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| 00:19.01 | file | omg The Cardigans |
| 00:19.19 | ruben23 | hi guys where do i add music moh on asterisk..? mp3 and wav.. |
| 00:20.10 | file | unless you compile the addon format_mp3 Asterisk will not be able to read MP3 files, but both are stored (by default) in /var/lib/asterisk/moh |
| 00:26.30 | ruben23 | <PROTECTED> |
| 00:38.18 | file | you can enable it by enabling it in menuselect of Asterisk, it is under Add-ons |
| 00:38.43 | file | it is ideal to use files in the formats you expect channels to be in, though |
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| 00:47.05 | ruben23 | ok, any idea how to covert my mp3 to wav that plays with asterisk somehow..? applications..? |
| 00:56.25 | WIMPy | sox |
| 00:56.44 | WIMPy | further information is available from uncle google. |
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| 01:34.55 | TazzNZ | hey file seems someone asked for rfc4904 support back in 2010 |
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| 01:47.27 | mjordan | TazzNZ: please don't take this the wrong way, but typically, unless someone contributes a patch, developers are unlikely to work it. At the end of the day, there's about 1000 open issues in the backlog - and the likelihood of one of us having free time to go off and do a random request that doesn't impact thousands of users just isn't that great. |
| 01:48.03 | mjordan | we operate heavily on that fundamental principle of open source: ideas are great, but code talks |
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| 08:30.39 | stian | has anyone seen this before: CID Superfecta: The DID passed from Asterisk is: s |
| 08:30.57 | stian | it's passing "s" as the DID, weirdly enough |
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| 08:37.29 | TazzNZ | stian: I have - I think it was something with a wrong macro or something |
| 08:39.58 | stian | TazzNZ: hmm it works now |
| 08:40.01 | stian | I think |
| 08:40.17 | stian | debugged it and it seems my /tmp directory was owned by root |
| 08:41.19 | stian | specifically error-output.txt |
| 08:41.28 | stian | changing that to asterisk:asterisk fixed it |
| 08:41.43 | stian | guessing there are some ownership issues :P |
| 08:43.13 | stian | I'm having massive issues with people trying to auth to my server though :( |
| 08:43.37 | stian | fail2ban takes care of it but it's unsetteling to see the log files |
| 08:48.30 | stian | I have port 5060-5070 forwarded to the pbx |
| 08:49.01 | stian | might not be a good idea, however the sip provider we're using needs to access on 5060 :/ |
| 08:49.08 | stian | guess I'm asking for trouble really |
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| 08:51.55 | stian | I guess it's probably not possible to avoid this? |
| 08:52.16 | stian | have super secure extension passwords and generated passwords for everything as well |
| 08:52.46 | WIMPy | If you only need it for your ITSP, why do you open the port for the world? |
| 08:53.12 | stian | ITSP? |
| 08:53.35 | stian | Sorry, I'm a noob when it comes to this stuff |
| 08:54.05 | WIMPy | Your provider |
| 08:54.10 | WIMPy | ~itsp |
| 08:54.10 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 08:54.23 | stian | Alright, now I know :) |
| 08:54.38 | TazzNZ | stian: and it's normally only port 5060 |
| 08:54.54 | TazzNZ | well - there is RTP too |
| 08:55.06 | stian | Well, they recommend port 5060 UDP forwawrded |
| 08:55.07 | TazzNZ | but that is different, sip is normally 5060 only |
| 08:55.14 | stian | mhm |
| 08:55.15 | WIMPy | That's ain another an muuuuch bigger range. |
| 08:55.22 | WIMPy | -a |
| 08:55.52 | stian | I'm looking at my router now and I have 5060:5070 forwarded |
| 08:56.05 | TazzNZ | I would do 5060 |
| 08:56.10 | stian | to the pbx that is |
| 08:56.12 | TazzNZ | and what ever you have in rtp.conf |
| 08:56.18 | TazzNZ | and only from your ITSP |
| 08:56.29 | WIMPy | It shouldn't matter since there's probably nothing listening on the other ports anyway. |
| 08:56.43 | stian | well, I'm being scanned |
| 08:56.46 | stian | every day |
| 08:56.48 | WIMPy | Yeah, the source is the point. |
| 08:56.55 | TazzNZ | stian: so am i |
| 08:57.02 | TazzNZ | and so is every sip server out there |
| 08:57.03 | WIMPy | That's pretty normal. |
| 08:57.11 | stian | ironically the IP's show up as my pbx IP too |
| 08:57.19 | WIMPy | Depends on where you are on the net. |
| 08:57.23 | stian | right |
| 08:57.32 | stian | I see that it's mostly Pakistan, and China etc |
| 08:57.45 | TazzNZ | stian: I would also recomment that you build a default context that "goes no where" |
| 08:57.48 | WIMPy | It almost never happens to me, but I see lots of attempts on other boxes. |
| 08:58.17 | stian | I guess I could only allow the ip's within my ITSP's range |
| 08:58.23 | stian | TazzNZ: what to you mean exactly? |
| 08:58.55 | WIMPy | Find the IP(s) of their server(s) and only allow those. |
| 08:59.05 | TazzNZ | stian: yeah - that is the better idea |
| 08:59.10 | TazzNZ | I have my open for a reason |
| 08:59.18 | stian | right, I have tried to ask them but meh they're not very responsive |
| 08:59.42 | WIMPy | I have them open as well. What's the point of using VOIP if it's not reachable? |
| 08:59.49 | stian | it's funny how they even don't allow useragent = asterisk. had to change it to cisco or something to make it register |
| 09:00.15 | stian | agreed |
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| 09:00.33 | WIMPy | If they don't want your money, go somewhere else. |
| 09:00.42 | stian | I plan to, eventually |
| 09:00.52 | stian | :) |
| 09:00.55 | TazzNZ | LOL |
| 09:00.57 | TazzNZ | that is sad |
| 09:01.17 | WIMPy | Are you in Germany? |
| 09:01.20 | stian | I need to speak to the developers, but instead I'm given the shit end of sales |
| 09:01.22 | stian | I'm in Norway :) |
| 09:01.37 | stian | but the ITSP is Swedish / Norwegian (support is swedish at least) |
| 09:01.46 | WIMPy | didn't know it's as bad elsewhere as well. |
| 09:01.53 | stian | It is |
| 09:02.24 | stian | They don't want people running their own servers, instead they say they support 3CX and other PBX's |
| 09:02.29 | stian | it's all about $$ anyhow |
| 09:03.06 | TazzNZ | WIMPy: I think ITSP *everywhere* are sh1t - they try and run SIP like it was the old Telco days |
| 09:03.15 | stian | ^^ |
| 09:03.17 | stian | bbl lunch |
| 09:03.22 | stian | thanks for the help though duys |
| 09:03.24 | TazzNZ | "Oh sorry - you didn't spend $$$$$$$$$$$$$$$$$$$$$$$$ on Product X, you can't connect to the network" |
| 09:03.28 | WIMPy | Well, if you use SIP how could you possibly not be shit? |
| 09:03.34 | stian | "we're soorry" |
| 09:04.06 | stian | oh btw |
| 09:04.11 | stian | I'm getting a lot of these: Sending fake auth rejection for device 5010<sip:5010@MYPBXIP5060>;tag=7e02c982 |
| 09:04.21 | TazzNZ | stian: yip - all normal |
| 09:04.35 | WIMPy | Just someone trying to hack you. |
| 09:04.35 | stian | alrighty |
| 09:04.38 | stian | Yeah |
| 09:04.41 | eirirs | stian: i use altibox's ITSP, no problem |
| 09:04.41 | TazzNZ | stian: Asterisk is sending a fake "user is right, password is wrong" |
| 09:04.42 | stian | It's quite disturbing :P |
| 09:04.50 | stian | eirirs: I need something for business :P |
| 09:05.05 | eirirs | stian: its altibox business, they dont offer ITSP for home users |
| 09:05.08 | TazzNZ | or something like that - reading up on Fake auth will tell you the whole story |
| 09:05.13 | stian | eirirs: even better |
| 09:05.17 | stian | I will TazzNZ |
| 09:05.29 | stian | Does this effect performance at all? |
| 09:05.31 | eirirs | stian: I use asterisk->altibox for my smb's |
| 09:05.40 | stian | sweet, eirirs I will look into it |
| 09:05.47 | TazzNZ | stian: nah - unless your * box is 100% used |
| 09:05.55 | stian | You don't by any chance have a good regex dialpattern laying around? :P |
| 09:06.03 | stian | TazzNZ: have TONS of overhead :) |
| 09:06.10 | stian | running on a quad core i5 lol |
| 09:06.13 | WIMPy | Dialplan patterns are not regex. |
| 09:06.15 | eirirs | stian: and they aint even charging us for it, only normal phone rates, but maybe its because we have altibox as ISP too. |
| 09:06.25 | stian | WIMPy: pattern matching, sorry |
| 09:06.40 | TazzNZ | stian: define tons of overhead ? |
| 09:07.16 | stian | we don't have many users on the system, and currently it sits at 4% load |
| 09:07.23 | eirirs | quad core i5 sounds overkill |
| 09:07.25 | stian | asterisk uses 4% |
| 09:07.33 | eirirs | we have a small hyper-v with 512mb ram for it |
| 09:07.52 | WIMPy | Wiki says Altibox is NO and DK, no SE. |
| 09:07.56 | TazzNZ | 4% is not tons of overhead :D |
| 09:08.01 | TazzNZ | anyways - bbib |
| 09:08.06 | stian | eirirs: it's a machine that runs VM's |
| 09:08.12 | stian | anyhow bbl lunch ! :D |
| 09:08.21 | eirirs | ahh lunch, thanks for reminding me |
| 09:08.28 | stian | yes indeed :) |
| 09:08.39 | WIMPy | Anyway, the trouble with chan_sip is that there is no parallelisms. Everything is done sequentially. |
| 09:09.23 | WIMPy | So too much shit can have a negative impact, even if you don't have noticable load. |
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| 09:10.33 | eirirs | stian: anyhow, my router rejects ALL incoming to port 5060 + rtp range from source other than altibox's PBX. |
| 09:10.43 | eirirs | so i aint that worried about security here |
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| 09:29.47 | andycol | hi Guys |
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| 09:30.29 | andycol | i have the monitor_exec running in the dialplan to run a perl script with some paramaters |
| 09:30.43 | andycol | but on the cli it shows it is running but does nothing unless i run it manually |
| 09:30.44 | andycol | [Jun 23 11:26:20] VERBOSE[9925] pbx.c: -- Executing [1@officehours:3] Set("DAHDI/i1/0114892000-1d69", "MONITOR_EXEC=/usr/local/parselog/update_mix_mixmonitor.pl 1403515556.64748 i-20140623072620-1.wav |
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| 09:55.24 | andycol | so monitor_exec is not working |
| 09:55.30 | andycol | any ideas why |
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| 10:04.40 | andycol | just ignores the command |
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| 13:43.03 | stian | does anyone have a US ITSP that they can recommend? |
| 13:43.11 | stian | s/have/know of |
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| 14:05.45 | cusco | hello folks |
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| 14:08.02 | cusco | we've compiled asterisk 11 the other day, so far so good.. now there was a module missing so we installed the libs, ran configure again, make .. and now can't load the module... state it was not compiled with the same compile-time options as this version of asterisk |
| 14:09.30 | Nivex | intriguing. this just hit wire: https://library.linode.com/communications/voip-services-ubuntu-12.04 |
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| 14:48.17 | Nugget | what's intriguing about it? |
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| 15:09.45 | qakhan | when every i use require_once("phpagi/phpagi.php"); in .php |
| 15:10.15 | qakhan | why require_once("phpagi/phpagi.php"); always call /etc/asterisk/phpagi.conf? |
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| 15:55.15 | jeremy_laine | file: a big thumbs up for taking on the DTLS-SRTP task |
| 15:55.42 | file | music is keeping me sane. |
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| 15:58.07 | jeremy_laine | file: that sounds like a brittle setup, what happens if the music stops? |
| 15:58.29 | file | I err on the side of not thinking of such dark things |
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| 16:43.27 | file | MUSIC ALL THE THINGS |
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| 16:49.43 | stian | ah damn |
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| 17:00.11 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 17:22.14 | mbowie | Has anyone had the pleasure of coming across the following issue with res_config_ldap? "res_config_ldap.c:973 realtime_multi_ldap: realtime retrieval requires at least 1 parameter and 1 value to search on." Google-fu is returning questions rather than answers. |
| 17:24.32 | mbowie | I'm *guessing* it's an issue parsing the query added by the addition of the callbackextension mojo... but that's so far without substantiation. :-p |
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| 17:28.40 | Katty | hi lads. |
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| 17:34.36 | TazzNZ | Hey Katty |
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| 17:35.12 | TazzNZ | mbowie: you'll need to show us the config for that |
| 17:37.32 | mbowie | TazzNZ: Fair enough... will start assembling it now. Is the "NZ" in your nick of any specific reference? |
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| 17:40.16 | andy09usa | TazzNZ: good time of the day, I found out that google voice does not work consistently on Asterisk 11.10.2, 11.9.0 version on all good! |
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| 17:46.58 | Katty | pokes [TK]D-Fender |
| 17:47.30 | [TK]D-Fender | prods Katty |
| 17:47.51 | Katty | did you have a fun weekend? |
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| 17:50.46 | [TK]D-Fender | hasn't ended, it's a long one here, and I took today off to make it a 4-day (tomorrow is a provincial holiday |
| 17:51.08 | Katty | lucky duck (= i'm surprised you're here! |
| 17:51.19 | Katty | well, no. i guess i'm not really surprised. |
| 17:51.32 | Katty | shouldn't you be watching soccer or something? |
| 17:53.18 | [TK]D-Fender | Soccer is just like wathcing grass grow... |
| 17:53.24 | [TK]D-Fender | ...except with soccer players on top. |
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| 17:55.54 | Katty | pats [TK]D-Fender |
| 17:56.12 | Katty | i don't know much about that sort of thing either. |
| 17:57.48 | [TK]D-Fender | For me the scores alone give me a basis of dislike. This is how I percieve hockey (as something I can respect), and basketball (not) |
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| 17:58.29 | [TK]D-Fender | Soccer: Italy CRUSHES France .... 1-0. One stupid goal, more of a fluke than anything. |
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| 17:59.38 | mbowie | extconfig.conf: http://pastebin.com/qgSFiNGT | res_ldap.conf: http://pastebin.com/KBzaYpTL | slapcat: http://pastebin.com/vgHF6CMP |
| 17:59.49 | [TK]D-Fender | Basketball: LA beats Phoenix .... 109-107. I could have watched the last 5 minutes and saved the rest of my time... |
| 18:01.08 | [TK]D-Fender | Hockey: Montreal defeats Boston .... 6-2. That SAYS something. Substantial reasonable difference in scoring. Actual actual without just running a counter. |
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| 18:12.15 | stian | burps |
| 18:12.36 | stian | yawn |
| 18:13.22 | stian | hockey is quite nice, not that I follow it |
| 18:13.30 | stian | but it's cooler to watch than soccer |
| 18:17.16 | mbowie | wonders if that's an ice joke. |
| 18:18.33 | stian | hah |
| 18:18.57 | stian | actually no, but hey I'll take it |
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| 18:38.27 | pjensen00 | I was wondering if the ARI is going to slowly overtake AMI as the supported async interface to asterisk in 12+. I'm just thinking of the future and if I should plan for the long term switch to ARI |
| 18:38.59 | file | they present different interfaces which do different things |
| 18:39.03 | file | (they aren't equivalent) |
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| 18:40.31 | file | perhaps a long way down the road functionality will exist in ARI which does what AMI does, but that remains to be seen and really even discussed |
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| 18:43.34 | SuperNull | if i want to allow any inbound call to be received from a friend without auth isnt it just as simple as adding a friend entry with an ip ? i keep getting auth declined ? |
| 18:46.22 | *** join/#asterisk Oggu (~Oggu@h-199-22.a137.corp.bahnhof.se) |
| 18:46.39 | [TK]D-Fender | SuperNull: You didn't tell it to not auth them... |
| 18:47.28 | SuperNull | okay. what am i missing here sir TK |
| 18:49.06 | [TK]D-Fender | insecure <-- |
| 18:49.22 | [TK]D-Fender | otherwise it WILL challenge them |
| 18:49.47 | *** join/#asterisk Dpunkt (~Dpunkt@p57A4ED7C.dip0.t-ipconnect.de) |
| 18:51.08 | Dpunkt | how can i set the c-flag (call answered elsewhere) on a call initiated when calling via call-file? |
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| 18:54.08 | mbowie | Is anyone here using LDAP realtime with 11.x (or 12.x for that matter)? Interested to see what I'm overlooking to get this "realtime_multi_ldap: realtime retrieval requires at least 1 parameter and 1 value to search on." ... or perhaps I should checkin on the ML instead. |
| 18:56.54 | [TK]D-Fender | Dpunkt: You will have to use a Lacal channel as your "Channel:" and set it int he dial you do yourself |
| 18:57.09 | Dpunkt | Thank you |
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| 19:02.47 | SuperNull | TK i got it, oddly required core restart before it took it instead of sip reload. |
| 19:04.29 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:07.02 | [TK]D-Fender | didn't require that./.. |
| 19:07.11 | [TK]D-Fender | That is just something you chose to do... |
| 19:07.20 | [TK]D-Fender | SIp reload would take that change |
| 19:09.45 | *** join/#asterisk Tod|Home (~TodWulff@unaffiliated/todwulff) |
| 19:14.54 | Tod|Home | can asterisk be configured to use ITSPs in an external gateway device (OBi202 in my case) as sip trunks? The ITSPs remotely provisioned the gateway and won't provide sip credentials to allow me to instantiate a trunk direct to their servers inside Asterisk (sumtin about unauthorized clients... :). As such, I am wondering if I can, via instantiating a 2nd GW:Asterisk channel on the device, loop thru and use the GW's other ITSPs as Asterisk SIP |
| 19:16.20 | *** join/#asterisk btcNeverSleeps (~user@109.129.187.206) |
| 19:16.34 | Cubber | I am building an asterisk system using gentoo and asterisk 11.8.1 from portage. I have a Wildcard AEX800 FXO card installed with 8 ports. I am testing it right now and am finding that if I call in through the FXO port the call goes fine, but after it hangs up if I call in a second time I get a loud shrill then after a few seconds it moves to the auto attendent. I am only testing with a single line plugged into port1. |
| 19:16.57 | Cubber | I have moved the line from port to port to test the other ports but only have access to a single phone line in my testing environment |
| 19:17.20 | Katty | runs around doing the airplane |
| 19:17.27 | Katty | neeeoorrrwwmmmm |
| 19:17.35 | Tod|Home | lol |
| 19:18.36 | Cubber | /etc/dahdi/system.conf file: http://bpaste.net/show/400709/ and /etc/asterisk/chan_dahdi.conf http://bpaste.net/show/400710/ |
| 19:18.40 | btcNeverSleeps | If I want to install SIP phones at home, what is the minimum number *physical* of things that I need to have? 2? Say a RaspberryPi with Asterisk/FreePBX and on SIP phone? Is that enough? |
| 19:19.01 | [TK]D-Fender | Tod|Home: It isn't a question if Asterisk can be configured to... can your DEVICE magically forward things between the two... |
| 19:19.18 | [TK]D-Fender | Tod|Home: At which point that device is largely a dumb ATA and not made to be used as some sort of proxy |
| 19:20.02 | [TK]D-Fender | btcNeverSleeps: You MIGHT want a switch. |
| 19:20.22 | [TK]D-Fender | btcNeverSleeps: but I supposed you could just use a crossover cable... |
| 19:20.23 | btcNeverSleeps | [TK]D-Fender: I have an ethernet switch and a computer of course ^ ^ |
| 19:20.45 | btcNeverSleeps | [TK]D-Fender: do I need a place to plug a phone line at all? |
| 19:20.49 | Tod|Home | [TK]D-Fender, yeah, I suspected that I may need to delve into GW functionality to see if it even supports doing obtuse stuff like that. I wanted to ask herein, just in case it had been considered and vetted previously by the audience. Thanks. |
| 19:20.51 | [TK]D-Fender | btcNeverSleeps: If you want to use * witha SIP phone.. you need something to run * on ... a Phone.... and some amount of wiring between them. |
| 19:21.28 | [TK]D-Fender | Tod|Home: It doesn't. This is a wild goose chase. They locked you out and that's pretty much the end of it |
| 19:21.38 | Tod|Home | nods |
| 19:23.20 | btcNeverSleeps | ok, maybe phrased in another way: if I want a SIP phone only to receive and give phone calls, do I need some hardware on which Asterisk (or something similar) would run or can I simply plug the SIP phone on the ethernet switch? |
| 19:33.46 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 19:35.15 | pjensen00 | file: thanks for the reply! |
| 19:36.12 | *** join/#asterisk Cubber (~ronny@mail.ronnybull.com) |
| 19:36.53 | [TK]D-Fender | btcNeverSleeps: It appears you don't understand what Asterisk itself is.... |
| 19:37.22 | *** join/#asterisk roentgen (~none@openvpn/community/support/roentgen) |
| 19:37.54 | stian | Haha sorry for laughing but your comment.. |
| 19:38.35 | [TK]D-Fender | btcNeverSleeps: A SIP phone places & receives calls from whatever you configure it to talk to. Where that leads is another matter. |
| 19:39.45 | Katty | ringringringringringringring, banana phone! |
| 19:39.53 | Katty | infobot: change your name back to jbot |
| 19:39.53 | stian | :D |
| 19:39.57 | Katty | infobot: banana phone |
| 19:40.04 | Katty | infobot: bananaphone |
| 19:40.04 | infobot | methinks bananaphone is cellular, modular, interactive-odular |
| 19:40.08 | stian | bananaphoone |
| 19:40.35 | pjensen00 | General question for PJSIP configuration. I am trying to route all default outbound traffic (dials made from dialplans) to my SIP server. Basically I want an entry in pjsip.conf for an endpoint that catches all my dials between my own SIP endpoints |
| 19:41.00 | pjensen00 | I'm not sure how to do this because right now the only way to make it work is the long hand way |
| 19:41.22 | pjensen00 | for example, I have 1234@customer5.whatever.com and 1234@customer6.whatever.com |
| 19:42.12 | Katty | infobot: banana phone is it's the best, beats the rest!!! https://www.youtube.com/watch?v=jkaKhFl2ocw |
| 19:42.12 | infobot | okay, Katty |
| 19:42.15 | pjensen00 | So I have manual entries for customerX@whatever.com where X is the specific customer. |
| 19:42.26 | Katty | infobot: banana phone |
| 19:42.27 | infobot | i heard banana phone is it's the best, beats the rest!!! https://www.youtube.com/watch?v=jkaKhFl2ocw |
| 19:42.37 | Katty | infobot: good bot. |
| 19:42.37 | infobot | Katty: :) |
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| 19:42.50 | Cubber | could the shrilling sound I am experience be the result of faulty hardware? Like the echo cancelation module on the FXO card? |
| 19:42.59 | pjensen00 | There any way to be able to just point them all instead of doing an individual entry for each one? |
| 19:44.02 | [TK]D-Fender | Cubber: no, more like improper disconnect supervision, signalling, etc |
| 19:44.38 | Cubber | [TK]D-Fender The input phone line is coming from a time warner cable modem |
| 19:45.15 | Cubber | [TK]D-Fender I know the end client has 3 time warner lines as well that they intend to plug into this card |
| 19:52.45 | Cubber | wq |
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| 20:29.29 | mjordan | pjensen00: there is an anonymous endpoint identifier that you can use |
| 20:29.51 | mjordan | pjensen00: it will match any un-matched requests and associate those requests with that endpoint |
| 20:30.39 | Qwell | I'm failing a bit here. Isn't there some app/func that can tell you whether an extension exists? |
| 20:31.18 | mjordan | er |
| 20:31.19 | Qwell | alternatively, something that can attempt to match a value to an extension pattern. such as 5551212 == NXXXXXX |
| 20:32.25 | Qwell | DIALPLAN_EXISTS! |
| 20:32.31 | [TK]D-Fender | yup |
| 20:32.49 | [TK]D-Fender | You could also use ChanIsAvail |
| 20:32.57 | Qwell | god no |
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| 20:45.18 | pjensen00 | mjordan: thanks, will give that a shot |
| 20:46.58 | TazzNZ | can anyone remember where I can get the script to detect the CPU type for Fax for Asterisk ? |
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| 20:51.50 | pjensen00 | mjordan: Tried this http://pastebin.com/JzKVH6Zz and it didn't work. I'm guessing I am taking your advice and doing the wrong things. :( |
| 20:56.10 | TazzNZ | I would have thought that Digium had a SIP URI contact... :) |
| 20:58.20 | pjensen00 | mjordan: Whooops, actually this is what I used http://pastebin.com/cRJcTwCw |
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| 21:02.29 | pjensen00 | I also tried this setup http://pastebin.com/rud8d6nh |
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| 22:14.45 | btcNeverSleeps | ok so I don't understand what Asterisk is. Can someone explain me? Say I want a SIP phone so I can both receive and give phonecalls to regular landlines numbers... Do I need a computer/Raspberry Pi/whatever running Asterisk (or any software similar to Asterisk)? |
| 22:18.53 | TazzNZ | btcNeverSleeps: Yes - Asterisk is one of the things you will need |
| 22:19.09 | TazzNZ | you will also need a way to connect Asterisk to the PSTN network |
| 22:19.18 | TazzNZ | be it via an ITSP or a Telephony card |
| 22:24.27 | btcNeverSleeps | TazzNZ: the way to connect it to the PSTN is the "thing" with a phone plug? |
| 22:25.44 | ChannelZ | Yes. Or via an ITSP (so, voip over the net) |
| 22:26.13 | [TK]D-Fender | [18:18]TazzNZbtcNeverSleeps: Yes - Asterisk is one of the things you will need <- not necessarily |
| 22:26.22 | ChannelZ | But if all you want is a phone you don't necessarily need asterisk |
| 22:26.37 | Tod|Home | nods |
| 22:27.13 | [TK]D-Fender | btcNeverSleeps: You can use physical lines if you have an usable interface to plug them into. Or you could get to the PSTN via an ITSP |
| 22:27.54 | btcNeverSleeps | and "an ITSP" means I need a device or it's just "all IP" and I buy some service from some company? |
| 22:28.10 | [TK]D-Fender | ~itsp |
| 22:28.10 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 22:28.12 | [TK]D-Fender | ^^^ |
| 22:28.18 | [TK]D-Fender | it is a SERVICE over the internet |
| 22:28.46 | btcNeverSleeps | I see, thanks. |
| 22:29.06 | [TK]D-Fender | So instead of you having a copper line from your local telco give you a "line" and "phone numner", this provider rents you that number and sends you calls placed to it, and lets you place calls |
| 22:30.03 | [TK]D-Fender | number* |
| 22:31.15 | btcNeverSleeps | And in the setup where I get a good old copper line too, what goes through that copper line when I'm using SIP phones? (I mean: what's VoIP then?) |
| 22:31.52 | [TK]D-Fender | VoIP is a protocol to pass calls between IP devices |
| 22:32.17 | [TK]D-Fender | There are GATEWAYS you can buty that will let you Uuse your copper phoneline with * and other VoIP devices |
| 22:32.25 | [TK]D-Fender | Linksys SPA-3102 for example |
| 22:32.51 | btcNeverSleeps | (basically I'm trying to troobleshoot a Raspberry Pi / RasPBX asterisk/FreePBX + SIP phones + issue at my girlfriend's SME after the "network admin" turned rogue and started blackmailing them) |
| 22:33.59 | TazzNZ | btcNeverSleeps: how about you tell us what the issue are ? then we can help from there |
| 22:34.06 | btcNeverSleeps | the setup is Asterisk (on the Raspberry PI) + SIP phones + a device from 3starsnet.com in which a copper line plugs)... And after a power outage the thing stopped working. |
| 22:34.58 | btcNeverSleeps | TazzNZ: sure but first I need to at least understand what are SIP phones :) (I'm a software dev, but although I'm no Un*x admin I'm quite confortable with Linux: I can build and configure servers [continuous integration, webapp, etc]) |
| 22:35.03 | [TK]D-Fender | Uncertain as to what that device is doing exactly. |
| 22:35.15 | *** join/#asterisk Defraz (~Defraz@205.185.92.237) |
| 22:35.40 | [TK]D-Fender | If they're using RasPBX then they are using it as a standard PBX for their phones... and using that "device (or service?) for PSTN access |
| 22:37.04 | btcNeverSleeps | [TK]D-Fender: there's a little physical device: it has a "WAN" port from which an ethernet cable goes to the main ethernet switch (16-port switch, for the PCs, printers, SIP Phones I suppose, etc.) and it has LAN port unused and a plug for what looks like a phone cable (which is used). |
| 22:37.38 | [TK]D-Fender | Where does the phone cable go? |
| 22:37.51 | btcNeverSleeps | so basically I managed to change the root password of the Raspberry's Linux system (by plugging the HD to another Linux system and changing /etc/passwd) |
| 22:38.23 | btcNeverSleeps | the phone cable I cannot check right now: it's 1am and I'm at home :) (but I took the Raspberry with me to make backups of the partition, change the passwords, admin emails, etc.) |
| 22:39.22 | *** join/#asterisk RaMcHiP (~RaMcHiP@wsip-98-190-149-42.ph.ph.cox.net) |
| 22:39.31 | RaMcHiP | Hello everyone1 |
| 22:39.37 | TazzNZ | hi RaMcHiP |
| 22:39.39 | btcNeverSleeps | RaMcHiP: hi |
| 22:39.41 | RaMcHiP | I am trying to get vtiger 6.1 to work with click to dial |
| 22:39.59 | RaMcHiP | It seems in my logs that it is trying to dial its own extension |
| 22:40.17 | RaMcHiP | I have tried everything in my context but nothing seems to work |
| 22:40.30 | RaMcHiP | I am going to put a pastebin file together here, one moment |
| 22:43.37 | RaMcHiP | http://pastebin.com/gnbpgiV6 |
| 22:43.42 | RaMcHiP | That is what I get when I click to dial |
| 22:44.27 | RaMcHiP | The to: in the link seems to actually get set as the callerid when it should be set as the # to dial |
| 22:44.37 | [TK]D-Fender | SIP/4911 gets called |
| 22:44.40 | [TK]D-Fender | On answer: |
| 22:44.46 | [TK]D-Fender | (Dial) Options: (SIP/4994&SIP/4999&SIP/4995, 60) |
| 22:44.55 | [TK]D-Fender | Those 3 different devices get called |
| 22:45.04 | RaMcHiP | Yeah, it is not supposed too. It seems like the Asterisk Connector for Vtiger is setting the Callerid wher TO should be |
| 22:45.07 | [TK]D-Fender | 2 aren't contactable |
| 22:45.22 | [TK]D-Fender | <PROTECTED> |
| 22:45.25 | RaMcHiP | Unfortunately that is a JAR file that is closed source... |
| 22:45.31 | btcNeverSleeps | another question: from a SIP phone, I can query its own IP. But can I query too the IP of the Asterisk device? (I don't know if my question makes sense) |
| 22:46.09 | btcNeverSleeps | RaMcHiP: a .jar file is "just" a .zip file (don't know if it helps) |
| 22:46.17 | [TK]D-Fender | btcNeverSleeps: Yes, if it has registered to Asterisk "sip show peers" |
| 22:46.22 | RaMcHiP | Ya but I need to edit source within it |
| 22:46.41 | [TK]D-Fender | RaMcHiP: We don't support their scripts anyway |
| 22:46.41 | RaMcHiP | Can you just edit a file within a jar and still have it work? |
| 22:46.53 | [TK]D-Fender | rqamWe can describe what it's doing... but we don't fix that here |
| 22:47.00 | RaMcHiP | I did not think so been banging my head on this for 3 days now figured I would ask ;) |
| 22:47.36 | RaMcHiP | Am I right that it seems like it setting the to: as the callerid? |
| 22:47.58 | btcNeverSleeps | [TK]D-Fender: that is very interesting (saving that whole discussion)... |
| 22:47.58 | RaMcHiP | when it rings out to the phone it shows the too # as the callerid |
| 22:48.40 | [TK]D-Fender | [18:47]RaMcHiPAm I right that it seems like it setting the to: as the callerid? <- no. We see nothing setting callerid in there. do you? |
| 22:51.24 | btcNeverSleeps | in Asterisk's /var/log/asterisk/security_log I see several lines like: "Registration from '"5050" <sip:5050@81.82.xxx.yyy:5060>' failed for '188.165.224.200:5077'" (where 81.82.xxx.yyy is the public IP of the office). |
| 22:51.48 | btcNeverSleeps | Shouldn't there be a firewall blocking port 5060 from IPs outside of the office's LAN? |
| 22:52.11 | [TK]D-Fender | Firewalls are YOUR job |
| 22:52.28 | [TK]D-Fender | that's OS / networking infrastructure |
| 22:53.09 | btcNeverSleeps | [TK]D-Fender: I know. I'm helping my girlfriend here (it's her own little company) after the previous network admin started blackmailing her. (Basically asking her for a raise, she said no and then he said: "I'm not giving any password unless you give me money") |
| 22:54.27 | btcNeverSleeps | My technical question is: would allowing only local IPs to connect to 5060 prevent Asterisk from working properly? Is there any reason that port 5060 seems to be accessible from the outside world? |
| 22:54.31 | [TK]D-Fender | So did you fix the PW? |
| 22:55.19 | [TK]D-Fender | And that PBX has 5060 forwarded to it.... one would think there is a REASON for it.. such as directly using an ITSP, having remote phones, etc.. |
| 22:55.28 | [TK]D-Fender | Go proves what it was set up to do |
| 22:55.42 | [TK]D-Fender | Because we can't tell you what's appropriate. It's not our server |
| 22:56.44 | btcNeverSleeps | [TK]D-Fender: oh I see. Maybe I could whitelist the ITSP's server(s) IP(s) on port 5060. |
| 22:57.10 | [TK]D-Fender | possibly. or range. |
| 22:57.13 | [TK]D-Fender | depends how they work |
| 22:57.36 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 22:57.37 | [TK]D-Fender | So far we don't know they are set up to talk direct because you mentioned this other device |
| 22:58.10 | btcNeverSleeps | [TK]D-Fender: I'll ask them (it's 3starsnet.com, from their website they seem to some Belgian company). |
| 22:58.40 | [TK]D-Fender | Or.. you could actually look at your PBX and see what's configured |
| 22:58.50 | [TK]D-Fender | [18:54][TK]D-FenderSo did you fix the PW? |
| 22:58.58 | btcNeverSleeps | what's the "PW"? |
| 22:59.04 | [TK]D-Fender | ... PassWord |
| 22:59.28 | btcNeverSleeps | well I did change the Linux system's root password (by modifying /etc/shadow ) |
| 23:00.14 | btcNeverSleeps | and now if I boot the device here I get the Debian / RasPBX prompt when I log in as root (but I haven't access to the ITSP / SIP phones here) |
| 23:00.48 | [TK]D-Fender | That thing is running FreePBNX <- |
| 23:00.52 | [TK]D-Fender | FreePBX <- |
| 23:00.59 | [TK]D-Fender | it's web-admin'd |
| 23:01.01 | btcNeverSleeps | which is a gui on top of Asterisk? |
| 23:01.06 | btcNeverSleeps | k |
| 23:01.07 | [TK]D-Fender | are you in on that as well? |
| 23:01.22 | [TK]D-Fender | Yes, it is a web-admin GUI for Asterisk |
| 23:02.06 | *** join/#asterisk cyborg-one (~cyborg-on@37.203.14.157) |
| 23:02.23 | btcNeverSleeps | I haven't tried yet: at the office the device wouldn't respond to neither ping nor browser request on port 80/443. So I did already modify the firewall by adding rules always accepting connection to/from the LAN on these ports (and accepting ping too). |
| 23:03.26 | btcNeverSleeps | I'll check if RasPBX comes with X and has a browser (I'm not on the same network), in which case I could boot it from here, without connecting it to anything, and see if I can get in. |
| 23:03.35 | [TK]D-Fender | No X |
| 23:03.38 | btcNeverSleeps | erf |
| 23:03.48 | [TK]D-Fender | SSh doesn't care where you are. |
| 23:04.01 | [TK]D-Fender | Tunnel the web interface to your client |
| 23:04.56 | btcNeverSleeps | yup but then I may need to modify my network settings here at home :-/ |
| 23:06.05 | btcNeverSleeps | if I can SSH in, then I can simply also point a browser to, say, 192.168.168.251 (I mean, once FreePBX is running) no!? |
| 23:08.56 | [TK]D-Fender | Read your client's instructions on SSH port tunneling |
| 23:09.20 | [TK]D-Fender | This creates a LOCAL port on your client that redirects to an ip:port on the remote end. |
| 23:10.03 | *** join/#asterisk imcdona (imcdona@2001:470:e916:3:a03a:7230:d912:453b) |
| 23:12.43 | btcNeverSleeps | Oh gotcha: you mean to log in *if* the firewall is refusing connection from the LAN? Don't know if SSH was allowed. It's an all-windows (besides the Raspberry) network. Next time I'll bring a Linux system with me. (I had no idea I'd find a Linux device beforehand) |
| 23:13.40 | btcNeverSleeps | weirdly enough X is installed on the Raspberry PI (not my call...) |
| 23:14.03 | TazzNZ | btcNeverSleeps: imho, running Asterisk on PI is a bad idea, unless you in a dev/test setup |
| 23:14.47 | [TK]D-Fender | You don't need to be running Linux. Use your SSH client on Windows/Whatever to connect to the Pi |
| 23:15.47 | btcNeverSleeps | TazzNZ: I hear you... But it's not my call: I didn't set that thing up. And it "failed" after a power outage. Don't know why yet. |
| 23:16.03 | TazzNZ | sure, I was just saying |
| 23:16.13 | btcNeverSleeps | It may be because the time is wrong (I see message about Asterisk refusing to start up due to the time setting not being correct) |
| 23:16.13 | *** join/#asterisk Dovid (~Dovid@185.23.174.1) |
| 23:16.14 | TazzNZ | so when you find that "network admin", shoot him :) |
| 23:16.34 | btcNeverSleeps | TazzNZ: my girlfriend is taking care of that :) |
| 23:16.43 | TazzNZ | cool |
| 23:18.19 | btcNeverSleeps | I'll call it a day for today: I'll go to sleep now and will come to bug you guys some more once I'm back at the office. |
| 23:18.25 | btcNeverSleeps | thanks to everyone for all the help |
| 23:18.31 | btcNeverSleeps | (things are a bit clearer now) |
| 23:19.07 | btcNeverSleeps | TazzNZ: btw one last question: why is it a bad idea to run Asterisk on a PI? |
| 23:19.33 | TazzNZ | purely due to CPU |
| 23:19.44 | TazzNZ | the PI will only transcode about 3 calls |
| 23:19.55 | btcNeverSleeps | TazzNZ: you mean simultaneously? |
| 23:20.08 | TazzNZ | if you are not transcoding, then you can do 100's via the PI |
| 23:20.15 | TazzNZ | yeah - all at once |
| 23:20.31 | btcNeverSleeps | TazzNZ: that's good to know. There are four SIP phones. |
| 23:20.44 | TazzNZ | but - from a platform point of view - the PI is not meant to replace a PC |
| 23:20.51 | TazzNZ | it a teaching tool |
| 23:20.55 | TazzNZ | it's* |
| 23:21.21 | TazzNZ | If you want small, look into the NUC or something like that |
| 23:21.35 | TazzNZ | something with a bit more CPU |
| 23:21.43 | btcNeverSleeps | but there's one or two unused PCs at the company: I could install RasPBX on one of them. I'll maybe do that once I first figure out how to make everything work again. |
| 23:21.50 | TazzNZ | something that is built to run more "24/7" than a PI |
| 23:22.15 | TazzNZ | btcNeverSleeps: that will be FreePBX or something, RasPBX is only for the PI |
| 23:22.39 | btcNeverSleeps | TazzNZ: oooh, gotcha |
| 23:23.27 | btcNeverSleeps | and if I manage to get RasPBX working again, it's easy to then "move" the setup to a real PC with FreePBX? |
| 23:23.53 | [TK]D-Fender | If the versions are the same, yes... or with a lot of work |
| 23:24.09 | Tod|Home | btcNeverSleeps, if you have a newer router that can handle the OpenWRT firmware, or Merlin's firmware, you may be able to get asterisk on it without having to purchase anything. I've got asterisk on my Asus RT-AC66U router. |
| 23:24.47 | Tod|Home | be warned, full CLI management, however - no prebuilt web ui flavors, however. |
| 23:24.56 | Tod|Home | (at least that I could find for my device) |
| 23:24.59 | btcNeverSleeps | I begin to suspect that after the power outage the PI lost the time setting and that that's the reason it's not working: Asterisk is refusing to start due to bogus time setting. |
| 23:25.57 | WIMPy | The PI doesn't have a clock. |
| 23:26.13 | btcNeverSleeps | so it's ntpd? |
| 23:26.38 | WIMPy | Possibly |
| 23:28.18 | btcNeverSleeps | ok really going to sleep now, it's nearly 1:30 am here in Brussels and I gat to work tomorrow. I'll come back to ask for help once I'm back at my girlfriend's office. See ya all. |
| 23:28.32 | btcNeverSleeps | is saving his entire #asterisk Emacs buffer :) |
| 23:29.06 | btcNeverSleeps | btcNeverSleeps does sleep once in a while after all |
| 23:30.40 | Tod|Home | btcNeverSleeps, freepbxnow works well as a bootable implementation, if you're gonna go the PC route. centos based and flawless for my pc test build. |
| 23:32.38 | [TK]D-Fender | freepbxnow <- this is not a "thing"z |
| 23:33.47 | Tod|Home | lol - it was asterisknow - |
| 23:34.57 | Tod|Home | but it is a freePBX implementation, so not totally out of context and incorrect.? :/ |
| 23:37.33 | [TK]D-Fender | It is an OS with * + FreePBX |
| 23:38.08 | [TK]D-Fender | As far as I've seen it is less regularly maintained that the ISO FreePBX themselves put out |
| 23:41.03 | Tod|Home | nods |
| 23:49.43 | *** part/#asterisk btcNeverSleeps (~user@109.129.187.206) |