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| 00:21.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), 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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| 07:19.55 | jkroon | for ExecIf(${cond1}&&${cond2}?TrueStatement():FalseStatement()) style statements, in that case I'm getting that if cond1 is 1 and cond2 is 0 that it still goes for TrueStatement ... is this expected behaviour? |
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| 09:43.54 | vs-temp12 | Hi there, I've been migrating an ancient asterisk setup to 15. Having some issues with get_data through AGI, seems the timeout is ignored and logs show "The FD we were waiting for has something waiting. Waitfordigit returning numeric 1". Ring any beels here |
| 09:43.57 | vs-temp12 | ? |
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| 16:23.00 | polysics | hey everybody! question for y'all: is there a way to play audio in a channel that is being monitored via EAGI, and keep executing the calling script? |
| 16:25.52 | sibiria | the only way is via MoH |
| 16:25.57 | sibiria | but i suspect it comes with some caveats |
| 16:26.19 | sibiria | read up on StartMusicOnHold in the asterisk wiki |
| 16:29.05 | polysics | will MoH keep sending the far-end audio through EAGI? |
| 16:45.14 | sibiria | EAGI doesn't send audio through |
| 16:45.21 | sibiria | it taps the inbound audio for analysis purposes |
| 16:45.56 | sibiria | but to answer your question, yes, MoH is a non-blocking operation |
| 16:46.03 | sibiria | you start it up and your dialplan continues |
| 16:47.18 | polysics | ok, so the inbound audio should still be there even when MoH is playing |
| 16:47.31 | sibiria | yes |
| 16:48.05 | sibiria | so f.e. you can run StartMusicOnHold() in your dialplan and then head over to an AGI script or just continue doing stuff in the dialplan itself |
| 16:48.51 | sibiria | as opposed to Playback() and Background() which will block until audio ends (or DTMF comes in) |
| 16:50.58 | polysics | that sounds like a good solution |
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| 16:59.12 | polysics | another more complex question: anyone knows if hte current SFU only support BUNDLE mode or also separate streams? |
| 17:00.11 | file | they are still separate streams, they just use the same underlying transport - and the SFU itself doesn't care, that's at the PJSIP/RTP level |
| 17:00.48 | file | it can be disabled if the webrtc option isn't used |
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| 17:04.01 | tonybaqain | anyone available for help ? |
| 17:04.14 | file | if you ask a question someone may answer, but there is no guarantee |
| 17:09.29 | polysics | file: the SDP sent back from Asterisk indicates a bundle |
| 17:10.06 | polysics | it replies with bundle and all traffic is on one port, so they are bundled - at least on the surface |
| 17:11.06 | file | polysics: in order to work with Chrome bundle is on if the webrtc option is used |
| 17:29.09 | file | moo |
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| 17:39.46 | josefig | hell guys, i'm getting this error on asterisk since today, i was trying to use asterisk calls from LAN to incoming trunk and then fwd to trunk out, https://paste2.org/YnawBeNG |
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| 17:55.12 | viebig | hi all! |
| 17:55.30 | viebig | have you ever used AMD module ? |
| 18:04.33 | [TK]D-Fender | Yes, people have used it before |
| 18:08.36 | viebig | I'm having a strage behavior. totalAnalysisTime is not respected |
| 18:09.25 | viebig | the app_amd timer seens to be right... but the dialplan take more than totlaAnalysis time to process |
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| 19:18.10 | igcewieling | Has anyone had issues changing the chan_sip timerb= setting? It seems hardcoded to 32000, even if set to something different in sip.conf. I'm using Asterisk 13. https://pastebin.com/HBN9gTC9 the non-timerb settings are just so I could verify something is getting set. |
| 19:18.26 | igcewieling | it seems to work if set on the peer but not globally |
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| 19:36.48 | Samot | Never really messed with it. And at this point, if that's how it works that is how it works. |
| 19:37.37 | Samot | But why would you want it to be 6400? |
| 19:37.48 | Samot | That's awfully low. |
| 19:46.32 | Samot | Ah, you dropped the T1. OK, that's about 6 restransmissions after the initial invite... |
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| 19:54.01 | josefig | I'm getting this error, it was working before this script, this is to execute call files: https://paste2.org/wsFCkOXL I read on some logs of this channel that caller ID was wrong written on the call extension but I'm not sure what can be the problem |
| 19:57.40 | josefig | found the issue, thanks |
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| 21:00.54 | igcewieling | Samot: As I understand it Timer B is how to long to wait for a response to an INVITE. Is a 100 Trying or 183 Progress considered a response? |
| 21:01.34 | igcewieling | If only an ACK is considered a response, then how can we have calls ring for more than 32 seconds? |
| 21:02.00 | Samot | It's how long it waits for a provisional response. So a 100 Trying.. |
| 21:02.51 | igcewieling | If I don't get a response to an INVITE in 2 seconds then something is seriously wrong. |
| 21:05.56 | igcewieling | In the pcaps I see a trying sent .007 seconds after the invite is sent, here is one .004 seconds after the invite, and another one .0016 second after the invite. |
| 21:07.39 | igcewieling | With time elapsed between sending the invite and get a Trying is usually under 40ms, I don't understand why it should wait 32,000ms before retransmitting. |
| 21:11.11 | Samot | It doesnt |
| 21:11.40 | Samot | T1 is how long to wait before retransmitting |
| 21:11.51 | Samot | So if you have 200 |
| 21:12.12 | Samot | Its 0, 200, 400, 800, 1600, 3200, 6400 |
| 21:12.38 | Samot | Since you want B to be 6400 |
| 21:12.47 | Samot | At 6400 it timesout |
| 21:13.11 | Samot | Thats 6 retransmissions after the inital invite |
| 21:13.29 | Samot | Initial* |
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| 21:24.11 | igcewieling | I'd rather it gave up if it doesn't get a response in 500ms, so the call can failover to the next destination. |
| 21:25.55 | Samot | Then Timer B should be 500ms |
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