknown)
16:14.12Zombie[2020-05-14 12:13:31] WARNING[20371][C-0000000a]: pbx.c:2927 pbx_extension_helper: No application 'Voicemail' for extension (public, stdexten-NOANSWER, 1)
16:14.14Zombiekefk
16:14.24ZombieuGH.
16:14.33SamotWhy are you using chan_console?!
16:14.48SamotHow is that even possible?!
16:15.32SamotZombie: Do: pjsip show transports
16:15.58Samotpjsip show settings
16:16.03SamotPastebin both of those
16:16.12Zombiepjsip show transports
16:16.12ZombieNo objects found.
16:17.35Zombiepjsip show transports
16:17.35ZombieNo objects found.
16:17.39Zombiehttps://pastebin.com/3d0LnH88
16:18.31SamotOK and now: sip show settings
16:21.18Zombiehttps://pastebin.com/kWEgYXsU
16:22.23igcewielinglooks like you are running chan_sip?
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16:26.04ZombieI think they are both loaded'
16:26.11Zombiebut I want to run pjsip
16:26.21igcewielingthen stop loading chan_sip
16:26.21Zombiethe old system has chan_sip
16:26.30Zombieokay
16:26.39ZombieI will add a noload to it
16:26.39igcewielingrun one or the other, not both.
16:27.43Zombiekefka*CLI> sip show settings
16:27.43ZombieNo such command 'sip show settings' (type 'core show help sip show' for other possible commands)
16:28.03SamotWell..
16:28.09igcewielinggood, that shows chan_sip is not loaded.
16:28.29SamotOK so is PJSIP listening on 5060?
16:28.32igcewielingPersonally, I'd upgrade and test, THEN upgrade to pjsip.
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16:31.52ZombieTransport:  transport-udp             udp      0      0  0.0.0.0:5060
16:32.58Zombie[2020-05-14 12:32:30] NOTICE[26530]: res_pjsip_session.c:3228 new_invite: Call from '6002' (UDP:192.168.0.132:59566) to extension '6001' rejected because extension not found in context 'internal'.
16:33.43wolrahI would agree with igceweiling, it's generally easier to get through an upgrade if you make as few changes at a time as possible, so if something goes wrong you have a much better idea where to look.
16:33.45igcewielingnow you are making progress.  Looks like you don't have an extension 6001 in the [internal] context in extensions.conf.
16:35.02Zombiecontext=from-internal
16:35.23igcewieling"internal" is not the same as "from-internal"
16:36.08igcewielingThat is like mixing up N2O and NO2.  very bad.
16:37.45ZombieWait.
16:37.59ZombieIsn't extensions.conf ued by chan_sip?
16:38.12igcewielingZombie: yes.  and by pjsip too!
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16:40.26igcewielingfrom-internal looks like a FreePBX system.
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16:44.42Zombieexten => 6000,1,Dial(SIP/6000)
16:44.42Zombieexten => 6001,1,Dial(SIP/6001)
16:44.42Zombieexten => 6002,1,Dial(SIP/6002)
16:44.47ZombieI have these.
16:44.57ZombieIs this not correct?
17:04.40SamotWell that's not correct if you're trying to dial PJSIP
17:05.16ZombieHow should it be formatted?
17:05.27SamotOK, this is documented.
17:05.46SamotThere are a few ways to Dial() PJSIP the simplest is Dial(PJSIP/6000)
17:05.56Zombiehttps://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
17:05.58ZombieOh.
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17:10.42igcewielingIt seems to me half your problem is not researching converting from chan_sip to pjsip enough.
17:16.52Zombie[2020-05-14 13:15:14] WARNING[8620]: res_pjsip_registrar.c:1073 find_registrar_aor: AOR '6001' not found for endpoint '6001'
17:16.55Zombierm.
17:21.16Samot1:10:43 PM <igcewieling> It seems to me half your problem is not researching enough. <-- Fixed it
17:21.36SamotLike I said, we just did this dance in the last few weeks with Zombie on the other system.
17:21.42SamotThat was a chore to get up as well.
17:21.55SamotAll the time we said, update the Asterisk version and go to PJSIP
17:22.02SamotAlso stop using users.conf
17:22.08SamotAll that fun stuff.
17:22.41ZombieThis build was not availible two weeks ago.
17:22.50SamotThe source was
17:23.06SamotPJSIP was available in v13
17:23.22SamotIt's been around since v12.
17:24.26igcewielingSwitching to pjsip is a pain in the ass and complicated.   However, it is worth the trouble in the long run.
17:24.42ZombieThats why I am doing this.
17:25.18igcewielinggo read the UPGRADE.txt file with the Asterisk source for whatever version you are using.
17:38.16ZombieI got them ringing each other
17:38.26ZombieThey immediately hang up on each others
17:46.57SamotOK now let's confirm something is actually happen.
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17:47.07SamotDo pjsip set logger on again and make another call
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20:12.24AsteriskRossHello, has anyone been able to filter CDR in cdr_adaptive_odbc.conf? I only want answered calls so I have tried filter disposition => ANSWERED and filter disposition != ANSWERED bit it doesn't appear to work
20:21.43seanbrightAsteriskRoss: can you pastebin your cdr_adaptive_odbc.conf file?
20:33.42AsteriskRossHi Sean, here it is https://pastebin.com/yLSMmN5v
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20:38.46seanbrighti thought i just glance at the code and tell you, but the code is pretty sucky
20:38.59seanbrightthe only thing i can think is that it is using the internal value for disposition instead of the pretty one
20:39.07seanbrightpotentially, that is
20:40.10AsteriskRossI also tried with billsec => 0 and != 0 but that doesn't work either
20:40.23seanbrightAsteriskRoss: try with 'filter disposition => 8'
20:40.29seanbrightif that doesn't work, i got nothin'
20:46.24AsteriskRossGood news! It works! Thank you very much once again :)
20:57.02seanbrightthat's... amazing
20:57.04seanbrightsure
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