IRC log for #asterisk on 20160803

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07:09.59woopstarGoodmorning all. So I've upgraded from 13.9.1 to 13.10 and after the upgrade is successfully applied, we're seeing that SQL queries trough ODBC is truncated in length. Anyone expirenced this ?
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07:58.05ruiedHello. In the dialplan, after I Hangup() a call I want to execute a system() script. I'm making a callback, when I dial to the pbx it matches my number and then it I want to Hangup() the call and execute a system script CallMeBack.call
07:59.01ruiedhow can I run system() after the PBX hangs up the call?
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08:00.49leon_somebody know why callee cannot send rtp frame( Null Frame ) in asterisk 13.10.0
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12:08.39XATRIXHi guys, can you advice why does my SIP software unable to connect to my asterisk ? https://paste.fedoraproject.org/400657/70225138/
12:09.14XATRIXI'm connecing behind 2 NATs,  WAN-----192.168.10.0/24(NAT)------192.168.137.0/24(NAT)----Softphone
12:10.37XATRIXAnd how does T38 crosses with SIP ? or is it completely different things ?
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12:16.33SamotWhy are you double NAT'ing yourself?
12:16.44SamotBecause now you have to deal with that.
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12:54.32XATRIXSamot: i don't that the point how it was done by the local adms
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15:51.16somepoortechanyone familiar with latency effecting call quality (specifically cpu ready)?
15:52.44scvsomepoortech what do you mean by cpu ready
15:52.46scvcpu usage?
15:53.22somepoortechas in I'm running asterisk on a vmware system and I have high ready times that impact call quality
15:53.58somepoortechI'd like to get some numbers as to what I can expect but everything I've read is more for network latency
15:54.19scvhow many cpus do you have assigned to your guest?
15:54.31somepoortechonly one
15:55.00somepoortechcpu utilization (ghz) is not the problem thankfully
15:55.34somepoortechits time waiting to execute instructions
15:56.19scvyes, it's scheduling related, not necessarily caused by load
15:56.31scvwhat is the hardware like? how many other guests are you running
15:57.38somepoortech12 physical cores (24 with HT) running 50 asterisk servers
15:57.57scvwhich model?
15:58.56somepoortechXeon e5-2630L @ 2.0ghz
15:59.03scvhm
15:59.08somepoortech2x of them
15:59.12scvyou shouldn't be seeing any capacity issues in an ideal situation
15:59.20scvi virtualize asterisk as well, ~50 per node
15:59.31scvbut i'm on X5450s still heh
15:59.41scvthose cpus should be more than capable
15:59.59somepoortechwhat are your ready times like?
16:00.16scvi'm on kvm so no easy stat to compare to
16:00.23scvbut my loads are usually under 2
16:00.36scvif you run top in a guest, do you see any st% reported
16:00.47scvlast percentage listed in the Cpu(s): line at the top
16:01.41somepoortech0.0 on the two heavy systems I picked
16:01.53somepoortechwait
16:01.58somepoortechvm level or host level
16:02.02scvvm level
16:02.09somepoortechok, that was vm
16:02.20scvthe linux console in esxi doesnt really give you much visibility
16:02.25scvsince it isnt really the host, it's another small vm
16:02.53scvanyway, 0 is good, means there isn't any noted cpu overcommit
16:03.32somepoortechactually... I'm off on my number I'm up to 60 vms now
16:03.33scvdo you have dahdi built with your systems or no? dahdi_test may be useful to see if there's timing issues occuring inside the vm
16:03.45somepoortechapparently me telling people we are overcommited hasn't stopped them
16:03.55scvhaha
16:03.59scvtypical :P
16:04.04somepoortechworse... I'm running freepbx on them
16:04.24scvyour setup is sounding more and more like the last guy i did consulting for...
16:04.26scv:p
16:04.49somepoortechapparently it's a common pitfall
16:04.52scvanyway, if it's fpbx you should have dahdi
16:07.51somepoortechIt has been 1000 milliseconds, and we got 1022 timer ticks
16:08.22somepoortechdid timing test 1024
16:08.38scvsounds about right
16:09.04scvis your issue persistent or transient?
16:09.18somepoortechtransient
16:09.45scvyou might need to setup some more detailed monitoring of your guests to get a better clue
16:10.13scvi'm inclined to think you might be getting hit with sudden bursts of load, from freepbx cronjobs
16:10.34scvwe used to run freepbx years ago and had issues like that, strange short periods of VQ problems at random intervals
16:10.53scvfix was just to stagger cronjob execution randomly for each vm
16:11.20somepoortechyeah... I really like how they had a cron to run 10k lines of php code
16:11.29somepoortechevery minute
16:11.46scvsounds like you're doing hosted pbx ?
16:11.52somepoortechyep
16:12.04scvhave you looked at any other options? freepbx is really quite awful
16:12.12somepoortechhave a good alternative?
16:12.18scvwe built ours in-house.. heh
16:12.34somepoortechI was considering that
16:12.43somepoortechbut I don't make pretty guis
16:13.07scvdo your customers actually access their systems?
16:13.19somepoortechyes :-(
16:13.24scvbrutal :s
16:13.42somepoortechalso I don't want to be personally responsible for setting up each customers phone
16:13.53scvoh, autoprov is a must
16:14.03TandyUKautoprov ftw
16:14.08TandyUKi use scopserv for our system
16:14.09scvit's actually a lot easier to integrate it when you spin your own panel
16:14.27TandyUKasterisk under the hood, but its a call centre scale, multi tenant platform
16:14.42scvis it mulit-tenant on a single asterisk instance?
16:14.47TandyUKyes
16:14.50scvrealtime?
16:15.01somepoortechI'm not sure if I can get away with a shared instance
16:15.04[TK]D-FenderNope.  It generates configs
16:15.08scvthat's good
16:15.10scvi was about to vomit
16:15.21[TK]D-Fendernot sure if some of their AGI's actually do DB work really...
16:15.21TandyUKwas gonbna say dunno what you mean by realtime
16:15.32scvi'm iffy on shared instances since the overhead of a vm is really minor
16:15.34TandyUKyes they do
16:15.42scvbut as long as its not realtime its acceptable :p
16:15.58TandyUKwhats realtime? direct editing of asterisks dialplan or something?
16:16.07scvdialplan in database
16:16.11scv+ peers/etc
16:16.31TandyUKah right
16:16.33scvits buggy and leaks memory like crazy
16:16.47scvi've seen several multi-tenant platforms that use a single instance and realtime and they're just terrible
16:16.50TandyUKkinda yes it is, but when i 'commit', it builds all the db values into actual asterisk configs
16:16.56scvyeah that's normal
16:17.00scvwe do it the same way
16:17.10scvso do a lot of other implementations
16:18.03scvi wish $DAY_JOB would listen to me about licensing our platform
16:22.54[TK]D-FenderScopServ was very polished looking when I had it and I'm sure they've only gotten better.
16:23.48[TK]D-FenderGiven the time & focus they put on it and the fact it isn't generically modular it lets them do a pretty solid job on what they do, so it's about as scalable as one could hope on a given instance
16:24.10somepoortechI'll check it out
16:24.11scvif we had a decent UI our product would be great, technically it handles every aspect of providing hosted pbx service, it just looks like an engineer made it
16:24.21scvwhy must UI be so difficult :p
16:24.40somepoortechsomething about pretty factor... maybe include some cat's on it and scroll effects
16:25.22scvAJAX-ify all the things
16:26.17scvhm, scopserv looks pretty retro
16:26.29scvfunctional but not "modern"
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16:26.42somepoortechI can already hear the complaints about it
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16:28.25scvwhat i'd really like to do is release our system as F/OSS but i doubt that'll ever happen :(
16:29.23somepoortechoops just dropped it on github... no idea how that could happen
16:29.36scv:D
16:30.19[TK]D-Fender"Accident" (tm)
16:31.40scvmaybe i can just take what i know already and start from scratch
16:31.47scvtheres plenty of demand for a freepbx alternative
16:31.58scvone that doesn't have NaN security issues
16:32.25somepoortechalso not a fan of the no stable branch
16:32.28TandyUKerm question about hangup codes... out of hours, we want to answer the line, play a message, and then hangup
16:32.45TandyUKdefault is "ISDN 16: Normal call clearing"
16:32.47scvif you're answering the line, the hangup code is just going to be normal clearing
16:33.04TandyUKok cool
16:33.10TandyUKeven though its voip not isdn ;)
16:33.18TandyUKsip*
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17:01.40[TK]D-Fenderdrmessano, https://www.reddit.com/r/Jokes/comments/4vybks/whats_heavier_a_ton_of_bricks_or_a_ton_of_feathers/
17:01.49[TK]D-Fenderdrmessano, Bird facts :)  sounds familiar!
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17:03.05WIMPyWhat "ton"?
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17:10.00[TK]D-FenderSince it's old British humour I'll assume Imperial instead of metric :p
17:10.35WIMPyLOL
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17:13.09drmessanoI was assured the bird was resting after a long squawk
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17:13.38WIMPyWas ist an angry bird?
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17:25.01jrunwhat is this in a dialplan? _#
17:25.05jrunor _*
17:25.19jrunwhere do i find the full list of especial characters?
17:25.47jrunthe wiki page (Pattern Matching) lacks explanation of those i mentioned.
17:26.28WIMPyThose are not special.
17:26.31WIMPy~book
17:26.31infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
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17:31.58[TK]D-Fenderthose are no different than # or * respectively.
17:32.28[TK]D-FenderAnd those are not "special" charaters.. they are literal.  As in the literal buttons on a standard dialpad
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18:03.05pa[TK]D-Fender, btw, i was wondering: is there any plan to get asterisk interoperate with google voice again?
18:04.08paah, it's about changing the ssl method?
18:04.17paso in theory the protocol is still there..
18:04.35[TK]D-FenderYes, Google is always inbetween killing it completely and not
18:05.33pabut atm it doesn't work with asterisk.. is there any trick one can do to get it working again?
18:05.52[TK]D-FenderThere is no "trick"
18:05.52papossibly without having to upgrade asterisk :-)
18:06.02[TK]D-FenderThere are only "set the channel driver up properly
18:06.10paaha
18:06.21paso you mean my old settings aren't good anymore?
18:06.29palet me check if i still get the ssl error
18:07.35pai get WARNING[24753]: res_xmpp.c:3633 xmpp_client_thread: JABBER: socket read error
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18:08.08pai have usetls = yes and usesasl = yes
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18:22.44pa[TK]D-Fender, according to https://www.youtube.com/watch?v=K7nhekOJgFM configuring isn't enough
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18:52.49paok rebuilding that module fixed the issue
18:52.58padrmessano, ^
18:53.05pashould someone stop by and ask again :-)
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21:18.44SpaceInvadersHey is there a "how to find a sip provider" web page to help noobs (like me) avoid all the sip scammers that seem to be out there and identify good providers that offer reasonable pricing and acceptable customer service?
21:20.28robmalIt's called google and you can check it out at http://google.com
21:20.50robmal(Scammers can't afford good SEM)
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21:24.08[TK]D-Fendervoip.ms
21:24.11[TK]D-Fendervitelity.net
21:24.17[TK]D-Fenderflowroute.com
21:24.21[TK]D-Fenderles.net
21:24.32[TK]D-Fenderthese are the common starters
21:24.36[TK]D-Fenderthen we add:
21:24.45[TK]D-Fenderbandwidth.com
21:25.08[TK]D-Fendercallcentric (.com?  .net?) don't recall witch
21:29.25SpaceInvadersthanks!
21:29.45SpaceInvadersrobmal apparently you haven't looked at google thats the entire reason I asked the question
21:30.07robmalI might have better google ;-)
21:30.25SpaceInvaderslol the 1st number I dialed (offered up by google) didn't know what sip was
21:31.31robmalWell...
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21:46.24SpaceInvadersDoes "Every paid LES.NET account receives a Flat-Rate 2 Channel DID!" mean 2 calls to 2 separate phone numbers at the same time?  or 2 calls to 1 number at the same time?  Or something else?
21:47.33SpaceInvadersOr does it mean after you have everything set up for in- and out- calls that they will allow direct dial to two extensions?
21:55.17[TK]D-FenderDID = phone number
21:55.20[TK]D-Fenderyou get 1 number
21:55.30[TK]D-Fenderthat supports 2 calls
21:55.44SpaceInvadersI found https://en.wikipedia.org/wiki/Direct_inward_dial
21:55.49SpaceInvadersI was just having trouble understanding.  Thanks!
21:56.42[TK]D-Fenderthat is also only INBOUND
21:57.01SpaceInvadersI was wondering about that.  Thanks, again :)
21:59.08scvbandwidth.com ftw :>
22:09.40*** join/#asterisk jr220 (~jamesruss@184-89-242-81.res.bhn.net)
22:12.41TandyUKWARNING[16260]: chan_sip.c:4086 retrans_pkt: Timeout on cbd7d754d31f8281b70e27df4f508127 on non-critical invite transaction.
22:12.59TandyUKis there any logging etc i can do which would tell me what 'cbd7d754d31f8281b70e27df4f508127' actually was
22:13.15TandyUKideally either the extension or remote ip which didnt respond
22:13.42[TK]D-FenderYou'll already see in the SIP debug as to what is being retransmitted
22:13.58TandyUKsip debug is off atm
22:14.08[TK]D-FenderThat is a failure of logic
22:14.09scvturning it on will shed some light then
22:14.20TandyUKid just like it if the error message itself was a little more useful
22:14.50[TK]D-Fenderand I wish the restaurant had strawberry ic-cream as an option.
22:14.56[TK]D-FenderNobody is happy
22:15.02TandyUKhaving ext or ip in the message, you could instantly see (without turning on debug) if it was the same phone having connection issues all the time, or random different phones
22:15.34[TK]D-FenderSIP debug it is...
22:15.51TandyUKNOTICE[16260]: chan_sip.c:29834 sip_poke_noanswer: Peer 'am203' is now UNREACHABLE!  Last qualify: 34
22:16.14TandyUKother errors include the extension name, so why not for this one :P
22:16.34TandyUKits just causing work havingto correlate logs in order to find out info thats already know when the message is logged
22:17.03[TK]D-FenderFeel free to post a feature request
22:17.17[TK]D-FenderThen again... this is chan_sip you're referring to
22:17.37[TK]D-FenderSo it's not going to happen unless you basically hire someone or someone feels like doing it for fun
22:17.49[TK]D-Fenderchan_sip = on Death Row
22:17.49TandyUKor i do it myself ;)
22:17.55[TK]D-FenderGo for it
22:18.10TandyUKwhats it being replaced by?
22:18.14[TK]D-Fenderpjsip
22:18.16rmudgettyep.  Patches welcome. :)
22:18.56[TK]D-Fenderstart 2.5 years ago
22:49.55*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
22:49.55*** mode/#asterisk [+o cresl1n] by ChanServ
23:00.55*** part/#asterisk kharwell (kharwell@nat/digium/x-klhdorjuruxumiqh)
23:01.20*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
23:10.45*** join/#asterisk hdon (~hdon@wsip-24-234-30-238.lv.lv.cox.net)
23:11.16hdonhi all :) how can i disable a feature code from the console?
23:11.25jameswfbut remember death row takes foreeeeeeeeeever
23:11.25scvwhich feature ?
23:11.47hdon# to transfer scv
23:11.58scvhdon edit the config and features reload
23:12.15scvor maybe you have to do a full reload
23:12.20scvnot sure if features has its own
23:13.53hdonscv, it seems to. but i notice it says "blindxfer=##" which seems odd because it only takes one pound to activate the feature
23:14.03jameswfstill waiting to deprecate the dialplan
23:14.08scvif you do features show
23:14.10scvwhat do you see?
23:14.11scvpastebin it
23:14.22hdonjameswf, Blind Transfer            #       #
23:14.27hdonoh ok
23:14.56hdonjameswf, http://ix.io/1awZ
23:15.19jameswf<.<      >.> why am I getting pinged?
23:16.44scvhdon: can you pastebin your full features.conf ?
23:16.52scvit may be failing to reload the file for some reason
23:17.01scvcan happen silently if you don't have sufficient verbosity on the console
23:17.45*** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net)
23:17.59hdonscv, hmm
23:18.12hdonscv, our configuration right now is a rescue from an ancient and now dead freepbx system
23:18.26hdonscv, we're working to build a similar configuration on more recent incrediblepbx
23:18.44hdonscv, if it isn't being loaded properly, is it possible that the feature code of "#" is the default?
23:19.46hdonscv, this line is the only occurrence of "blindxfer" string in my /etc/asterisk
23:20.00scv# is the default
23:20.05scvthe left column in features show
23:20.06scvis the default
23:20.10scvand the right column is the current value
23:20.10hdonscv, ahh ok. good thinking then it probably isn't even being loaded..
23:20.13scvits likely
23:20.16scvfreepbx makes the worst possible configs
23:20.21scvi'm very sorry for your loss
23:20.21hdonis there a command i can use to just change it now without reloading configuration?
23:20.26scvno, config is the only way afaik
23:20.26hdonscv, yeah i hate it
23:20.30hdonscv, i'm glad that system is dead
23:20.32scv:)
23:20.35scvgood riddance
23:21.33*** join/#asterisk KaliLinuxGR (~alexandro@188.117.250.184)
23:24.37hdonscv, so i have changed it successfully to "##"
23:24.44hdonscv, but weirdly... i can't activate it now
23:25.08hdonscv, i don't think anybody uses this so i don't think it matters, but ... am i misunderstanding that "##" means dialing "#" twice consecutively?
23:25.20scvno, that's correct
23:25.42scvall my deployments use ## so not sure why it isn't working for you
23:25.47scvperhaps do dtmf debugging?
23:26.33hdonthis is a console output mode?
23:26.37hdonoutput level?
23:26.37scvyea
23:26.47hdoncore set verbose dtmf ?
23:46.34*** join/#asterisk klow (~textual@96.81.150.137)
23:49.25*** join/#asterisk pchero (~pchero@109.70.54.56)

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