IRC log for #asterisk on 20210304

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03:31.14igcewielingstays perfectly still and listens for the "h323....h323....h323...." call of the dangerous creature called a Time Sucker
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05:23.02igcewielinga Chupahora
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15:48.38Kobazso, this is weird.  https://dpaste.com/7R6RYWTNS
15:48.49KobazWaitForRing() never gets the ring
15:51.02filering and ringing are two separate things, ring is from analog land if I recall
15:51.23KobazAaah
15:51.53Kobazoddly enough this works in older asterisk's
15:52.19KobazI changed the flow to use SendDTMF and WaitDigit, works well
15:53.06Kobazoh wait, that's not the same one i used in 11
15:53.17KobazOkay so, the analog thing makes sense
15:53.33Kobazgood to know
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16:54.08gregfHi, How can I make it so my extension ring X number of times before going to voice VoiceMailMessage? I haven't figured that out yet, very new.
16:54.19gregfI have it now so it just goes directly to voicemail I did figure that much out
16:54.52gregfon asterisk 16 btw
16:59.42igcewielingYou can't.
16:59.59igcewielingYou can, however, make it ring X number of seconds.
17:00.53gregfOk, Well ring period?
17:01.13gregfI think I need to use Dial?
17:04.15gregfyep, it was dial... that took me way to long to figure out.
17:05.50gregfand Dail does take ring-time in  seconds btw...
17:06.24KobazUSA is 6 seconds per ring
17:06.36gregfKobaz: ty
17:06.37Kobazcore hsow application Dial
17:06.49Kobazyou can give a timeout parameter to Dial()
17:06.53Kobaz*core show
17:07.36KobazIn North America (excluding Mexico, Central America and parts of the Caribbean), the standard audible ringing tone is a repeated cadence of a two-second tone and four seconds of silence. The signal is composed of the frequencies 440 Hz and 480 Hz.[3]
17:07.41Kobazhttps://en.wikipedia.org/wiki/Ringing_tone
17:12.17igcewielingThat doesn't really apply to VoIP.
17:12.32KobazHmm?
17:13.33igcewielingWith analog, the ring cycle is done by the local CO, which is pretty standard.
17:13.40Kobazright
17:14.00igcewielingWith VoIP, it is handled by the remote phone, the local phone, the remote carrier, or the local carrier.
17:14.34Kobazwell haha yeah, but the ring cadance is typical
17:14.46Kobazacross those, (for sane systems)
17:15.58igcewieling*shrug*  My ring cadence is Triiplet-Low or something like that.
17:16.23Kobazthat's for receiving calls
17:16.24igcewielingI don't think the user actually specified Ring or Ringback
17:16.37Kobazright, i'm talking about ringback
17:17.05igcewielingI was referring mostly to ringing.
17:20.56csavinovich_Hello all, I have a question: How can I call a function immediately on lifting up the handset? Been a long time since I've used the dialplan. Off the top of my head I know I would do it using AMI and I would capture the triggered event, something "line Up" me thinks.  Will anyone know of a much simpler solution?
17:21.12igcewielingI'm weird.    When making a call I care about how many seconds it takes to go to Voicemail.   When receiving a call about the number of times the phone rings so I know how much time I have to answer it.
17:21.37igcewielingcsavinovich_: analog or SIP?
17:22.35igcewielingI'm weird.    When making a call I care about how many seconds it takes to go to Voicemail.   When receiving a call about the number of times the phone rings so I know how many rings I have to answer it.
17:24.17Kobazsome people want 'rings'
17:24.35csavinovich_<PROTECTED>
17:24.59Kobazactually a lot of people want rings, every manager I talk to at customer sites who helps design the systems we install in terms of call flow. they say something like "after 4 rings, this needs to go to voicemail"
17:25.35Kobazcsavinovich_:  you have an ATA like a cisco SPA or something like that, you need to set up the hot line feature, where picking up the handset dials an extension on the sip server
17:25.59igcewielingcsavinovich_: If analog on Digium or Sangoma, that is configured in chan_dahdi.conf immediate=yes.
17:26.22Kobazigcewieling: ..."most of the phones are actually analog attached to one of those boxes that do analog to digital"...
17:26.33igcewielingfor SIP or analog connect to a SIP adapter, that is handled by the SIP device.   Usually called "hotline" or "warmline".
17:26.39Kobazyup
17:30.16Samot"After 4 rings" roughly equals a 24 second ringtime.
17:30.46Kobazright
17:30.54Kobaz...just replying to: <igcewieling> I'm weird.    When making a call I care about how many seconds it takes to go to Voicemail.
17:31.14Kobazthe flip side is most office people I've run into count rings, not seconds
17:31.53SamotSIP systems do not do it in "rings"
17:31.59KobazWe know this
17:32.07SamotSo how you present that to the end user is up to you.
17:32.16SamotYes, I have users that say "rings"
17:33.04Kobazringalingaling
17:33.36SamotI mainly service hotels, so having special dialing rules for various types of phones is common place.
17:35.07KobazIt's always really interesting to get a feel for the use cases.  We get involved in custom app builds and really complex routing and tracking systems, which is why I run into all this weird stuff I do
17:35.30SamotAlong with "Comes from outside, ring one way", "comes from guest, ring another way", "comes from the outside doors, ring another way"
17:35.44Kobazsteps back and waits for Samot's response to "weird"
17:35.59SamotShrug.
17:36.01Kobazhehe
17:36.13SamotI've serviced all sorts of industries.
17:36.21SamotI'm use to special use cases.
17:36.48KobazSpeaking of special, I may have found a DoS for Asterisk-16, requires AMI
17:37.04SamotI mean, University of Michigan and pretty much 80% of the universities in Michigan at one point.
17:37.11KobazAh
17:37.23igcewielingSwap out the admin with a different one.  AMI should not be exposed on the internet. 7-)
17:37.28SamotHospitals, state buildings (reps/congress)..etc. etc.
17:37.59Kobazigcewieling: i mean like, if someone has developer access via AMI, and sends requests in a certain way, it's more like accidental DoS
17:39.15KobazAll I know so far is that I broke my asterisk pretty badly just by using cli and ami together.  More details to follow once I get the info in place and I read up on the current security discussion process
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18:06.52csavinovich_@igcewieling: It is an Adtran SIP device with analog phones connected to it.  It seems to send out an off hook message, anyone knows how Asterisk receives this message?
18:07.12igcewielingshow us the message it sends.
18:07.47igcewielingwithout hotline or wamrline setup, the Adtran won't send anything to the SIP server until it has collected all expected digits.
18:09.03csavinovich_Don't know it yet  :(...   I guess I will have to wait until I receive the documentation for it.  I would venture guess it will be a SIP message  :(
18:14.04igcewielingum, Adtran has extensive documentation online.
18:16.01csavinovich_yes, I will be receiving some logs soon.  Thank you igcewieling,  you cleared it up :)
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18:28.00igcewielingJust ssh/telnet into it or access it with a web browser.   you can't do what you want to do without access to the device.
18:33.30KobazIs it normal that module reload res_pjsip.so blocks new calls?
18:33.36Kobazabout 800 endpoints
18:34.04SamotHavent seen it and I got more then 1000
18:34.26Kobazfor me it will not process a new call until the reload is done
18:34.44SamotHow long does that take?
18:34.57Kobazabout 5 seconds
18:35.01Kobazsometimes 10
18:35.27SamotI dont think it should take that long.
18:35.39SamotId have to test
18:40.06SamotYeah, it was basically instant.
18:40.24Kobazhmm
18:40.41Kobazlet me compare reading plain text vs extconfig
18:41.04SamotOh you're using realtime?
18:41.15Kobazsemi realtime
18:41.30Kobazit's loading a static config by selecting from a db function
18:41.32SamotYou have stuff in the database you're calling with realtime, yes?
18:42.00SamotSo reloading res_pjsip.so is loading data from a database?
18:42.10Kobazcorrectr
18:42.30SamotSo you're using the pjsip_endpoints via realtime?
18:42.34Kobazno
18:42.42Kobazit references them in memory once it's loaded
18:43.05SamotSo how is reloading res_pjsip.so calling from a database?
18:43.11Kobazvia extconfig
18:43.26*** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
18:43.28Kobaz[settings] pjsip_wizard.conf => pgsql,pbx,asterisk.asterisk_exten_config('PJSIP_WIZARD')
18:43.55SamotSo yes, realtime.
18:44.29SamotI use static only, so it was almost instant.
18:45.06Kobaz# explain analyse select * from asterisk.asterisk_exten_config('PJSIP_WIZARD');   Execution Time: 241.959 ms
18:45.25Kobazso, that should be the only overhead
18:47.07SamotWell can't really help. realtime never had an impactful benefit for me.
18:47.26Kobazi just like not relying on real files
18:47.44Kobazbut, something I did notice, is that the query is run 10 times, during the cource of the reload
18:47.53Kobaz*course
18:48.03Kobazso that's going to add 2.4 seconds to whatever else it's doing
18:48.07SamotThen you wouldn't be "semi-realtime"
18:48.17Kobazyeah
18:48.27KobazIn Asterisk-11 it was the same as loading a static file
18:48.30Kobazit would query it once
18:48.31SamotSee realtime has a performance cost.
18:48.45SamotI don't think that cost is worth it.
18:48.55Kobazi'm going to run this through my exporter and make a static file and compare
18:49.10fileand if you were using chan_sip it may do the same, but chan_sip is not sorcery and PJSIP, and thus is different
18:49.12KobazWell, this was never al issue, even with 2000 endpoints, it would be, instant, like you said
18:49.15SamotI'm reloading 100% more endpoints at a fraction of what you are doing.
18:49.20Kobazfile: yeah I'm finding that out
18:49.49KobazSamot: i essentially have a ~200ms reload time using this method and chan_sip, so i never had a performance issue
18:51.53Kobazhence, never having to do anything differently
18:52.04KobazBut yeah, I might need to change this around then
18:53.21Kobazhere's the issue i'm having
18:53.43Kobazres_pjsip is taking so lone to reload, and while it's loading it hangs onto a lock on module_list
18:54.05Kobazand then I have other things in AMI that check for certain thigns on startup, like which modules are loaded, and then they block
18:54.25SamotI'm just going to say moving from chan_sip to chan_pjsip is not a parallel  move.
18:54.26Kobazand then other things block like pj subscription lists, and then everything grinds to a halt sometimes
18:54.32KobazSamot: of course
18:54.40KobazI'm just saying why i never had performance issues
18:55.30SamotWell I doubt that. Just at a different level.
18:55.42Kobaz?
18:55.44SamotBut yeah, you're going to be hunting these things down for a bit.
18:55.52Samotchan_sip had limitations.
18:55.54KobazWhat level would that be?
18:55.55KobazWell yeah
18:56.04SamotIt can't handle the number of sessions pjsip can.
18:56.18Kobazfair enough, I never had that high of a session load
18:56.18SamotOr with the CPU/memory performance pjsip can
18:56.20filechan_sip was a known factor, chan_pjsip isn't
18:57.11KobazYeah, I know our own caveats of chan_sip very well, and can easily design around that
18:57.27KobazSamot: yup, reloading 800 endpoints with a static file is, almost instand
18:57.31SamotI think part of this is the fact that there were so many kludges for chan_sip performance they became standard.
18:57.38Kobazand no long-term locks being held
18:58.06SamotSo people by default were already accounting for chan_sip issues when doing setups because that's just what had to be done.
18:58.08seanbrightso just use chan_sip
18:58.16Kobazallrightey, refactoring here we go
18:58.26Kobazseanbright: yes! that's the answer!  </kidding>
18:58.49seanbrightis it not?
18:59.11KobazAt least I have all the hooks already for extension changes and writing a file would be literally adding one line of code
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19:42.13TandyUKhey guys, Im interested which android apps you use for sip calls, and maybe why.. (and yes i know theres like a billion writeups i could google)
19:42.27Kobazeither linphone or cloud soft phone
19:44.05TandyUKdo you know if they integrate properly when using bluetooth? (ie while driving - auto answer and/or voice dial, vi the bluetooth device)
19:44.20Kobazno idea
19:44.26Kobazlinphone is free, you can try it
19:44.36TandyUKfair enough :)
19:45.12TandyUKit bugs the hell out of me that samsung can nerf the native sip functionality and google let them
19:45.30TandyUKhttps://support.google.com/phoneapp/answer/2811843?  << Thats how hard it _should_ be lol
20:13.41SamotIt's an open source OS
20:13.53SamotBy that sheer fact, Google let them.
20:14.24SamotAnd honestly, everything I've read on the native SIP stack in Android ain't great.
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21:28.52TandyUKSamot: at least it worked in the past though, without samsung causing me to jump through hoops to undo whatever it is theyve done to stop it working
21:29.12TandyUKserves me right for buying a samsung device i guess
21:33.42SamotI'm sure Samsung is just one of the manufacturers who disable the native SIP dialer.
21:34.12SamotWho knows, depending on the carrier they could request it be disabled.
21:35.02SamotSamsung+ATT != Samsung+T-Mobile when buying from ATT or T-Mobile.
21:38.59igcewieling*shrug* Use Bria, for a softphone it is pretty good.
21:50.08TandyUKfwiw, this is on a bought-direct-from-samsung phone, which I was hoping would not have such nerfs
21:50.29TandyUKbut yeah, playing with various softphone apps is what im doing now :)
21:51.00TandyUKC Sip Simple  i think was the last android one I used
22:05.58Kobazoh
22:06.05Kobazrandom question. account.$reg_num.codec.opus.para=
22:06.08Kobazwhat's para ?
22:06.12Kobaz...for yealink phones
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23:03.27Kobazi broke it
23:03.37Kobazhttps://dpaste.com/5T7K32CMM
23:07.53Kobazguess i'll need more core debug
23:26.47Kobazaaaaah
23:26.50Kobazyou need to match dialplan
23:26.55Kobazi added debugging to res_parking
23:27.06KobazERROR[9793][C-0000003a]: Parking lot 'c10000' -- Failed to find extension '701' in context 'parkedcalls_c10000'
23:27.21Kobazi guess you need to populate the context with your parking destinations ahead of time
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