IRC log for #asterisk on 20120926

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00:21.55aster1skBack
00:21.58aster1skYou around.
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00:22.20aster1skwoops, forgot to ctrl+x
00:31.22lorsungcuwat do
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02:02.23PlatinumAnyone have experience with asterisk replacing audix?
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02:50.06Maxus2Hi People, Does anyone know if there is a module to allow cdr logging via curl? and if so where is it?
02:51.31ectospasmcdr_curl was a project I was working on with Corydon76-home, don't know if it ever got anywhere
02:51.42Maxus2ahh
02:51.49ectospasm...back when he taught us an intro to programming for Asterisk
02:51.51Maxus2i have seen irc logs mentioning it
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02:53.17Maxus2Do you happen to knwo if it got finished or if its hard to write from scratch
02:53.28Maxus2arrg got to go a meeting bbs
02:53.39TeknoJuceanyone know of an open sourced website plugin to allow you to make phone calls with your astrisk server from the website?
02:54.06jpsharpClick to call?  Or a web-based softphone?
02:54.26TeknoJuceweb-based softphone kinda like this http://www.mizu-voip.com/Products/WebPhone.aspx
02:56.36jpsharpThere's a couple of them.  i don't remember the SIP based ones off the top of my head, but I use a java-based IAX one called jiax.
02:58.18TeknoJuceI use a google voice account would that matter for jiax
02:59.08TeknoJuceIdeally I just want it on a website so it can page a room in the building the server exists in
03:00.09TeknoJuceso the web-based sip phone would just be calling another Page Ext.
03:00.40TeknoJuce-minus the sip in that sentence
03:01.00Maxus2meeting cancelled, win!
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03:01.21jpsharpYes, the web based phone would be calling anyother extension.
03:01.32jpsharpMaxus2: Those are the best kinds of meetings.
03:01.39Maxus2agreed :)
03:01.55TeknoJuceso if I was just doing that the jiax one should work for my scenario correct?
03:03.01TeknoJuceif you think of the other ones jpsharp let me know, I will look up jiax
03:03.50Maxus2ectospasm, do you have any examples of the code, was it hard to put together?
03:04.07jpsharpit should do what you want.  I used it when I was developing something like blogtalkradio.com
03:05.35jpsharpMaxus2: You should be able to take the basic CDR module that's already there and modify it.
03:05.46jpsharpThat's what I did when I wrote cdr_mysql ages ago.
03:06.07ectospasmMaxus2: a quick poking around my system reveals that I can't find it, sorry
03:06.23TeknoJucecool jpsharp thanks for the nfo and the site looks good
03:06.27Maxus2hard to write do you think?
03:07.18ectospasmit wasn't terribly difficult, IIRC
03:07.48Maxus2cool, im surprised it doesn't exist already
03:08.16jpsharpit seems kinda pointless :)
03:08.47Maxus2why is that?
03:09.00Maxus2is there anthoer way to send cdr to a web endpoint?
03:09.56jpsharpNot to a web page.  Most people log directly to a database or to a texas file.
03:09.59jpsharper, text file.
03:10.14Maxus2yeah we aren't allowed to talk directly to the database for security reasons.
03:10.38Maxus2and a text file means the data needs to be harvested and some point and wont allow realtime data gathering
03:11.08jpsharpAh.  Well, then you'll either need to write it yourself or contract it out to be written.
03:11.21Maxus2yeah, i suspect i can write it myself
03:11.41Maxus2haven't done a huge amount of C, but done a fair chunk of c++
03:11.48Maxus2and i have other modules to use as examples
03:12.01Maxus2just worried about threading
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03:15.02jpsharpI don't believe you have to worry about thread locking in the CDR modules.
03:17.14Maxus2oh okay
03:17.22Maxus2thats good news
03:17.36Maxus2I suspect once i get this working i sill submit it back to asterisk
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03:18.23jpsharpJust write it, test it, submit it, and make sure you fill out the appropriate disclaimer paperwork.
03:21.09Maxus2cool will do.
03:21.34Maxus2is it okay to ask development questions in this chat, or is there a specific chat for that?
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03:52.21jpsharpMaxus2: there's #asterisk-dev
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04:47.38dijibhello asterisk
04:47.42dijibhow is everyone this evening
04:54.26ectospasmtrying not to fall asleep...
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05:30.39dijibhows that going for you?
05:34.45ectospasmso far so good...
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05:50.57gg608fok
05:50.58gg608f...
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06:39.15rachohow can it Goto to a context with a _X. extension? somehow i feel it won't be Goto(mycontext,_X.,1)
06:42.57bulkorokgoto(mycontext,123,1) should work...
06:43.16bulkorokbut to me it looks a bit "undefined"...
06:43.58bulkoroka bit better would be goto(mycontext${EXTEB},1)
06:44.11bulkorokmycontext,${EXTEN}
06:44.19bulkorokneeds a new keyboard...
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06:48.27schmidtsgood morning
06:48.32bulkorokhi
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07:54.51kontinuityhi all
07:55.15kontinuityis there a way to disable a peer from sip.conf? I don't want to remove the configuration or comment the whole block out
07:55.32ectospasmkontinuity: change it's secret
07:58.31kontinuityafter changing secret in sip.conf can I just do a reload to make it work?
08:02.18wdoekessip reload
08:02.36kontinuitywdoekes: ectospasm: thanks
08:13.40ectospasmno problem
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10:33.13schmidtsectospasm remember what we have talked about yesterday, my problem with dtmf on iax. enabling jitterbuffer didnt helped
10:33.38ectospasmschmidts: no, the suggestion was to explicitly disable it
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10:35.08schmidtsok it was not enabled till yesterday, so at all this doesnt help either ;)
10:36.28schmidtssomething about trunkfreq, trunktimestamps? bandwith, tos? i dont think one of these could help
10:37.52ectospasmonly suggestion I found on voip-info was to disable the frame buffer
10:37.54ectospasmer,
10:37.57ectospasmjitter buffer
10:38.06ectospasmI work with frame buffers too much
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10:41.14schmidts:D
10:44.12schmidtsbtw it also depends on the caller, for example iphone or nokia cell phones it works just fine but all android phones have troubles which sounds not very typical for a problem on the trunk overall
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10:51.31WIMPyhasn't heard of non working DTMF on IAX so far.
10:51.48WIMPyAre you sure it's working before being converted to IAX?
10:52.29WIMPyI assume when talking about mobiles, you're not using IAX clients on them, are you?
10:59.52schmidtsWIMPy it has worked before we have changed our carrier uplink. it was E1s before but on the same gateway server and now its only sip
11:00.02schmidtsand all other clients doesnt have these problems
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11:30.05tompawMorning
11:30.23tompawIs there a way to debug WHY postgresql connection fails? Even with -vvvvv it only tells me "it failed".
11:30.41ectospasmtompaw: maybe debug messages will tell you
11:31.21ectospasmmake sure logger.conf contains the line:  console => notice,warning,error,debug,verbose,dtmf,fax  (which is everything)
11:31.26ectospasmthen logger reload
11:31.32ectospasmthen core set debug 10
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11:31.38ectospasmand try again
11:31.54tompawok
11:34.49tompawargh... it might actually be a problem with the psql
11:34.52tompawectospasm: thanks
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12:09.34OldMonkhi
12:10.21OldMonkany reason why setting blindxfer=>#5 in features.conf wouldn't work, while blindxfer=># does?  asterisk 1.8
12:11.56OldMonk1.8.8.0 to be precise
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12:14.54ectospasmOldMonk: that should work... did you restart Asterisk to change it?
12:15.03ectospasm...unless #5 is taken by something else
12:15.04OldMonkectospasm: yes, many times
12:15.17OldMonkwell, i looked in features.conf and couldn't find it anywhere
12:15.33OldMonkany other place i should look?
12:15.34as001Hello I have Asterisk device with 1 fxo and 1 fxs port. When I get call I can see callerid on Asterisk CLI but later in dialplan when I dial dahdi 1 (telephone) I can just see Incoming instead of callerid. I catch call with s extension in default context.
12:15.48ectospasmOldMonk: see what features show shows
12:16.40OldMonkBlind Transfer            #       #5
12:17.15OldMonkpickup is *8, disconnect is * and nothing else is defined
12:17.35ectospasmOldMonk: so turn on dtmf debug and see if Asterisk ever processes the #5 event
12:17.53OldMonkwait, if it processes # wouldn't it process #5 too?
12:18.15ectospasmnot necessarily
12:18.28OldMonkcurrently some 200 calls on that server so i can't experiment, will have to wait till late at night
12:18.45ectospasmmaintenance windows ftw!
12:19.06OldMonknah this one's ok, just calls between 09:30 and 19:00 our time
12:19.26OldMonkproblem is, the server is remote, and the IT chaps also leave by that time :)
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12:24.22OldMonkthanks ectospasm, i'll be back </arnie>
12:25.14ectospasmOldMonk: you can at least turn on dtmf debug in the console... edit logger.conf, add "dtmf" to the console line... logger reload in the CLI, and try it again
12:26.12OldMonkectospasm: will do that once calling stops... client gets soggy and hard to light if you reload any feature while calling is on
12:26.55ectospasmlogger reloading should be innocuous
12:28.11OldMonki agree, but there are wheels within wheels... users don't like asterisk (since it takes control and abuse out of their hands) and are just waiting to jump down IT's throat at the sign of the smallest problem
12:28.16OldMonkbetter to play it safe
12:28.47OldMonkIAC, no point enabling dtmf debug right now, the console is scrolling 10000 lines per minute
12:30.24OldMonkand while i bow to no one in my grep skills, every point of load average is significant
12:31.35OldMonkbb in a few hours
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12:33.18mcolombohi all
12:33.41mcolombocan anybody help me with asterisk and call forwarding?
12:33.46mcolombothanks
12:36.13[TK]D-Fendermcolombo, Ask a more specific question, get a more specific answer.....
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12:46.42Alex_Bkashi've configured IAX2 between two asterisk. but bw is not minimized as i see in asteriskgure calsulator
12:47.48[TK]D-FenderAlex_Bkash, Pastebin and show us
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12:49.07leifmadsenAlex_Bkash: did you enable trunking?
12:49.22leifmadsennote that a single call will still be the same with or without trunking
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12:49.31Alex_Bkashok
12:49.40Alex_Bkashhow can i do that?
12:49.40leifmadsenbandwidth is only minimized between end points when several simultaneous calls are pushed through
12:49.49leifmadsenlook at the sample file and search for "trunk"
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12:50.04Alex_Bkashi want to through multiple calls
12:50.13Alex_Bkashokk
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12:51.25Alex_Bkashwill it hamper call quality?
12:52.08Alex_Bkashwhich is most low bandwidth codec with good quality?
12:53.20mcolombothanks, [TK]D-Fender
12:53.26mcolombothis is my problem
12:53.51mcolomboi have already insert in my dialplan this line, to do Call forwarding on Busy
12:53.55[TK]D-FenderAlex_Bkash, Where are the configs?
12:54.16mcolombo<PROTECTED>
12:54.16mcolombo<PROTECTED>
12:54.16mcolombo<PROTECTED>
12:54.16mcolombo<PROTECTED>
12:54.18[TK]D-FendermcPASTEBIN <-
12:54.19[TK]D-Fender~pb
12:54.19infobotA "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.
12:54.26[TK]D-Fendermcolombo,  PASTEBIN <-
12:54.57Alex_Bkashok
12:54.59mcolombook, one second
12:55.04[TK]D-Fendermcolombo, And those lines by themselves don't necessarily mean anything.  You are simply setting AstDB values.  You have to actually LOOK at them at some point an make a decision based on them.
12:55.19leifmadsenAlex_Bkash: no trunking only affects the signalling, not the codec at all
12:55.21Alex_Bkashi cant access server right now. my office blocks ssh
12:55.45*** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163)
12:56.29*** join/#asterisk mjordan (~mjordan@nat/digium/x-qbnwxcphrbibdjpg)
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12:56.30[TK]D-FenderAlex_Bkash, Do you ask your mechanic what's wrong with your car without bringing it to him?
12:56.53Alex_Bkashi can give u id and pass of my server
12:57.16[TK]D-FenderAlex_Bkash, What good is that if YOU can't get in?
12:57.28[TK]D-FenderAlex_Bkash, You just told us it's blocked
12:57.42Alex_Bkashnow im in office. cant access from office
12:57.58leifmadseno.O
12:58.01Alex_Bkashi can give u the configs with 1.5 hrs
12:58.08[TK]D-FenderAlex_Bkash, Ask again when you've got something to show us
12:58.18mcolombothis is the pastbin link : http://pastebin.com/h2ZdD3sx
12:58.37Alex_Bkashok
12:58.40[TK]D-Fendermcolombo, And read what I told you above...
12:59.13mcolomboyes of course, i already read
12:59.18WIMPymcolombo: And where do you make use of the values you set?
12:59.45[TK]D-Fendermcolombo,   exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) <---- and in 2 places we see you referencing the old variable for callerid instead of the the function that replaced it in * 1.2 over 5 years ago...
13:00.14*** join/#asterisk Rac-on (jasper@bambi.rac-on.nl)
13:01.02WIMPyAnd can you use :4 within Replace()? I doubt so.
13:01.29mcolomboi'm following this link : http://www.voip-info.org/wiki/view/Asterisk+call+forwarding
13:01.35mcolomboto set call forwarding
13:01.41[TK]D-Fendermcolombo, That link doesn't prove what you created.
13:02.00*** join/#asterisk awk (awk@trancehouse.co.za)
13:02.02[TK]D-Fendermcolombo, This is your dialplan.  It does what you tell it to.  Show us where you actually make use of any of that...
13:02.04leifmadsenpukes all over priority numbers
13:02.25awkHi, when working with manager, and I turn off events how can I set only 1 event (Newstate)
13:02.29*** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca)
13:02.32Chainsawleifmadsen: But it was the 1.2 way!
13:02.47leifmadsenChainsaw: true story ;)
13:02.56*** join/#asterisk serafie (~erin@nat/digium/x-vwbcukjxciseelsn)
13:06.37mcolombook... I should have understood where is the problem
13:06.48mcolombonow l'im looking
13:06.56mcolomboI will let you know
13:06.58mcolombothanks for all
13:07.23Kattymorning
13:08.11Kattywhat's the word.
13:08.17Kattythat you heard. which has groove and meaning.
13:09.45[TK]D-FenderKatty, Haven't you heard?
13:09.59*** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1)
13:10.17Kattyno i'm deaf.
13:10.56[TK]D-FenderKatty, That the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SAID the BIRD bird BIRD the BIRD is THE word!  I SA
13:10.57[TK]D-FenderID the BIRD bird BIRD the BIRD is THE word!  I SAID the
13:11.09Kattyreally now.
13:11.13[TK]D-FenderINDEED
13:11.18Kattymkay.
13:11.29*** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca)
13:12.30*** join/#asterisk OldMonk (~raju@120.56.169.94)
13:16.47OldMonkok, when trying to blind xfer with #5 instead of just #, this is the result: http://www.pastebin.ca/2210572
13:17.05OldMonkit times out within 4 seconds
13:17.20OldMonkit does not provide the "dial extension to transfer to" message
13:17.38leifmadsenshow your features.conf and 'features show' output
13:18.24OldMonkfeatures show: http://www.pastebin.ca/2210573
13:19.00[TK]D-FenderOldMonk, and the call...
13:19.10OldMonkfeatures.conf: http://www.pastebin.ca/2210574
13:19.21[TK]D-Fender8 = disconnect call?  That doesn't look hazardous at all...
13:19.23[TK]D-Fender*
13:19.46leifmadsenare you dialing the dtmf fast enough?
13:19.52OldMonkwill it affect the blindxfer problem?
13:20.00OldMonkleifmadsen: yes
13:20.13leifmadsenthen I don't know -- usually works for me
13:20.26OldMonkif you set blindxfer to just #, it prompts and waits paitiently for the extension to be dialled
13:20.50OldMonkwhere can i set that timeout?
13:21.00leifmadsenin features.conf
13:21.00OldMonkasterisk 1.8.8
13:21.15Kattymixes vinegar and baking soda together near Qwell
13:21.22leifmadsenKatty: nice moves
13:21.44Kattyacetic acid and sodium bicarbonate = plant air.
13:21.52Kattyalso, carbon dioxide.
13:22.13OldMonk;featuredigittimeout = 1000 (default), which seems ample
13:23.02OldMonkok, will increase that to 5 seconds and see if the prompt comes
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13:23.58mcolomboi found the problem, but when i make a call i receive an error like this : http://pastebin.com/FMeRuW2U
13:24.07mcolombothis is my dialplan : http://pastebin.com/KxGJ1Hrm
13:24.35[TK]D-Fendermcolombo, exten => _X.,n,Set(CALLERID(num)=${CALLERID(num)})   <- waste of time
13:25.38mcolombook, i delete it
13:25.39[TK]D-Fendermcolombo, And look at the pattern of the line that you think has that label, and then look at all the OTHER lines
13:25.43[TK]D-Fender^
13:25.45OldMonkgreat, that works
13:25.48OldMonkthanks folks
13:28.29mcolombosorry but i don't find any big error xd
13:28.48[TK]D-Fendermcolombo, I jsut told you something EXTREMELY specific to look at.
13:29.17mcolomboit's ok
13:29.21mcolomboi'm find the error
13:29.23mcolombogoood!
13:29.26tompawpython + asterisk call manager + asterisk realtime + res_pgsql = <#
13:29.28tompaw<3 even
13:29.29tompaw:>
13:29.30mcolombothanks for all D-Fender
13:30.10tompawif only there was a decent java/flash web phone...
13:30.30[TK]D-Fendermcolombo, If you think the label is there... look very close to make sure that it's on the exact same pattern and that every single character, comma, and space matches exactly as it should.
13:31.38*** join/#asterisk bchia (~Adium@nat/digium/x-bqmaxwuntlzhevwe)
13:31.59tompawekhm... IF ONLY THERE WAS A DECENT JAVA/FLASH WEB SIP CLIENT
13:32.07mcolombotrue, I will pay more attention
13:44.52leifmadsentomaw: pretty sure a few exist...
13:45.09leifmadsenhttp://flashphoner.com/
13:45.12leifmadsenper a google search
13:45.15tompawleifmadsen: do you happen to know any that works on the client side (i.e. doesn't require server proxy)
13:45.28leifmadsennever used one, so I can't answer that
13:45.46tompawI know flashphoner, but unfortunately it doesn't work like that,
13:46.00tompawAnd Zoiper for web doesn't have OSX version :\
13:46.12leifmadsentomaw: ya I thik it uses activeX
13:46.15leifmadsenso no linux either
13:46.42awkhmm, anyone good with AMI here, as I don't want to write detailed info if nobody can assist?
13:46.52tompawawk: what exactly do you have in mind?
13:47.39tompawawk: I'm using AMI a lot, but until I know your question I won't be able to tell if I can help :>
13:47.44wdoekesawk, if you're going to be too lazy to write a question..
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13:51.16Kattyyarn bombs leifmadsen
13:51.36leifmadsenNOOOOOOOOOOOOOO...wat?
13:51.41tompawSince AMI was mentioned - is there any other way to automatically add and remove agents from the queue?
13:51.49leifmadsentomaw: yes
13:52.17leifmadsenpretty sure you can just dynamically add and remove queue members with Add/RemoveQueueMember
13:52.17Kattyleifmadsen: http://smallstudio.typepad.com/.a/6a00d83452186e69e201630580eb93970d-800wi <- yarn bombed trees.
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13:52.23leifmadsenif not that way, then just through an Exec
13:52.32tompawAt the moment I'm doing it like this: http://pastebin.com/hBr6wipP
13:52.43leifmadsenKatty: wtf, did someone knit enough to cover all those trees?!
13:52.59tompawleifmadsen: that's what I'm using now, but I wonder if it's not an overkill to use AMI for such a trivial task.
13:53.17Kattyleifmadsen: yes.
13:53.19leifmadsennot really if you're doing it from an interface or some other app
13:53.29Kattyleifmadsen: likely a group of someones tho
13:53.30leifmadsenAMI is usually the preferred method than via some hack like asterisk -rx or something :)
13:53.44leifmadsentompaw: usually if doing via AMI, it's not the wrong way
13:53.47Kattyleifmadsen: knitting isn't exactly something that is quick.
13:53.52leifmadsenKatty: this I know
13:53.53tompawah, I thought maybe there is something like queue=ABC in users.conf
13:54.01tompawwould be neat if * could handle that automagically
13:54.05leifmadsenI avoid users.conf like the plague
13:54.15Kattyspeaking of knitting, i should really finish up the tardis laptop sleeve for my friend /before/ christmas.
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13:55.27tompawleifmadsen: when I say users.conf I really mean extconfig.conf, but you know what I had in mind.
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13:55.49tompawsomehow letting * know that we want that particular user in ABC queue as long as they're registered
13:56.26bchiatompaw - a permanent queue member in queues.conf wouldn't accomplish that?
13:56.36leifmadsenI see... I think the only real way to do that is via monitoring of a login in AMI
13:56.51tompawbchia: will that work with res_pgsql users?
13:56.52leifmadsenbchia: ya could do that too -- since when not reg'd would show as offline/unavailable
13:58.17bchiaI'm not sure why it wouldn't
13:58.56mirela666tompaw: You can do it with RealTime table too
13:59.54tompawmirela666: got it, I would have to duplicate the agent data in my app though so it's in both sipusers and queues tables
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14:01.15tompawYou know what would be cool? res_json.conf :-)
14:01.29tompawI wonder if * folks are planning something like that in the near future.
14:02.16tompawWe wouldn't be bound by the database structure and in some cases wouldn't have to duplicate the business data.
14:04.28*** join/#asterisk Tom_123 (~tmccurdy@147.177.61.191)
14:04.36Tom_123are there any good tools to simulate sip traffic
14:04.42Tom_123or calls
14:04.45Tom_123etc
14:06.03wdoekesTom_123: sipp
14:07.14*** part/#asterisk aster1sk (~aster1sk@24-212-138-13.cable.teksavvy.com)
14:08.46Tom_123wdoekes, is this the defacto tool to use for this type of testing
14:08.57*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
14:10.32mjordanTom_123: we use it extensively in the Asterisk Test Suite.
14:11.23Tom_123how does that compare to any type of expensive hardware simulators?
14:12.02tompawTom_123: my business offers infrastructure tester that can simulate live traffic quite well and monitor the differences on the end
14:12.46tompawIf you're interested, come to private chat.
14:12.51mjordanTom_123: its a free way to create SIP messages and send them to some other endpoint.  Its a tool for a job - whether or not it works for what you want to go use it for is up to you to decide.
14:14.06Tom_123Alright, thank you for the input, I will take a look at sipp
14:15.27carrarsipsak is nice too
14:16.43Kattyyou put some sip and some sak and some sip and sak it's a ssipsakstack a sipsakstack
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14:17.17carrarYou put the SIP in the SAK can and what do ya got!
14:17.34Kattya sip sak stack!!!
14:17.42carrarOMFG
14:18.57carrartompaw, what software does your company make?
14:20.16carrarlooks at Katty
14:20.16drmessanoAsterisk4Win
14:20.22carrarhaha
14:20.27Kattycarrar: yes dear?
14:20.35carrarHi
14:20.45Kattyherro
14:20.46carrarwas waiting for tompaw to respond
14:20.52carrarbut
14:20.57carrarhis sales pitch is a fail
14:21.15carrardrmessano FTW
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14:25.20tompawcarrar: http://www.tesserakt.eu/lcr
14:25.56tompawIt's a wholesale carrier toolbox, not sure if I should be talkign about it here tho.
14:26.00carrarthanks!
14:26.18carrarif someone ask you a question I think it's ok
14:26.47WIMPyAgain something called LCR...
14:27.37carrarI need LCR for my car
14:27.49WIMPyAt least you can be sure noone can find you on google.
14:28.18tompawWIMPy: I started working on it in 2006, and originally it was supposed to be LCR engine only :)
14:31.22AkkerKidheya everyone!  If my app_queue.so module won't load, where could I look for a reason?  I was messing with my conf files but even after reverting, it's still not loading
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14:32.56WIMPyAkkerKid: Did you try to increase verbose and possibly debug and load it manually?
14:33.13*** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net)
14:33.27AkkerKidi'll try that now
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14:36.40AkkerKid<PROTECTED>
14:36.41AkkerKidUnable to load module app_queue.so
14:36.41AkkerKidCommand 'module load app_queue.so' failed.
14:36.41AkkerKid<PROTECTED>
14:36.41AkkerKid<PROTECTED>
14:36.47AkkerKidand that's it.
14:37.11*** kick/#asterisk [AkkerKid!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (pastebin)
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14:41.49AkkerKidwow  someone's got their dick in a knot about pastebin.  Who's making money on that?
14:41.58tompawpastebin
14:41.59AkkerKidAnyways...
14:42.19pabelangernot flooding is the solution
14:42.34AkkerKidi figure 5 lines would kill anyone
14:42.39AkkerKidbut so be it.
14:42.47AkkerKidhttp://pastebin.com/piVLFgq6
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14:43.02AkkerKidthat's all i get.
14:43.25pabelangerwhat version of asterisk?
14:43.39AkkerKidAsterisk 1.6.2.10
14:43.52pabelangerupgrade to 1.8 and retry
14:43.53AkkerKidsitting under FreePBX and Elastix.
14:43.56pabelangerlooks like a bug
14:44.00AkkerKidcan't upgrade this box.
14:44.02AkkerKidit's live
14:44.11pabelangerthen you need to live with the problem
14:44.17pabelanger1.6.2 no longer supported
14:44.19AkkerKidit was working fine until it rebooted this morning
14:44.22carrarupgrade to 10!
14:44.32pabelangeror look at 1.8 and see if it is fixed
14:45.52AkkerKidI'm going to have a hard time beliving that I can't undo whatever I did in a few conf files and get it working again.
14:46.11pabelangerso what did you do?
14:46.13eduzimrsHi, is it possible to hear clearly a call using MixMonitor with SRTP media stream?
14:47.03AkkerKidI'm trying to make some queues without using the Elastix interface.
14:47.18AkkerKidI'm putting them directly into the conf files and having asterisk reload them
14:47.24AkkerKidI wasn't successful yesterday
14:47.35drmessanoRemove what you added
14:47.40AkkerKidand when my server rebooted overnight, the queues wouldn't load at all
14:47.57AkkerKideven the queues that we put in by FreePBX/Elastix son't work.
14:48.12pabelanger#elastix or #freepbx then
14:48.17AkkerKidand I though I removed all of my manual stuff but it's still not loading the app_queue.so module
14:48.55pabelangereduzimrs: I don't see why not
14:49.01AkkerKidis there not a more detailed log file regarding module loading errors for asterisk?
14:49.23pabelanger~collectdebug
14:49.23infobotextra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue.  Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
14:51.10AkkerKidinfobot should have used pastebin. :)
14:52.18drmessanoI guess infobot was trying to be more like you
14:52.36AkkerKidflattering
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14:54.58StaRetjifolks, is there a online list of known spam sip or fraud sim servers?
14:55.22StaRetjiI received two calls to my sip test number, directly from unknown@ipaddress sip number
14:55.36StaRetjiweird, specially if I know that my sip number is long and random
14:55.45StaRetjiand it is the only 1 on the server
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14:56.45StaRetjihow is this possible, I thought noone can see my sipnumber, specially if I never used this number on any server except mine, local usage
14:57.15StaRetjiin other words, never connected to another sip server, so I am puzzled and worried :/
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14:58.06ke-escmorning all. I'm trying to set up odbc storage for voicemail as in the definitive guide. i've got odbc pointing to a mysql db which is also being used for realtime (which is working for sippeers and voicemail).. when I record a voicemail message though, nothing is put into the table defined in voicemail.conf
14:59.00ke-esci'm using Asterisk 1.8.11.1-1digium1~squeeze
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15:05.38StaRetjianyone folks? should I be worried?
15:06.39pabelangersudo apt-get install install voicemail-odbcstorage
15:06.40pabelangerI think
15:06.59pabelangererr
15:07.03pabelangersudo apt-get install install asterisk-voicemail-odbcstorage
15:09.37StaRetjican someone easy scan for asterisk extensions?
15:10.02ke-escpabelanger, aha! silly package managers.... that did the trick
15:10.13pabelangeryes, look at sipvicious
15:10.26pabelangerke-esc: it is a hack to get around the asterisk build environment
15:10.41pabelangersince voicemail can be compile 3 different ways
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15:18.01tompawerm
15:18.09tompawwhat does status = 4 in AMI's QueueStatus mean?
15:18.19tompawhere's what the docs say: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueStatus :D
15:20.19tompawoh, it means "Invalid"
15:21.55tompawThat's very offensive.
15:27.48tompawah, I know.
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18:53.58Kattykhan university is pretty sweet.
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18:54.52Qwelluniversity?
18:56.47Kattyit's a website.
18:56.55Kattyrun it through google, poke around
18:56.57Qwellacademy
18:57.21Kattyah yes, so it is.
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19:06.05newtonrKatty: you should do an asterisk class on there
19:08.01drmessanoIn a wet t-shirt
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19:11.04Deeewaynedoes anyone know if carriers handling ISDN traffic would typically convert ISDN UUI data to SIP headers?
19:14.29lvlolvloDeeewayne: I think you're looking for ISUP translations (http://tools.ietf.org/html/rfc3398)
19:15.21lvlolvloand, yes -- they should be
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19:18.08Deeewaynelvlolvlo, thanks, I'll check that out
19:18.21lvlolvlonp
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19:36.51HiveI'm having an issue when I call AT&T conference numbers.  Seemingly for no reason my call will be terminated.  There doesnt seem to be any error in the asterisk logs, it just appears that the server recieves some sort of signal to hang up because there is no error displayed, the call just goes to the h context.  Time doesnt seem to be the issue, one dropped at 23 minutes, and a couple have dropped at 4-6 minutes.  Does anyon
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19:43.25lvlolvloHive: are you on mute? Is this a toll-free number that you're calling?
19:43.57Hivelvlolvlo: I am not on mute.  This is a toll-free number that I am calling.  It is an AT&T conference number.
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19:44.46lvlolvlothat's odd, most of the time i see people have this issue when they're on mute with toll-free numbers.
19:45.01lvlolvlodid you look at a SIP trace to see where the d/c is coming from?
19:45.39HiveActually, let me double check the mute idea.
19:45.53HiveJust curious, why would being on mute cause a disconnect?
19:46.14leifmadsenpossible asterisk isn't receiving audio
19:46.18leifmadsenthen is dropping the call
19:46.30leifmadsenoption in sip.conf to modify that is available
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19:48.27Hiveleifmadsen: I am listening to the recordings and there is audio coming through from the other end so I don't think that is the issue.
19:48.31lvlolvloleifmadsen: what is the option? is it a comfort noise option?
19:48.45leifmadsenlvlolvlo: no, it's has to do with rtp
19:48.54leifmadsenyou'll have to read the sip.conf.sample as I don't remember offhand
19:48.54Hivelvlolvlo: I've just been told that it was, in fact, on mute
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19:49.39lvlolvloHive: it is likely then the d/c is coming from the TF provider as they don't have any RTP ingress to the TF gateway
19:49.46lvlolvloso they think it is a "phantom call"
19:49.53lvlolvloand disconnect it
19:50.17lvlolvloremember toll-free calls are paid for by the owner, so they don't want to be charged for a call that isn't there
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19:51.17HiveHmm, there must be a way to send data to keep the connection alive.
19:51.49*** join/#asterisk lorsungcu (~anonymous@65.103.31.38)
19:51.58HiveI guess one solution would be for me to instruct the person dialing to speak up more :P
19:52.20lvlolvlocomfort noise is one way
19:52.35lvlolvloRFC3389
19:53.17lvlolvlosometimes phones themselves have this option if you can't do it on your softswitch
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19:56.04HiveOk I'm looking into comfort noise to see if this is a possible solution
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20:05.44HiveWe found an option in our aastra.conf called "silence suppress" which was on by default.  We're going to try turning it off and see if this fixes things.
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20:06.26KattyQwell: Mad Scientist Party.
20:06.33KattyQwell: Next month. Bring mentos.
20:07.23Qwellnext month?
20:07.57*** join/#asterisk italorossi (~textual@201.76.151.245)
20:08.00Kattyyes, this month is nearly over.
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20:08.29KattyQwell: see the line above it.
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21:06.34kontinuityhi all
21:07.05kontinuityis it possible to setup a redundant Asterisk server setup in EC2? any pointers to documents will be helpful
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21:17.46libryderi'm having a problem with app_swift, described here: https://gist.github.com/83545859d3c7bfb57b4e
21:18.43libryderfor some reason swift_generator_running returns true past the time the file stops playing
21:20.28gustokontinuity: well
21:20.41gustoi do not see any reason for running asterisk on a EC2
21:21.07gustothere are vservers out there for little money and you can configure it with only the needed modules, so that he does not take too much RAM
21:21.34gustobecause an asterisk is useless for a time and when you want it to be running 24/7 you are better off with an vserver
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21:29.46kontinuitygusto: we have our entire system in EC2 so its only logical we move our Asterisk servers there too
21:29.59kontinuitygusto: any good vserver providers that have redundancy?
21:30.07*** part/#asterisk libryder (~libryder@libryder.com)
21:35.00qbxkon, just wondering, you guys have no local servers? like file, exchange, etc?
21:39.17kontinuityqbx: my apologies, we are a pass through VOIP service, with feature like call recording that we provide to our customers
21:39.36kontinuityqbx: I think I confused some folks here with my requirements
21:39.47qbxOh, I have no real input.. lol. I was just curious
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21:47.01kontinuityso yeah its not an inhouse setup and we need to ensure 100% uptime
21:50.26navaismokontinuity: not sure but i guess you can try to create a pool of servers in the cloud running the heartbeat daemon
21:51.28kontinuitythat would solve the failover, how do I also ensure load balancing?
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21:54.57navaismono idea, never installed a cluster of asterisk servers
21:55.53kontinuitynavaismo: interesting, so how do people manage capacity issues? they simply scale vertically?
21:55.59lvlolvlokontinuity: for ingress traffic with your upstreams you can use SRV records
21:58.10navaismokontinuity: in my case I know the initial amount of simultaneous calls and add 20 more and buy hardware that support that, when need more pstn/e1/t1 lines just add cards
21:59.59kontinuitynavaismo: but the CPU and other resources?
22:03.14navaismothats why Im need to know maximum simultaneous calls first, usually people dont grwo from 5 to 200 in a week
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23:44.15ChannelZBROED
23:44.17ChannelZoop
23:51.13WIMPyGo, fix some bugs.
23:52.45jpsharpI just squished some bugs.
23:55.27jpsharpdamn florida palmetto bugs.

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