IRC log for #asterisk on 20171005

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00:42.24pclerieHello all! I am new to Asterisk and inevitably I need a little bit of help.
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00:44.08kunwon1pclerie: ask your questions and those who see them will likely do their best to answer
00:44.23pclerieI've set up a test system and I'm using a GrandStream 1625 IP phone to test. The phone appears to work only some of the time. Just after a restart of Asterisk I can receive and place call. After a while, I can receive but every outgoing call fails.
00:44.49kunwon1do you have sip qualify set?
00:45.03pclerieAs a matter of fact I do.
00:45.13kunwon1i think it's a different setting i'm thinking of, actually
00:45.15kunwon1hang on
00:45.40pclerieI replaced the GS with a Polycom 331 and that appears to work correctly.
00:46.09pclerieThe system on Ubuntu 16.04 and it's version 13.1.
00:46.57SamotSounds like an issue with the phone not handling the auth properly for the call
00:47.51pcleriesamot: why would it work some of the time then?
00:48.02SamotI don't know.
00:48.21SamotThere could be various reasons. Can't really tell without seeing it fail.
00:50.14kunwon1pclerie: some tips- run asterisk -r -vvvvv and watch the console to see if there is any evidence - and after the problem manifests and you're unable to dial, see if you see anything on the console
00:50.24kunwon1and is this pjsip or chan_sip?
00:51.52pcleriekunwon1: chan_sip. Asterisk won't start with pjsip. But that's another story for now.
00:52.20kunwon1well, chan_sip is much easier to troubleshoot
00:52.29kunwon1sip show peer XXX where XXX is your peer
00:52.36kunwon1when the problem manifests
00:52.43kunwon1that could give hints as well
00:53.06kunwon1and if you want to use pjsip, use the version bundled with asterisk
00:53.11kunwon1using separate pjsip is deep magic
00:53.36pcleriekunwon1: I've used the console with up to 5 v's and sip debug (or something). I've seen the registration packets come and go but I'm not too sure yet about the details of that.
00:53.59kunwon1sip debug is less helpful to people who aren't intimately familiar with sip
00:54.20kunwon1the console output should only show you things that are meaningful, sip debug will show you a crapload of stuff that is not
00:54.33[TK]D-FenderActually the opposite
00:54.44[TK]D-Fendergeneric verbose is just that .. generic
00:54.51kunwon1at the moment you attempt a call, the output from the console is important
00:55.01[TK]D-Fenderif doesn't show you where the screwup is and often barely alludes to the cause
00:55.53pclerieI think I would agree with that.
00:56.51[TK]D-Fender<pclerie> I've set up a test system and I'm using a GrandStream 1625 IP phone to test. The phone appears to work only some of the time. Just after a restart of Asterisk I can receive and place call. After a while, I can receive but every outgoing call fails.
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00:57.05[TK]D-FenderWHY does it fail?  You'd have to look at the packets for the attempt and response
00:59.23pcleriefender: I've tried that. I've tried matching packets to what is happening. But as near as I can tell, there are simply missing packets when it fails that are present when it works. Which is kind of expected... :-(
00:59.48kunwon1do you have the latest firmware for your phone?
01:00.07pclerieOK! I was sort of hoping it was a known issue... :-)
01:00.37kunwon1it probably is, but there aren't enough details to determine
01:01.17pclerieThen I should go back and gather more info...
01:01.58kunwon1console output, sip debug, and sip show peer are useful for others to help you to determine the issue
01:02.04kunwon1phone config too
01:02.22pclerieOK! Will do and will revert ASAP.
01:02.24pclerieThanks all!
01:02.25[TK]D-FenderForget configs right now
01:02.44[TK]D-FenderThat's like seeing a picture of a car before a crash.  You don't see the fact that it gets driven into a wall
01:02.58pclerie:-D
01:02.59[TK]D-FenderGet proper debug for a filure
01:03.07[TK]D-FenderYou want an autopsy?  Give us a body
01:03.11pclerieGotcha!
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08:12.46sotozHi, recently I sent an email at the asterisk-dev list. http://lists.digium.com/pipermail/asterisk-app-dev/2017-October/000892.html
08:13.00sotozIf anyone has an answer on that I would be grateful. Thanks for checking.
08:18.10drmessanoWell, there is a -dev channel
08:38.59sotozsorry my mistake. The message was in the asterisk-app-dev list :)
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13:43.19_8bitsHi, how to set System() result to variable in dialplan?
13:43.24_8bitsI try same => n,Set(test=${System(/var/www/rankinis/test.php)})
13:43.36_8bitsbut it doesnt work :<
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13:49.54sibiriais that php-file executable?
13:50.01sibiriais it +x, does it have a proper shebang?
13:52.00sibiriause SHELL function instead i think
13:52.04[TK]D-FenderSystem isn't a function
13:52.09sibiriaif you need the result and not just the exit status
13:52.14[TK]D-Fenderand functions are all uppercase
13:52.16sibiriathe output*
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13:52.35[TK]D-Fender2 mistakes from the start
13:52.50[TK]D-FenderSystem() is an app and retuns nothing.
13:52.55[TK]D-FenderSHELL() is a function and can
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13:54.21_8bitssibiria, i get pbx.c:4358 ast_func_read: Function System not registered
13:54.31_8bitsbut without Set - System is working fine
13:55.16_8bitsERROR[5736][C-000a1dd2]: pbx.c:4358 ast_func_read: Function Shell not registered
13:55.27sibiriaSHELL
13:55.30sibirianot Shell
13:56.01_8bits**** me....
13:56.09_8bitsI was trying solve this like for 2 hours
13:56.18_8bitsand always tried Shell not SHELL
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14:25.40[TK]D-Fender<[TK]D-Fender> System() is an app and retuns nothing. <----------------------
14:25.49[TK]D-FenderAPPLICATION
14:26.07[TK]D-Fender<[TK]D-Fender> and functions are all uppercase <--------------------------
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14:48.38domlaz11hi guys, trying to understand this part of log
14:48.49domlaz11[2017-10-04 11:46:06] VERBOSE[12928][C-00005ab4] pbx.c: Executing [**301@from-internal:4] Hangup("SIP/303-00005caf", "") in new stack
14:48.49domlaz11303 tried to pickup the call from 301
14:48.56domlaz11and why its saying hangup?
14:49.39SamotOne line of log doesn't tell anything 98% of the time.
14:50.03SamotThat just says the Hangup happened.
14:50.12SamotEverything BEFORE IT tells you why it would be hungup.
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14:50.54SamotYou asked this yesterday
14:51.01SamotYou were given multiple answers yesterday
14:51.55SamotIncluding how to determine which side initiated the hangup.
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15:06.38domlaz11sorry it wasnt the same call
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15:19.07SamotIt doesn't mean the method for checking for the information is different.
15:19.16domlaz11i have one question
15:19.31domlaz11when i see the call coming in, is there a reference number i can look at when a trunk has left the bridge
15:20.06domlaz11"SIP/GoldLineOut-00005ca9    Is the 0005ca9 a unique ID of the call?
15:21.45SamotNo.
15:21.53SamotIt's the trunk name
15:22.02SamotThe unique ID is different
15:22.58domlaz11what it look like?
15:23.10domlaz11[2017-10-04 11:45:05] VERBOSE[12842][C-00005ab1] pbx.c: Executing [in@sub-record-check:3] ExecIf("SIP/GoldLineOut-00005ca9", "10?Set(FROMEXTEN=5143276372)") in new stack
15:23.16domlaz11this is the first line i can see the phone number
15:23.26SamotWhat?
15:23.37SamotFirst you have to call on it in the dialplan to "see it"
15:23.47SamotIt's a channel variable.
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15:24.52domlaz11in the full log i cannot have a way to figure out when trunk left the bridge and associate it with the call i am lookin at?
15:25.21SamotIt's in there.
15:25.37SamotThe problem is, you don't understand how to interrupt/read the log.
15:25.52SamotBecause you have no idea what any of it really means.
15:25.52domlaz11thats why i am asking
15:26.01SamotYou need to go and learn Asterisk's dialplan.
15:26.06domlaz11ok
15:26.18SamotYou need to understand how dialplan works in order to understand what the log output means.
15:27.23SamotEven more, you're using FreePBX..
15:27.40domlaz11yes i am
15:27.46Samotso not only do you have to understand dialplan, you have to understand the dialplan *it* created
15:28.32domlaz11i started to read the book asterisk the feature of telephony but i am not done yet
15:28.45SamotReading it is a good start
15:28.55domlaz11i usally understand what i read
15:29.11SamotBut *just* reading it doesn't give you real-time experience.
15:29.27SamotBecause the book doesn't teach troubleshooting
15:29.45domlaz11i have to start somewhere
15:29.50SamotYes
15:29.52domlaz11the way you talk i should quit
15:30.01SamotBut supporting customer systems is not the place.
15:30.18SamotYou need to have a system you can *literally* break.
15:30.20domlaz11its small systems
15:30.25SamotDoesn't matter.
15:30.33SamotPeople are paying you for an expectation of service.
15:30.41SamotAnd knowledge, experience, skillsets.
15:30.56SamotNot to have you come into IRC and ask random people to fix their issues.
15:31.05domlaz11you are very special my friend
15:31.23SamotYou want to learn Asterisk, I 100% encourage that.
15:31.30SamotOn your own time and dime.
15:31.47SamotNot at the expense of unknowing end users.
15:32.34domlaz11thats not the point
15:32.43SamotOh?
15:32.47SamotWhat is the point?
15:32.48domlaz11i am not asking you to fix my issue
15:32.55domlaz11just asking question
15:32.59SamotNo, you're asking me to explain it.
15:33.06SamotYou can't figure out how something happened.
15:33.12SamotYou need to go and learn.
15:33.15SamotWe had this talk.
15:33.29domlaz11what is the purpose of this channel
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15:33.35SamotI said don't do calculus before you learn algebra..
15:33.43SamotTo help.
15:33.49domlaz11many people use freepbx without knowing asterisk
15:34.02domlaz11i am learning both
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15:34.15SamotWhile having people paying you services.
15:34.22SamotThat you have to ask questions about.
15:34.34SamotWhy do you need to know why the hangup or when the hangup happened?
15:34.38domlaz11no
15:34.57SamotWhy did you need to ask the question you asked?
15:35.17domlaz11because apparently when she tried to pickup the call no one was there
15:35.28domlaz11i was wondering when the call ended from the sip trunk side
15:35.29SamotAnd the issue yesterday?
15:35.39SamotBecause it's the same trunk.
15:35.40domlaz11ok let it go
15:35.42SamotSame customer.
15:35.51SamotSo you're asking a question because your customer has an issue
15:35.52domlaz11im ok thanks
15:35.57SamotAnd  you don't know how to answer it.
15:36.12SamotSo you're asking us.
15:36.17SamotSo you can tell your customer.
15:36.39domlaz11nope
15:36.44domlaz11i can ask the senior
15:37.02domlaz11i just wanted to have some tips
15:37.06domlaz11not you to tell me what to say\
15:37.21domlaz11im not using you to get money
15:37.24domlaz11thi sis ridiculous
15:37.26[TK]D-FenderSo far what you've shown doesn't tell us anything about that call
15:37.43domlaz11it's ok i will stop
15:38.28[TK]D-Fender<domlaz11> in the full log i cannot have a way to figure out when trunk left the bridge and associate it with the call i am lookin at? <- Channel name
15:39.23domlaz11thanks fender
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20:37.15SoBlindWolfDo you guys know if there is a free automated voice assistant for FreePBX 13 and Asterisk 13? Such as Speech Recognition for voice prompt?
20:38.52SoBlindWolfSorry version 14 for both
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22:12.27eharrisis there a way to make Asterisk cdr log inbound calls that ring for only a few seconds, not long enough to be answered by voicemail?  There don't seem to be cdr records generated for those.
22:12.56[TK]D-FenderYou can log whatever you want in your dialplan
22:13.28[TK]D-FenderAnd if you wnt to log unanswered calls there is a setting in CDR for that too
22:13.59eharrishave a keyword for me to lookup?
22:14.15[TK]D-Fenderthose are the keywords
22:14.20eharrisah, I see it.  Thanks.
22:14.33[TK]D-FenderThat's what the sample configs are for
22:14.35[TK]D-Fenderread them
22:15.13[TK]D-FenderThey are all commented in plain writing
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23:56.54hdonhi all :) when my asterisk sends SIP INVITE i'm not getting the Contact header that I want sent. what's the dialplan interface to the Contact header? is that Set(CALLERID(...)=...) ?

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