IRC log for #asterisk on 20210101

00:03.05*** join/#asterisk zopsi (zopsi@2600:3c00::f03c:91ff:fe14:551f)
00:08.47*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
00:39.00void09anyone around to help me set up my freepbx ? simple setup, dial in a SIP voip number (works) and then dial out through a SIM (set up as a SIP gateway, from a goip box)
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01:40.00n0tizI'd like to wish everyone a happy new years!
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11:56.14*** join/#asterisk n0tiz (~n0tiz@82-69-15-38.dsl.in-addr.zen.co.uk)
12:19.09*** join/#asterisk dongs (~dongs@max.butt.care)
12:24.57dongshmm, what can i do to debug peer to peer IAX2 connection? i've stopped asterisk and did nc -u -p 4569 and i can see the client trying to connect, i've done iax2 set debug on and i can see REGREQ and CTOKEN or wahtever, but client doenst seem to be registering.
12:25.14dongsi'm connecting android phone (iax2 softphone) -> asterisk by iax2.
12:29.37*** join/#asterisk n0tiz (~n0tiz@82-69-15-38.dsl.in-addr.zen.co.uk)
12:34.37dongsi actually don't understand what is the problem with IAX registration. i can see the incoming registration in iax2 debug but it doesn't generate a warning or notice or anything.
13:03.57*** part/#asterisk dongs (~dongs@max.butt.care)
13:16.16*** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net)
13:33.04void09Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
13:33.09void09goip32/goip               192.168.100.198                             Yes        Yes            5060     OK (2 ms)
13:33.26void09this is what i get when i do sip show peers in asterisk console
13:33.36void092 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]
13:36.33*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
13:43.18SamotWhat is the actual problem
13:45.25void09well, I have this in FreePBX
13:45.44SamotAnd?
13:46.14void09https://pastebin.com/raw/1GqcmcTy
13:46.30void09sip trunk
13:46.45void09no dialed number manipulation
13:46.54SamotWhat?
13:47.08SamotYou need to do that in dialplan
13:47.13void09in sip trunk settings. there are no dialed number manipulation settigns
13:47.22void09where is dialplan ?
13:47.37SamotThat sip config is junk
13:47.58SamotWell FreePBX manages that
13:48.27void09https://imgur.com/a/psm0bQC
13:48.33void09this is my freePBX config
13:49.17SamotThat is a screenshot of a trunk
13:49.18void09i have another pjsip trunk that works, I can dial it. Inbound route is set to IVR
13:49.20void09yes
13:49.25void09I am show you my config
13:49.26SamotJust one tab
13:49.40void09all 3 tabs
13:50.16SamotI see the tabs
13:50.28SamotWhere are the screenshots
13:50.39void09they're all 3 in the same link
13:51.03void09in IVR I have digit 3 bound to Destination "DISA"
13:51.13void09I enter my password, it's good
13:51.21void09I get dial tone
13:51.33*** join/#asterisk Dovid (~dovid@ool-4356e81f.dyn.optonline.net)
13:51.57void09then i try to dial the number and it says "Your call cannot be completed as dialed. Please check the number and dial again"
13:52.01void09what am I missing ?
13:52.09void09from goip32 I can make test calls just fine
13:52.42void09== Spawn extension (from-internal, 07, 7) exited non-zero on 'Local/07@from-internal-00000004;2'
13:52.42void09<PROTECTED>
13:52.42void09<PROTECTED>
13:52.42void09<PROTECTED>
13:53.04DovidI just joined but did you try looking at a SIP debug? Have a look at sngrep
13:53.28SamotWhy are there local channels involved?
13:53.41void09this is what it says on asterisk console
13:53.51void09local channels?
13:54.01void09PBX and GoIP box are on the same network
13:54.11void09192.168.100.198/199
13:54.37SamotHow is this call being made?
13:56.22void09well I enter 3 (the digit to access the DISA), then enter password 1234#
13:56.32void09then I just dial the number 07XX XXXXXX
13:57.18void09by dialing the voip phone number in the other sip trunk, which works
13:58.40void09https://imgur.com/a/KxRZv02
13:59.40SamotWhat other SIP trunk?
13:59.43void09IVR https://imgur.com/a/ivQio2u
13:59.59void09the other SIP Trunk (ClickPhone) is a voip number I added. I can dial it in no problem
14:01.02SamotShow an actual sip debug of the failed call
14:01.15void09just the text in the asterisk console ?
14:01.20void09or where do I see this debug
14:03.00void09https://pastebin.com/raw/6Zty3uKa
14:04.02void09missed the first line: https://pastebin.com/raw/S9PB4tdu
14:10.10void09just trying to call to sip clickphone, and go out through sip goip (the device with the sim card)
14:10.34void09spent 6 hours already on this and I can't figure it out, can't find tutorial
14:13.29*** join/#asterisk ckb (~ckb@unaffiliated/ckb)
14:23.19SamotThat's not a SIP debug
14:23.24Samotasterisk -r
14:23.28Samotsip set debug on
14:23.38SamotMake the call out the goip device
14:30.13*** join/#asterisk tripleslash (~triplesla@unaffiliated/imsaguy)
14:40.00Samotvoid09: Well?
14:58.56SamotWell it's good to see 2021 is keeping the classic hits rolling.
15:03.22*** join/#asterisk paulgrmn (~paulgrmn@c-98-250-183-21.hsd1.mi.comcast.net)
15:05.14*** join/#asterisk dongs (~dongs@max.butt.care)
15:05.47dongswhy is asterisk messages spamming in syslog?
15:06.04dongsin addition to /var/log/asterisk/message AND console?
15:06.40SamotWhat messages?
15:07.06dongsasterisk[675]: [Jan  2 00:06:38] #033[1;33mNOTICE#033[0m[804][C-00000096] and so on
15:07.11dongsincluding malformed ansi codes!
15:07.19dongsit looks like same contents thats going on console.
15:07.39dongsive checked logger.conf and made syslog => error (also tried none, etc) and no effect.
15:07.51dongsalso commented out syslog => completely, also same.
15:08.25SamotWAnt to show more than one line?
15:08.32SamotGive a bit more of context
15:09.17dongshttp://bcas.tv/paste/results/b1gJsX49.html
15:09.30dongsi know the cause of the errors, i don't mind.
15:09.36dongsi just do NOT want them in syslog
15:09.41dongsthey're already logged to asterisk messages.
15:10.02SamotWhat is causing the errors?
15:10.08SamotAnd why aren't you fixing it?
15:10.32dongssome idiots repeatedly scanning my SIP server trying to make calls from IPs that are behind nat or something and cant actually receive a SIP response back
15:10.45SamotSo why don't you block them?
15:10.55SamotInstead of turning off logging that tells you bad things are happening
15:10.59dongsrandom IPs all the time, I dont care enough about the spam, i just dont awnt it double logged
15:11.10dongsits already logged to asterisk's own log
15:11.15dongsi don't want the spam in SYSLOG
15:11.20SamotIt's not SPAM
15:11.29SamotIt's the system telling you have bad things happening
15:11.35SamotLogging is not spam.
15:11.40dongsof course it is, its spam beacuse /var/log/asterisk/messages already has same stuff!
15:11.43SamotYou could secure your network properly
15:11.46dongsand thats wehre I will see it if I wanted to see it
15:11.47SamotThat would stop this.
15:12.51SamotWhat do you have in the logger config?
15:13.05dongsconsole => notice,warning,error
15:13.07dongsthats all.
15:14.02dongsmy astlogdir points to /var/log/asterisk
15:18.06void09Samot: just a second : )
15:19.45void09Samot: https://hastebin.com/ezonehatig.yaml
15:20.53dongsso.... where do i check why its double logging to syslog...
15:21.42void09this is the output of the followinv events: dialing in the jsip voip number trunk. getting greeted by the IVR voice. pressing 3 to get to disa. entering password followed by "#". Just entering 10 digit phone number
15:27.58*** join/#asterisk CatCow97 (~mine9@c-73-96-109-206.hsd1.or.comcast.net)
15:30.19Samotvoid09: There's no call in that debug.
15:30.35void09well maybe that's the problem then ? :-s idk
15:30.38dongsSamot: i think i figured it out. systemd for some insane reason captures asterisk stdout and sends it to syslog when its running as a service.
15:30.49void09the chinese "engineer" from the goip support is not online now
15:30.52dongsso this is more of a dumb ubuntu/systemd problem than asterisk
15:31.02SamotSure.
15:31.11SamotI don't have that issue on Ubuntu.
15:31.24dongsyou are probably not using their packaged version either.
15:32.34SamotNope. I don't like being that far behind in Asterisk releases.
15:33.19dongsi just upgraded my stuff from something I setup > 10 years ago. s o i dont mind either way. my requirements are simple anyway.
15:36.22dongsgot it. had to set StandardOutput=null and StandardError=null inside the unit file. i can still connect with -r so all is good.
15:36.32dongsallright, problem solved, was not asterisk issue at all.
16:03.58dongsi doubt anyone cares but I've reported it as a bug to ubuntu here https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1909816 and I do think its a bug due to double-logging and not stripping ansi codes. but yeah whatever it works so problem solved.
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