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02:00.12 | DannyA | if i enable ari in the conf file, then do module reload res_ari.so, is that enough to enable and allow connections? or do i need to restart asterisk? |
02:12.32 | DannyA | i keep getting connection refused, even when testing locally using wscat |
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02:33.42 | kevinnn | hi all, is anyone here? I am in a bit of a crisis over here and I need help! |
02:34.15 | kevinnn | So we have an asterisk box that admittedly has not been updated in a long time. Thankfully I took a backup of this system |
02:34.37 | kevinnn | upon apt-get update, apt-get upgrade something went wrong |
02:35.29 | kevinnn | all of the tones I play on my speakers from asterisk have stopped playing correctly. asterisk log shows nothing wrong... but the client systems are all showing a log like this: |
02:35.55 | kevinnn | https://pastebin.com/KciT1JxZ |
02:36.12 | kevinnn | so I took the backup and brought it back online and the speakers work perfectly |
02:36.38 | kevinnn | the version of asterisk did not get upgraded it is still on the 13 branch (yes I know that is still really old) |
02:36.43 | kevinnn | can anyone please help! |
03:08.47 | kevinnn | is anyone here? |
03:45.18 | DannyA | doesn't that tell u that the issue is with whatever got upgraded on the system when u did apt update? |
03:45.47 | DannyA | doesn't sound like an asterisk issue. sounds like some kind of system program, something related to audio or io or something like that |
03:46.09 | DannyA | you need to figure out what got upgraded when u do apt upgrade and take it upgrade by upgrade to find the source of the problem |
03:46.23 | DannyA | but that's just my guess |
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07:21.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); 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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13:26.30 | DannyA | hey all. how can i reload pjsip_wizard? I just added some endpoints, and after i do pjsip reload, and then pjsip show endpoints, i don't see them there |
13:34.57 | file | "module reload res_pjsip_config_wizard.so" potentially |
13:44.23 | DannyA | got it |
13:44.23 | DannyA | thanks |
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16:51.09 | velix | pjsip is working now on my system. Once you figured it out, it works. |
16:51.35 | velix | This AoR, auth and identify stuff is pretty complicated. |
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16:53.21 | seanbright | not really |
16:53.48 | seanbright | and that is why res_pjsip_config_wizard exists |
16:54.12 | velix | seanbright: Yeah, but you still need to setup stuff in pjsip for this. |
16:54.23 | velix | Transportation etc. |
16:54.31 | igcewieling | pjsip is easier once you get the hang of it. None of that friend/peer/user sillyness. |
16:55.19 | velix | igcewieling: That's true. |
16:55.34 | velix | But really annoying is: once you mix up a config parameter, the endpoint or registration is gone. |
16:55.43 | velix | it doesn't say: "hey, this config is wrong in this type". |
16:55.50 | velix | Even not with debug and verbose set to 99 |
16:55.56 | velix | or logger and history = on |
16:57.28 | velix | Why can't I make Asterisk use an STUN server without ICE? My softphone uses STUN without ICE... the ICE server on the other side has some auth issues. |
16:58.01 | file | because noone has added support to use a STUN server for discovery of external address for the transport |
16:58.50 | igcewieling | I find that the easiest solution is to get a public IP for your asterisk server, then you can avoid all this sillyness. |
16:59.44 | velix | file: Ah, I thought it was my fault by not finding the setting. |
17:00.24 | velix | igcewieling: Did so. I'm just doing some debugging. |
17:02.13 | velix | We had 1000 calls processed toda on a single core VPS. |
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17:02.20 | velix | it never was over 5% cpu. |
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17:02.29 | velix | Asterisk is damn light |
17:07.00 | Kobaz | velix: 1000 calls doesn't say much, if it was one at a time. with no concurrency |
17:07.18 | velix | Kobaz: 8 in parallel max |
17:07.25 | Kobaz | try handling 1000 *simultaneous* calls, and that will show you if your setup is solid |
17:07.41 | Kobaz | You can run 8 calls on a raspberry pi |
17:07.42 | Kobaz | heh |
17:07.46 | velix | Hehe, I'll never reach this. |
17:07.57 | velix | Kobaz: I've got a RPi 1 for my heating at home. |
17:08.09 | sibiria | my 1 ghz quad-core Arm test platform (a bit weaker than an RPi 4) can manage 150 simultaneous calls + random dial plan functionality without hitting 100% cpu |
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17:08.30 | [TK]D-Fender | velix, I didn't think the Pi put out enough heat for that.... |
17:08.38 | Kobaz | haha |
17:08.40 | velix | [TK]D-Fender: ^^ |
17:08.41 | [TK]D-Fender | clearly needs to reconsider his energy usage... |
17:08.48 | Kobaz | [TK]D-Fender: ba da ding! |
17:09.10 | velix | I've coded a nice conference management. PHP directly writes to Asterisk's DB. |
17:09.12 | Kobaz | [TK]D-Fender: i heated my garage for one winter with my friend's antminer s9's |
17:09.18 | velix | DID support and stuff. |
17:09.36 | Kobaz | [TK]D-Fender: he paid for the electricity, and i got free heat |
17:09.44 | velix | Google's Voices have won my sound quality test. I've created an account. Since we are students, we got it for free. |
17:11.35 | velix | Really good quality. Nobody was able to detect, it computergenerated. |
17:11.44 | velix | it > was < ... |
17:18.12 | Samot | Google Voice is already free. |
17:18.25 | Samot | Regardless of your education status. |
17:18.28 | Kobaz | he's talking about text to speech |
17:18.39 | Samot | Ah. |
17:19.03 | Kobaz | I have my velix translator hat on |
17:19.27 | igcewieling | Kobaz: does it break every time you mess with it? |
17:19.30 | velix | Oh sorry. Yes. The Wave... what was its name? |
17:19.37 | velix | Neurowave? |
17:19.43 | igcewieling | test-to-speach or TTS |
17:19.52 | igcewieling | text-to-speach or TTS |
17:19.54 | velix | igcewieling: yeah, TTS ... but it has a special name |
17:20.12 | Kobaz | igcewieling: does what break? |
17:20.29 | velix | Wavenet! |
17:20.39 | igcewieling | velix: don't worry. Google will rename or discontinue the service in few months like most of their other projects. |
17:20.45 | velix | hehe |
17:20.55 | velix | Yeah, Google Hangouts SIP support was cancled. That's sad. |
17:21.07 | igcewieling | Kobaz: your velix translator hat. If it doesn't break every time you touch it, it can't be a velix hat. |
17:21.15 | Kobaz | igcewieling: oh, haha |
17:21.42 | Kobaz | velix: so you got the firewall/network/whatever issues fixed? |
17:21.49 | Kobaz | the remote site is actually responding properly now? |
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17:22.34 | velix | Kobaz: Nope. I gave up, it doesn't seem to be my fault. We've purchased 8 flatrates for "land line" numbers and 12 hours for cell network. |
17:22.44 | Kobaz | right, and that works perfectly? |
17:22.48 | velix | yep. |
17:22.57 | Kobaz | yeah, so you see what i mean finally |
17:23.04 | Kobaz | what i've been saying since two weeks ago |
17:23.08 | Kobaz | something is wrong with that other system you're talking to |
17:23.32 | velix | Kobaz: Funny thing: I've got many working calls without forwarding :DD |
17:23.46 | velix | He didn't want to tell me what the forwarding is for. |
17:23.53 | Kobaz | their system is broken |
17:23.58 | velix | Also think so. |
17:24.06 | Kobaz | and it doesn't seem like they understand why |
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17:24.43 | velix | It's a innovaphone hardware gear with a GUI. Not CLI & config, as pure Asterisk. |
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18:17.41 | Kobaz | soooo, with SIP TLS, would there still be limitations on the size of a SIP INVITE with chan_sip ? |
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18:34.42 | Samot | There's limitations regardless of driver. |
18:35.07 | Kobaz | default linphone sends a pretty massive SDP |
18:35.15 | Kobaz | every codec is enabled |
18:35.28 | Samot | So disable the ones you don't need. |
18:35.38 | Kobaz | right, yeah, updating our provisioning |
18:35.51 | Kobaz | I wasn't sure if switching to SIP TLS would address the size issue |
18:38.23 | Samot | UDP is 65,535 bytes |
18:38.35 | Kobaz | the RFC does not specify a hard upper limit on the size of the SDP |
18:38.50 | Kobaz | well, this is way less than 64k |
18:38.54 | Samot | If the packet is within 200 of the path MTU it needs to be TCP or follow the RFC for fragmenting |
18:39.29 | Kobaz | [2020-04-02 14:29:53.949] failed to extend from 256 to 371 |
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19:32.41 | Testeree | hi |
19:33.31 | Testeree | has anyone ever experimented with binaural rendering feature. does it work? |
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20:30.06 | Nivex | I just stumbled across https://wiki.asterisk.org/wiki/display/AST/AudioSocket . Looks neat, but are there any plans to support other sample rates? Most of my kit can do wideband audio. |
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21:33.41 | MLC | Any having trouble with the hold time announced in a queue being way off? I can tell just by looking that the hold time is > 5 minutes, but when a caller enters the queue it is telling them "30 seconds". |
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