IRC log for #asterisk on 20171020

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01:54.17elvisthedjHello - Is it possible to call an agi from within an agi? e.g., I have a script (perl) and I need to use google's asr and would rather not have to integrate the speech recognition into my existing, but simply call it.
01:54.52SamotWhy would you call an AGI from an AGI?
01:55.18SamotYou're using AGI to execute a script that does dialplan functions...
01:55.35SamotIf you're already calling a file via AGI, call the other file via the main file called.
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01:58.44elvisthedjI have an IVR that uses tts and voice recognition as opposed to DTMF's.  My menu.agi that I wrote used googles old api and there is a new working perl agi already written that I'd like to use instead of rewriting my script.  My agi now will prompt a user and then go to a sub &getinput() or whatever... My subs don't work with the current api, so instead of rewriting the routines, I'd just like to call
01:58.50elvisthedjto the already written agi from gethub (also in perl)
01:59.51SamotIf you want to call on other scripts via AGI you either need to have AGI calls for them all..
02:00.06SamotOr call a single master file that includes/calls on the other  files.
02:01.16elvisthedjgotcha.. I'm not a perl guru, but I'm guessing I'll just #include the file and then do something->like->this.  I'll stop googling for agi help and start googling for perl help :)
02:01.21elvisthedjthank you, sir
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07:18.41LiuYanusing asterisk-java to GetVar in HangupEvent (), I can't get the variable value in this event. But I can get value in NewState event while State is Ringing. Does variables get cleaned in Hangup?
07:19.34LiuYanI tried put '_' or '__' infront of variable name, but I got same result.
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08:19.07_8bitsexten => _out-X.,3,Set(CALLERID(all)=${IF($[${CALLERID(all)} = 37037]?370371234:${CALLERID(all)})})
08:19.10_8bitswhis this is not working?
08:19.35_8bits[Oct 20 11:15:37] WARNING[49818][C-00005fdd]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting '-' or '!' or '(' or '<token>'; Input:
08:19.35_8bits"" <3705489465> = 37037
08:19.54*** join/#asterisk UOL (~king@111.68.105.235)
08:19.56UOLhello
08:22.59_8bits<_8bits> exten => _out-X.,3,Set(CALLERID(all)=${IF($[${CALLERID(all)} = 37037]?370371234:${CALLERID(all)})})
08:23.00_8bits<_8bits> whis this is not working?
08:23.00_8bits<_8bits> [Oct 20 11:15:37] WARNING[49818][C-00005fdd]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting '-' or '!' or '(' or '<token>'; Input:
08:23.00_8bits<_8bits> "" <3705489465> = 37037
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08:59.15UOLsip trunk unreachable
08:59.22UOLsometimes its reachable
08:59.35UOLafter some mins its shows un-reachable
09:12.25UOLReceived response: "Forbidden" from
09:16.38TandyUKthe other end doesnt like you for some reason
09:17.34UOLthe other end is VOIP providers
09:17.37UOLsip providers
09:18.03UOLno restriction from their side
09:18.34UOLReceived response: "Forbidden" from '<sip:924232560865lbo@10
09:18.45UOL924232560865lbo is my providers username
09:18.59TandyUKright so youre being told "forbidden"
09:19.06UOLand am trying to call with extension 7001
09:19.09TandyUKask provider why they are giving you a forbidden response
09:19.48TandyUKor do a trace
09:19.54UOLhow?
09:19.59TandyUKanalyse the sip packets and what they contain
09:20.08TandyUKwireshark / tcpdump
09:20.16UOLmethod pleas
09:20.21UOLam totally new in asterisk
09:20.39*** join/#asterisk DonkeyDong (~theplague@unaffiliated/donkeydong)
09:21.07TandyUKor i nasterisk, sip set debug on (for everything) or sip set debug peer <extension or ip> to limit to just one peer
09:21.32TandyUKof if newer asterisk, pjsip ......
09:26.28UOLif i create pjsip trunk
09:26.40UOLhow can i check its reachable or not?
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09:34.02Phil-Workasterisk Ringing() seems to time out "Spawn extension exited non-zero" after 180 seconds
09:34.07Phil-Workany idea how to raise this value?
09:50.15wdoekes_8bits: you'll want to test CALLERID(num), not CALLERID(all)
09:50.50wdoekes_8bits: the expression parser gets this: '$[' '"" <3705489465> = 37037' ']'
09:51.04wdoekeswhen you want it to get this: '$[' 3705489465  = 37037' ']'
09:51.21wdoekeser: '$[' '3705489465 = 37037' ']'
09:55.43UOL[2017-10-19 08:36:13] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (35ms / 2000ms)
09:55.44UOL[2017-10-19 08:38:17] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 35
09:55.45UOL[2017-10-19 08:46:09] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (39ms / 2000ms)
09:55.46UOL[2017-10-19 08:47:13] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 39
09:55.47UOL[2017-10-19 08:56:15] NOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (40ms / 2000ms)
09:55.48UOL[2017-10-19 08:59:19] NOTICE[1948]: chan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 37
10:13.49*** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net)
10:14.46UOLhow to fix it
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10:38.42*** join/#asterisk AsteriskRoss (259d3426@gateway/web/freenode/ip.37.157.52.38)
10:42.53pchero_workSeems like firewall, network or sip device problem.
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11:07.50sibiriaUOL: do you happen to have high packet loss to that "trunk"?
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11:22.16_8bitstry to ping to that trunk ip
11:22.27_8bitsto check if it is online when it goes unreachable
11:30.33UOLfirewall is off
11:30.52UOLi dont know about packets
11:30.57UOLhow can i check them?
11:33.13pawieckiUOL: I missed your post, what's the problem?
11:33.45UOLNOTICE[1948]: chan_sip.c:24444 handle_response_peerpoke: Peer 'multinet-outgoing' is now Reachable. (36ms / 2000ms)
11:33.58UOLchan_sip.c:29963 sip_poke_noanswer: Peer 'multinet-outgoing' is now UNREACHABLE! Last qualify: 38
11:34.46UOLthat's the issue
11:40.23pawieckiUOL: It's a SIP Trunk, right? Is it a remote host? Is your internet connection stable?
11:41.50fileproviding a log with "sip set debug on" will show if the remote is actually not responding or not
11:44.43*** join/#asterisk sekil (~sekil@nat-73.net011.net)
11:46.49UOLyes its a sip trunk with voip providers
11:48.07UOLi can ping voip providers ip, proxy , gateway from my asterisk machine which i created on vmware
11:49.25UOLmy sip trunk settings are
11:49.26UOLtype=peer
11:49.26UOLdomain=multinet.pri
11:49.26UOLfromdomain=10.99.34.216:5083
11:49.26UOLusername=xxxxxxxxxxxxxxxx
11:49.26UOLfromuser=xxxxxxxxxxxxxxxx
11:49.26UOLsecret=xxxxxxxxxxxxxxxx
11:49.27UOLhost=125.209.93.196
11:49.28UOLport=5083,5060, 5160
11:49.29UOLinsecure=port,invite
11:49.29UOLoutboundproxy=125.209.93.196:5083
11:49.29UOLdisallow=all
11:49.30UOLallow=ulaw,alaw,gsm
11:49.30UOLqualify=yes
11:49.48pawieckiUOL: as file mentioned above, set debug on that trunk and show us what's happening when trunk goes unreachable
11:53.08UOLhow to on the debug ¿
11:53.14UOLsip set debug on ¿
11:53.19UOLthat command?
11:53.49pawieckiUOL: yes, and you should know that if you administer asterisk server.
11:54.15pawieckiuhh I mean no, set the debug only on that trunk
11:54.19UOLam new for asterisk
11:56.04UOLsip set debug ip 125.209.93.196:5083 peer multinet-outgoing
11:56.07UOLam trying this
11:56.09UOLbut failed
11:56.26pawiecki'sip set debug peer multinet-outgoing' should be ok
11:56.29*** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net)
11:57.11UOLRetransmitting #2 (no NAT) to 125.209.93.196:5083:
11:57.12UOLOPTIONS sip:125.209.93.196 SIP/2.0
11:57.12UOLVia: SIP/2.0/UDP 10.99.34.216:5060;branch=z9hG4bK24264759
11:57.12UOLMax-Forwards: 70
11:57.12UOLFrom: "Unknown" <sip:924232560865lbo@10.99.34.216>;tag=as17ae315f
11:57.13UOLTo: <sip:125.209.93.196>
11:57.13UOLContact: <sip:924232560865lbo@10.99.34.216:5060>
11:57.13UOLCall-ID: 33a354ff2aa6d7e32d125cf07249ecd7@10.99.34.216:5060
11:57.15UOLCSeq: 102 OPTIONS
11:57.16UOLUser-Agent: FPBX-13.0.192.19(13.12.1)
11:57.16UOLDate: Fri, 20 Oct 2017 11:57:00 GMT
11:57.16UOLAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
11:57.17UOLSupported: replaces, timer
11:57.18UOLContent-Length: 0
11:57.25UOL5060?
11:57.38UOLi have set this to 5083
11:57.45UOLthen why 5060
11:57.50UOLhow can i change it?
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11:58.37pchero_workHow did you set to 5083?
11:59.19pchero_worktype sip show settings
11:59.32UOLfromdomain=10.99.34.216:5083
11:59.37UOLlike this
12:00.05pchero_worktype sip show setting
12:00.18filethose options would not have an impact on getting a response to that OPTIONS
12:00.32fileAsterisk is sending it out and not getting a response back, but only sometimes
12:00.47UOLGlobal Settings:
12:00.48UOL----------------
12:00.48UOLUDP Bindaddress: 0.0.0.0:5060
12:00.48UOLTCP SIP Bindaddress: Disabled
12:00.48UOLTLS SIP Bindaddress: Disabled
12:00.48UOLVideosupport: No
12:00.48UOLTextsupport: No
12:00.50UOLIgnore SDP sess. ver.: No
12:00.51UOLAutoCreate Peer: Off
12:00.51UOLMatch Auth Username: No
12:00.51UOLAllow unknown access: Yes
12:00.51UOLAllow subscriptions: Yes
12:00.52UOLAllow overlap dialing: Yes
12:00.53UOLAllow promisc. redir: No
12:00.54UOLEnable call counters: No
12:00.54UOLSIP domain support: No
12:00.54UOLPath support : No
12:01.03pchero_workOkay..
12:01.04fileand use something like pastebin.net instead of copy/pasting tons of output into here
12:01.16UOLopss
12:01.17UOLok
12:01.20UOLsorry
12:02.00pawieckifile: hmm i'm not sure, but is that setting valid: "port=5083,5060, 5160" ?
12:02.12fileit's not
12:02.19fileit parsed it as '5083'
12:03.37pawieckiUOL: correct that setting - set a single port for SIP commnication.
12:04.01UOLok
12:04.10UOLwell
12:04.13UOLnow am trying to call
12:04.18UOLbut not happening
12:04.26UOLnor any error on debug mode
12:05.14pawieckiUOL: maybe, let's start from the begining - Did you manage to setup your SIP Trunk that you could call in and out?
12:05.27*** join/#asterisk sekil (~sekil@nat-73.net011.net)
12:06.25UOLyes
12:06.28UOLi created
12:06.54pawieckiUOL: no private messages please.
12:09.48UOLokey
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12:17.39UOLpawiecki
12:17.39UOLhttps://pastebin.com/FrFPkVCn
12:17.46UOLtrunk is ok now.. but calling issue
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12:21.05pawieckiUOL: did you get instructions on how to configure trunk to your ITSP?
12:22.13*** join/#asterisk billxx (49958984@gateway/web/freenode/ip.73.149.137.132)
12:24.27UOLi configured the trunk
12:24.43UOLultinet-outgoing/9242325 125.209.93.196 No No 5083 OK (18 ms)
12:24.50UOLtrunk is reachable now
12:25.01UOL2017-10-20 17:24:05] WARNING[1948][C-00000067]: chan_sip.c:23862 handle_response_invite: Received response: "Forbidden" from '<sip:924232560865lbo@10.99.34.216:5083>;tag=as2d9ea1c8'
12:25.02UOLScheduling destruction of SIP dialog '597bb423575ff82c4049f1d462cbf847@10.99.34.216' in 6400 ms (Method: INVITE)
12:29.33pawieckiUOL: Is '924232560865lbo' a valid number?
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14:23.03*** join/#asterisk FarhaadN (~Farhad@82.99.206.194)
14:27.07FarhaadNi have to elastix pbx , in one of them(elastix1) i have pstn card and config , i want to configure this elastix(elastix1) as a pstn gateway , and send call to other elastix(elastix2) , and calls from elastix2 out from pstn over elastix1
14:28.38FarhaadNi create trunk in 2 pbx , incoming calls is ok
14:28.46FarhaadNbut outgoing call not working
14:28.59FarhaadNchan_sip.c:24052 handle_response_invite: Failed to authenticate on INVITE to '"9998" <sip:9998@192.168.200.2>;tag=as0ef8cf3b'
14:29.20FarhaadNthis error accourd
14:29.32FarhaadNanyone can help me?
14:29.48SamotThere's no elastix support here.
14:30.04Samot1) It's a GUI
14:30.07Samot2) It's dead.
14:30.43FarhaadNi now
14:30.52FarhaadNi know
14:31.03FarhaadNbut i need help
14:31.07*** join/#asterisk RovingWriter (~RovingWri@unaffiliated/rovingwriter)
14:31.09SamotYou always do.
14:31.17SamotYou need to hire someone that knows this stuff.
14:34.12FarhaadNso u cant help?
14:34.30SamotIRC/Forums should not be the defacto support channel for voip providers.
14:34.38SamotI don't get how this has become of thing lately.
14:34.50SamotYou have an auth failure.
14:35.00SamotIt's like any other auth failure with a peer.
14:35.03fauxalliance^^small time issue
14:35.20SamotThis is like Asterisk 101/SIP 101.
14:35.40fauxallianceauth failures and one way audio... 80% of the requests.
14:35.54fauxallianceand filing a bug report will not help.
14:36.07SamotIt's not even that.
14:36.16fauxallianceprovider..
14:36.17fauxalliancei know
14:36.25SamotIt's the 80% of those request mainly come from 90% "tech support people"
14:36.26FarhaadNwhy authentication is need?
14:36.31fauxalliancecommercial entities looking for _free_ handholding and spoonfeeding.
14:36.43SamotBecause you didn't configure the system to not use auth.
14:36.55[TK]D-FenderExcept when you do.
14:36.58FarhaadNthis eroor accourd in elastix2
14:37.19SamotThen start looking there.
14:37.25[TK]D-FenderFarhaadN, your SIP peers do not agree.
14:37.30SamotAnd get someone that knows Asterisk
14:37.31[TK]D-Fenderyou can't jsut look a one side
14:37.36FarhaadNi just need outcall from elastix2 to elastix1
14:37.41SamotOK
14:37.52SamotYou just need to fix the auth issue for that to happen.
14:38.07SamotThis isn't the first time we've had to help you peer two PBX systems.
14:38.20SamotI'm not sure why you need that help more than one or two times.
14:38.41SamotYou should have taken notes, remembered some of the things we've walked you through.
14:38.55FarhaadN[TK]D-Fender : i config every trunk with this doc : https://www.elastix.org/docs/elastix-pstn-card-gateway-3cx/
14:39.33FarhaadNSamot: another time, i was call 2 extentions in 2 pbx
14:39.41FarhaadNi want*
14:39.51FarhaadNand use iax peer
14:39.56SamotDoesn't matter.
14:40.00[TK]D-FenderThat guide has some stupid stuff in there
14:40.11SamotWell that guide is also for the other side being 3CX
14:40.30SamotSo completely irrelevant if you don't understand that core concepts to begin with.
14:40.51FarhaadNSamot: i just config elastix side from doc
14:40.53SamotSo a lack of knowledge and a guide that isn't even for this type of setup.
14:41.04SamotYou need to configure the OTHERSIDE PROPERLY
14:41.28SamotYou can't just take a doc for A and Z and then make two A's that will connect.
14:41.49FarhaadNno, config two side
14:41.51SamotSpecially when Z isn't the same as A so how A connects to Z would not be how A connects to another A
14:41.58FarhaadNbut only incoming work
14:42.14SamotAsterisk <--> 3CX NOT THE SAME AS Asterisk <--> Asterisk
14:42.21SamotThey are two DIFFERENT PBX systems.
14:42.30SamotThey handle crap completely different.
14:42.47SamotHow you send a call from Asterisk to a 3CX peer is not going to be the same as Asterisk to Asterisk.
14:43.01SamotThere will be differences.
14:43.28[TK]D-FenderDepending how the * end was intended to work work in that first scenario
14:43.43davidbowlbyfeels for Samot
14:43.46davidbowlbylol
14:44.04FarhaadNwait i screen shot that
14:44.40davidbowlbyHave you tried turning it off and back on again?
14:45.12SamotAgain..
14:45.29davidbowlbylol
14:45.33SamotThis is basically "Be my support team" for someone working at a VoIP provider.
14:45.50SamotSome provider is paying FarhaadN to be a support tech.
14:46.00davidbowlbyI'm sitting here writing my library to talk to AMI and this mofo is using you guys to support his platform
14:46.00SamotWho ends up in here for about 90% of what needs to be handled.
14:46.48SamotI'm writing an Asterisk/FreePBX module for WHMCS right now.
14:46.54SamotSo yeah, I get that feeling.
14:47.08[TK]D-FenderIf he can't deal with a basic SIP peer setup between 2 servers he shouldn't be administering a PBX for anyone, let alone a company providing those services to others
14:47.17SamotYup.
14:47.23SamotBut that seems to be a thing now.
14:47.38davidbowlbyworse, he can't even google his way into a solution or example
14:47.40Samot"Tech support" people just using IRC/Forums as their source of knowledge.
14:47.59davidbowlbyit's just as ugly in the language channels
14:48.14davidbowlbyppl begging for help with their homework
14:48.16SamotYeah...
14:48.21*** join/#asterisk rmudgett (rmudgett@nat/digium/x-maprkrtdjuawxego)
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14:48.24SamotBut writing a bad PHP for a website..
14:48.33FarhaadNhttp://ibb.co/dvQ3Y6
14:48.34FarhaadNhttp://ibb.co/gLpxt6
14:48.37davidbowlbyor "what's the best practice" turns into we're doing it for you
14:48.45SamotIsn't having crappy 911/emergency routing that breaks calls for life and death needs.
14:48.57SamotVoIP != Web Hosting
14:48.57davidbowlbySamot, very true!
14:49.02NirSGuys, everybody starts somewhere - the guy maybe a little clueless right now - but every newbie is
14:49.20SamotHaving Asterisk+Billing software does not make you a VoIP provider.
14:49.24davidbowlbyNirS, but you play with it, you READ, you GOOGLE, you TEST
14:49.35SamotNirS: FarhaadN is NOT A NEWBIE
14:49.39SamotThat's the point.
14:49.51SamotRepeat offender is the right term.
14:50.18NirSoh, in that case, off with his head and RTFM
14:50.21Samot1) Not a newbie 2) Has a job as VOIP SUPPORT TECH
14:50.37davidbowlbywow lol
14:50.47SamotThis isn't Gary who just got stuck or wants to learn Asterisk.
14:51.03SamotThis is a person who went out and is getting paid to be a support tech with Asterisk knowledge.
14:51.16SamotWho doesn't appear to having a lot because they are always in here asking for help.
14:51.17NirSI can point him to packt publishing and read the two books I published, but that would be self promoting, wouldn't it ?
14:51.24davidbowlbyFarhaadN, cook up some ViMs and play, bro
14:51.36davidbowlby*VMs
14:51.57[TK]D-FenderNirS, Technically, but if it's valid then I  couldn't fault the recommendation
14:52.01FarhaadNi cant underestand every your word
14:52.17SamotFarhaadN: We're saying get someone that knows this stuff.
14:52.45SamotFarhaadN: You need to either seriously start learning or tell your bosses this is beyond you and they need to get someone that knows Asterisk.
14:52.49FarhaadNcan u help me? or only to make fun of
14:53.03SamotCan't we do both?
14:53.17SamotThere is a *cost* for my time, you know.
14:53.19[TK]D-FenderFarhaadN, Fix your peers
14:53.33SamotBeing the butt of the jokes is rather a cheap cost.
14:53.48FarhaadNSamot: every time i enter this channel you did that
14:53.51[TK]D-Fenderand look at the actual call.  You showed nothing which implies you didn't REALLY look in the first place.  Especially based on that single line you considered relevant
14:54.05SamotFarhaadN: Because the point is still the same.
14:54.05FarhaadN[TK]D-Fender: what peer?
14:54.12SamotOh FFS.
14:54.14[TK]D-FenderBOTH
14:55.16[TK]D-FenderIt's a bad sign when a guy shows up applying to be to be a commercial truck driver ... and then asks "what's this steering wheel you speak of?"
14:55.56FarhaadNi dont know whats the meaning of fix your peers
14:56.16FarhaadNwhats wrong
14:56.25FarhaadNcontext, type
14:56.30FarhaadNor other thing
14:56.37SamotHow do we know?
14:56.41SamotAnd how should we know?
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14:56.45SamotThey aren't OUR BOXES
14:56.50SamotYou have shown NOTHING
14:57.00SamotAfter all the times you've been helped..
14:57.08SamotWe still have to DRAG DETAILS FROM YOU
14:57.19SamotYeah, I'm done.
14:57.24SamotYou guys have fun with this.
14:57.59FarhaadNSamot: i dont know what you need to can help
14:58.13SamotI don't need anything.
14:58.19SamotI'm not helping.
14:58.33FarhaadNSamot: yea you just for make fun
14:58.53SamotI've helped you in the past.
14:59.16FarhaadNafter any word you want to tell me
14:59.19SamotYou've made zero improvements over the last year or so I've helped you.
14:59.43SamotIt's always the same steps over and over again. Having to explain each of them all the time.
14:59.46SamotYou're not learning.
14:59.54SamotI'm not enabling that anymore.
15:00.31FarhaadNSamot: so you silence when i ask a question
15:00.37FarhaadNand let other help me
15:01.07SamotI'm not stopping others.
15:01.12SamotThey are free to help you.
15:01.52voipmonkGood luck
15:04.09FarhaadNSmote: tanx for be quiet in the future
15:04.26*** part/#asterisk FarhaadN (~Farhad@82.99.206.194)
15:04.31SamotLOL
15:04.45voipmonkHe smote himself
15:04.59SamotYup.
15:05.07SamotHe realized no one was going to do his job.
15:05.33SamotIn the next 12 hours, watch.
15:05.49SamotHe'll be back hoping to ask when "fresh" people are around.
15:06.15davidbowlbyHi group, can you just put money in my pocket?  Thanks
15:06.17davidbowlbylol
15:06.25SamotI just don't get how this is a thing now?
15:07.02davidbowlbyI think it's been like this since the early days of IRC
15:07.06SamotWith some many advances in tech I don't get how these providers can't get qualified people in the door.
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15:07.13voipmonkYes you do, it's pure laziness
15:07.15SamotNaw..
15:07.22SamotIt's gotten worse.
15:07.36SamotMy main client base *is* providers.
15:07.42voipmonkReading is becoming a super power like sense that used to be common
15:07.50davidbowlbyI assume it's gotten worse in this channel with so many trunk providers being out there
15:07.51SamotWhen then end up with guys like FarhaadN
15:07.55Samot*they
15:08.06SamotI usually end up getting guys like FarhaadN fired.
15:08.12davidbowlbybut it is global, I blame the 140 character society
15:08.31SamotOne place had a guy that did all his setups via copy and paste what he found on google.
15:08.39SamotIf he found a blog that said X could be done.
15:08.47SamotHe would tell customer they could do X
15:08.52SamotWithout actually testing it.
15:08.54SamotTrying it
15:08.55davidbowlbylol
15:09.00davidbowlbyyeah, I can see that
15:09.15davidbowlbyhell, I worked for a managed service provider and their sales ppl screwed us all the time like that
15:09.19SamotThen the owners are wondering why clients are bitching them out..
15:09.28davidbowlbyheheh yeah
15:09.32SamotWhy they are spending sooo much money on support...
15:09.45Samot^^ Biggest question providers have.
15:09.50Samot"Why is support so costly?"
15:10.02Samot"Because your team takes 2 hours to do 15 minute tasks"
15:10.23[TK]D-FenderThat's a gnerous round-upt to 15...
15:10.29SamotYes.
15:10.29[TK]D-Fendera peering setup shouldn't take 5
15:10.37SamotBut that includes updating the ticket.
15:10.40davidbowlbyhaha
15:10.47SamotI mean as a support tech
15:10.52SamotFrom open to close 15 minutes.
15:11.05SamotAnd yeah, that's more a high average for most tasks.
15:11.16[TK]D-FenderStill quite generous, but he does type kinda slow...
15:11.17*** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net)
15:11.30davidbowlbysounds like this guy has been in here a lot, you'd think he would document some of the solutions you folks have given him
15:11.39SamotBut if you have to google or log into IRC each time a client can't register their endpoint..
15:11.44SamotYou're bleeding money.
15:11.45voipmonkJust search the log
15:11.52SamotHemorrhaging.
15:12.20SamotBut then again..
15:12.33[TK]D-FenderWe've got a number of people who've been around here for YEARS (some 7+) who are still absolutely clueless
15:12.39SamotI spent over 15 years being a one man tech support team to 50K+ end users.
15:12.41[TK]D-Fenderthe most basic stuff
15:13.12SamotSo being efficient kinda became required.
15:15.05SamotThe ideal tech support team should be spending the majority of their time working on revenue generating stuff.
15:15.15SamotAnd handling very few real issues.
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15:50.30davidbowlbywas talking to a coworker about how computers used to come with programming books
15:50.50davidbowlbyand now they come with icons and pictures to show you how to plug it in
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18:54.37nafg__Hi
18:55.13nafg__Does asterisk still prefer audio files to be in elm?
18:55.22nafg__Err autocorrect
18:55.28voipmonkelm?
18:55.32nafg__Sln
18:55.51nafg__Phone autocorrected it
18:56.00nafg__:D
18:57.01nafg__Does asterisk still prefer sln
18:57.48[TK]D-Fender* prefers audio files to be in the codec of the call
18:57.55nafg__It says that online but most asterisk information online is really old
19:00.00voipmonkT/P https://wiki.asterisk.org/wiki/display/AST/Asterisk+Audio+and+Video+Capabilities     Transcode  / Pass through
19:01.35nafg__I'm using flowroute for did. Docs say "Supported media codecs are G.711-ulaw and G.729"
19:02.38[TK]D-Fender<[TK]D-Fender> * prefers audio files to be in the codec of the call <---
19:03.26nafg__Right so does that mean it should be one of those two?
19:03.51[TK]D-FenderYou should know what codec you rstricted your trunk to and have files to match
19:04.04[TK]D-FenderIf yuo allw multiple then yuo should have those files in ALL of those formats
19:05.20nafg__I don't want to buy g729
19:05.32davidbowlbythe buy bandwidth
19:05.33davidbowlbyhehe
19:05.38davidbowlby*then
19:05.55davidbowlbyflowroute uses G.711-ulaw
19:06.01davidbowlbythat's what I save mine in
19:06.07nafg__Valid point, but I don't need too much bandwidth for this
19:06.27[TK]D-FenderSo use the otehr
19:06.43nafg__What is g711 ulaw in ffmpeg flags and file extension terms?
19:06.49davidbowlbyand I paid for 729, just don't use it a lot for inbound
19:07.04davidbowlbyI dunno, I use Audacity and upload the files
19:07.28davidbowlbycause I'm lazy
19:07.37nafg__Let me Google it :)
19:07.46davidbowlbythere ya go
19:07.49davidbowlbya scholar!
19:09.09*** join/#asterisk DanB (~DanB@clt-195.192.201.144.ip-anschluss.net)
19:10.52nafg__https://superuser.com/q/670561/146949
19:11.47nafg__A commenter there says: @meda If you use .wav as an extension, ffmpeg automatically guesses that you want a WAV container wrapping your PCM audio. If you do not want that, and instead need raw audio data in a .ulaw file, you need to use -f mulaw to force ffmpeg to use the PCM mu-law output format. – slhck Nov 7 '13 at 6:20
19:12.33nafg__So to be clear, g711-ulaw is codec AND a container format?
19:12.52nafg__And does asterisk care about its extension?
19:13.18nafg__How much does the container format matter to asterisk?
19:15.48[TK]D-Fendernafg__, "asterisk convert audio files" <- GOOGLE IT
19:16.20[TK]D-FenderAnd be open to using other common tools for this job, including SOX, and if * can already read yuor base file format, ussing * itself
19:17.00nafg__What does using asterisk itself mean
19:19.56[TK]D-Fenderuse ASTERISK to convert the files
19:20.24[TK]D-Fenderif it can read the source you can convert to any other codec it supports
19:20.41nafg__You mean on the fly?
19:20.51[TK]D-FenderNo.
19:20.57[TK]D-Fenderthat is what we are trying to AVOID
19:21.08[TK]D-Fenderconvert it as a file.
19:21.11nafg__Right. So, then how
19:21.16[TK]D-Fendervia CLI
19:21.20[TK]D-Fenderlook at your command list
19:22.03nafg__That's not an option. I need something my prompt management app can automate
19:23.23[TK]D-FenderWe don;t knwo what app you're using or what it can do.  EVERYTHING sounds to be external from it right now
19:23.29[TK]D-Fendersince you're already lookingat ffmpg
19:23.33[TK]D-Fenderto do the work.
19:23.46nafg__It's my own app
19:24.37[TK]D-FenderSo coding it to use * doesn't sound any different than any other method really
19:24.41nafg__I can just spawn ffmpeg and pipe its stdin and stdout
19:24.52[TK]D-FenderSo pick whatever you want
19:25.43nafg__I can have ffmpeg in my docker container, asterisk is in a different one
19:26.28nafg__I already picked one, and I asked a question before, and you just gave me an unhelpful phrase to google
19:27.15nafg__In quotes it returns one result about digium online converter
19:27.25[TK]D-Fender"ffmpeg ulaw conversion" <- tried that?
19:27.35[TK]D-FenderI see a pile of * related links right away
19:27.45nafg__And without, I get all the ancient voip-info pages I've gone through etc etc
19:28.11[TK]D-Fender<PROTECTED>
19:28.28nafg__Of course the are * related links. Let me repost my question
19:28.45nafg__If you don't know the answer, just say so, telling me to google is not helpful
19:28.47[TK]D-FenderIf syntax changed maybe nobody's documenting it perfectly for you anymore and you'll just have to read up on the minor tweak.
19:29.02nafg__3:11 PM <nafg__> A commenter there says: @meda If you use .wav as an extension, ffmpeg automatically guesses that you want a WAV container wrapping your PCM audio. If you do not want that, and instead need raw audio data in a .ulaw file, you need to use -f mulaw to force ffmpeg to use the PCM mu-law output format. – slhck Nov 7 '13 at 6:20
19:29.12nafg__3:12 PM <nafg__> So to be clear, g711-ulaw is codec AND a container format?
19:29.19nafg__That's my first question
19:29.38nafg__3:12 PM <nafg__> And does asterisk care about its extension?
19:29.44nafg__Was the 2nd
19:29.57nafg__3:13 PM <nafg__> How much does the container format matter to asterisk?
19:30.06nafg__Was the 3rd question
19:30.23*** join/#asterisk FarhaadN (~Farhad@82.99.206.194)
19:31.28nafg__These are things I've spent lots of time googling and there is tons of conflicting, outdated (or likely outdated), or just plain unclear information
19:32.17nafg__I come to IRC after I'm exhausted from googling, hoping to actually talk to experts
19:32.29FarhaadNSamot: for your information , this problem is not my fault, and all of my configuration is that correct, and problem acourd becuase in 2 pbx there is same extention
19:33.13FarhaadNplease when you dont know about somthing , no tell any word from that
19:33.19FarhaadNtnx
19:33.45*** part/#asterisk FarhaadN (~Farhad@82.99.206.194)
19:33.51nafg__[TK]D-Fender: it has nothing to do with syntax. Do you know the answer to any of those three questions?
19:35.26[TK]D-FenderI don't see anything referring to ulaw as being in a container
19:36.03seanbrightnafg__: 1) no, it is not a container format. 2) if it's ulaw, it should have an .ulaw, .mu, .ul, .ulw, or .pcm extension. 3) n/a
19:37.06nafg__[TK]D-Fender: then how do you interpret what the commenter is saying?
19:37.36seanbrightwav is a container format
19:37.58[TK]D-Fenderthey were talking about wav when that word came up
19:38.12[TK]D-Fenderthey then said if you wanted a raw ulaw do X ....
19:38.25[TK]D-Fenderso they drew a clear separation between the two
19:38.33SamotROFL
19:38.45SamotI love when those happen.
19:39.11nafg__seanbright: right. The answerer already had the flag for the ulaw codec but the commenter said that's insufficient
19:39.30nafg__Because then it will be ulaw codec in wav container
19:39.57nafg__Did you guys read the context?
19:40.14seanbrightno, because i don't care
19:40.16nafg__https://superuser.com/q/670561/146949
19:40.22seanbrightis your question "how to convert an audio file to ulaw?"
19:40.29seanbright"... with ffmpeg?"
19:40.54nafg__Partially
19:41.20nafg__I'm also trying to understand things better
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19:42.26nafg__seanbright: could just look at the superuser page (very short) and tell me if you're disagreeing with the commenter or how I'm misunderstanding him
19:42.53nafg__could *you
19:43.35seanbright"ffmpeg -i input -codec:a pcm_mulaw -f mulaw output.ulaw"
19:43.50seanbrightthat looks right-ish, but i don't know anything about ffmpeh
19:43.56seanbrightffmpeg*
19:44.55nafg__seanbright: so if mulaw isn't a container format what's the -f for
19:46.04nafg__-f mulaw
19:46.12seanbrighti dunno?
19:46.17seanbrighttry 'man ffmpeg'
19:46.22seanbrightor /join #ffmpeg
19:46.38nafg__-f is for the container format
19:46.44seanbrightok?
19:46.52seanbrightagain - i don't care
19:47.03seanbrightfor once i agree with [TK]D-Fender
19:47.05seanbrightuse google
19:47.07seanbrighttry it out
19:47.59[TK]D-Fenderseanbright, c'mon we agree a lot more than that!
19:48.19nafg__Anything I'm asking here, I've tried googling etc already
19:48.51[TK]D-FenderHave you tried taking one of the provided ulaw files and using one of your tools to look at it and give you that answer?
19:49.32nafg__Unless "container" is the wrong word. But there's definitely codec vs format
19:49.48seanbrightnafg__: what is your source file format?
19:50.11nafg__wav... Whatever recorder.js produces
19:50.26seanbrightso you don't know?
19:50.33seanbrightbecause as you know... wav is a container format
19:50.53nafg__Not sure offhand
19:51.10seanbrightok, let me install ffmpeg real quick so that i can answer this question
19:51.43nafg__You can view the man page at https://linux.die.net/man/1/ffmpeg
19:51.51seanbrightlol
19:51.52nafg__Thanks
19:51.58seanbrightthis is trolling level 1000
19:52.05davidbowlbyuse audacity
19:52.06nafg__?
19:52.06davidbowlbylol
19:52.23seanbrightnafg__: why are you unable to use the man page and figure this out yourself?
19:53.18nafg__I don't know. Do you think I'm too dumb?
19:53.52[TK]D-Fender-f fmt
19:53.52[TK]D-Fender<PROTECTED>
19:53.59[TK]D-Fenderwell this seems to force format
19:54.08nafg__Really now
19:54.10[TK]D-Fenderthat doesn't imply "container" to me....
19:54.22nafg__Ok, it's not codec though
19:54.24seanbright[TK]D-Fender: no no no, that is clearly the container flag
19:54.28seanbrightas indiciated by nafg__
19:54.30[TK]D-FenderSo forcing a format for your output file ... seems sane enough so far
19:54.39[TK]D-FenderWhat do you feel is suspcious there?
19:54.51nafg__3:49 PM <nafg__> Unless "container" is the wrong word. But there's definitely codec vs format
19:55.05davidbowlbyI like pizza
19:55.08nafg__Who said anything is suspicious
19:55.11[TK]D-FenderSo.... what's your concern with it?
19:55.14[TK]D-FenderDoes it work?
19:55.14davidbowlbysince we're trolling
19:55.24davidbowlbypizza always works
19:55.30nafg__I think I have a misunderstanding about audio file binary layout
19:55.43davidbowlby11pepperoni011
19:55.53seanbrightthe the binary layout of the audio file make any difference?
19:55.57seanbrightdoes the*
19:56.05nafg__Are the 3 things? Codec, format, container format?
19:56.20davidbowlbywhispers "audaaaaaaacityyyyy"
19:56.21nafg__That's what I'm trying to learn about right now
19:57.10nafg__davidbowlby: can you make me a docker image that I can script audacity headless?
19:57.27davidbowlbynafg__ you said headless
19:57.33davidbowlbyno one wants to live like that
19:57.45davidbowlbysets trolling = 0
19:57.55nafg__Servers apparently do :)
19:58.05[TK]D-FenderNo-one GETS to live like that.... for more than a handful of seconds
19:58.31davidbowlbyso you're gonna sit here and tell us you are creating a docker image to do fancy stuff and can't try/fail your way through ffmpeg
19:58.44[TK]D-FenderNone of us are going around container-ing  our solutions in bite-sized pieces using docker and insisting on ffmpeg.
19:58.48seanbrightnafg__: this is what i just tested:
19:58.54seanbrightffmpeg -i input.wav output.au
19:59.05[TK]D-FenderThe list of people we can say have been in that camp so far .... = 1 person
19:59.26seanbrightnafg__: so you said that was partially your question. what is the rest?
20:00.57nafg__And what was the result?
20:01.11seanbrighta ulaw file named output.au
20:01.14seanbrightnafg__: so you said that was partially your question. what is the rest?
20:01.50nafg__So apparently yes, there are 3 different things: https://en.m.wikipedia.org/wiki/Audio_file_format#Format_types
20:02.29nafg__> It is important to distinguish between the audio coding format, the container containing the raw audio data, and an audio codec.
20:03.15seanbrightnafg__: so you said that was partially your question. what is the rest?
20:03.36seanbrighttrying to wrap up this painful interaction
20:03.42nafg__So I guess ffmpeg's -f is about the "audio coding format," which I confused with container formats
20:05.05nafg__seanbright: hence my perceived contradiction between you saying ulaw is not a container format, and the commenter implying it's not only a codec
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20:06.02nafg__In any case it's still two things, ulaw the codec and ulaw the format and those are not necessarily hand in hand?
20:06.38nafg__It's possible to have a file with ulaw codec and other format, and vice versa?
20:07.19nafg__And is it only the codec that matters to asterisk?
20:07.32davidbowlbynafg__ I dunno, try it and find out
20:08.21nafg__Yeah... Was hoping to find people here at a deeper understanding of how asterisk works
20:08.33davidbowlbyexperience is the best understanding
20:08.53davidbowlbywon't you know after testing?
20:09.14nafg__Also because I think asterisk makes assumptions based on a files extension
20:09.30davidbowlbyand two years later when talking to someone who is having the issue, be like, "oh yeah, I had to try this, that, and the other thing, but it turns out.."
20:09.40nafg__davidbowlby: yeah you know you're right, who needs IRC, let everyone using their own experience
20:10.01davidbowlbyyou've asked and you've gotten the answers available
20:10.04davidbowlbywhy keep proddin
20:10.35nafg__No, I was misunderstood because I used the wrong word,
20:10.49nafg__so I asked the corrected question
20:11.09davidbowlbythe folks who responded suggested they don't use ffmpeg, so you're on your own there.
20:11.26nafg__The question was not about ffmpeg sir
20:11.34nafg__I don't know why you think that
20:11.41davidbowlbyit was how to use ffmpeg to achieve your goal
20:11.45nafg__No
20:12.00nafg__It was about what rules asterisk follows
20:12.13nafg__What aspects of files it cares about
20:12.36davidbowlbyto the end of using ffmpeg...
20:12.37nafg__I know ffmpeg syntax. That was never the question
20:12.43davidbowlbyok
20:13.13nafg__No. To the end of learning asterisk more deeply
20:13.31davidbowlbyif you wanna get deep with asterisk, read the code :P
20:13.33davidbowlbyhaha
20:14.03nafg__Yeah, who doesn't love reading C
20:14.19davidbowlbydocker boys, I assume
20:15.07davidbowlbyhey man, it's Friday, lift your spirits and drop your packets
20:15.30fileAsterisk does not support containers and determines the underlying codec based on the file extension, with the files themselves being raw uncontainerized frames or payloads
20:15.45davidbowlbydamn, that was sexy talk right there
20:15.51davidbowlbyanswered your question like *boom*
20:16.14filedrops mic
20:16.51drunkdavidbowlbydamn nick char limit
20:16.57seanbright15:40 <@seanbright> is your question "how to convert an audio file to ulaw?"
20:16.57seanbright15:40 <@seanbright> "... with ffmpeg?"
20:16.58seanbright15:40 < nafg__> Partially
20:17.04seanbright16:11 < nafg__> The question was not about ffmpeg sir
20:17.04seanbright16:11 < nafg__> I don't know why you think that
20:17.14seanbrightgood times
20:17.21drunkdavidbowlby[16:15:30]  <@file>Asterisk does not support containers and determines the underlying codec based on the file extension, with the files themselves being raw uncontainerized frames or payloads
20:17.25drunkdavidbowlbyI can do that too, see
20:17.26seanbrightis exhausted all of a sudden
20:17.33filegives seanbright a cookie
20:17.41drunkdavidbowlbytakes another sip
20:17.57drunkdavidbowlbyI'm gonna have some bugs to fix tomorrow!
20:18.24nafg__seanbright: if you can't reconcile that, maybe sleep on it
20:20.49seanbrightanything else or are you good now?
20:20.56nafg__file: ok thanks. But what about what Wikipedia calls "audio file coding" (distinct from codec; IIUC corresponds to ffmpeg -f flag) -- how much does that matter to asterisk?
20:21.32seanbrightthe only thing that matters to asterisk is the extension
20:21.52seanbrightif it's .ulaw, it will assume it is ulaw data and try to read it as such
20:22.00nafg__Ok thanks
20:22.21drunkdavidbowlbyyou're gonna sit there and tell me if the ext is .ulaw, it's going to *ASSUME* ulaw?!!?!
20:22.45drunkdavidbowlbyand I guess next you're gonna tell me if it is .mp3 it's going to assume mp3
20:22.55filea codec is the name of the encoding and the process, the file coding is the result of the process
20:22.56drunkdavidbowlbyis shook
20:23.05seanbrightmp3 is a container format
20:23.26nafg__Interesting observation (IIUC) - asterisk correlates extension to codec, while ffmpeg correlates extension to format (by default)
20:26.51drunkdavidbowlbyneeds another drink after reading that
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20:34.19drunkdavidbowlbywhy is there a new account code event?
20:34.27drunkdavidbowlbyI assume for billing?
20:34.59drunkdavidbowlbyahh nm, imagine that, looked it up my damn self.
20:35.07drunkdavidbowlbysee how that's done?
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20:54.49voipmonkclones drunkdavidbowlby
20:55.32drunkdavidbowlbylooks at clone... good lookin guy
20:56.13voipmonkwas gonna send him to Jules but he would come back and bslap me for thinking of that as a solution... Clones lives matter
20:56.55drunkdavidbowlbyI heard there was a whole war about it
20:57.28voipmonk[J]oules
20:58.26voipmonkmanifests food inside Restaraunt, ready for pickup - bbl
20:59.08voipmonkRestauhhh hell
20:59.45drunkdavidbowlbylol
20:59.54drunkdavidbowlbylots of folks have problems with that one
21:00.09drunkdavidbowlbyeven if sober
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