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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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09:25.05 | AL13N_lappy | drmessano: 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. |
09:25.59 | AL13N_lappy | in my case, i had tried with the 18.0.0-rc2 and found a weird packaging issue: |
09:29.22 | AL13N_lappy | in 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... |
09:29.41 | AL13N_lappy | does anyone have ideas on this? |
09:35.51 | AL13N_lappy | ah, 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? |
09:40.24 | markong | AL13N_lappy, exactly my point, that was foggy and unnecessary sarcasm, very prone to generate confusion. |
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09:51.37 | kjetilho | in combination with: 17:07 <drmessano> Wouldn't want you to not have bugs to chase down |
09:51.48 | kjetilho | I don't see how this can not be understood as sarcasm |
09:52.15 | kjetilho | (this was the very next line, without anything coming in between the two statements.) |
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10:15.10 | markong | OP 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. |
10:20.56 | kjetilho | excellent. |
10:33.07 | Ravenheart | hi |
11:15.27 | sibiria | howdy |
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11:23.13 | AL13N_lappy | kjetilho: tbh: due to double negation, i kinda had to read this 3 times to get the meaning... sorry |
11:24.45 | AL13N_lappy | so, what about this patch for removal of AC_HEADER_STDC ? do you guys know a place where this can be talked about/submitted? |
11:24.51 | kjetilho | no need to be sorry! :) |
11:25.54 | file | the wiki has a guide for contributing a patch, https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process |
11:26.02 | kjetilho | their repo mirror on Github says: Use gerrit.asterisk.org |
11:26.49 | AL13N_lappy | this 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? |
11:26.54 | AL13N_lappy | file: thanks |
11:27.23 | file | menuselect will automatically build everything that is to be built by default, with dependencies met |
11:27.39 | file | it can also be controlled from the CLI, https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options |
11:29.33 | AL13N_lappy | file: thanks |
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13:26.11 | drmessano | kjetilho: Thank you for the validation. Let's make sarcasm great again. |
13:27.01 | kjetilho | double negatives are best avoided, though |
13:27.30 | drmessano | Unless fully intended |
13:27.50 | kjetilho | no, I mean for readability |
13:27.59 | drmessano | Seems like a big deal over nothing since additional explanation was provided |
13:28.07 | kjetilho | yes, agreed |
13:28.53 | drmessano | I think markong just likes to be offended. He may be French, idk. |
13:29.22 | drmessano | I'll check back in with him later |
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13:30.51 | kjetilho | let it lie, please |
13:41.15 | markong | drmessano, 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. |
13:42.37 | drmessano | markong: 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. |
13:43.17 | drmessano | I go to a sleep, wake up, guess who's been talking shit? Get over yourself |
13:43.33 | drmessano | One damn comment you didn't like and you're triggered for life. |
13:43.46 | drmessano | Grow up |
13:47.26 | markong | drmessano, WTF you're talking about? last time I've highlighted your nick was yesterday! Speaking of trolls...would you please stop? It's enough. |
13:48.18 | drmessano | Yep, and I walk away again and again you've highlighted me. Putting you on ignore you worthless fuck |
13:49.10 | drmessano | Bye markong markong markong |
13:49.42 | markong | Childish idiot. |
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13:55.50 | Samot | markong: Are you packaging a distro? |
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14:12.00 | igcewieling | I think I shall leave for a while. too much drmessano. |
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14:17.54 | Samot | Yeah, that's the problem. |
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14:53.11 | AL13N_lappy | hmm well... i feel like i've triggered something i shouldn't have... |
14:54.20 | AL13N_lappy | anyway: when building asterisk, i come accross something like this: |
14:54.20 | AL13N_lappy | "The existing menuselect.makeopts file did not specify |
14:54.20 | AL13N_lappy | <PROTECTED> |
14:54.20 | AL13N_lappy | <PROTECTED> |
14:54.20 | AL13N_lappy | <PROTECTED> |
14:55.05 | AL13N_lappy | the menuselect.makeopts does NOT include app_voicemail_odbc ... why is it complaining about something that's NOT in there? |
14:55.54 | file | menuselect.makeopts specifies what should not be built |
14:56.25 | AL13N_lappy | how should i mark it NOT to be built? do i need to prefix it with '-' or something? |
14:57.29 | AL13N_lappy | file: wait, do you mean that all the stuff in there is what NOT to build? |
14:57.36 | file | yes. |
14:58.04 | AL13N_lappy | huh, then i guess i was doing it the other way around, thanks; it kinda makes sense now... |
14:58.19 | AL13N_lappy | file: what about that bottom part on FAILED stuff? what does that mean? |
14:58.35 | file | that question lacks context and I can not answer |
15:00.05 | AL13N_lappy | file: eg: in that file, there's also MENUSELECT_DEPSFAILED=MENUSELECT_CHANNELS=chan_pjsip |
15:00.22 | AL13N_lappy | does that mean that this chanpjsip doesn't work due to failed deps or something? |
15:01.13 | Samot | AL13N_lappy: You triggered nothing. |
15:01.15 | file | yes. |
15:01.24 | AL13N_lappy | ok, thanks |
15:07.46 | AL13N_lappy | file: 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? |
15:08.05 | file | it will list in the make menuselect interface itself |
15:11.30 | AL13N_lappy | file: ic, thanks |
15:12.07 | AL13N_lappy | i 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... |
15:12.19 | AL13N_lappy | but i found some stuff in makedeps that's wrong |
15:12.24 | AL13N_lappy | i'm testing that now |
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15:45.41 | AL13N_lappy | file: ok, i couldn't do interactive, but i figured out dependency tree |
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18:36.36 | Posterdati | hi |
18:36.39 | Posterdati | please help |
18:37.45 | Posterdati | I'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 #...) |
18:38.12 | Posterdati | I checked with nslookup yyy.zzz.it and the server is unreachable |
18:38.40 | Posterdati | could be a provider side problem? Thanks for help |
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19:22.07 | Dovid | @file: Why the -1 for https://gerrit.asterisk.org/c/asterisk/+/15066? Becuase it's not ready? |
19:22.13 | file | yes |
19:22.26 | file | "Marking -1 per comment of not to review" |
19:23.02 | Dovid | file: 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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19:37.30 | AL13N_lappy | file: 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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19:38.12 | file | codec_opus as provided by Sangoma is distributed as a binary module due to potential patent concerns |
19:39.38 | AL13N_lappy | pfff, 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... |
19:39.51 | file | you are free to use other implementations |
19:39.58 | AL13N_lappy | file: also, if you build against the opus library, then you don't actually contain it... so? |
19:40.13 | AL13N_lappy | file: ic |
19:41.19 | AL13N_lappy | file: 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? |
19:41.39 | file | the codec_opus module as provided by Sangoma is closed source, I can not comment on any other implementation |
19:41.49 | AL13N_lappy | ok, thanks |
19:42.11 | AL13N_lappy | file: do you know of another implementation or can point me to one? |
19:42.29 | file | I can't comment on such things |
19:42.51 | AL13N_lappy | oh, you work for Sangoma? |
19:42.54 | AL13N_lappy | ok, fair enough |
19:43.02 | file | I do. |
19:43.08 | AL13N_lappy | file: thanks for all the packaging help though |
19:44.38 | AL13N_lappy | file: 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? |
19:44.46 | file | yes. |
19:44.47 | AL13N_lappy | file: or at least, that was my understanding |
19:44.52 | AL13N_lappy | ok, thanks |
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20:34.33 | DannyA | does anyone happen to have the firmware to convert a mitel phone to sip? |
20:40.57 | drmessano | DannyA: Which device is it? |
20:41.19 | DannyA | 5330 |
20:43.40 | drmessano | You can reboot the phone with the volume down key pressed. |
20:43.40 | drmessano | Once you are in the phone configuration go to âPhone modeâ |
20:43.40 | drmessano | then âProtocolâ and then you can change the Phone Mode from Minet to SIP. |
20:46.16 | drmessano | DannyA: ^ |
20:46.33 | DannyA | let me try hang on (thank u!) |
20:46.41 | seanbright | that'll be $499 |
20:53.13 | drmessano | $1 for my time and $498 for knowing what to googlwe |
20:53.15 | drmessano | $1 for my time and $498 for knowing what to google |
21:28.24 | Kobaz | is there a way to turn off connnected line updates in chan_sip |
21:28.56 | Kobaz | chan_sip is sending UPDATE and the remote peer does not take kindly and is sending back 500 internal server error |
21:29.07 | Kobaz | not sure if you can... on chan_sip |
21:38.07 | Kobaz | of course pjsip can do it |
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22:53.02 | markong | Samot, no. |
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23:27.12 | jeev | there 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 |
23:27.12 | jeev | the 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 ? |
23:27.17 | jeev | it's a complete load of crap |
23:28.34 | Kobaz | well this is fun |
23:29.14 | Kobaz | AMI -> ModuleCheck -> Module: res_pjsip Returns: Response:Success Version: |
23:30.39 | Kobaz | jeev: transfer tracking is something I spent a year on getting 100% right on asterisk 1.8 |
23:30.49 | Kobaz | It's never been an easy problem to solve |
23:31.03 | Kobaz | It's for sure easier on new-er asterisk's |
23:31.18 | Kobaz | But by no means is straightforward at all |
23:31.45 | Kobaz | But, it's 150% doable using Asterisk, you just need to be a developer to do it |
23:32.14 | jeev | so 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 ? |
23:33.17 | Kobaz | correct |
23:33.26 | Kobaz | many developers aren't software engineers |
23:33.33 | Kobaz | most developers are not, in fact |
23:34.06 | Kobaz | web 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 |
23:35.36 | jeev | yea, 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. |
23:35.42 | jeev | when we're talking about five figures per year |
23:35.51 | Kobaz | I'll do it for 5 figures a year |
23:36.00 | Kobaz | Our software tracks that no problem at all |
23:36.38 | jeev | eh it's not just that |
23:37.55 | Kobaz | and the issue is... |
23:38.22 | jeev | it's an entire suite that communicates with salesforce, stores recordings on (ONLY amazon s3,. because wasabi isn't worth using apparently) |
23:39.15 | Kobaz | ah |
23:39.25 | Kobaz | we've done some integrations with salesforce |
23:40.13 | jeev | things seem ok, except transfers which is fatal |
23:41.35 | Kobaz | but to answer your question, Asterisk, especially modern Asterisk, is more than capable of handling that particular need |
23:42.19 | jeev | yea i wish i knew exactly what to say, how to say it. it's a joke |
23:56.04 | AL13N_lappy | speaking 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? |
23:58.27 | AL13N_lappy | is there a SIP message that changes source or callerid or something while E never hung up? |
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