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| 00:21.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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 |
| 00:47.41 | Nivex | igcewieling: when I first moved to NC from OH, I asked what day the tornado siren tests were. Noone had any clue what I was on about. |
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| 15:24.57 | andresmujica | dear team, I wonder if anyone could help me with two questions. 1.- is it possible to send the outbound leg from a .call file to a different context instead of dialing directly from the .call file itself. and 2.- I had an issue with inband audio through a sip trunk, when my asterisk sends the call to a different operator it cannot detect when a voicemail answers the call and left a cut message. |
| 15:30.08 | [TK]D-Fender | There is no "outbound leg" |
| 15:31.01 | [TK]D-Fender | a call file dials the Channel: however you defined it and upon answer eitehr executes 1 single dialplan application, or dumps that channel into the dialplan in the context,exten,prio you defined |
| 15:31.43 | [TK]D-Fender | You choose the Channel, you choose the destination. This is 100% explicit and your job to fill in. |
| 15:38.32 | [TK]D-Fender | 2. pretty much all "call" audio is "inband". As for detecting if voicemail is what answers that is your job to do in your call processing. This involves initial detection plus detecting when it is finished. |
| 15:39.12 | sibiria | andresmujica: asterisk's built-in AMD is unfortunately not so efficient |
| 15:39.35 | sibiria | and "the best" commercial solution out there is rubbish |
| 15:39.40 | sibiria | (as is the support they offer) |
| 15:40.22 | igcewieling | Asteria, if they are still around has an AMD implementation for Asterisk. No cheap, not cheap at all. |
| 15:40.39 | sibiria | we had to resort to writing our own AMD, which incidentally has over 97% accuracy on a set of ten thousand calls which "the best" commercial option hits about 75% on |
| 15:43.39 | sibiria | "the best" commercial option is Lumenvox, btw. |
| 15:43.45 | sibiria | i would not recommend them or their products to my worst enemy |
| 15:44.03 | sibiria | shit product, shit developers, shit tech support |
| 15:45.30 | sibiria | they charged us $1200 when we reported bad performance and zero detection of certain AMD beeps |
| 15:45.49 | sibiria | provided them with packet dumps, audio recordings etc., and they couldn't solve or improve their product at all to detect the beeps |
| 15:46.18 | sibiria | then held our account "hostage" (by rejecting all operative requests) until the $1200 payment for no support at all was made |
| 15:47.02 | sibiria | never before seen or heard of anything like it |
| 15:50.39 | Samot | AMD works pretty well for me. |
| 15:52.07 | sibiria | it works well for what it does in a technical sense - count words |
| 15:52.16 | sibiria | counting words just isn't the most efficient solution |
| 16:00.44 | Samot | No, it does more than that. |
| 16:01.33 | sibiria | no, it counts words, including this or that amount of silence or greeting |
| 16:01.44 | sibiria | it does no frequency or waveform analysis of any kind |
| 16:01.55 | sibiria | no beep detection involved |
| 16:02.24 | sibiria | neither by sample comparison nor fourier transform |
| 16:03.28 | Samot | OK. |
| 16:09.22 | craigify | sibiria, you all wrote an in house voicemail detector? impressive. |
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| 16:09.52 | andresmujica | thanks for your input guys⦠I guess I'll start with amd detection in the meantime |
| 16:10.49 | andresmujica | and take a look into the dial leg from the .call file |
| 16:11.47 | sibiria | craigify: it's not simple but at the same time it's not nearly as complicated as it may sound. it does real-time frequency analysis of the audio stream to see if there's something specific that peaks the way a static tone does |
| 16:12.35 | sibiria | in contrast, counting words is a more complicated analysis |
| 16:21.59 | craigify | cool |
| 16:29.06 | craigify | so you all are really interested in the tone |
| 16:32.12 | sibiria | we used to employ asterisk's own AMD (which only counts words), and no matter what various configuration of that we used, the detection rate wasn't good enough |
| 16:32.19 | sibiria | we simply placed too many "prank calls" if you understand |
| 16:32.44 | sibiria | by using both word counting and beep detection, we're at over 97% accuracy |
| 16:32.58 | sibiria | (word counting picking up the slack where the beep doesn't come through because of poor audio) |
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| 20:03.15 | LunaLovegood | Got a problem again. When I receive a fax on, for example +15553565648, it sometimes has time to go through the dialplan up to Dial(PJSIP/my_ata) before detecting that it's a fax and jumping to the 'fax' extension. Then, in the fax extension i decide that I need to send that fax to PJSIP/my_ata after all, I call Dial(PJSIP/my_ata) again, but now it's busy because it was already ringing. What can I d |
| 20:03.21 | LunaLovegood | o? |
| 20:03.24 | LunaLovegood | *do |
| 20:05.43 | LunaLovegood | When I decide to ring a line on which there's only a fax machine, I'd like to be able to disable fax detection before calling Dial(). Is there a variable I can set or something, that would disable detection afterwards? |
| 20:06.10 | LunaLovegood | From the dialplan I mean. |
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| 20:11.47 | LunaLovegood | When the dialplan is waiting on a Dial() call and decides halfway through to jump to the 'fax' extension, is the call terminated or is there a way, from the 'fax' extension, to do nothing and wait for the first Dial() call to connect? |
| 20:12.10 | LunaLovegood | That'd be a solution to my problem too I guess. |
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| 21:01.57 | igcewieling | LunaLovegood: 1) it would only jump to a fax extension if faxdetect is enabled. Disable faxdetect and it should work better. |
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| 21:34.00 | meyou_ | found some guide that seems to basically have me running killall -HUP fop_server to make voicemail sync with fop2 password |
| 21:35.01 | meyou_ | doesn't seem to work though, is fop_server supposted to resync passwords upon sighup? |
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| 22:08.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.20.0 (2018/03/15), Standard: 15.3.0 (2018/03/15); 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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