IRC log for #asterisk on 20201019

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04:31.55*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.36.0 (2020/09/03) 16.13.0 (2020/09/03) Standard: 17.7.0 (2020/09/03); 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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09:25.05AL13N_lappydrmessano: markong: i'm sorry to revisit the topic, but tbh, i didn't spot the sarcasm until it was talked about... emotions are hard to pick up virtually and communication errors are easy to happen. But, it may also be due to me not having english as the first language, which i should mention there are a lot of people like that.
09:25.59AL13N_lappyin my case, i had tried with the 18.0.0-rc2 and found a weird packaging issue:
09:29.22AL13N_lappyin configure.ac there's AC_HEADER_STDC (which is obsoleted by gnu doc), but it seems to fail, or rather, the resulting $ac_cv_header_stdc seems not to have "yes" in it, even with glibc-devel and stdc++-devel installed...
09:29.41AL13N_lappydoes anyone have ideas on this?
09:35.51AL13N_lappyah, I read some more on it. it appears gnu is advocating removing this in favor of checking the header files you need; which happens just below this invocation... so, it seems this check is completely unneeded... is there a patch i can upstream to you guys?
09:40.24markongAL13N_lappy, exactly my point, that was foggy and unnecessary sarcasm, very prone to generate confusion.
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09:51.37kjetilhoin combination with: 17:07 <drmessano> Wouldn't want you to not have bugs to chase down
09:51.48kjetilhoI don't see how this can not be understood as sarcasm
09:52.15kjetilho(this was the very next line, without anything coming in between the two statements.)
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10:15.10markongOP confirmed that he didn't spot the supposed sarcasm and thus did get away confounded as I did. Whatever it was, that's been a failure as a matter of facts. For me case is CLOSED.
10:20.56kjetilhoexcellent.
10:33.07Ravenhearthi
11:15.27sibiriahowdy
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11:23.13AL13N_lappykjetilho: tbh: due to double negation, i kinda had to read this 3 times to get the meaning... sorry
11:24.45AL13N_lappyso, what about this patch for removal of AC_HEADER_STDC ? do you guys know a place where this can be talked about/submitted?
11:24.51kjetilhono need to be sorry! :)
11:25.54filethe wiki has a guide for contributing a patch, https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
11:26.02kjetilhotheir repo mirror on Github says: Use gerrit.asterisk.org
11:26.49AL13N_lappythis menuselect thing, i get it for users, but for distro's ; it's kind of a PITA... as a distro, we generally want to build all modules we have requirements for, or we disable some in configure if there is licensing issues. Is there a way to get this menuselect things to just build as much as possible? some kind of non-interactive way?
11:26.54AL13N_lappyfile: thanks
11:27.23filemenuselect will automatically build everything that is to be built by default, with dependencies met
11:27.39fileit can also be controlled from the CLI, https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options
11:29.33AL13N_lappyfile: thanks
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13:26.11drmessanokjetilho: Thank you for the validation.  Let's make sarcasm great again.
13:27.01kjetilhodouble negatives are best avoided, though
13:27.30drmessanoUnless fully intended
13:27.50kjetilhono, I mean for readability
13:27.59drmessanoSeems like a big deal over nothing since additional explanation was provided
13:28.07kjetilhoyes, agreed
13:28.53drmessanoI think markong just likes to be offended.  He may be French, idk.
13:29.22drmessanoI'll check back in with him later
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13:30.51kjetilholet it lie, please
13:41.15markongdrmessano, instead of keeping highlighting my nick, why you just don't go on to entertain the channel with your useless sarcasm? that would be appreciated because I could easily ignore it.
13:42.37drmessanomarkong: you've highlighted me more times now than I can count when I have been away.  You keep going and going.  You're a useless troll.
13:43.17drmessanoI go to a sleep, wake up, guess who's been talking shit?  Get over yourself
13:43.33drmessanoOne damn comment you didn't like and you're triggered for life.
13:43.46drmessanoGrow up
13:47.26markongdrmessano, WTF you're talking about? last time I've highlighted your nick was yesterday! Speaking of trolls...would you please stop? It's enough.
13:48.18drmessanoYep, and I walk away again and again you've highlighted me.  Putting you on ignore you worthless fuck
13:49.10drmessanoBye markong markong markong
13:49.42markongChildish idiot.
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13:55.50Samotmarkong: Are you packaging a distro?
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14:12.00igcewielingI think I shall leave for a while.  too much drmessano.
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14:17.54SamotYeah, that's the problem.
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14:53.11AL13N_lappyhmm well... i feel like i've triggered something i shouldn't have...
14:54.20AL13N_lappyanyway: when building asterisk, i come accross something like this:
14:54.20AL13N_lappy"The existing menuselect.makeopts file did not specify
14:54.20AL13N_lappy<PROTECTED>
14:54.20AL13N_lappy<PROTECTED>
14:54.20AL13N_lappy<PROTECTED>
14:55.05AL13N_lappythe menuselect.makeopts does NOT include app_voicemail_odbc ... why is it complaining about something that's NOT in there?
14:55.54filemenuselect.makeopts specifies what should not be built
14:56.25AL13N_lappyhow should i mark it NOT to be built? do i need to prefix it with '-' or something?
14:57.29AL13N_lappyfile: wait, do you mean that all the stuff in there is what NOT to build?
14:57.36fileyes.
14:58.04AL13N_lappyhuh, then i guess i was doing it the other way around, thanks; it kinda makes sense now...
14:58.19AL13N_lappyfile: what about that bottom part on FAILED stuff? what does that mean?
14:58.35filethat question lacks context and I can not answer
15:00.05AL13N_lappyfile: eg: in that file, there's also MENUSELECT_DEPSFAILED=MENUSELECT_CHANNELS=chan_pjsip
15:00.22AL13N_lappydoes that mean that this chanpjsip doesn't work due to failed deps or something?
15:01.13SamotAL13N_lappy: You triggered nothing.
15:01.15fileyes.
15:01.24AL13N_lappyok, thanks
15:07.46AL13N_lappyfile: so, now i have a similar warning about "codec_opus", but i want to build opus support. is there a way i can found out exactly what is missing to get codec_opus? in some way?
15:08.05fileit will list in the make menuselect interface itself
15:11.30AL13N_lappyfile: ic, thanks
15:12.07AL13N_lappyi don't have an easy way to do the menuselect itself, since i'm testbuilding on one of our distro-nodes, which are not interactive, so...
15:12.19AL13N_lappybut i found some stuff in makedeps that's wrong
15:12.24AL13N_lappyi'm testing that now
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15:45.41AL13N_lappyfile: ok, i couldn't do interactive, but i figured out dependency tree
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18:36.36Posterdatihi
18:36.39Posterdatiplease help
18:37.45PosterdatiI'm experincing this error in asterisk [Oct 19 20:36:53] NOTICE[482043]: chan_sip.c:16046 sip_reg_timeout:    -- Registration for '+xxx@yyy.zzz.it' timed out, trying again (Attempt #...)
18:38.12PosterdatiI checked with nslookup yyy.zzz.it and the server is unreachable
18:38.40Posterdaticould be a provider side problem? Thanks for help
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19:22.07Dovid@file: Why the -1 for https://gerrit.asterisk.org/c/asterisk/+/15066? Becuase it's not ready?
19:22.13fileyes
19:22.26file"Marking -1 per comment of not to review"
19:23.02Dovidfile: Ok. The code as is only supports one error code (e.g. a 404). I wanted it to support multiple like 404, 503 etc. having it just support a 404 is good only for me. once were creating a patch it should work for others as well
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19:37.30AL13N_lappyfile: what's with all these binary blobs that are being fetched, like codec_opus? i thought opus was opensource? and no patent? why can't we build just build it from source and/or link it to the upstream?
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19:38.12filecodec_opus as provided by Sangoma is distributed as a binary module due to potential patent concerns
19:39.38AL13N_lappypfff, almost anything is "potential patenting"... for our distribution, it's actually not counted as patent, so, we are perfectly able to do so, AND, there's countries where patents don't even count too...
19:39.51fileyou are free to use other implementations
19:39.58AL13N_lappyfile: also, if you build against the opus library, then you don't actually contain it... so?
19:40.13AL13N_lappyfile: ic
19:41.19AL13N_lappyfile: so their codec_opus is not linked to libopus? so, their code for codec_opus is not opensource then? or is the source for that available?
19:41.39filethe codec_opus module as provided by Sangoma is closed source, I can not comment on any other implementation
19:41.49AL13N_lappyok, thanks
19:42.11AL13N_lappyfile: do you know of another implementation or can point me to one?
19:42.29fileI can't comment on such things
19:42.51AL13N_lappyoh, you work for Sangoma?
19:42.54AL13N_lappyok, fair enough
19:43.02fileI do.
19:43.08AL13N_lappyfile: thanks for all the packaging help though
19:44.38AL13N_lappyfile: just for clarity, the codec_opus module is for en/de-coding stuff to and from opus, if both sides have opus, then passthrough is used which doesn't use this module, right?
19:44.46fileyes.
19:44.47AL13N_lappyfile: or at least, that was my understanding
19:44.52AL13N_lappyok, thanks
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20:34.33DannyAdoes anyone happen to have the firmware to convert a mitel phone to sip?
20:40.57drmessanoDannyA: Which device is it?
20:41.19DannyA5330
20:43.40drmessanoYou can reboot the phone with the volume down key pressed.
20:43.40drmessanoOnce you are in the phone configuration go to “Phone mode”
20:43.40drmessanothen “Protocol” and then you can change the Phone Mode from Minet to SIP.
20:46.16drmessanoDannyA: ^
20:46.33DannyAlet me try hang on (thank u!)
20:46.41seanbrightthat'll be $499
20:53.13drmessano$1 for my time and $498 for knowing what to googlwe
20:53.15drmessano$1 for my time and $498 for knowing what to google
21:28.24Kobazis there a way to turn off connnected line updates in chan_sip
21:28.56Kobazchan_sip is sending UPDATE and the remote peer does not take kindly and is sending back 500 internal server error
21:29.07Kobaznot sure if you can... on chan_sip
21:38.07Kobazof course pjsip can do it
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22:53.02markongSamot, no.
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23:27.12jeevthere is a site that does analytics, answered calls, dialed calls and all that wonderful stuff, which extension did it, you can monitor multiple extensions per employee and all seems to be ok other than just one or two calls not showing and then we realized that we can't actually monitor transferred calls, if the receptionist answers 1,000 calls and transfers 980 calls, it won't show that
23:27.12jeevthe employees that were transferred to were on the phone. the company tells me that asterisk isn't open enough for that (using ami), how much of a load of crap is this ?
23:27.17jeevit's a complete load of crap
23:28.34Kobazwell this is fun
23:29.14KobazAMI -> ModuleCheck -> Module: res_pjsip    Returns: Response:Success   Version:
23:30.39Kobazjeev: transfer tracking is something I spent a year on getting 100% right on asterisk 1.8
23:30.49KobazIt's never been an easy problem to solve
23:31.03KobazIt's for sure easier on new-er asterisk's
23:31.18KobazBut by no means is straightforward at all
23:31.45KobazBut, it's 150% doable using Asterisk, you just need to be a developer to do it
23:32.14jeevso you would think a company that sells a 'powerful product' can surely ask it's developers to work with ami's messages and figure out the path forward, right ?
23:33.17Kobazcorrect
23:33.26Kobazmany developers aren't software engineers
23:33.33Kobazmost developers are not, in fact
23:34.06Kobazweb pages and querying the ping status of phones is entirely different than orchistrating infrastructure work to properly track state and status and report on correctly
23:35.36jeevyea, i've got no development skills, it's sad, i know it's one of the hardest things out there but a company that claims to have "developers" has no exuse not to have this feature.
23:35.42jeevwhen we're talking about five figures per year
23:35.51KobazI'll do it for 5 figures a year
23:36.00KobazOur software tracks that no problem at all
23:36.38jeeveh it's not just that
23:37.55Kobazand the issue is...
23:38.22jeevit's an entire suite that communicates with salesforce, stores recordings on (ONLY amazon s3,. because wasabi isn't worth using apparently)
23:39.15Kobazah
23:39.25Kobazwe've done some integrations with salesforce
23:40.13jeevthings seem ok, except transfers which is fatal
23:41.35Kobazbut to answer your question, Asterisk, especially modern Asterisk, is more than capable of handling that particular need
23:42.19jeevyea i wish i knew exactly what to say, how to say it. it's a joke
23:56.04AL13N_lappyspeaking of transfers, how do i get this right? if you have a guided transfer from receptionist R, from outside O to employee E: E should see R as source first, and then after transfer he should see O, so he can redial directly to outside?
23:58.27AL13N_lappyis there a SIP message that changes source or callerid or something while E never hung up?
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