00:07.24 | *** join/#asterisk akp55 (~akp55@c-73-148-15-31.hsd1.va.comcast.net) |
00:18.58 | *** join/#asterisk Kritnich1 (~Kritnich@kritni.ch) |
00:19.58 | Zombie | I have a headache I'd like to discuss, I'm trying to extend my Asterisk server to include a family member I am in an Alliance with. |
00:20.24 | Zombie | He has his own Asterisk Server like mine. |
00:20.49 | Zombie | We have a StrongSwan IPSec VPN between us. |
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00:43.21 | Samot | Ok |
00:45.41 | Samot | Zombie |
00:45.57 | Samot | Was there more that? |
00:46.08 | Samot | +to |
00:47.15 | Zombie | Yes, we have a situation where I can hear him, adn he can't hear me. |
00:51.14 | Samot | Need to see a call with no audio |
01:02.04 | Zombie | I cannot call him at all. |
01:02.11 | Zombie | Hang on. |
01:08.02 | Zombie | https://pastebin.com/pGZRF22C Here is an attempt to call the Allied network. |
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01:15.57 | Samot | Is that a completed call with no audio? |
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01:31.20 | Zombie | Thats a rejected outgoing call. |
01:32.34 | Zombie | Its not going through because the Allied Asterisk server has not registered those devices. |
01:32.43 | Zombie | <PROTECTED> |
01:32.51 | Zombie | So they get dropped. |
01:40.14 | Samot | OK |
01:40.26 | Samot | That is unrelated to the no audio. |
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01:57.47 | Zombie | I have two separate problems |
01:57.55 | *** join/#asterisk nonasuomy (~nonasuomy@kodi/staff/TeknoJuce) |
01:58.53 | Zombie | Samot: I believe I have a wrong DID Number on Outgoing calls. |
02:00.15 | Zombie | [2021-01-28 20:56:59] ERROR[8558]: res_pjsip.c:3593 ast_sip_create_dialog_uac: Endpoint 'codenet': Could not create dialog to invalid URI 'sip:172.16.253.30/7100'. Is endpoint registered and reachable? |
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02:16.06 | Samot | That is invalid |
02:16.53 | Samot | Why is there a / in that? |
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02:26.02 | Zombie | That was at the other guy's request |
02:26.35 | Zombie | It's since been fixed. |
02:32.35 | Samot | So why show it? |
02:34.44 | Zombie | I hadn't figured out he goofed. |
02:34.59 | Zombie | I can show you my pjsip.conf |
02:35.06 | Zombie | Would that help? |
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02:36.57 | Zombie | https://pastebin.com/xCjmWE34 |
02:37.26 | Samot | No |
02:42.42 | Samot | Are you going to provide something relevant to the no audio issue you asked about over 90 minutes ago? |
02:43.07 | Zombie | Hold on. |
02:43.27 | Samot | Uhm |
02:43.52 | Samot | Ive been waiting for 90+ minutes. I have been holding on.. |
02:44.02 | Zombie | I'm trying to get the other guy to play ball. |
02:44.25 | Zombie | He is what the hold up, is as he is not following orders. |
02:44.38 | Samot | Ironic |
02:45.11 | drmessano | lol |
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02:58.46 | Samot | And why do you need both inbound and outbound auth while using identity to match against an IP? |
02:59.07 | Zombie | We got it working |
02:59.16 | Zombie | zBut I still can't call him. |
02:59.27 | Samot | First, you shouldn't really be doing both inbound/outbound auth on the same endpoint. |
02:59.42 | Samot | Second, you're doing IP auth. |
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03:01.45 | Samot | Zombie: You need to show a failed outbound call. Not with verbose debugging but with the pjsip logger on so we can see what is happening. |
03:01.56 | Samot | Zombie: Please don't take forever to provide this. |
03:03.39 | Zombie | Understood. |
03:03.47 | Zombie | I'm getting the other guy coming |
03:05.33 | Zombie | He can't get into the channel because he isn't registered and the channel is +R |
03:06.17 | Samot | He is not needed for you to make a failed called |
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03:15.00 | Zombie | There he is. |
03:18.37 | Zombie | https://pastebin.com/y81chF0N |
03:18.45 | Zombie | There you go, Samot. |
03:20.33 | Samot | 10:01:44 PM <Samot> Zombie: You need to show a failed outbound call. Not with verbose debugging but with the pjsip logger on so we can see what is happening. |
03:20.46 | Samot | What was that you said earlier about people not following directions? |
03:21.13 | Zombie | I enabled the pjsip logger and shuiut off the debugger |
03:21.19 | Zombie | Let me re-check. |
03:21.26 | Samot | That's not what you just pasted. |
03:22.51 | Zombie | core set debug 0 |
03:22.51 | Zombie | Core debug was 1 and is now OFF. |
03:24.09 | Zombie | pjsip set logger verbose on |
03:24.09 | Zombie | PJSIP Logging to verbose has been enabled |
03:24.16 | Zombie | It gives me no output. |
03:24.55 | Samot | pjsip set logger on |
03:25.02 | Samot | That's it. |
03:27.09 | codebase7 | try calling me. |
03:27.28 | codebase7 | Err.. Sorry wrong window. |
03:27.50 | Samot | Well that's what he should be doing |
03:27.56 | Samot | With the proper debug tools enabled. |
03:28.35 | Zombie | https://pastebin.com/ykSu2Ttk |
03:28.46 | Zombie | There. |
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03:30.00 | Samot | Why are you authing the calls? |
03:30.16 | Samot | This is a direct peer between two system over a VPN. |
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03:30.39 | Samot | Because this is failing to auth properly with user details. |
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03:31.16 | Zombie | I don't want him to be able to make calls remotely using my ITSP. |
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03:32.07 | Samot | How can he? |
03:32.14 | Samot | You don't allow it. |
03:32.29 | Samot | But this is *HIS PBX* auth YOUR CALL |
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03:32.48 | Samot | HIS PBX is challenging the call and you're failing to auth properly. |
03:32.53 | Zombie | He can call my PBX, |
03:33.00 | Samot | Sigh |
03:33.04 | Samot | I f'ing know. |
03:33.13 | Zombie | but how am I failing to authenticate? |
03:33.16 | Samot | You are sending a call to HIM |
03:33.25 | Samot | Correct? |
03:33.28 | Zombie | Yes. |
03:33.30 | Samot | You're trying to call him? |
03:33.32 | Samot | Right |
03:33.34 | Samot | And that doesn't work |
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03:33.39 | Zombie | Yes. |
03:33.45 | Samot | It doesn't work because HE is challenging the call for auth |
03:33.49 | Samot | And it is failing. |
03:34.00 | Samot | This is due to a configuration of the other PBX |
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03:34.23 | Samot | Or you not having the right outbound_auth details. |
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04:20.30 | Zombie | Samot: We are arguing amongst each other. |
04:21.40 | Samot | We are? |
04:26.20 | Zombie | Codebase and I, |
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04:34.36 | drmessano | ð¿ |
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07:13.14 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.0, 16.16.0 (2021/01/21) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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13:04.12 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.0, 16.16.0 (2021/01/21) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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19:11.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.0, 16.16.0 (2021/01/21) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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21:12.11 | catphish | does anyone happen to have any experience persuading shoretel phones to work with asterisk? |
21:43.10 | Samot | Nope |
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22:42.15 | catphish | i got a little stuck because they have a proprietary undocumented provisioning config file, hopefully i can grab one from an original provisioning server, will see how far i get |
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