IRC log for #asterisk on 20200113

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13:35.26davlefouHi, where i can buy asterisk g729 liscence?
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13:43.01SamotPeople still need to use that?
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14:15.36ReinhildeSamot: you'd do well to not question it.
14:15.46seanbrightdavlefou: https://www.sangoma.com/asterisk/software/g729-codec/
14:15.56seanbrightnear the bottom of the page
14:16.36SamotI'd be well either way.
14:18.32Reinhildeseanbright took the correct approach to the situation here.
14:18.58SamotAsking if people still need to use an old codec isn't really that off..
14:19.01SamotIt's 2020.
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14:21.36ReinhildeBy the same argument, I could ask why, in 2020AD, there are still vegetarians in the world if someone who is a vegetarian asks me how to make their diabetes less severe. But I don't. I reformulate my dietary information around the fact that the person does not wish to eat meat.
14:22.12SamotOK.
14:22.21SamotYou're making a mountain of of this but that's fine.
14:22.22seanbrightevery ITSP i work with offers G.711 and G.729, so there is definitely still a need
14:22.38seanbrightReinhilde: you'd do well to call it CE and not AD
14:22.49ReinhildeTechnically correct is the best sort of correct
14:22.54seanbrightsorry, cou ldn't resist
14:23.04Reinhildethe terms are synonymous and, sadly, have identical meaning
14:23.09SamotYes, carriers and ITSP's will have g.711 and g.729 because those are standard PSTN codecs.
14:24.02Reinhilde... Last I checked, 729 was not a standard DPSTN codec. I could be wrong, but if I'm not mistaken, only the mu and alpha laws of 711 are truly standard.
14:24.09seanbrighti guess i just don't understand the "old codec" comment
14:24.09Reinhildeand both CE and AD need to be thrown into the memory hole
14:24.16Reinhildeseanbright: me neither
14:24.20SamotIt's simple..
14:24.54Samot10-15 years ago when you had a shitty 1.5 DSL and that was the best you could get...g729 was need for those trying to get the most out of their SIP trunk as a solution to their PRI needs.
14:24.58SamotYou needed g729.
14:25.11Reinhildeif my ITSP offers me Speex8, and I have a bandwidth concern that means I should use that instead of the mu and alpha laws of 711, I will do exactly that.
14:25.16SamotNow, the average bandwidth connection is 50x that's.
14:25.19SamotNow, the average bandwidth connection is 50x that.
14:25.50ReinhildeSamot: This isn't Skype, Discord or Telegram. Up arrow does not edit the previous message. It duplicates the previous message and edits the duplicate.
14:26.00SamotI know.
14:26.03SamotJFC. I know.
14:26.06SamotThanks for pointing that out.
14:26.11ReinhildeJFC this entire conversation.
14:26.17SamotYou started it.
14:26.23ReinhildeA clusterfuck and a shitshow if I've ever seen one.
14:26.28seanbrightthis level of pedantry is so off-putting
14:26.37davlefouseanbright, tkx, i have finish to find! To buy is in the botton of the page!!!
14:26.53seanbrightdavlefou: yes indeed
14:26.56SamotAsking why someone is still choosing g729 as their primary codec these days isn't that bad of a question.
14:27.05SamotHell there actually could be valid reasons.
14:27.11ReinhildeSamot: Not everyone lives in South Choseon.
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14:29.12ReinhildeSome people who have out-of-LAN subscribers still have subscribers who are on lines that can't deliver a consistent 100kbits/s symmetrical (that's the amount of data you need for 711 mu/alpha, as well as for FM-quality Opus)
14:32.57SamotOK so the argument on the table now is that people with VoIP don't have providers that can't have 80Kbps consistently for a call over g711? I'm sorry but if your Internet connection can't handle 80Kbps for a single call, then you need better Internet because nothing is going to work right.
14:34.20Reinhilde"You need better internet." So sayeth the privileged man of the urban West.
14:34.56SamotSo every place out side of North America has inferior infrastructure?
14:34.59SamotThe EU?
14:35.15SamotYou know the EU where they are making POTS go away.
14:35.19ReinhildeApparently the EU is not part of the Occident anymore.
14:36.00ReinhildeAnd by the way I think making the APSTN go away is a terrible decision that will lessen resilience against things like, oh I don't know, NUCLEAR FUCKING WAR?
14:36.19SamotNo one said the PSTN was going away.
14:36.29ReinhildeYou just did.
14:36.30SamotI said the EU is getting rid of POTS.
14:36.36ReinhildeAnd I said APSTN, not PSTN.
14:36.44Reinhilde06:35:15 < Samot> You know the EU where they are making POTS go away.
14:36.45SamotPOTS = Plain Ole Telephone Service i.e. copper wires.
14:37.02SamotI don't know what an APSTN is, sorry.
14:37.07Reinhildeanalogue PSTN
14:37.15SamotThe PSTN is the PSTN.
14:37.16Reinhildecontract with DPSTN
14:37.30SamotPublic Switched Telephone Network. Nothing states the method in which that is done.
14:37.43SamotSIP has been the backbone of the PSTN for some time now.
14:37.56ReinhildeAgain, irresponsible decision that will lessen resilience against things like nuclear war.
14:38.44Reinhildeseanbright: I hope you aren't on a keto diet, because you are going to need roughly 20 gallons of popcorn.
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14:39.39SamotI like how a copper base PSTN will survive a nuclear war vs SIP.
14:39.58SamotSince you know, infrastructure.
14:40.13Reinhildeyes, the infrastructure would be melted either way
14:40.51Reinhildebut I get the sense that an earlier type of PSTN, fully analogue in nature, would be a metric shitton easier to rebuild, after all of the hard disks have been fried with EMPs.
14:41.04SamotAhh.
14:41.16ReinhildeSamot: Would you say that my estimate of how much popcorn seanbright needs is accurate?
14:41.17SamotYou've clearly never worked directly in the infrastructure part.
14:42.19seanbrightnever underestimate the simplicity of /ignore
14:42.24seanbrightgame changer
14:42.48Reinhildeseanbright: That's a rather polite way to say something completely unbecoming of a channel operator.
14:43.10Samotseanbright: I just like to say, Reinhilde has /ignore'd me a few times in the past but here we are with them commenting on things I say out of the blue.
14:43.25ReinhildeI wish my terminal supported upside down text.
14:43.57ReinhildeIt would be quite a ways more convenient either than /ignore'ing people or seeing their text unmodified.
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15:23.01drmessano🍿
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15:23.17igcewielingis that an ice cream cone?
15:24.03drmessanoThe PSTN is neither public nor is it switched
15:26.41drmessanoPrivately-Owned, Paid-Access, Routed Telephone Network is more accurate
15:27.19drmessanoand I also don't think "wiped hard drives" is going to be the concern when EMP takes out the POPARTN
15:28.56drmessanoIf EMP hits my routing infrastructure I am not going to be freaking out about the configs
15:29.54SamotAlso if a nuclear strike was to happen, guys like drmessano would take over our crap anyways.
15:30.36SamotBecause emergency services/communication becomes the top priority.
15:32.04drmessanohttps://www.cisa.gov/government-emergency-telecommunications-service-gets
15:33.06drmessanoI keep my GETS/WPS access within reach
15:35.16jkisterif anyone didnt see my t38 issue in history over the weekend, pls let me know.  still confuzzled.
15:36.02SamotIs this the two Asterisk boxes sending calls between each other?
15:36.06jkisteryes
15:36.11jkisterand audio fallback works fine
15:36.19SamotShow the configs
15:36.24jkistersure, which would you like
15:36.26SamotFor the two peers.
15:36.27jkisterill just give all
15:36.34jkisterudptl/rtp/sip ?
15:37.19drmessanoJust pick the least important line in each and pastebin it
15:37.30drmessanoSamot: Accurate ^ ?
15:38.40jkistergrep '^;' *.conf ?
15:44.11igcewielingperhaps just the sip.conf file.
15:45.24SamotThere is a peer on each PBX.
15:45.36SamotThat serves as the trunk, those are the configs I want to see.
15:46.40Samotbvn1 and jk-pbx1 from the looks of the old pastebin.
15:49.17jkisterwww.kister.net/jeremy/tmp/t38
15:50.51jkisteri did flip directmedia on both hosts just for giggles based on my old experience @ https://asterisk-users.digium.narkive.com/MwxT2ibd/t-38-interop-with-metaswitch that Kevin figured
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15:52.17jkisterusing spandsp instead of digium fyi -- used to work fine until i messed something up during upgrades
15:53.58drmessanoWhat did you upgrade?  That is an important starting point
15:55.19jkisters4 from 15.3 to 16.7 - pbx1 from 13.27 to 13.30
15:55.47jkisteri put back old source and its still broke, so i dunno if i messed up some config ?? i did it late at night and didnt back up
15:55.58jkisterand dont remember exactl what i might have changed to make something work
15:56.35SamotOK
15:56.43SamotSo pbx1 is sending a fax to s4?
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16:25.25jkistercarrier
16:25.26jkisteris
16:25.28jkistersending call
16:25.29jkisterto
16:26.13jkisters4, s4 sending to pbx1
16:30.48jkisterof note, if i faxdetect=yes on s4 (just for fun), it cannot negotate t38 from carrier either
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16:38.43igcewielingSince you don't want fax detect, that doesn't matter much
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17:11.46jkisterover weekend i was looking at s4 debug saying "Capabilities: us - (ulaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)" in one of the invites
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17:33.59igcewielingyou should enable a codec
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18:06.56jkisterwell yeah, ulaw is there.
18:07.16jkisteraudio fallback works
18:07.26jkisterthe other debugs show ulaw in peer from s4
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18:59.55deependI've got a remote user connected with VPN and phone keeps becoming unreachable and reboots. It only seems to reboot when he's actually on the phone, and the phone call is working just fine, can hear him just fine etc.
19:00.05deependhere's some relevant logs of this happening: https://bpaste.net/raw/3DPQ
19:00.24deependany ideas what's going wrong here? anything I can do to fix this issue?
19:00.42[TK]D-FenderIf a phone actually reboots then that's entirely the phone's fault
19:01.59deependI think the phone is programmed to reboot when it loses it's registration? could be wrong about that
19:02.17seanbrightwhat kind of phone?
19:02.21deependthe red light on the phone comes on and then after a few it reboots, however sometimes it will regain it's connection and light goes off
19:02.26deependit's an aastra 9143i
19:02.30[TK]D-FenderNo
19:02.36[TK]D-FenderThat is not a normal thing anywhere
19:02.52deependHave tried swapping it out with a known working phone and had same results
19:02.55seanbrightRTT: 1641.293 msec
19:03.02seanbrightseems a touch on the high side
19:03.05deependyes he's quite far away geographically
19:03.09[TK]D-FenderAnd what model was that phone?
19:03.15deependaastra 9143i
19:03.23igcewielingset your qualify for 10000
19:03.26[TK]D-Fenderso both the same...
19:03.39deependyes same phones all around
19:03.51[TK]D-FenderSo ... that model/firmware has issues
19:04.03igcewielingcould it be a duplicate IP issue?
19:04.08deependigcewieling: so that would be a lower time than default?
19:04.09seanbright1641 seconds is a 27 minute RTT... isn't the sun only 8 light minutes away?
19:04.16deependthat's msec seanbright
19:04.22deepend1.641 seconds
19:04.24seanbrightyes it is
19:04.28seanbrighti'll show myself out
19:04.36deependit's monday no worries
19:05.11deependqualify time lower would mean more chances of success before expiration?
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19:16.15igcewielingdeepend: no, much longer.   you don't want your phone to go unreachable
19:18.02deependso in /etc/asterisk/pjsip.aor.conf I see qualify_frequency=60 for each endpoint
19:18.10deependwhich would be 60 seconds, and you're saying bump that way up
19:18.34deependbut I'm running freepbx and this seems to be autogenerated, not sure where I'm supposed to actually change this
19:19.52SamotThis phone is powered by a power supply?
19:19.58[TK]D-FenderQualify changes nothing
19:20.13[TK]D-FenderThat's how often * tries to check up on the phone, something the phone doesn't care about.
19:20.17[TK]D-FenderNEXT
19:20.21deependyes Samot power brick not using PoE
19:21.33igcewielingweird, I thought I was on #asterisk
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19:21.43SamotOK. So is it really just rebooting when the user is on the phone or is it rebooting more than that but it is only noticed due to being on the phone.
19:22.27deependI'm only seeing the `is now Unreachable` and such when the user is actually on the phone. fairly confident it is not rebooting except when on phone
19:23.33deependpjsip conf looks like: https://bpaste.net/raw/B67Q
19:23.43SamotWell as it has been pointed out, Asterisk and its qualifying have nothing to do with the phone rebooting.
19:23.44deependpjsip.aor.conf that is
19:23.54SamotAgain, nothing to do with the phone rebooting.
19:24.30deependI guess my question is really: what is causing this phone to become Unreachable?
19:24.41igcewielingdeepend: wrong question.
19:24.46SamotYes.
19:24.51igcewielingright question: why is the phone rebooting
19:24.51SamotWhy is the phone rebooting?
19:25.05SamotBecause that could be the cause of the unreachable problem.
19:25.27SamotWhen you swapped phones, did you swap the power supply too?
19:25.28deependit become unreachable before rebooting
19:25.36deependsometimes it toggles back/forth multiple times
19:25.45SamotRight because the phone could be dying down and not responding.
19:26.00deependnot sure on power supply, I can ask. the phone does not reboot suddenly, it indicates a problem before rebooting
19:26.02SamotThe phone becomes unresponsive and power cycles.
19:26.03igcewielingdeepend: the phones becomes unreachable BECAUSE IT STOPS RESPONDING.
19:26.21deependbut then why can i still talk with them?
19:26.49deependwe can maintain a conversation while the phone become 'Unreachable' up until the phone actually reboots
19:27.02SamotBecause Asterisk keeps trying.
19:27.12SamotAnd then will fully drop the device location.
19:27.24seanbrightdeepend: does the phone itself maintain any logs?
19:27.33seanbrighti'm not familiar with aastra phones
19:27.38SamotA bad power supply can cause this issue.
19:27.39igcewielingdeepend: it is not impossible for audio to work and sip to stop working.
19:27.39deependI believe it has some limited logs on the web interface
19:27.48seanbrightdeepend: i would start there
19:28.02SamotA bad network cable  or the phone losting its IP...
19:28.12Samotlosing*
19:28.21igcewielingsomething this weird could be a under powered power supply.
19:28.21deependbut the phone still works is the most confusing part
19:28.37igcewielingdeepend: you said the phone is rebooting.  That is not working.
19:29.05deependyes but it works up until that point, ie still connected and audio working, and then after a reboot it works again like normal
19:29.09seanbrightdeepend: different model, but: https://community.freepbx.org/t/aastra-6735i-phone-just-randomly-reboots/24517
19:29.27deependyes I've found that post
19:29.35igcewieling"after a reboot", does that mean the phone is rebooting on its own, or does that mean you are rebooting the phone?
19:29.36deependphone checks config changes at 0300 each morning
19:29.49seanbrightdeepend: ok, i would check the phone logs to see if there is anything interesting there
19:29.55seanbrightgod speed
19:30.02deependok I will take a look
19:31.24[TK]D-Fender<deepend> I guess my question is really: what is causing this phone to become Unreachable? <- quite possibly the same broken firmware that's causing it to reboot entirely.
19:31.56deependit's running the same firmware as all the other phones
19:34.40[TK]D-FenderAnd the same thing as the other phone you tried replacing with it.
19:34.45[TK]D-Fenderit's a fail
19:35.36deependso reachable/unreachable is based on qualify response
19:35.55seanbrightcorrect
19:35.57deependis it only used on dynamic contact?
19:36.02deependor always used?
19:36.09seanbrightso you're not going to check the logs on the phone i guess
19:36.32deependit's quite limited
19:36.36deependcrash report from last october
19:36.49deependI configured the UDP remote logging to forward to my server
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23:31.48FuriousGeorgehey all...  having some issues with execif syntax.  the error is syntax error: syntax error, unexpected '=', expecting $end; Input:
23:32.01FuriousGeorgeshould i have quotes around the F maybe?
23:33.16FuriousGeorgeor is it complaining about the first =, since it mentions Input: 6
23:36.14*** join/#asterisk GoldenBear (~gb@5.253.206.70)
23:37.03*** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-bagmyusihdqxylcs)
23:41.32FuriousGeorgei changed the true side to be a noop and it still complains
23:59.27FuriousGeorgeok, i needed quotes around the variable and the value in the first part

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