| 00:13.11 | Samot | Swap them |
| 00:13.43 | Samot | It has to hit 6000 first and then it is answered it will dial the other number |
| 00:13.50 | Samot | Same thing just in reverse |
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| 01:21.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); 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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| 15:30.20 | wasanzy | hello |
| 15:30.36 | wasanzy | anyone done AI with asterisk? |
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| 15:45.16 | [TK]D-Fender | * isn't a tool for this |
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| 15:45.48 | [TK]D-Fender | so the intelligence" part is all up o you |
| 15:48.22 | wasanzy | ok |
| 15:48.32 | wasanzy | but have you done something like that? |
| 15:49.01 | [TK]D-Fender | You are talking in completely vague terms. |
| 15:49.18 | [TK]D-Fender | There is no GOAL. There is no direction. No intent. No application attached to this. |
| 15:50.31 | wasanzy | my question is have you done any AI solution with Asterisk? say a call center like |
| 15:51.03 | [TK]D-Fender | You haven't described what qualifies as such a thing |
| 15:51.09 | [TK]D-Fender | and "in a call center" means nothing |
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| 16:02.05 | f3nr1l | Thx Samot |
| 16:02.14 | f3nr1l | Hello |
| 16:05.36 | f3nr1l | I tried to swap them without success. |
| 16:05.48 | f3nr1l | channel originate PJSIP/0123456789@SIPTRUNK extension 6000 <---- that works |
| 16:06.32 | f3nr1l | but when i try to swap the external with the internal it fails |
| 16:13.32 | [TK]D-Fender | And you aren't showing what you're trying and the actual attempt |
| 16:17.41 | f3nr1l | hi [TK]D-Fender |
| 16:18.49 | f3nr1l | I'm trying to call an external number and to pass it to an extension (a confbridge actually) while it's still ringing |
| 16:19.58 | [TK]D-Fender | You can't do it specifically at that time. |
| 16:20.17 | f3nr1l | ok |
| 16:20.47 | [TK]D-Fender | You CAN use a local channel on the call-out leg, answer that before the actual Dial() takes place which will trigger the other side to call your inside user. |
| 16:21.05 | f3nr1l | yesterday, Samot told me "[20:13] <Samot> Swap them |
| 16:21.05 | f3nr1l | [20:13] <Samot> It has to hit 6000 first and then it is answered it will dial the other number" |
| 16:21.27 | [TK]D-Fender | but if that call is rejected (never actually rings) then your agent is going to get a dead-air call that might stop almost instantly, etc |
| 16:21.37 | [TK]D-Fender | You said to reverse the direction |
| 16:21.52 | [TK]D-Fender | you did NOT say anything about "still ringing",. |
| 16:21.59 | f3nr1l | Sorry, english is not my mothertongue |
| 16:22.14 | [TK]D-Fender | you can do it BEFORE the actual attempt is made. but you could get hit with a busy which should kill the other side, etc. |
| 16:22.47 | [TK]D-Fender | I suppose on the destination side you COULD put a 2-3 sec delay before dialing them just to ensure that it didn't abort instantly |
| 16:22.55 | [TK]D-Fender | results may vary |
| 16:23.18 | f3nr1l | could you please give me a example of syntax ? |
| 16:24.28 | [TK]D-Fender | Local/12345678@dialplancontext/n |
| 16:26.09 | f3nr1l | dialplancontext is the context the trunk is in ? |
| 16:27.08 | [TK]D-Fender | "trunk" isn't a thing. |
| 16:27.12 | [TK]D-Fender | DIALPLAN CONTEXT. |
| 16:27.17 | [TK]D-Fender | it is exactly what it says. |
| 16:27.58 | [TK]D-Fender | you are dilaing into dialplan itself and when that side gets answered it will start up the other leg which is also dialplan and they will get bridged when that 2ns leg answers |
| 16:28.10 | [TK]D-Fender | Whatever they are doing at that time will be what the other experinces. |
| 16:30.04 | Samot | Please dont quote me out of context |
| 16:31.01 | Samot | The original question in which I replied to had nothing about conf bridge in it. |
| 16:31.26 | Samot | It was expressly about Originate. |
| 16:31.31 | f3nr1l | I'm sorry Samot, didn't intend to say that you gave me a wrong answer |
| 16:31.53 | Samot | In the context of having a conf bridge involved.. |
| 16:31.57 | Samot | It is the wrong answer. |
| 16:32.23 | f3nr1l | a conf bridge, or a meetme... |
| 16:32.50 | f3nr1l | I'm testing with an extension |
| 16:33.44 | f3nr1l | [TK]D-Fender, the config you just gave me works pretty good ! |
| 16:34.11 | f3nr1l | channel originate Local/01234567@from-internal/n extension 6000 |
| 16:34.27 | f3nr1l | 1: 6000 rings |
| 16:34.46 | f3nr1l | 2: I pick it up |
| 16:35.07 | f3nr1l | THEN (after a few secs) 01234567 rings |
| 16:35.40 | Samot | Which is what I said to do. |
| 16:36.02 | Samot | Because the original question was that 01234567 rings first then rings 6000 after they pick up |
| 16:36.10 | Samot | So that's why I said "reverse it" |
| 16:37.00 | f3nr1l | Samot, I have no doubt about that. I'm just saying that I was too newbie|dumb tu get it when you told me |
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| 16:45.45 | f3nr1l | now I'm gonna replace extension 6000 with 6099@ext-meetme or app confbridge |
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| 16:52.50 | f3nr1l | [TK]D-Fender, Samot thanks a million for your help ! |
| 16:52.57 | f3nr1l | It works like a charm |
| 16:53.16 | f3nr1l | channel originate Local/0123456789@from-internal/n extension 86000@ext-meetme |
| 16:53.58 | f3nr1l | from the confbrigde, i can hear the call ringing. |
| 16:54.42 | [TK]D-Fender | You shouldn't if that's typicaly FreePBX dialplan |
| 16:54.51 | [TK]D-Fender | Which is also something you should be specific about. |
| 16:54.57 | [TK]D-Fender | because that's THEIR dialing rules |
| 16:55.29 | [TK]D-Fender | And doesn't normally answer the call-out leg so you could hear ringing. |
| 16:55.55 | f3nr1l | I'm sorry, i didn't understand |
| 16:57.12 | f3nr1l | did you mean that I shouldn't change asterisk's default behaviuor ? |
| 16:57.44 | [TK]D-Fender | <[TK]D-Fender> because that's THEIR dialing rules <---- |
| 16:57.48 | [TK]D-Fender | You aren't using Asteriwsk. |
| 16:58.01 | [TK]D-Fender | FREEPBX is using asterisk, and you seem to be using the dialplan THEY generate |
| 16:58.09 | f3nr1l | correct |
| 16:58.51 | f3nr1l | I'm testing on a freepbx box |
| 16:58.55 | [TK]D-Fender | You can buy a box full of LEGO and buil dwhatever you want. You bought a "medieval castle" LEGO set and are following the instructions. Don't be surprised when you don't end up with a custom castle in the end |
| 16:59.26 | f3nr1l | but yes, eventually that shall run on a compiled by hand asterisk-on-ubuntu box |
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| 19:53.29 | patata | Hello guys, I am facing some anoying issue with Asteris 15.1.5. I can "file convert" from an to ulaw, g729, alaw, g723 ... but GSM fails at all !! I checked and I have loaded codec_gsm.so ... |
| 19:53.50 | patata | any sugestion ? |
| 19:57.46 | [TK]D-Fender | Show us the attempt, the files, etc |
| 19:57.53 | [TK]D-Fender | proof that the modules are loaded..... |
| 19:59.31 | patata | Command 'file convert /var/lib/asterisk/sounds/es/noaccounten.wav /var/lib/asterisk/sounds/es2/noaccounten.gsm' failed. |
| 20:00.12 | patata | sip*CLI> module show like gsm |
| 20:00.12 | patata | Module Description Use Count Status Support Level |
| 20:00.12 | patata | format_gsm.so Raw GSM data 0 Running core |
| 20:00.55 | [TK]D-Fender | that's FORMAT, not CODEC |
| 20:01.06 | [TK]D-Fender | completely separate |
| 20:01.31 | patata | <PROTECTED> |
| 20:01.31 | patata | <PROTECTED> |
| 20:01.36 | [TK]D-Fender | Format is used to READ a file, Codec is to be able to translate to/from |
| 20:01.46 | [TK]D-Fender | the codec module is not loaded |
| 20:01.51 | [TK]D-Fender | we see only "format" |
| 20:01.53 | [TK]D-Fender | no good |
| 20:01.58 | [TK]D-Fender | Other output is meaningless |
| 20:02.21 | [TK]D-Fender | * KNOWS that 5 should be GSM. Doesn't mean the module to SUPPORT it is loaded |
| 20:02.37 | [TK]D-Fender | try to load it manually and see what you get |
| 20:02.45 | patata | ok, just a sec |
| 20:03.00 | [TK]D-Fender | if that works go look at your modules.conf to see why it might not have loaded like you expect it to have |
| 20:03.27 | patata | module load codec_gsm |
| 20:03.27 | patata | Loaded codec_gsm |
| 20:03.27 | patata | <PROTECTED> |
| 20:03.27 | patata | <PROTECTED> |
| 20:03.27 | patata | <PROTECTED> |
| 20:03.27 | patata | sip*CLI> file convert /var/lib/asterisk/sounds/es/noaccounten.wav /var/lib/asterisk/sounds/es2/noaccounten.gsm |
| 20:03.30 | patata | Converted /var/lib/asterisk/sounds/es/noaccounten.wav to /var/lib/asterisk/sounds/es2/noaccounten.gsm in 28ms |
| 20:03.43 | patata | yes, you was right, was not loaded for some reason, I will recheck all config. |
| 20:03.52 | patata | now the issue is fixed once I loaded it manually |
| 20:04.37 | patata | thank you very much |
| 20:04.51 | [TK]D-Fender | So something has happened with the way modules are being loaded. double check your configs, etc |
| 20:05.12 | [TK]D-Fender | And you're welcome. Glad it is something that would appear to be a minor thing to fix |
| 20:07.03 | patata | [TK]D-Fender, whas something stupid from me ... I had noload =>codec_gsm.so in some file, now I removed and restarted this Asterisk, that was |
| 20:07.23 | [TK]D-Fender | Yup, silly kind of thing to have in your way... |
| 20:07.42 | patata | yeah, sorry |
| 20:07.43 | [TK]D-Fender | But at least it stood out. |
| 20:08.16 | [TK]D-Fender | Clear mistakes are at least quickly solvable ones! |
| 20:41.42 | vo1pbx | maybe /var/log/asterisk/full |
| 20:41.50 | vo1pbx | just guessing |
| 20:42.04 | vo1pbx | there you can get a bit of `tail` |
| 20:42.51 | vo1pbx | given enough eyes, all bugs are shallow, and [TK]D-Fender has a lot of eyeballs. |
| 20:43.10 | [TK]D-Fender | wrong channel.... |
| 20:44.04 | vo1pbx | all i'm inferring is that you can spot a misconfiguration in plaintext from forty yards away. |
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| 22:11.50 | cervajs2 | i have problem with random hanging calls . client - chrome,jssip. asterisk 13.18 + chan_pjsip. i see these messages in log: res_http_websocket.c: Web socket closed abruptly and tcptls.c: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe . any ideas what can be the cause? |
| 22:12.58 | cervajs2 | s/hanging/hangup/ |