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| 00:23.33 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 09:01.27 | LiuYan | https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure <-- This table struction seems old (such as `nat` field), is there an updated version of realtime tables ? |
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| 15:54.52 | CJSL | anyone have any recommendations for DID service provider ? |
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| 15:57.05 | Reinhilde | callcentric for small users, anveo direct for large users |
| 15:57.25 | CJSL | thank you! i will check out callcentric :) |
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| 21:32.06 | Reinhilde | Emergent one way audio. Only configuration change is restarting Asterisk (because it got stuck) |
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| 21:36.37 | Samot | That is vague. |
| 21:37.13 | Samot | Please provide more details of the issue. |
| 21:39.16 | Reinhilde | What I know is that it worked before I had to restart Asterisk (because it was having trouble with the DNS server - hostname lookups were failing, so it), and now it doesn't work. I've had this problem before, but it was when I was using a SIP proxy. I've stopped using that (now I use Asterisk directly). |
| 21:40.22 | Reinhilde | What I also know is that the remote host does not see the address of the Asterisk in the SDP, and for all intents and purposes appears not to receive the SDP what-so-ever. |
| 21:42.01 | Samot | So this is inbound one way? |
| 21:42.12 | Samot | Which side can't hear the other side? |
| 21:43.02 | Reinhilde | I can't hear Asterisk. I have to assume Asterisk can hear me. |
| 21:43.14 | Reinhilde | but that's not an assumption I should really make |
| 21:44.37 | Reinhilde | Confounders include a change in nat= mode (so I should probably mention actually that the third config change was a new user account) |
| 21:44.56 | Reinhilde | sip reloads on my system typically take upwards of 8 minutes |
| 21:46.01 | Samot | Then you have something seriously wrong with your system. |
| 21:47.20 | Samot | Is this phone on the same local network as Asterisk? |
| 21:53.26 | Reinhilde | No. |
| 21:53.36 | Reinhilde | It appears that the issue is that the 'nat=' field did not include the term 'comedia.' |
| 21:53.42 | Reinhilde | (I'm using chan_sip, not pjsip) |
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| 21:54.07 | Reinhilde | bonjour [TK]D-Fender |
| 21:57.54 | Samot | And have you checked the router where the phone is? |
| 21:58.01 | Samot | You're having one way audio issues at the phone. |
| 21:58.11 | Samot | That generally means you have a NAT issue there. |
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| 22:09.59 | Reinhilde | Samot: well it's working now. |
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| 22:27.19 | Samot | Yes, intermittent one way audio is an indicator of NAT issues where the phone is at. |
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