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| 01:15.38 | gregf | Having a weird problem, I switched to tls/srtp. I got it working, so everything was connected, my endpoint was registered, but soon as my sip phone was switched over to srtp I had no audio. pjsip debugger looked normal, I could see a normal call flow from what I could tell. I just wasn't hearing anything. Does anyone know what could cause this, |
| 01:15.38 | gregf | where I should start looking? |
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| 01:35.13 | igcewieling | What might be the pjsip version of chan_sip's auth_options_requests= setting? |
| 01:35.26 | Kobaz | interesting question |
| 01:40.08 | igcewieling | I don't think it is related, but wanted to try it. I'm getting this when sending out an options request. I'm authing by IP only. https://paste.nyigc.net/view/3be640e2 |
| 01:45.24 | Kobaz | oh weird |
| 01:46.44 | igcewieling | I'll check my configs, maybe it is authing by something else |
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| 02:13.36 | Samot | authenticate_qualify |
| 02:13.36 | Samot | If true and a qualify request receives a challenge response then authentication is attempted before declaring the contact available. |
| 02:14.41 | igcewieling | thanks. I was searching for "option". |
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| 07:27.12 | Cyrillax | gregf: 1) Restart asterisk with debug=9 and verbose=9 in /etc/asterisk.conf 2) enable sip debugging with 'sip set debug on' and 'pjsip set logger on'. 3) place a test call and pastebin the logs, and I will take a look at it |
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| 13:50.35 | gregf | Cyrillax I ended up getting it late last night, My D60 needed media_encrypt="sdes" enabled in it's xml file. |
| 13:51.05 | gregf | Thank you though |
| 13:51.33 | Cyrillax | gregf np, well done :) |
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| 15:59.02 | igcewieling | I can't figure out if our sales people are sadistic bastards or just incompetent. |
| 16:00.54 | file | or both? |
| 16:05.09 | igcewieling | One of the scheduled a customer to be moved to our hosted service, but neglected to request the service actually be setup. Now we have a tech on the way to convert them over. |
| 16:18.32 | Samot | Sales Engineers. |
| 16:18.59 | igcewieling | That is what we need. |
| 16:19.25 | Samot | It's how we avoided the problems you have. |
| 16:19.46 | Samot | Which seems to be a break down in communication between the two departments. |
| 16:20.14 | igcewieling | In this case, the order should not have proceeded until the checklist was complete. That is a management issue. |
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| 23:25.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); 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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