| 15:18.51 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:18.51 | *** 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: 13.26.0 (2019/4/4) 16.3.0 (2019/4/4), 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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| 16:28.22 | sibiria | MLC: you could play the set of digits with background/playback |
| 16:29.03 | MLC | if the delay is in the audio files, that wouldn't speed it up any |
| 16:29.46 | sibiria | no i wouldn't |
| 16:30.50 | sibiria | it* |
| 16:31.08 | sibiria | i thought you were shifting digits off of a variable and calling saydigits in succession |
| 16:31.19 | sibiria | saydigits itself can take a long string of numbers, too. same thing |
| 16:31.46 | MLC | I'm passing it a full 10 digit phone number |
| 16:32.22 | sibiria | then you're left with having to trim the audio as suggested |
| 16:32.37 | MLC | yes |
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| 18:25.10 | *** topic/#asterisk by bford -> 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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| 21:32.26 | Toerkeium | hello guys, any help on why this dialog ends with a "Not Accepted Here"? https://pastebin.com/vqgRiGXV |
| 21:32.54 | Toerkeium | extension 88888 registers within the context of the IVR, which has setup the digit 2 to a queue |
| 21:32.59 | file | not configured for WebRTC? |
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| 21:34.38 | Toerkeium | file: it's a script which was working with chan_sip, then I setup an extension with pjsip, and it's registering well.. the same happens if I setup the extension on a softphone |
| 21:35.14 | file | registration is the same whether it's WebRTC or not, but if the configuration isn't right then it'll fail for WebRTC and produce that response. |
| 21:35.52 | Toerkeium | file: but I've tested it with a softphone, the same result |
| 21:36.43 | file | I'd suggest providing the actual endpoint configuration. |
| 21:38.58 | Toerkeium | file, at the end is the endpoin configuration https://pastebin.com/GYZU3VzH |
| 21:39.46 | file | that configuration is not saneish |
| 21:40.06 | file | media_use_received_transport should be yes |
| 21:40.14 | file | you also have media_encryption specified twice |
| 21:40.31 | file | you also need ice_support |
| 21:40.33 | Toerkeium | sorry, what's saneish? |
| 21:40.44 | file | you also need use_avpf and rtcp_mux |
| 21:41.10 | Toerkeium | ouch, that many options, I'll change them now |
| 21:41.49 | file | https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients#ConfiguringAsteriskforWebRTCClients-PJSIPEndpoint,AORandAuth |
| 21:43.46 | Toerkeium | it's calling now file !! I get no audio, perhaps is the ice servers I've disabled in the script |
| 21:43.49 | Toerkeium | thank you very much! |
| 21:46.09 | Toerkeium | now with rtcp_mux it works |
| 21:46.33 | Toerkeium | I'll read about it, thank you, I could finish my day with some progress |
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