IRC log for #asterisk on 20151211

00:24.13*** part/#asterisk kharwell (kharwell@nat/digium/x-jvlkqcvvmshikfun)
00:51.39*** join/#asterisk jasonwert (~wert@71.89.137.28)
00:59.20*** join/#asterisk fstd (~fstd@unaffiliated/fisted)
01:02.59*** join/#asterisk Iamnacho (~Iamnacho@ip72-213-62-18.om.om.cox.net)
02:06.56*** join/#asterisk axisys (~axisys@unaffiliated/axisys)
02:07.06*** join/#asterisk axisys (~axisys@unaffiliated/axisys)
02:55.06*** join/#asterisk SeRi (~wtf@unaffiliated/seri)
02:56.59SeRiteam, I have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN)
03:13.34SeRiby the way this is just to my vmail
03:13.40SeRinormal calls work just fine
03:18.06EllenorWTF?!
03:18.09EllenorThat's weird.
03:19.33EllenorSeRi: Either the phone's fuckered, the program itself is fuckered (but PLEASE, for the sake of Our Flying Spaghetti Monster, upgrade to the latest LTS) or you've fucked a configuration line somewhere.
03:20.10SeRiEllenor: LOL, I saw that one comming from a mile away. I am working on the upgrade but I am not all there yet.
03:20.38SeRiEllenor: Its odd as hell... I can make calls just fine
03:20.57EllenorSeRi: did you try turning it off and on again? (semi-serious)
03:21.06SeRiis just calling the vmail that sounds choppy from this phones
03:21.16SeRiEllenor: LOL, Yes many times.
03:21.28SeRiI can try soft phones with the same issue
03:21.42SeRiEllenor: ^^
03:22.25EllenorSeRi: Tried reinstalling from scratch?\
03:23.05SeRiEllenor: Meh... Not worth it. I have a dev system with Asterisk 11 on it that I am migrating my dial plan too...
03:23.15EllenorOkay
03:23.49SeRiEllenor: Thanks though.... I just been lazy as shit to migrate... I am that one that thinks if it works... Why fuck with it... lol
03:24.40EllenorSeRi: Hopefully this will put you in low gear and heave the trottle mode.
03:24.43Ellenorthrottle*
03:25.09SeRiEllenor: +1
03:25.17EllenorSeRi: eh?
03:25.34EllenorSeRi: also stick around the IRC, it's much more fun when you chat AND give/take support. :P
03:26.28SeRiEllenor: Oh I know all about that... ;) I am an all timers around here, Just work and Family are taking my time now days...
03:26.39SeRiold timers*
03:26.44SeRifor got the sed replace command
03:26.47SeRiLOL
03:29.02SeRilet me try tis
03:29.18SeRis/tis/&this/g
03:29.24SeRicrap
03:29.25SeRilol
03:29.27SeRialmsot
03:29.50SeRis/almsot/almost/g
03:29.53SeRinice
03:30.12SeRigithub to the rescue... :)
03:30.25Ellenors/cue/uce/
03:30.28Ellenorwoops
03:39.43*** join/#asterisk ganbold (~ganbold@173.244.215.173)
03:40.59*** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl)
04:26.58*** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net)
04:31.15*** join/#asterisk juned (~juned@202.131.119.122)
04:46.50carrarbwhahah  http://imgur.com/gallery/gGjMGGg
06:18.31*** join/#asterisk vader- (~Adium@pool-71-175-67-97.phlapa.fios.verizon.net)
06:24.10*** join/#asterisk elitas (~elitas@213.226.135.203)
06:33.13*** join/#asterisk bof22 (~Thunderbi@LPuteaux-656-1-131-218.w193-251.abo.wanadoo.fr)
06:48.42*** join/#asterisk tparcina (~tomo@212.92.200.41)
07:07.14*** join/#asterisk doome_ (~doome@94-21-135-221.pool.digikabel.hu)
07:32.33*** join/#asterisk goddva (~goddva@77.40.154.242)
07:34.37*** join/#asterisk TheKingOfChaos (~kenneth@185.98.99.40)
07:36.34TheKingOfChaosHi i am trying to config ISDN30 from TDC in denmark i am using TE133F i cant get it to change from T1 to E1
07:38.13*** join/#asterisk hehol (~Adium@ip-178-203-97-214.hsi10.unitymediagroup.de)
07:45.32*** join/#asterisk mirton (~jazz@mirtoff.ru)
07:47.13*** join/#asterisk ganbold (~ganbold@173.244.215.173)
07:50.40*** join/#asterisk Zogot (~Adium@185.21.52.255)
08:06.01*** join/#asterisk Kobaz (~kobaz@its.kobaz.net)
08:11.25*** join/#asterisk defsdoor (~andy@cpc73037-sutt4-2-0-cust62.19-1.cable.virginm.net)
08:17.00*** join/#asterisk K0HAX (~K0HAX@2604:8800:187:1::20)
08:18.05*** join/#asterisk pchero_work (~pchero@109.70.54.56)
08:45.36*** join/#asterisk doome_ (~doome@82.150.48.146)
08:48.42*** join/#asterisk evil_gordita (robert@ip70-188-55-129.rn.hr.cox.net)
09:15.07*** join/#asterisk sekil (~sekil@78.24.104.73)
09:19.26*** join/#asterisk guest1009 (~bounceman@185.32.9.250)
09:19.51guest1009Howdy! How does Asterisk scale with more cores? We have four cores which seems to work pretty hard currently, can we increase them to 8 and asterisk will cope well?
09:21.04guest1009we have about 60 active dialogs and work load about 3.5 - 4.0 on all four cores
09:21.31guest1009source htop
09:31.22guest1009I believe the h323 codec is using some cpu
09:35.40*** join/#asterisk rsanting (~rsanting@217.21.195.34)
09:52.14*** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at)
09:52.50WIMPyTheKingOfChaos: Either you chang the jumper ot there's a module parameter to override it.
09:53.25TheKingOfChaostherte is not any jumper on the card
09:53.42WIMPyguest1009: H.323 is not a CODEC. So it's something else.
09:59.51guest1009Ok WIMPy, ofc you are correct.
10:01.07guest1009When using Asterisk as  b2bua can I pass a sip header from one leg to another? In this case P-Asserted Identity
10:01.18guest1009Is there a smooth way of doing this?
10:01.38WIMPyIf you enable them on both peers they would pass automagically.
10:02.53guest1009same => n,sipaddheader(P-Asserted Identity: ${SIP_HEADER(P-Asserted Identity)})
10:02.56guest1009something like that?
10:03.18WIMPyNot neccessay. Just enable them in sip.conf.
10:03.51guest1009I am unsure of what you are talking about, I mean I did not know you handle headers in sip.conf
10:04.21WIMPyYou configure chan_sip there.
10:04.37WIMPyAnd it already does support PAI.
10:05.14WIMPySee 'trustrpid' and 'sendrpid'.
10:05.18*** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire)
10:05.29guest1009if there is no PAI, will it be ignored? So we do not send an empty PAI
10:06.02WIMPyIf it wasn't there, it would use the 'from'.
10:06.02guest1009Asterisk does nothing when it receives a P-Asserted-Identity header. It ignores it totally no matter what settings you use for "trustrpid" or "sendrpid". It does not copy it from an inbound call leg to an outbound call leg for a bridged SIP-to-SIP call.
10:06.05guest1009:(
10:06.28WIMPyWho says so?
10:06.36guest1009voip-info, I know... they cannot be trusted
10:06.44guest1009Can you confirm this is false statment?
10:06.47WIMPyDid you take a look at the values I gave you?
10:07.28WIMPyLook at sip.conf or wiki.asterisk.org.
10:08.26WIMPyYou have to do it manually if you want it in addition to what's in the 'from'.
10:08.47WIMPyBut PAI is definitely supported.
10:09.08guest1009yeah we have different values in from and pai. From field show the actual number while PAI sends anonymous.
10:09.33WIMPyFunny combination.
10:10.00guest1009Does that mean we need dialplan hacking?
10:10.21WIMPyDepends on what exactely you need.
10:10.58guest1009We need the PAI to copy from incoming leg to outgoing, no matter what.
10:11.55WIMPywell, I guess your version might be better then.
10:18.56guest1009yeah, we just need to check if the PAI header actually exists
10:19.09guest1009before doing anything, or we will be sending an empty PAI header. I do not know about the consequences
10:26.40*** join/#asterisk catphish (~catphish@unaffiliated/catphish)
10:31.55*** join/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65)
10:31.56*** part/#asterisk c0rnoTa (~c0rnoTa@91.221.232.65)
10:36.04*** join/#asterisk saratogga (~saratogga@166.170.47.229)
10:42.59*** join/#asterisk Lachezar (~lachezar@hosting.prolet.org)
10:45.46*** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl)
10:48.26LachezarIs it possible to create SIP users in Asterisk that are not numeric (extensions)?
10:49.28catphishLachezar: yes
10:49.40catphishLachezar: i always name mine with people's usernames
10:50.07catphishthere's no need to name extensions the same as their dialplan numbers
10:50.54WIMPyUsing numbers will only make it easier for attackers.
10:51.24Lachezarcatphish: Hm. So I suppose the requirement to have it all-numeric is somewhat Elastix-specific?
10:51.41catphishi don't know anything about elastix i'm afraid
10:51.59WIMPyThere isn't even a need to have your extensions numeric.
10:52.31catphishbut i use usernames for my SIP peers, then use numbers in the dialplan (although as WIMPy says technically those dont have to be numbers either)
10:53.59Lachezarcatphish: OK. Let me restart: I have access to an Asterisk that has a trunk, and various internal extensions (connected to Hardware phones). I'd like to have a second SIP account for every extension, and send a Message to that account on incoming call.
10:54.44catphishLachezar: you want 1 number to go to 2 SIP accounts
10:55.06WIMPyWhat kind of message?
10:55.25catphishprobably best to avoid the word "extension" and talk about either SIP accounts or numbers, the 2 things are quite separate
10:55.40*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
10:56.28Lachezarcatphish: No, 1 number goes to one account, but when an incoming call comes Asterisk should send a SIP message to the companion account.
10:56.38LachezarWIMPy: MessageSend(…)
10:57.08Lachezarcatphish: Sorry. I'm pretty new at this. I still have no idea what an extension is.
10:57.25WIMPyWell, just do that. However you will probably have a hard time to find someone who knows how to do it with elastix.
10:57.52WIMPy"Extension" is something you can dial.
10:58.00catphishLachezar: just change the dialplan to add the MessageSend() command to your number
10:58.04catphishi have no idea how to do that though
10:58.42WIMPyYou can hang out in #elastix and hope someone else joins in.
10:58.59WIMPyBut it might make more sense to scrap that.
10:58.59LachezarI see.
10:59.04WIMPy~elastix
10:59.04infobothmm... elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org
10:59.23WIMPyOh, no rant?
10:59.44catphishlol
10:59.54WIMPyIf you want a GUI, go for the original FreePBX.
10:59.56catphishdon't elastix themselves have support?
11:00.21WIMPyDon't know. Is it still alive?
11:00.34catphishi have no idea, i develop my own commercial asterisk gui
11:01.00LachezarOK. Asterisk (hopefully) question: I have to use WebSocket+SIP to connect. However the internal Web Server is only set to handle HTTP, no HTTPS, which is why it fails when used from a https:// site. Is there any settings that would make the WebSocket endpoint handle HTTPS?
11:01.37WIMPyProbably tlsenable or something similar.
11:01.54WIMPywould never try to use Asterisk as a web server.
11:02.43LachezarWIMPy: Asterisk is not the web-server, but WebSocket is the only way to connect to it from a Browser.
11:03.15catphishdon't browsers have built in RTP?
11:03.34catphish"webrtp", i don't know if that supports SIP
11:03.45WIMPyIt's called something more specific, IIRC.
11:03.47Lachezarcatphish: RTP is what handles AUDIO/VIDEO. WebSocket is what handles SIP
11:03.54WIMPyThat's the one.
11:04.04catphishLachezar: that makes sense then, cool
11:04.33catphishi assumed webrtp had its own setup protocol, or used SIP
11:04.45catphishbut i guess using websocket makes sense too
11:04.49*** join/#asterisk guest1009 (~bounceman@185.32.9.250)
11:05.28LachezarWIMPy: thank you! There is a tlsenable, but was in another config file. Now I can see what I have to look for.
11:05.33guest1009WIMPy: you know if there is a way to check if a SIP HEADER has a value? We was talking about the PAI example earlier, I was thinking that you could check its length with LEN and then do a simple if statement
11:05.38guest1009${SIP_HEADER(P-Asserted-Identity)}
11:05.38WIMPyNot sure if ebsockets, the way they are, are usefull for anything.
11:05.53guest1009I was wondering if there is a better way
11:06.57WIMPyguest1009: Not unless SIP_HEADER returns somehting that isn't documented.
11:09.39*** join/#asterisk troyt (~troyt@c-24-10-220-108.hsd1.ut.comcast.net)
11:43.13*** join/#asterisk bof22 (~Thunderbi@LPuteaux-656-1-131-218.w193-251.abo.wanadoo.fr)
11:43.29*** join/#asterisk TheKingOfChaos (~kenneth@185.98.99.40)
11:59.57*** join/#asterisk samneo (~samneo@123.63.166.181)
12:10.06*** part/#asterisk Brian- (~brian98@unaffiliated/brian98)
12:13.54*** part/#asterisk rsanting (~rsanting@217.21.195.34)
12:42.50*** join/#asterisk Jesterboxboy (~Thunderbi@chello080109194026.3.graz.surfer.at)
12:46.48*** join/#asterisk vinrock (~vin@unaffiliated/vinrock)
12:59.19*** join/#asterisk fstd (~fstd@unaffiliated/fisted)
13:11.47*** join/#asterisk Abus56rus (~Abus56rus@smtp.oren-sms.ru)
13:22.42*** join/#asterisk bof22 (~Thunderbi@LPuteaux-656-1-131-218.w193-251.abo.wanadoo.fr)
13:23.11*** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net)
13:23.13LachezarHm. I'm feeling quite queezy. Tried copy-pasting part in sip.conf for an extension XXX trying to add a CTRL-XXX as SIP account… Failing miserably(
13:24.18*** join/#asterisk K0HAX (~K0HAX@shellhost.home.englehorn.com)
13:34.57*** join/#asterisk sekil (~sekil@78.24.104.73)
13:37.01*** join/#asterisk Iamnacho (~Iamnacho@ip72-213-62-18.om.om.cox.net)
13:43.43*** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca)
13:43.48Lachezarslaps his forehead hard: transport must have ws and wss at the same time.
14:02.55*** part/#asterisk juned (~juned@202.131.119.122)
14:05.24*** join/#asterisk vinrock (~vin@unaffiliated/vinrock)
14:05.25*** join/#asterisk klictel (~Claude@modemcable250.76-70-69.static.videotron.ca)
14:05.35*** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com)
14:20.04*** join/#asterisk brad_mssw (~brad@66.129.88.50)
14:20.33*** join/#asterisk azerus (~badass@unaffiliated/badass)
14:28.45catphishi wish people would kindly stop trying to brute force my PBX hosts :)
14:41.02*** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl)
14:44.08*** join/#asterisk azerus (~badass@unaffiliated/badass)
14:44.16igcewielingcatphish: let them in, send them to a special context
14:45.03catphishi'd just send all their calls to my desk
14:45.13catphishunfortunately they're invariably not humans
14:45.50*** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net)
14:46.37*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
14:47.14igcewielingexactly
14:48.39*** join/#asterisk GHerbstman (171f2361@gateway/web/freenode/ip.23.31.35.97)
14:51.32GHerbstmanHello all, I am having trouble with queues in Astersik 13.5. Both my static and dynamic phones show a status of invalid. Can anyone help me to start determining the cause?
14:52.54[TK]D-FenderShow us
14:52.55[TK]D-Fender~pb
14:52.58infobotA "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.
14:52.59[TK]D-Fender^^^
14:55.48GHerbstmanWhat info would be helpful? show queue all members are like this: ******* (Local/1076@from-queue/n from ********) (ringinuse enabled) (Invalid) has taken no calls yet
14:56.24GHerbstmanBoth static and dynamic members are always invalid.
14:57.19*** join/#asterisk kolko (~kolko@46.48.58.17)
14:57.41[TK]D-FenderThat is not a "phone"
14:57.48[TK]D-Fenderthat is a queue memeber with a lcaol channel device
14:58.21[TK]D-FenderAnd is usually an issue if you are targeting a dialplan exetnsion that doesn't eexist or if chan_local.so wasn't pre-loaded before app_queue
14:59.13igcewielingfrom-queue?   sounds GUIish
14:59.34*** part/#asterisk catphish (~catphish@unaffiliated/catphish)
14:59.46*** join/#asterisk kharwell (kharwell@nat/digium/x-udndrphqlvztgqly)
15:02.05GHerbstmanThis is an up to date recent install of the FreePBX Distro 10.13.66. I have confirmed chan_local.so is loaded. The associated extension 1076 does exist and is otherwise fully functional.
15:02.05[TK]D-FenderMake sure your modules.conf has a "preload => chan_local.so"
15:02.11*** join/#asterisk cyford (~support@c-73-137-1-6.hsd1.ga.comcast.net)
15:02.15[TK]D-FenderPREloaded
15:02.16[TK]D-Fender^^^^^^^^^^^^^^^^^^^^
15:02.34[TK]D-Fenderand take this into their channel as things get over-written in GUYI-land
15:02.44*** join/#asterisk zerohalo (~zerohalo@2601:199:4200:d92e:cc19:4593:30f3:c525)
15:03.45GHerbstmanpreload = chan_local.so is the second line in the modules section of modules.conf
15:04.18LachezarWhy my SIP clients are receiving messages when connected via SIP/WS, but not when connected via SIP/WSS ?!? DAmN!
15:05.23GHerbstmanthanks D-Fender, are you suggesting I run this by the FreePBX people?
15:08.40*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
15:10.06LachezarMeh. Setting-up WSS in Asterisk has adverse effect. Having external SSL tunnel over the WS works better…
15:11.16*** part/#asterisk lumasepa (~gestoip@193.145.124.30.local.ull.es)
15:13.12*** join/#asterisk CyberJacob (~CyberJaco@bouncer.bluesapphiremedia.co.uk)
15:13.17CyberJacobIs there a way to have a queue that doesn't get answered untill an agent picks up?
15:16.16[TK]D-FenderGHerbstman: "module reload app_queue.so"
15:16.39[TK]D-FenderCyberJacob: "core show application queue" <-
15:17.48GHerbstmanModule 'app_queue.so' reloaded successfully. All members still invalid, no error messages.
15:18.23*** join/#asterisk onebree (d1bf0df7@gateway/web/freenode/ip.209.191.13.247)
15:18.26[TK]D-Fenderunload
15:18.27onebreeHello?
15:18.28[TK]D-Fenderthen load
15:18.49onebreeTesting. Is this working?
15:19.03CyberJacobonebree: yes?
15:19.11onebreeThank you
15:19.24onebreeI have a question on whether something is possible with Asterisk 13
15:19.46onebreeIs it possible to do 1:N video conferencing with Asterisk 13?
15:20.06[TK]D-Fender* only does "follow the speaker"
15:20.13[TK]D-Fenderonebree: ^
15:20.45onebreeThat means, the speaker will be displayed to all, correct?
15:20.51[TK]D-Fenderyes
15:20.55onebreeThank you.
15:21.26onebreeWhat I should say is -- video conferencing with 2+ participants.
15:22.19onebreeWe are looking to incorperate WebRTC as well, through sipML5. But the tutorial provided by asterisk is audio conferencing only. We want audio + video
15:22.59[TK]D-Fenderonebree: Which I've just answered
15:23.06*** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view)
15:24.02onebreeD-Fender I appreciate your answers. But how would I go about setting up video conferencing? I cannot find tutorials on it
15:24.11onebree[TK]D-Fender I appreciate your answers. But how would I go about setting up video conferencing? I cannot find tutorials on it
15:24.36[TK]D-FenderAllow the codecs on all parties.  use a conferencing feature
15:25.05[TK]D-FenderThere shouldn't be a guide for video "conferencing" since conferencing isn't the key aspect
15:25.07[TK]D-Fenderit's just video
15:25.08onebreeI am new to asterisk and have been assigned this project for work. Forgive my novice-ness
15:25.11[TK]D-Fenderthere is no "mixing" of it.
15:27.20onebreeSo I do not need external programs like "app conference", as recommended on voip-info ?
15:28.56GHerbstmanload Queue, I received a couple messages: Member has an invalid ringinuse value, using [the queue] value. Second, Error parsing persistent member string for '8001' (penalty), Then: Queue members successfully reloaded from database.
15:29.00igcewielingonebree: the voip-info is terribly out of date.  Check the official docs or the book.
15:29.43onebreeigcewieling the book covers only up to v11. I need v13 because we plan to incorperate webrtc
15:30.46igcewielingonebree: best of luck.
15:30.58klictelHello all. Is there a channel specific to SIP in asterisk?
15:31.00onebreedoes that mean it is not possible?
15:31.45onebreeigcewieling: does that mean it is not possible?
15:31.55[TK]D-Fenderonebree: Link it exactly please
15:32.16[TK]D-Fenderklictel: No.  Just Asterisk.  And you're already here
15:32.33onebree[TK]D-Fender: Link what? The voip-info page? Or something else?
15:32.40[TK]D-Fenderyes
15:33.11klictelThanks
15:33.34onebreehttp://www.voip-info.org/wiki/view/Asterisk+video
15:34.11onebreeI cannot find many sources detailing how to enable video conferencing for v13. Again, I am new to Asterisk, and have been assigned this task at work
15:34.21igcewielingonebree: https://wiki.asterisk.org/wiki/display/AST/Video+Telephony
15:34.27[TK]D-Fenderjunk
15:34.33igcewielinghttps://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge
15:35.31igcewielinghttps://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support and https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
15:35.55igcewielingonebree: if you keep using voip-info you'll never get this working.
15:37.01onebreeigcewieling: I HAVE seen the sipML5 tutorial. Right now audio webrtc conferencing works, but I did not set it up. (Supervisor did).
15:37.17igcewielingonebree: we are not here to do your job.
15:37.33onebreeI was just asking if you were aware of tutorial links or whatnot.
15:37.46onebreeAs I said, I am new to Asterisk
15:38.01[TK]D-Fendernothing to "tutorial" for this
15:38.06[TK]D-Fenderits TWO dumb settings
15:38.21[TK]D-Fender(maybe one or two more if pjsip is used)
15:38.33onebreeWell, sorry for taking up your time then
15:38.38igcewieling[TK]D-Fender: I think he wants to use WebRTC.
15:38.52[TK]D-FenderCodecs are cordecs
15:38.56[TK]D-Fendercodecs*
15:39.02onebreeYes, we want to use webrtc for video conf
15:39.30[TK]D-FenderSo go allow general videosupport and the codecs in your peers
15:40.27onebreeSo: sip.conf > [general] videosupport=yes
15:40.38onebreeThen just adding the codecs per user?
15:41.50[TK]D-Fenderthat's what I said
15:42.03[TK]D-Fenderthe SAME codec for video.
15:42.12[TK]D-Fender* will NOT transcode so they all ahve to support the same
15:42.19onebreeI am just making sure.
15:43.05onebreeDo I need to configure in Asterisk which codec to use? Say user A supports 3 and user B supports only 1 of those.
15:43.33[TK]D-Fender[10:42][TK]D-Fenderthe SAME codec for video. <-----------------
15:43.35[TK]D-FenderONE
15:43.43[TK]D-FenderAll the same.
15:43.45[TK]D-FenderClear?
15:43.46igcewielingonebree: if the devices use different codecs they can't conference.
15:44.01klictelonebree try to move away from ambiguity in your first attempt... just use one codec, the same
15:44.24onebreeigcewieling: That is good to know, thank you
15:44.52igcewielingonebree: you have an interesting few weeks ahead.
15:45.08onebreeI ask this because the docs list 3 available codecs. How do I know which ONE I want?
15:46.05onebreeigcewieling: I know I do. As I said before, we are going to incorperate sipML5 client-side, and do the CRUD actions for a conference in Rails.
15:46.53igcewielingheh, you have an interesting few months ahead
15:47.25onebreeigcewieling: Why did you change that to months? Anything I should be aware of?
15:48.43[TK]D-FenderonebreeI ask this because the docs list 3 available codecs. How do I know which ONE I want? <- the best one your bandwidth can afford
15:49.21onebree[TK]D-Fender: Thank you, I will talk to my supervisor about that decision then.
15:52.44GHerbstmanD-Fender: is there anywhere I have more detailed information on the cause of the invalid status of the queue member?
15:56.08[TK]D-FenderI just gave you the only 2 things
15:56.20[TK]D-Fenderremove the members and try to re-add them
16:01.41*** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-hfxlmiklfjmkmjlj)
16:02.51*** join/#asterisk jasonwert (~wert@75-134-81-98.static.aldl.mi.charter.com)
16:13.47*** join/#asterisk bof22 (~Thunderbi@LPuteaux-656-1-131-218.w193-251.abo.wanadoo.fr)
16:17.27*** join/#asterisk bof22 (~Thunderbi@LPuteaux-656-1-131-218.w193-251.abo.wanadoo.fr)
16:19.28*** join/#asterisk pchero (~pchero@109.70.54.56)
16:32.55*** join/#asterisk vader- (~Adium@50.232.174.194)
16:33.19*** join/#asterisk karelk (~karel@31.10.157.57)
16:34.39*** join/#asterisk DragonAzul (~DragonAzu@187.208.29.196)
16:37.54*** join/#asterisk pjensen00 (~per@ip-69-178-218-71.far.ideaone.net)
16:41.16*** part/#asterisk DragonAzul (~DragonAzu@187.208.29.196)
16:41.48*** join/#asterisk eofster (~eofster@213.61.153.26)
16:45.54*** join/#asterisk mjordan (mjordan@nat/digium/x-ikvmepwqgbyimdqa)
16:45.54*** mode/#asterisk [+o mjordan] by ChanServ
16:45.56*** join/#asterisk vader- (~Adium@50.232.174.194)
16:51.55GHerbstmanThanks D-Fender for your help!
16:54.01*** join/#asterisk F2Knight (~F2Knight@c-50-139-86-39.hsd1.or.comcast.net)
16:56.24[TK]D-FenderWorking now?
17:06.43stefan2742 days since last segmentation fault on any of the 150 asterisk installations, it's a new record!
17:08.51pjensen002 days for me.  But it was our B2BUA's fault.  :(
17:08.53stefan27looks like I made about 10 jira tickets this year, maybe in 2016 I dont have to make any if everything runs this smooth
17:09.04*** join/#asterisk phonebuff (~chatzilla@c-98-249-178-74.hsd1.fl.comcast.net)
17:10.21*** join/#asterisk wonderworld (~ww@ip-84-119-184-180.unity-media.net)
17:11.57karelkI am logging cdr into mysql
17:12.18karelkthere are lots of usefull information about call, such as clid, duration, etc
17:12.37karelkbut I would like to log the IP address as well
17:12.48karelkfrom which the call came
17:12.52karelkis that possible ?
17:26.54*** join/#asterisk samneo (~samneo@115.184.112.232)
17:32.17igcewielingyes
17:35.53[TK]D-Fenderhttps://botbot.me/freenode/asterisk/search/?q=cdr
18:08.24SeRiI have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN)
18:08.29SeRiBy the way all calls to the outside world work just fine from this wireless phones the issue is only calling my vmail..
18:08.32SeRiI have an odd issue that has been haunting me for a while... I am using Asterisk 1.8.32.3 and all works well EXCEPT when I call my voicemail from any phone other than my Polycom 670 the audio is choppy. IE: My Gigaset or my WiFi SIP Phone. Any ideas? This is all internal to me (LAN)
18:08.37SeRiBy the way all calls to the outside world work just fine from this wireless phones the issue is only calling my vmail..
18:09.02SeRiops
18:09.04SeRisorry about that
18:09.37SeRiI had the sceen up and didnt see the paste :P
18:10.21lvlinuxHave you tried upgrading? 1.8 is gone the way of the dodo...
18:11.45lvlinuxwhat sort of hardware are you running * on?
18:12.18lvlinuxIf it's underpowered, and your codecs are mismatched, it could be transcoding the voicemail files and stressing the CPU? Just a possiblity, but you didn't say what you were running it on.
18:12.49igcewielingSeRi: you're not doing something silly like allowing more than one codec for your phones?
18:12.56lvlinuxThat probably wouldn't apply unless you are using some sort of low power embedded system though.
18:13.45igcewielinglvlinux: IIRC 1.8 had some issues with transcoding to/from Siren, that could cause choppyness.
18:14.03lvlinuxyep sure could
18:15.37igcewieling1.8 was EOL'd 2015-10-21 so it isn't THAT out of date.
18:15.37*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
18:16.30igcewielinghttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
18:16.50[TK]D-FenderEOL is a year after the cutoff for "no more bugfixes"
18:17.14[TK]D-FenderSo nothing that is "broken" has been fixed for over a year
18:17.21[TK]D-FenderDitch 1.8
18:17.23[TK]D-FenderNo excuses
18:17.36[TK]D-Fenderbut, but, but, = BS
18:42.31SeRiigcewieling: I am allowing G722 and g711
18:43.59igcewielingtry only allowing ulaw
18:44.18igcewielingor, if you are outside usa/canada, alaw
18:44.22SeRiigcewieling: whats odd is that I can make calls just fine is just dialing my vmail from this specific phones.
18:44.32SeRiigcewieling: I did. Is the same issue
18:44.49SeRiigcewieling: I only allowed g722 yesterday for testing but previously was only g711
18:52.45*** join/#asterisk vinrock (~vin@unaffiliated/vinrock)
18:54.13igcewielinggets revenge on a customer by giving them exactly what they asked for
18:54.41robmalHe said 'fuck me' and you did?
18:56.10igcewielingNo, I asked them to confirm the MAC of the polycom phone they needed a change to.  I asked them confirm the MAC, since the MAC they gave me did not match the extension they gave me.   They confirmed it was the right MAC, so I made the change.  I knew it wasn't the MAC, but I didn'tfeel like arguing with them.   10 mins later they e-mail saying oh, yes, here is the correct MAC.
18:57.23robmalWell, not my kind of revenge i see.
19:06.08MaliutaLapigcewieling: nice move
19:06.27MaliutaLapigcewieling: now just to burn them with fire for being idiots
19:38.36*** join/#asterisk pchero (~pchero@109.70.54.56)
19:41.35*** join/#asterisk vader- (~Adium@50.232.174.194)
19:50.55*** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0)
20:02.14EllenorSeRi: Upgrade now, is what is being said here.
20:28.47*** join/#asterisk saratogga (~saratogga@12.172.251.103)
20:36.48*** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica)
20:48.34Cuzner-= 7527 extensions (32214 priorities) in 4642 contexts. =-
20:48.42Cuzneri think it's time to clean up the dialplan LOL
20:53.56*** join/#asterisk saratogga_ (~saratogga@mobile-166-171-184-221.mycingular.net)
21:05.37SeRiEllenor: I understand that I need to upgrade but thats not the solution.
21:05.53*** join/#asterisk jonno11 (~Jon@cpc1-walt12-2-0-cust582.13-2.cable.virginm.net)
21:32.19EllenorSeRi: It is. :\
21:46.19*** join/#asterisk _-Jon-_ (~jon@unaffiliated/--jon--/x-4089069)
21:46.22*** join/#asterisk spicyramen (~Adium@216.239.45.81)
21:46.28_-Jon-_Hello all!
21:48.21*** join/#asterisk GHerbstman (171f2361@gateway/web/freenode/ip.23.31.35.97)
21:51.16*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
22:00.40*** join/#asterisk italorossi (~Adium@209.49.207.17)
22:17.38*** join/#asterisk F2Knight (~F2Knight@m8e0536d0.tmodns.net)
22:39.53EllenorHello all.
22:40.38*** join/#asterisk gusto (~gusto@85.233.61.60.dynamic.cablesurf.de)
22:42.12*** join/#asterisk CeBe (~CeBe@dhcp-213-62.vpn.tu-berlin.de)
22:58.32*** join/#asterisk timotheus1 (~roost@unaffiliated/roost)
23:10.00*** join/#asterisk membiblio (~membiblio@108.32.57.2)
23:12.04wyounghi Ellenor, how are you?
23:12.22Ellenorwyoung: Not bad. I wonder what the highest posted speed limit is in your country.
23:12.29EllenorWhatever country that may be.
23:12.36EllenorWell, highest you've observed.
23:14.02*** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor)
23:15.24wyoungEllenor: Why do you wonder this for?
23:15.38Ellenorwyoung: Curiosity.
23:15.41rrittgarnFor a while Montana (United States) didn't have speed limits - due to too many stupid people from other states they now have a limit of 80 on some roads. Texas currently holds the highest of 85MPH in the US
23:16.01rrittgarnpersonally 70 around chicago/illinois for highest observed
23:16.50wyoungEllenor: ok, well when your curiosity becomes on topic I will answer you :)
23:17.48Ellenorwyoung: meh
23:18.03Ellenorrrittgarn: If the US were to metricate the roads, the speed limit on Texas' 85 roads would go down from 85mph to 135km/h, a reduction of just over 1 mile legally travellable per hour, if they wanted to keep signs practical, or it would stay at 136.794km/h as it is now to the nearest 3 dec places.
23:18.05EllenorOw.
23:18.07EllenorHe left.
23:31.09wyoungEllenor: so, how about that asterisl hey
23:31.13wyoungasterisk*
23:31.26Ellenormah
23:31.30EllenorI gave up on * long ago
23:31.35Ellenorhack on top of hack tbh
23:32.26wyoungEllenor: ok, so lets move to another channel then :)  what are you interested in now?
23:35.32*** join/#asterisk Iamnacho (~Iamnacho@ip72-213-62-18.om.om.cox.net)
23:49.24*** part/#asterisk kharwell (kharwell@nat/digium/x-udndrphqlvztgqly)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.