IRC log for #asterisk on 20150904

00:02.54xochilpiliWIMPy, yes, i forgot to paste here the ${PWD} and ${input} respectivly; i dont understand as well, what about 's' extension; i mean, im reading about a macro with 's',1,Dial(${ARG2},20); but my question is; in my extension [100] in sip.conf, i have to put my [macro-test] in context option in [100] sip.conf in order to use macro-test?
00:05.24*** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com)
00:05.27WIMPyMacros are deprecated.
00:06.13WIMPyThe context specified in your spi peers will be the one where cqlls from that peer will go to initially.
00:06.32WIMPyWhat you do from there on is entirely up to you.
00:07.01WIMPyYou can Goto() another context or extension, you can include => other contexts or just do you thing there.
00:08.56xochilpiliyes, i get it, my doubt is that if a [test-context] what do i need to put if i just want to playback(vm-goodbye) once if gotoIf success; i mean, in Cli i see that gotoif is working correctly, but in the goto(test-context) i dont understand what should i have to put; exten => s?
00:17.20wyoungWIMPy: I have a few macros, should I replace them?
00:17.49*** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com)
00:18.39xochilpilioh i get it !
00:18.47xochilpilithis is working! ASterisk Rocks!
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00:25.31*** mode/#asterisk [+o mjordan] by ChanServ
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00:47.38wyoungxochilpili: It is that
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11:22.20*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
11:22.42*** join/#asterisk marlinc (~marlinc@ip2.weert.li.nl.cvo-technologies.com)
11:22.58_fuzWhat will happen if asterisk is hitting a line like this:
11:23.19_fuzexten => _330X.,n(sendcid),NoOp(${CALLERID(num)}})
11:23.39_fuzwith a wrong syntax?
11:23.40Chainsawbrgtt: If you have the log files, potentially.
11:24.11_fuzignore it? stop reading the section? or something else?
11:24.21brgttI have logs on /var/log/asterisk but I cant see the numbers I have called there.
11:24.37brgttI know because of VoIP provider logs.
11:24.42Chainsawbrgtt: Check whether CDR has been configured.
11:24.54Chainsawbrgtt: As that is what you are after really. Call Dispatch Records. Who called what number.
11:25.22brgttWhere can I check this?
11:25.47Chainsawbrgtt: Check whether /etc/asterisk/cdr.conf exists.
11:26.30brgttI dont have cdr.conf :(
11:26.37brgttSo its impossible to me?
11:27.33Chainsawbrgtt: No cdr_mysql.conf either?
11:27.55brgttthere is no cdr at all in asterisk folder
11:28.17Chainsawbrgtt: Might be able to infer it from other logs in /var/log/asterisk, but it will not be as convenient.
11:28.54brgttI tried to look for the numbers we have called there and I cant find it
11:29.02brgttSo I believe I dont have it on my logs
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11:31.26Chainsawbrgtt: Shorten the number by removing any leading & trailing zeroes when you grep.
11:32.13brgttI tried a grep with the last 5 numbers. Didnt find anything
11:32.43brgttIs it possible to see what was the last time any username (for us 4 digits) was used?
11:33.50brgttBecause we have a clue on one of our usernames was set with a really easy password
11:35.23brgttChainsaw: hey I really appreciate your help!
11:36.17Chainsawbrgtt: It seems the log level is relatively low. As such it may not be possible to answer that.
11:36.24Chainsawbrgtt: Three things you need to do, right now.
11:37.01Chainsawbrgtt: 1) Change those weak passwords. 2) Configure CDR logging. 3) Increase your log level and make sure you rotate the log files to cope with the increased amount of information.
11:37.24brgttChainsaw: thanks dude
11:38.58brgttthe password was already changed. I will work on enabling CDR log now
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11:40.06brgttcdr show status on my CLI says it is enabled
11:44.37ChainsawBut you have no /etc/asterisk/cdr.conf; that suggests you have a GUI of some kind on top. Which one?
11:46.42carrar_fuz: Looks at raw packets from y our capture server
11:46.53brgttI dont have GUI
11:48.17Chainsawbrgtt: If CDR is configured, that is good news. As it will have the information you seek.
11:48.51carrarerr that was ment for brgtt, not _fuz, sorry _fuz
11:49.32Chainsawbrgtt: What backends are registered for CDR?
11:49.50_fuzcarrar: you can do anything today :-)
11:50.01carrarWell I am in the FUTURE
11:50.07carrarFri Sep  4 20:50:07 JST 2015
11:50.32_fuztell me the numbers 50/50 :-)
11:50.57carrar28 8 7 19 16 7   10
11:51.31brgttI could see we are receiving a brute force now
11:51.43brgttWe just blocked an IP from palestina and it stopped
11:52.15brgttWe also saw they trying to connect with the username with a previous easy password
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11:52.46carrarbrgtt: Welcome to the internet :)
11:52.54brgtthahahaha
11:53.14*** join/#asterisk bulkorok (~Adium@89.245.151.228)
11:53.33carrarMight want to setup the security logging and fail2ban at the very min
11:53.55carrarand fire whoever set your current system up :)
11:53.58ChainsawOr just whitelist ranges with your users on.
11:54.00brgttthanks for tips
11:54.12brgttWe will block all the ips and allow just the office
11:54.15ChainsawAnd refuse to talk to the rest of the internet.
11:54.21ChainsawAt least for now.
11:54.47Chainsaw(It certainly did the CPU load on my Asterisk kit a lot of good)
11:54.50carrarWell you don't have to get too nazi with your filtering
11:55.24brgtttoo nazi
11:55.25brgtthahahaha
11:56.20brgttI am going to have lunch now
11:56.25brgttty all for your help
11:56.29brgttI love IRC
11:56.30brgtt:D
11:56.36carrarIRC is Horrible
11:56.43brgttdont say that :(
11:56.45carrarhaha
11:56.57carrarI sucks all your time
11:57.13carrarerr
11:57.17carrarIT
11:57.32carrarIRC consumes important time
11:58.33carraractually IRC is great
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11:59.25carrarWithout IRC it would be just me in my mothers basement
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12:00.28carrarpokes Katty
12:01.10_fuzWARNING[15566] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_GE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
12:01.11_fuz<PROTECTED>
12:01.31KattyShh it's nap time still
12:01.50_fuzI have some of these messages in the log
12:02.17Kattyhugs carrar, goes back to sleep
12:02.48_fuzand a lot of these statements: 8001,n,GotoIf($[${Calls_In_Queue} >= ${Queue_Max_Len}]?100)
12:02.51carrar!!
12:03.58_fuzis there a way to find out which of those lead to the error?
12:04.37carrarperhaps uninstall FreePBX and install asterisk from source
12:07.09_fuzfor the moment i cant. the machine is production
12:07.31carrarhttp://community.freepbx.org/
12:07.50_fuzic
12:08.35carrarThats a FreePBX issure
12:08.37carrarissue
12:09.20_fuzdo you know what asterisk will do if it hits a non valid syntax line? ignore the rest or what action is taken?
12:09.24carrarMight also try the #freepbx channel
12:09.33_fuzi hear you
12:09.38_fuzthanks
12:10.21*** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net)
12:11.17carrar_fuz: Depending on the severity of the issue it may do nothing to all the way to crash
12:12.46carrarIf only Asterisk had been written in erlang with self recovery code ;)
12:13.31carrarMorning TK!!
12:16.07*** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it)
12:17.25[TK]D-Fender_fuz : is there a way to find out which of those lead to the error? <- cleary the first variable is BLANK in those cases
12:20.24_fuzis this the case when there are no surrounding double quotes? Like in this statement: exten => s,n,GotoIf($[${Calls_In_Queue} >= ${Queue_Max_Len}]?100)
12:21.05_fuzi am on 1.2.16, maybe you remember, tk
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12:27.22[TK]D-Fenderthe variable is blank...
12:27.29[TK]D-Fendermake sure it isn't.
12:27.47[TK]D-Fenderexten => s,n,GotoIf($[0${Calls_In_Queue} >= 0${Queue_Max_Len}]?100)
12:27.52[TK]D-FenderThat may also do it
12:28.01[TK]D-Fenderthe 0's should just get calc'd in
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12:31.32_fuzI got it. Will it be possible also to set the variable initial value at the beginning of the dial plan? Like Calls_in_Queue=0? I do have statements  in extensions, which alter the value later.
12:34.44[TK]D-FenderGood idea depending on how it is used
12:36.55_fuzSet(Calls_In_Queue=${QUEUEMEMBERCOUNT(xyz)})
12:37.08_fuzit is set this way ^
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12:38.36_fuzbut just in a lot of extensions. nevertheless the variable has the same name in every extension, so i guess it is not possible since it is a local variable for each extension, right?
12:40.09[TK]D-Fenderhuh?
12:42.52*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
12:44.53_fuzthe variable "Calls_in_Queue" is set in several extensions, always based on the QUEUEMEMBERCOUNT of each specific queue. If I initialize it on the very top, will it affect every instance of the variable, even if these are in the "scope" of particular extensions?
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12:46.21[TK]D-FenderAre you sure it's actually getting SET?
12:46.31[TK]D-FenderBecause I wouldn't think that function would return "nothing"
12:46.46[TK]D-FenderAnd there is no such thing as "scope" in the dilplan
12:48.54*** part/#asterisk tnt (~tnt@osmocom/tnt)
12:49.44_fuzNo I am not sure it is set under all circumstances, so I would prefer the solution to initialize the value with 0 at the very beginning if you  also think this is worth a try.
12:50.21[TK]D-FenderIf you're not sure your actually setting it you have a bigger problem
12:50.27[TK]D-Fendergo look at your dialplan properly
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12:56.21_fuzthis dialplan is big. I try to avoid applying your solution on each line the first impression could be BLANK. I have not checked every occurence of the SET-Statement there, this is why I said I am not 100% sure about. If the initial value is set, than I can be sure that the value is set without checking every single line where SET() ist called, right?
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13:00.37[TK]D-Fenderhuh?
13:00.46[TK]D-FenderTaht function should not be retuning "northing"
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13:03.29[TK]D-Fendermake sure that it is actually being set in all extensions that could lead to be it being evaluated
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13:31.48brgttSo.. We werent hacked. The VoIP provider got hacked and are refunding us now. :)
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13:33.33mutilatorhah
13:33.41carrarwhich provider?
13:33.48mutilatorrefunding you how much
13:34.05mutilatoror just removing the charges?
13:34.47brgttrefunding on credit card
13:34.52brgttvoicheap
13:34.59brgttvoipcheap
13:35.02brgttis the name
13:35.30carrarBut I thought you said they were making calls through your system?
13:35.53mutilatorstolen credentials
13:35.56mutilatorusing his acct
13:35.59brgttI dont know how
13:36.00mutilatormy guess...
13:36.43brgttVoipcheap said it their fault and they will refund us. They also asked us to create a new account and stop using our current one.
13:37.03carraror better yet, switch providers :)
13:37.11brgtthahaha
13:37.14brgttmaybe
13:37.29brgttanyone that is better than my company being hacked :D
13:37.33brgttanyway
13:42.56wyounghi
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13:46.22rie:)
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13:48.42alex999hello .... can someone tell me why I do receive in log: r all the extensions
13:48.42alex999<alex999> and presecestate dows not work
13:49.08alex999WARNING[24657] presencestate.c: No provider found for label CustomPresence
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14:11.11ipalmerhello all, a quick question is asterisk 13 32bit or 64 bit
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14:11.30[TK]D-Fenderyes
14:11.49[TK]D-FenderIt is however you compiled it <------------------
14:11.59alex999yes compiled
14:12.02alex99964 bit
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14:12.58alex999asterisk 13
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14:14.02ipalmerok thankyou, apologies for the stupid question
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14:32.41todwetsprockhello
14:34.30todwetsprockquestion: i imported my speeddials by openpbx->asterisk phonebook, all fine. the phonebook shows up there nicely. now id like to use the speeddial, but no effort. whats wrong ?
14:35.46todwetsprock*i cant use it in only dialing my speedialnumber with my phone :(
14:37.22[TK]D-FenderOpenpbx isn't supported here....
14:37.49[TK]D-Fenderlast I heard is they were a fork of FreePBX from way back and probably isn't supported there either....
14:47.37todwetsprocki get "silence/1&cannot-complete-as-dialed&check-number-dial-again"
14:48.38todwetsprocki mean freepbx, sry
14:48.47[TK]D-Fenderit isn't getting matched by anything and is falling through
14:48.53[TK]D-FenderYou'll have to ask in their support channel
14:49.16todwetsprockyea... :)
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15:36.28alex999Hello Can someone help me with this:
15:36.31alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority 1 to ext-local
15:36.32alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority 2 to ext-local
15:36.32alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority 3 to ext-local
15:36.32alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority 4 to ext-local
15:36.32alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority 5 to ext-local
15:36.32alex999[2015-09-04 17:09:15] WARNING[12693] presencestate.c: No provider found for label CustomPresence
15:36.34alex999[2015-09-04 17:09:15] VERBOSE[12693] pbx.c: Added extension '605' priority -1 to ext-local
15:37.28alex999this is the conf:
15:37.34alex999exten => 605,hint,PJSIP/605,CustomPresence:605
15:38.22alex999what does it mean "no provider found"
15:39.11alex999what should be as provider?
15:40.12[TK]D-FenderPASTEBIN
15:40.15[TK]D-FenderDo NOT flood in here
15:40.18[TK]D-Fender~pb
15:40.18infobotA "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.
15:40.20[TK]D-Fender^^^
15:41.12alex999Ok sorry
15:48.48todwetsprockarr somebody knows if mysql -> load data infile ... is applicable on already filled tables, such the data there will be there in the end, too ?
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15:49.37todwetsprockdont wanna destroy my database ...
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16:10.48alex999can someone help me please.... i am stuck with blf
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16:24.35todwetsprockhave a lot of fun !
16:25.34alex999thanks
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17:00.17liquidempireIs there a way to subvert the enforcement of RFC 5922 within pjsip?
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19:23.15fileuh oh
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20:07.00PenguinHere's a video for your viewing enjoyment.  http://800notes.com/videos/n2GZu0HK_gU/i'm-an-indian-too-telemarketer-prank-call-12
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20:20.28ldchello :o what is a dlink horstbox?
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21:04.25[TK]D-Fenderldc, imminently Google-able
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21:17.05Demon_VoIPHi. What AMI event i can find out the IP address of the registering client in res_pjsip? Chan_sip has the field in event "PeerStatus".
21:18.01Demon_VoIPand UA if possiable too :)
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21:23.54lvlinuxI need help with MySQL CDR connection failing:
21:24.06lvlinuxi'm using odbc
21:24.37lvlinuxand sometimes when a call comes in it gives an error saying it can't connect to the mysql server
21:25.09lvlinuxsays trying to reconnect
21:26.16lvlinuxthen it says it's connected. I'm not sure if it actaually logs the call or not.
21:27.04lvlinuxThe mysql server is on the same machine as the Asterisk instance, so I know it isn't a connectivity issue.
21:27.15lvlinuxAnyone know where I should look to troubleshoot?
21:28.26Demon_VoIPfound. I can use AMI event "successfulauth" to access remoteaddress in case of res_pjsip
21:40.54lvlinuxAfter checking, it does go ahead and log the call.
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21:45.34lvlinuxHere is a pastebin of the console output right after the call hangs up and it tried to log the CDR: http://pastebin.com/etzYSpWw
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22:34.21jeffspeffwhat phones are recommended for use with TLS and SRTP? or are those even the recommended methods for offering secure & encrypted voip?
22:41.43lvlinuxjeffspeff: I like the Yealink phones, particularly the T46G. TLS/SRTP is the standard way of doing encrypted voip. There is also ZRTP but I don't know if there are any hardware phones that support it.
22:42.00lvlinuxBut there are many other good phone models that do work well with TLS/SRTP.
22:42.34jeffspefflvlinux, thanks. How would you compare the yealink phones with polycom's ip 335 and similar models?
22:43.03jeffspeffi've heard of yealink before, just haven't used them or talked to anyone who uses them with *. we use only polycom's right now.
22:43.35lvlinuxBuild quality is probably similar, or maybe slightly better with the Polycom, but the Yealinks have a lot more features (bluetooth and other fun junk).
22:44.22lvlinuxI haven't used models other than the T46G though, so I can't speak for them.
22:44.47jeffspeffhow are they on price-point? I can get an ip 335 (standard 2 line grayscale display non-touch) for about $130 with the power adapter.
22:46.01lvlinuxI think the T46Gs are about $169 or so, but you get g.722, color screen, 6 or more lines, BLF, etc.
22:46.04jeffspeffgoogle shows the t46g going for between $144 and $267
22:46.24jeffspeffwho do you purchase yours through?
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22:46.45jeffspeffthose prices seem to vary quite a bit. i try to stay away from sellers just price gouging.
22:46.52lvlinuxi can't remember who i got them from, it was a reseller through an ebay listing but they were brand new.
22:47.02jeffspeffok, cool. thanks
22:47.03lvlinuxyeah I think around 169 is the correct range
22:47.07lvlinuxnp
22:47.12lvlinuxgtg have a great evening!
22:47.22jeffspeffyou too
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