IRC log for #asterisk on 20210201

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01:21.07pearce15https://www.irccloud.com/pastebin/xQsEt8vO/
01:21.34pearce15Been getting this message from a client trying to register with TLS, not sure where to begin.
01:21.34pearce15<PROTECTED>
01:21.35pearce15pjproject: <?>:  SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336151576> <SSL routines-ssl3_read_bytes-tlsv1 alert unknown ca> len: 0 peer: 10.0.0.51:5061
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01:29.24drmessanoAre you using self-signed certs?
01:45.25pearce15@drmessano Yes.
01:46.05drmessanoread the message
01:46.09drmessano"unknown ca"
01:46.24drmessanoSo the client is trying to validate the cert
01:50.14pearce15Thanks, so I generated the client cert/keys and generated a PEM file with the CA cert to the phone. That should work right? Not sure what else I am missing here.
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01:53.17drmessanoIf the phone has the whole chain, sure
01:53.18drmessanoBut
01:53.30drmessanoMost client have an option to ignore the cert validity
01:53.40drmessanoIf you're using a self-signed, you may as well check it
01:53.52drmessanoIt's not like it's doing anything more than securing the channel at a basic level
01:53.59drmessanoThere's really no trust level to assure here
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04:45.05avbfile: https://github.com/avbdr/asterisk-res_json
04:47.07avbsorry, it took me a bit to get the final code without cJson library. I think im missing one more code review to fix couple small memleaks and this week ill work out licensing question with an initial module developer
04:49.24avbin what form do you want me to get this permission to upstream the code?
04:49.44avbsome kind of an email to the mailing list or?
04:49.52avbthank you
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07:30.22ZombieCan I ask an insane question?
07:30.54ZombieWould there be a way to interconnect a Skype call with a SIP Conference call?
07:31.06ZombieOr Discord...
07:38.26drmessanoYou can pay $7 a month or so add SIP to Skype
07:38.29drmessanoper channel
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10:58.09fileavb: license agreements are signed on https://issues.asterisk.org/jira
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13:02.42MarkS-Hello, short question. Is it possible to completely move extensions.conf to the database? without having to set anything in extensions.conf for every context
13:08.29SamotYou would need to look into Real-Time.
13:10.42MarkS-I did and could not find any examples/documentation for this part
13:25.25avbthey are all over
13:25.42avbhttps://www.voip-info.org/asterisk-realtime-extensions/
13:25.45avbfor example here
13:29.51filesuch an action has consequences, though
13:30.10fileyou tightly couple your phone system to another process - the database
13:34.55nbjoergfile: sounds like something a phone company would do, just without COBOL
13:35.46avbfile: i always do :)
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13:50.11MarkS-I want to create a web interface for it, so file isn't the thing I'm hoping for and looking for an alternative
13:50.40MarkS-< avb> https://www.voip-info.org/asterisk-realtime-extensions/ << that seems to still require every context to be in the file
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14:02.07SamotSo why does a GUI need real-time?
14:05.58MarkS-when clients change something they want to get the change at that time
14:06.11MarkS-And the gui will be on a central system
14:06.21MarkS-(not on the asterisk server)
14:06.36MarkS-so a database connection from the gui to the asterisk server is easier than file access
14:09.18SamotYou do realize that pretty much all the GUI based PBX systems using Asterisk do not rely on Real-Time?
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14:14.14TandyUKid suggest totally decoupling your web interface and the config...  or have fun when a user does somethign stupid, and kills your pbx
14:15.05TandyUKalso needs multiple layers of snaity checking what $user tries to do imho
14:15.12TandyUKsanity*
14:17.47MarkS-Samot: this will be something created in house and yes it will have sanity checking where possible
14:18.35SamotMarkS-: I understand that. I'm just pointing out that major projects don't use Real-Time.
14:24.58MarkS-We at least need something else later this year than what we are doing now (manually changing extensions.conf). We already have all users in the database (and we will not change that).
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14:26.15SamotNo one said a database cant be used to store data
14:27.39nbjoergwell, I'd say one major advantage of using a database is that you don't need to reinvent a parser
14:28.19SamotWhat parser?
14:28.29nbjoergfor extensions.conf
14:28.43nbjoergor you depend on noone else touching things which is asking for trouble as well
14:28.56SamotYou mean to generate it?
14:29.31nbjoergSamot: dealing with existing config files falls into one of two categories. understanding them or requiring everyone else to keep their hands off them
14:29.41nbjoergSamot: both are problematic requirements
14:29.56nbjoergso manipulating structured data certainly is attractive
14:30.07SamotSane for the database
14:30.14SamotSame
14:30.37SamotYou put something wrong in the DB you are in the same spot
14:30.44nbjoergyou don't parse free text for the database
14:31.00SamotYou still need to make DB calls
14:31.09SamotMore of them. There is a cost
14:32.38nbjoergthere are tradeoffs, sure. but dealing with a database (and one that enforces at least some basic constraints) makes things a lot less fragile for an UI
14:33.13SamotNo one said the GUI required anyone to touch configs
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14:43.33avbdatabases is the best what happened with an IT world :)
14:44.02avbbut I do not recommend using dialplans from the  database
14:44.06avbin short its a mess
14:44.44avbthe best shot to make it is to utilize regular dialplans + func_odbc + curl
14:45.33avbfunc_odbc is a bit funky with default settings, but its very handy
14:46.11avbI would suggest to read via func_odbc and to write its better to utilize rest api with curl
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14:46.38avbfunc_odbc on a high load can lead to the deadlocks on write
14:48.21SamotWell my original intent was just pointing out that a GUI management system does not require the use of Real-Time.
14:48.33avbnot at all, correct
14:48.35SamotAs just about all major projects don't use it.
14:48.40avbhuh
14:48.50avbi think so
14:48.57SamotFreePBX is the biggest.
14:49.01SamotIt doesn't use it.
14:49.20SamotThere are limitations to Real-Time.
14:49.25avbi wouldnt consider freepbx as an example of the best solution :)
14:49.47SamotSo to give everyone a real PBX experience you would either have to not do Real-Time or do pieces outside of Real-Time
14:50.06SamotAnd what you would consider an example of a best solution?
14:51.06avbif we are talking about boxed solutions, Thirdlane
14:52.28avbthey dont use realtime too if you will ask :-D
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14:52.58avbrealtime works pretty good for the queues,extensions,vms,etc
14:53.05avbbut not for the dialplan
14:53.07SamotYes, I know Thirdlane doesn't do it.
14:53.16Samotqueues have limitations.
14:53.31avbasterisk queues is a mess even without a realtime :-D
14:54.12avbrealtime queues work fine, its a queue members which a problem
14:54.57avbif you pull queues lists from the db and then using QueueMember* apps to manage agents, it works pretty good
14:55.07SamotWell that would then be a limitation to using queues.
14:55.17avbthe only problem that if you have like 1000 queues -- asterisk would die :(
14:55.21avbyes, indeed
14:55.54avbwhat i been saying that if you know the bottlenecks, it works pretty good :)
14:55.55SamotSo again, an offering like Thirdlane or FreePBX would have to do a bit in Real-Time and the rest static.
14:56.15SamotInstead of trying to support two different methods...
14:56.49avbplus i do not see any pos of managing dialplan out of the dialplan since its just a pure mess
14:57.16avbits waaaay easier to make a static dailplan and use curl/odbc to  manage the data
14:57.32SamotAnd as I said before, there's a cost to that
14:57.42SamotSo I guess it all depends on the costs you are willing to pay.
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20:58.08catphishi'm having fun trying to make some shoretel sip phones work with asterisk this week, no idea if this is a worthwhile endeavour but it's interesting, it has a proprietary config and claims to be SIP, but it's TLS only, think i'm getting there
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22:13.31catphishis it possible for TCP-SIP to use the same connection for requests in both directions? it seems like this would be essential for NAT, but i can't work out how to configure it in PJSIP
22:14.16catphishi specifically mean, when a telephone device with no fixed address registers, i'd like the TCP connection it initiates to be used for SIP requests to that device
22:18.29SamotDo you want them to exclusively use TCP?
22:18.57SamotYou just need to setup a TCP based transport and statically set that transport in the endpoint.
22:19.43catphishyes, i didn't set the transport in the endpoint, i will try that, thanks
22:20.27filerewrite_contact=yes which cause connection reuse
22:26.05catphishfile: that worked, thank you, it also deletes the contact when the connection is terminated which is cool
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22:36.10catphishlooks like what i actually needed was "sip outbound" which isn't supported in asterisk 16, but should be in 18, i'll upgrade
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22:50.52catphishoh, i don't think this is supported for inbound connections :(
22:55.34catphishone more question: is it possible to rewrite the contact and not tell the registering device you've done it?
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23:36.53catphishat this point, i fear this phone's sip implementation is just too bespoke to work with asterisk, it registers with a fake contact (a sips: address that doesn't listen) and expects connection reuse, but will disconnect if the contact header is rewritten, it then sends NOTIFY messages that expect an OK response, but asterisk rejects them because they have no contact header
23:38.24SamotThey were designed for a specific platform.
23:38.44SamotSo thats a downside
23:39.09catphishindeed, they are sip, but not designed for compatibility
23:39.30catphishi was just hoping i could breathe some new life into them
23:40.01catphishbut i think it would involve hacking asterisk and i don't have that much dedication this
23:40.14catphishthanks for the help anyway
23:42.17SamotYup

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