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| 01:19.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); 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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| 09:46.50 | sotoz | hello |
| 09:47.22 | sotoz | how can I delete all cached media files from my filesystem the proper way? |
| 09:47.50 | sotoz | if I just delete them from /tmp then asterisk is complaining on its logs for every single item and the cpu goes 100% |
| 09:48.12 | sotoz | and there is no way to invalidate/delete all cache items from the CLI |
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| 10:17.51 | Gugge | sotoz: did you try core clear media-cache? |
| 10:18.20 | sotoz | Gugge: yes, you need to specify each file individually |
| 10:18.29 | sotoz | and I have 20k files that need to be deleted |
| 10:18.34 | Gugge | not accoding to the docs |
| 10:18.44 | sotoz | ? |
| 10:18.51 | Gugge | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Project+-+URI+Media+Playback#Asterisk14Project-URIMediaPlayback-Caching |
| 10:18.58 | Gugge | "3 items purged." |
| 10:19.16 | sotoz | this is not implemented |
| 10:19.21 | sotoz | its discussion about a feature |
| 10:19.46 | Gugge | then do all the files in a loop :) |
| 10:20.01 | sotoz | how? |
| 10:20.11 | Gugge | how do you clear a single file? |
| 10:20.53 | sotoz | media cache delete filename.wav |
| 10:21.29 | Gugge | for i in <all files>; do asterisk -rx "media cache delete $i"; done |
| 10:21.47 | sotoz | hmmm |
| 10:22.01 | sotoz | you're right |
| 10:22.06 | sotoz | stupid me :) |
| 10:23.41 | sotoz | thanks man :) |
| 10:23.56 | sotoz | I'll write a bash script and put it on a cronjob |
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| 10:29.40 | bittis | if you restart asterisk does the cache purge itself? |
| 10:33.19 | bittis | random question, is there any place i can get visio stencils from? |
| 10:33.28 | bittis | for asterisk |
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| 16:44.13 | jamesaxl | hi |
| 16:46.43 | jamesaxl | I would like to know if I do not inherit MixMonitor, the call transfer will create another file ? |
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| 16:52.44 | jamesaxl | [TK]D-Fender: Good evening |
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| 17:48.45 | ChannelZ | So, is the proper way to set a jitter buffer on the remote channel of an outgoing extension to use the 'b' option of Dial() to call a subroutine and use the JITTERBUFFER function from there? |
| 17:53.24 | [TK]D-Fender | You can't add options to a Dial that's already placed |
| 17:53.29 | [TK]D-Fender | the command was called |
| 17:53.36 | [TK]D-Fender | It's already a done deal |
| 17:55.20 | ChannelZ | "b(....): Before initiating an outgoing call, 'Gosub' to the specified location using the newly created channel." |
| 17:56.12 | [TK]D-Fender | So if you are actually calling that already .... then go use the function. |
| 17:56.38 | [TK]D-Fender | it executes against that channel. It will do whatever it does |
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| 18:09.23 | jamesaxl | I still do not understand how to configure asterisk to transfer calls correctly, for example 'Client' Calls to operator A and after the operator A transfer the calls to B. and I want MixMonitor to record 1st wav => from_client_to_operator_A.wav and 2nd wav => from_client_to_operator_B.wav |
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| 18:20.14 | jamesaxl | [TK]D-Fender: ping |
| 18:23.48 | [TK]D-Fender | I don't have your answer. If I did I've have said something |
| 18:24.57 | jamesaxl | [TK]D-Fender: thank you very much |
| 18:25.47 | radicaldev | anyone ever have issues with loading res_pjsip because of undefined symbol pjsip_tls_transport_start2 in asterisk 15? |
| 18:28.17 | radicaldev | jamesaxl: Probably need to make the feature code somehow call stopmixmonitor, then mixmonitor with a new filename |
| 18:28.26 | radicaldev | then do the transfer |
| 18:38.37 | gtrmtx | if i have inbound routes A, B, and C going to extensions A, B, and C respectively, is it possible to have one time group/condition that routes each of them independently to their respective locations? |
| 18:43.59 | [TK]D-Fender | gtrmtx, Keep FreePBX questions for their channel |
| 18:44.04 | [TK]D-Fender | GUI's aren't supported in here |
| 18:44.42 | gtrmtx | [TK]D-Fender, ah. what would be an example of something that would be supported? |
| 18:45.10 | gtrmtx | anything CLI? |
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| 18:51.04 | radicaldev | anything asterisk related. dialplans, features, application usage, etc. FreePBX automates in a FreePBX way, so it's best to ask them for help doing something in FreePBX |
| 18:51.24 | radicaldev | automates all of that* |
| 18:53.37 | gtrmtx | gotcha |
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| 19:37.14 | jamesaxl | radicaldev: thank you very much |
| 19:37.31 | jamesaxl | gtrmtx: stop using freepbx, it is creazy |
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| 19:50.07 | gtrmtx | jamesaxl, what would you suggest i use instead? |
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| 19:58.41 | Samot | FreePBX is fine. |
| 19:59.59 | jamesaxl | gtrmtx: just asterisk alone |
| 20:00.20 | jamesaxl | Samot: FreePBX does not support postgresql |
| 20:00.40 | Samot | OK. |
| 20:01.52 | Samot | It uses MySQL/MariaDB |
| 20:04.00 | Samot | For people, like me, that don't use postgresql it's not an issue. |
| 20:04.15 | Samot | I still use mysql with Asterisk |
| 20:05.11 | Samot | Using Asterisk vs something like FreePBX is a choice. |
| 20:05.20 | jamesaxl | Samot: I like native asterisk, And I like to write dials by hands. |
| 20:05.27 | Samot | OK. |
| 20:05.31 | Samot | Again, that is a choice. |
| 20:05.44 | jamesaxl | of course |
| 20:05.45 | Samot | But let's look at it another way.. |
| 20:06.13 | Samot | You're still using Asterisk. |
| 20:06.24 | Samot | It's just how you are managing it. |
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| 20:07.54 | jamesaxl | Samot: yep that it right, in my way, I like writing dials in file, while sips, queues in database, sometimes I prefer lua for dials sometimes just native |
| 20:08.40 | jamesaxl | As I am programmer I use #include a lot for dials. |
| 20:08.49 | Samot | OK. |
| 20:09.15 | Samot | You made the statement of not using FreePBX because "it's crazy" |
| 20:09.34 | Samot | It's not really a correct statement. |
| 20:10.35 | jamesaxl | Samot: I really meant GUI, because the icons are closed, and you can by mistake delete drunk if you want to edit it |
| 20:10.55 | jamesaxl | trunk* |
| 20:11.46 | jamesaxl | 2nd thing, people that are using Freepbx, they do not know what is asterisk. |
| 20:14.34 | Samot | That is also a highly incorrect statement. |
| 20:14.49 | Samot | I use FreePBX and I know Asterisk. |
| 20:15.38 | jamesaxl | you vs people. |
| 20:16.27 | Samot | [TK]D-Fender as well. |
| 20:16.50 | Samot | Quite a lot of people use FreePBX and have a strong knowledge of Asterisk. |
| 20:18.02 | jamesaxl | in France people speak french, but they are some people speak other language, it does not mean they not all France people speak french |
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| 20:22.19 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 20:22.49 | Samot | True. Just like all the people that use Asterisk does mean they all understand Asterisk. |
| 20:23.05 | Samot | s/does/doesn't |
| 20:29.19 | mub | Â I would say that most people using FreePBX are somewhat aware of the backend |
| 20:29.54 | mub | I mean, it says "Asterisk" right there on the dashboard |
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| 21:34.06 | drmessano | ;hah |
| 21:34.23 | drmessano | 15:11:46 <jamesaxl> 2nd thing, people that are using Freepbx, they do not know what is asterisk. <-- FALSE |
| 21:34.27 | drmessano | A great many do |
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| 22:25.20 | salviadud | When I do core show channels I get to see some channels that don't even exist |
| 22:25.41 | salviadud | Like a temporary chan for DAHDI/1-1 |
| 22:25.54 | salviadud | Does it clean itself up? |
| 22:26.00 | salviadud | It's weird... |
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