IRC log for #asterisk on 20210126

00:00.23grummund_another option would be to have asterisk re-write callerid name based on a table lookup, maybe.
05:49.51*** join/#asterisk infobot (ibot@96-86-209-99-static.hfc.comcastbusiness.net)
05:49.51*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.0, 16.16.0 (2021/01/21) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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07:03.39iliusI'm one of the guys that does the sucky static XML directory that my Polycom phones get from provisioning.
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08:52.51grummund_It has the sucky static XML directory service, and i got it to work, but the phone won't use that for lookups on incoming calls.
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11:30.12ckbhow different is the ARI between 13 to 16?
11:31.20ckbfor reference: I'm only trying to get a list of pjsip extensions
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11:38.42filenot very different at all in that regard, but ARI has no PJSIP specific stuff - it just allows you to list endpoints, https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Endpoints+REST+API
11:40.23ckbfile: that's where I ended up. But I'm -really- lost in this regard.
11:41.22fileAMI is how people query for things.
11:41.34ckband the ARI is?
11:41.45filefor writing telephony applications
11:42.10fileexample: writing your own voicemail application
11:42.14fileor writing your own call queueing application
11:42.24ckbso if I wanted to create a web app to handle calls?
11:42.33file"handle calls" has many different meanings
11:42.38fileif you mean third party call control, then AMI
11:42.38ckbI'm just asking questions that I'm not tackling right now.
11:43.19ckbso the AMI is what I would want to use to list pjsip extensions (regardless if no devices are registered)?
11:43.27fileyes.
11:43.41ckband does the AMI and ARI use the same API credentials?
11:43.48fileno.
11:44.09ckbcan I get a bit more details on this?
11:44.23filehttps://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4817239
11:44.41filehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+16+AMI+Actions
11:45.12ckband is the AMI available in 13 as well?
11:45.19fileyes.
11:46.17ckbnow we have FreePBX running on top of asterisk... and the GUI allows us to turn on the ARI / AMI(I think) ports while FreePBX has it's own firewall (I think)?
11:46.32fileI don't do FreePBX, so don't know its interaction
11:46.57ckbokay, that's fine. does Asterisk have it's own firewall (iptables)?
11:47.12fileno.
11:47.40ckbgot it. and is there a way to set up authentication on asterisk using the CLI?
11:47.46ckbfor the AMI/ARI?
11:47.50filein the Asterisk CLI? no.
11:47.55fileit uses .conf files.
11:48.32ckbokay so the same auth for ARI won't affect AMI?
11:48.45filethey are configured in different files, and are different systems
11:48.52ckbgot it. thank you.
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16:08.15ckbHow can I add an AMI or ARI user? is there a sample I can reference? I know it's through the config files.
16:11.04Samotckb: Have you tried the wiki?
16:11.33ckbSamot: I have tried to wiki
16:11.48ckbI have CLI access, but not webaccess to FreePBX to create a user
16:11.55SamotWait
16:11.57ckbor even open the port
16:12.00SamotThis is FreePBX?
16:12.07SamotThen you need to do things properly in the GUI.
16:12.28ckbI, being a developer, am trying to skip FreePBX
16:13.10ckbI am literally brand new to Asterisk and FPBX having a CLI
16:13.55SamotThis isn't something new.
16:13.59ckbI started trying to modify the database...
16:14.00SamotAnd you should be using the GUI.
16:14.16ckbI'm trying to build a web app to do what FPBX does, tbh
16:14.28ckbbranding, I guess...
16:14.56SamotSo you're trying to recreate what FreePBX does
16:15.02SamotIn you're own GUI.
16:15.07ckbso I've bandaid'd around to get what I want... but what I really want in the AMI and ARI
16:15.19ckbPartially, yes
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16:15.43ckbs/in/is
16:16.06SamotOh is this the "I got a 100 PBX systems" thing?
16:16.08ckbmy boss failed to tell me there was API's I could utilize.
16:16.15ckbYES
16:16.23SamotOK and they are all running FreePBX?
16:16.28SamotOr just pure Asterisk?
16:16.33ckbyes 13-16
16:16.36ckbFPBX
16:16.38SamotTo which?
16:16.44SamotOK so why are you not using the GUI?
16:17.03ckbbecause I'm trying to build my OWN GUI
16:17.08SamotWhy?
16:17.23SamotWhy aren't you taking the OSS software, forking it and then building off that?
16:17.31ckb60 year-old people won't understand what the FPBX GUI does
16:17.40ckbOSS?
16:17.42SamotAnd they'll understand yours?
16:17.48SamotOpen Source Software
16:17.49ckbThey have so far?
16:18.23ckbIt started with SMS with the carrier, then got into FPBX
16:18.35ckband I was having to dig through the asterisk DB
16:18.43SamotWell you need to understand a lot more to this.
16:18.54ckbthat's why I'm here asking questions.
16:18.56SamotNot just how to pull data via AMI/ARI or from the database.
16:19.50SamotSo you're trying to do all this on a FreePBX box?
16:19.50ckbI've set up pub/priv keys for the servers to access said 100 FPBX servers
16:20.24ckbI have root access from my front facing servers to these servers to access the CLI (asterisk -rx)
16:20.32SamotOK
16:20.33SamotJFC.
16:20.38ckbHELP
16:20.40SamotFreePBX does things ITS OWN WAY
16:21.00ckbIt's not just a GUI on top of Asterisk?
16:21.02SamotIt breaks up the normal Asterisk .conf files into multiple files for holding configs
16:21.45SamotYou need to understand how FreePBX works and how it manages Asterisk before you start just messing with things.
16:22.05SamotOtherwise you could make changes outside of FreePBX and the next time FreePBX is reloaded/restarted your changes go away
16:22.20ckbI already hit these road blocks
16:22.27ckbhence me trying to figure out AMI/ARI
16:22.44SamotThat isn't going to be the end all answer.
16:22.57ckbit's just the "bandaid" as I said
16:23.35SamotYou need to understand both FreePBX and Asterisk if you're trying to make a PBX to do what FreePBX does.
16:23.36ckbthis is, again, why I'm reaching out for help and understanding
16:23.47SamotAnd I'm giving you help.
16:23.53SamotYou need to start reading things.
16:24.03ckbI have read A LOT.
16:24.05SamotLooking at wikis, sample files
16:24.18SamotYou need to look at the actual FreePBX code
16:24.30SamotTo see how things are being done if you want to mess with the 100 FreePBX systems.
16:24.57ckbI think ultimately, I need to know how VoIP works
16:25.04SamotOh dear god
16:25.14ckb<PROTECTED>
16:25.51ckbI have a super basic understanding, but I built a FTP server in C++ over TCP/IP, so I'm not completely useless.
16:26.14SamotAnd that has nothing to do with any of this.
16:26.21ckbUDP, yes.
16:26.37SamotFTP != SIP
16:26.47ckbyes I know.
16:27.03SamotSo what is this application of yours supposed to do?
16:27.11ckbbut I understand network connectivity and how devices communicate.
16:27.43ckbRight now, I'm using Bandwidth for SMS. But now I'm trying to expand it to the voice side.
16:27.53SamotSo what is this application of yours supposed to do?
16:28.55ckbI am trying to get call logs (right now mysql.cdr query) and system status (cli: "pjsip show contacts" and "sip show peers")
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16:29.40ckbfor a list of extensions (online or offline [regardless of devices registered])
16:30.10SamotThat's it?
16:30.14ckbyes.
16:30.34ckbeventually probably a web portal to answer/create calls
16:30.44ckbBUT that's in the roadmap
16:31.37ckbso right now I'm only using the CLI to find extensions, but if no devices are listed, pjsip show contacts doesn't list it.
16:31.54ckbpjsip show endpoints does, but very hard to parse
16:32.28ckb^^^^ this is the hangup
16:33.06SamotWhy aren't you just pulling that from the database?
16:33.22ckbwell the database is so large, and I don't know it well
16:33.25SamotWhy do you need to us AMI to show the endpoints that are already in the database.
16:33.41SamotWell it sounds like with an app like yours, you should be learning it
16:33.58ckb.... this is why I'm asking questions
16:33.59Samot'We want to make a replacement for FreePBX to do what FreePBX does'
16:34.04SamotNo.
16:34.05SamotGD.
16:34.13SamotYou need to put some footwork in on this
16:34.34SamotNo one, I mean NO ONE, is going to sit here and hold your hand and explain the database schema or other details like that to you.
16:34.54ckbOk.
16:35.11ckbHard to understand a schema when I don't even know the process.
16:35.20SamotWell..
16:35.27ckbI didn't even know the diff between sip and pjsip before 3 days ago
16:35.27SamotA smart project leader would then get someone that does
16:36.26ckbdidn't even know there was DAHDI
16:36.51SamotWell basically at this time and moment, you're the wrong one for the project.
16:37.19SamotMaybe six months down the road after you've actually learned some stuff about Asterisk and FreePBX at some deeper levels, you could be the right one.
16:37.21ckbThanks for the positivity.
16:37.31SamotIt's called being realistic.
16:37.39SamotAnd I said *at this moment and time*
16:37.48SamotAs in today.
16:38.14SamotYou lack the needed understanding and knowledge of all the pieces involved in making a PBX GUI for Asterisk.
16:38.30ckbInstead of criticizing my current knowledge and wasting time, you could provide me other knowledge? I don't know.
16:38.46SamotWhat knowledge would you like to me to lay down on you?
16:38.47ckbIt's a simple GUI I'm asking for.
16:39.05ckbI'm not trying to replace FPBX by anymeans.
16:39.06drmessanoHave you tried scouring the wiki?
16:40.08ckbI've parsed sip show peers and pjsip show contacts and TRIED pjsip show endpoints but I was pointed to the AMI rather than the CLI
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16:40.58drmessanoThe CLI is not an API
16:41.31drmessanohttps://wiki.asterisk.org/wiki/display/AST/Asterisk+16+AMI+Actions
16:41.39ckbdrmessano, It's been less than a week I've been trying to figure this out. I have 100+ FPBX servers that I need to turn the AMI/ARI on for and set up credentials.. but I don't have the FPBX credentials to even dig around atm
16:42.03SamotWhy don't you?
16:42.18drmessanoSo let me get this straight
16:42.18ckbMy boss failed to mention there was an ARI/AMI
16:42.33ckbhe thought it was "out of date" and there "was no documentation"
16:42.40SamotWhy don't you have access to the FreePBX boxes?
16:42.46Samotckb: That's bullshit.
16:42.48ckbI have access to the boxes.
16:42.57ckbI have root access to all the boxes.
16:42.59SamotThere's docs, there's sample files. It's just plain bullshit.
16:43.08SamotWhy don't you have GUI ACCESS?
16:43.17ckbHe's not a software guy, as I am.
16:43.24ckbI don't have the password
16:43.25drmessanoYou don't understand how this works, so you're asking on IRC.. but you can't actually apply anything that's being discussed here, because you don't have access.  So not only are you asking for ELI5 stuff, but you can't even understand the ELI5 part because you have nothing to reference.  Dude, this could go on forever
16:43.30SamotWhy don't you?
16:43.37drmessanoGet access, read the wiki, ask questions when you have something to reference
16:43.39SamotWhy don't you have GUI access to the PBX systems you manage?
16:44.05ckbI have everything BUT
16:44.19ckbRoot, mysql root...
16:44.39drmessanoSo nothing then
16:44.44drmessanoCLI and DB access
16:44.46SamotSo instead of logging into the GUI and using the ARI manager
16:44.50drmessanoNot the actual API
16:44.58SamotWhich does all this stuff for you with pointy/clicky things.
16:45.01ckbCLI and DB yes
16:45.07drmessanoSo nothing then
16:45.09drmessanoGet access, read the wiki, ask questions when you have something to reference
16:45.19SamotYou want us to explain to you all the steps you'll need to take to setup an ARI account via the backend you don't underestand?
16:45.20ckbbut I'm trying to make an app to do the "pointy/clicky" things
16:45.29drmessanoThis is like being a windows user and asking someone to explain Linux without you using it
16:45.41drmessanoGet access, read the wiki, ask questions when you have something to reference
16:45.59ckbI can set up access to the ARI or AMI via config files myself?
16:46.04SamotI don't understand why there isn't a test/sandbox FreePBX for all this
16:46.10drmessanoThey'll be overwritten
16:46.14drmessanoSo it's useless
16:46.16ckbSamot: I have said the same shit
16:46.19drmessanoYou need access
16:46.35Samotckb: So this is a test to mess with PRODUCTION SYSTEMS?
16:46.48Samotckb: You're going to be testing all this BS on live systems
16:46.50Samot?
16:46.58drmessanoA core developer pointed you to the wiki
16:47.03ckbI am already dealing with live systems
16:47.09Samot.......
16:47.12ckbdrmessano, I have the wiki
16:47.17SamotWhat kind of mickey mouse show is this?
16:47.22drmessanoNow get access and apply the concepts
16:47.36ckbsighs
16:47.40ckbafks
16:48.09drmessanoThis is "Tell me how this works without me looking at it"
16:48.10Samothttps://usercontent.irccloud-cdn.com/file/1QAIr4JA/IMG_3039.JPG
16:52.45Samotckb: You know this would go a lot better if this was a "I want to do Y, I did X for it but I got Z results. What did I miss?" style conversation.
16:53.27SamotAnd then you can show work and we can help.
17:15.25drmessanoI guess that was my point as well.  "Tell me how this works so I know what to do when I get in there" is wrought with basic, questions like "what happens if I turn the wheel left" that are easily answered being behind the wheel.
17:19.29igcewielingToo harsh?  'I figured we'd "do a sales" and ignore stuff we don't understand and figure it out during the install.'
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19:14.54KobazHey umm... never really dug into this.  Polycom phone number display on the screen shows as 1231231234... anyone know if there's a formating option to auto-format this as XXX-XXX-XXXX
19:27.10igcewielingKobaz: As far as I know, there isn't.  You'll have to use CONNECTEDLINE to set a CallerID you format yourself in the dialplan.
19:32.24igcewielingAsterisk might filter out the -.
19:33.57igcewielingI just set this via an AGI: SET VARIABLE CONNECTEDLINE(all) "VMAX Toll Riverside, CA <1-909-701-7758>"
19:33.57igcewieling<PROTECTED>
19:34.27igcewielingasterisk pjsip.conf
19:34.31igcewielingoops
19:38.43Kobazah okay
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23:17.31iliusERROR[10226]: res_pjsip_session.c:2053 resolve_refresh_media_states:  PJSIP/<redacted>-00000076: Attempting to remove stream 1:audio-1 but it doesn't exist anywhere.
23:17.41iliusWARNING[10226]: res_pjsip_session.c:2296 sip_session_refresh:  PJSIP/<redacted>-00000076: Unable to merge media states
23:17.52ilius👆I get this error every once in a while when a call is answered between two polycom phones and the channels are bridged, but it isn't fatal and both audio streams work.
23:18.18iliusMy Google-fu can't seem to find out what this actually means, besides finding the code itself at github.  I'm not sure how to fix whatever this is.
23:18.26iliusDo any of you know off the top of your head? :D
23:19.34iliusThis is Asterisk 16.8-cert5.
23:23.52filehave you tried Asterisk 16.16.0?
23:26.40iliusI've been sticking with certified versions to hopefully stay with the most stable branch of code I can get.
23:27.34filenormal branches are "stable"
23:27.39fileand numerous people use them
23:28.49iliusI also have several patches to app_queue.c and the certified release schedule is more my speed. :D
23:29.03fileso be it
23:30.13iliusI guess I could try 16.16 without my patches just to see if the error is there.
23:30.41iliusI guess this isn't something you've seen or can tell what's going on off the top of your head, so it must be an obscure error.
23:31.01filehaven't seen it at all.
23:31.50iliusThank you sir.
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