| 00:00.23 | Helenah1 | I've set up a PBX, and when I try to auth, the console reports "Wrong password", we've checked and re-checked the configuration file for pjsip over and over again, and we've tried two different SIP clients and re-entered the password many times also. |
| 00:05.13 | *** join/#asterisk Katty (uid62315@gateway/web/irccloud.com/x-nnenvehgurktscex) |
| 00:06.04 | Katty | bloop |
| 00:06.49 | [TK]D-Fender | Mew. |
| 00:07.12 | Katty | hi fenderbender |
| 00:07.35 | [TK]D-Fender | Ms. Kitty-Katty |
| 00:07.38 | Samot | Helenah1: You need to show debugs. |
| 00:08.00 | Samot | Helenah1: Starting with the results from 'pjsip set logger on' |
| 00:08.07 | Katty | [TK]D-Fender: how've you been? |
| 00:08.19 | [TK]D-Fender | Katty, SSDD. You? |
| 00:08.45 | Katty | [TK]D-Fender: i've been fantastic (= |
| 00:09.10 | Katty | [TK]D-Fender: you still doing music stuffs? |
| 00:10.19 | [TK]D-Fender | Katty, Sound tech mostly. Played only 1 show all year personally.... |
| 00:11.36 | Katty | [TK]D-Fender: just not feeling up to it? too much work? |
| 00:12.55 | [TK]D-Fender | Issues with out drummer. |
| 00:13.10 | [TK]D-Fender | I'm working on 2 other groups right now to see about getting back out and playing... |
| 00:15.04 | wyoung | linux_user: GUI interfaces have nothing to do with Asterisk though. |
| 00:15.35 | [TK]D-Fender | wyoung, that topic left for #freepbx a while ago |
| 00:15.46 | [TK]D-Fender | wyoung, And headed further and further sideways |
| 00:16.22 | wyoung | [TK]D-Fender: hehe, OK :) Sorry, I just gotback so I was answering questions asked a few hours ago :) |
| 00:16.53 | Samot | wyoung: Let that be. It was taken to #freepbx and devolved from there. |
| 00:19.18 | wyoung | [TK]D-Fender: You are still extension 300 on my Asterisk setup. |
| 00:19.47 | [TK]D-Fender | wyoung, funny .. I just finished letting go of my domain name :) |
| 00:20.02 | [TK]D-Fender | wyoung, And I haven't really had a home PBX in quite some time |
| 00:20.30 | wyoung | :) so no more aocomputing? |
| 00:22.46 | [TK]D-Fender | Haven't needed it ina while |
| 00:26.13 | Katty | [TK]D-Fender: well i hope you can find a drummer |
| 00:26.29 | [TK]D-Fender | We've got a new one we're breaking in, but it's taking forever. |
| 00:26.53 | [TK]D-Fender | But in that band I'm on keys .. the 2 new ones I'm on guitar which I'd rather get going faster honestly... |
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| 01:37.31 | rue_mohr | Samot, well, ok, I'll go 64bit with it then |
| 01:38.20 | rue_mohr | so, to be clear, asterisk isn't supported on 32bit hardware anymore... yes? |
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| 02:03.13 | sibiria | rue_mohr: it is |
| 02:03.21 | sibiria | it builds and runs fine on a 32 bit platform |
| 02:05.11 | Samot | It's just everything else that is giving up on 32bit support. |
| 02:12.16 | rue_mohr | how about that error tho |
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| 02:41.00 | Samot | No clue. Havent done 32bit installs in years |
| 02:48.38 | rue_mohr | well, the dahdi kernel module doesn't compile for 32bit |
| 02:48.49 | rue_mohr | so I'd say somewhere support was dropepd |
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| 08:55.35 | Helenah1 | Hi, lets say I got two PBXs and I want to link them both together, is there a way to easily make all the extensions available one PBX available on the other? |
| 08:55.59 | Helenah1 | I wanted to do something like Dial(PJSIP/pbX_name/extension) |
| 08:56.09 | wyoung | Helenah1: do it! |
| 08:56.22 | Helenah1 | wyoung: I already have tried. |
| 08:56.33 | Helenah1 | Apparently it isn't valid. |
| 08:56.43 | wyoung | oh |
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| 10:23.03 | file | it's PJSIP/exten@pbx |
| 10:30.27 | Helenah1 | file: Was that in response to my question? |
| 10:30.58 | Helenah1 | Cause I'm banging my head trying to get two PBX's unified. |
| 10:31.12 | file | yes |
| 10:31.33 | Helenah1 | My issue is, is I'm trying to avoid a NAT situation by registering two boxes at both ends of a VPN tunnel. |
| 10:32.15 | Helenah1 | file: So I don't need to do a SIP registration to achieve this, I just dial a SIP address? |
| 10:33.21 | file | if the address is static you don't have to... |
| 10:42.07 | Helenah1 | file: Do I still need a SIP registration? |
| 10:44.13 | file | a SIP registration tells a remote server your IP address and port so you can be contacted |
| 10:44.23 | file | whether a SIP registration is needed or not depends upon the deployment |
| 10:44.39 | file | if your IP address and port is dynamic, then a registration would be needed |
| 10:44.49 | file | if it is static and doesn't change then it is not strictly needed |
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| 10:58.24 | Helenah1 | file: Thank you very much, things work! |
| 11:15.00 | Helenah1 | I'm sure I remember you could phone a SIP address without registering a client... |
| 11:15.12 | Helenah1 | I'm trying to do this to test the security of one of my PBX systems. |
| 11:27.20 | Helenah1 | Is my PBX open to the WAN? |
| 11:33.57 | TandyUK | only if youve opened the relevant ports (or dont have a firewall) |
| 11:35.43 | Helenah1 | TandyUK: I mean... can one phone exten@mypbx without being registered? |
| 11:36.01 | Helenah1 | to the PBX itself |
| 11:43.05 | file | registration is not required in SIP, depending upon configuration then dialing can occur |
| 11:44.57 | TandyUK | Helenah1: with your vpn you were talking about, if your pbx is 10.20.30.1, then depending if your configuration lets them, the other pbx could call ext201@10.20.30.1 |
| 11:45.57 | TandyUK | they might even map eg '501' as a speeddial to 'ext201@10.20.30.1' so 'other pbx' can call youre extension 201 by dialling 501 |
| 11:46.10 | TandyUK | but its all then up to what your pbx does when it recieves that invite |
| 11:46.25 | TandyUK | ie, how youve configured it.. you may have allowed it, you might not |
| 12:17.40 | Helenah1 | TandyUK: Is this what I need to be reading to ensure INVITE requests are not allowed? https://www.voip-info.org/asterisk-sip-insecure/ |
| 13:26.26 | TandyUK | i thought you said you wanted that |
| 13:26.54 | TandyUK | make your mind up... do you want your 2 asterisk boxes each end of the vpn to be able to call each others extensions, or do you want to force them to register to each other first? |
| 13:28.33 | TandyUK | i suspect you probably want insecure:yes on the context you create for the opposite ends asterisk |
| 13:28.50 | TandyUK | just dont open either of your asterisk boxes to the world period |
| 13:32.41 | Samot | Helenah1: You're using PJSIP, that link at voip-info.org is invalid for what you're doing. It's for Chan_SIP. |
| 13:44.10 | Samot | Helenah1: For PJSIP you'll want to use the Identify section with the match= setting for the IPs you want to allow to send calls. |
| 13:44.28 | Samot | Well, accept calls from. |
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| 15:33.00 | Samot | HAHAHAHAHAHAHAHAHAHAHA |
| 15:33.04 | Samot | OMG |
| 15:33.08 | Samot | drmessano: https://www.reddit.com/r/VOIP/comments/du7v6g/irresponsible_users/ |
| 15:33.41 | Samot | " irresponsible users who are leaving their phones off the hook and letting all of the dial tone leak out of the system." |
| 15:34.48 | Samot | I just hate when all my dial tone leaks out of the system. Takes forever to refill it. |
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| 15:48.06 | sibiria | it's also expensive |
| 15:48.21 | sibiria | it used to be cheap per gallon in the 90s, but not these days |
| 15:56.47 | drmessano | hahahhahah |
| 15:57.34 | Samot | I know right? |
| 15:57.40 | Samot | The replies are great too. |
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| 15:58.39 | Samot | "Don't worry, you can make your own fresh dial tone from scratch using 100% recycled electrons by installing sox on your computer" |
| 16:29.49 | drmessano | Sad thing is, I don't know if he's kidding or not |
| 16:31.48 | Samot | I know. |
| 16:31.53 | Samot | He did give an example. |
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| 18:46.51 | jkroon | is anyone else experiencing a problem with 13.29.1 ./configure? |
| 18:46.55 | jkroon | checking for uuid/uuid.h... yes |
| 18:46.56 | jkroon | ./configure: line 14364: syntax error near unexpected token `else' |
| 18:46.56 | jkroon | ./configure: line 14364: `else' |
| 18:54.21 | jkroon | never mind, seems to be autoconf issue. |
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