IRC log for #asterisk on 20180310

00:02.01*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
00:04.25*** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu)
01:10.57*** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com)
01:14.38*** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net)
01:19.21*** join/#asterisk infobot (ibot@rikers.org)
01:19.21*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
01:27.36*** join/#asterisk degenerate (~degenerat@S0106689e199caaf4.no.shawcable.net)
02:05.07*** join/#asterisk tzafrir (~tzafrir@62-90-199-247.barak.net.il)
02:09.47*** join/#asterisk SunTsu (~miyamoto@unaffiliated/suntsu)
02:30.05degenerateAnyone here use the Bria desktop app?
02:30.43degenerateI've been pulling my hair out trying to get it to play nicely with my microphone.  I can see the microphone is working in system preferences -> sound -> input
02:30.57degeneratebut when i select it from bria, i get nothing. just a volume bar stuck at 50% :(
02:43.35*** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu)
02:45.01*** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co)
03:44.13*** join/#asterisk radicaldev (~kvirc@c-73-76-132-192.hsd1.tx.comcast.net)
03:44.17radicaldevFender, are you around?
03:45.08radicaldev[TK]D-Fender: Ping!
03:50.12radicaldev[TK]D-Fender: If you accept donations, I owe you one.
04:04.52wyoungdegenerate: All I know about is asterisk and snom.
04:27.46*** join/#asterisk Typhon (~Typhon@ipservice-092-218-210-028.092.218.pools.vodafone-ip.de)
04:36.32*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
04:46.44*** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com)
04:49.29*** join/#asterisk Oatmeal (Suzeanne@gateway/vpn/privateinternetaccess/suzeanne)
04:55.15*** join/#asterisk bmg505 (~leon@196-210-23-168.dynamic.isadsl.co.za)
05:10.57*** join/#asterisk jkroon (~jkroon@165.16.204.174)
05:11.07*** join/#asterisk LowJack (~cojack@ip-58.188.freevpn.it)
06:47.06*** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co)
06:57.01*** join/#asterisk areski (~areski@37.223.225.58)
06:57.18*** join/#asterisk ganbold (~ganbold@173.244.215.173)
07:57.27*** join/#asterisk areski (~areski@37.223.225.58)
07:58.03*** join/#asterisk Oatmeal (Suzeanne@gateway/vpn/privateinternetaccess/suzeanne)
08:51.06*** join/#asterisk ih8wndz (jwpierce3@2600:3c00::f03c:91ff:fe15:fddf)
09:31.30*** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca)
09:36.05*** join/#asterisk Worldexe (~Worldexe@95-107-33-134.dsl.orel.ru)
09:59.31*** join/#asterisk miralin (~Thunderbi@91.237.94.7)
10:09.14*** join/#asterisk jkroon (~jkroon@41.13.92.233)
10:42.54*** join/#asterisk Asterisco (~catan@151.54.33.218)
10:42.57Asteriscohi!
10:43.25Asteriscoi need do some anonymous outgoing calls
10:43.38Asteriscoin my country i can do it with #31# prefix
10:43.52Asteriscoin USA it's possible use *67
10:44.08Asterisconow i'm trying do it with asterisk
10:44.18Asteriscois it possible?
10:49.50SamotYes.
10:49.59SamotThose are dialable extensions..
10:50.04SamotThat route
10:50.12SamotAnd do things on the system that accepts them.
10:50.24SamotYou have Asterisk, you can set your CallerID to however you'd like.
10:51.31AsteriscoSamot i use a pstn gateway
10:51.43Asteriscois not possible use a CallerID
10:51.57SamotOK
10:52.05SamotSo you know how your set anonymous calls?
10:52.12SamotYou block transmitting your callerid
10:52.12Asteriscoexten => _3X.,n,Dial(${IAX_EUROPTICAL}/#31#${EXTEN},30)
10:52.19SamotOK.
10:52.25Asteriscoi'm trying this
10:52.33Asteriscobut don't work
10:52.38SamotDoes the provider accept that as a valid code?
10:52.48Asteriscoi don't recive nothing in the server with pstn gateway
10:52.58SamotI don't know what you mean
10:53.05Asteriscothe server is mine
10:53.10SamotDoes your provider that you are dialing out to..
10:53.13Asteriscothe iax server is mine
10:53.25SamotWho is EUROPTICAL?
10:53.40SamotWhere does that Dial() send the call?
10:53.48SamotTo a local device on Asterisk or to your PSTN provider?
10:54.27Asteriscoi call from home to my office (EUROPTICAL)
10:54.36Asteriscoin europtical there is a pstn gateway
10:54.49Asteriscoi need do a anonymous call using #31# prefix....
10:55.09Asterisco...from pstn gateway that is in my office (EUROPTICAL)
10:55.51Asteriscoin [fromiax] i need put this: exten => _#31#3X.,n,Set(chiama=${EXTEN}) ??
10:56.21SamotOK dude.
10:56.26SamotIt's Asterisk.
10:56.38SamotYou need to setup your own Feature Codes on it for them to do something.
10:56.49SamotYou need to make #31# do something
10:57.07Asteriscoon Europtical server?
10:57.12SamotIn order to have an anonymous calls you need to set your OUTBOUND CallerID to private.
10:57.15SamotYes.
10:57.23SamotIf that is the server processing the call, yes.
10:57.43SamotFeature (star) codes are things the PROVIDER programs.
10:57.51SamotNot all providers have the same feature codes these days
10:58.12Asteriscoi live in italy
10:58.13SamotThe old POTS codes from the 80's/90's are some what obsolete.
10:58.17Asteriscomy provider is tim
10:58.17SamotOK
10:58.27Samoti don't know what providers in Italy do
10:58.29Asteriscoi can do anonymous call with prefix #31#
10:58.35Samotor what feature codes they use.
10:58.35SamotOK
10:58.55SamotSo that's THEIR system going "Oh it starts with #31#, hide the callerid"
10:59.03SamotSo now you need to make Asterisk do the same thing.
10:59.19Asteriscoin europtical extensions ...
10:59.27Asteriscoin outgoing i put this: exten => _#31#3X.,1,Dial(SIP/991${EXTEN}@9001) ;
10:59.29SamotThese aren't global features that just work
10:59.39SamotDude
10:59.46SamotYou have to set the CALLERID TO PRIVATE
10:59.55SamotJust doing a Dial() isn't ENOUGH
11:00.03Asterisco?
11:00.13SamotYou have to set CALLERID(name) and CALLERID(num) to private settings.
11:00.27Asteriscosamot... i use a patton gateway
11:00.32SamotOK
11:00.36SamotThen look how it's done.
11:00.41Asterisco?!?!?!?!
11:00.46SamotYou can't ask how to set callerid to private..
11:00.55Asteriscois not possible start a call with #31# ^^
11:00.56SamotThen say "I can't set callerid I have a patton gateway"
11:00.56Asterisco?
11:00.59SamotFFS.
11:01.04SamotWhat part are you not listening?
11:01.13Samot#31# are JUST DIGITS
11:01.36Asteriscowhy my server don't recive the calls ???
11:01.41SamotThe voice service processing your outbound call, in this case Asterisk, needs to know what to do when they accept those digits.
11:02.07Asteriscoright...
11:02.19Asteriscoin my home server i put:
11:02.39Asteriscoexten => _3X.,n,Dial(${IAX_EUROPTICAL}/#31#${EXTEN},30)
11:02.47SamotAGain
11:02.59Asteriscoso when i call a number that start with 3...
11:03.07SamotWho is "IAX_EUROPITCAL"?
11:03.11Asterisco...i pass the call to iax_europtical
11:03.15SamotOK.
11:03.20SamotSo stop.
11:03.22SamotRight there..
11:03.24AsteriscoIAX_EUROPTICAL is my server in my office!
11:03.45SamotAll you're doing is sending #31#3XXXXXXXXX
11:03.49SamotThat's all you're doing
11:03.57SamotThe callerid is not being set to private.
11:04.14Asteriscoi'm dooing a call with #31# prefix!
11:04.18SamotYou have no dialplan, that you've shown, that sets the CallerID
11:04.21Asterisco<PROTECTED>
11:04.21SamotSO WHAT?
11:04.26Asteriscoi put this extension:
11:04.27Asteriscoexten => _#31#3X.,1,Dial(SIP/991${EXTEN}@9001)
11:04.37SamotWhat part of #31# is a PROVIDER SETTING?
11:04.42SamotIs hard to follow?
11:04.50SamotIn this case YOU ARE THE PROVIDER
11:04.57AsteriscoYES i'm!
11:05.15SamotYou're calling from one Asterisk Box to another Asterisk Box
11:05.25AsteriscoYES
11:05.32SamotSo the ASterisk box that makes the call, that's the box that needs to handle the #31#
11:05.36SamotDo you understand?
11:05.42Samot#31# is not magically
11:05.46Samotgoing to work
11:05.52SamotYOU HAVE TO PROCESS THE DIGITS
11:05.56SamotAND DO STUFF
11:06.04Asteriscoi need check that is a call that i need "anonimize"
11:06.11SamotJust like the Telco/Provider's do
11:06.17SamotRIGHT
11:06.25SamotAnd that's what #31# is for.
11:06.29Asteriscoi use #31#
11:06.32SamotIt tells you that you need to do shit.
11:06.35SamotOK
11:06.41Asteriscothat  is the same that use TIM (my provider)
11:06.46SamotSo make your dialplan to match on that
11:06.49SamotOK
11:06.54SamotBut TIM actually is doing stuff.
11:06.57SamotYou're not
11:07.01SamotYou're missing entire steps.
11:07.01Asterisco???
11:07.11SamotOK, I'm going to try this one more time.
11:07.16SamotListen closely.
11:07.23Asteriscoi try
11:07.29SamotWhen you send a call to TIM as #31#34567890
11:07.36SamotTIM looks at the call..
11:07.47SamotSEES #31# as the first FOUR DIGITS
11:08.01SamotAnd the system then goes "I need to set the CallerID to Private for this call"
11:08.08SamotIt sets your CallerID to private
11:08.14Asteriscook
11:08.15SamotThen it dials out to 34567890
11:08.25Asteriscook
11:08.41SamotSince you are only sending the call to your ASTERISK box from your phone...
11:08.47SamotThat calls your OTHER ASterisk box..
11:09.01SamotThen you need to make Asterisk do the same thing...
11:09.15AsteriscoSamot ...
11:09.16SamotMatch on calls that start with #31# and then set the CallerID values to PRIVATE
11:09.18Asteriscoone moment
11:09.51Asteriscoin my home i need add anonymize all cellular phone
11:10.05Asteriscoat my home i call number 34567890
11:10.20Asteriscoi need call it from my office server
11:10.26SamotOK
11:10.28Asteriscoso...
11:10.33SamotSo
11:10.34SamotFFS.
11:10.36SamotAgain..
11:10.45Asteriscoi add the prefix #313# and past to my office server the call
11:10.51Asterisco#31#
11:10.56SamotWhen you dial #31#34567890 from the phone connected to your Asterisk box....
11:11.06SamotThat is being sent to Asterisk..
11:11.13AsteriscoSamot
11:11.18SamotWhat?
11:11.20AsteriscoI NEED call 34567890
11:11.34SamotI have no idea what you are talking about
11:11.35Asteriscoand automaticaly add #31# prefix
11:11.42SamotOK.
11:11.44Asteriscoand past to my office server
11:11.45SamotSo then do that in Asterisk.
11:11.47SamotOK
11:12.10Asteriscomy office server need do this cal to privider by the patton gateway
11:12.11SamotYou need to do this shit on your Asterisk box that is MAKING THE CALL
11:12.39Asteriscomy asterisk box is in my home right?
11:13.08Asteriscoin [outgoing] extensions of my home box
11:13.11Asteriscoi put this:
11:13.12Asteriscoexten => _3X.,n,Dial(${IAX_EUROPTICAL}/#31#${EXTEN},30)
11:13.40Asteriscoi call 3456789 and my home box add #31# and need past it to my office server
11:13.42SamotOK
11:13.45Asteriscoso where is the error?
11:13.47SamotThat sends the call to your Office right?
11:13.52AsteriscoNO
11:13.55SamotOK
11:14.05Asteriscoin my office i don't recive anything
11:14.14SamotFFS.
11:14.19SamotLet me try to ask this again
11:14.25SamotYou need to give a clear answer...
11:14.41SamotWHO IS the destination IAX_EUROPTICAL?
11:14.53SamotIs that your Office PBX?
11:14.59Asteriscoi repeat:
11:16.53AsteriscoI (call number 34567890) -> (home asterisk box that add #31# prefix and use iax_europtical) ->Iax_europtical recive the call #31#3456789 and send it to my patton gateway --> that resend the call to my provider TIM and do the call #31#3456789
11:17.18Asteriscoi think that is linear my explain
11:17.20Asteriscoor not?
11:17.45SamotOK
11:17.48SamotSo what is the issue then?
11:18.00SamotWhen you say it's not working, what is actually not working?
11:18.51Asteriscoin my home when i do a call add #31# BUT don't resend the call to my OFFICE BOX
11:19.15Asterisconow can you see my configuration with me?
11:19.38Asteriscoin my HOME configuration ... in outgoing i put this: exten => _3X.,n,Dial(${IAX_EUROPTICAL}/#31#${EXTEN},30)
11:19.38SamotSo you're sending the Dial() to your Office PBX?
11:19.48SamotBut the office PBX never gets the call?
11:19.55Asteriscoexactyly
11:19.58SamotOK
11:20.07SamotSo you need to show a debug of you attempting the call
11:20.14SamotI don't know why it's not working..
11:20.23SamotYou need to show an actual call attempt
11:20.29SamotWhat do the logs say?
11:20.31Asteriscoi think that is a problem of hastag char
11:20.35Asterisconothing
11:20.36SamotNo, it's not.
11:20.38Samot# is fine
11:20.47SamotSo your home Asterisk never sees the call?
11:20.56Asteriscoone moment
11:21.13Samotasterisk -rvvvvvvvvvvvvv
11:21.15Samotmake call
11:21.17SamotShow output
11:24.31Asterisco[Mar 10 12:23:46] WARNING[10173][C-00000001]: chan_sip.c:33735 process_crypto: Ignoring crypto attribute in SDP because RTP transport is insecure
11:24.55SamotNope.
11:24.58SamotShow a whole call
11:25.06SamotBecause I'm not guessing line by line
11:25.20Asteriscook
11:27.11Asteriscohttps://nopaste.xyz/?f14a431cfad2b0c8#XRnZrY83NKK8G+oW+nA71abCt/q4DUUsRW9VYfgoHP4=
11:27.43Asteriscothis is my HOME asterisk log
11:28.10Samot<PROTECTED>
11:28.10Samot<PROTECTED>
11:28.11Samot<PROTECTED>
11:28.22SamotSo now you need an Office PBX log
11:28.30SamotBecause your Office PBX is denying the call.
11:28.51Asteriscowith asterisk -rvvvvvvv i don't see nothing
11:29.02SamotOn the OfficePBX?
11:29.06Asteriscoyes
11:29.10SamotDon't know
11:29.23SamotBut it's not completing some how..
11:29.30SamotYour Home PBX is dialing out.
11:29.36Asteriscoi know
11:29.53Asteriscohow can i do a complete debug?
11:29.55SamotWell if the Home PBX shows the call being dialed..
11:30.03SamotAnd the Office PBX doesn't show anything...don't know
11:30.08Asteriscoso i try to see if my office box recive something
11:30.11Samotasterisk -rvvvvvvvvvvvvvvv
11:30.18Samotiax set debug on
11:31.03Asteriscoiax2 set debug on
11:31.20SamotOK
11:31.35SamotI don't use IAX
11:35.27AsteriscoTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REJECT
11:35.27Asterisco<PROTECTED>
11:35.27Asterisco<PROTECTED>
11:35.40Asteriscoi think that we have found the problem
11:35.41Asterisco:\
11:37.27Asteriscobut is strange
11:37.44Asteriscobecouse if i don't use #31# it works perfectly
11:37.48SamotRight
11:37.59SamotBecause you're incoming context is setup for that
11:38.16SamotDid you set up a match in your incoming context with #31# to accept calls?
11:38.53Samot6:06:46 AM <Samot> So make your dialplan to match on that
11:39.07SamotThat whole part of "Make Asterisk match"
11:39.15wyoungSamot!
11:40.00Asterisco<PROTECTED>
11:40.35SamotSigh.
11:40.36SamotYes.
11:40.44SamotThe box that is rejecting the call for not having a match
11:41.03Asteriscohow can i do it?
11:41.06kodomoHi folks! Is there anybody who has TLS running under * 13 with pjsip 2.7.1 ? I've got something that looks like a bug to me, but that I was hoping might be a misconfiguration, because I'm just migrating from chan_sip to pjsip and might be missing a mandatory configuration item.
11:41.16Asteriscoi don't put nothing in incoming context! :)
11:42.22Asteriscoexten => _#31#3X,1,Answer()
11:42.23Asteriscoexten => _#31#3X.,1,Dial(SIP/991${EXTEN}@9001)
11:42.31Asteriscoso it's right?
11:42.50kodomoSo I'd need either an example of a truly working tunnel configuration (+ info whether [endpoint] sections are relevant to the matter), or I'll need to open a bug report (forum post was to no avail, so far...)
11:44.32Samot6:42:23 AM <Asterisco> exten => _#31#3X,1,Answer() <-- Fix your match
11:44.46Samot_#31#3X is not going to match
11:44.50SamotYou're missing the .
11:45.43Asteriscoexten => _#31#3X.,1,Answer()
11:45.44Asteriscoexten => _#31#3X.,1,Dial(SIP/991${EXTEN}@9001)
11:45.46Asteriscoright?
11:47.16Samotyes
11:47.26SamotWell no
11:47.37SamotThey both can't be priority 1
11:47.52Asteriscooh :)
11:47.59Asteriscoexten => _#31#3X.,1,Answer()
11:47.59Asteriscoexten => _#31#3X.,n,Dial(SIP/991${EXTEN}@9001)
11:49.10AsteriscoNOTHING TO DO
11:49.30AsteriscoCAUSE           : No such context/extension
11:49.31Asterisco<PROTECTED>
11:50.14SamotIs it in the proper context?
11:55.24Asteriscoohmygod! :D
11:55.32Asteriscothe context was fromiax
11:55.36Asteriscoand not incoming! :)
11:55.57Asteriscoi'm relly stupid
11:58.57SamotMight explain why things worked when you said [incoming] was empty
11:59.29SamotBecause that's not what is being used.
12:01.39kodomoNo one here, who uses TLS w/ pjsip under * 13?
12:06.25*** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com)
12:07.55*** join/#asterisk nede (~nede@194.74.31.178)
12:08.14SamotWell..
12:08.20SamotYou mentioned it may be a bug
12:08.34SamotAnd that you either need an example of something or need to open a bug report..
12:08.56SamotHowever, you have shown nothing to support either of those statements.
12:09.20kodomohttps://community.asterisk.org/t/pjsip-tls-trouble/73813
12:09.35kodomoTell me what you'd like to be shown :)
12:11.34kodomoIf I had a configuration that is known to work I could check my config against it. I'm still hoping I might just not be savvy enough to set pjsip up correctly
12:11.36SamotYour configs
12:11.51SamotShow your tls transport settings
12:11.53kodomoThat's the reason I hesitate filing a bug report for now
12:12.05kodomoSamot: It's in the posting
12:12.25kodomo"My [transport] reads"
12:13.12SamotWhat is in the PEM?
12:13.22kodomokey and certificate
12:13.23SamotWhat files are in the PEM?
12:13.29SamotIt is a public cert?
12:13.31kodomogenerated by * 1.8
12:13.36Samot.....
12:13.42SamotIs it a public cert?
12:14.30nedeHi, does anyone know if pjsip or cel_odbc has a column character limit logging cel events to MySQL in 14.0? I'm having trouble with the extra column (JSON) being truncated to 100 characters for ATTENDEDTRANSFER/BLINDTRANSFER. It's not a limit set by the column definition, it seems to be asterisk itself.
12:14.52kodomoSamot: I'm unsure what you refer to as 'public' cert.
12:15.09SamotDoes it have a public CA
12:15.14SamotComodo, Thawte, etc
12:15.20SamotOr did y ou generate it yourself?
12:15.28kodomoSamot: It's the key and cert my chan_sip configuration uses for successful sips /srtp connections
12:15.44kodomoSamot: no - not signed by any other authority
12:15.45SamotDid you generate the crt file yourself
12:15.55Samotor did you give it to a company to generate it?
12:15.58kodomoSamot: I did - with the tools * 1.8 provided
12:15.59SamotOK
12:16.02SamotSo it's a private cert.
12:16.08kodomoyes
12:16.12SamotRegenerate it.
12:16.16SamotSo it's current.
12:17.55kodomoSamot: Cando, will take a while to look up the procedures... I generated it years ago...
12:18.29kodomoPlease mind that the cert is not expired (working in chan_sip) and that the error message is not concerned about the cert, but the cipher suites
12:19.05kodomoThe error message is "no shared cipher suites", although pjsip tells me it supports the configured suite and it is being offered by the client
12:19.56kodomoDo you suspect that the error message may be not specific (as in 'the root cause lies elsewhere')?
12:20.09kodomo...regenerating cert anyway...
12:21.44kodomo.oO(thank you for answering, anyhow, btw)
12:24.02*** join/#asterisk ih8wndz (jwpierce3@2600:3c00::f03c:91ff:fe15:fddf)
12:33.57SamotAsterisco: Please stop PMing me.
12:34.03SamotI don't read them.
12:34.07kodomo?
12:34.10kodomoah
12:34.31SamotSTOP PMING ME
12:34.58Asteriscooh excuse me
12:35.23AsteriscoSamot.. in my office asterisk box...
12:35.30Asteriscoin [fromiax] context
12:35.38Asteriscoi've put this:
12:35.48Asteriscoexten => _#31#3X.,1,Dial(SIP/991${EXTEN}@9001)
12:36.07Asteriscobut when i try to call from my home a cellular line i recive this:
12:36.48Asterisco<PROTECTED>
12:37.09Asterisco[2018-03-10 13:31:13] WARNING[6377][C-00000002]: chan_sip.c:23875 handle_response_invite: Received response: "Forbidden" from '"15 Giuseppe snom" <sip:9001@10.0.0.1>;tag=as043d3669'
12:37.09Asterisco<PROTECTED>
12:42.59Asteriscocould be the patton that not accept #31# ?
13:08.31kodomoSamot: regenerated keys and self-signed certs as lined out in https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
13:09.00kodomoSamot: as suspected - unfortunately there's no change in the situation (chan_sip works, pjsip does not)
13:09.12kodomosame issue 'no shared cipher'
13:11.03SamotDid you install the cert on the device trying to use it?
13:11.10SamotIt's a private signed cert.
13:11.17SamotThe device needs the cert to use it.
13:11.23SamotIt just can't download it.
13:11.26kodomoSamot: yes
13:11.34SamotYou need to show an actual call
13:11.39Samotasterisk -rvvvvvvvv
13:11.46Samotpjsip set logger on
13:12.24kodomoSamot: with chan_sip? It's working... with pjsip there's only exactly one message output, as no SIP dialogue is started.
13:12.31kodomoIt's the one I put into my post
13:12.43kodomoWARNING[9241]: pjproject:0 <?>: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> len: 0
13:13.01SamotThat's not what I asked for.
13:13.07SamotYou need to show an actual call
13:13.10Samotasterisk -rvvvvvvvv
13:13.12Samotpjsip set logger on
13:13.25Samot^^ Do that now. Pastebin the results.
13:16.42kodomoSamot: before I do that - two caveats: You will get a lot of clutter of devices that are _not_ configured with the correct client certificate, as I only generated one for the device I'm debugging with (it's a lot of effort to roll back, otherwise and the devices are not all local to where I am)
13:16.54SamotOK
13:17.09SamotUntil I can see what the device is sending
13:17.16SamotI can't tell you anything
13:17.30SamotI need to see the transaction.
13:17.45SamotOne line of the log isn't enough for this type of issue.
13:18.18kodomoSamot: You won't be able to discern which message is coming from the device I'm currently debugging with as there is no SIP dialoge (as it's the TLS handshake failing with abovementioned message - which btw. is trunkated in my post - it reads "err: <336109761> <SSL routines-ssl3_get_client_hello-no shared cipher> len: 0"
13:18.48SamotOK then I can't help.
13:18.51SamotI can't see debugs
13:18.55SamotThen I can't see what the issue is.
13:18.56kodomoSamot: that being said: how do I do pjsip logger on without a console  (-cvvvvvvvvvv)?
13:19.05SamotYou can't
13:19.05kodomoSamot: that being said: how do I do pjsip logger on without a console  (-cvvvvvvvvvv doesn't start a console)?
13:19.09SamotIt's a console command
13:19.24SamotEither follow the troubleshooting steps I provided or not.
13:19.56kodomodarn - I'm distracted by local events... sorry - again: how do I do pjsip logger on with -rvvvvvvv (no console)
13:20.08SamotYou can't.
13:20.17SamotI'm not wasting time talking in circles.
13:20.24SamotEither follow the steps and provide what I asked for
13:20.28SamotOr don't.
13:20.28kodomoSamot: then I didn't get your instructions
13:20.34SamotYes you did.
13:20.36kodomoSamot: they need to be clear to follow
13:20.40SamotReally?
13:20.55Samot8:13:10 AM <Samot> asterisk -rvvvvvvvv
13:20.55Samot8:13:12 AM <Samot> pjsip set logger on
13:20.59SamotMake a damn call.
13:21.04SamotPastebin the results.
13:21.15kodomoSamot: so - I do asterisk -rvvvvvvvv and redirect the output to a file to subsequently paste in pastebin
13:21.25SamotI have now spent almost 3 hours in here with two different issues with people fighting me over shit.
13:21.49SamotYou copy and paste it from the scrollback buffer.
13:21.50kodomoSamot: appreciated - but I'm not fighting you and not shitting you
13:22.04SamotRun the debug
13:22.11SamotCopy and paste the results to paste bin
13:22.16SamotPost the pastebin link
13:22.19SamotIt's not hard.
13:26.17kodomoSamot: ok - local interrupt handled - concentrated again... pastebin: https://pastebin.com/VXUdGPNT
13:26.29kodomo1249 dialogue not relevant (unencrypted phone)
13:26.48kodomothe one line from the client in question is the last one - when registering
13:28.04kodomothe phone does _not_ register, as the TLS connection doesn't work, as I'm getting this very single one line I mentioned (which doesn't make sense when I look at the pcap trace!)
13:28.35kodomoThat's why I'm trying to confirm that _anybody_ managed TLS to get running with the versions I'm running
13:29.08kodomoIf so, I was hoping to get a glimpse at their configuration to see whether I was missing something... there is no usable debug output on behalf of asterisk
13:29.23SamotOK.
13:29.25kodomoand the one line I get is refuted by my pcap trace of the dialogue
13:29.29SamotSo I asked for something.
13:29.32SamotIt was a debate
13:29.38kodomoAnd I provided
13:29.41SamotAnd what I finally get isn't even close to what I asked for
13:29.46SamotIt has no information
13:29.58SamotWhich means the device didn't send jack to the PBX
13:30.02SamotOr the PBX never got it.
13:30.05kodomoSamot: tell me how to get more info, then - I did exactly what you wanted
13:30.12kodomoThe device is not registering
13:30.15SamotOK.
13:30.22SamotShow it trying to register
13:30.26SamotMake it register.
13:30.28SamotOr try to
13:31.23kodomoSamot: That's what you're seeing - if there is supposed to be more output, I'd need what to configure to make asterisk even more verbose (getting only this one nonsensical line is what is exactly what is unnerving me)
13:31.49kodomoI can show you the pcap trace
13:32.27kodomothere you see SYN - SYN ACK - ACK - TLSv1 Client Hello  and asterisks TLS answer
13:32.45kodomoand then of course a TCP connection shutdown
13:33.03SamotShow this pcap
13:33.06kodomono SIP message is ever sent because the TLS connection doesn't get established
13:33.09kodomook - sec...
13:33.12SamotUhm.
13:33.15SamotThat's not how it works.
13:33.16SamotAT ALL
13:33.28SamotIn order for Asterisk to reject the message it first must see it
13:33.34SamotWith means a SIP packet hit it.
13:34.01SamotThe pjsip logger would have outputted the packet.
13:34.16kodomoSamot: it's not rejecting the device registration, * is refusing to communicate with the client because (* claims) there is no shared crypto cipher.
13:34.37SamotShow the pcap.
13:34.55kodomosec - will find a way to poste it someplace...
13:34.56SamotAlso, I will say this again..
13:35.05SamotIn order for Asterisk to do something with your request..
13:35.08SamotIt has to REACH ASTERISK
13:35.14SamotWhich means the SIP packet is there.
13:35.18SamotAsterisk is PROCESSING IT
13:35.22fileSamot: nope.
13:35.28SamotOh so that's changed?
13:35.47filefor connection based protocols a connection has to be established in order for traffic to flow
13:35.53SamotThe only time I never saw a packet is when openssl had issue with the cert
13:36.12fileTLS negotiation is more than just certificate, it can fail for different reasons
13:36.18SamotI know.
13:36.30SamotI've been asking for some proof for almost two hours.
13:36.45SamotJust like that IAX issue.
13:36.53SamotI spent more time talking about it then seeing actual proof.
13:37.02SamotThat could be actually worked with.
13:38.33filekodomo: have you also configured a priv_key_file?
13:39.31kodomohttp://kronos.ran-dom.org:2080/1244_pjsip.trace
13:40.14kodomofile: no, as with chan_sip * was happy with the .pem containing both, I assumed pjsip would be content with it as well
13:40.40kodomofile: but the error message is not about the certificate - it's specifically about a lack of overlap in crypto ciphers
13:40.59fileI understand that but the wiki has a priv_key_file explicitly mentioned and I'm not sure internally in PJSIP whether its TLS transport is that nice
13:41.10fileso trying to match the wiki one as close as possible is best
13:41.15fileor rather, explicitly configured
13:41.33kodomofile: ack, will configure the transport accordingly - give me a sec.
13:41.42SamotWhich he said he did when he regenerated the cert.
13:41.51SamotSo you didn't follow the steps that you said you followed?
13:42.20Samot8:08:32 AM <kodomo> Samot: regenerated keys and self-signed certs as lined out in https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
13:42.36kodomoSamot: I regenerated everything, I configured it the way it worked w/ chan_sip - could we please reduce the adrenaline and outrage? I'm not trying to waste your time, but am cooperating
13:42.43SamotWait, what?!
13:42.56SamotYou configured like chan_sip?!
13:43.00SamotThey are not the same.
13:43.03kodomoSamot: mind the tutoial is from * 1.8 (no pjsip) and chan_sip didn't need the explicit key config item
13:43.08SamotFollow the PJSIP
13:43.12SamotDude.
13:43.16SamotYou just said you used the WIKI
13:43.23SamotWhich has ALL THE INSTRUCTIONS
13:43.42kodomoSamot: there's a wiki on cert creation (* 1.8) and one on pjsip - naturally, they don't align...
13:43.43SamotPJSIP and Chan_SIP are two sets of different instructions on that page.
13:43.49SamotNo.
13:43.51kodomoeither way... proceeding to configure
13:43.53SamotBecause they aren't the same
13:44.01SamotJust like configuring endpoints.
13:44.02SamotNot the same
13:47.15*** join/#asterisk SoBlindWolf (~SoBlindWo@go.pcshost.co)
13:48.19kodomofile: set crt and key individually and something changed: I'm still unable to register, but I'm not getting the error message about the cipher anymore
13:48.43kodomofile: the error message seems indeed to have been misleading and unspecific
13:49.02filedid you also remove the explicit cipher list, or did you keep it?
13:49.03kodomofile: however, I'm not getting any output anymore... need to prepare another pcap trace to see what's going on
13:49.12kodomofile: kept it
13:49.23fileremove it, start from the most basic config first
13:49.45SamotYeah.
13:49.53SamotDon't get fancy until the basics work.
13:49.57kodomofile: shall I remove the method as well?
13:50.02SamotNo.
13:50.04filemethod is fine
13:50.05Samotmethod is needed.
13:50.12SamotOr should be there..
13:50.21SamotIt's set to tlsv1 so that should be fine.
13:51.13SamotIn fact, someone had a handshake issue earlier this week with PJSIP/TLS
13:51.25kodomostill no message - will prepare a pcap trace and look at it...
13:51.25SamotAnd they were missing the method= setting in the transport.
13:51.37SamotWhen they added it, it fixed the handshake issue.
13:51.40kodomoSamot: that was me, in fact
13:51.43SamotOK
13:51.47Samotyou said it fixed your issue?!
13:52.05kodomoSamot: that's how long I've been at my pjsip migration (and it's a measure of my personal frustration with it ;) )
13:52.15SamotOK
13:52.25SamotBut we fixed this handshake issue earlier this week
13:52.34SamotSo what did you do between then and now to make it come back?
13:52.44SamotAlso, you didn't have everything in one PEM
13:52.49kodomoSamot: It changed the situation - which was I was content with at the moment... afterwards I had the cipher suite message which gave me the impression I knew what to look for.
13:53.06SamotOh yeah, it was also a PUBLIC cert.
13:53.11SamotSo.
13:53.18SamotYou changed the cert from public to private
13:53.28SamotYou removed the key setting and put all three in the pem
13:53.36kodomoSamot: I always and every only had a self-signed cert
13:53.39SamotThen went and did chan_sip stuff...
13:53.40SamotOK
13:53.44SamotSo how did we fix this
13:53.52SamotAnd now, like 5 days later it is broken again?
13:53.58SamotWhat did you do in the last five days?
13:54.04kodomoSamot: it was never working for me.
13:54.08SamotSigh.
13:54.27*** join/#asterisk qxork (~qxork@unaffiliated/qxork)
13:54.30kodomoSamot: but after inserting the method I had an error message I understood the meaning of
13:54.48kodomoSamot: only that it turned out to be misleading, which is why I reappeared
13:55.06kodomoSamot: I _do_ first attempt to figure out stuff by myself before asking for help ;)
13:55.18SamotWell..
13:55.29SamotYour transport config is now different than earlier this week
13:55.49SamotYou didn't have key, ca and crt in one pem
13:55.55kodomoSamot: some people also say that I occasionally _do_ learn ;)
13:55.57SamotYou had key and then the ca and crt in a pem.
13:56.12SamotSo let me ask you this..
13:56.21SamotDid you recall  me helping you earlier this week?
13:56.33SamotBecause, I didn't put your nick to that issue....
13:56.37SamotBut if you did..
13:56.40SamotAnd knew I helped you
13:56.59SamotMentioning that this is the thing we did earlier this week would have been more helpful.
13:57.01*** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.67)
13:58.31kodomofile: heh - I know why I don't receive a message anymore... the tls transport is no longer created and the port is closed...
13:58.46SamotOh JFC.
13:58.47kodomofile: sec... will cross check my current transport def against the wiki
13:59.03filecheck the console output at start, it'll likely tell you why it couldn't be created
14:00.34*** join/#asterisk znoteer_ (~znoteer_@69-196-143-108.dsl.teksavvy.com)
14:03.18kodomofile: fixed... and "hooray" - I'm registered... attempting call...
14:03.41fileyou probably want rewrite_contact=yes if you haven't enabled it on the endpoint
14:04.57kodomofile: ok... attempting a call I get an 488 - unacceptable error... but I guess that's something else...
14:05.04fileit is.
14:05.22kodomo...actually - sure... my extensions haven't been updated... they're trying to call sip channels (chan_sip)
14:05.35filecodec incompatibility, or trying to do SRTP without having it configured
14:05.57kodomofile: I'd guess it's the extensions table
14:06.08kodomo*checking the rewrite_contact parameter*
14:07.06kodomofile: I'd recommend changing the error message that I got previously - it's really misleading
14:07.12kodomoshall I create a bug report?
14:07.20fileit's a message from PJSIP itself, likely raised from OpenSSL
14:07.35fileOpenSSL tends to not... be helpful... in error messages it tells you
14:07.56kodomo(and it may be worthwile to point out explicitely the need to split key and cert in the chan_sip->pjsip migration part of the wiki
14:07.59kodomo)
14:08.19kodomoappreciated :-|
14:08.21fileyou can file an issue if you want, it'll just get triaged, no idea when anything would be done about it
14:08.51kodomofile: ok - will do - maybe it'll at least someone like me in the future in the right direction
14:11.47kodomofile: besides... I noted something odd in the * forum I posted to: the error message I had and posted got tructaced (but only when displaying it).
14:12.12kodomoWhen I edit the posting it reads "WARNING[9241]: pjproject:0 <?>:                           SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> <SSL routines-ssl3_get_client_hello-no shared cipher> len: 0"
14:12.30fileyou have to use the pre-formatted text option or Discourse will attempt to interpet it
14:12.30kodomoWhen I read the post it reads “WARNING[9241]: pjproject:0 <?>: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336109761> len: 0”
14:12.47kodomofile: ack
14:13.24*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
14:16.07kodomoSamot: file: Thank you for the support - really appreciated!
14:16.58SamotYup.
14:17.14*** join/#asterisk jkroon (~jkroon@197.96.224.13)
14:17.21SamotJust in the future, if you come in on a Monday and then leave but the issue wasn't fixed or still is a problem...
14:17.42kodomoI'll notify you the next time about it, ack ;)
14:17.46SamotWhen you come back on Saturday, mention that you got helped on Monday and a little recap.
14:17.53SamotNot that.
14:18.05SamotOther people who were in the channel might recall seeing what happened before...
14:18.24SamotOr won't try to treat this like nothing has been done before.
14:18.34SamotAnd start from step 1 when you left off on step 4
14:36.24Asteriscosamot... the problem was the  GXW410X Grandstream!
14:36.42AsteriscoOutgoing Call Dial Plan: {x+}
14:36.47Asteriscogrrrrrrr
14:58.10*** join/#asterisk jamesaxl (~James_Axl@109.172.62.242)
15:06.30AsteriscoSamot... i've resolved! :)
15:06.54Asteriscothe problem was the Grandstream gateway
15:07.11Asteriscothat accept only number from 0 to 9
15:07.47Asteriscowith { [x#*]+ } now i can use also * and #
15:07.48Asterisco:)
15:39.10*** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-pcbqvmqummwkrcjd)
15:39.10*** mode/#asterisk [+o bford] by ChanServ
15:49.33*** join/#asterisk robink (~quassel@unaffilated/robink)
15:50.08*** part/#asterisk nede (~nede@194.74.31.178)
15:58.53*** join/#asterisk daemonwrangler (~daemonwra@168.235.108.29)
16:18.53*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
16:42.12*** join/#asterisk Samot (sid133316@gateway/web/irccloud.com/x-idpckytlmvdyxrgu)
16:51.20*** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye)
16:59.34*** join/#asterisk jamesaxl (~James_Axl@109.172.62.242)
17:47.26*** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye)
17:59.02*** join/#asterisk paulgrmn__ (~paulgrmn@c-68-34-113-42.hsd1.mi.comcast.net)
18:16.05*** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye)
18:17.33*** join/#asterisk gregs (sid160074@gateway/web/irccloud.com/x-wvagmzhzrdmziexj)
18:29.34*** join/#asterisk bariskayar (97fa0efc@gateway/web/freenode/ip.151.250.14.252)
18:48.33*** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt)
18:58.28*** join/#asterisk u0m3_ (~u0m3@188.27.122.117)
19:39.11*** join/#asterisk jkroon (~jkroon@165.16.204.174)
20:47.56*** join/#asterisk Typhon (~Typhon@ipservice-092-211-215-226.092.211.pools.vodafone-ip.de)
21:10.39*** join/#asterisk Catatronic (~corrupt@unaffiliated/catatronic)
21:17.04*** join/#asterisk Dirk23 (Dirk23@diehildebrands.de)
21:35.06*** join/#asterisk areski (~areski@37.223.225.58)
21:35.11*** join/#asterisk ghoti (~paul@75.98.206.5)
21:36.24*** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com)
21:40.54*** join/#asterisk Oatmeal (Suzeanne@gateway/vpn/privateinternetaccess/suzeanne)
22:05.09*** join/#asterisk jamesaxl (~James_Axl@109.172.62.242)
22:42.36*** join/#asterisk paulgrmn_ (~paulgrmn@c-68-34-113-42.hsd1.mi.comcast.net)
23:17.41*** join/#asterisk startledmarmot (~startledm@cpe-75-82-221-87.socal.res.rr.com)
23:32.55*** join/#asterisk simbalion (~simba@45.77.108.203)
23:33.30simbalionHi, what would be the best choice for a GUI for Debian 9? The freePBX documentation hasn't been updated since Jessie (Debian 8) so I'm thinking maybe the community has moved on to something better
23:35.44simbalionor maybe I should skip using a GUI entirely?
23:50.24*** join/#asterisk robink (~quassel@unaffilated/robink)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.