IRC log for #asterisk on 20170801

00:00.04[TK]D-Fenderthat too
00:02.07_carmexwell i haven't tested inbound calling
00:02.17_carmexbut, i did have everyrhing working on a different provider
00:02.22_carmexit just stopped when i swapped carriers
00:06.02_carmexso this problem is outbound only
00:27.52*** join/#asterisk _carmex (~wsmith@173-31-142-124.client.mchsi.com)
00:30.33lvlinuxdoes ${CHANNEL(endpoint)} exist with chan_sip, or just pjsip?
00:31.25*** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com)
00:35.15*** join/#asterisk phix (~threat@C-61-69-201-111.for.connect.net.au)
01:00.34[TK]D-FenderNo, the problem is no "outbound only"
01:00.40[TK]D-Fenderthat calling IN to * is failing
01:00.52[TK]D-Fenderendpoint = pjsip term
01:01.04[TK]D-Fenderthat's what peer is for in SIP
01:01.08[TK]D-Fenderchan_sip that is
01:02.26*** join/#asterisk phix (~threat@C-61-69-201-111.for.connect.net.au)
01:23.24_carmexi got extension to extension working
01:23.25_carmex:|
01:42.14*** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com)
01:46.05*** join/#asterisk mahlon (~mahlon@martini.nu)
01:47.07_carmex[TK]D-Fender, still around?
01:48.01[TK]D-Fenderfor now
01:49.07_carmexno longer getting 404. nothing really changed
01:49.14_carmexgetting 408 and 601
01:49.37_carmexcore restart now is the best way to "refresh" settings?"
01:51.23_carmex*603
01:52.42Samot408 is a timeout error
01:53.17Samot603 is a declined
01:54.53_carmexthat's what i'm reading
01:55.00_carmexbut i'm not finding what causes that
01:55.01SamotShow it
01:55.04Samot?pb
01:55.07Samot~pb
01:55.07infobotpastebin is probably a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
01:55.10_carmexone sec
01:55.13SamotShow the debug with these errors
01:55.14_carmexi think i have an idea
01:55.29SamotBefore you do anything
01:55.36SamotShow what are getting now
01:55.43Samot+you
01:58.36_carmexhttps://pastebin.com/iNhARYpG
01:58.41_carmexi got the same
01:58.44_carmexboth times
01:59.14SamotSo you were wrong with both your errors
01:59.19SamotYou said 408 and 601
01:59.24SamotBut you mean 480 and 603
01:59.24_carmex408 and 603
01:59.28SamotNo.
01:59.30_carmex480*
01:59.31_carmexyeah
01:59.42SamotSIP/2.0 480 Temporarily unavailable <-- 480
02:00.14_carmexokay, so i'm sitting here idling.. nothing is going on with either phone. and i'm getting random SIP messages from ext 201 to 202 ?
02:00.50SamotOK..
02:00.50SamotSo
02:00.52SamotNow.
02:00.56SamotShow a WHOLE call.
02:01.11SamotFrom the moment you dial from the phone to the moment you end the call.
02:01.13_carmexnot sure what you mean. i copied everything
02:01.17SamotNo.
02:01.18_carmexit ends by itself
02:01.20SamotThere's no INVITE
02:01.54_carmexweird. so now my debug in the terminal is flooding
02:01.56SamotYou might have to adjust your scrollback buffer so you have a bigger scrollback
02:01.58_carmexrandom extensions
02:02.07SamotYou're exposed to the Internet
02:02.14_carmexfrom 7008 to 7008
02:02.14SamotSo people are jamming you
02:02.17_carmexoh
02:02.24SamotLike, lock down the PBX
02:02.25_carmexjokes on them. doesn't even work for me
02:02.27_carmex:D
02:02.55_carmexmight be why i'm getting forbidden? for the flooding
02:03.01_carmexlet me install iptables, one sec
02:06.06[TK]D-Fenderno
02:06.12[TK]D-FenderThat is all a waste of time
02:06.22[TK]D-FenderYou're not even looking at your call
02:06.41SamotWell that's the thing.
02:06.45SamotI told him to get a new call..
02:06.55_carmexatm i can't see anything but random extensions
02:06.57SamotAnd sip debug showed him being hammered now.
02:07.02_carmex^
02:07.12[TK]D-Fenderhttps://pastebin.com/iNhARYpG
02:07.16SamotRight
02:07.18[TK]D-FenderYou have debug for ONE STUPID PEER
02:07.28Samot10:00:51 PM <Samot> Now.
02:07.28Samot10:00:55 PM <Samot> Show a WHOLE call.
02:07.28Samot10:01:11 PM <Samot> From the moment you dial from the phone to the moment you end the call.
02:07.43[TK]D-FenderYou are not looking at the call going OUT
02:07.46Samot10:01:54 PM <_carmex> weird. so now my debug in the terminal is flooding
02:07.49[TK]D-Fender"sip set debug on" <-
02:07.53SamotFFS man
02:07.57SamotWe just went over this
02:08.06_carmexthat's what i typed
02:08.07SamotHe went to turn it on again and he is being attacked.
02:08.10[TK]D-FenderNo, it isn't
02:08.17[TK]D-Fenderyou restricted it to a peer/ip
02:08.25[TK]D-FenderOtherwise we see the shit after your DIAL
02:08.26[TK]D-Fender^
02:08.28SamotThen I can't see the WHOLE CALL
02:08.31[TK]D-Fender== Using SIP RTP CoS mark 5
02:08.31[TK]D-Fender<PROTECTED>
02:08.38[TK]D-FenderCause that other side WAS called
02:08.41_carmexi dont even k now how to do that .... that would make it easier
02:08.45[TK]D-FenderPackets were SENT
02:08.55[TK]D-Fender<[TK]D-Fender> "sip set debug on" <-
02:08.56*** join/#asterisk u0m3 (~u0m3@5-12-122-119.residential.rdsnet.ro)
02:09.02[TK]D-FenderSHOW us you setting it and placing a new call
02:09.03_carmexthat's the only command i used
02:09.10[TK]D-Fendernonsense
02:09.13_carmexi can't see anything now... it's being flooded
02:09.16[TK]D-Fenderwe'd have seen the INVITE * generated
02:09.29[TK]D-FenderI don't care what you thinkk your eyes see
02:09.35[TK]D-Fendergrab MORE then you need
02:09.35_carmexlol
02:09.37Samot_carmex: Is this machine behind NAT?
02:09.43_carmexSamot, no
02:09.58[TK]D-FenderYou are placing calls
02:10.01[TK]D-FenderWe see them there
02:10.11[TK]D-FenderI don't care how much ELSE gets mixed in there
02:10.19SamotDude, I don't want to sift through hackers attempting to make calls.
02:10.23[TK]D-FenderRefusing to look is flat out stupid and wasting everyone's time
02:10.30[TK]D-FenderI WILL sift though it
02:10.40SamotThat's great.
02:10.44SamotIt doesn't work now
02:10.47SamotIt hasn't worked yet
02:10.47[TK]D-FenderI want someone to get their hands off their nuts and PROVE they are folling directions
02:10.50[TK]D-Fenderand show a call
02:10.52_carmexi don't think you realize the amount of flooding i'm getting. i can't even scroll at the moment
02:10.52SamotSo we can fix him from being raped.
02:10.58SamotThen we can get to the call.
02:11.02[TK]D-Fenderyou don't HAVE to scroll
02:11.15[TK]D-FenderPutty can capture the ENTIRE stupid buffer in TWO CLICKS
02:11.21_carmexenlighten me
02:11.21[TK]D-Fenderwithout giving a shit about "scrolling"
02:11.36[TK]D-FenderRight click on the TITLE BAR > COPY ALL TO BUFFER
02:11.40_carmexboom
02:12.34[TK]D-FenderI expect to SEE the command being issued, a shit-tone of other crap, AND the complete call attempt in there
02:12.40[TK]D-Fendermake sure your buffer is BIG ENOUGH
02:13.36*** join/#asterisk thagreen (~cloudchas@ip68-102-50-198.ks.ok.cox.net)
02:13.58thagreendoes anyone know anything about remote forwarding random numbers
02:14.06thagreenplease msg me
02:14.14SamotWhat do you mean by that thagreen?
02:14.15[TK]D-Fendergive a clearer description
02:14.30[TK]D-Fender"remote forwarding" and what is a "random number"?
02:14.48thagreensay a phone i dont physically have in my hand
02:15.03_carmexhttps://pastebin.com/QC0H9Hzm
02:15.05[TK]D-Fender"in your hand" isn't a "thing"
02:15.14[TK]D-Fenderand what is "random number"?
02:15.16thagreenmessage me
02:15.20drmessanoROFL
02:15.21[TK]D-FenderJust type it HERE
02:15.46thagreenhow can i forward a cell phone thats is not in my pocket house card whatever
02:15.53thagreenremote forwarding
02:15.53[TK]D-FenderIf you can't use real words clearly here then yuo could be wasteing people's time PM'ing trying to get the SAME answers
02:16.09[TK]D-FenderIf you're going to join a communications channel you should capable of communicating
02:16.19thagreenalright
02:16.51thagreenso remote forwarding a cell phone not in my possession
02:16.56_carmexmy buffer is 20K lines...so hopefully that's enough
02:17.04_carmex;)
02:17.07drmessanothagreen: You want to forward someone elses phone?
02:17.13[TK]D-Fender"sip show peer trunk"
02:17.29thagreenmy wifes yes
02:17.37[TK]D-Fenderthagreen, You don't get to do that
02:17.40drmessanoForward it where?
02:17.50[TK]D-FenderNoone gives you the power to fuck with other peoples phone's magically
02:17.51thagreenis there anyway possible to remote forward it to say a house phojnew
02:18.03thagreenusing like a pbx service or something
02:18.06[TK]D-FenderNo, it's THEIR phone
02:18.07drmessanothagreen: #1 This isn't an Asterisk anything
02:18.16drmessano#2 She can forward it from the phone
02:18.18[TK]D-Fender"PBX service" doesn't give you control over their phone
02:18.19drmessanoAsk her to do it
02:18.29[TK]D-FenderMy cell phone doesn't care about you
02:18.30thagreennever mind
02:18.33[TK]D-FenderNeither does hers
02:18.37drmessanonever mind what?
02:18.58thagreenis there anyway possible to remote forward a cell phone
02:19.10drmessanothagreen: That's hacking, so no
02:19.10[TK]D-FenderYes.  Be a tech in the cell phone company <-
02:19.20drmessanothagreen: Ask your wife to do it
02:19.26[TK]D-Fenderbecause that COMPANY controls where calls to her go
02:19.26drmessanoFrom the phone
02:19.33[TK]D-FenderOr her on the phone
02:19.42thagreenyour missing what im saying i guess
02:19.47drmessanoNope
02:19.48[TK]D-FenderYou don't to remote control that any more than you get to remote control other driver's cars on the freeway
02:19.51*** join/#asterisk tzafrir (~tzafrir@206.167.44.157)
02:19.53drmessanoYou want to forward it
02:19.58drmessanoSo go on the phone
02:20.00[TK]D-FenderYou want to manipulate things you have no control olver
02:20.01drmessanoand forward
02:20.04thagreenI DONT HAVE THE OPHONE
02:20.06[TK]D-Fenderyou can't trol random people's phones
02:20.07thagreenthats what your missing
02:20.10thagreenpay attentsion
02:20.12drmessanoIm not missing it
02:20.12[TK]D-FenderI get that
02:20.13drmessanoSee
02:20.20[TK]D-Fenderand you don't get that if you don't ahve the phone you'
02:20.21drmessanoIm making a point
02:20.22thagreenyes you can
02:20.24[TK]D-Fenderre FUCKED
02:20.29drmessanoWE CANT HELP YOU HACK YOUR WIFES PHONE
02:20.30thagreenmy partners done it
02:20.38[TK]D-FenderGo ask your partner
02:21.07[TK]D-Fenderbecause there is no magic "make someone's cellphone do something" protocol or command stack out there for Joe Blow
02:21.07Samot_carmex: <--- Reliably Transmitting (no NAT) to 185.22.155.15:5074 --->
02:21.07SamotSIP/2.0 603 Declined
02:21.15[TK]D-Fenderotherwise you could fuck EVERYBODY'S lives up
02:21.17Samot_carmex: Your provider is declining the call.
02:21.30thagreenyou have to forward certain ports or something
02:21.38drmessanoROFL
02:21.40_carmexSamot, could that be because of the flooding? i read it was a decline from the provider...but idk why it would be declining
02:21.42drmessanoDude, no
02:21.49drmessanoForward ports?
02:21.49[TK]D-FenderJoe blow does *NOT* get to randomly impose his will over someone else's cell phone
02:21.51Samotthegreen: You have to forward the number from the CARRIER
02:21.59SamotIn a cell phone's case.
02:22.05[TK]D-Fenderthagreen> you have to forward certain ports or something <- cell phones don't HAVE "ports"
02:22.31Samotthagreen: You have to have access to the phone number account in some way to do this.
02:22.37_carmex[TK]D-Fender i think the new pixel has a usb-c port
02:22.44Samotthagreen: Which means the password/PIN
02:22.53[TK]D-Fender_carmex, <[TK]D-Fender> "sip show peer trunk"
02:23.05_carmex[TK]D-Fender idk what that means
02:23.08[TK]D-Fender_carmex, he doesn't POSSESS the phone
02:23.10Samotthagreen: Do not PM me.
02:23.16SamotI did not ask you to.
02:23.18_carmex[TK]D-Fender i was joking about the port
02:23.28[TK]D-Fender_carmex, It means type the line into * CLI
02:23.32SamotIf I wanted to speak privately to you I would have take your offer on a PM.
02:23.33_carmexgotcha
02:23.34_carmexone min
02:23.57_carmex[TK]D-Fender, "peer trunk not found"
02:24.06[TK]D-Fender<PROTECTED>
02:24.08Samotthagreen: I don't think you understand. I'm agree with the others. You cannot do what you want.
02:24.16[TK]D-FenderThen what's this junk you're Dial()'ing?
02:24.24Samotthagreen: While there may be a way to do it, it crosses lines I do not cross.
02:25.00*** join/#asterisk phix (~threat@C-61-69-201-111.for.connect.net.au)
02:26.08_carmexSamot, i am now not getting a failed call. it's just hanging on "calling"
02:26.13Samot_carmex: What extension are you dialing from?
02:26.22SamotWhat did you assign the device?
02:26.24_carmexSamot, 202
02:26.26SamotOK
02:26.27[TK]D-Fender[TK]D-Fender> Then what's this junk you're Dial()'ing?
02:26.28[TK]D-Fender^^^^
02:26.43SamotRemember that part where I said his system was being attacked.
02:26.45SamotAnd it wasn't secure
02:26.50SamotSo it's LETTING IT ALL IN
02:26.50[TK]D-Fenderirrelevent
02:26.56SamotThat's ALL The crap
02:27.03[TK]D-FenderHis dialplan isn't working because he's dialing shit that he didn't DEFINE
02:27.05[TK]D-Fender^
02:27.08SamotOK.
02:27.14[TK]D-FenderI don't give a shit how much other crap is happening
02:27.17_carmexi'm just dialing random local numbers
02:27.22[TK]D-FenderWhy do HIS dials fail?
02:27.26[TK]D-FenderBecause he fucked up
02:27.31_carmextrue dat
02:27.42[TK]D-FenderDIALING A PEER HE NEVER DEFINED
02:27.56[TK]D-FenderSo ... why did you put "trunk" there when you didn't DEFINE it?
02:28.05_carmexhowever, the same config worked on the prior carrier? nothing has changed
02:28.15[TK]D-Fender<PROTECTED>
02:28.29[TK]D-FenderYou don't HAVE a peer named [trunk]
02:28.41[TK]D-FenderSo "it worked before" must have bee ANOTHER CONFIG FILE
02:28.51[TK]D-FenderBecaus that shit doesn't EXIST right now
02:29.18[TK]D-Fender<_carmex> [TK]D-Fender, "peer trunk not found"
02:29.26[TK]D-FenderIt's. Not. There.
02:29.39[TK]D-FenderAnd that's YOUR dial command.  You put whatever you have in there
02:30.27_carmexweird
02:30.32[TK]D-FenderNo.
02:30.34_carmexi'll just change it to 202
02:30.39[TK]D-FenderThere's nothing "wierd" about this
02:30.46_carmexi beg to differ
02:30.50[TK]D-FenderYou have a Dial() to call something you never defined
02:31.02[TK]D-FenderThere is no [trunk]
02:31.06[TK]D-Fenderyou are trying to dial it
02:31.13[TK]D-Fenderthat is a stright up fail
02:31.22[TK]D-Fender+a
02:31.49[TK]D-Fender-- Executing [2024561212@from-internal:3] Dial("SIP/202-00000005", "SIP/+2024561212@trunk" <- "202" isn't going to make [trunk] magically exist
02:32.28_carmexbut that's what it's called in sip.conf?
02:32.37Samot_carmex: Please shoe your sip.conf
02:32.45SamotMask _any_ secrets only.
02:32.52Samot~pb
02:32.53infobotwell, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
02:34.47_carmexok
02:34.48_carmexone sec
02:34.50[TK]D-Fender<_carmex> but that's what it's called in sip.conf? [trunk]
02:34.58[TK]D-FenderOr it would be .. if it EXISTED properly
02:35.09[TK]D-FenderYOU should have made that.
02:35.12[TK]D-FenderWhy are you asking?
02:35.18[TK]D-FenderYou don't seem to have a clue what you are DIALING
02:35.24[TK]D-FenderYou need to serious go read the book
02:35.40[TK]D-FenderYou don't know where any of the basic sections belong or how they interact
02:35.53_carmexthat's why i asked you <3
02:36.06[TK]D-FenderYou don't get to shove arbitrary stuff into your dialplan and expect it to know what to do
02:36.24[TK]D-FenderYou skipped all of the "learning" part
02:36.42[TK]D-Fenderand we shouldn't be explaining that part.  that's what guides including that free book are for
02:37.34[TK]D-FenderThe same as walking into a C++ programming channel  writing purly invented syntaxand asking why it doesn't work, never having read the basics
02:37.50[TK]D-Fender" I typed a word ... what should it mean?"
02:37.54[TK]D-FenderNOT  a good start
02:38.22_carmexSamot, https://pastebin.com/ymN9AR0X
02:39.26_carmex[TK]D-Fender, we'll be the best of friends when this is all over
02:40.06Samot_carmex: 1) That is the most convoluted trunk for Flowroute I have ever seen.
02:40.19Samot2) You do not have a peer actually called [trunk]
02:40.22[TK]D-FenderThat TRUNK you shoved in your dial means nothing
02:40.27[TK]D-Fenderand you can't explain why it's there
02:40.45_carmexaccurate
02:41.11SamotYou do not need all that crap
02:41.40[TK]D-FenderIf you don't know how to to call Dial then you have real problems
02:41.52[TK]D-FenderGo read the book and read Dial()'s instructions
02:41.52SamotNot even Flowroutes outdated Asterisk samples are that complex.
02:41.54_carmexi've been kind of frankensteining it over the last  couple of days from flowroute support articles
02:42.02SamotDude
02:42.09SamotThey have a sample config
02:42.12[TK]D-FenderDon't sit in the driver's seat and ask "what's a brake"?
02:42.14SamotRight there in your portal
02:42.22SamotLinky link right with your creds and shit.
02:42.24_carmexyeah, but when it didn't work i opened a ticket
02:42.28SamotOK
02:42.32SamotYour trunk is insane
02:42.34[TK]D-FenderYou're copying random things you're clearly not doing the homework to understand
02:42.41[TK]D-FenderAnd since that's the level of commitment here....
02:42.43SamotAnd you're calling on the wrong peer name in your Dial string
02:42.44[TK]D-FenderI'm out.
02:43.10_carmexright, so i changed it in the dialplan
02:43.12_carmexto 202
02:43.24SamotWhy would you dial 202 from 202?
02:43.37SamotThe trunk name is flowroute-trunk
02:43.39SamotNot trunk
02:44.31SamotChange your dial string that you want to go out to Flowroute from @trunk to @flowroute-trunk
02:44.49_carmexokay, i did that and now it's getting the dialtone and failing
02:45.01SamotBut you need to do some serious looking at that machine
02:45.13_carmexgetting hammered atm
02:45.16*** join/#asterisk phix (~threat@C-61-69-201-111.for.connect.net.au)
02:45.32SamotThere is no point in making that trunk work
02:45.42_carmexmachine*CLI> sip show peer flowroute-trunk
02:45.42_carmexPeer flowroute-trunk not found.
02:45.43SamotWhen you are opened like a trailer park prom date
02:45.57_carmexlol
02:48.45SamotAt this point, you make that trunk work and it can send out calls...
02:48.53SamotAnd you have nothing really in place for the rest...
02:49.00SamotYou're just asking for trouble.
02:49.15Samothttps://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample
02:49.22Samot^^^ Go read it
02:49.29SamotThat has all the settings for peers
02:49.32SamotWhat they do
02:49.46SamotExamples for trunks, dial strings, device peers
02:51.55_carmexcool
02:51.56_carmexthanks
02:52.06_carmexi've been looking for something like thois
02:52.08_carmex*this
02:52.35SamotThe install instructions for Asterisk cover the "make samples" option
02:52.57SamotWhich creates all samples for all the configs.
02:53.11SamotWhat you were just lined to
02:53.14SamotWhat you were just linked to
02:55.40_carmexugh. it just completed a call. and now it won't again
02:57.52_carmexcall established \o/
03:00.33_carmexthanks Samot
03:00.56lorsungcunp
03:42.44*** join/#asterisk phix (~threat@C-61-69-201-111.for.connect.net.au)
03:53.17*** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com)
03:58.08iheartlinuxI have one peer connected via chan_sip, my provider (flowroute). 5060 is not open. I am able to call in. Is this because the peer is registering to my provider, and establishing a route that way?
03:59.08lorsungcuiheartlinux: that is correct
03:59.17lorsungcualthough calling it a 'route' is misleading
03:59.48iheartlinuxwhat would be the proper term?
04:00.02lorsungcuyour router is hanging onto that connection started by the registration, and subsequent messages are replied to on the port it NATs to
04:00.30lorsungcuroute is sort of reserved for a route. what you're doing is NAT.
04:01.45iheartlinuxI don't have nat turned on in chan sip. My server is on a VPS with a public ip. My end devices are connected via vpn'ed/pjsip
04:02.08lorsungcuare you using freepbx?
04:02.21iheartlinuxvanilla asterisk
04:02.37iheartlinux13.15.1
04:02.49lorsungcu'turning nat on' implies your PBX is behind NAT.
04:03.02lorsungcuyou aren't, so you don't need it
04:03.35lorsungcuwhatever you're using as a firewall evidently has an established/related rule, and that's handling this.
04:04.18lorsungcusame concept without the layer of NAT
04:04.50iheartlinuxshorewall, no 5060 is off. flowroute is using ip (not sip reg)
04:05.19lorsungcuyou just said it was registering.
04:05.28lorsungcugoes homwe
04:06.08iheartlinuxwell, it is a peer
04:06.47iheartlinuxor friend rather
04:07.33drmessanoThat's still a peer
04:07.49iheartlinuxI know
04:08.13drmessanoThere's no such thing as a SIP 'friend'
04:10.12iheartlinuxwell, it's set to "type = friend"
04:24.18iheartlinuxok, so I changed my provider to peer, which ofc required opening 5060. Just trying to fully understand asterisk, thanks
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04:31.30snadgeasterisk14 is the bees knees
04:31.51snadgewell not really, but it works with our crusty old agi code.. amazing ;)  .. last functioning version was 11
04:32.07snadgetheres a few little side cases, but i dont particularly feel like investing effort in a legacy platform
04:32.14snadgethats probably going to be replaced by freeswitch
04:32.18snadge(not my choice)
04:32.35drmessanoI guess you lost me
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05:23.05[TK]D-Fender<iheartlinux> ok, so I changed my provider to peer, which ofc required opening 5060. Just trying to fully understand asterisk, thanks <- no
05:23.09[TK]D-Fenderthat has nothing to do with anything
05:25.30iheartlinuxtype peer sends calls correct?
05:25.34iheartlinux[TK]D-Fender:
05:25.45[TK]D-FenderBOTH
05:31.58iheartlinuxit didn't work correctly without opening port 5060 (outgoing was fine, incoming was not). If I had type set to friend, asterisk created both a user & a peer for my provider. which allowed for bi-directional comm. My provider is sending my DID's vi IP. So opening port 5060 with [type = peer] allowed for bidir
05:35.30[TK]D-Fenderif your port isn't open then packets aren't making it in
05:35.36[TK]D-Fenderthat is regardless.
05:36.01[TK]D-Fenderpeer is bidirectional as well
05:36.16[TK]D-FenderYou don't seem to understand what the types actually mean.
05:36.24iheartlinuxIt worked with type = friend and my incoming port 5060 blocked.
05:36.41[TK]D-FenderYou're mixing things
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05:36.50[TK]D-Fenderpacketsd don't magically get through because of the TYPE
05:36.54[TK]D-Fenderthats your router
05:37.05[TK]D-FenderAnd if you have a KEEP ALIVE keeping the port open
05:37.17[TK]D-FenderOr are REGSITERING which does the same
05:37.33[TK]D-FenderSo saying you switched to "ip" ....
05:37.50[TK]D-Fenderthere goes a chunk of the regular things forcing it open at all
05:37.59[TK]D-FenderType has nothing to dow ith it
05:38.34iheartlinuxwhat does my local router have to do with this?
05:40.03drmessanoUh
05:40.07drmessanoEverything
05:40.21iheartlinuxmy pbx is on a vps, connected to a public IP. my phones are connecting via pjsip on a totally different port. im only using sip for flowroute
05:41.40drmessanoiheartlinux: Have you ever looked at SIP DEBUG?
05:42.06iheartlinuxyes I have it open.
05:42.11[TK]D-FenderNone of that shows us anything useful so far
05:42.32drmessanoiheartlinux: All of that communication is viewable
05:42.36drmessanoThere is no magic
05:44.59iheartlinuxhttps://old.trunkmasters.com/pastebin/typepeer5060open
05:45.31drmessanoThats not SIP Debug
05:45.47iheartlinuxasterisk -Rvvvvvvvvvd
05:45.54drmessanoThats not SIP Debug
05:45.55iheartlinuxdoes that not show debug info
05:45.58iheartlinuxok
05:46.01drmessanoIt does not
05:46.13drmessanosip set debug on
05:46.29iheartlinuxCore debug is still 1
05:46.39drmessanoJFC
05:46.46drmessano*****Thats not SIP Debug******
05:46.58drmessanoSIP DEBUG != Core Debug
05:47.03drmessanoSIP DEBUG != Verbose CLI
05:47.07iheartlinuxok
05:47.22iheartlinuxgotit
05:51.10iheartlinuxhttps://old.trunkmasters.com/pastebin/typepeer5060open
05:55.56[TK]D-FenderThere we don't even see the start of a call
05:56.59iheartlinuxPJSIP/212-0000000f answered SIP/flowroute-00000007
05:58.25[TK]D-FenderThere we don't even see the start of a call
06:00.59iheartlinuxPJSIP/212-0000000f answered SIP/flowroute-00000007
06:01.01iheartlinuxhttps://old.trunkmasters.com/pastebin/typepeer5060open
06:01.02drmessanoiheartlinux: You're selling this?
06:01.49drmessanoYou really should learn a lot more about debugging before selling people a vital public utility
06:02.00drmessanoJust sayin
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06:02.47iheartlinuxappreciate the input
06:06.04[TK]D-Fenderpiping is making reading a stupid mess
06:06.09[TK]D-FenderWith all the ANSI garbage
06:06.16[TK]D-Fender[0K[2017-08-01 00:53:43] [1;31mWARNING[0m[22255][C-00000001]: [1;37mast_expr2.fl[0m:[1;37m470[0m [1;37mast_yyerror[0m: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
06:06.25[TK]D-FenderYou've got broken expression in your dialplan
06:06.39[TK]D-Fenderand I'm not sure the point of what you're showing
06:07.07iheartlinuxNo point, it's working
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06:29.35[TK]D-FenderWell one line for sure isn't
06:32.59iheartlinuxsame => n,GotoIf($[${DB(blockcaller/${CALLERID(num)})} = "1" ]?blocked)
06:33.17iheartlinuxwhich I grabbed from here: https://gist.github.com/warewolf/58dc06bbe0548f75c8a2
06:35.06iheartlinuxit blocks unwanted calls, but your right
06:40.21iheartlinuxfixed it: same => n,GotoIf($["${DB(blockcaller/${CALLERID(num)})}" = "1" ]?blocked)
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08:02.28FuriousGeorgehi everybody
08:03.33FuriousGeorgeyou may remember me from educational films such as "three minus two equals negative fun!"
08:07.42FuriousGeorgekidding asside -- and fully realizing that this is not an * related question -- is SIP video ever gonna really work without a b2bua doing all the heavy lifting?
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08:18.13FuriousGeorgeno expert here, but seems to me that so long as video support is passthru only (i assume this is true with every sip server, not just the ones I know), then it will always be dependant on clients playing nice
08:19.03FuriousGeorgewhich is to say that there are infinite possibilities, and (no math major here) it is guaranteed never to really work.  nothing can be done server side
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12:47.44SamotFuriousGeorge: You do not need a B2BUA to do video calls. They can be done endpoint to endpoint.
12:56.18fileit also depends on the definition of 'work', both sides certainly have to negotiate things
12:56.36SamotWell, of course.
12:56.44SamotAnd deal with NAT
12:57.03SamotAnd all sorts of other things that making a B2BUA needed.
12:57.13Samots/making/make/
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13:10.02darkunderlordok this is a long shot, but anyone ever have issues with remote phones not coming back up on a Cisco EasyVPN? I still need to packet capture, etc but these are sip phones at a remote location. ONly difference is that this location is on EasyVPN style Cisco connection, rather than traditional site to site VPN.
13:10.24darkunderlordit can take sometimes an hour or two after network comes back up for them to reregister with asterisk
13:10.37darkunderlordbut all other traffic works fine, no firewalling, etc
13:13.53eric_hilldarkunderlord, depending on your site to site VPN connection settings, the SA might still be sitting in one of the two VPN endpoints.
13:14.22eric_hillAssuming your phones might be on a different subnet, one spurious SA would cause the phones not to work while everything else works fine.
13:16.41darkunderlordwhat's odd is today the ones on wireless are connected but wired aren't Maybe this resides somewhere else.
13:18.56eric_hillIs your EasyVPN running in network extension mode or client mode?
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13:26.01jbrouwersHey guys, I'm not getting RecordingFinished events from ARI. What could be the cause of that?
13:26.33darkunderlordok so subscribes are getting to me, but here is the sip debug for extension 430
13:26.34darkunderlordhttps://pastebin.com/Pe89mu9A
13:27.10darkunderlordwe have a whole network on the other side, not just the one phone. So maybe network extension? :)
13:29.19fileit's not getting the response that Asterisk is sending
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13:31.50darkunderlordfile: we don't block any ports though and eventually it just starts working. but it can be anywhere between 30 minutes and a couple hours. Usually this is after the internet provider goes down and comes back up
13:32.05fileokay, but the second SUBSCRIBE Is a retransmission
13:32.11filewhich means it either didn't get the response, or it ignored it
13:32.15darkunderlordhmm ok
13:32.48darkunderlordso maybe my phone needs a setting changed if it's ignoring it, Hmm thanks.
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13:47.07darkunderlordhow can I direct sip debug to a file?
13:50.50eric_hilldarkunderlord, you have an SA, likely on your side, still sitting in the VPN box that's encrypting the responses being sent back to the phones with the wrong tunnel information.
13:51.18eric_hilldarkunderlord, after some time, that SA finally expires and the new (correct) one takes precedence.
13:52.04eric_hilldarkunderlord, until that timeout occurs, any traffic headed back to the site is being encrypted with the old key, sent to the site, and disposed of because it doesn't match an active SA.
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13:58.05darkunderlorderic_hill: thanks
14:01.11darkunderlorderic_hill: is it weird though that it's only a problem for 3 of 6 phones?
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14:04.33eric_hilldarkunderlord, I'm guessing based on my past experiences with site to site VPN's and no information about your subnet layout or log files that might indicate another issue.
14:05.16eric_hilldarkunderlord, start with the SA.  If that's not the problem, pick apart another piece and prove or disprove it.
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14:13.00darkunderlordany other way to test 5060 to one of my phones from here? Like how you telnet to smtp servers to test? :)
14:14.20darkunderlordnvm :)
14:14.35eric_hilldarkunderlord, netcat is a glorious tool.
14:15.02ChainsawI like nc -v
14:15.08ChainsawAt least it explicitly says "yes, that port opened".
14:15.16ChainsawUnlike telnet.
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14:27.49darkunderlordsame shit happens with same phones connected to a PJSIP server iwth entirely different version of asterisk. so odd. :D
14:28.46eric_hillI've been fighting a losing battle with PJSIP on a new server, so meh.  :)
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15:03.10darkunderlordif I force a new IP statically on the phone, it is allowed to register. very odd
15:04.29igcewielingdoes the source port change?
15:04.50igcewielingah, nevermind
15:06.30eric_hilldarkunderlord, are you using NAT between the sites?
15:08.05eric_hilldarkunderlord, for that matter, what is the phone?  Cisco?  Polycom? Mitel? Grandstream?
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15:44.42polysicshello! anyone is familiar with UniMRCP isntallation, please? I can't get it to compile (it never works tbh)
15:45.19polysicshere is what I get: https://gist.github.com/lpradovera/9bf1be1889e94d02fb67aef8f97edcc4
15:45.43polysicsAsterisk is 13.8 certified, UniMRCP, deps and Asterisk module are checked out from master
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15:46.01polysicsI was wondering if I am chasing something that might just not work or I am just doing something wrong
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15:50.56[TK]D-Fenderif you're not paying Digium for support you shouldn't be running that
15:52.10polysicsI am not even sure the company that asked me to do this install is paying them or not
15:52.16polysicsI didn't install Asterisk on the box :D
15:52.22polysicsI can go check
15:52.37polysicsassume they do, they have been running on -cert for a long time
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15:54.46polysicsI am just not sure if I this is a "you are doing it wrong" thing or a "it will never work" type :D
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16:07.11darkunderlorderic_hill: there's have to be some sort of NAT I'd think, if it's two diff subnets. Also they are Cisco SPA525G2 phones. It's gotta be a cisco thing
16:17.18eric_hilldarkunderlord, doesn't have to be NAT if it's a L2L tunnel.  But NAT has timeouts, tunnels have timeouts, and Cisco has bugs.  __it hapens.
16:18.46darkunderlorderic_hill: could it be something with SIP inspection? Seems like there is some smart table somewhere that needs flushed or something
16:20.17drmessanoWhat is the router onsite?
16:20.30darkunderlordboth are ASA's
16:20.54voipmonkdid you install the prereqs polysics  ?
16:22.31darkunderlordwhat's that? My router guy us at lunch :D
16:25.03drmessanohuh?
16:25.54darkunderlordnot sure what the prereqs polysics is
16:27.44RovingWriterdarkunderlord, he was talking to another person here, named polysics. check the nick list
16:28.16darkunderlordlol ok
16:29.44drmessanodarkunderlord: You really need an IRC client with a nicklist and scrollback
16:30.02darkunderlordI do, I just need a brain too :)
16:30.11darkunderlordI'm using Konversation
16:30.23drmessanoOh wow
16:31.12drmessanoKonversation is like the Koutlook Kexpress of IRC clients
16:31.15drmessanoWell played
16:31.40darkunderlordhahaha
16:32.08darkunderlorddrmessano: what do you suggest?
16:33.40drmessanoIRCCloud
16:34.18darkunderlordI don't want to pay, but I guess I can make my employer pay :)
16:34.49drmessanoYou dont HAVE to pay
16:35.33drmessanohttps://www.irccloud.com/pricing
16:35.40drmessanoBut it's a better experience if you do
16:35.42igcewielingI use Pidgin, the IRC support is good enough for me.
16:35.58drmessanoigcewieling: That's worse than Konversation
16:36.30darkunderlordhaha yeah I've used pidgin too
16:37.33drmessanoIRCCloud is great.. it's the only way to take IRC with you on the go
16:37.35salviadudirssi ftw
16:39.20igcewielingXchat was OK when I used it many years ago.
16:39.31drmessanoIt's no longer developed
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16:39.39salviadudigcewieling, try hexchat
16:39.47salviadudforked from xchat
16:40.17darkunderlordsalviadud: I use that on my win box sometimes
16:40.18igcewielingsalviadud: suggest it to darkunderlord.   I already have an IRC client.
16:40.33salviaduddarkunderlord, hexchat bro
16:41.22igcewielingMost of my messaging is via XMPP.   Not sure why I hang out here anymore. 8-|
16:41.24drmessanoYou know you can use IRCCloud FREE, you just idle disconnect after 2 hours
16:41.32drmessanoigcewieling: /quit
16:41.56salviadudigcewieling, admit it, you are oldschool
16:42.10drmessanoHe just likes to hang around and troll
16:42.14drmessanoSo he'll never leave
16:42.27igcewielingsalviadud: I admit I'm something of a luddite.
16:43.21drmessanoA luddite that works in telephony and is currently chatting on IRC with a computer
16:43.30drmessanoThat's.. rich
16:43.37drmessanoAnywho
16:50.39RovingWriteri still use mIRC
16:50.43RovingWriterso i can be a l33t scripter.
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17:03.47lvlinuxirssi/znc/bitlbee takes care of IRC, XMPP, Google Hangouts/Talk, Twitter for me!
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17:09.22learathanyone have a sample pjsip.conf for flowroute?
17:09.39lvlinuxi do, wait just a sec
17:10.00learathlvlinux: Thanks!
17:14.25lvlinuxhttp://paste.debian.net/979239/
17:14.28lvlinuxthere you go
17:14.44lvlinuxnow, that's assuming you are using IP auth
17:14.55lvlinuxif you need user/pass/registration it will be a little different
17:15.14learathhm.  I don't need a registration?
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17:16.48lvlinuxdo you have a static ip?
17:16.54learathyes
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17:16.59lvlinuxthen you don't need it
17:17.33lvlinuxregistration tells the provider where to find you, so if you're ip changes, the registration says "i'm here now at this ip---use it to send me calls"
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17:18.29learathhm.  does not like endpoint=flowroute
17:19.41lvlinux??
17:20.04lvlinuxthat config is in use on a box i have
17:20.46learathAsterisk 14.0.0?
17:20.59lvlinux13.something, but it should be the same
17:21.22lvlinuxshould work just by copy/paste
17:21.51lvlinux(well, you'll have to change the transport name from sip-udp to whatever your trasnport is named though)
17:22.05learathyeah
17:22.11learathmine matched yours pretty closely already.
17:23.03lvlinuxI believe something else is wrong if it's complaining...
17:23.14learathentirely possible :)
17:23.31learathI'm trying to port a... version 11? 12? config
17:23.35learath(pre-pjsip)
17:23.44learathat this point, I suspect moving to pjsip was a mistake.
17:23.44learathmeh
17:24.30lvlinuxNOOOOOOOO
17:24.35lvlinuxpjsip rocks!
17:24.43learathsure, but it's a huge change
17:25.01lvlinuxyes, but well worth it in my opinion, and my experience.
17:25.30RovingWriterwhy is it better?
17:25.30learathI have no opinion.
17:26.11fileif your endpoint or something is wrong, it does tell you when it's loaded
17:26.30learaththat's how I found out endpoint=flowroute was no good.
17:26.35lvlinuxRovingWriter: lots of improvements---check the docs.
17:27.00lvlinuxlearath: by "no good" you mean "not found" or what?
17:28.21learathlvlinux: erm.  lemme see if I can grab the error
17:28.24lvlinuxi bet there's a typo somewhere
17:28.26learathit might have scrolled off.
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17:28.37learathpossible
17:28.49learath[Aug  1 17:18:05] ERROR[1]: config_options.c:720 aco_process_var: Could not find option suitable for category 'flowroute' named 'endpoint' at line 223 of
17:29.21learathhah yep.  you are 100% right.
17:29.29learathit's "flowroute-trunk" not flowroute.
17:29.37lvlinuxhehe
17:30.56darkunderlordok trying this irccloud thing out :D
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19:58.32darkunderlordsure, everyone goes quiet now that I'm trying a new client :D
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20:22.18darkunderlordI've been doing this shit for years and somehow still don't understand how my IAX trunking works lol
20:30.54lvlinuxdarkunderlord: we're not getting your messages with irccloud, especially about your ignorance of IAX.
20:30.57lvlinuxlol
20:31.40darkunderlordhahaha thanks!
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21:29.52lvlinuxAny suggestions for an open source router/firewall distro that runs on standard x86 hardware? I'm considering ditching pfSense since it appears Samot was right about it dropping/mangling my RTP packets for no good reason, and without notification.
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22:51.37[TK]D-Fenderlvlinux, Buy an actual dedicated router
22:51.53salviadudlvlinux, sonicwall
22:51.56[TK]D-Fenderlvlinux, Wasteing a full PC for this only very rarely makes any sense
22:52.02[TK]D-FenderLOL ....
22:53.21salviadud[TK]D-Fender, ummm, sonicwall is useful if you're planning on using deep packet inspection.
22:53.29salviadudOtherwise, it might be a bit too much.
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23:01.14[TK]D-Fenderif you want DPI use MITM Proxy
23:01.23[TK]D-FenderAnd you can do that in software separate from your firewall
23:01.42[TK]D-FenderBecause SonicWALL is a giant rip-off
23:02.44Samotlvlinux: routerboard.com
23:02.51SamotMikrotik is the way.
23:03.00[TK]D-FenderIt's a very good option for sure
23:03.11[TK]D-FenderCertainly hard to need more for 99% of users
23:03.23lvlinuxbut it isn't open source
23:03.39lvlinuxand it's not "wasting" a PC when it's a thin client and useless for anything else :-D
23:03.54[TK]D-FenderYou had us at "useless" :p
23:03.56SamotOK..
23:03.58SamotWait..
23:04.15SamotYou have an issue with Mikrotik because it's not open source?
23:04.20lvlinuxI don't need DPI, just standard routing/firewall, and VERY good QoS. and full ipv6 support
23:04.26SamotOK.
23:04.27[TK]D-FenderIt'll cost more to power, more hassle and more things likely to go wrong.
23:04.29SamotMikrotik
23:04.38SamotGet a hEX or hAP lite
23:04.43SamotLess than $50
23:04.51SamotAll the goodies you'd like inside it.
23:04.56SamotAnd it *WORKS*
23:05.04lvlinuxmy thin client uses 5W at idle---that isn't much power, and only about 14W when full blast.
23:05.14SamotWhat
23:05.26lvlinuxDoes Mikrotik support fq_codel
23:05.27lvlinux?
23:06.45SamotNo.
23:06.49SamotIt does it better.
23:06.56SamotIt has Queues for that kind of stuff
23:07.01SamotAnd other features.
23:07.02lvlinuxwrong---there is no better
23:07.07SamotOK.
23:07.10lvlinuxqueues isn't good enough
23:07.14SamotOK.
23:07.18lvlinux(for me anyway)
23:07.36SamotSure.
23:07.58SamotJust go get a OpenWRT or DDWRT device then.
23:08.13SamotIf you're not interested in a real router.
23:10.01lvlinux"real" is quite subjective, and really the "real" routers are no more "real" than the goofy consumer stuff anymore, at least not unless you buy a cisco 6500.
23:10.32lvlinuxtheyre all just software running on mostly plain hardware, with sometimes a few enhancements here and there in hardware.
23:10.37lvlinuxbbl
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23:27.39drmessanoFOSS firewalls on consumer hardware are fine for hobby use
23:27.45drmessanoSO if that's all that is needed, go for it
23:27.56drmessanoI prefer something that works and that I don't have to think about
23:29.40robmalOh no, it's that 'don't use asterisk if you're not a pro' talk again :-/
23:30.53SamotNo.
23:31.21SamotThis is the "My pfSense router is mangling my outbound packets for no reason, gotta a better suggestion for a router"
23:31.36SamotAnd when one was given the response was "Oh, it's not open source. Can't use it"
23:32.07snadgemikrotik, ubiquiti, asus
23:32.09SamotFollowed by asking about something Open/DDWRT and pfSense do as a "feature"
23:32.19SamotAnd if it doesn't do that, then forget it.
23:33.16SamotSo no this is not a "Don't use it if your not a pro" talk
23:33.24drmessanoOh I think it's funny
23:33.24SamotThis is a "Use some common sense" talk.
23:33.30drmessanoLike
23:33.34drmessano"PFSense does X"
23:33.46drmessanoBut it does it 110% slower
23:34.03drmessanoSo whatever gains there could be from feature X are lost in implementation
23:35.28drmessanofq_codel in pfsense sounds like taping rubber pads to square wheels hoping to round them out
23:36.07drmessanoand when it goes "faster" than it did, you can brag about the performance
23:36.35drmessanoWhile anyone else with a $50 Mikrotik are just sitting back passing packets like it's a Tuesday
23:36.44robmalSo you still can't help people but you're now vendor specific? ;-)
23:36.59drmessanohuh?
23:37.20Samotrobmal: The original question was "recommend a router that's not pfsense to me"
23:37.38SamotThen they only wanted routers that did pfsense stuff..
23:37.59SamotWhich means, they'll probably have the _exact_ _same_ _problems_
23:38.19drmessanoSamot: I think he's just trolling me
23:38.27robmaldrmessano: Just a bit ;-)
23:38.29snadgeif pfsense is the answer, then you're asking the wrong question.. unless its.. what shouldn't i use for a router
23:38.32snadge:P
23:38.41drmessanorobmal: Why don't you just put me on ignore?
23:38.49drmessanosnadge: Indeed
23:39.05robmaldrmessano: Why should i?
23:39.22drmessanorobmal: Obviously i'm bothering you
23:39.56robmalNot at all.
23:46.03RovingWriterwellllppp, about that time of the day to leave the office
23:46.24drmessanoYeah, trolls are coming out
23:47.13RovingWritermade an auto-scaling group full of webservers and mariadb servers, to scale a website automatically when we do our next product launch
23:47.23RovingWriterso that'll do it for todays work
23:47.41RovingWriterand i didn't use pfsense's haproxy plugin.
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23:48.27drmessanoI prefer PFSense's LOLProxy

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