IRC log for #asterisk on 20160914

00:01.22[TK]D-FenderAnd saying that doesn't add anything to the conversation.  No change in methods, no testing, no added description about what's on each end of the call.  Nothing.
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00:01.53[TK]D-FenderI recommend correcting any (and or ALL) of these points.
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01:02.57KattyBloop
01:04.38[TK]D-FenderKatty, Mew.
01:08.12KattyHerro fenderbender
01:10.31[TK]D-FenderMsPrettyKittyKatty
01:10.34[TK]D-FenderHowgoesit?
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01:36.26KattyIt goes. You?
01:36.44KattyAny major life changes in the last 1.5yrs?
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01:49.03[TK]D-FenderNope.
01:49.50[TK]D-FenderSSDD
01:57.04KattyGood!
01:57.59KattyNot much change here either. Moved into a town house in April. Probably start looking for a house to buy in another year or two
01:58.32[TK]D-FenderSlow & steady keeps up with the glaciers
02:00.37KattyYep
02:02.11[TK]D-FenderUpgrading homes is a thing...
02:04.46KattyIs it?
02:06.23[TK]D-Fenderyup.  Change of lifestyle with more property upkeep, independent taxes, full control, full responsibility
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02:14.44KattySounds like adulting is involved
02:17.05KattyTrying to keep the adulting to a minimum
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07:45.45DanQuinneyadulting sucks
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09:18.23wyounghey gang
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12:24.57kim_bruninghello, in my sip tcpdump connecting to xs4all, in a REGISTER request, I get           From: <sip:0858769839@sip.xs4all.nl>;tag=as7bad8b24  ...       To: <sip:0858769839@sip.xs4all.nl> my version is  User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
12:25.03kim_bruningis this actually what's supposed to be happening
12:25.31kim_bruningI have a 2nd provider as well, and that logs in differently, so I'm a tad confuzzeled how that's possible
12:28.42wyoung<PROTECTED>
12:31.41*** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net)
12:31.57[TK]D-Fenderkim_bruning, Donèt see why not.  What is the actual problem?
12:32.17wyoung[TK]D-Fender: Lets pal it
12:32.21wyoung[TK]D-Fender: TF2?
12:32.30wyoungFTP
12:38.03kim_bruningI can't register with my upstream provider
12:38.28kim_bruningwell, I could until yesterday. I may have unwittingly updated asterisk
12:38.30[TK]D-FenderShow us the full attempt
12:39.05[TK]D-Fender"my version is User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3" <- this is not what I'd call "updated"
12:39.18[TK]D-Fender1.8 = DEAD.  so is 10 & 12
12:39.18kim_bruningdebian
12:39.39[TK]D-Fenderthey have 11 in their repos
12:39.44[TK]D-FenderThat is not an excuse
12:40.00[TK]D-FenderAlso not a reason for a failure though...
12:40.18[TK]D-Fenderjust kills any support and means you're massively behind on fixes
12:40.30kim_bruninghttp://pastebin.com/TZFcXZXi
12:40.36kim_bruningsuggest update to 11?
12:40.45[TK]D-FenderFull attempt from * CLI w/ SIP debug
12:40.47[TK]D-Fendernot external
12:40.48NivexI miss debian backport
12:41.17kim_bruningfair enough. let's see
12:43.40kim_bruninghttp://pastebin.com/TNbZFgez
12:43.44kim_bruningthis what you're looking for?
12:44.13kim_bruningor if not, what should I be doing instead?
12:46.00kim_bruningthe cli version looks very similar to the tcpdump
12:46.30[TK]D-FenderAnd there is NO response from them at all?
12:46.41kim_bruningindeed not, they have a 401 in their logs
12:46.55kim_bruningand that's all they tell me
12:47.01kim_bruningand no response
12:47.05[TK]D-Fenderthen something is blocking that
12:47.23[TK]D-Fendercheck your firewalls on the server / routers & confirm your WAN IP on the contact
12:47.31[TK]D-FenderAs well as any required forwarding
12:47.33kim_bruningyou'd think so, though a different provider *is* responding correctly (and is -in fact- connecting)
12:47.36[TK]D-FenderContact: <sip:s@82.94.48.148:5060>
12:47.37[TK]D-Fender^
12:47.54kim_bruningthat port is open and responding
12:47.57[TK]D-FenderA different provider might ignore what you send for the return.
12:48.07kim_bruningcould be!
12:48.44[TK]D-FenderDoes look like the IP your IRC connection is from....
12:49.23[TK]D-FenderThere is some reason why the packet isn't making it back if they say they got the REGISTER and sent the 401.
12:49.32[TK]D-Fenderask for a confirmation of where they were sending
12:49.36[TK]D-Fenderget their dump
12:49.43[TK]D-Fenderand then start tracing backwards from there
12:50.06kim_bruningI already asked... and they didn't provide further info ^^;;;
12:50.19kim_bruningso trying other avenues first
12:51.14kim_bruningand traffic to port 5060 is coming through (tcpdump on this end,  and remote box nc <ip> 5060 )
12:51.49kim_bruningif I update to .11 , is it likely my config will bork?
12:54.06kim_bruningincidentally, if I test the connection with zoiper, I can immediately log in.
12:54.13kim_bruning(which I should have mentioned earlier)
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12:58.21[TK]D-FenderSomehting else is wrong
12:58.22[TK]D-Fenderand itès not the * version
12:59.02whizzikim_bruning: Do you get any reply from Xs4all from your register attempt?
12:59.49kim_bruningwhizzi: I do not, not in asterisk
13:00.26whizzithen there’s a network related issue
13:01.08[TK]D-Fender[TK]D-Fender> then something is blocking that
13:01.08[TK]D-Fender<[TK]D-Fender> check your firewalls on the server / routers & confirm your WAN IP on the contact
13:01.14whizzieither NAT problem or a too strict firewall that doesn’t allow RELATED connections to return
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13:01.31whizzi[TK]D-Fender as always, you are right ;-)
13:02.08whizziXs4all will see a 401 being sent to your box, but that one never comes through
13:02.28whizzithis 401 has the nonce code in it, allowing Asterisk to register using this code
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13:02.43whizziif that’s okay, combined with the password, you are registered
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13:02.50kim_bruningright
13:03.19kim_bruning<scratches head>
13:03.22PhrohdohHello everyone, I just discovered asterisk late yesterday and I only have a single question so far. Is it possible for one user to directly leave another user a voicemail?
13:03.49kim_bruningmy WAN IP is ok
13:03.53zafPhrohdoh, yes
13:04.07Phrohdohzaf: Excellent, thank you for the quick response. :-)
13:04.33PhrohdohDoes anyone here run asterisk on a raspi? I saw a project just for that yesterday and am curious to know if that is viable?
13:04.59zafdon't expect to run a call center on it, but for small installs, yes
13:06.10[TK]D-FenderPhrohdoh, It's your system, it'll do whatever you configure it to do.
13:06.10PhrohdohGreat to hear. I would only throw it on my pis so that I may learn the system a bit. Of course a full install on a much more suited machine will follow. Thanks!
13:06.48Phrohdoh[TK]D-Fender: While I don't doubt that I have _never_ done anything with a system like Asterisk.
13:06.58[TK]D-Fenderpi = puny.  If you only inttend ona tiny # of connection it could do the job if you aren't transcoding, etc
13:07.10[TK]D-Fender~book
13:07.10infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
13:07.12[TK]D-Fender^^^
13:07.19[TK]D-FenderTime to read up & learn about it
13:07.24PhrohdohGreat I will do that!
13:08.22PhrohdohHm the online version of the book focuses on 1.8 but given the title for this channel I assume that is quite dated. Would that be correct?
13:08.57zaf4th edition is more up to date
13:10.51PhrohdohThanks again - http://asterisk-service.com/downloads/Asterisk-%20The%20Definitive%20Guide,%204th%20Edition.pdf
13:21.00kim_bruningwell, turning nat on, means I have xs4all listed in my peers
13:21.02kim_bruningthat's a start
13:22.12kim_bruningasterisk sends a message, then gets a reply from xs4all
13:22.26kim_bruningSIP/2.0 200 OK
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13:22.41kim_bruningbut can't call out for some reason <scratches head>
13:23.06[TK]D-FenderConsidered showing us?
13:23.28kim_bruningcan call in
13:23.30kim_bruninghang on
13:23.44whizzi*keeps hanging*
13:23.55kim_bruningok, what would you like to see
13:24.01[TK]D-Fendergrabs a big stick
13:24.14kim_bruningcall in works, I figure I probably left something silly in the sip.conf now
13:24.24[TK]D-Fender#piñatatime
13:25.22kim_bruninghttp://pastebin.com/jLUxN0nB
13:25.31kim_bruningI'm showing, I'm showing!
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13:28.16whizzi_see what you did [TK]D-Fender
13:28.51wyounghi
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13:30.23whizzikim_bruning: Do you see an attempt calling out on the xs4all trunk?
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13:31.08kim_bruning<PROTECTED>
13:32.10whizzicheck your from-header, rpid and callerid
13:32.42[TK]D-Fenderkim_bruning, Where's the CALL?
13:34.14PhrohdohSo I think I got some misinformation yesterday. Asterisk itself is not a distribution of GNU/Linux, it is instead a suite of tools that were built to run on a linux system, correct?
13:36.01kim_bruningPhrohdoh: correct
13:36.14kim_bruninggeh, I'm getting traffic in between
13:38.00[TK]D-FenderYes, but there are distros made exclusively for the purpose of deploying an * based system
13:38.05[TK]D-FenderMostly GUI driven though
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13:42.34kim_bruningI have sip debug on ....
13:42.40kim_bruningso ... trying to figure where this started
13:44.19[TK]D-Fenderreboots to take updates....
13:45.45kim_bruninghttp://pastebin.com/Gr9Wf9Bf
13:45.48kim_bruningthis should cover it
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13:56.02[TK]D-Fenderkim_bruning, you aren't showing the INVITE tiself
13:56.26kim_bruningI might have gotten it now
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14:00.59kim_bruninghttp://pastebin.com/up0Htrk6
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14:05.29[TK]D-Fender^MESC[Kvosserver*CLI> ^MESC[0KReliably Transmitting (NAT) to 194.109.16.16:5060:
14:05.36[TK]D-FenderYour provider should not be behind NAT
14:06.17kim_bruningpreferably ^^;;
14:06.20[TK]D-Fenderbut that isn't the immediate problem
14:06.27kim_bruning*nod*
14:06.34PhrohdohIn the context of this project what is a 'trunk'?
14:06.38[TK]D-FenderThre is another auth issue
14:06.57[TK]D-FenderPhrohdoh, A vague term often misused
14:07.37[TK]D-FenderPhrohdoh, But typically meant to be a tech that allows you to connect to the PSTN for in and/or outbound calls to the PSTN
14:08.11kim_bruningD-Fender: I got to the point where I can see it's 403ing me when I try to connect
14:08.21kim_bruningbut I'm not entirely sure why
14:08.29[TK]D-FenderWhere "SIP trunk" is used to describe call/registration auth setup to use an account with a SIP provider
14:08.40[TK]D-Fenderthe term also applied to wired connections
14:08.45PhrohdohThanks, that makes a little more sense. That is quite vague though.
14:08.47[TK]D-Fenderlike analog and digital land-lines
14:09.30[TK]D-FenderIt is sometimes used in terms of talking aboutt the receiving end specifically, other times referring toa config setup on your side
14:09.38[TK]D-Fenderit's all aboutt context
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14:14.18PhrohdohWhich distro would you all suggest to host an asterisk installation? I could spin up a vm on my current machine, but I wouldn't be able to dedicate any 'real' amount of resources to that, nor can I afford another machine currently.
14:16.05PhrohdohI should have prefaced that with what I'd like to build.
14:16.20Phrohdoh(note that I am reading the book currently, too)
14:17.00kim_bruninghere's my config. am I doing something obviously wrong?  http://pastebin.com/hah3ntTm
14:17.53kim_bruningcontinues looking
14:18.07PhrohdohI asked about voicemail earlier and would like to build a system so that a user may send another user a voicemail directly. But ideally the wav would be prerecorded so the user wouldn't have to actually speak, just select a wav and a recipient. Of course the user interactions could be built into a web interface, or a command line tool, etc (which i
14:18.08Phrohdohsn't directly related to asterisk, I assume). So my understanding is that I require very little to do this.
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14:22.45kim_bruningpokes the isp
14:23.38kim_bruningD-fender: hmm, anything I can read up on or track down ? ^^;;
14:23.50[TK]D-FenderPhrohdoh, Everything depends on your needs.  the GUI distros offer basic cookie-cutter approaches to a variety of aspects of traditional closed PBX's
14:24.05kim_bruningPhrohdoh: I'm told debian not so great ;-)
14:24.30[TK]D-Fenderthis includes how you dial between phone devices (treated as "people"), processing inbound vs outbound calls, queues, IVRs , etc
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14:24.39kim_bruning(because it installs a really old asterisk)
14:24.43[TK]D-Fenderkim_bruning, your decript PACKAGES are not great
14:24.54[TK]D-Fenderkim_bruning, And they at least HAVE 11 available... but you're not using that
14:25.06kim_bruning<PROTECTED>
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14:25.42kim_bruningI'll be testing that on a test box later
14:26.11kim_bruningthough currently I'm working through auth stuff
14:28.45Phrohdoh[TK]D-Fender: Well, I understand that everything depends on my needs but that is why I have laid them out. I am probably not asking the best questions because I am completely new to dealing with a PBX/Asterisk. My day job is a health insurance platform and I'd like to integrate automated voicemail sending under certain conditions. Of course asteris
14:28.45Phrohdohk wouldn't be aware of the consumer's conditions. I'll do some more reading over the week.
14:30.32Phrohdohkim_bruning: Ha ok thanks. I'd like to not go with a cookie-cutter GUI-oriented solution because I plan to really learn the tools and eventually customize them as my vision changes, but perhaps that'll be the easiest way for me to get started.
14:31.15PhrohdohI am heading out for now though. Thank you for all the help so far!
14:31.30[TK]D-FenderPhrohdoh, " integrate automated voicemail sending under certain conditions" <- you'll want to express this call flow in proper detail to see if the GUI offers the means to your goals including this and to determin which paths might serve you best
14:38.25kim_bruninghmm, asterisk is sending local ip over net instead of wan ip <scratches head>
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14:57.44seik0Hi! Is it possible to translate DIALSTATUS and HANGUPCAUSE when we do Dial(Local/localchan-ext), and in localchan-ext we do "real" Dial(SIP/123) ?
14:58.20seik0Because now when Dial(SIP/123) returns DIALSTATUS=NOANSWER, what to do next?
14:59.10seik0in main dialplan DIALSTATUS  became CHANUNAVAIL, and HANGUPCAUSE=0
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15:03.59seik0but maybe in newer asterisk releases this behaviour has already changed
15:04.14seik0I use old 1.4 version
15:12.08kim_bruningso setting nat=no now works too?
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15:13.03kim_bruningit's gotta be something really simple, and I'm just not seeing it ^^;;;;
15:18.37[TK]D-Fenderseik0, You'd have to show us an actual call to see what's happening.
15:18.53[TK]D-Fenderseik0, And then .... you know you should not be on anything other tthan 11 or 13 by now
15:19.14[TK]D-Fenderkim_bruning> so setting nat=no now works too? <- depends WHERE
15:19.37seik0we talked about old versions at least 5 times here already =)
15:19.48seik0but this is not as simple
15:21.05[TK]D-FenderI guess it's a question of how long you can stay on an ancient insecure and unsupported platform.
15:21.10seik0many things stopped working when we tried to jump from 1.8 to 10, few days were not enough to fix all, so it was delayed till better days )
15:21.24[TK]D-FenderAnd every year you put it off the situation gets worse and stupider to be in.
15:21.46[TK]D-Fender1.8 = DOA.  10 = DOA
15:22.06[TK]D-FenderBetter days never happen
15:22.11[TK]D-FenderYou're still where you started
15:23.08seik0reality is cruel
15:23.42[TK]D-FenderWe keep saying "later".  That's not reality, that's a conscious choice not to do the work.
15:24.00[TK]D-Fendereither way... if you want any input on whatt you've got... then show it
15:24.26seik0no, it is choise of kind "there are more important things to do", and even better idea of "do not touch while it works"
15:25.30Nivexahh, I love me some "there are more important things to do". Seems like the "new shiny" always ends up being more important. The question is: when your phone system comes crashing down, will you get fired? That's how important it is.
15:26.23kim_bruningyou know what would be funny? If my sip.conf was actually perfectly in order, and the actual bug was in extensions.conf
15:27.04[TK]D-FenderAs in not dialing the correct format of # in the first place?
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15:30.42seik0Nivex, when system is crashing - it is one thing. When for new asterisk release you need new version of drivers (and it is very serious), new OS kernel (and new OS all the way) - it is another.
15:31.46seik0ok, let's stop for now
15:32.50[TK]D-Fender<[TK]D-Fender> either way... if you want any input on whatt you've got... then show it
15:34.42seik0ok, ok, i'm working
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16:03.13seik0It's "pseudo", just simplified description, because making it to work in real dialplans may take few hours of work: http://pastebin.com/yuSYgL89
16:04.22seik0is it ok?
16:07.23[TK]D-FenderYou can't be getting "noanswer" from that
16:07.39[TK]D-Fenderand waiting an hour for broken pseudo-code was a waste of everyone's time
16:12.48seik0what do you mean "I can't be getting "noanswer"" ? Because channels was answered?
16:14.45seik0mainly the question is whether is is possible to translate HANGUPCAUSE out from Local channel, and  this pseudo-dialplan is just an explanaition of what I  mean
16:15.41seik0so for first to dive into making real plans I asked if it is expected to work or not
16:16.16seik0if it is not expected, then you can always say that it may have been already fixed or smth
16:17.41seik0btw, I already found some patches that sets hangupcause for local channel, though I expect some caveats
16:19.57[TK]D-FenderYou use Hangup() to set the pass-back for this, not messing with the variable directly
16:22.56seik0it works for for IAX2 call, but doesn't for call to Local channel
16:23.29[TK]D-FenderCome back with something real to compare
16:23.40seik0ok, I'll be back then
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17:08.41igcewielingseik0: look at https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance and "core show function MASTER_CHANNEL"
17:08.54igcewielingthat might help with chan_local stuff, I don't know.
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23:52.32KattyHello my asterisk does not work at all, how to fix
23:55.54WIMPyhands Katty a light
23:55.57WIMPyBurn it
23:56.13Katty<PROTECTED>
23:57.48WIMPyWho knows. Using smoke signals might become fashionable again.

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