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| 00:18.37 | *** 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: 13.28.1 (2019/09/05) 16.5.1 (2019/09/05), Security Only: 15.7.4 (2019/09/05); 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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| 03:02.22 | Katty | Hmm |
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| 04:28.34 | [TK]D-Fender | Katty: Mew. |
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| 07:06.14 | Reinhilde | Mew. |
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| 09:42.10 | p1ke_ | An operator is trying to connect to my Asterisk server and the operator's server sees my server offline because in OPTIONS I send the following header Accept-Encoding: text/plain. Is there any way to make Asterisk stop sending that header? I'm using Asterisk 13 |
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| 09:54.51 | sibiria | sounds like you're using pjsip |
| 09:55.05 | sibiria | chan_sip doesn't send that header. no idea if you can configure pjsip to not send it along |
| 09:56.31 | sibiria | accept-encoding is a standardized response to an OPTIONS request |
| 09:56.32 | sibiria | it should be in there |
| 09:56.39 | sibiria | maybe ask your provider to fix their shit(tm) |
| 09:59.21 | p1ke_ | Yes I'm using pjsip and also tried to decrypt the SIP RFC and it looks like accept-encoding should be in the OPTIONS request. I don't think that anything will happen if I tell the operator to fix their shit lol. Thanks anyway! :) |
| 10:02.41 | sibiria | maybe stick around to see if someone who knows pjsip better than me has any ideas |
| 10:03.31 | file | there is no config option, but it's a single line in the code: https://github.com/asterisk/asterisk/blob/master/res/res_pjsip/pjsip_options.c#L248 |
| 10:04.35 | p1ke_ | Thank you so much! I will recompile Asterisk without that line of code |
| 10:31.43 | karelk | I have upgraded to asterisk 16 |
| 10:32.00 | karelk | I have notices, these 3 modules don't exist anymore: |
| 10:32.02 | karelk | bridge_multiplexed.so |
| 10:32.02 | karelk | chan_bridge.so |
| 10:32.02 | karelk | func_audiohookinherit.so |
| 10:32.21 | karelk | have these been replaced by other modules ? |
| 10:33.06 | file | they were replaced by core things, or aren't applicable anymore. |
| 10:40.44 | karelk | so everything should work without these modules ? |
| 10:40.52 | file | yes. |
| 10:40.54 | karelk | or do I need to change my .conf files as well ? |
| 10:41.15 | file | any applicable upgrade changes would be in the UPGRADE.txt document |
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| 14:18.04 | qakhan | Hi all, when i do "core show channels verbose" there is channel stuck. here is an example https://pastebin.com/4rBkj1HN |
| 14:18.20 | qakhan | I have to manually hangup it |
| 14:18.40 | qakhan | what is the reason of stuck channel |
| 14:26.20 | Samot | qakhan: Silly question but did you look at the channel? |
| 14:27.05 | qakhan | what is silly about the question ? |
| 14:28.53 | Samot | Because it should have been part of the debug details you just provided. |
| 14:29.04 | Samot | I was being sarcastic. |
| 14:31.14 | qakhan | channel does not stuck always. it is random |
| 14:31.49 | qakhan | so i dont know which call will be stuck or not. |
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| 14:32.21 | Samot | Then you need to find out that is and then show us stuff. |
| 14:32.31 | Samot | Showing use non-stuck channels is pointless. |
| 14:34.15 | karelk | does the asterisk CLI support some kind of "command line editing" ? |
| 14:34.26 | karelk | I mean can I use some useful keyboard shortcuts as I would in bash ? |
| 14:34.51 | karelk | for instance: history-search-forward |
| 14:35.11 | qakhan | Samot i am showing you stuck channel |
| 14:35.12 | karelk | ctrl + w |
| 14:35.35 | qakhan | dont you see duration? its been 67:53:11 |
| 14:35.56 | Samot | Sigh. |
| 14:36.08 | Samot | Dude, then show us the channel details. |
| 14:36.51 | igcewieling | You don't set a max duration on all your outgoing calls? |
| 14:37.27 | Samot | Well more importantly, you're provider doesn't? |
| 14:37.27 | qakhan | no |
| 14:37.41 | Samot | 12 hours is like the defacto. |
| 14:37.49 | igcewieling | I use 6 hours for an absolute timeout. |
| 14:37.54 | Samot | Right. |
| 14:38.05 | Samot | I don't even go that high for some customers. |
| 14:38.25 | igcewieling | qakhan: I'd also look at the physical phone since Asterisk thinks it is still on the call. |
| 14:38.59 | Samot | Yeah but this is his little setup with all the AMI management of calls. |
| 14:39.11 | Samot | Who knows if this channel is still being managed by the phone at this point. |
| 14:39.14 | igcewieling | Oh, wow. he is screwed. |
| 14:41.16 | Samot | I'm still waiting for the channel details that I first implied should have been provided with the other output and then flat out asked for. |
| 14:42.13 | igcewieling | Samot: stop helping until the info is provided. |
| 14:49.22 | kharwell | karelk: It might depend on the backing shell? I've found some bash ones to work. Like ctrl+a (goto start of line), ctrl+e (goto end of line), ctrl+k (kill/cut after cursor), ctrl+y (yank/paste line), ctrl+p (previous command), ctrl+n (next command). Others? |
| 14:49.47 | kharwell | unfortunately ctrl+r doesn't work, nor ctrl+d |
| 14:49.50 | igcewieling | um, asterisk uses gnu readline for cli |
| 14:52.23 | karelk | igcewieling, gnu readline is using /etc/inputrc as config file |
| 14:52.27 | karelk | and that works in bash |
| 14:52.33 | karelk | but not in asterisk |
| 14:52.51 | igcewieling | interesting. you might want to look a the asterisk code |
| 14:52.58 | karelk | for instance history-search-backward works in bash , but no in asterisk |
| 14:53.15 | karelk | same with crtl+r |
| 14:54.38 | kharwell | yeah not all of the commands work, and I haven't looked into why |
| 14:55.11 | karelk | another thing, I am trying to make asterisk prompt red |
| 14:55.13 | karelk | export ASTERISK_PROMPT="%C31%H*CLI%#" |
| 14:55.26 | karelk | but I get just the escape codes, instead of color |
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| 14:59.15 | karelk | my prompt looks like this: [1;31mvoip*CLI>[1;0m |
| 14:59.19 | karelk | instead of red |
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| 15:38.04 | igcewieling | If anyone is looking for PoE injectors, these people have some inexpensive ones. https://www.adafruit.com/category/342 |
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| 18:53.50 | FinboySlick | Hello people. I haven't used asterisk in a while but I have a project that might call for it. Would it be possible with something like chan_alsa to monitor an audio source for noise (noise detection) and forward that noise to a call if it is detected? |
| 18:54.36 | FinboySlick | I was going for custom code and what not at first but I figured all the basic elements of what I'm trying to do pretty much exist in Asterisk already. |
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