| 00:11.54 | josefig | one question guys, how can I modify the default context and it doesn't matter If I modify something on web I want the context cannot be overrwrite it. Is it possible ? specifically i'm modifying something in all calls executed so I want manage something to all calls |
| 00:25.59 | *** join/#asterisk infobot (ibot@208.53.50.136) |
| 00:25.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.0 (2018/09/05) 16.0.0-rc2 (2018/09/12), Standard: 15.6.0 (2018/09/05); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 03:47.23 | [TK]D-Fender | josefig, your question doesn't make sense |
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| 04:36.14 | hexhaxtron | Hi! I got about 600 users on my system. Can I allow them all to use Asterisk and everyone have their own SIP address? |
| 04:38.37 | [TK]D-Fender | You already have them there... |
| 04:38.53 | [TK]D-Fender | Not sure what it is you're looking to do that you aren't already doing.... |
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| 05:15.33 | hexhaxtron | [TK]D-Fender, http://paste.debian.net/1042144/ |
| 05:15.42 | hexhaxtron | I can't start the daemon... |
| 05:17.41 | [TK]D-Fender | go run it manually and see where it craps out |
| 05:17.46 | [TK]D-Fender | <PROTECTED> |
| 05:22.57 | hexhaxtron | [TK]D-Fender, http://paste.debian.net/1042145/ |
| 05:23.52 | [TK]D-Fender | <PROTECTED> |
| 05:23.57 | [TK]D-Fender | more verbose |
| 05:27.11 | hexhaxtron | [TK]D-Fender, http://paste.debian.net/1042146/ |
| 05:27.18 | hexhaxtron | There is a Segmentation fault in the end. |
| 05:28.32 | [TK]D-Fender | I don't see that failure and a command prompt from where you left off |
| 05:29.03 | hexhaxtron | I'm going to try Asterisk from the repositories. |
| 05:31.30 | [TK]D-Fender | wellyou are also using an unreleased verions on an unreleased branch. There could be legitmate issues, but we haven't seen your current status in full. |
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| 10:13.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.0 (2018/09/05) 16.0.0-rc2 (2018/09/12), Standard: 15.6.0 (2018/09/05); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 16:39.18 | jhord | Anyone using 16.0.0-rc2, yet? I have an endpoint with a valid aor/contact that shows 'unavailable' for the endpoint, but 'avail' for the contact. Was wondering if anyone else had seen this. |
| 16:40.22 | file | There was a report against 13 for real time, but otherwise nothing has been reported for that |
| 16:41.25 | jhord | This has been working in 15.2.2. I know 15.4 changed some stuff with qualification but I thought that was just startup? |
| 16:41.38 | file | It was a complete rewrite of options support. |
| 16:50.42 | igcewieling | I'm stuck with v13 for a while 8-( |
| 16:51.52 | rmudgett | igcewieling: The OPTIONS rewrite is also in the last two 13 releases |
| 16:52.25 | igcewieling | oh! that is cool, I was just thinking of updating from 13.20 to 13.23 |
| 16:53.16 | igcewieling | I can upgrade past v13 when/if Sangoma updates their transcoding card drivers to support Asterisk > v13. |
| 17:00.01 | igcewieling | With chan_sip if there is an unknown option in sip.conf it is ignored. I see pjsip has stricter syntax, would unknown options be rejected? "It is important to know that PJSIP syntax and configuration format is stricter than the older chan_sip driver. When in doubt, try to follow the documentation exactly, avoid extra spaces or strange capitalization. Always check your logs for warnings or errors if you suspect something is wrong." |
| 17:01.36 | file | yes, when used in .conf then unknown are rejected |
| 17:02.06 | file | realtime is lenient, it filters out unknowns |
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| 17:15.49 | igcewieling | drat. |
| 17:23.21 | rmudgett | all the better to catch typos and stray dogs |
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| 19:07.45 | jameswf | Random question.... all these people on raspi (specifically the ones writing scripts).... WTF do they compile and not use packages |
| 19:26.52 | jhord | Custom options? We compile asterisk for specific features and have our own patchset. No packages for that. |
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| 20:26.55 | jhord | So I can see the AOR/endpoint come online and available for one extension, but when it tries to synchronize a 2nd endpoint, I see a message stating that the first extension has no AORs feeding it. |
| 20:27.40 | jhord | I don't see anything in between stating that the AOR changed state, but I think sip_options_synchronize_endpoint() is taking it offline when it runs for the 2nd extension. Is that intended? |
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| 21:41.54 | jonesmeier | hi, is it possible to log the PIN that has been received when someone tries to enter a meetme conference ? |
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| 22:30.41 | DapOrp | Anyone here familiar with IBM's Watson Text To Speech service? You feed it some text through their api and it returns a sound file? I'm having issues trying to employ that TTS service's returned file and getting Asterisk to play it out over the phone. |
| 22:33.09 | jonesmeier | not sure if you have looked at this: https://www.voip-info.org/convert-wav-audio-files-for-use-in-asterisk |
| 22:38.13 | DapOrp | hmm. yeah I'm prolly dealing with a format issue. |
| 22:41.43 | jonesmeier | are you on linux ? |
| 22:43.04 | DapOrp | yeah, it's linux. |
| 22:43.44 | jonesmeier | 'exiftool' or 'mediainfo' can give some info on what kind of file you have |
| 22:44.21 | jonesmeier | and ffmpeg should be able to do any kind of conversion, that's all i got.. |
| 22:46.54 | DapOrp | Thanks, I'll see if I can figure this out. :) |
| 22:47.04 | jonesmeier | ok good luck :) |
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