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| 06:02.25 | *** 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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| 16:21.22 | grummund_ | is clearing out sample conf files from /etc/asterisk |
| 16:22.38 | grummund_ | whould appreciate if anyone could take a look at this list - http://paste.debian.net/1181403/ |
| 16:23.21 | grummund_ | is there anything there worth keeping? |
| 16:23.30 | sibiria | yes, almost all of it |
| 16:24.15 | sibiria | if you "make samples" you'll get a decent starting point to alter from |
| 16:24.35 | sibiria | the default config files offer the bonus of being quite complete documentation as well |
| 16:27.27 | grummund_ | it's for a basic home setup with a few extensions and voicemail. |
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| 18:27.42 | igcewieling | Here is a reasonably minimal list of config files which I use. This is not the absolute minimum config files, bit most of the rarely used ones are not listed. https://pastebin.com/3ghzPK0Z |
| 18:29.37 | igcewieling | There are also a few extra like codec_sancoma which almost nobody uses, but I use it. |
| 18:33.44 | grummund_ | igcewieling: thanks. |
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| 19:56.07 | avb | igcewieling: just curious, is there any cons of codec_sangoma then old g729 codecs from asterisk.hosting.lv? |
| 19:56.49 | igcewieling | avb: well, codec_sangoma requires an expensive transcoding card. |
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| 19:57.14 | igcewieling | I got the cards back in 2008 or so, when there was no legal software g729. |
| 19:58.00 | igcewieling | I have no interest in running unlicensed codecs |
| 19:58.39 | avb | :) |
| 19:58.51 | avb | well, hw module would save you cpu |
| 19:59.05 | avb | software stuff always comes at a cost of the cpu resources |
| 19:59.18 | avb | regardless if its a digium paid license or free one |
| 19:59.41 | avb | this days g729 is not that actual |
| 19:59.46 | igcewieling | avb: indeed. These days virtually all of our customers have enough bandwidth to use ulaw or g722 so we don't have many g729 calls. |
| 20:00.02 | avb | everybody can run g711 over the worst internet connection |
| 20:00.47 | igcewieling | Almost everyone. I still have a few customers with a 1.544M T-1 running Frame Relay. |
| 20:00.57 | avb | the only person I kknow who use g729 is a friend of mine. he always travel to the weird countries where there is no decent internet but lot of nice and cheap girls lol |
| 20:01.05 | avb | omg :-D |
| 20:01.18 | sibiria | the sooner g.729 dies the better |
| 20:01.20 | avb | i havent heard about thos for a long time |
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| 21:14.12 | grummund_ | endpoint is defined with 'disallow=all' and 'allow=alaw,ulaw' |
| 21:14.28 | grummund_ | this is to receive direct sip calls |
| 21:14.56 | grummund_ | it works but is using gsm codec, why? |
| 21:15.18 | file | define "is using gsm codec" |
| 21:15.31 | grummund_ | ok. |
| 21:16.20 | grummund_ | codec_gsm.so and format_gsm.so must be loaded or the incoming call is rejected. |
| 21:16.40 | file | are you playing back sound files? |
| 21:17.09 | grummund_ | call to/from my other sip trunks work fine, it's just this incoming-only provider. |
| 21:18.17 | grummund_ | only if it would timeout to voicemail (if that's what you mean) but it doesn't get that far. |
| 21:18.30 | file | then you'd need to provide the actual console log |
| 21:18.35 | file | as well as a SIP trace |
| 21:19.57 | grummund_ | in theory then with 'disallow=all' and 'allow=alaw,ulaw' it shouldn't be doing that? |
| 21:20.19 | file | that controls the SDP negotiation for codec, yes |
| 21:20.30 | file | "it shouldn't be doing that" depends on what exactly is being done |
| 21:21.02 | file | it very well may be negotiating at ulaw, but you are doing something which results in a sound file being played which then transcodes from gsm due to a lack of ulaw sound files being present |
| 21:23.15 | grummund_ | that's unlikely, but i can check. |
| 21:23.18 | avb | i have a feeling that your allow/disallow parameter is not applied to this extension |
| 21:23.37 | file | I can only guess without the info I stated |
| 21:23.54 | grummund_ | sure, i'll paste something. |
| 21:32.39 | grummund_ | file: sorry that seems to be a red-herring, calls are now succeeding without codec_gsm.so or format_gsm.so loaded. |
| 21:36.14 | grummund_ | sometime |
| 21:44.26 | grummund_ | sip provider is sending from ip outside their published range that's why. |
| 21:55.10 | grummund_ | Is 'disallow=all; allow=alaw,ulaw' a reasonable choice to work with sip providers? or what is? |
| 21:56.13 | grummund_ | Also, these seem to be the required modules: codec_alaw.so, codec_a_mu.so, codec_ulaw.so |
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| 23:24.57 | igcewieling | It is almost never useful to have both ulaw and alaw enabled. |
| 23:25.46 | igcewieling | gruetzkopf: don't call prey to being a premature optimizer. |
| 23:26.06 | igcewieling | ..er.. gruetzkopf: don't fall prey to being a premature optimizer. |
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