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00:54.00 | igcewieling | rpifan: I had a similar issue. |
00:55.35 | igcewieling | I stopped fighting it and used init.d files. At least CentOS seems to work fine that way. |
01:07.57 | rpifan | so weird where did u get the files from |
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01:55.09 | igcewieling | "make config" in the Asterisk source directory |
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04:20.55 | james4545 | Hi guys, I'm new with Asterisk, need help |
04:23.10 | clarjon1 | Don'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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04:30.06 | james4545 | I 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. |
04:52.43 | igcewieling | That sounds a lot like telemarketing or scam system. |
04:53.58 | james4545 | Telemarketing |
04:54.08 | drmessano | lol |
04:55.07 | drmessano | I always recommend 'rm -rf /*' on telemarketing systems |
04:55.14 | drmessano | Good, 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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17:35.37 | rpifan | <PROTECTED> |
17:35.45 | rpifan | i thought that was set up in either sip or extensions |
17:35.54 | rpifan | but i deleted the contents and its till trying to register |
17:36.08 | rpifan | ive totally forgotten a lot about asterisk but where do i delete those extensions |
17:36.26 | file | the file does not control whether a remote endpoint attempts to register or not, just whether it is accepted |
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17:38.10 | rpifan | so my vps is already being scanned by bots for open connections? |
17:38.49 | file | entirely possible? |
17:39.29 | rpifan | hm |
17:39.51 | rpifan | <PROTECTED> |
17:39.51 | rpifan | May 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' |
17:40.00 | rpifan | it seems its not starting |
17:40.05 | rpifan | the service wont start |
17:41.09 | rpifan | sudo asterisk -rvvv |
17:41.10 | rpifan | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
17:41.52 | igcewieling | did you install from a package? |
17:42.30 | rpifan | nop |
17:43.13 | rpifan | https://blog.here-host.com/install-asterisk-debian-9/ |
17:43.16 | rpifan | i used that guide |
17:48.18 | rpifan | i am only able to connect to the cli if i run command asterisk and then asterisk -rvvv etc |
17:48.27 | rpifan | does the cli need to be called indepedently |
17:48.32 | rpifan | is asterisking running the background |
17:48.43 | drmessano | You can't run asterisk -rvvvv unless Asterisk is running |
17:52.14 | rpifan | but isnt the systemd unit file supposed to start asterisk |
17:52.35 | rpifan | Loaded: loaded (/etc/systemd/system/asterisk.service; enabled; vendor preset: enabled) |
17:52.35 | rpifan | <PROTECTED> |
17:52.36 | rpifan | <PROTECTED> |
17:52.36 | rpifan | <PROTECTED> |
17:52.36 | rpifan | <PROTECTED> |
17:52.36 | rpifan | <PROTECTED> |
17:52.38 | rpifan | <PROTECTED> |
17:52.42 | rpifan | <PROTECTED> |
17:52.51 | rpifan | it says that but if i try to connect to asterisk -rvv it says its not running |
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18:07.48 | igcewieling | I'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? |
18:08.18 | igcewieling | contact=sip:sip.provider.com:5060 but sip.provider.com returns 3 IPs. |
18:08.53 | igcewieling | I also get back a 603 and I'd like to not retry with the additional IPs. |
18:09.36 | Samot | It's called an SRV record. |
18:11.31 | Samot | Do you only see one INVITE on the outbound calls? |
18:11.48 | Samot | How is you Dial() formatted? |
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18:12.49 | igcewieling | $agi->exec("Dial", "PJSIP/$dnis@clearip"); |
18:13.25 | Samot | And does it show three contacts or just one? |
18:13.44 | igcewieling | <PROTECTED> |
18:13.45 | igcewieling | _sip._tcp.sip.clearip.com has no SRV record |
18:13.58 | Samot | So you're using TCP? |
18:14.29 | igcewieling | one contact. Contact: clearip/sip:sip.clearip.com:5060 |
18:14.37 | igcewieling | yes. they only support TCP and TLS. |
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18:15.30 | Samot | So you're seeing a call go out, you get a 503 and then a new Dial() is issued? |
18:15.53 | igcewieling | no. nothing extra is seen in the CLI. |
18:16.05 | file | there's no option to control failover behavior - it follows RFC3263 |
18:16.14 | igcewieling | the provider sees 3 invites. |
18:16.21 | igcewieling | file: thanks. |
18:16.38 | igcewieling | My current solution is to pick one of the IPs and hope the server stays running. 8-| |
18:17.48 | Samot | Yeah they don't have an SRV but they are returning 3 IPs off the A record |
18:18.14 | file | if anyone is ever curious if you ramp up core debug, the resolver will tell you precisely what it is doing |
18:18.23 | Samot | True dat. |
18:18.40 | Samot | Did that when I was having NAPTR issues. |
18:18.48 | igcewieling | The provider allows me to change the sip code they return so it isn't a big deal. |
18:19.19 | igcewieling | this is a robocall/scam "rbl" type of service over SIP |
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