00:08.55 | *** join/#asterisk Akuma (~Akuma@apn-31-1-20-182.dynamic.gprs.plus.pl) |
00:31.41 | snadge | people want to trunk all kinds of shit.. its amazing.. even cisco call centre manager |
00:31.52 | *** join/#asterisk troyt (~troyt@2601:7:6202:211:44dd:acff:fe85:9c8e) |
00:32.07 | snadge | so apparently it makes business sense to be certified with that kind of garbage ;) |
00:32.52 | Penguin | What does that mean, to trunk? |
00:33.17 | snadge | maybe thats an aussie term? .. to connect a pbx up to a voip provider |
00:34.03 | snadge | actually thats not quite correct.. its just to send and receive calls through a server |
00:34.23 | snadge | aka trunking |
00:46.26 | *** join/#asterisk Ankoran (~zerolegen@207-118-224-74.dyn.centurytel.net) |
00:46.45 | Ankoran | can someone recommend a good linux softphone that does both iax2 and sip? |
00:54.14 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
01:00.49 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
01:01.14 | *** join/#asterisk jzaw (~jzaw@2001:8b0:7:0:5054:ff:fe8e:3b24) |
01:03.06 | [TK]D-Fender | Ankoran, You'll find them on the 3rd shelf behind your, right next to the unicorns. |
01:03.31 | Ankoran | gee thanks |
01:10.05 | ChannelZ-Wk | I've never used one in Linux |
01:12.44 | Ankoran | it is alright i have zoiper and sflphone for linux i do not have windows machines here |
01:13.20 | ChannelZ-Wk | yeah was going to say I thought Zoiper did IAX |
01:13.40 | Ankoran | having issues with SIP right now anywyas |
01:13.46 | Ankoran | got 2 systems side by side |
01:14.22 | Ankoran | they both register but 1 can try to call the other butit just rings and goes to voicemail after the specified duration |
01:14.28 | Ankoran | the other cannot try to call |
01:14.41 | Ankoran | <PROTECTED> |
01:15.46 | ChannelZ-Wk | bad IP/bad DNS/not registered/ |
01:16.03 | Ankoran | yeah i just noticed the other one is not registering but not sure why just yet |
01:16.17 | Ankoran | maybe i had a typo in sip.conf since they are both computers right next to each other |
01:17.07 | Ankoran | i might be using the wrong sip info too i am used to asterisk 1.4 been a long time since I used this |
01:18.27 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
01:19.56 | Ankoran | when i try to register with the one having issues i do not even see anything in the asterisk cli |
01:25.08 | [TK]D-Fender | If you've enabled SIP debug and still see nothing then either things aren't pointed right, or you have a blockage somewhere |
01:28.33 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
01:28.54 | Ankoran | i think i am just having an issue with that computer |
01:29.10 | Ankoran | ubuntu has been giving me problems lately so i am testing using my other phone with zoiper |
01:30.28 | Ankoran | yeah my phone registered just fine now i can start to see what is going on with sip |
01:32.00 | Ankoran | the zoiper app really sucks it crashes on both my phones |
01:42.12 | *** join/#asterisk Milos_ (~Milos@pdpc/supporter/student/milos) |
01:49.35 | *** join/#asterisk Ankoran (~zerolegen@207-118-224-74.dyn.centurytel.net) |
01:50.09 | Ankoran | ok 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 |
01:52.45 | Ankoran | but when i use linphone on both only 1 sided audio |
02:02.14 | Ankoran | i do not seem to be getting any audio on my windows phone though from the system |
02:02.57 | Ankoran | using 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-Fender | SIP debug holds the answer.... |
02:04.37 | [TK]D-Fender | (most likely) |
02:06.16 | *** join/#asterisk u0m3 (~u0m3@109.96.153.17) |
02:08.11 | Ankoran | how do i enable that? |
02:09.44 | Ankoran | ok i got sip debut on all |
02:09.46 | Ankoran | i made a call |
02:10.09 | Ankoran | not sure what i am looking for i see unknown control stuff |
02:11.13 | Ankoran | and on the side that i hear audio i am hearing my own audio back |
02:14.07 | Ankoran | ok i fixed that somehow on this linux system my default input was monitor of output |
02:18.06 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
02:39.11 | *** join/#asterisk lorsungcu_ (~lorsungcu@216.84.98.129) |
02:54.14 | *** join/#asterisk Valduare (~Valduare@108-198-116-80.lightspeed.mdsnwi.sbcglobal.net) |
02:56.49 | *** join/#asterisk lorsungcu_ (~lorsungcu@216.84.98.129) |
02:58.36 | *** join/#asterisk areski (~areski@80.174.128.97.dyn.user.ono.com) |
03:01.50 | *** join/#asterisk frek818 (~frek818@172.56.30.119) |
03:24.41 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
03:31.40 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
03:33.15 | *** join/#asterisk ZerOlegend (~ZerOlegen@207-118-224-74.dyn.centurytel.net) |
03:33.29 | ZerOlegend | is it still possible to integrate google voice to asterisk? |
03:34.36 | *** join/#asterisk blee (~blee@72.189.166.247) |
03:45.44 | ZerOlegend | in 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? |
03:46.40 | WIMPy | The UPGRADE* documents thell you what to watch out for when upgrading. |
03:47.04 | WIMPy | Also a look in to current samples is often a good idea. |
03:47.30 | ZerOlegend | i guess i should of done make samples |
03:47.33 | WIMPy | Does the message no longer include the new options? |
03:47.48 | ZerOlegend | figured i could just compile and use all my old config files |
03:48.11 | WIMPy | Great. If you built from source, you still have the sample files there. |
03:48.23 | ZerOlegend | im not sure i even need that line i am looking at i guess an older tutorial on integrating google voice |
03:48.31 | WIMPy | More or less, yes. |
03:48.31 | ZerOlegend | i want to use my GV for incoming calls |
03:48.45 | ZerOlegend | hoping this is not a lost cause on this project |
03:48.45 | WIMPy | That's a SIP option. |
03:48.57 | ZerOlegend | what is GV? |
03:49.05 | ZerOlegend | or the insecure=very? |
03:49.11 | ZerOlegend | i already removed that deprecated line |
03:49.12 | WIMPy | Should still work, but noone knows for how long. |
03:49.24 | WIMPy | insecure= is a SIP thing. |
03:49.33 | ZerOlegend | yeah google is quite private about their plans for GV every since they bought it. I wish grand central never sold it |
03:49.39 | ZerOlegend | oh yeah WIMPy i knew that |
03:49.52 | WIMPy | ok |
03:49.57 | ZerOlegend | was more or less looking in here to see if someone had an example syntax of what replaced it before I dug through documentation |
03:50.43 | WIMPy | Well, I gave you the files where you find it easily. |
03:51.26 | ZerOlegend | indeed and thank you |
03:51.34 | ZerOlegend | I was just clarifying my intent asking here hehe |
03:52.09 | ZerOlegend | for the time being im just removing that line as it is most likely not needed for my implementation |
03:52.14 | WIMPy | I'm sure others remember the syntax, but they might be doing other things right now. |
03:52.54 | ZerOlegend | it 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 |
03:53.26 | WIMPy | Maybe you're hanging around at the wrong times. |
03:53.40 | WIMPy | It's generelly rather quiet here on weekends. |
03:53.59 | ZerOlegend | well until today I have not been in here in many years |
03:54.12 | WIMPy | ok |
03:54.15 | ZerOlegend | I used to use freeswitch for my stuff in the past but got tired of it |
03:54.21 | ZerOlegend | I prefer AEL over XML |
03:54.45 | WIMPy | prefers dialplan over xml. |
03:54.52 | ZerOlegend | amen to that |
03:55.10 | ZerOlegend | the only reason i ever went over to FS in the first place was because asterisk was having issues with meetme |
03:55.13 | ZerOlegend | long time ago |
03:55.26 | WIMPy | I couldn't get any sensible integration with real phones in FS. |
03:55.27 | ZerOlegend | it wouldnt work on a system that did not have some sort of hardware i forget what it was |
03:55.46 | ZerOlegend | so i ended up using asterisk for my dialplan and freeswitch for conferencing |
03:55.48 | WIMPy | MeetMe is history. We have ConfBridge now. |
03:55.56 | ZerOlegend | yeah i saw that |
03:56.00 | ZerOlegend | hopefully it is easy |
03:56.30 | ZerOlegend | not 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 |
03:56.33 | WIMPy | It is. Lots to configure, but almost everything is optional. |
03:56.58 | ZerOlegend | excellent |
03:57.15 | ZerOlegend | asterisk sure has come a long way |
04:07.50 | WIMPy | ... in a long time. |
04:13.55 | ChannelZ | asterisk love you long time |
04:34.15 | *** join/#asterisk frek818 (~frek818@172.56.30.119) |
04:53.54 | Ankoran | bummer that google voice integration guide did not work for me |
04:54.12 | Ankoran | it was just to set it up using google voice callback using IPKall or callcentric |
04:55.32 | Ankoran | im reading on nerdvittles that if you have a google account previous to them getting rid of xmpp that it still works |
05:03.57 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
05:05.47 | lorsungcu_ | anyone familiar with mdadm? |
05:06.43 | WIMPy | The manual :-) |
05:07.06 | lorsungcu_ | yes :/ |
05:07.33 | lorsungcu_ | upgraded to centos 6.5, apparently udev does something much more different |
05:08.05 | WIMPy | udev is evil. |
05:08.11 | lorsungcu_ | i am gathering |
05:08.12 | WIMPy | But it might be the kernel itself. |
05:09.46 | Ankoran | seems a lot of people use centos for asterisk is there some specific reason for that? |
05:10.18 | lorsungcu_ | this install is freepbx; distro uses it. |
05:11.12 | Ankoran | oh ok |
05:11.21 | Ankoran | funny i am setting up freepbx right now but on my ubuntu system |
05:11.50 | WIMPy | Too many F-words... |
05:11.53 | Ankoran | the tutorial i found for google voice integration from nerdvittles is based on freepbx so i figured i better use it |
05:12.07 | Ankoran | haha WIMPy |
05:13.04 | *** join/#asterisk linocisco (~linocisco@61.4.76.65) |
05:13.54 | *** join/#asterisk linocisco (~linocisco@61.4.76.65) |
05:16.45 | linocisco | is asterisk ok with Ruby and MongoDB? |
05:17.04 | lorsungcu_ | yes |
05:17.05 | WIMPy | Where's th link? |
05:17.08 | linocisco | can asterisk DB be non-SQL DB like MongoDB? |
05:18.48 | [TK]D-Fender | What "Asterisk DB"? |
05:19.16 | Ankoran | as far as i know asterisk only works with sql based database engines |
05:19.18 | [TK]D-Fender | AstDB was BDB and then SQLite |
05:19.37 | linocisco | Ankoran, ok . thanks |
05:19.49 | [TK]D-Fender | that is a vague question |
05:20.02 | WIMPy | The question is where or for what purpose. |
05:27.39 | linocisco | there aare many asterisk based settop box from China these days. Any idea ? any reliable brands? |
05:31.56 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
05:37.21 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
06:22.41 | *** join/#asterisk Zer0legend (~zerolegen@207-118-224-74.dyn.centurytel.net) |
06:23.03 | Ankoran | well I got freepbx installed but now the web gui seems to have stuff missing :( |
06:23.38 | WIMPy | #freepbx |
06:23.49 | Ankoran | thanks |
06:23.57 | Ankoran | hopefully they are not all idle |
06:36.25 | ChannelZ | keep hoping |
06:51.07 | *** join/#asterisk linocisco (~linocisco@61.4.76.65) |
06:52.32 | linocisco | hi 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? |
06:53.45 | WIMPy | Well, if you route calls through some random server you hacked before, Caller ID is unlikely to work. |
06:57.34 | linocisco | WIMPy, I am sorry . Does that answer my questions? |
06:57.50 | WIMPy | Partially |
06:58.29 | WIMPy | What else do you want to know? |
07:02.42 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
07:09.55 | linocisco | WIMPy, what is CLI and what is NON-CLI route? I know CLI is command line interface only. |
07:10.09 | linocisco | WIMPy, where did whole saler gots those routes from whom? |
07:11.14 | WIMPy | Or CaLler Id. |
07:13.07 | linocisco | WIMPy, 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? |
07:14.02 | WIMPy | Routes where caller ID works or doesn't work. |
07:22.24 | linocisco | WIMPy, how did they get those numerous routes from where? Like public IP registering at ICANN? ? |
07:22.57 | WIMPy | I don't understand the question? |
07:23.53 | linocisco | WIMPy, there are many voip whole sales company they have A-Z routes for multiple destinations. Where did they get those routes to resell? |
07:24.48 | WIMPy | From other resellers or by making lots of contracts with lots of telcos. |
07:28.24 | linocisco | WIMPy, ok. thanks |
07:51.38 | *** join/#asterisk ChannelZ (channelz@burner.com) |
08:31.03 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
08:40.09 | *** join/#asterisk riess82 (~riessma@62-47-136-212.adsl.highway.telekom.at) |
08:41.51 | *** join/#asterisk drmessano^ (~nonya@pdpc/supporter/active/drmessano) |
08:45.26 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-iqqpjnxcjhwxatlo) |
09:02.08 | *** join/#asterisk Davlefou (~davlefou@unaffiliated/davlefou) |
09:02.19 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
09:20.06 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
09:31.12 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
09:40.38 | *** join/#asterisk frek818 (~frek818@172.56.30.119) |
10:15.53 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
10:28.02 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
10:31.12 | *** join/#asterisk frek818 (~frek818@172.56.30.119) |
10:48.11 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
11:01.31 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
11:02.02 | wonderworld | Any 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. |
11:05.24 | WIMPy | sngrep |
11:05.39 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
11:05.55 | WIMPy | Absolutely 0 documentation, but it looks really nice. |
11:10.20 | *** join/#asterisk wolrah (~wolrah@24.239.210.140) |
11:10.29 | *** join/#asterisk frek818 (~frek818@172.56.30.119) |
11:47.33 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
12:08.54 | *** join/#asterisk CeBe (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
12:19.04 | wonderworld | i 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? |
12:33.07 | *** join/#asterisk areski (~areski@80.174.128.97.dyn.user.ono.com) |
13:00.50 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
13:07.22 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
13:30.39 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
13:30.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
14:26.12 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:8219:34ff:fecf:17f0) |
14:28.01 | *** join/#asterisk CeBe (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
14:48.50 | *** join/#asterisk zamba (marius@flage.org) |
14:50.43 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
14:51.38 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
15:31.55 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
15:39.31 | *** join/#asterisk riess82 (~riessma@62-47-136-212.adsl.highway.telekom.at) |
15:46.39 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
16:08.48 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
16:21.37 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
16:46.37 | *** join/#asterisk Valduare (~Valduare@108-198-116-80.lightspeed.mdsnwi.sbcglobal.net) |
17:06.49 | *** join/#asterisk lorsungcu_ (~lorsungcu@71.39.97.57) |
17:14.06 | *** join/#asterisk jetlag (~jetlag@pool-71-168-194-254.cmdnnj.east.verizon.net) |
17:38.51 | *** join/#asterisk michael_work_ (~michael_w@bzq-82-168-31-134.red.bezeqint.net) |
17:58.03 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
17:59.13 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
18:06.34 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
18:32.58 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
18:35.47 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
18:36.37 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:8219:34ff:fecf:17f0) |
18:51.05 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
18:51.33 | *** join/#asterisk spditner (~simon@76-10-131-126.dsl.teksavvy.com) |
18:57.15 | *** join/#asterisk popers (bdbf4a8b@gateway/web/freenode/ip.189.191.74.139) |
19:00.40 | popers | hello 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 |
19:01.01 | popers | does anybody how to solve this problem? |
19:03.44 | *** join/#asterisk ShapeShifter499 (~Raansu@unaffiliated/shapeshifter499) |
20:07.14 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
20:49.16 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
20:51.41 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
20:53.19 | *** join/#asterisk Valduare (~Valduare@108-198-116-80.lightspeed.mdsnwi.sbcglobal.net) |
20:58.05 | *** join/#asterisk areski (~areski@80.174.128.97.dyn.user.ono.com) |
21:17.47 | *** join/#asterisk dxd828 (~dxd828@dsl-dynamic-77-44-45-38.interdsl.co.uk) |
21:19.17 | *** join/#asterisk tparcina (~tomo@cpe-109-60-1-42.st3.cable.xnet.hr) |
21:20.25 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
21:24.54 | WIMPy | Am I missing somethingn or is 'manager set debug on' broken? I don't see anyting. |
21:25.29 | ChannelZ | I don't think it shows you raw traffic |
21:26.05 | ChannelZ | so you only get output when commands acctually happen. |
21:26.20 | WIMPy | Nope. I get absolutely nothing. |
21:27.03 | ChannelZ | tries |
21:27.03 | WIMPy | Is there some extra thing in logger.conf I overlooked or something? |
21:32.52 | ChannelZ | hmmm.. something does seem weird. It won't even tell me debug is on or off when I switch it. |
21:33.32 | WIMPy | Ok, so you see as much as I do. |
21:33.42 | WIMPy | Or as little, rather. |
21:35.19 | ChannelZ | yeah. Looking at the code trying to figure out why it's not even showing me the state. |
21:39.52 | ChannelZ | It doesn't actually look like manager puts out anything.. |
21:40.59 | WIMPy | The only ting I see with normal debug is unref_mansession. |
21:41.29 | ChannelZ | Actually.. I'm not even sure where the main manager thread is that listens |
21:41.45 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
21:41.46 | *** mode/#asterisk [+o Qwell] by ChanServ |
21:42.37 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-103.dynamic.qsc.de) |
21:47.18 | WIMPy | Looks like debug 9 helps to get manager events. |
21:47.28 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-kpbxrbgcrfppkxuz) |
21:47.29 | ChannelZ | oh I see. "manager set debug" returns the status, but it doesn't say anything when you say "on" or "off" |
21:49.03 | ChannelZ | yeah just doesn't look like there is a system in place for manager debug like it works for say AGI debug |
21:50.15 | ChannelZ | probably easier to just wireshark it than dig through normal asterisk debug |
21:50.25 | ChannelZ | (that high) |
21:50.50 | WIMPy | Trouble is I need it in sync with other debug. |
21:51.05 | WIMPy | But verbose and debug each at 9 should be ok. |
21:55.26 | *** join/#asterisk vorona (~admin@93-80-32-223.broadband.corbina.ru) |
21:56.44 | vorona | Hi all! |
21:58.17 | vorona | I need help with exotic configuration asterisk =) |
21:58.23 | Penguin | ROFLs for a minute |
21:58.43 | Penguin | I absolutely love my torture system for telemarketers. |
21:59.01 | Penguin | I just listened to one where the guy finally caught on that he was talking to a machine. |
21:59.11 | Penguin | I heard him sigh right as he disconnected. |
22:00.58 | Penguin | I guess he suspected something was up, because he stopped talking for a minute. |
22:01.27 | Penguin | And the machine would still talk after a while of silence. |
22:01.40 | Penguin | He sighed and hung up. I win again! |
22:02.02 | WIMPy | How long did you hold him? |
22:02.17 | Penguin | This one was 2.483333min |
22:02.34 | Penguin | I had one woman who stayed on for nearly 5 entire minutes. |
22:03.26 | Penguin | That's a long time to talk to a machine that only has about six unique phrases. |
22:03.35 | vorona | Who can help me? |
22:03.47 | Penguin | vorona: Did you ever bother to ask a question? |
22:03.48 | WIMPy | vorona: noone. |
22:03.53 | Penguin | I sure didn't see a question. |
22:03.58 | WIMPy | Not unless you ask something. |
22:05.08 | Penguin | It's like when someone asks me if I can do a favor for them. |
22:05.21 | Penguin | How do I know if I do it if I don't know what it is that needs done?! |
22:05.32 | vorona | Ok, I need change domain on INVITE header, but in sip.conf any changes didn't change this header =( |
22:05.50 | Penguin | What did you change in sip.conf to try to get it to change? |
22:06.14 | vorona | fromdomain, domain, realm |
22:06.46 | Penguin | Do you need it to change on just one peer that talks to one specific place? |
22:07.17 | vorona | Just one peer |
22:08.01 | Penguin | And fromdomain didn't change what needed to be changed? |
22:08.49 | vorona | YYes, it changes just 'contact' and 'from' |
22:08.50 | [TK]D-Fender | Only changes the "From:" |
22:09.22 | vorona | But INVITE is <exten>@<hostname> |
22:09.26 | Penguin | and 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 |
22:11.06 | vorona | [TK]D-Fender, I tried turn off srvlookup and add record to hosts file for domain with needed host ip address |
22:11.21 | vorona | in that situation all works |
22:11.27 | Penguin | fromdomain only sets the From: when you send a call to the other side. |
22:11.35 | vorona | but I need srvlookup |
22:12.03 | WIMPy | So you need a domain that's different from the host= ? |
22:12.14 | vorona | WIMPy, yes |
22:12.33 | WIMPy | Then use outboundproxy. |
22:13.23 | vorona | What I must enter in outbondproxy domain or host? |
22:13.48 | WIMPy | What you really want to connect to. And put the domain in host= . |
22:14.36 | vorona | I tried it, but this configuration didn't works. sip peer was wrong address |
22:15.01 | WIMPy | You probably need outbountproxy=...,force |
22:15.25 | vorona | Hmmm, one second |
22:15.39 | vorona | What enter to host? |
22:15.40 | Penguin | You can put the SRV name in the outbound proxy. |
22:16.05 | vorona | Same outbondproxy? |
22:17.36 | WIMPy | outboundproxy= the host you want to connect to. And put the domain in host= . |
22:18.30 | vorona | host=sip.beeline.ru outbondproxy=msk.sip.beeline.ru,force |
22:18.36 | vorona | not working |
22:18.46 | vorona | ip of peer is ip of domain |
22:19.25 | vorona | and not reacheble |
22:19.41 | WIMPy | You are a little vague in your desctription. |
22:20.46 | vorona | Specifically? |
22:21.58 | vorona | server, witch listen - msk.sip.beeline.ru |
22:22.07 | vorona | domain is sip.beeline.ru |
22:23.20 | vorona | provider whants to get INVITE as INVITE <number@sip.beeline.ru>, but my aster send <number@msk.sip.beeline.ru> |
22:23.34 | WIMPy | then your example looks good. |
22:23.39 | vorona | and provider answer with 403 code |
22:23.46 | WIMPy | Try to restart asterisk. |
22:24.03 | WIMPy | sip reload doesn't always update everything. |
22:24.11 | Penguin | fromdomain changes the From: part of the INVITE. |
22:24.27 | vorona | I use module reload |
22:24.49 | WIMPy | Not sure that's any better. |
22:24.53 | WIMPy | module unload/load does work however. |
22:25.14 | vorona | Restarted, trying to call |
22:25.52 | vorona | Not working |
22:26.39 | vorona | Penguin, I need change INVITE, with from and contact no problems |
22:28.48 | vorona | Is this option in sip.conf =) |
22:30.11 | vorona | I need option like "todomain" |
22:30.46 | Penguin | I believe that option is "host" |
22:30.56 | WIMPy | Indeed |
22:31.27 | Penguin | fromdomain is used to change the "From:" field. |
22:31.30 | Penguin | nothing else. |
22:32.23 | Penguin | If you want to send calls to sip.beeline.ru, set host=sip.beeline.ru |
22:32.39 | vorona | Penguin, I think so, but this option changes realy address of peer, what I'm not needed |
22:32.58 | Penguin | host=sip.beeline.ru <------- |
22:33.00 | Penguin | Did you do it? |
22:34.00 | vorona | yes, with this option peer unreacheble, sip.beeline.ru and msk.sip.beeline.ru has different addreses |
22:34.28 | Penguin | Set your outboundproxy to the address you want to connect to. |
22:34.47 | vorona | Penguin it has no effects |
22:34.53 | Penguin | Show me. |
22:34.54 | Penguin | ~pb |
22:34.54 | infobot | A "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. |
22:37.06 | vorona | http://pastebin.com/LV2g4NsN |
22:37.26 | vorona | OpenWrtMSK*CLI> sip show peers |
22:37.27 | vorona | Name/username Host Dyn Nat ACL Port Status |
22:37.27 | vorona | 201/201 172.16.117.33 D 5062 OK (65 ms) |
22:37.27 | vorona | b_trunk/vorona@sip.beelin 195.239.253.100 N 5060 UNREACHABLE |
22:37.39 | WIMPy | Where is the ",force"? |
22:38.01 | WIMPy | >>You probably need outbountproxy=...,force |
22:38.11 | vorona | I chenged it |
22:38.24 | vorona | It had no effects |
22:38.39 | WIMPy | Did you restart? |
22:38.57 | vorona | yes |
22:39.18 | Penguin | I don't see the defaultuser specified and I don't see where you have a secret. |
22:39.29 | ChannelZ | it's a secret |
22:39.36 | vorona | Yes =) |
22:39.41 | Penguin | canreinvite and directmedia? |
22:39.42 | Penguin | Remove canreinvite |
22:39.50 | Penguin | directmedia is the correct name for that. |
22:40.44 | Penguin | And I can't read Russian, so I can't read their web site to see if they have an asterisk configuration example. |
22:41.01 | Penguin | AND... |
22:41.11 | Penguin | It's "outboundproxy" |
22:41.21 | Penguin | You have to spell it correctly for it to work. |
22:41.41 | WIMPy | That should definitely help. |
22:42.05 | WIMPy | It's always good if someone can actually read :-) |
22:42.20 | Penguin | I recommend copy/paste. |
22:42.41 | Penguin | That way you can blame it on the person instructing if you get it wrong. |
22:42.58 | Penguin | "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) |
22:45.58 | WIMPy | Wouldn't it be great if chan_sip told you about crap in your configs? |
22:47.36 | file | chan_pjsip does, and will refuse to load a broken configuration |
22:48.56 | vorona | chan_sip passed my config, changes in "outboundproxy" has no results in peer ip now ip of domain |
22:49.41 | Penguin | outboundproxy is the name or address or the host you'll send your requests to. |
22:50.06 | vorona | I know, but it change nothing |
22:50.50 | ChannelZ | your provider is retarded |
22:51.15 | vorona | ChannelZ, yeah, I think so |
22:51.22 | WIMPy | Those two hosts resolve to the same IP anyway. |
22:51.57 | vorona | provider's support cant tell me even peer used tcp |
22:52.37 | vorona | WIMPy, whitch hosts? sip... and msk...? |
22:52.41 | Penguin | wimpy: 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) |
22:53.59 | Penguin | And they don't seem to have any SRV records either. |
22:54.20 | WIMPy | Isn't it great that we check different things? |
22:56.17 | vorona | http://pastebin.com/w8NdqHTd |
22:56.42 | Penguin | Since those two names resolve to the same IP, there's no sense in setting outboundproxy since host will take care of it. |
22:57.11 | vorona | http://pastebin.com/LiMDvDi1 |
22:57.27 | Penguin | That's an internal DNS server. |
22:58.01 | WIMPy | Ugly |
22:58.11 | Penguin | You can put any information you want into an internal DNS server. Including invalid information. |
22:58.26 | vorona | This my router, whith forwards all request to provider's DNS |
22:58.54 | vorona | all information from provider's DNS |
23:00.24 | vorona | WIMPy, agree |
23:00.24 | Penguin | http://pastebin.com/3KwC9YmN |
23:01.19 | vorona | Penguin, nice |
23:02.08 | Penguin | I 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 |
23:02.33 | vorona | one moment, i check my dns |
23:03.13 | ChannelZ | kqfp9 |
23:03.21 | Penguin | ? |
23:03.46 | ChannelZ | bollocks. |
23:04.08 | ChannelZ | there went half a password |
23:04.16 | vorona | http://pastebin.com/93BurMx1 |
23:04.30 | vorona | This my dns |
23:04.51 | Penguin | I'll check those for the SRV records. |
23:05.43 | Penguin | _sip._udp.sip.beeline.ru. 8501 IN SRV 20 0 5060 global.sip.beeline.ru. |
23:06.36 | vorona | Penguin, exactly |
23:07.49 | Penguin | global.sip.beeline.ru. 2340 IN A 195.239.253.100 |
23:09.36 | Penguin | msk.sip.beeline.ru. 4887 IN A 10.25.0.50 |
23:09.45 | vorona | yes |
23:09.55 | Penguin | Do not use msk.sip.beeline.ru for anything unless you are on the same LAN segment as that host. |
23:10.06 | Penguin | It has a private address, so do not use that hostname. |
23:10.29 | WIMPy | fears that's the idea... |
23:10.54 | vorona | Penguin, do you mean, that I must enter ip in host? |
23:10.56 | WIMPy | Do you have a route there? |
23:11.17 | Penguin | Interesting question. |
23:11.37 | vorona | http://pastebin.com/dP6719Ba |
23:12.24 | Penguin | Are you on the same LAN as the ITSP host? |
23:12.53 | vorona | Penguin, no, but host is reacheble |
23:13.01 | WIMPy | A little hard to read, but yes, it looks like you're supposed to use private IPs. |
23:13.11 | Penguin | can't read Russian. |
23:13.39 | WIMPy | I don't even have a font to display it. |
23:13.46 | WIMPy | Not that it would make any difference. |
23:13.55 | vorona | Penguin, sorry, http://pastebin.com/QYEFxDf9 |
23:14.18 | Penguin | that's the same thing. |
23:15.20 | Penguin | I'm more interested in why 10.0.0.0 is reachable. |
23:15.53 | WIMPy | Why not? |
23:16.09 | vorona | Penguin, on my router I hame 2 connection: IPoE and L2TP |
23:16.34 | Penguin | You have a VPN to beeline.ru? |
23:16.40 | vorona | IPoE 10.93.16.9 and L2TP - external |
23:17.12 | vorona | Penguin, no, I have VPN FROM beeline.ru to the internet |
23:18.00 | vorona | http://pastebin.com/16bpW7W5 |
23:18.34 | WIMPy | Ouch |
23:18.45 | vorona | WIMPy, yeah =) |
23:19.18 | Penguin | This doesn't make much sense to me. |
23:21.21 | vorona | Strange, but X-lite with same options sent INVITE with domain, not host |
23:21.39 | WIMPy | Now 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. |
23:21.48 | scv | lol |
23:21.49 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
23:21.58 | ChannelZ | haha for serious |
23:22.21 | Penguin | I'm surprised they can reach your servers to attempt cracking them open. |
23:24.24 | vorona | Many D-link routers with RU suffix even has special 'Russian PPTP' and 'Russian L2TP' connections =) |
23:24.46 | scv | similar situation in austria from what i understand |
23:25.29 | WIMPy | My provider also uses private IPs for VOIP. |
23:26.17 | vorona | WIMPy, but< I think, your provider not uses public fqdn for private IP =) |
23:26.19 | Penguin | Are you on the Germany IntraNet? |
23:26.37 | WIMPy | They do |
23:26.56 | WIMPy | No. It's independant of the internet connection. |
23:27.17 | WIMPy | And it would be a spanish intranet :-) |
23:27.35 | vorona | WIMPy my DNS undependant of L2TP too |
23:27.52 | vorona | 10.x.x.x reacheble without it |
23:28.04 | Penguin | For asterisk, would you put the public hostname/IP into the outboundproxy and the private hostname/IP into the host setting? |
23:28.34 | WIMPy | The other way round. |
23:29.16 | vorona | Penguin, you mean host=msk.sip.beeline.ru outboundproxy=sip.beeline.ru ? |
23:29.27 | WIMPy | But as the private IPs are only resolvable on an DNS that's on a private IP as well, I have to use IPs there. |
23:29.44 | Penguin | I 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) |
23:30.13 | WIMPy | I'm pretty sure you wouldn't have a chance anyway. |
23:32.10 | Penguin | rolls over |
23:32.28 | vorona | INVITE sip:8xxxxxxxxxx@msk.sip.beeline.ru |
23:32.29 | vorona | ( |
23:34.13 | scv | noot noot |
23:34.52 | Penguin | You want it to say not msk.sip but just sip, right? |
23:35.29 | vorona | yes |
23:35.36 | Penguin | Do you have a fromuser= value in sip.conf? |
23:37.17 | vorona | http://pastebin.com/kT8cN8uC |
23:42.47 | vorona | is there any way to cancel srvlookup on peer, but not global? |
23:43.18 | Penguin | I guess you set a host name that doesn't have an SRV record. |
23:44.14 | vorona | It has SRV |
23:44.58 | vorona | http://pastebin.com/dagBmCas |
23:47.19 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
23:49.03 | vorona | Now in Msk 2:48 AM I very want to sleep... |
23:49.12 | vorona | Thanks all for help |
23:49.23 | vorona | I'll continue it tomorrow |
23:50.27 | vorona | bye all! |