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| 01:10.52 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.1.1, 16.15.1 (2020/12/22) 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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| 22:47.43 | Kobaz | soooo |
| 22:47.54 | Kobaz | still having sip-tls issues |
| 22:48.16 | post-factum | good! |
| 22:48.33 | Kobaz | hah |
| 22:48.33 | Kobaz | yeah |
| 22:48.45 | Kobaz | https://dpaste.com/8X6426HF4 |
| 22:48.51 | Kobaz | I'll paste up some sips as well |
| 22:49.01 | Kobaz | it just bails in the middle of a call |
| 22:49.13 | Kobaz | with two way audio.. asterisk just says 'i give up' |
| 22:56.01 | grummund | Hi, sip credentials are repeated in different contexts, e.g. "register => user:secret@host", then we have "host=...", "fromuser=...", "username=..." and "secret=..." |
| 22:57.14 | grummund | is there a way to declare these just once say as variables or constants which can be referenced? |
| 22:57.21 | Samot | The secret in the peer is for inbound registrations |
| 22:57.30 | Samot | The secret in the register string is for outbound registrations |
| 22:57.46 | Samot | Also, you shouldn't be using Chan_SIP anymore. |
| 22:58.02 | Kobaz | https://dpaste.com/GRDYQL8C8#line-739 |
| 22:58.14 | Kobaz | Samot: unfortunately, I might be needing to switch back to chan_sip |
| 22:58.30 | Kobaz | pjsip's support for multiple transports is broken in several situations |
| 22:59.10 | Kobaz | so, anyway... like 739 of that dpaste... asterisk just sends a BYE out of the blue |
| 23:00.12 | Kobaz | or maybe i'll need to set up kamillio to proxy |
| 23:00.24 | Kobaz | and have asterisk just use a single sip stack and sip-udp |
| 23:01.30 | Kobaz | call hangs up in exactly 30 seconds |
| 23:02.42 | Kobaz | sounds like some sort of timer |
| 23:08.10 | grummund | Samot: that's all noted, but is there a way to avoid the duplicates? |
| 23:09.11 | file | Kobaz: set external_signaling_port to 5061 on the transport? |
| 23:09.37 | Kobaz | oh |
| 23:09.44 | Kobaz | that might do it |
| 23:11.39 | Kobaz | made it past 30 seconds this time |
| 23:12.15 | Kobaz | file: thanks |
| 23:12.54 | file | yes, because now the softphone can send the ACK to Asterisk so Asterisk knows the 200 OK was received |
| 23:13.03 | Kobaz | right yeah makes sense |
| 23:16.59 | Kobaz | [2021-01-03 18:16:41.039] WARNING[29831]: pjproject: <?>: SSL 6 [SSL_ERROR_ZERO_RETURN] (Read) ret: 0 len: 65535 |
| 23:17.02 | Kobaz | any idea what that is? |
| 23:17.37 | file | when it comes to that stuff reading wireshark and having intimate knowledge of TCP is generally needed |
| 23:17.47 | Kobaz | yeah i can poke through that |
| 23:18.20 | Kobaz | i've gotten those 0 reads in pjsip or chan_sip, and across multiple sites with different network environments and all that |
| 23:18.25 | Kobaz | never really looked into it |
| 23:23.28 | Kobaz | although the biggest difference is that in pjsip it spits out an error, and in old asterisk chan_sip it drops the call |
| 23:38.36 | Samot | Kobaz: So is the call working now? |
| 23:38.46 | Samot | Kobaz: And show more around the SSL error |
| 23:40.03 | Kobaz | Samot: that's it... it's out of the blue... would need core debug to get something more |
| 23:40.51 | Kobaz | https://dpaste.com/7V3YKDZEF |
| 23:41.00 | Kobaz | But, the context isn't consistent |
| 23:41.23 | Kobaz | It can be during a call, after a call... before a call.... no call. registered.... and it might or might not happen |
| 23:41.41 | Kobaz | Samot: well... one useragent is working |
| 23:42.03 | Kobaz | Samot: testing with Acrobits cloudphone, sip-tls. Now trying microsip.. and microsip is dropping after 1.5 minutes |
| 23:42.21 | Kobaz | cloudphone handled a 12 minute two way audio call no issue |
| 23:43.04 | Kobaz | more like ~1 minute actually. looking at the actual durations |
| 23:43.18 | Kobaz | 1 minute 4 seconds.. drop... and 1 minute 6 seconds drop |
| 23:44.56 | Samot | Show your transport for this |
| 23:46.52 | Kobaz | https://dpaste.com/39VR2GWMX |
| 23:48.47 | Kobaz | microsip definitely having an issue, hmm |
| 23:55.20 | Samot | sslv23? |
| 23:55.28 | Samot | Why is this set to sslv23? |
| 23:56.01 | Kobaz | Oh, I don't know... Pulling from examples |
| 23:56.12 | Samot | But that's what you have configured? |
| 23:56.13 | Kobaz | This is my first serious dive into asterisk tls |
| 23:56.22 | Samot | OK, SSL is dead. |
| 23:56.24 | Kobaz | Yeah it's configured that way |
| 23:56.25 | Samot | It's TLS now. |
| 23:56.33 | Kobaz | k |
| 23:56.36 | Kobaz | yeah makes sense |
| 23:56.38 | Samot | TLS1.2 is the current standard |
| 23:56.52 | Samot | Now if your softphones are current they should support that |
| 23:56.58 | Kobaz | that I know, but this is from an example... just literally trying a million things trying to get things to work |
| 23:57.06 | Samot | What example? |
| 23:58.27 | Kobaz | I've went through so many... I would have to do some google-fu |
| 23:58.37 | Kobaz | Anyway, let me set that to something more modern |