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| 00:23.27 | catphish | i have another question: can a pjsip channel sanely act like a FXS? by which i mean can it answer a sip invite but then behave like an analog line that hasn't dialled anything yet, playing a dial tone, and accepting dtmf into a dialplan? i'm sure this could be hacked together, but is this a standard thing to do at all? |
| 00:28.20 | catphish | this may seem an odd request, the reason is that i'm considering whether users might get a better experience with ATAs if they dialled asterisk immediatly when they went off hook, and asterisk took over the process |
| 00:36.06 | [TK]D-Fender | That is not a SIP thing period. PJSIP is no different |
| 00:36.17 | [TK]D-Fender | Normally you dial a completed number b the time the client is hapy with it |
| 00:36.54 | [TK]D-Fender | Otherwise you need the phone to literally dail on "anything" then hit an IVR regtardles of what then send and emulate the experience |
| 00:41.22 | sibiria | you can simulate the experience with some dial plan magic |
| 00:42.11 | catphish | thanks, i thought i could simulate it with a dialplan context, but wondered if i was missing anything more native / standard |
| 00:42.33 | catphish | hopefully this won't be necessary and ATA hardware will handle it well enough |
| 00:43.10 | [TK]D-Fender | ATA = FXS devide that DOES the dialtone, etc |
| 00:43.26 | Samot | Yeah, why does Asterisk need to do that? |
| 00:44.26 | catphish | my thought was only that it would avoid the need to manage dialplans on ATA devices, particularly where they may vary between customers |
| 00:45.14 | [TK]D-Fender | Maybe you should follow standards rather than let each invent their own and have that imposed on your hardware |
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| 00:46.01 | catphish | standards? |
| 00:50.23 | Samot | You can either do it in the dialplan of the device |
| 00:50.27 | Samot | Or you do it in Asterisk. |
| 00:50.43 | Samot | You really shouldn't be letting everything just pass through. |
| 00:51.30 | catphish | can ATAs send INVITEs for each digit and get back a "this number isn't complete yet"? |
| 00:51.38 | Samot | No. |
| 00:51.40 | catphish | shame |
| 00:51.43 | Samot | That's not how it works. |
| 00:51.50 | Samot | That's not even how POTS works. |
| 00:52.16 | catphish | what did you mean by "Or you do it in Asterisk"? |
| 00:52.22 | Samot | In the dialplan. |
| 00:52.48 | catphish | oh, well sure, the issue is purely the ATA knowing when to send the invite |
| 00:53.07 | Samot | It sends the INVITE when its digit timeout is reached |
| 00:53.16 | [TK]D-Fender | So enable the auto-dialout feature if it has it |
| 00:54.10 | catphish | timeout isn't a nice user experience imo, so i'm hoping i can avoid that (at least nationally) with good dialplans |
| 00:54.18 | Samot | What do you mean? |
| 00:54.31 | Samot | People pick up the phone, they dial digits, it goes out. |
| 00:54.49 | Samot | If you want to limit the timeout, you make it that way in the device dial patterns. |
| 00:55.00 | catphish | thy dial digits, then they have to wait before the call it initiated |
| 00:55.07 | Samot | What do you mean? |
| 00:55.20 | Samot | They dial the digits, it gets sent, the call is processed, they are hearing ring back |
| 00:55.34 | Samot | Or whatever that might happen. |
| 00:55.37 | catphish | you just said it, It sends the INVITE when its digit timeout is reached |
| 00:55.48 | Samot | Right? |
| 00:55.50 | Samot | And? |
| 00:55.52 | catphish | ie they have to dial fast, and then wait |
| 00:55.55 | Samot | Even POTS had digit timeouts |
| 00:56.05 | Samot | Then you set the patterns. |
| 00:56.17 | catphish | right, and that's what this conversation is about |
| 00:56.30 | catphish | whether those patterns could be managed in asterisk |
| 00:56.36 | Samot | OK. |
| 00:56.43 | catphish | or whether i have to push them out to the ATAs |
| 00:56.53 | Samot | The pattern in the phone is "10digits, send immediately" |
| 00:57.07 | Samot | Not if that person has the right to call that destination. |
| 00:57.08 | catphish | right, but sadly it's nowhere near that simple |
| 00:57.13 | Samot | You still need to deal with that. |
| 00:57.25 | catphish | but hopefully i can come up with one that works for everyone |
| 00:57.33 | Samot | It's not that hard. |
| 00:57.36 | Samot | Seriously. |
| 00:57.46 | Samot | I've been doing this for almost two decades. |
| 00:58.25 | Samot | How fast you want the device to send digits is up to you. |
| 00:58.31 | Samot | The standard timeout is 3 seconds. |
| 00:58.41 | Samot | In a lot of devices. |
| 00:58.52 | catphish | 3 seconds is reasonably sane |
| 00:59.16 | catphish | worst case i can probably match some numbers to dial instantly, and make shorter ones have to time out |
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| 01:00.51 | catphish | uk phone numbers are an unforgivable mess |
| 01:01.09 | Samot | But what you're suggesting is that when the device goes off hook it "hotlines" to Asterisk. |
| 01:01.15 | Samot | It still needs to dial something |
| 01:01.19 | Samot | It just does it auto dial |
| 01:01.47 | Samot | So now it dials 000 and Asterisk takes the call. Now everything that happens is in a call |
| 01:01.55 | catphish | yes, i was assuming the ATA was configured to auto-dial into something |
| 01:01.59 | Samot | Or you're giving them an IVR to enter things into. |
| 01:02.17 | catphish | yeah, that was the hack i was considering, but thought i'd ask for thoughts on it |
| 01:02.33 | catphish | frankly, i hate the idea, so i think i'm going to put it out of my mind :) |
| 01:30.06 | catphish | so, i have one more question before i sleep, i need to associate an area code with each of my sip endpoints so that when they dial a local number, i know what area they mean, am i best off 1) having a context per customer (endpoint) 2) a context per area code 3) some kind of file based lookup of this information? or is this one of those things that asterisk simply has no opinion on and i need to work out what works for me? |
| 01:32.15 | catphish | i suspect a context per area code is going to be the most efficient solution for now, unless there's a way to have pjsip set arbitrary channel variables from its own endpoint config, i looked but didn't find such a thing |
| 01:33.39 | Samot | catphish: I would use the AstDB. |
| 01:33.54 | catphish | can i push data into astdb statically? |
| 01:34.05 | Samot | catphish: When user A makes a call load their prefix |
| 01:34.12 | Samot | Yes. |
| 01:34.49 | catphish | well that's ideal if i can do that, i love astdb, but have only ever stored data in it dynamically, not from my config |
| 01:35.24 | Samot | I write out numerous entries for users/tenants. |
| 01:37.33 | catphish | how do you populate it? |
| 01:38.00 | catphish | i found someone doing it with individual asterisk shell commands |
| 01:41.33 | catphish | yep, people seem to suggest just sending a script to asterisk -rx with "database put" commands |
| 01:41.42 | catphish | seems reasonable enough |
| 01:43.43 | Samot | AMI |
| 01:47.21 | catphish | oh ok, that is extremely useful, i think for this requirement i will find a way to do it with static config (ie more contexts), but being able to manage astdb via ami will very likely be useful for other things later |
| 01:48.18 | catphish | thanks for your help today! it's great to talk through configs sometimes, if only to realise why they're a bad idea :) |
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| 10:55.06 | fling | Can you invite me to #voip? |
| 10:55.19 | fling | Should I use linphone or prefer another client? |
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| 18:32.05 | igcewieling | Any ideas what might cause Asterisk to send a SIP 480 reply? |
| 18:39.17 | Samot | User is busy |
| 18:40.43 | Samot | Well Asterisk could use that four no answer |
| 18:40.53 | Samot | For* |
| 18:46.09 | igcewieling | Return a busy when there is no answer? |
| 18:47.39 | igcewieling | Our hosted platform returns a 486 instead of a 404 on calls to DIDs not loaded into the system. 8-| |
| 18:47.59 | igcewieling | Anyway, I'll track the cause of the 480 eventualy, I was just wondering if there is a common cause. |
| 18:48.24 | Samot | 480 is Temporarily Unavailable |
| 18:48.35 | Samot | Some devices send that when in DND |
| 18:49.07 | igcewieling | At least with Asterisk / FreePBX there is a chance to making a change. Our Bicom hosted service runs EVERYTHING inside an encrypted php AGI. |
| 18:49.24 | igcewieling | Gawd, I hate that system. |
| 18:49.33 | Samot | So this was an unallocated number? |
| 18:49.54 | igcewieling | in my 480, I don't know yet. for the 486 on hosted, yes. |
| 18:52.10 | igcewieling | for now I'll send a busy on 480 |
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| 19:58.17 | towser | how do I find the ip address of my server when running astrisk on deepin? |
| 20:01.58 | Samot | The same way you find it without Asterisk running on the server. |
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| 20:04.18 | towser | well I did try ifconfig but the boradcast ip didn't access the gui when I went to it |
| 20:05.17 | file | Asterisk has no GUI. |
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| 20:23.13 | Kobaz | goooooey |
| 20:23.26 | Kobaz | towser: what gui are you using? perhaps freepbx? |
| 20:24.12 | towser | Kobaz, I'm trying to add a gui |
| 20:24.22 | Kobaz | sounds fun |
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| 20:35.51 | igcewieling | generally you get the hostname, then lookup the IP of the hostname using DNS. |
| 20:36.38 | igcewieling | towser: don't create a general use GUI for Asterisk. Use FreePBX. The only case where that is not the case is where you have specific tasks/applications for which you want a custom gui. |
| 20:37.29 | Kobaz | towser: what's your use case? |
| 20:38.50 | towser | trying to install a freepbx to use with astrisk |
| 20:41.32 | igcewieling | use the distro and be done with it. |
| 20:52.30 | towser | the question is. can the distro be used as regular linux as well [normal desktop pbx in backgorund[ |
| 20:52.41 | Samot | No. |
| 20:52.44 | Samot | The distro is a distro |
| 20:52.53 | Samot | Like deepin |
| 20:53.15 | Samot | However, the original issue was about finding the IP of the server. |
| 20:53.34 | Samot | That has nothing to do with Asterisk or FreePBX. If you can't find the IP of the server, you have bigger issues to work out. |
| 20:58.11 | catphish | can i trigger a pjsip reload (ie reload endpoints, auths, aors) via AMI? |
| 20:59.12 | igcewieling | Perhaps you are asking if you can run CLI commands from a script? |
| 20:59.14 | catphish | i guess i need to choose between "reload" or "command(pjsip reload)" |
| 20:59.59 | catphish | well ami has a native reload command that (i assume) does a core reload, but i now realise i can run arbitrary cli commands, so i can just reload it with the "pjsip reload" command |
| 20:59.59 | igcewieling | like this, for example: $response = $asm->command("sip show peers"); |
| 21:00.37 | catphish | igcewieling: thanks, so in my case: command(pjsip reload) |
| 21:01.54 | Samot | No. |
| 21:01.56 | igcewieling | pjsip reload isn't really a command. |
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| 21:02.14 | Samot | module reload res_pjsip.so |
| 21:02.26 | Samot | Or the pjsip module you need to reload |
| 21:02.53 | igcewieling | perhaps you aliases it in cli_aliases.conf to module reload res_pjsip.so |
| 21:03.04 | Samot | *CLI> pjsip reload qualify |
| 21:03.04 | Samot | aor endpoint |
| 21:03.08 | catphish | really? that command works for me, it reloads all the pjsip modules |
| 21:03.18 | Samot | pjsip reload *is* a command, just not for what you think. |
| 21:03.24 | catphish | oh okay |
| 21:03.41 | Samot | pjsip reload qualify aor <aor> |
| 21:03.44 | Samot | or |
| 21:03.53 | Samot | pjsip reload qualify endpoint <endpoint> |
| 21:04.05 | catphish | i specifically want to reload pjsip.conf, specifically *all* endpoints, aors and auths |
| 21:04.16 | Samot | module reload res_pjsip.so |
| 21:04.44 | catphish | ideal, thanks |
| 21:04.49 | Samot | Or like igcewieling said, cli_aliases.conf and make a command that does that plus any other pjsip modules you want to reload. |
| 21:04.56 | Samot | Like mwi, etc. |
| 21:05.09 | catphish | pjsip reload seems to reload all of them |
| 21:05.09 | Kobaz | towser: so you *are* using freepbx. that was my first question |
| 21:05.30 | towser | Kobaz, no I have deepin with astrisk installee |
| 21:05.40 | Kobaz | a what? |
| 21:05.42 | towser | installed* |
| 21:05.45 | Samot | *CLI> pjsip reload |
| 21:05.45 | Samot | No such command 'pjsip reload' (type 'core show help pjsip reload' for other possible commands) |
| 21:05.52 | Samot | That by itself does nothing. |
| 21:05.58 | Samot | Unless it's been aliased. |
| 21:06.02 | Kobaz | yeah module reload res_pjsip |
| 21:06.08 | catphish | Samot: weird, maybe it's aliased in my default config |
| 21:06.12 | Kobaz | which, is a little silly IMO |
| 21:06.19 | Samot | That's not a default setting |
| 21:06.23 | Samot | Which means someone had to do it. |
| 21:06.28 | Kobaz | pjsip reload would be more expected |
| 21:06.33 | igcewieling | "cli show aliases" will tell you. |
| 21:06.35 | Samot | Unless this is some odd distro based release of Asterisk. |
| 21:06.51 | catphish | pjsip reload module reload res_pjsip.so res_pjsip_authenticator |
| 21:07.01 | catphish | it's an ubuntu package |
| 21:07.47 | catphish | <PROTECTED> |
| 21:07.56 | catphish | there it is, part of ubuntu's default config |
| 21:08.10 | catphish | good to know |
| 21:08.37 | catphish | i'm using ubuntu's default config as a starting point, i plan to strip it back one file at a time as i develop |
| 21:08.54 | Kobaz | Samot: what's the reason for all the extra module reloads... doesn't reloading res_pjsip do the trick |
| 21:09.12 | Samot | No. |
| 21:09.25 | Kobaz | interesting |
| 21:09.34 | catphish | i'm still dangerously on the fence between using realtime, and using a static file for my pjsip config |
| 21:09.45 | Kobaz | I use static-realtime |
| 21:09.50 | igcewieling | you don't want to use realtime. |
| 21:10.05 | Kobaz | my configs are generated from the db, but they are statically loaded |
| 21:10.13 | catphish | igcewieling: any particular reason? it appears to work very well, and i'm definitely using it for ps_contacts |
| 21:10.24 | igcewieling | write a little script to generate pjsip configs from the database. |
| 21:10.43 | Samot | There is a cost |
| 21:10.43 | igcewieling | catphish: qualify doesn't work until the first packet from the endpoint arrives. |
| 21:10.47 | Samot | We've talked about this. |
| 21:11.07 | Samot | The is a resource cost for hitting the database for everything. |
| 21:11.11 | catphish | igcewieling: are you sure? i'm sure i tested this and it does |
| 21:11.26 | catphish | ie it loads all contacts in at startup, my config depends on this in fact |
| 21:11.52 | Samot | Loading all the contacts doesn't mean they qualify |
| 21:12.00 | igcewieling | perhaps static realtime is different. Samot, Kobaz any comments? |
| 21:12.02 | catphish | my very reason for putting ps_contacts in the database is so that peers are immediately reachable after a failover to a different server |
| 21:12.19 | Samot | You're confusing things. |
| 21:12.32 | catphish | actually this is easy to test, i will kill and start asterisk and see if it qualifies |
| 21:12.39 | Samot | Being able to send a request to the endpoint is not the same as Asterisk qualifying it. |
| 21:12.47 | catphish | i know :) |
| 21:12.51 | Samot | OK. |
| 21:13.12 | igcewieling | then immediatly do a "pjsip show contacts" |
| 21:13.57 | igcewieling | that shows you the qualify status in a simple format |
| 21:15.12 | catphish | my setup is a mess right now, but i'll retest that asap |
| 21:29.48 | catphish | igcewieling: tested, the contacts load immediately from realtime, they show up as "NonQual" for the first qualify period, then all qualify |
| 21:32.53 | catphish | interestingly, peers not qualifying until first packet is why i stopped using pjsip with chan_sip about 10 years ago, but i'd definitely say it's fixed now |
| 21:33.09 | catphish | *stopped using realtime with chan_sip |
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| 21:34.25 | Kobaz | igcewieling: static realtime is no different than hard coded config files in terms of end-result functionality |
| 21:34.44 | Kobaz | all the behefits of realtime-db-generated configs, with none of the drawbacks of having just static files |
| 21:34.57 | Kobaz | you could generate the static files of course from the db and do it that way |
| 21:34.59 | Samot | In other words, it's word salad. |
| 21:35.11 | Kobaz | but i like the consistency of having one and only one official configuration |
| 21:35.17 | Samot | Because anything that stores config details in a database then writes out static configs is "static realtime" |
| 21:35.23 | Kobaz | no |
| 21:35.43 | Kobaz | Samot: static-realtime is specifically enabled by using extconfig.conf |
| 21:35.53 | Samot | OK. That's mixing the two |
| 21:36.04 | Kobaz | Everything else, ie: generating config files, is just that... generating config files |
| 21:36.30 | Kobaz | some things are more realtime than others, ie: voicemail is true realtime |
| 21:36.43 | Samot | So.. |
| 21:36.45 | Kobaz | using extconfig... and then pjsip/sip/extensions.conf etc, needs a reload to trigger anything interesting |
| 21:36.54 | Samot | I have customers who's configs haven't changed in years. |
| 21:37.00 | Samot | Is there a benefit to realtime for that? |
| 21:37.19 | Kobaz | It completely removes the possability of hand-editing config files, so i would say yes |
| 21:37.30 | Kobaz | you can be assured that your database matches your "config files" |
| 21:37.33 | Samot | No, I can store information in a database and write it out to a file |
| 21:37.48 | Kobaz | and then those files can be later hand-edited, and be out of sync with the db |
| 21:37.57 | catphish | the benefit to me is that guarantee that the config matched the database, and only having to push config to one place in a cluster |
| 21:38.25 | Samot | OK |
| 21:38.31 | Samot | So I have multiple servers, right now. |
| 21:38.34 | Kobaz | yup |
| 21:38.42 | Samot | All using static config files with data stored in a database. |
| 21:38.44 | catphish | but on the other hand, since "the database" isn't actually my primary data source, (another database is), i'm actally swaying towards generating and pushing out files |
| 21:38.54 | Samot | I can push that data from the database to any of those servers. |
| 21:38.57 | Samot | WRite out the configs |
| 21:39.29 | Kobaz | sure |
| 21:39.36 | catphish | right, i think my use case is more liek Samot describes, i have a central management database, then i want to push config out to multiple clsusers (in my case 2 servers per cluster in pairs) |
| 21:39.37 | Kobaz | Samot: are you the only one with access to these machines? |
| 21:39.42 | Samot | No. |
| 21:39.58 | Samot | But you're confusing non technical problems with technical solutions. |
| 21:40.02 | Kobaz | Do you have a well-defined process for updates? |
| 21:40.07 | Samot | Yes. |
| 21:40.14 | Kobaz | Okay, so, you're fine then |
| 21:40.22 | Samot | So basically this is a solution for idiots. |
| 21:40.26 | Kobaz | No |
| 21:40.31 | Kobaz | It just removes a step |
| 21:40.41 | catphish | the downside to realtime is that i'd have to actually synchronize 2 databases (the central database, and the asterisk database) which is more complicated than just writing a replacement static config |
| 21:40.48 | Kobaz | 16:40:35 <seanbright> Stop argueing with Kobaz |
| 21:40.50 | Samot | What? |
| 21:41.03 | Samot | catphish: I write out data to config files, I write out data to AstDB |
| 21:41.18 | Samot | catphish: This is also how major PBX systems using Asterisk work. |
| 21:41.37 | Kobaz | Samot: Some are worse than others |
| 21:41.52 | Samot | The AstDB is not needed for static configs |
| 21:41.58 | Kobaz | Samot: Ie: Xorcom with 1000 extensions takes like, 5 minutes to perform a reload |
| 21:42.05 | Samot | That's a problem on their end |
| 21:42.08 | catphish | Samot: yeah, that's making a lot of sense to me too as i trial both options, the only thing i'm in some doubt about is how to accurately sync the astdb |
| 21:42.11 | Samot | I don't have those problems. |
| 21:42.17 | Kobaz | our system can reload 2000 extensions in less than 10 seconds |
| 21:42.20 | Samot | How to sync it? |
| 21:42.23 | Samot | You write to it |
| 21:42.37 | catphish | Samot: write it in what way? |
| 21:42.47 | catphish | i mean, what mechanism? |
| 21:42.50 | Kobaz | A lot has to do with how they generate their dialplan, it's all like c++ template-style. They generate the same code for every extension |
| 21:42.51 | Samot | AMI |
| 21:42.56 | Samot | I said this already. |
| 21:43.02 | Samot | Asked/answered. |
| 21:43.03 | catphish | Samot: so just push a script that replaces every value? |
| 21:43.16 | Samot | How you decide to do that is up to you |
| 21:43.27 | Samot | You only want to modify things being modified at the moment, do it that way |
| 21:43.35 | catphish | the only issue there is old/abandoned values |
| 21:43.37 | Samot | Want to just push everything out on an update, do it that way |
| 21:43.39 | catphish | i don't want to flush it |
| 21:43.45 | Samot | What? |
| 21:43.59 | *** join/#asterisk gerhard7 (~gerhard7@86-87-238-48.fixed.kpn.net) |
| 21:44.06 | Samot | You don't want to update the old data? |
| 21:44.17 | Samot | What do you need to flush? |
| 21:44.19 | catphish | what happens with deleted values? when i generate a new config file, deleted values disapear on their own |
| 21:44.33 | catphish | but with astdb, deleted data would have to be actively removed, right? |
| 21:44.37 | Samot | These are two separate processed. |
| 21:44.40 | Samot | These are two separate processes. |
| 21:45.28 | catphish | so, the thing i love about config files, is that entries deleted from my database simply don't appear in a newly generated config file |
| 21:45.31 | Samot | A user logs into the portal. |
| 21:45.39 | Samot | They want to set themselves to DND. |
| 21:45.50 | Samot | They push a button, the AstDB is updated to show they are DND |
| 21:45.59 | Samot | I push an AMI command to update their devicestate |
| 21:46.09 | Samot | Done. |
| 21:46.14 | catphish | and if that user account is deleted, the astdb entry that they're DND stays forever? |
| 21:46.23 | Samot | Uhm. |
| 21:46.26 | Kobaz | Samot: does your dnd survive asterisk restarts? |
| 21:46.31 | Samot | I remove a user, it sends delete commands. |
| 21:46.33 | catphish | or do you have a method to actively purge it? |
| 21:46.35 | catphish | ok |
| 21:46.36 | Samot | It's the AstDB |
| 21:46.40 | Samot | Of course. |
| 21:47.22 | Samot | It's an sqllite database guys |
| 21:47.35 | Samot | Do those lose all their data when things are restarted? |
| 21:47.40 | catphish | i thought it was berkley |
| 21:47.40 | Kobaz | right yeah, custom device state saves in astdb |
| 21:47.47 | Kobaz | catphish: depends on the asterisk version |
| 21:47.55 | Samot | It hasn't been for years. |
| 21:48.01 | Kobaz | 1.8 uses berkley, 11+ use sqlite |
| 21:48.04 | Kobaz | I don't recall what 10 uses |
| 21:49.25 | Samot | See I can understand the use case of catphish on real time |
| 21:49.37 | Samot | But at the same time, this is also the reason I use Kamailio. |
| 21:49.59 | Samot | It just handles this scenario way better. |
| 21:50.24 | catphish | the truth is that kamailio is a complexity that i would love to avoid, and having tested asterisk 16, pjsip, and realtime, i'm convinced that realtime contacts is plenty good enough |
| 21:50.36 | Samot | It could be. |
| 21:50.59 | Samot | But honestly dude, after some of your suggestions lately complexity is not the hill you want to die on with that arguement |
| 21:51.17 | Kobaz | heh |
| 21:51.23 | Samot | I defer to the whole hotlining Asterisk conversation from yesterday. |
| 21:51.38 | Kobaz | hotlining? |
| 21:51.50 | catphish | wasn't asterisk literally designed to do that with PCI FXS cards? |
| 21:52.10 | Samot | Then you'll need to do that with FXS. |
| 21:52.21 | Kobaz | catphish: Samot just likes to argue |
| 21:52.22 | catphish | i realise it's unusual with SIP, but i maintain that it's not insane :) |
| 21:52.26 | Samot | But you're also conflating PBX features with POTS service. |
| 21:52.37 | Kobaz | you can hotline with SIP |
| 21:52.39 | Samot | FFS. |
| 21:52.39 | Samot | JFC. |
| 21:52.40 | Samot | That wasn't the point at all. |
| 21:52.42 | Samot | AT ALL |
| 21:52.45 | Kobaz | Set your SIP phone to 'autodial on pickup' |
| 21:52.58 | Samot | It was to also wait to accept digits from the user |
| 21:53.10 | Kobaz | WaitExten() or Read() |
| 21:53.12 | Samot | So if they wanted to make a regular phone call, the device would autodial into an IVR. |
| 21:53.15 | Kobaz | or whatever |
| 21:53.19 | Samot | FOR HOME SERVICES |
| 21:53.25 | Samot | AS A REPLACEMENT FOR POTS |
| 21:53.28 | Samot | JFC. |
| 21:53.29 | Kobaz | Yelling doesn't make it right |
| 21:53.31 | catphish | Kobaz: the question i asked yesterday was whether i could have my ATA dialdial SIP on pickup, then defer to an asterisk dialplan to process digits |
| 21:53.39 | Samot | Not something a user wants to experience. |
| 21:53.39 | Kobaz | catphish: sure can |
| 21:53.45 | catphish | Kobaz: as a relacement for POTS for home users |
| 21:53.46 | Samot | That was my whole comment about the hotlinging. |
| 21:54.04 | catphish | my idea was slated :) |
| 21:54.16 | Kobaz | [2021-02-14 16:52:21] <Kobaz> catphish: Samot just likes to argue |
| 21:54.20 | Samot | Because not telecom company uses IVRs for every call. |
| 21:54.35 | Kobaz | Samot: that's a pretty baseless assertion |
| 21:54.41 | Samot | ... |
| 21:54.44 | Kobaz | I'n sure there's at least one that does |
| 21:54.46 | Samot | It's 20 years of experience at telecoms. |
| 21:54.53 | Samot | But OK. |
| 21:55.05 | Kobaz | 'No telecom'? Remember, Don't use 'never' and 'always' |
| 21:55.11 | catphish | in any case, i agree that if i can get the dialplan into the ATAs accurately, that will be the better choice |
| 21:55.18 | catphish | so i'll try that first |
| 21:55.36 | Samot | Kobaz: Please don't argue on behalf of poorly done things. |
| 21:55.50 | Samot | Yes, there are exceptions. They are at the low end of the bar. |
| 21:55.54 | Kobaz | I'm just playing devils advocate here. You can't assume 100% of anything for anywhere doing anything |
| 21:55.56 | Samot | The wrong way to go. |
| 21:56.08 | Samot | No but in the context of catphish starting a voip service |
| 21:56.15 | Kobaz | Yeah don't do that |
| 21:56.18 | Samot | Let's not encourage bad practices. |
| 21:56.19 | Kobaz | The whole ivr thing sounds... weird |
| 21:56.20 | Samot | FFS. |
| 21:57.45 | catphish | anyway, as said, i'm only interested in maximizing reliability and customer experience in simulating a POTS, so will be keeping things largely as simple as possible |
| 21:58.36 | Samot | So you can do fun things like offer custom ringing. |
| 21:58.50 | Kobaz | why do you want to simulate pots again? |
| 21:58.52 | Samot | So when a caller is calling my number, instead of ringing they hear my custom ringing. |
| 21:58.58 | Samot | POTS is a level of service. |
| 21:59.02 | Kobaz | the ATA does a pretty good job of that |
| 21:59.11 | Samot | POTS is a level of service. |
| 21:59.17 | Samot | Plain Ole Telephone Service |
| 21:59.19 | Kobaz | right |
| 21:59.23 | Kobaz | we know what it means |
| 21:59.23 | catphish | Kobaz: indeed, and and ATA will be the main component in doing so |
| 21:59.34 | Kobaz | the ATA will play dialtone and ringing |
| 21:59.40 | Kobaz | ...and all that fun stuff |
| 21:59.47 | Samot | You totally are missing what I just said |
| 21:59.59 | Kobaz | Yes you can do custom ringing, why do you want to? |
| 22:00.01 | Samot | Some mobile services allow you, as the customer, to set a custom ringback |
| 22:00.15 | Samot | So that when a friend calls you, they hear that ringback |
| 22:00.18 | Kobaz | You can do custom ringing without an ivr |
| 22:00.22 | Kobaz | yeah true, but you don |
| 22:00.23 | Samot | I know |
| 22:00.27 | Kobaz | 't need an ATA for that |
| 22:00.27 | Samot | I know that |
| 22:00.37 | Samot | I was suggesting that as a service feature |
| 22:00.40 | Kobaz | Right |
| 22:00.42 | Kobaz | Fair enough |
| 22:01.07 | Samot | If you want to provide basic phone service for people and have some extra features, that is one thing you can do |
| 22:01.17 | catphish | Kobaz: likely an ATA built into a fiber OTU, the aim here is to replace customer copper lines with fiber ones |
| 22:01.36 | Samot | I'm not sure how the UK market is but trying to be a residential provider in the US is a self abuse action. |
| 22:01.41 | Kobaz | yeah, speaking of fiber |
| 22:02.19 | Kobaz | Verizon went and sent a letter to one of our customers, we're upgrading your pots to fiber, they subsequently shut off the pots lines and never installed the fiber, so they have no fax line anymore |
| 22:02.23 | catphish | basically dumb phone lines for people who are giving up their POTS but want to keep their phone line |
| 22:02.33 | Samot | I left the residential market 7 years ago... |
| 22:02.49 | Samot | I understand that but all the major players offer it |
| 22:02.51 | Kobaz | We are strictly non-residential |
| 22:03.00 | Samot | And will bundle it with services like Fiber/TV |
| 22:03.14 | catphish | i've been doing hosted business PBX for 10 years, but i've just moved to a FTTH provider, so naturally i'm looking at phone provision |
| 22:03.15 | Samot | Making it really cheap if not even close to free |
| 22:03.33 | Samot | well if you are the one also offering fiber, that's the bonus. |
| 22:03.36 | catphish | but i haven't been that hands on with it since chan_sip stopped being cool |
| 22:03.37 | Samot | the voice is a add on |
| 22:03.53 | catphish | so making sure to relearn everything |
| 22:04.12 | Kobaz | chan_sip was always a bit of a pain |
| 22:04.13 | Samot | Because I'm guessing at the end of the day if I had a choice of fiber or voice with you, you'd want me to take fiber. |
| 22:04.42 | catphish | so now i only sell fiber internet, voip is just a necessity |
| 22:05.04 | Samot | Yes, see if I was an ISP I would do residential services again. |
| 22:05.18 | Samot | But I'm not an ISP so not going to waste the effort on it. |
| 22:05.22 | catphish | its not so bad when it's properly designed |
| 22:05.33 | Samot | It's about market saturation. |
| 22:05.36 | catphish | i wouldn't do home voip if i wasnt doing the fiber |
| 22:05.42 | Samot | ^^^^ |
| 22:05.44 | Samot | Exactly |
| 22:06.57 | catphish | so anyway, i just need simple lines with an inbound number, and voicemail, asterisk is overkill, but it does a good job of voicemail |
| 22:07.18 | catphish | and deploying in mysql replicate realtime pairs seems the simplest configuration to me |
| 22:08.39 | Kobaz | i would highly recommend not using mysql |
| 22:08.44 | catphish | i think i'll come down to using static config files, pushed to both servers in the pair, and database replication only for contacts, and voicemail messages |
| 22:08.59 | catphish | why not mysql? i'm extremely comfortable with mysql optimization and replication |
| 22:09.13 | Kobaz | Postgres > * |
| 22:09.35 | catphish | postgres is supposed to be good, but i don't have the experience with it that i do wit mysql |
| 22:10.04 | Kobaz | Mysql has been used for plenty of heavy lifting, but i've had a long history of my pet peves and issues with it |
| 22:10.25 | Kobaz | But if you want to use what your comfortable with, that's fine |
| 22:10.51 | catphish | it's not perfect, i have had some issues with its optimization, but not enough that i ever learned postgres |
| 22:11.07 | Kobaz | i've had my issues like, silent truncation and all that stupid stuff |
| 22:11.13 | catphish | and this was only when running many-TB SaaS databases on it |
| 22:11.17 | Kobaz | you can turn on warnings now for that, but it drives me nuts it's not standard |
| 22:11.35 | Kobaz | https://sql-info.de/mysql/gotchas.html |
| 22:11.36 | catphish | yeah, luckily silent truncation is very much considered a legacy mode now, that was awful |
| 22:12.38 | Kobaz | Some of these things have been fixed since, but... just, irks me like Visual Basic 5-6 was a complete rewrite |
| 22:12.56 | Kobaz | and Python 2>3... but anyway.... </rant> |
| 22:13.05 | catphish | yeah, makes sense, i strongly suspect postgres was better from day 1 |
| 22:13.30 | catphish | but i know i can make mysql (mariadb now) work well for me |
| 22:13.43 | Kobaz | It's been built with enterprise in mind, whereas mysql half way through was like oh, we need to half-ass bolt on some enterprise stuff... like you need a new storage engine to support transactions |
| 22:14.02 | catphish | at least it's not as bad as PHP :) |
| 22:14.15 | catphish | in which doing anything properly was an afterthought :) |
| 22:14.47 | catphish | i'm not familiar with python at all, so the 2/3 thing is just a mild annoyance for me when i get the wrong one |
| 22:15.06 | Kobaz | as a developer it sucks |
| 22:15.20 | Kobaz | They were like, f--- backwards compatability, sucks to be you |
| 22:17.18 | catphish | on a more on-topic note, i have just one outstanding question about my config - how to get arbitrary data about SIP clients (specifically their numeric area code) from my central database into the dialplan, it would be lovely if i could upload a database file, right now the best ideas i have are 1) put every peer into a context based on its area code 2) push all the mappings into astdb and look up in dialplan 3) push mappings into mysql and |
| 22:17.18 | catphish | look up in dialplan |
| 22:17.57 | Kobaz | context based on area code? |
| 22:18.03 | catphish | when a client phones a local number, i need to map it to a national number, based on *their* area code, which sadly isn't a fixed length at the start of their callerid |
| 22:18.06 | igcewieling | set_var in pjsip.conf won't help? |
| 22:18.09 | Kobaz | are you trying to do, on-net is free, type stuff? |
| 22:18.27 | catphish | Kobaz: not billing related, just want to allow "short" dialing |
| 22:18.33 | Kobaz | catphish: use includes |
| 22:18.36 | igcewieling | HAHAHAHHA! |
| 22:18.52 | igcewieling | where do you live, montana? |
| 22:18.59 | Kobaz | our routing system is like this. we have a 'class of service', which includes all the routes they can dial.. like 1XXXXXXXXXX and XXXXXXX type stuff |
| 22:19.06 | catphish | Kobaz: use includes? do you mean use a different context for each area? |
| 22:19.13 | Kobaz | so i would define a new context like, customerid-route-ld7 |
| 22:19.30 | Kobaz | and they would have their own ld7 route, which would prefix +1212 or whatever their area code is |
| 22:19.34 | Kobaz | let me show you |
| 22:19.40 | catphish | i can use a context for each customer, that would solve it |
| 22:19.51 | catphish | but the dialplan will be needlessly huge and need reloading |
| 22:19.51 | Kobaz | correct |
| 22:20.03 | Kobaz | you need to reload anyway typically |
| 22:20.06 | Kobaz | depending on what you're doing |
| 22:20.13 | Kobaz | unless you do a lot of stuff dynamically |
| 22:20.29 | catphish | you're right, i will probably need to reload it anyway |
| 22:20.51 | igcewieling | Write your dialplan so it doesn't need to keep changing. |
| 22:21.06 | catphish | i can go either way |
| 22:21.11 | Kobaz | https://dpaste.com/E6EPXA4V7 |
| 22:21.16 | catphish | i can make my dialplan totally dynamic (ie do it doesn't need changing) |
| 22:21.38 | Kobaz | so, if this customer needs international calling, add an include in the routes list for intl |
| 22:21.42 | Kobaz | otherwise, they can't |
| 22:22.19 | igcewieling | I block international calling by setting a flag in a database table then checking that flag in the dialplan. |
| 22:22.36 | *** join/#asterisk sinaowolabi (~Sina@102.134.114.1) |
| 22:22.40 | Kobaz | lots of ways to skin the cat |
| 22:22.41 | catphish | for example, my [inbound] can dial PJSIP customers simply by looking up their contacts by phone number PJSIP_DIAL_CONTACTS($EXTEN) |
| 22:22.54 | igcewieling | That is the sort of thing I meant by "write your dialplan so you don't need to reload the dialplan all the time" |
| 22:22.58 | catphish | there's no need to hardcode every customer |
| 22:23.35 | igcewieling | Kobaz: 's way is a very traditional way, using includes and contexts. |
| 22:23.43 | Kobaz | yup |
| 22:23.46 | catphish | similarly, voicemail can be looked up in mysql do see if it's enabled or not |
| 22:23.50 | Kobaz | we started doing it this way 12 years ago, so |
| 22:23.54 | Kobaz | it's kind of baked in |
| 22:24.02 | catphish | this means it kinda makes sense to do the area codes from the database too |
| 22:24.21 | igcewieling | catphish: why not set the area code in the pjsip endpoint/ |
| 22:24.28 | Kobaz | it works pretty well actually |
| 22:24.31 | catphish | igcewieling: oh, you mentioned that, how do i do it? |
| 22:24.32 | Kobaz | even with enormous systems |
| 22:24.35 | catphish | that would be better |
| 22:24.36 | Kobaz | the reloads happen pretty quick |
| 22:25.05 | catphish | if i can set arbitrary channel variables in pjsip endpoints, i'll be happy |
| 22:25.15 | igcewieling | see set_var |
| 22:25.18 | Kobaz | igcewieling: it depends what you want to optimzie for |
| 22:25.29 | Kobaz | igcewieling: i like the idea of having as much stuff in memory to avoid db lookups as possible |
| 22:25.38 | igcewieling | I use set_var to specify a failsafe callerid for the peer. |
| 22:25.42 | Kobaz | igcewieling: it really helps with high-concurrency |
| 22:25.52 | catphish | oh yeah https://www.asterisk.org/did-you-know-you-can-use-variables-in-pjsip-conf/ |
| 22:26.00 | catphish | i couldn't find that when i looked before |
| 22:26.15 | catphish | igcewieling: thanks, that 100% solves my area code problem |
| 22:26.38 | catphish | set_var=AREACODE=01234 |
| 22:26.42 | Kobaz | right, you can set variables and then use them in dialplan |
| 22:26.45 | igcewieling | Kobaz: I run an agi when the call arrives and when it ends. Since I'm doing at least a few database lookups anyway, I don't worry TOO much about it. |
| 22:27.02 | Kobaz | igcewieling: yeah we're using AGI as well for most call-routing related things |
| 22:27.47 | igcewieling | My AGI sets a bunch of dialplan variables which the dialplan uses -- much like FreePBX. |
| 22:28.02 | catphish | just need to decide where to store "voicemail enabled", something the users can change by dialing a code, and i'll be done |
| 22:28.08 | Kobaz | we're the opposite, dialplan is the glue that puts steps from AGI to AGI |
| 22:28.30 | Kobaz | like dialplan is the necessary evil in between calls to AGI, heh |
| 22:30.06 | igcewieling | catphish: could you instead check for the existance of the mailbox in the dialplan. |
| 22:30.46 | Kobaz | yeah |
| 22:30.53 | Kobaz | VM* functions |
| 22:31.04 | catphish | i think the mailbox will always exist, it's just a simple enabled/disabled flag |
| 22:31.29 | igcewieling | have you read the output of "core show functions" and "core show applications"? |
| 22:31.53 | catphish | i have not, i will give it a go |
| 22:38.51 | *** join/#asterisk paulgrmn (~paulgrmn@c-98-250-183-21.hsd1.mi.comcast.net) |
| 22:39.58 | KaiHerlemann | Hi. I added an announcement to play a custom recording. If I play it from our 1st extension, it works. If I play it from the 2nd extension (per follow me â time condition or destination "if not reachable"), it says in the log "file.c: File custom/exmp does not exist in any format". What could be reason for this? |
| 22:40.29 | KaiHerlemann | It's the same announcement. |
| 22:42.33 | Kobaz | codec most likly |
| 22:43.05 | Kobaz | example: first extension is using ulaw and the second extension is using gsm... you might have custom/exm.ul but not custom/exm.gsm |
| 22:43.44 | Kobaz | for highest compatability, use .sln files or .wav files, or make sure you convert to each codec format you intend to use |
| 22:44.16 | Kobaz | and .wav being pcm 8khz mono... considering .wav is just a container for what could be many different formata |
| 22:45.09 | Kobaz | you can use 'core show channel xxxxx' to see what codec(s) are currently being used |
| 22:45.29 | Kobaz | throw the channel in a Wait(10) to keep it open long enough to inspect |
| 22:45.38 | Kobaz | or something similar |
| 22:46.35 | KaiHerlemann | It was the same announcement, so also the same file. (Actually two different files, but I tried it out with the same file/announcement on both extensions to find out where's the problem) |
| 22:47.11 | Kobaz | also, could be permissions |
| 22:47.25 | Kobaz | make sure asterisk has read permissions to the file and +x to the entire path to get to that file |
| 22:47.50 | Kobaz | I don't recall if file.c will report back permission issues or just not-exist |
| 22:50.16 | KaiHerlemann | Second message in the log was "file.c: Unable to open custom/AB (format (alaw)): No such file or directory", means actually the same like the first message. |
| 22:50.24 | KaiHerlemann | This was immediately after the first one. |
| 22:50.27 | igcewieling | If you were using Asterisk and not FreePBX, I'd suspect you added the file extension in the Playback |
| 22:50.54 | KaiHerlemann | No, I use FreePBX. |
| 22:51.04 | Kobaz | KaiHerlemann: at this point... use dpaste and show your dialplan, and show ls -al on your custom/ files |
| 22:51.18 | Kobaz | KaiHerlemann: you'll be better served in #freepbx |
| 22:51.23 | Kobaz | !freepbx |
| 22:51.29 | Kobaz | ~freepbx |
| 22:51.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:51.56 | Kobaz | it uses asterisk, but there's many many many layers added |
| 22:56.28 | KaiHerlemann | Kobaz: thanks for the advice⦠yes, I was in the past also sometimes unsure if I should ask in the FreePBX or asterisk forum. There were problems in which case I had to add something extensions_custom.conf, it's not always clear if I need to choose a solution like that. |
| 22:59.51 | KaiHerlemann | I ask usually at first in the asterisk community if I'm not sure, because the very basic function is based on Asterisk. |
| 23:00.05 | Kobaz | yeah, but... freepbx is controlling what asterisk is doing |
| 23:00.33 | igcewieling | If you do it in the GUI, then it is "FreePBX" |
| 23:02.00 | Samot | So the files exist right? |
| 23:02.03 | KaiHerlemann | What would you advise in such a case? It's not always clear to me at first whether I need a custom-made solution, I notice that after work deeper into it/after ask. |
| 23:02.08 | Kobaz | KaiHerlemann: basic functions are not always so obviously basic if there's configuration settings or other environmental things going on, and it's typically more complicated than "this one function dosn't work" |
| 23:02.09 | KaiHerlemann | Samot: yes |
| 23:02.16 | Samot | In the right format? |
| 23:02.35 | KaiHerlemann | Samot: yes⦠it works with the other extension |
| 23:02.42 | Kobaz | KaiHerlemann: did you do what i suggested before? check your channel for the codecs in use |
| 23:03.28 | Samot | What format where these files uploaded in? |
| 23:03.45 | Kobaz | when you're checking code and such, when one thing works and the other thing doesn't and it seems everything matches and is all the same... there's the problem.. the thing that's not working has a difference of *some kind* |
| 23:04.18 | Samot | Well the errors say the files couldn't be find in the right or any format. |
| 23:04.25 | Kobaz | right |
| 23:04.28 | Samot | That looks more like something on the system |
| 23:04.32 | Samot | Not codecs. |
| 23:04.56 | Samot | KaiHerlemann: ls -l /var/spool/asterisk/sounds/custom |
| 23:04.57 | Kobaz | it could be codecs for sure, one device plays the announcement and the other does not |
| 23:05.07 | Samot | The system can't find the file. |
| 23:05.12 | Samot | That's what the errors state |
| 23:05.24 | Samot | One can't be found in ulaw, so sure a codec issue in a form. |
| 23:05.31 | Samot | The other wasn't found in any format. |
| 23:05.33 | Kobaz | it also could be playing a different file.. something is different |
| 23:05.42 | Kobaz | yeah true, maybe some non-printable character in there |
| 23:05.46 | Samot | Let's see if the files exist and what formats. |
| 23:05.50 | Samot | KaiHerlemann: ls -l /var/spool/asterisk/sounds/custom |
| 23:06.00 | Kobaz | Samot: yeah I asked for that a while ago |
| 23:06.07 | Kobaz | KaiHerlemann: dpaste ^^^^ |
| 23:06.30 | Samot | I will look after dinner |
| 23:07.54 | KaiHerlemann | Samot: the folder doesn't exist at all, do you mean /var/lib/asterisk/sounds/custom? |
| 23:08.12 | Samot | Yes |
| 23:08.14 | Samot | Sorry |
| 23:08.22 | Samot | Good catch |
| 23:09.15 | KaiHerlemann | Samot: there's also no file, the files are in /var/lib/asterisk/sounds/de_DE/custom/ |
| 23:09.44 | KaiHerlemann | right language is set for the extension⦠but this is probably FreePBX-specific ;) |
| 23:10.55 | Kobaz | KaiHerlemann: show the console right before the error |
| 23:12.55 | KaiHerlemann | Kobaz: I read what you wrote regarding codecs, I just didn't believe the codec is the problem, because it's the same file/announcement (and DAHDI card, although not the same ISDN line)⦠I didn't want to ignore you ;) |
| 23:13.10 | Kobaz | Okay thanks |
| 23:13.13 | Kobaz | Just sanity checking |
| 23:13.43 | KaiHerlemann | Kobaz: the line in the log before that was "pbx.c: Executing [s@app-announcement-2:5] BackGround("DAHDI/i2/0015128053848-12", "custom/AB,nm") in new stack" |
| 23:13.56 | KaiHerlemann | Oh shit, I should xxxx that |
| 23:14.01 | Kobaz | too late |
| 23:14.29 | Kobaz | but anyway... so, and what's your channel language set to? |
| 23:14.31 | igcewieling | no, you shouldn't. phone numbers are not secret. |
| 23:15.17 | KaiHerlemann | igcewieling: on the blacklist are anyway already 60 numbers ;) |
| 23:15.17 | Kobaz | yeah but, some troublemaker could very well go through the log for #asterisk and dial known numbers to be a pain |
| 23:15.25 | KaiHerlemann | *numbers = extensions |
| 23:15.37 | igcewieling | Kobaz: how would that be different than random scammers calling you? |
| 23:16.17 | Kobaz | it's not |
| 23:16.30 | Kobaz | just one less source of random scammers |
| 23:17.08 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 23:19.39 | KaiHerlemann | Kobaz: one moment please, I check the channel language⦠but FreePBX says, it's set to "German (Germany)" |
| 23:20.09 | Kobaz | it sounds like it's playing from custom/AB and not de_DE/custom/AB |
| 23:25.46 | KaiHerlemann | It think I need to execute "core show channels" to see the channel name? |
| 23:26.17 | KaiHerlemann | Then enter the channel name in "dialplan show chanvar"? |
| 23:26.44 | Kobaz | you can see channel variables/settings via 'core show channel ...' |
| 23:27.56 | Samot | So almost 30 minutes later I still haven't seen a list of files. |
| 23:29.40 | KaiHerlemann | Samot: "AB.wav dd.wav test-vw.wav" |
| 23:30.08 | Samot | So the files didn't exist. |
| 23:30.17 | Samot | AB did but not in ulaw. |
| 23:31.28 | KaiHerlemann | sorry, I had here several advices how to solve it, because of that I didn't posted the list of files. |
| 23:31.37 | KaiHerlemann | (Thanks for the advices!) |
| 23:31.47 | KaiHerlemann | *post |
| 23:32.34 | Samot | <PROTECTED> |
| 23:32.52 | KaiHerlemann | exmp = example in this case |
| 23:33.03 | Samot | Sigh. |
| 23:33.13 | Samot | So this was the only problem? "file.c: Unable to open custom/AB (format (alaw)): No such file or directory" |
| 23:39.03 | KaiHerlemann | In the case of the other extension (it works on this one), the log says: "file.c: <DAHDI/i1/xxxxxxx-19> Playing 'custom/AB.slin' (language 'de_DE')" |
| 23:39.11 | KaiHerlemann | Such a file doesn't exist, but it work.s |
| 23:39.42 | KaiHerlemann | Probably I really should ask in the FreePBX channel. |