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| 00:04.37 | rpifan | well ive tried just about every setting on aserisk to get the softphone to work and it wont |
| 00:04.53 | rpifan | whats weird is that i use the same softphone and wifi to connect to a call centric number that works flawless |
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| 01:01.56 | igcewieling | You want to use fail2ban to block those sip scanning scripts. |
| 01:10.33 | rpifan | igcewieling, you are right i had it installed long ago |
| 01:14.28 | igcewieling | rpifan: here are some iptables rules, the important ones are SIP-SCAM |
| 01:14.31 | igcewieling | https://pastebin.com/50PsmMsN |
| 01:14.40 | igcewieling | SIP-SCAN that is. |
| 01:15.15 | igcewieling | that blocks some of the most common scanners |
| 01:16.29 | rpifan | thanks |
| 01:16.34 | rpifan | the current scammer is someone in NL |
| 01:16.43 | rpifan | there is an email to contact them if there is abuse |
| 01:16.46 | rpifan | i should do that |
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| 01:34.53 | DannyA | hi all. what is the method by which you would send a 302 to your carrier? |
| 01:36.03 | DannyA | like, a call comes in, and i "don't want the call" and want the carrier to "bounce" them somewhere else. how do i reply with a SIP REFER or a 302? |
| 01:36.05 | igcewieling | did you try googling for asterisk send 302 |
| 01:36.20 | DannyA | yes. i promise you i always try to find answers on my own before i come here. |
| 01:37.08 | igcewieling | Try the Transfer application with out answering |
| 01:37.11 | DannyA | please have a little more faith in me after all this time. i'm not someone who just immediately asks questions because i am lazy. i am respectful of and cognizant of the fact that when i ask something, people need to expend effort to answer. I'm not trying to waste anyone's time. I always appreciate this group. |
| 01:37.18 | DannyA | :-) |
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| 03:05.33 | DannyA | ok, another one i can't figure out: can i "dial into" an asterisk system by using a sip url with credentials? |
| 03:06.26 | DannyA | let's say i have a phone number with twilio, and when you dial that number it's programmed to forward your call to a SIP url. can asterisk accept a call that's not pre-registered the way an endpoint is registered at all times? |
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| 03:22.44 | DannyA | do i need to define an entry in sip.conf? |
| 03:38.23 | drmessano | Every call is to/from a SIP URI |
| 03:39.27 | DannyA | right. so if im dialing into asterisk from a vanilla client that just has a url field, what am i passing in? an extension with username/password that's defined in pjsip_wizard? |
| 03:40.17 | drmessano | If you choose to have a sip peer that requires auth, sure |
| 03:41.33 | drmessano | If I want to dial 101@yourpbx from elsewhere on the internet, you need to either allow anonymous SIP, or I need to match a peer on your PBX, by IP or auth |
| 03:41.59 | drmessano | registration has nothing to do with calls IN to your PBX |
| 03:42.32 | DannyA | gotcha |
| 03:42.55 | drmessano | A phone registered to your PBX registers to tell the PBX where to send the calls |
| 03:42.56 | DannyA | so as a test, if i want to NOT require auth, and allow anonymous, is that a config in the peer itself? |
| 03:43.11 | drmessano | When it dials, it's making a SIP URI call to the PBX |
| 03:43.17 | DannyA | so just, an entry in pjsip_wizard kinda like the endpoints are configured there? |
| 03:43.36 | drmessano | There a PJSIP option to allow guests |
| 03:43.56 | drmessano | and your default context must have a path to the extension somehow |
| 03:44.11 | drmessano | That's basically it |
| 03:48.50 | DannyA | ok. will try that now. thanks. |
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| 20:50.16 | *** 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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