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| 03:14.14 | snuff-work | mm.. i know we can 'write' to a file from the dialplan using record.. is there a way to grab a list of 'files' in a location natively.. rather than a direct 'shell' ls cmd? |
| 03:18.06 | [TK]D-Fender | Not in direct dialplan |
| 03:18.11 | [TK]D-Fender | so sheel out, AGI, etc |
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| 03:24.18 | snuff-work | mm.. shame.. not too much effort i guess.. (basically i have a 'recording' extension, they decide if they happy with it then hangup.. i'd then like someone else to just listen to them and 'pick' one) |
| 03:36.08 | snuff-work | mm.. but system() only returns success/fail |
| 03:36.15 | snuff-work | not output =*( |
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| 04:40.14 | [TK]D-Fender | System isn't a function returning anything |
| 04:40.24 | [TK]D-Fender | SHELL is |
| 04:40.41 | [TK]D-Fender | but it's a question of what kind of results you expect to have to parse |
| 04:41.02 | [TK]D-Fender | Which makes this a job probably best suited to AGI |
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| 05:03.17 | snuff-work | mm i think i'll stick to standard dialplan.. string parse from 'ls' isn't that hard |
| 05:07.19 | [TK]D-Fender | Let us know how that works out... |
| 05:08.53 | snuff-work | mm.. not too bad with func SHIFT / POP.. |
| 05:09.01 | snuff-work | and a while loop |
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| 05:46.42 | snuff-work | yep.. set(array=${SHELL(ls /var/lib/asterisk/sounds/recordings | tr '\r\n' ',')}) then while($["${SET(var=${SHIFT(array)})}" != "") |
| 05:47.17 | snuff-work | can has nice playback while loop from directory etc =) |
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| 10:33.40 | Guest0193 | Hello I need ideas. Running asterisk 13.24 and have been having wierd app_queue behaviour. Suddenly from nowhere asterisk stops sending calls to the queue. The only thing I see in the logs are this => [2019-10-08 11:45:12] VERBOSE[23980][C-00005025] pbx_realtime.c: Executing [sipext010@from-voip-provider:10] Queue("SIP/trunk-000091e0", âInit,xX,,,300,,,,,") |
| 10:34.12 | Guest0193 | Then it goes silent in the phone. Trying to run asterisk -rx 'queue show Init' causes asterisk to freeze briefly. Module reload app_queue does not work. Only a service restart clears it. |
| 10:34.20 | Guest0193 | Not sure how I would proceed in this issue. |
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| 13:28.01 | Guest0193 | bump for justice |
| 13:34.36 | Reinhilde | no |
| 13:45.34 | Guest0193 | Kek |
| 13:48.31 | [TK]D-Fender | What's the question? |
| 13:51.45 | Samot | Guest0193: This isn't a forum. "Bumping for justice" will not help because those without a scroll buffer or joined after your question was asked, will not know the question. |
| 13:52.00 | Samot | As highlighted by [TK]D-Fender just now. |
| 13:54.24 | Samot | Guest0193: How long does it take after your service restart for the Queues to lock up again? |
| 13:54.52 | [TK]D-Fender | Oh he's bumping for someone else..... |
| 13:55.27 | Samot | No. |
| 13:55.43 | Samot | He asked a question about RealTime queues locking up like 3.5 hours ago |
| 13:56.25 | Samot | Unlike a forum which a bump would raise the topic to the top of a summary this doesn't have the benefit of the original question being available for new comers. |
| 13:56.35 | Samot | Who wouldn't have it in the history buffer. |
| 14:01.52 | sibiria | am i misremembering something about a change between Ast 13 and 16 introducing an option allowing the dial time of an outbound call (e.g. "WaitTime" in a call file) to be "reset" or extended when/if a 180/Ringing was returned? |
| 14:02.25 | [TK]D-Fender | He asked a question about "your" implying someone else. Just incorrectly worded? |
| 14:04.34 | Samot | 6:33:41 AM <Guest0193> Hello I need ideas. Running asterisk 13.24 and have been having wierd app_queue behaviour. Suddenly from nowhere asterisk stops sending calls to the queue. The only thing I see in the logs are this => [2019-10-08 11:45:12] VERBOSE[23980][C-00005025] pbx_realtime.c: Executing [sipext010@from-voip-provider:10] Queue("SIP/trunk-000091e0", âInit,xX,,,300,,,,,") |
| 14:04.34 | Samot | 6:34:14 AM <Guest0193> Then it goes silent in the phone. Trying to run asterisk -rx 'queue show Init' causes asterisk to freeze briefly. Module reload app_queue does not work. Only a service restart clears it. |
| 14:04.34 | Samot | 6:34:21 AM <Guest0193> Not sure how I would proceed in this issue. |
| 14:17.18 | [TK]D-Fender | I'll take that as a "yes". |
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| 16:43.58 | file | my PJSIP hour of fun at AstriCon is now at 52 slides |
| 16:52.09 | Reinhilde | fun! |
| 16:52.41 | file | added one about endpoint identification, since it differs from chan_sip in how it works/configuration/some other things |
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| 16:58.59 | hecman | whats the bset asterisk distro |
| 16:59.28 | file | Asterisk doesn't have a distro, it runs on various ones such as CentOS, Ubuntu, Debian, and more |
| 17:00.10 | hecman | i mean a distro made specifically for asterisk |
| 17:00.43 | Reinhilde | I personally run it on plain ol Debian :-) |
| 17:00.44 | [TK]D-Fender | FreePBX ISO |
| 17:00.56 | Reinhilde | [TK]D-Fender: what's cool? |
| 17:00.56 | [TK]D-Fender | NEXT!!@@!@ (c) BKW |
| 17:00.59 | hecman | [TK]D-Fender| i see that now, thanks |
| 17:01.23 | hecman | devuan is old debian |
| 17:02.08 | Reinhilde | no |
| 17:02.16 | Reinhilde | only debian is 'plain ol debian' |
| 17:02.22 | Reinhilde | yes, I use it with the red hat crap init |
| 17:03.15 | hecman | gotcha |
| 17:03.59 | hecman | what im trying to do is test pushing CEL data to mysql, but i know nothing about asterisk so far |
| 17:04.08 | hecman | i guess thats possible, right? |
| 17:06.02 | Reinhilde | i'd stick to basics until you're more up to snuff |
| 17:06.34 | [TK]D-Fender | CEL is already stored in MySQL last I heard... |
| 17:06.57 | hecman | interesting |
| 17:07.11 | file | yeah, for FreePBX I believe it is |
| 17:08.13 | hecman | but for a non FreePBX system (assuming its new enough) i would just need to add an extra config file? |
| 17:08.23 | hecman | and obviously setup the db and such |
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| 17:15.29 | cervajs2 | is it possible to use function curl and send POST with body as application/json? tnx |
| 17:16.57 | hecman | curl -H "Content-Type: application/json" -X POST -d '{}' "https://cervajs2" ????? |
| 17:17.21 | cervajs2 | it looks like CURLOPT(header)Â can be used only for read result |
| 17:17.36 | cervajs2 | i mean curl function in asterisk dialplan |
| 17:17.44 | hecman | oh, sorry :) |
| 17:18.04 | cervajs2 | https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_CURL |
| 17:22.17 | hecman | it does look like it will always do application/x-www-form-urlencoded ? |
| 17:26.48 | hecman | cervajs2| doesnt look like it has anyway to set a random header? https://github.com/asterisk/asterisk/blob/master/funcs/func_curl.c |
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| 17:28.32 | hecman | cervajs2| could you do it with a system command? |
| 17:29.54 | cervajs2 | probably yes |
| 17:35.46 | cervajs2 | maybe i can modify func_curl.c. it can be easy implement needed header - https://stackoverflow.com/questions/11973994/json-requests-in-c-using-libcurl/12938692#12938692 |
| 17:38.10 | hecman | easy besides compiling and deploying... |
| 17:40.40 | hecman | better to try system command first so you can avoid all of that |
| 17:44.21 | hecman | what are the Ram requirements for SNG7-FPBX-64bit-1904-2.iso |
| 17:44.44 | Samot | There are none. |
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| 17:48.56 | hecman | so, like 2GB of ram is enough? |
| 17:49.21 | hecman | setting up a VM |
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| 21:22.20 | *** topic/#asterisk by file -> Join us at AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.0 (2019/10/08) 16.6.0 (2019/10/08); 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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| 22:05.56 | bf_ | Stupid question: I have a SwitchPI and use RJ11 to connect its FXO port to the RJ11 on my router. Is this the right cabling for asterisk to automatically answer incoming calls? |
| 22:06.41 | bf_ | I have been trying out different combination of cables and adapter for two hours and now I am a bit lost :( |
| 22:10.11 | Samot | bf_: That's going to be a SwitchPI question. We can't answer it here. |
| 22:10.32 | Samot | But FXO/FXS is RJ11. |
| 22:11.34 | bf_ | Samot: Thanks for your response. They are using Asterisk GUI on the switchpi so I figured people might be familiar with it here, but thanks for the pointer |
| 22:11.48 | Samot | There is no Asterisk GUI. |
| 22:12.28 | bf_ | Oh, sorry. I saw "Asterisk-GUI - an Asterisk configuration interface" "Copyright (C) 2009-2011, Digium, Inc." in the source code |
| 22:13.42 | Samot | Yes, there used to be a GUI option back in the day. But it was more of an add-on choice. |
| 22:13.51 | Samot | The fact they are using something that old.. |
| 22:14.51 | bf_ | Haha, okay I did not know that |
| 22:15.33 | bf_ | But when I connect to the asterisk console it says version 13.20 which seems to be LTS |
| 22:16.25 | bf_ | Could you by chance point me somewhere that has a tutorial on how to configure asterisk to answer a simple hello world to incoming analog RJ11 calls? I just want to debug my setup and you sound like someone who could know of something :) |
| 22:17.31 | Samot | You have to look at the SwitchPI documentation. |
| 22:19.26 | bf_ | Hm it seems they focus their docs on using the GUI |
| 22:19.43 | bf_ | I'd be happy to follow some low-level commands as I know my way around configuring linux stuff |
| 22:20.03 | bf_ | Most tutorials seem to assume a sip config though |
| 22:21.12 | bf_ | *sip login data from your provider |
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| 22:37.52 | bf_ | Ok I found out that I can use lsdahdi to find out which of the 4 rj11 adapter slots are actually used |
| 22:37.56 | bf_ | thats something positive |
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