IRC log for #asterisk on 20141206

00:08.55*** join/#asterisk Akuma (~Akuma@apn-31-1-20-182.dynamic.gprs.plus.pl)
00:31.41snadgepeople want to trunk all kinds of shit.. its amazing.. even cisco call centre manager
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00:32.07snadgeso apparently it makes business sense to be certified with that kind of garbage ;)
00:32.52PenguinWhat does that mean, to trunk?
00:33.17snadgemaybe thats an aussie term? .. to connect a pbx up to a voip provider
00:34.03snadgeactually thats not quite correct.. its just to send and receive calls through a server
00:34.23snadgeaka trunking
00:46.26*** join/#asterisk Ankoran (~zerolegen@207-118-224-74.dyn.centurytel.net)
00:46.45Ankorancan someone recommend a good linux softphone that does both iax2 and sip?
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01:03.06[TK]D-FenderAnkoran, You'll find them on the 3rd shelf behind your, right next to the unicorns.
01:03.31Ankorangee thanks
01:10.05ChannelZ-WkI've never used one in Linux
01:12.44Ankoranit is alright i have zoiper and sflphone for linux i do not have windows machines here
01:13.20ChannelZ-Wkyeah was going to say I thought Zoiper did IAX
01:13.40Ankoranhaving issues with SIP right now anywyas
01:13.46Ankorangot 2 systems side by side
01:14.22Ankoranthey both register but 1 can try to call the other butit just rings and goes to voicemail after the specified duration
01:14.28Ankoranthe other cannot try to call
01:14.41Ankoran<PROTECTED>
01:15.46ChannelZ-Wkbad IP/bad DNS/not registered/
01:16.03Ankoranyeah i just noticed the other one is not registering but not sure why just yet
01:16.17Ankoranmaybe i had a typo in sip.conf since they are both computers right next to each other
01:17.07Ankorani might be using the wrong sip info too i am used to asterisk 1.4 been a long time since I used this
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01:19.56Ankoranwhen i try to register with the one having issues i do not even see anything in the asterisk cli
01:25.08[TK]D-FenderIf you've enabled SIP debug and still see nothing then either things aren't pointed right, or you have a blockage somewhere
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01:28.54Ankorani think i am just having an issue with that computer
01:29.10Ankoranubuntu has been giving me problems lately so i am testing using my other phone with zoiper
01:30.28Ankoranyeah my phone registered just fine now i can start to see what is going on with sip
01:32.00Ankoranthe zoiper app really sucks it crashes on both my phones
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01:50.09Ankoranok it was just bad software phone i guess i got 2 way audio working just fine with sip between 2 local systems one on linphone the other using zoiper
01:52.45Ankoranbut when i use linphone on both only 1 sided audio
02:02.14Ankorani do not seem to be getting any audio on my windows phone though from the system
02:02.57Ankoranusing the demo features i dial 500 and it connects and the CLI says its playing but i hear nothing, if i use the windows to call linux i can hear windows on linux but windows does not hear linux
02:04.22[TK]D-FenderSIP debug holds the answer....
02:04.37[TK]D-Fender(most likely)
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02:08.11Ankoranhow do i enable that?
02:09.44Ankoranok i got sip debut on all
02:09.46Ankorani made a call
02:10.09Ankorannot sure what i am looking for i see unknown control stuff
02:11.13Ankoranand on the side that i hear audio i am hearing my own audio back
02:14.07Ankoranok i fixed that somehow on this linux system my default input was monitor of output
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03:33.29ZerOlegendis it still possible to integrate google voice to asterisk?
03:34.36*** join/#asterisk blee (~blee@72.189.166.247)
03:45.44ZerOlegendin sip.conf i pasted my old sip.con from asterisk 1.4 and when i reload it says insecure very is not a valid line so what is the current format for that?
03:46.40WIMPyThe UPGRADE* documents thell you what to watch out for when upgrading.
03:47.04WIMPyAlso a look in to current samples is often a good idea.
03:47.30ZerOlegendi guess i should of done make samples
03:47.33WIMPyDoes the message no longer include the new options?
03:47.48ZerOlegendfigured i could just compile and use all my old config files
03:48.11WIMPyGreat. If you built from source, you still have the sample files there.
03:48.23ZerOlegendim not sure i even need that line i am looking at i guess an older tutorial on integrating google voice
03:48.31WIMPyMore or less, yes.
03:48.31ZerOlegendi want to use my GV for incoming calls
03:48.45ZerOlegendhoping this is not a lost cause on this project
03:48.45WIMPyThat's a SIP option.
03:48.57ZerOlegendwhat is GV?
03:49.05ZerOlegendor the insecure=very?
03:49.11ZerOlegendi already removed that deprecated line
03:49.12WIMPyShould still work, but noone knows for how long.
03:49.24WIMPyinsecure= is a SIP thing.
03:49.33ZerOlegendyeah google is quite private about their plans for GV every since they bought it. I wish grand central never sold it
03:49.39ZerOlegendoh yeah WIMPy i knew that
03:49.52WIMPyok
03:49.57ZerOlegendwas more or less looking in here to see if someone had an example syntax of what replaced it before I dug through documentation
03:50.43WIMPyWell, I gave you the files where you find it easily.
03:51.26ZerOlegendindeed and thank you
03:51.34ZerOlegendI was just clarifying my intent asking here hehe
03:52.09ZerOlegendfor the time being im just removing that line as it is most likely not needed for my implementation
03:52.14WIMPyI'm sure others remember the syntax, but they might be doing other things right now.
03:52.54ZerOlegendit is no big deal thank you for replying I appreciate it most the time I come around ask a question and its just dead silence or I get told things referring to unicorns
03:53.26WIMPyMaybe you're hanging around at the wrong times.
03:53.40WIMPyIt's generelly rather quiet here on weekends.
03:53.59ZerOlegendwell until today I have not been in here in many years
03:54.12WIMPyok
03:54.15ZerOlegendI used to use freeswitch for my stuff in the past but got tired of it
03:54.21ZerOlegendI prefer AEL over XML
03:54.45WIMPyprefers dialplan over xml.
03:54.52ZerOlegendamen to that
03:55.10ZerOlegendthe only reason i ever went over to FS in the first place was because asterisk was having issues with meetme
03:55.13ZerOlegendlong time ago
03:55.26WIMPyI couldn't get any sensible integration with real phones in FS.
03:55.27ZerOlegendit wouldnt work on a system that did not have some sort of hardware i forget what it was
03:55.46ZerOlegendso i ended up using asterisk for my dialplan and freeswitch for conferencing
03:55.48WIMPyMeetMe is history. We have ConfBridge now.
03:55.56ZerOlegendyeah i saw that
03:56.00ZerOlegendhopefully it is easy
03:56.30ZerOlegendnot that I really have a need for it these days but got some projects with some other web guys and might come in handle for calls
03:56.33WIMPyIt is. Lots to configure, but almost everything is optional.
03:56.58ZerOlegendexcellent
03:57.15ZerOlegendasterisk sure has come a long way
04:07.50WIMPy... in a long time.
04:13.55ChannelZasterisk love you long time
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04:53.54Ankoranbummer that google voice integration guide did not work for me
04:54.12Ankoranit was just to set it up using google voice callback using IPKall or callcentric
04:55.32Ankoranim reading on nerdvittles that if you have a google account previous to them getting rid of xmpp that it still works
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05:05.47lorsungcu_anyone familiar with mdadm?
05:06.43WIMPyThe manual :-)
05:07.06lorsungcu_yes :/
05:07.33lorsungcu_upgraded to centos 6.5, apparently udev does something much more different
05:08.05WIMPyudev is evil.
05:08.11lorsungcu_i am gathering
05:08.12WIMPyBut it might be the kernel itself.
05:09.46Ankoranseems a lot of people use centos for asterisk is there some specific reason for that?
05:10.18lorsungcu_this install is freepbx; distro uses it.
05:11.12Ankoranoh ok
05:11.21Ankoranfunny i am setting up freepbx right now but on my ubuntu system
05:11.50WIMPyToo many F-words...
05:11.53Ankoranthe tutorial i found for google voice integration from nerdvittles is based on freepbx so i figured i better use it
05:12.07Ankoranhaha WIMPy
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05:16.45linociscois asterisk ok with Ruby and MongoDB?
05:17.04lorsungcu_yes
05:17.05WIMPyWhere's th link?
05:17.08linociscocan asterisk DB be non-SQL DB like MongoDB?
05:18.48[TK]D-FenderWhat "Asterisk DB"?
05:19.16Ankoranas far as i know asterisk only works with sql based database engines
05:19.18[TK]D-FenderAstDB was BDB and then SQLite
05:19.37linociscoAnkoran, ok . thanks
05:19.49[TK]D-Fenderthat is a vague question
05:20.02WIMPyThe question is where or for what purpose.
05:27.39linociscothere aare many asterisk based settop box from China these days. Any idea ? any reliable brands?
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06:23.03Ankoranwell I got freepbx installed but now the web gui seems to have stuff missing :(
06:23.38WIMPy#freepbx
06:23.49Ankoranthanks
06:23.57Ankoranhopefully they are not all idle
06:36.25ChannelZkeep hoping
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06:52.32linociscohi all, newbie questions. I am not clear about CLI and NON-CLI routes. what is difference? some companies in the world said they have A-Z routes for whole sales. where did they get those routes?
06:53.45WIMPyWell, if you route calls through some random server you hacked before, Caller ID is unlikely to work.
06:57.34linociscoWIMPy, I am sorry . Does that answer my questions?
06:57.50WIMPyPartially
06:58.29WIMPyWhat else do you want to know?
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07:09.55linociscoWIMPy, what is CLI and what is NON-CLI route? I know CLI is command line interface only.
07:10.09linociscoWIMPy, where did whole saler gots those routes from whom?
07:11.14WIMPyOr CaLler Id.
07:13.07linociscoWIMPy, i didn't ask about calling ID unless CLI and NON-CLI routes are defined as Caller ID. white, grey,black any color else for routes?
07:14.02WIMPyRoutes where caller ID works or doesn't work.
07:22.24linociscoWIMPy, how did they get those numerous routes from where? Like public IP registering at ICANN? ?
07:22.57WIMPyI don't understand the question?
07:23.53linociscoWIMPy, there are many voip whole sales company they have A-Z routes for multiple destinations. Where did they get those routes to resell?
07:24.48WIMPyFrom other resellers or by making lots of contracts with lots of telcos.
07:28.24linociscoWIMPy, ok. thanks
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11:02.02wonderworldAny reccomendations for a text or ncurses based tool to debug SIP traffic in console? I really like the SIP module in wireshark GUI and am looking for something similar in textmode.
11:05.24WIMPysngrep
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11:05.55WIMPyAbsolutely 0 documentation, but it looks really nice.
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12:19.04wonderworldi want to compile asterisk 13 with pjsip support. do i need to have to care about the pjsip version used or will the latest release work?
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19:00.40popershello guys, im here because im trying to make my asterisk work with websockets but im getting an error of Rejecting secure audio stream without encryption details: audio 58306 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
19:01.01popersdoes anybody how to solve this problem?
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21:24.54WIMPyAm I missing somethingn or is 'manager set debug on' broken? I don't see anyting.
21:25.29ChannelZI don't think it shows you raw traffic
21:26.05ChannelZso you only get output when commands acctually happen.
21:26.20WIMPyNope. I get absolutely nothing.
21:27.03ChannelZtries
21:27.03WIMPyIs there some extra thing in logger.conf I overlooked or something?
21:32.52ChannelZhmmm.. something does seem weird.  It won't even tell me debug is on or off when I switch it.
21:33.32WIMPyOk, so you see as much as I do.
21:33.42WIMPyOr as little, rather.
21:35.19ChannelZyeah.  Looking at the code trying to figure out why it's not even showing me the state.
21:39.52ChannelZIt doesn't actually look like manager puts out anything..
21:40.59WIMPyThe only ting I see with normal debug is unref_mansession.
21:41.29ChannelZActually.. I'm not even sure where the main manager thread is that listens
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21:47.18WIMPyLooks like debug 9 helps to get manager events.
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21:47.29ChannelZoh I see. "manager set debug" returns the status, but it doesn't say anything when you say "on" or "off"
21:49.03ChannelZyeah just doesn't look like there is a system in place for manager debug like it works for say AGI debug
21:50.15ChannelZprobably easier to just wireshark it than dig through normal asterisk debug
21:50.25ChannelZ(that high)
21:50.50WIMPyTrouble is I need it in sync with other debug.
21:51.05WIMPyBut verbose and debug each at 9 should be ok.
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21:56.44voronaHi all!
21:58.17voronaI need help with exotic configuration asterisk =)
21:58.23PenguinROFLs for a minute
21:58.43PenguinI absolutely love my torture system for telemarketers.
21:59.01PenguinI just listened to one where the guy finally caught on that he was talking to a machine.
21:59.11PenguinI heard him sigh right as he disconnected.
22:00.58PenguinI guess he suspected something was up, because he stopped talking for a minute.
22:01.27PenguinAnd the machine would still talk after a while of silence.
22:01.40PenguinHe sighed and hung up.  I win again!
22:02.02WIMPyHow long did you hold him?
22:02.17PenguinThis one was 2.483333min
22:02.34PenguinI had one woman who stayed on for nearly 5 entire minutes.
22:03.26PenguinThat's a long time to talk to a machine that only has about six unique phrases.
22:03.35voronaWho can help me?
22:03.47Penguinvorona: Did you ever bother to ask a question?
22:03.48WIMPyvorona: noone.
22:03.53PenguinI sure didn't see a question.
22:03.58WIMPyNot unless you ask something.
22:05.08PenguinIt's like when someone asks me if I can do a favor for them.
22:05.21PenguinHow do I know if I do it if I don't know what it is that needs done?!
22:05.32voronaOk, I need change domain on INVITE header, but in sip.conf any changes didn't change this header =(
22:05.50PenguinWhat did you change in sip.conf to try to get it to change?
22:06.14voronafromdomain, domain, realm
22:06.46PenguinDo you need it to change on just one peer that talks to one specific place?
22:07.17voronaJust one peer
22:08.01PenguinAnd fromdomain didn't change what needed to be changed?
22:08.49voronaYYes, it changes just 'contact' and 'from'
22:08.50[TK]D-FenderOnly changes the "From:"
22:09.22voronaBut INVITE is <exten>@<hostname>
22:09.26Penguinand only as a client.
22:09.46[TK]D-Fender"INVITE" I suspect might require changing the host, and setting an outboundproxy to ensure it actually goes to the right place
22:11.06vorona[TK]D-Fender, I tried turn off srvlookup and add record to hosts file for domain with needed host ip address
22:11.21voronain that situation all works
22:11.27Penguinfromdomain only sets the From: when you send a call to the other side.
22:11.35voronabut I need srvlookup
22:12.03WIMPySo you need a domain that's different from the host= ?
22:12.14voronaWIMPy, yes
22:12.33WIMPyThen use outboundproxy.
22:13.23voronaWhat I must enter in outbondproxy domain or host?
22:13.48WIMPyWhat you really want to connect to. And put the domain in host= .
22:14.36voronaI tried it, but this configuration didn't works. sip peer was wrong address
22:15.01WIMPyYou probably need outbountproxy=...,force
22:15.25voronaHmmm, one second
22:15.39voronaWhat enter to host?
22:15.40PenguinYou can put the SRV name in the outbound proxy.
22:16.05voronaSame outbondproxy?
22:17.36WIMPyoutboundproxy= the host you want to connect to. And put the domain in host= .
22:18.30voronahost=sip.beeline.ru outbondproxy=msk.sip.beeline.ru,force
22:18.36voronanot working
22:18.46voronaip of peer is ip of domain
22:19.25voronaand not reacheble
22:19.41WIMPyYou are a little vague in your desctription.
22:20.46voronaSpecifically?
22:21.58voronaserver, witch listen - msk.sip.beeline.ru
22:22.07voronadomain is sip.beeline.ru
22:23.20voronaprovider whants to get INVITE as INVITE <number@sip.beeline.ru>, but my aster send <number@msk.sip.beeline.ru>
22:23.34WIMPythen your example looks good.
22:23.39voronaand provider answer with 403 code
22:23.46WIMPyTry to restart asterisk.
22:24.03WIMPysip reload doesn't always update everything.
22:24.11Penguinfromdomain changes the From: part of the INVITE.
22:24.27voronaI use module reload
22:24.49WIMPyNot sure that's any better.
22:24.53WIMPymodule unload/load does work however.
22:25.14voronaRestarted, trying to call
22:25.52voronaNot working
22:26.39voronaPenguin, I need change INVITE, with from and contact no problems
22:28.48voronaIs this option in sip.conf =)
22:30.11voronaI need option like "todomain"
22:30.46PenguinI believe that option is "host"
22:30.56WIMPyIndeed
22:31.27Penguinfromdomain is used to change the "From:" field.
22:31.30Penguinnothing else.
22:32.23PenguinIf you want to send calls to sip.beeline.ru, set host=sip.beeline.ru
22:32.39voronaPenguin, I think so, but this option changes realy address of peer, what I'm not needed
22:32.58Penguinhost=sip.beeline.ru   <-------
22:33.00PenguinDid you do it?
22:34.00voronayes, with this option peer unreacheble, sip.beeline.ru and msk.sip.beeline.ru has different addreses
22:34.28PenguinSet your outboundproxy to the address you want to connect to.
22:34.47voronaPenguin it has no effects
22:34.53PenguinShow me.
22:34.54Penguin~pb
22:34.54infobotA "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
22:37.06voronahttp://pastebin.com/LV2g4NsN
22:37.26voronaOpenWrtMSK*CLI> sip show peers
22:37.27voronaName/username              Host            Dyn Nat ACL Port     Status
22:37.27vorona201/201                    172.16.117.33    D          5062     OK (65 ms)
22:37.27voronab_trunk/vorona@sip.beelin  195.239.253.100      N      5060     UNREACHABLE
22:37.39WIMPyWhere is the ",force"?
22:38.01WIMPy>>You probably need outbountproxy=...,force
22:38.11voronaI chenged it
22:38.24voronaIt had no effects
22:38.39WIMPyDid you restart?
22:38.57voronayes
22:39.18PenguinI don't see the defaultuser specified and I don't see where you have a secret.
22:39.29ChannelZit's a secret
22:39.36voronaYes =)
22:39.41Penguincanreinvite and directmedia?
22:39.42PenguinRemove canreinvite
22:39.50Penguindirectmedia is the correct name for that.
22:40.44PenguinAnd I can't read Russian, so I can't read their web site to see if they have an asterisk configuration example.
22:41.01PenguinAND...
22:41.11PenguinIt's "outboundproxy"
22:41.21PenguinYou have to spell it correctly for it to work.
22:41.41WIMPyThat should definitely help.
22:42.05WIMPyIt's always good if someone can actually read :-)
22:42.20PenguinI recommend copy/paste.
22:42.41PenguinThat way you can blame it on the person instructing if you get it wrong.
22:42.58Penguin"I didn't misspell it.  I copied what you told me!"
22:44.15*** join/#asterisk CeBe (~CeBe@port-92-200-92-103.dynamic.qsc.de)
22:45.58WIMPyWouldn't it be great if chan_sip told you about crap in your configs?
22:47.36filechan_pjsip does, and will refuse to load a broken configuration
22:48.56voronachan_sip passed my config, changes in "outboundproxy" has no results in peer ip now ip of domain
22:49.41Penguinoutboundproxy is the name or address or the host you'll send your requests to.
22:50.06voronaI know, but it change nothing
22:50.50ChannelZyour provider is retarded
22:51.15voronaChannelZ, yeah, I think so
22:51.22WIMPyThose two hosts resolve to the same IP anyway.
22:51.57voronaprovider's support cant tell me even peer used tcp
22:52.37voronaWIMPy, whitch hosts? sip... and msk...?
22:52.41Penguinwimpy: I didn't even check that.  He said they were different, and I believed him.
22:53.16*** join/#asterisk posixninja (~posixninj@ec2-54-71-138-140.us-west-2.compute.amazonaws.com)
22:53.59PenguinAnd they don't seem to have any SRV records either.
22:54.20WIMPyIsn't it great that we check different things?
22:56.17voronahttp://pastebin.com/w8NdqHTd
22:56.42PenguinSince those two names resolve to the same IP, there's no sense in setting outboundproxy since host will take care of it.
22:57.11voronahttp://pastebin.com/LiMDvDi1
22:57.27PenguinThat's an internal DNS server.
22:58.01WIMPyUgly
22:58.11PenguinYou can put any information you want into an internal DNS server.  Including invalid information.
22:58.26voronaThis my router, whith forwards all request to provider's DNS
22:58.54voronaall information from provider's DNS
23:00.24voronaWIMPy, agree
23:00.24Penguinhttp://pastebin.com/3KwC9YmN
23:01.19voronaPenguin, nice
23:02.08PenguinI tried that from my own location as well as a server in Pakistan.  The results are the same: no SRV records for sip.beeline.ru
23:02.33voronaone moment, i check my dns
23:03.13ChannelZkqfp9
23:03.21Penguin?
23:03.46ChannelZbollocks.
23:04.08ChannelZthere went half a password
23:04.16voronahttp://pastebin.com/93BurMx1
23:04.30voronaThis my dns
23:04.51PenguinI'll check those for the SRV records.
23:05.43Penguin_sip._udp.sip.beeline.ru. 8501  IN      SRV     20 0 5060 global.sip.beeline.ru.
23:06.36voronaPenguin, exactly
23:07.49Penguinglobal.sip.beeline.ru.  2340    IN      A       195.239.253.100
23:09.36Penguinmsk.sip.beeline.ru.     4887    IN      A       10.25.0.50
23:09.45voronayes
23:09.55PenguinDo not use msk.sip.beeline.ru for anything unless you are on the same LAN segment as that host.
23:10.06PenguinIt has a private address, so do not use that hostname.
23:10.29WIMPyfears that's the idea...
23:10.54voronaPenguin, do you mean, that I must enter ip in host?
23:10.56WIMPyDo you have a route there?
23:11.17PenguinInteresting question.
23:11.37voronahttp://pastebin.com/dP6719Ba
23:12.24PenguinAre you on the same LAN as the ITSP host?
23:12.53voronaPenguin, no, but host is reacheble
23:13.01WIMPyA little hard to read, but yes, it looks like you're supposed to use private IPs.
23:13.11Penguincan't read Russian.
23:13.39WIMPyI don't even have a font to display it.
23:13.46WIMPyNot that it would make any difference.
23:13.55voronaPenguin, sorry, http://pastebin.com/QYEFxDf9
23:14.18Penguinthat's the same thing.
23:15.20PenguinI'm more interested in why 10.0.0.0 is reachable.
23:15.53WIMPyWhy not?
23:16.09voronaPenguin, on my router I hame 2 connection: IPoE and L2TP
23:16.34PenguinYou have a VPN to beeline.ru?
23:16.40voronaIPoE 10.93.16.9 and L2TP - external
23:17.12voronaPenguin, no, I have VPN FROM beeline.ru to the internet
23:18.00voronahttp://pastebin.com/16bpW7W5
23:18.34WIMPyOuch
23:18.45voronaWIMPy, yeah =)
23:19.18PenguinThis doesn't make much sense to me.
23:21.21voronaStrange, but X-lite with same options sent INVITE with domain, not host
23:21.39WIMPyNow I know why the russians try to hack their way in to our servers. They are just too frustrated by their ISPs/ITSPs. Quite understandably.
23:21.48scvlol
23:21.49*** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net)
23:21.58ChannelZhaha for serious
23:22.21PenguinI'm surprised they can reach your servers to attempt cracking them open.
23:24.24voronaMany D-link routers with RU suffix even has special 'Russian PPTP' and 'Russian L2TP' connections =)
23:24.46scvsimilar situation in austria from what i understand
23:25.29WIMPyMy provider also uses private IPs for VOIP.
23:26.17voronaWIMPy, but< I think, your provider not uses public fqdn for private IP =)
23:26.19PenguinAre you on the Germany IntraNet?
23:26.37WIMPyThey do
23:26.56WIMPyNo. It's independant of the internet connection.
23:27.17WIMPyAnd it would be a spanish intranet :-)
23:27.35voronaWIMPy my DNS undependant of L2TP too
23:27.52vorona10.x.x.x reacheble without it
23:28.04PenguinFor asterisk, would you put the public hostname/IP into the outboundproxy and the private hostname/IP into the host setting?
23:28.34WIMPyThe other way round.
23:29.16voronaPenguin, you mean host=msk.sip.beeline.ru outboundproxy=sip.beeline.ru ?
23:29.27WIMPyBut as the private IPs are only resolvable on an DNS that's on a private IP as well, I have to use IPs there.
23:29.44PenguinI can't make any sense out of the beeline stuff, so I'm not even going to try.
23:30.06*** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil)
23:30.13WIMPyI'm pretty sure you wouldn't have a chance anyway.
23:32.10Penguinrolls over
23:32.28voronaINVITE sip:8xxxxxxxxxx@msk.sip.beeline.ru
23:32.29vorona(
23:34.13scvnoot noot
23:34.52PenguinYou want it to say not msk.sip but just sip, right?
23:35.29voronayes
23:35.36PenguinDo you have a fromuser= value in sip.conf?
23:37.17voronahttp://pastebin.com/kT8cN8uC
23:42.47voronais there any way to cancel srvlookup on peer, but not global?
23:43.18PenguinI guess you set a host name that doesn't have an SRV record.
23:44.14voronaIt has SRV
23:44.58voronahttp://pastebin.com/dagBmCas
23:47.19*** join/#asterisk moke (~moke@unaffiliated/moke)
23:49.03voronaNow in Msk 2:48 AM I very want to sleep...
23:49.12voronaThanks all for help
23:49.23voronaI'll continue it tomorrow
23:50.27voronabye all!

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