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| 00:18.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); 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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| 05:37.39 | russell-- | total newbie question: i want to do some testing of old analog dialup modems (one calling, one answering), i have an ooma "landline" now, and in my initial flailing, i couldn't successfully get my modem to train over that line. i got a grandstream gxw408 and tried port-to-port dialing, and also couldn't find a setting that allowed the modems to train. just picked up a sipura spa-2100 in a freebie bin, |
| 05:37.45 | russell-- | and after a factory reset, can't seem to get even a dialtone on the lines (but it does react to **** commands). any ideas what i'm doing wrong? |
| 05:38.40 | russell-- | gxw4008* |
| 05:41.28 | [TK]D-Fender | typically those devices only give dialtone if you're registered. |
| 05:42.54 | SwedeMike | russell--: it's not uncommon for VoIP to not support modems/faxes. |
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| 08:37.48 | alexises_ | Hello, I have a dunny question, asterisk agi did have timeout ? |
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| 09:31.22 | dan_j | I've got the following hanguphandler which stops mix monitor and then executes a php agi script to move and file the recording away. |
| 09:31.32 | dan_j | https://www.irccloud.com/pastebin/7Cz1Ayzf/ |
| 09:33.31 | dan_j | However, I noticed that occasionally, some recordings cannot be moved and yesterday saw that when StopMixMonitor() returns, it does not mean that it successfully stopped. It simply means that the request has been sent to the MixMonitor thread. |
| 09:35.22 | dan_j | So I added sleep(2); into the MoveRecording.php script, but no matter how long I set it to, "End MixMonitor Recording" always appears in the CLI and Logs after the MoveRecording.php has completed. |
| 09:36.08 | dan_j | I've also tried using the asterisk dialplan Wait() command after StopMixMonitor, but that causes the hanguphandler to exit without processing any subsequent commands. |
| 09:40.14 | sibiria | dan_j: you can send a command along with MixMonitor which will be executed after the recording is done. that might work for you. |
| 09:41.05 | sibiria | what we do is that we simply batch-move recordings from monitor/ when the files have been untouched for N minutes, by using a separate service script that trawls the directory with find(1) |
| 09:41.55 | sibiria | which is definitely a leaner method than firing up PHP once for every recording |
| 09:42.36 | dan_j | sibiria: MixMonitor is started by an AMI agent. I used to use the command option but can't remember why we removed it. |
| 09:43.46 | FarhaadN | hi all , how i can figure high cpu load issue on my asterisk? |
| 09:43.46 | FarhaadN | on debian 8 and asterisk 13.21.1 |
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| 09:48.40 | sibiria | dan_j: we do something along these lines: find /var/spool/asterisk/monitor/ -name "*.wav" -amin +3 |
| 09:49.06 | sibiria | to get a list of recordings that haven't been touched in 3 minutes, which is a clear sign they're done |
| 09:49.29 | sibiria | then the shell script just moves said files out of the way |
| 09:50.16 | dan_j | Thanks. I'll try it out. |
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| 11:02.59 | saint_ | after a couple of years of leaving asterisk, i m back in the game with a new partner ... who suggest freepbx over asterisk . for basic small installs (sip trunk, 20 phones, crm connected, ivr, voicemail, email notifs).. can freepbx handle it ? I know i would have more flexibility with asterisk.. but for small simple systems, any feedback welcome please |
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| 15:37.19 | [TK]D-Fender | saint_, That comparison isn't exactly valid |
| 15:37.27 | [TK]D-Fender | FreePBX is a GUI for CONFIGURING Asterisk. |
| 15:37.44 | [TK]D-Fender | So that's like asking, "What's better, a wheel, or a car?" |
| 15:37.46 | *** join/#asterisk robmal (r@wporzo.pl) |
| 15:38.25 | [TK]D-Fender | And the difference in the end result you can have is really self-explanatory |
| 15:39.01 | igcewieling | There might be hope yet. If enough insurance companies refuse to pay out for losses from hacking, maybe people still start actually caring about security "Hackers used phishing emails to break into a Virginia bank in two separate cyber intrusions over an eight-month period, making off with more than $2.4 million total. Now the financial institution is suing its insurance provider for refusing to fully cover the losses. According to |
| 15:39.02 | igcewieling | a lawsuit filed last month in the Western District of Virginia" |
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| 16:00.37 | jfindley | Hello, I'm trying to get transfers working in Asterisk 15. When I add t to the dial string, I get the 'transfer' prompt when I press #1, but as soon as I press another digit it tries to transfer to that digit when I really want to transfer to a 4 digit extension. How can I specify the number of digits I expect before transfering? The documentation is unclear on this. |
| 16:01.52 | [TK]D-Fender | It's based on the dialplan context |
| 16:02.12 | [TK]D-Fender | What phone are you doing this with? |
| 16:02.14 | jfindley | so it'll fire at the first match? |
| 16:02.40 | jfindley | it's a moto E4+ =) |
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| 16:19.57 | jfindley | [TK]D-Fender - That did it. Thank you! |
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| 18:52.49 | KingKeA | someone who has no experience at all in asterisk, but would like to get started - do you have a suggested resource? |
| 18:53.09 | rmudgett | ~book |
| 18:53.09 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:53.19 | KingKeA | ty |
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| 20:19.15 | Henson | Hi everyone, I'm having a heck of a time figuring out why my MWI is very delayed. On an old install, it worked just fine. Now with a new install it takes around 30 minutes for the MWI to show up after getting a message. With a different install the MWI never responded quickly. I've tried all kinds of sip.conf and voicemail.conf options, but nothing works... |
| 20:20.45 | Henson | using "voicemail show users for default" I can see that Asterisk notices right away when a message is present or not. The phones also have the MWI set correctly when I reset the SIP-analogue gateway which causes a reregistration of the phones. But otherwise the SIP NOTIFY messages indicating the MWI state are sent to the phones after a long delay. |
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| 20:21.43 | Henson | I'm using Asterisk 13.14.1 |
| 20:22.30 | [TK]D-Fender | Upgrade and test on current. |
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| 20:52.01 | Henson | upgrading to 13.22.0 didn't help, any other ideas? |
| 20:54.13 | igcewieling | make sure dns isn't being delayed or timing out. Asterisk does dns lookups at the oddest places. |
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| 22:51.28 | KingKeA | Is this a help desk? |
| 22:51.36 | KingKeA | If I have issues? |
| 22:51.52 | [TK]D-Fender | depends |
| 22:51.56 | file | this is a community chatroom, if you have a question someone may answer |
| 22:51.58 | [TK]D-Fender | ^ |
| 22:52.02 | [TK]D-Fender | Much more accurate |
| 22:52.15 | Ellenor | this is a potato room |
| 22:52.21 | Ellenor | i should leave lest i get snackish |
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