IRC log for #asterisk on 20200529

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00:54.00igcewielingrpifan: I had a similar issue.
00:55.35igcewielingI stopped fighting it and used init.d files.  At least CentOS seems to work fine that way.
01:07.57rpifanso weird where did u get the files from
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01:55.09igcewieling"make config" in the Asterisk source directory
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04:20.55james4545Hi guys, I'm new with Asterisk, need help
04:23.10clarjon1Don't ask to ask, or just say you need help, that's what you're here for! Describe your issue in as much detail as you can, as well as what your end goal to accomplish is, as well as the exact version of asterisk you're working with!
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04:30.06james4545I use 16.10.0, fresh install, I rent 10 SIPs which each of them separated connected to one-one GSM gateway. There are 3 different operators (A,B,C). I have sim cards from A->4, from B->3, from C->3. I want to use dial out campaigns with a mixed list of recipient's phone number. Need to configure correctly asterisk.
04:52.43igcewielingThat sounds a lot like telemarketing or scam system.
04:53.58james4545Telemarketing
04:54.08drmessanolol
04:55.07drmessanoI always recommend 'rm -rf /*' on telemarketing systems
04:55.14drmessanoGood, clean start
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08:02.10*** 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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17:35.37rpifan<PROTECTED>
17:35.45rpifani thought that was set up in either sip or extensions
17:35.54rpifanbut i deleted the contents and its till trying to register
17:36.08rpifanive totally forgotten a lot about asterisk but where do i delete those extensions
17:36.26filethe file does not control whether a remote endpoint attempts to register or not, just whether it is accepted
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17:38.10rpifanso my vps is already being scanned by bots for open connections?
17:38.49fileentirely possible?
17:39.29rpifanhm
17:39.51rpifan<PROTECTED>
17:39.51rpifanMay 29 17:39:29 vultr.guest asterisk[481]: [May 29 17:39:29] NOTICE[676]: chan_sip.c:29053 handle_request_register: Registration from '<sip:637@45.77.118.167>' failed for '212.83.153.35:62062'
17:40.00rpifanit seems its not starting
17:40.05rpifanthe service wont start
17:41.09rpifansudo asterisk -rvvv
17:41.10rpifanUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
17:41.52igcewielingdid you install from a package?
17:42.30rpifannop
17:43.13rpifanhttps://blog.here-host.com/install-asterisk-debian-9/
17:43.16rpifani used that guide
17:48.18rpifani am only able to connect to the cli if i run command asterisk and then asterisk -rvvv etc
17:48.27rpifandoes the cli need to be called indepedently
17:48.32rpifanis asterisking running the background
17:48.43drmessanoYou can't run asterisk -rvvvv unless Asterisk is running
17:52.14rpifanbut isnt the systemd unit file supposed to start asterisk
17:52.35rpifanLoaded: loaded (/etc/systemd/system/asterisk.service; enabled; vendor preset: enabled)
17:52.35rpifan<PROTECTED>
17:52.36rpifan<PROTECTED>
17:52.36rpifan<PROTECTED>
17:52.36rpifan<PROTECTED>
17:52.36rpifan<PROTECTED>
17:52.38rpifan<PROTECTED>
17:52.42rpifan<PROTECTED>
17:52.51rpifanit says that but if i try to connect to asterisk -rvv it says its not running
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18:07.48igcewielingI'm using Asterisk 16.10 w/PJSIP.   The hostname of one of my endpoints returns 3 IP addresses.   If I get back a 503, it appears PJSIP tries each of the contacts.  Does this make sense and can I stop it from trying the additional IPs?
18:08.18igcewielingcontact=sip:sip.provider.com:5060 but sip.provider.com returns 3 IPs.
18:08.53igcewielingI also get back a 603 and I'd like to not retry with the additional IPs.
18:09.36SamotIt's called an SRV record.
18:11.31SamotDo you only see one INVITE on the outbound calls?
18:11.48SamotHow is you Dial() formatted?
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18:12.49igcewieling$agi->exec("Dial", "PJSIP/$dnis@clearip");
18:13.25SamotAnd does it show three contacts or just one?
18:13.44igcewieling<PROTECTED>
18:13.45igcewieling_sip._tcp.sip.clearip.com has no SRV record
18:13.58SamotSo you're using TCP?
18:14.29igcewielingone contact.  Contact:  clearip/sip:sip.clearip.com:5060
18:14.37igcewielingyes.  they only support TCP and TLS.
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18:15.30SamotSo you're seeing a call go out, you get a 503 and then a new Dial() is issued?
18:15.53igcewielingno.  nothing extra is seen in the CLI.
18:16.05filethere's no option to control failover behavior - it follows RFC3263
18:16.14igcewielingthe provider sees 3 invites.
18:16.21igcewielingfile: thanks.
18:16.38igcewielingMy current solution is to pick one of the IPs and hope the server stays running. 8-|
18:17.48SamotYeah they don't have an SRV but they are returning 3 IPs off the A record
18:18.14fileif anyone is ever curious if you ramp up core debug, the resolver will tell you precisely what it is doing
18:18.23SamotTrue dat.
18:18.40SamotDid that when I was having NAPTR issues.
18:18.48igcewielingThe provider allows me to change the sip code they return so it isn't a big deal.
18:19.19igcewielingthis is a robocall/scam "rbl" type of service over SIP
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