IRC log for #asterisk on 20100830

00:02.26*** join/#asterisk beccara (~chatzilla@2403:d200:2:0:69fe:fefc:ea44:7e06)
00:02.42drmessanoOk, so what is the proper target for SVN to get dahdi updates in the same vein as say svn'ing the 1.6.2.x branch asterisk
00:05.15drmessanoLooks like for DAHDI it's trunk or specific releases
00:05.25drmessanono "branches"
00:06.13drmessanoOk, for linux-complete their isn't
00:06.41drmessano"there"
00:07.29beccaraAnyone here who knows there way around dialplans?
00:07.41drmessanoI'm sure a lot of people do
00:07.44drmessano~ask
00:07.44infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
00:09.47beccaraok, http://pastebin.com/LVzzDkRh I have the extended CDR info set but it doesnt work when the call jumps to BUSY,NoAnswer,Cancel etc
00:10.27beccaraI assume that its because S-BUSY etc dont jump to the h parts
00:11.52[TK]D-Fenderbeccara: Where is the call to look at?
00:12.06beccarahuh?
00:12.28[TK]D-Fenderbeccara: So far that is indeed "assuming" and not "looking".  Also you should consider at what point CDR access gets cut off as well as to what parts of it are READ-ONLY and which aren't
00:12.29beccarayou want a console dump?
00:12.47[TK]D-Fenderbeccara: So time to start looking in detail
00:13.22beccaraokie dokie
00:14.05*** join/#asterisk NEEDINGHELP123 (Mordi@v58.sgsvr.com)
00:14.38NEEDINGHELP123hi guys, looking at developing a logistical routing platform for asterisk (or , so) ... wondering if there are any existing solutions i can take ideas / samples from
00:14.59beccara[TK]D-Fender: http://pastebin.com/k3zuu2yE
00:15.16beccarathats a working call that does a full CDR
00:20.26[TK]D-FenderNEEDINGHELP123: Sorry, could you be a little more vague please?
00:21.30[TK]D-Fenderbeccara: And what is in the CDR record?
00:21.47beccarathe full cdr
00:21.58[TK]D-Fender....
00:22.12beccarawhen a call fails its not includeing the extra info i specifiy
00:22.25beccaralike the hangupcause
00:22.52[TK]D-FenderWhere do I SEE THIS?
00:23.07beccarain the pastebins?
00:24.35[TK]D-Fenderbeccara: I am not seeing you showing things in a coherent manner.  Complete call debug with matching CDR OUTCOME.  FAILED call with ITS CDR.
00:25.06[TK]D-Fenderbeccara: I can't tell where the bits and pieces match up and I don't see any CDR currently....
00:25.13NEEDINGHELP123[TK]D-Fender yes, i am interested in receiving calls into asterisk, and instead of sending them directly to one sip peer, instead load balancing them, or such 'round robin' to created
00:25.14beccaradont worry
00:25.33NEEDINGHELP123i am willing to pay a decent  amount for such a script also
00:25.35[TK]D-FenderNEEDINGHELP123: Huh?
00:26.49NEEDINGHELP123[TK]D-Fender ... i have incoming calls via SIP or OOH323 channel in asterisk, i need to distribute them equally to sip users, ie: my agents connected
00:27.09[TK]D-FenderNEEDINGHELP123: sure...
00:27.17[TK]D-FenderNEEDINGHELP123: "core show application queue" <-
00:27.20beccaraCan anyone tell me how to make the s-BUSY section of this dialplan http://pastebin.com/LVzzDkRh jump to the h section of the context after it the s-BUSY section is complete?
00:27.43[TK]D-FenderBecc where do we see it FAIL to do so?
00:28.20beccaraits not jumping, there is nothing to see
00:28.28[TK]D-Fenderbeccara: where do we see it FAIL to do so? <---
00:29.19beccarahere is the fail "" it displays nothing
00:29.27NEEDINGHELP123[TK]D-Fender i cannot make alot of sense of it, i have also read the WIKI
00:30.24*** join/#asterisk jakent (~john@soleil.johnkent.mooo.com)
00:30.33[TK]D-FenderNEEDINGHELP123: define your queue in queues.conf.  make sure it has the members you want, strategy, etc.  Send your caller into the queue
00:30.47[TK]D-Fenderbeccara: Show us the failed call.
00:32.18beccarahttp://pastebin.com/aCyb3t6k
00:32.52beccaraignore the nosuch context error
00:34.50[TK]D-Fenderbeccara: Why would I?  That immdediately trashes your call.
00:34.56NEEDINGHELP123[TK]D-Fender understood, and if i use the random strartgery
00:35.12NEEDINGHELP123how do i set a call limit
00:35.15NEEDINGHELP123just with call-limit= ?
00:35.19[TK]D-FenderNEEDINGHELP123: That doesn't sound very "balanced" to me....
00:35.33NEEDINGHELP123if one call limit has been reached, will it continue to try on the next?
00:35.55[TK]D-FenderNEEDINGHELP123: and yes if it doesn't reach the first member, it will move on to others
00:36.09NEEDINGHELP123ok, doesnt reach = for any reason? including clal limit
00:36.11NEEDINGHELP123call* sorry
00:36.31[TK]D-FenderNEEDINGHELP123: For whatever reason
00:36.39NEEDINGHELP123[TK]D-Fender are you ofay with the queue scripts in asterisk? if so, would you consider writing me a small change for a fair amount of $?
00:36.55[TK]D-FenderNEEDINGHELP123: What change?
00:37.15[TK]D-FenderNEEDINGHELP123: and what is a "Queue script"?
00:37.32[TK]D-FenderNEEDINGHELP123: Queue() ,_ 1 LINE in your dialplan.  The end.
00:37.39NEEDINGHELP123yes,
00:37.45NEEDINGHELP123well i want a custom script written up, an dif your ofay with it
00:37.49NEEDINGHELP123i would appreciate the help and be willing to pay
00:37.51NEEDINGHELP123that is fair no?
00:37.56[TK]D-FenderNEEDINGHELP123: What is a "custom script"?
00:38.06NEEDINGHELP123well, a script that does a certain set of things that i want it to do
00:38.12NEEDINGHELP123that you can also advise me the best way of doing
00:38.26NEEDINGHELP123[TK]D-Fender can we move to PM?
00:38.27[TK]D-FenderNEEDINGHELP123: Might be nice to know that you aren't asking for something * already does and that it is even possible.
00:38.39NEEDINGHELP123no, it is possible
00:39.05beccara[TK]D-Fender: that is called after it exit's the contect
00:39.11beccaracontext
00:39.13[TK]D-FenderNEEDINGHELP123: You don't seem to know the functioning of what you already have.  You may want torefrain from jumping to the conclusion that what you need qualifies as "special" and requires any reall amount of work.
00:39.45[TK]D-Fenderbeccara: If you leave the Ccontext, "H is not longer THERE!
00:39.54beccaraexactly
00:40.17[TK]D-Fenderbeccara: then I guess you better put that code wherever it may be called.. which in this case it OUTSIDE of your macro
00:40.37beccarawhen the call flows normally it will hit H but if the call hits S-BUSY/NOANSWER/etc it wont be called
00:41.06[TK]D-Fenderbeccara: Then maybe you should just JUMP to that code directly.
00:41.14beccarawhich is what i asked
00:41.23[TK]D-Fenderbeccara: So Goto(h,1)
00:41.26beccarahttp://pastebin.com/LVzzDkRh How do I make S-BUSY hit the N section
00:41.38beccarathank  you
00:41.42[TK]D-Fenderbeccara: "N section"?
00:41.55beccaraexten => h,1,set(CDR(hangupcause)=${HANGUPCAUSE})
00:41.59beccarasorry H sec tion
00:43.37[TK]D-Fenderbeccara: then the Goto should work
00:44.09*** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net)
00:47.52beccaraSweet, That worked
00:50.57ChannelZYay!  It's time for cake!
00:51.49ChainsawChannelZ: Huge success!
00:52.07ChannelZI can't wait for Feb
00:52.47ChainsawYeah, I'm glad it's coming out for PS3 immediately. I hate waiting.
00:53.20ChannelZI'm really lousy with game controllers.  I tried to play it on the PS but am suck at it
00:53.45ChainsawI have the orange box. I have to be honest, I got it just for Portal. I haven't even touched HL2 yet.
00:54.31ChannelZA friend bought the orange box and brought it over last weekend, but I prefer it on the PC.
00:54.47ChannelZI guess I just don't play enough games on the PS3 because I find it difficult
00:55.28ChannelZespecially shooters, I just can't get accurate with the control sticks.
00:56.08ChainsawPC gaming is dead to me. Requires expensive unstable operating systems and rootkits.
00:56.26ChainsawOh, sorry. DRM / anti-cheat software. You can't call them rootkits anymore.
00:57.40ChannelZI'm all over Little Big Planet though, love that
00:59.43ChainsawI still need to get that one day :)
01:01.23ChannelZit's pretty fun, very creative
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02:18.47jimi_What does this error mean?  pbx.c:3114 pbx_extension_helper: No application 'SetVar' for extension
02:20.13[TK]D-Fenderjimi_: Exactly what it says.  SetVar is not a valid application. And hasn't been since * 1.2
02:20.49jimi_[TK]D-Fender, Where is SetVar set? In extensions.conf?
02:22.44[TK]D-Fenderjimi_: Yes
02:28.03jimi_[TK]D-Fender, What should SetVar be replaced w/ ?
02:28.21[TK]D-Fenderjimi_: Set
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02:31.09russellbor MSet(), depending on your needs (most likely Set(), though)
02:32.04jimi_What about DigitTimeout?
02:32.13jimi_and SetCallerID ?
02:32.20jimi_i just migrated my 1.2 system to 1.6
02:32.29russellbdid you read the UPGRADE*.txt files?
02:32.37russellbif not, please do so.
02:32.56russellbas it covers these exact issues -- things you need to be aware of when upgrading between major versions
02:34.59[TK]D-Fenderjimi_: You are dealing with some tragic 1.0 grade dialplan there.
02:35.09[TK]D-Fenderjimi_: Go read all of the upgrade.txt's since
02:35.15jimi_ty
02:35.17jimi_love
02:35.26russellbheh
02:35.47[TK]D-Fenderlets the hate flow through him ... yes, yesssssss.......
02:36.50[TK]D-FenderTHESE AREN'T THE DIALPLAN APPS YOU ARE LOOKING FOR
02:38.31russellba dialplan checker that checks for stuff like that would probably be nice ...
02:39.00[TK]D-Fender[22:18]<jimi_>What does this error mean? pbx.c:3114 pbx_extension_helper: No application 'SetVar' for extension <- checked if it was valid... says "no" :)
02:39.24[TK]D-Fenderrussellb: Anyone with syntaxt that old kinda gets what they deserve :)
02:39.28russellbyeah, but i mean without having to run it
02:39.37russellb./check-dialplan extensions.conf
02:39.42russellband it spits out all references to old stuff
02:39.51russellbi think that idea is about 5 years too late, though.
02:40.23[TK]D-Fenderrussellb: No, now we've created so much more for it to look for :)
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03:54.07Alton35I heard advice not to telnet into AMI (port 5038) too many times.  Has that problem been fixed?  Could I telnet in once for each call, like 50-100 times?
03:54.08Nuggettelnet is eeeeeeevil!
03:54.36Alton35Or better to run one of those proxies for that or use ActionId and different logic?
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03:56.32joobieAlton35, afaik it's a performance issue by going into the AMI so many times so frequently
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03:56.58Alton35hmm, the response time is not a problem, as long as it doesn't kill the system
03:57.39joobieit's a load issue
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03:57.52joobieso killing the system depends on what you're doing
03:57.57joobieuse a proxy
03:57.59joobieplay it safe
03:58.12Alton35does asterisk spend a lot of time polling the telnet connection then?  must be something like that.
03:58.16joobiepush your multiple connections to the proxy and you can keep the load off the ami
03:58.42Alton35ok, I'd probably code to where I didn't need the proxy then, I don't like too many dependencies,
03:58.43[TK]D-FenderAlton35: If you're using a 1.6+ branch you should be fine.
03:58.45Alton35but it's certanily an option.
03:58.56Alton35why so?  different code for this telnet host?
03:59.06[TK]D-FenderAlton35: Older version did have more serious issues with high connection loads, and AstManProxy is still a viable choice
03:59.13Alton35ok,
03:59.20joobieahh
03:59.34Alton35well, you'd be proud of me, I don't have it all implemented and everything, but it works fine, my program does my little dialing and stuff just fine.
03:59.46Alton35just need to parse all the possible returns thoroughly.
04:00.12Alton35great to know; I'll keep an eye on the system load and proceed apace.
04:02.35*** join/#asterisk martyn-dev (ba1cb672@gateway/web/freenode/ip.186.28.182.114)
04:02.45martyn-devHi asterisk people :)
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04:08.25martyn-devjoin http://tinyurl.com/speedwayrock
04:09.46Kyoshwhy?
04:10.47martyn-devis jst OT.
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04:57.14ruben23hi guys
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05:42.14ChannelZaloha
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05:58.40rue_housepeopel are here, say something more interesting or contravercial
06:03.09ChannelZI am sucking on my own tit-taaays!
06:06.38ChannelZSo shocking it's stopped some people's network connections
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07:34.25tehrabbittp3nguin around?
07:35.29ChannelZDo you see him?
07:36.24kaldemarhe might be disguised as a s3agull
07:38.16tehrabbittlol i suppose not 0_o....  gah i keep missing him lol
07:40.58tehrabbittwell heres a question for anyone who might be able to help...  I recently moved my * server from one location to a  new location... same internal IP range, different router (WRT54G which *should* work with SIP lol)... anyway, I noticed i'm not reciving inbound calls anymore though I can place calls outbound
07:41.15tehrabbittis that just a setting I need to change?
07:42.04ChannelZdo you register with your ITSP?
07:43.00tehrabbittNope I don't think so... hold on lemme look at the IAX2.conf
07:43.00ChannelZmake sure you have 5060 incoming forwarded to your * box and also the range of ports in rtp.conf
07:43.15tehrabbittAH fowarded ports... would that do it?
07:43.17ChannelZoh you're using IAX... well then port 4569
07:43.44tehrabbittChannelZ: gonna try fowarding that and see if it works
07:44.12kaldemaris it really IAX2 you're using? you mentioned SIP first.
07:45.04ChannelZhe might actually be using tin cans and string
07:46.11tehrabbittChannelZ: Actually it's two imaginary tin cans that don't need string :-p jkjk
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07:46.35tehrabbittkaldemar: I'm gonna be switching some stuff over to SIP i think... channelz might remember the huge router headache I had for idk, 3 weeks
07:46.40schmidtsmorning all
07:46.52tehrabbittfinding out that the router I had just did NOT support SIP at *all*
07:46.58ChannelZSIP is much more of a headache than IAX
07:47.23tehrabbittChannelZ: Agreed, but i'm thinking of using SIP as inbound for a PAP2 device
07:47.37schmidtsrouter which "support" sip will allways make more headache than if its not supported
07:47.39tehrabbittwell PAP2 located at a friend's house
07:47.42ChannelZSIP = 2+ ports needing forwarding, and ports needing outgoing access not under your control.  IAX = 1 port each direction
07:47.52tehrabbittschmidts: Linksys 160n
07:48.03tehrabbittsupports "VoIP" and you can't disable it
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07:48.21schmidtsthats the worst thing of all, if you cant turn this feature off
07:48.43schmidtsyou dont neet forwarding if you just do nat and no firewall
07:48.57schmidtsatleast the pap2 can send keep alive packages to keep the port open
07:49.39tehrabbittschmidts: yea, it was the crappiest router EVER...  even flashed DDWRT which caused more headaches (wasn't supposed to support DDWRT)...  the problem was the phone would ring, i could hear them, but they couldn' thear me
07:49.50tehrabbittalso DMTF wouldn't work over IAX, SIP, or anything with that router
07:49.54ChannelZYou need it for *incoming* unless you can support STUN and I don't think Asterisk does that well (it was removed in later versions if that tells you anything)
07:50.26schmidtsi have around 5k peers out behind several routers, most of them are cisco and zyxel and i dont have a stun server either ;)
07:51.06tehrabbittlol  yea, idk ChannelZ's advice was to "move your server" so I did haha like 2 months later 0_o  but i'm going to finally get it working, i think
07:51.07tehrabbittlol
07:51.09schmidtsthere is no need for port forwards, just simple nat, thats why is say all sip ALG types are worst
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07:52.01tehrabbittschmidts: well basically it quote on quote "handled all VoIP traffic using QoS and packet shaping"
07:52.11tehrabbittwhich screwed everything up lol
07:52.33schmidtsyou will need port forwards when you device doesnt register to your server, but if it does the nat port is allready there and just needs to keep open, the same thing on rtp traffic
07:53.31ChannelZExcuse me, my advice was what?
07:54.12tehrabbittChannelZ: my server was in a location where I couldn't physically connect it directly to the modem, so you said "Either move the server or get a new router, or don't use SIP"
07:54.24ChannelZWere you the one trying to like hop through 2 routers in your grandmas house?
07:54.33tehrabbitthaha aunt's house, but yes
07:54.47ChannelZok I remember now
07:54.59tehrabbittnow it's connected directly to a port of a WRT54G running stock firmware, no funny filters, on a FiOS line
07:55.38schmidtssry channelz i have missunderstood you ;)
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07:56.22ChannelZWell assuming FiOS (verizon?) isn't blocking random things, IAX is still easier to setup than SIP
07:56.30ChannelZif your ITSP supports IAX
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07:58.51tehrabbittFiOS line is completely unblocked apparently with no packetshaping too
07:59.49Kyoshfios rocks
08:00.51tehrabbittKyosh: what plan do you have?  50/20 here
08:01.27Kyoshno dude i have the basic
08:01.43Kyoshas basic as possible
08:01.44Kyosh20/5
08:01.59Kyoshand i get...20/5
08:02.01Kyoshalways
08:02.55tehrabbittah
08:02.57tehrabbittlol
08:03.20tehrabbitti've seen 50/10 here once :-\  then again it was an issue with the old router that was here before I upgraded to the WRT54G
08:03.35tehrabbittstupid belken garbage lol
08:03.36Kyoshu using their actioncrap router?
08:03.50tehrabbittNope, linksys off of the cat5 connection
08:04.10tehrabbittthe actiontek routers are actually just acting as a MoCA bridge in my house
08:04.30Kyoshyou have both moca and ethernet turned on?
08:04.33tehrabbittgiving the 2nd floor full 100Mbit stable connections
08:04.53tehrabbittKind of... MoCA is being handeled by the Actiontek not the ONT
08:05.02Kyoshmy ONT can have either ether or moca, but not both active
08:05.05tehrabbittethernet is turned on at the ONT and going to the linksys
08:05.19tehrabbittthe linksys then gives IPs out to all the cable boxes in the house via MoCA
08:05.23Kyoshmoca is not handled by the ont?  then what?
08:05.30tehrabbittwhich is nice because I can ping each box internally :-D
08:05.45Kyoshfrom the fiber to the ont, from the ont to either actioncrap or ...
08:05.49tehrabbittand can add additional actiontek boxes for more ethernet in any room using MoCA
08:06.03Kyoshooooo
08:06.10tehrabbittFiber to the ONT... ONT to Linksys... Linksys to Actiontek and to a Cisco Switch
08:06.13Kyoshyou got the actioncrap connection via ethernet
08:06.13tehrabbittActiontek to MoCA
08:06.17Kyoshnice
08:06.25tehrabbittActioncrap is hosting a MoCA bridge
08:06.28Kyoshgot ya
08:06.31Kyoshsmart move
08:06.39tehrabbittyupp no real "routing" being done for the exception of MoCA
08:06.43Kyoshgotta get the actioncrap off the ont, first thing
08:07.05tehrabbittwell the ONT is also connected to the same "MoCA" network for On Demand... it's kinda hard to explain lol
08:07.11tehrabbittit's a "Device" of the linksys router
08:07.19Kyoshno dude i understand
08:07.23Kyoshno worries
08:07.34Kyoshi got my directv ondemand going over my fios
08:07.40Kyoshso strange
08:08.15Kyoshbrb
08:08.15tehrabbitthaha nice
08:08.19tehrabbittkk
08:09.32gr0mitcan anyone recommend a good provider of geographic DID in spain please?
08:17.50tehrabbittChannelZ: I win the idiot award again for the night :-p  I forgot to change the IP for the DNS on the server from the old location
08:18.02schmidts:D
08:19.43tehrabbittsad part is I remember setting it up to use a hostname.domain.com ratehr than just an IP so if I ever moved the server I could just change DNS... yet I completely forgot haha
08:19.51tehrabbittthough opening those ports probabbly helped things
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08:34.13tehrabbittanyway i'm goin to bed, night everyone
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08:36.32Dovidnight
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08:40.08ruben23hi guys is asterisk http currently not available..?
08:40.47kaldemarby asterisk http you mean...?
08:41.32Kyoshruben, the website?
08:44.08ruben23Kyosh: when i do wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz  ---> its not downloading..
08:44.38xheliox2010-08-30 04:44:33 (1.55 MB/s) - “dahdi-linux-complete-2.3.0.1+2.3.0.tar.gz” saved [2001427/2001427]
08:44.41xhelioxno problem here.
08:46.31ruben23<PROTECTED>
08:47.17Kyoshworks fine for me
08:47.23ruben23http://pastebin.com/NieUHcdz
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08:50.03ruben23are  there other option i can get it..?
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09:03.34tzafrirhmm.... ipv6-related problems with downloads.asterisk.org ?
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09:28.01EmleyMoorIs there a good, reasonably priced supplier of AEX410P cards in the UK?
09:29.45EmleyMoorParticularly after a 400E, though a 401E would do at a push
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09:43.32tzafrirruben23, still having problems?
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09:50.34ruben23tzafrir: it just pause for around 2 minutes before it goes to download it, but it did download.
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10:33.25knarflyI could use some advice on how to read the asterisk log files
10:39.56schmidts<PROTECTED>
10:40.23schmidtswhat in special do you want to know?
10:42.53knarflyI need to know what this means...it's from my /var/log/asterisk/cdr-csv/Master.csv file
10:42.53knarfly"","xxxxxxxxxx","011972599544327","default","xxxxxxxxxx","SIP/100.100.100.100-00000004","SIP/my_voip_provider-00000005","Hangup","","2010-08-30 04:37:55",,"2010-08-30 04:37:58",3,0,"NO ANSWER","DOCUMENTATION","1283143075.4",""
10:43.29knarflythe IP address 100.100.100.100 is actually the IP address from my ISP...so where did this call come from?
10:46.44schmidtshave a look at your cdr_custom.conf file where your asterisk config files are.
10:47.52schmidtsper default its the clid, src, dst, context, channel, destinationchannel,lastapp, lastdata, start, answer, end ....
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10:49.35schmidtswhich means "xxxxxxxxxx" called 011972599544327 in context default on channel xxxxxxxxxx to channel SIP/100.100.100.100-00000004 but it really depens what dose your cdr_custom.conf file say
10:52.14knarflyI enabled logger.conf last night too. the full file shows something interesting at the time one of the illegals attempts happened. two sip registrations failed but then the call was attempted. since my voip_account is now empty of credits the call failed.
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10:53.57schmidtsif this two fields "SIP/100.100.100.100-00000004","SIP/my_voip_provider-00000005" are the normal channel and dstchannel it whil mean, an incoming call was going out again to your provider
10:54.19schmidtsyou should split up the incoming and outgoing context to avoid such things
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10:54.54upbmaybe he was totally owned :p
10:56.11knarflyyes, that is true. but then the question becomes how do the callers gain access to my asterisk server. the sip.conf accounts are only a few and I don't see the calls being used by any of the existing extensions
10:56.48knarflyupb: can you elaborate on that point?
10:58.35schmidtsmaybe its just a forked ip thing
10:59.39schmidtssending an invite package witn an forked ip, which looks like the ip of your voip provider to dial a number which would be called over the trunk to your voip provider will cause this
11:00.31knarflythe entry in the full file shows a bunch of IP's trying to register at once...is that what you mean?
11:00.54schmidtsnope if an account was used, the channel will show this
11:01.20schmidtsit could also be a transfered call, but you will see this in your log file
11:02.23drmessanoDo you have allowguest=yes?
11:02.34knarflythat's the problem. I'm not finding anything in the log files which shows one of the existing sip extensions being used...how else could they gain access?
11:02.59knarflyno allowguests
11:03.15schmidtsthats pending how your extension.conf looks like
11:03.38schmidtsas i said with forked ips it would be possible
11:05.20knarflyok thanks.
11:05.39schmidtsshow us your extensions.conf then we can say what could happens
11:07.10upbhmm forked ip you mean forged ?:P
11:11.41schmidtsfaked
11:12.28schmidtsyes forged, thx ;)
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13:20.22Kattyderrrkaadurrr
13:22.19beekwaves to Katty
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13:34.27Kattyhugs on beek
13:35.15[TK]D-FenderKatty: Mew.
13:35.22Katty[TK]D-Fender: mew.
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13:37.00Kattyso my date turned out to be a fox/glenn beck supporter.
13:37.29pigpensmart date.  Wise, wise person.
13:38.11drmessanoDid you stab him and hide the body, writing it off as another Fox viewer removed from the gene pool?
13:38.21Kattyno i just tried to remain polite.
13:38.40Kattybut it was difficult to not laugh at him.
13:39.23drmessanoThat's almost as bad as being on team Jacob and sitting across from an Edward fan
13:39.35[TK]D-Fenderdrmessano: HERESY!
13:39.40drmessanoThinking the whole time "Vampires, lol"
13:39.52Kattymore like, sparkles, lol
13:39.57drmessanolol
13:40.09pigpenI've already converted my quota of Liberals today.  So someone else will have to fall on this one.
13:40.12drmessanoDid your date sparkle?
13:41.13drmessanopigpen:  What the hell is conservative doing in an OPEN SOURCE IRC channel?   Are you a patent troll or do you work for big brother and are observing how the "other side" lives?
13:41.21Kattydrmessano: no. he didn't sparkle.
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13:41.33[TK]D-Fender[09:40]<drmessano>Did your date sparkle? <- diamonds in eyes = rocks for brains
13:42.32Kattyhis accent was atrocious
13:44.42drmessanoHe was probably married anyway
13:44.49drmessanoYou know how THOSE people are
13:44.53pigpendrmessano, I know many devs, kernel, strongswan, gentoo, that are Conservative.
13:44.58pigpenKatty, what accent?
13:45.04pigpenie: nationality?
13:45.33pigpenI am a Texan.  Other than my "Ya'll" you would never know.
13:45.40Kattypigpen: TN
13:45.42drmessanopigpen:  It doesn't surprise me that Gentoo devs are a bunch of Conservatives.
13:45.55Kattypigpen: i've met people from alabama that sound better.
13:46.04Kattyno offense to alabama, of course.
13:46.18*** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman)
13:46.37pigpenKatty, yeah, me too.  Some have strong drawl.
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13:48.12pigpendrmessano, wow.  Classifying people rather generally today eh?  You sound like the NAACP.
13:48.13Kattyit's not attractive.
13:48.40drmessanopigpen s/today/every day/
13:48.55drmessanoThat's pretty much how I roll
13:49.19pigpendrmessano, well, at least you are consistent.
13:49.24pigpenthat is a good quality.
13:51.32drmessanoSorry, little distracted here.. Sitting in the Fox News chatrooms telling everyone Reagan is dead and watching them all get emo and quit.
13:51.45Kattyhehehe
13:52.45pigpendrmessano, Well, to be fair to the channel, this topic should be dropped.  I won't get into this as I am probably one of the last you want to debate.
13:53.27tzafrirNothing would suprise me about Gentoo people. Name their distro after a file manager...
13:53.44drmessanopigpen:  Your aggression isn't necessary
13:53.50Kattysimmer down boys.
13:55.14pigpenAggression?  Wow, next you will be calling me racist.  It's time to drop this.  Bring it up somewhere else.  This is not the purpose of the channel.
13:55.36drmessanopigpen:  Seriously, you don't know me. Enough.
13:56.01drmessanopigpen:  No need to go on and on with your characterization of me.
13:56.13pigpenOk, well, I quess I won't be hanging around in here today.  Have a nice day libtard.
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13:57.28drmessanoYes, and people wonder why conservatives have a bad name.
13:57.32drmessano:/
13:57.45Kattytestosterone spill in aisle 5
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13:58.27drmessanoKatty:  That was Estrogen
13:58.30[TK]D-Fenderhates all people equally :)
13:58.54adyn[TK]D-Fender: I aggree
13:58.56drmessanoI Googled that through a series of tubes
13:59.03drmessanoRIP Ted Stevens
13:59.14Kattyoh?
13:59.22Kattyaww i could have made a friend )=
13:59.50radenKATTY !!!!!!!!!!!!!!!!!!!!!!!
13:59.54raden:D :D :D
14:00.57Kattyohai :>>>
14:01.01Kattyhugs raden
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14:02.52radenhugs Katty
14:02.58radenwhats happening :) ?
14:04.21rue_houseexperts are getting shorter with noobs and people have been throwing not-nice stuff, it ended with hugging....
14:06.03Kattyraden: feeling all meh about my current dates.
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14:07.00anny__hey all
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14:07.16[sr]hi
14:07.17[sr]:)
14:07.27Kattyherroes.
14:09.44radenKatty, current dates ?
14:16.05[TK]D-Fenderraden: Mostly prunes :p
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14:19.37creativxgod damn fc4
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14:24.46doolittleworkhi there i need some help please, i need to get some data captured by asterisk whiles in a call and write it to a mysql database, i want to use phpagi
14:25.07Kattyi need some help too.
14:25.09doolittleworkwhere can one get more info or howto to learn how it works?
14:25.10Kattyi'm out of caffeine.
14:25.12Kattyvolunteers?
14:25.24doolittleworklol whats up Katty i can try
14:26.16Kattysee above.
14:26.33doolittleworki just joined above empty
14:26.52Katty09:25 < Katty> i need some help too. 09:25 < Katty> i'm out of caffeine.
14:27.01doolittleworklol
14:27.28beekfires up the grinder and french press and brews katty a steaming mug of French Roast.
14:27.42Kattywoo!
14:27.43Katty<3
14:27.47doolittleworkcaffeine or ""milk"" seeing u a "KATTY"
14:29.27Dovidanyone have an issue with EYE Beam/X-Lite that on getting the 200 OK the EYE Beam/X-LITE sends a BYE ?
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14:31.13Kattydoolittlework: no need, i don't need tryptophan
14:31.17[TK]D-FenderDovid: Perhaps you should show us
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14:32.48Kattydoes milk have iron?
14:33.20doolittleworkKatty: no its white iron is grey in colour
14:33.29Dovid[TK]D-Fender. It’s a very wierd issue. not so much involving Asterisk. If I have Eye Beam -> Proxy -> Gateway when the proxy sends the 200 OK and the IP for the RTP is the gateway then the EYE Beam hangs up. If I do Eye Beam -> Proxy -> SIP Provider and in the 200 OK it gets the IP of the SIP provider then there is no issue. it seems to be an issue with the audio which makes no sense
14:33.33Katty/facepalm
14:34.57Kattyhmm. 1 cup of whole milk contains .1mg of iron.
14:35.49doolittleworkDovid: check your codex
14:36.19Doviddoolittlework: I know codecs is not an issue since there is early media. If there was a codec issue I would never get the early media
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14:43.27Kattyglomps leifmadsen
14:43.33Kattyleifmadsen: how're you dear
14:43.39leifmadsenI do not exist!
14:43.43Kattylies.
14:43.52leifmadsenback shortly, I have 15 mins to get a phone setup :)
14:43.59Kattykbai
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14:44.29pathis it possible to Dial to both 'ingroups' through AGI ? like using SIP&
14:45.10pathhttp://pastebin.com/dc4RhGmj something like that
14:46.55paththat could work like this ? http://pastebin.com/X63KJtec
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14:48.43[TK]D-Fenderpath: One of those just calls an AGI.  The otehr calls another one after.  Whats the point ehre?
14:48.45[TK]D-Fenderhere*
14:49.03Kattynarwhals.
14:49.23[TK]D-Fenderpath: And what is with the "&" in there?  This isn't a DIAL command
14:49.33[TK]D-FenderKatty: Horny :)
14:50.27Katty[TK]D-Fender: i think i missed the reference.
14:50.35Katty[TK]D-Fender: oh.
14:50.38Katty[TK]D-Fender: nevermind.
14:51.05[TK]D-FenderKatty: Zoologically challenged you are! hhmmmmmmMMM!?!? </yoda>
14:51.11Kattyhurrrduuuhurrr
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14:51.42Kattyhi Chainsaw
14:51.54ChainsawHi Katty :)
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15:35.31doolittleworki dont know if my mind just snapped, but it looks like there is a agi command for all the applications you can call in asterisk, please let me know if i am trying to blow smoke up my own &(*@ or is this correct?
15:36.15doolittleworki take it all the asterisk commands is written in agi?
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15:41.54Qwelldoolittlework: huh?
15:41.55CoolCat2012hi all!
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15:42.23CoolCat2012what i need to have the command cdr mysql status avaiable?
15:42.39CoolCat2012(im tring to enable the cdr stuff, but no luck so far)
15:42.48QwellDo you have asterisk-addons installed?
15:43.20CoolCat2012Qwell yes, i just installed it. but in the cli it still no accept this commanda! :/
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15:44.50WIMPyIt's monday!
15:45.28EmleyMoorWIMPy: Agreed - but a bank holiday
15:45.57WIMPyLooks like a normal monday in #asterisk.
15:48.20CoolCat2012module not active.
15:48.32CoolCat2012should i add any entry in the modules.conf?
15:49.28QwellCoolCat2012: no.  you need a config file for the cdr_addon_mysql
15:50.04fleawow asterisk compile on ppro's didnt take nearly as long as i expected
15:50.06CoolCat2012cdr_mysql.conf ?
15:50.26QwellCoolCat2012: yes
15:52.48CoolCat2012i have it ok. strange.
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15:55.12CoolCat2012i dont have this lib cdr_addon_mysql.so
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15:58.21fleahi all> been planning a ivr project, just installed asterisk, reading reading reading ... first dive just 'hi' :)
15:59.07KattyAHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHHH
15:59.09Kattyasplodes.
15:59.12Kattyk, all better.
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16:10.02mesfetHi. Please could you tell me a way to define a queue with linear strategy, where the members are dynamic? To better explain, sometimes the incoming queue should be routed to extensions 11, then 12, then 13, then 14,   while sometimes the incoming call should be routed to 15, 12, 13, 14 because extensions 15 take the control over 11.
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16:22.31doolittleworkhow does one install text2wav?
16:22.39EmleyMooris looking for a place to buy an AEX410E in the UK
16:22.51fleadoolittlework its in the festival package
16:23.05doolittleworkthx flea
16:23.44doolittleworkhow does one check if an app is installed "show application festival" does not work
16:24.25fleadoolittlework i was referring to festival as in an os package
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16:26.19doolittleworki see that i have festival installed but i can not seem to get this -->   $agi->text2wav('what the hell why does this not work'); to WORK?
16:28.22EmleyMoorI think it's text2wave
16:28.42*** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db)
16:28.57fleadoolittlework, maybe not related, but check the path of the text2wave script from festival
16:29.08fleamine was placed here:  /usr/share/doc/festival-1.96_beta/examples/text2wave
16:29.27EmleyMoor/usr/bin/text2wave on Debian
16:30.03fleamy portage is intentionally way out of date
16:30.32drmessanoUsing a 486?
16:31.20flea<-- ?
16:31.32fleapentium pro's
16:32.33doolittleworkflea: where would i find this script
16:32.44fleadoolittlework what os/distro?
16:32.50doolittleworkcentos
16:32.53doolittlework5
16:33.12fleaoh no, i have to put on my rpm hat
16:33.28doolittleworklol
16:33.28flearpm -qil festival | grep text2
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16:36.42doolittleworkflea i found it under /usr/bin/text2wave do i need to edit it?
16:37.14doolittleworkwhere do you need to specify the path in asterisk?
16:37.23fleai have just finished installing asterisk, i know nothing about it yet
16:37.39doolittleworkare u new to asterisk?
16:37.43fleaday 1
16:37.59doolittleworklol and you learning me about festival
16:38.10doolittleworkgood on u
16:38.24fleai dont have a subscriber line so i'm not sure how well i can test asterisk
16:39.13doolittleworki can setup a sip trunk for u
16:39.18fleamy goal is to use asterisk with pri but i dont have this yet :/
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16:39.56doolittleworkwhere is the flea hopping from?
16:39.58fleahmm doolittlework i would be interested in this i think for test.  but i must do my diligence and read docs for some days first
16:40.05fleausa, nc
16:40.16doolittleworkpri big install?
16:40.22fleasmall
16:40.55doolittleworkjust shout if ou need help people on this site rocks
16:41.10drmessanoYes.  We do.
16:41.54doolittleworkkissing assssssssss for future help
16:42.07Naikrovekheh
16:42.19drmessanoThat won't help
16:42.24doolittleworklol
16:42.36Naikrovekpeople will help you whether or not you kiss ass.  if you do your own legwork then we help more than if you kiss ass
16:42.37Naikrovekactually
16:42.37drmessanoThis is #asterisk.. first time you say something stupid, you're going into the dog food
16:42.46Naikrovekkissing ass will get you nowhere in here
16:42.50Naikrovekfor the most part
16:42.55*** join/#asterisk cusco (~trilili@213.63.137.210)
16:42.57cuscohi...
16:43.13drmessanoKissing ass doesn't help, but bribes are always welcome
16:43.14bougymanis fred here?
16:43.26cuscothere is someone with a 3cx softphone, that everytime he enters a dtmf in app Read() every digit gets doubled up
16:43.30cuscoinstead of 13, 1133 shows up
16:43.35drmessanoI accept paypal, amazon payments, and beer
16:43.53bougymancusco: go into 'advanced' on the 'connection' settings and turn off 2 of the 3 dtmf types
16:43.57cuscowhat is most common cause for dtmf's to be doubled
16:44.05doolittleworki see i have a festival.conf file but it is blank is this the reason my text2wav is not working in my agi?? kissy kissy
16:44.06bougymanit sends like 3 diff dtmf on every keypress, by default.
16:44.07drmessano^^^^
16:44.12bougymanjust went through this with 3cx
16:44.15cuscobougyman: I found dtmf options, supporting inband and sipinfo
16:44.23bougymanyes, choose just one.
16:44.32cuscobut if softphone supports several is it not good?
16:44.36fleatelephony is not even my area, by day i'm unix, but i want to get into this for own project for fun
16:44.37bougymanthere should be 3 on 3cxphone4
16:44.38drmessano..
16:44.48drmessanoIt's sending ALL of them
16:44.52drmessanoTurn off all but ONE
16:44.54cuscoahm...
16:44.58cuscoOK
16:45.00cusco:)
16:45.00bougymanwhat drmessano said.
16:45.09doolittleworkwq!
16:45.16doolittleworklol
16:45.31drmessanoSupporting multiple DTMFs types is great.. it's just doing it at the SAME TIME
16:45.37drmessanoshoots 3CX
16:45.55bougymanwell, the phone switch should narrow them down to one.
16:46.00bougymanbut I don't know many which do.
16:46.04cuscodrmessano: got it
16:46.14cuscoasterisk shouldn't support them all then :P
16:46.16cuscobug!
16:46.38drmessanoYeah, DTMF type should be negotiated, not implemented shotgun style
16:46.52cuscobougyman: there was a third option yes
16:47.00drmessanoWait
16:47.02cuscoI think I have one of them disabled in asterisk
16:47.14drmessanoInband, SIP Info, and YES?
16:47.54cuscofirst one was som payload 101 selected
16:48.03cuscodon't care, inband is working dine
16:48.03drmessanoThat's the most horrible boolean ever
16:48.04cuscofine
16:48.13bougymanpayload 101, sip info, and inband are the options.
16:48.18cuscoyep
16:48.18drmessano"We support, SIP, IAX2, and VoIP"
16:48.26cuscolol
16:48.56cuscoby the way, anyone uses any soft phone web based cross platform?
16:49.01cusco(no activex)
16:49.35bougymanwaiting for styrophone before I try that.
16:49.41bougymani don't like the flash phones.
16:49.44bougymannor the java ones.
16:49.50cuscowhat else is there=
16:49.51cusco?
16:49.54bougymanstyrophone will be a browser plugin.
16:49.55drmessanoSilverlight
16:50.03bougymanhttp://styrophone.com/
16:50.04drmessanolol
16:50.07cuscolol
16:50.18bougymanthey announced at cluecon, but still haven't released the beta
16:50.27cuscoI tryed mozz iax a plugin as well from mozilla
16:50.31bougymanthey demoed it, i think it's politics (open source vs non) now.
16:50.32drmessanoVaporware
16:50.38drmessanoHow long ago was cluecon?
16:50.42bougymandrmessano: no, it was demoed.
16:50.47bougymanbeginning of august.
16:50.53drmessanoYes, Vaporware
16:50.55bougymanthey said the license was undecided.
16:50.58drmessanoThats 4 weeks
16:51.13doolittleworkwhat is the agi->send_text("what the hell"); used for?
16:51.17bougymanthey use it internally at the co who developed it, it's not vapor.
16:51.33drmessanoIf they can't put out a Beta in 4 weeks, they should just park at the Cemetery
16:52.01bougymangoogle seems to have gizmo5 integrated, i never did see what they used for that.
16:52.06doolittleworki can not send any data to my eybeam phone, does anyone know of of a phone that supports sending messages tooo?
16:52.06bougymanis it pure websockets?
16:52.17bougymanif so, are they making people use chrome/safari?
16:52.44bougymandoolittlework: zoiper, 3cx (business version), miranda, um... others.
16:53.00doolittleworkbougyman: thx
16:53.06cuscodoolittlework: twinkle does
16:53.51cuscoI agree with drmessano, it will become vapourware if nothing happens soon
16:54.25doolittleworkhow does these phones compare to eyebeam?
16:54.52cuscoyou should try them and decide for yourlsef
16:55.08doolittleworkbuzy with zioper will update
16:55.16drmessanodoolittlework:  yes, no, yes, yes, no, yes, no, yes
16:55.28drmessanodoolittlework:  Those are my comparisons
16:55.40*** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu)
16:56.54bougymancusco: to me vaporware means no one uses it.
16:57.01bougymanif they use it internally it defies the definition.
16:57.02*** join/#asterisk kalimc (~mcurry@proxy.hostopia.com)
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16:57.46drmessanoVaporware is a word used to describe products, usually computer hardware or software, that were not released on the date announced by their developer, or that were announced months or years before their release
16:58.28WIMPyOr were announced many times but nevver surfaced.
16:58.46bougymanah, but it's never been delayed.
16:59.02bougymanso even by wikipedia's definition it's not in the category.
16:59.16bougymanit's neither been delayed nor months.
16:59.30kalimcis there a place in the documentation that explains all the possible results of the 'Status: ' field that resides in the dial file 'Archive'?  (files residing in /var/spool/asterisk/outgoing_complete).  I see "Complete" but wondering what other status or fields could reside in it
16:59.58drmessanoWell, the key term is "released".. Really doesn't matter if they're using it internally or using it to wallpaper the bathroom
17:00.04coredumbHello there, i'm doing basic testing with a voip handset + SIPdroid, i can call from both sides but can't hear anything in the handset hearing... Any idea how to troubleshoot this?
17:00.52Qwellcoredumb: use an app that actually works.
17:00.55Qwellsipdroid is terrible.
17:00.59*** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net)
17:01.41coredumbQwell: oh it has worked flawlessly for me with external providers... Any advice on "another" good SIP app for android ?
17:02.30kalimcSorry guys, I just found the answer for my question here- http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+callfiles.txt
17:02.32kalimcthanks
17:02.55doolittleworkwhat is the protocall called for sending text to channels?
17:03.13bougymancoredumb: linphone seems to work well on the droids i've seen.
17:04.00coredumbok let's say i still have the problem with linphone where should i look out?
17:04.37Kattyso what can i do with a free weight to help my triceps.
17:04.45Qwelllift it
17:05.12Kattyhurddahurr
17:05.18Kattypost gifs.
17:05.54Naikrovekpushups work triceps
17:05.58Naikrovekamong other things
17:06.15Kattycan't do any of those.
17:06.17Kattytoo weak
17:06.29Naikrovekthen lay on back and push weights up into the air
17:06.41Naikrovekif you can do it more than 12 times in one go, get heavier weights
17:06.53Kattylike bench press?
17:06.55Naikrovekyes
17:07.01Naikrovekbut free weight, dumbel in each hand
17:07.09Naikrovekfocus only on moving your elbows
17:07.20Kattybut like bench press, right? not from the sides up
17:07.26Naikrovekyes
17:07.28Kattyk
17:07.43Kattyi can do girly pushups, will those help?
17:07.45doolittleworki am learning phpagi need some help please why would this not work  http://pastebin.com/N7qUTZaz
17:07.47*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
17:07.49Kattyor is that a waste of effort?
17:07.49Naikrovekhold weights near your neck then extend arms, bending mostly elbow
17:07.53*** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net)
17:08.05Naikrovekyes girly pushups will help, but not as much as the normal variety
17:08.15doolittleworkagi comes back with result -1
17:08.26Kattygirly pushups or bench press thingy?
17:08.34Kattywhich should, theoretically, be more effective?
17:09.01Kattyi could throw punches with free weights too
17:09.04Naikrovekif you want to focus on triceps and nothnig else, the on-your-back-bench-press-with-dumbells thing will be better
17:09.06Naikrovekyes puches
17:09.07coredumbbougyman Qwell ok i have the exact same problem with Linphone... I can call from/to both sides, but can't get anything to the handset...
17:09.22bougymanthis may be a handset issue.
17:09.24NaikrovekKatty: anything that takes effort for you to straighten your arm at the elbow will use triceps
17:09.27bougymanhas there been an update?
17:09.33Kattyooooooooh
17:09.36Kattyi didn't know that
17:09.49KattyNaikrovek: i've been doign dips
17:09.51coredumbbougyman: nope i just changed my provider to register on my local asterisk
17:09.52*** part/#asterisk mesfet (~mesfet@host165-3-static.25-87-b.business.telecomitalia.it)
17:10.01KattyNaikrovek: but i think i've outgrown them
17:10.11*** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt)
17:10.11bougymancodec mismatch?
17:10.13NaikrovekKatty: the elbow is moved through its range of motion by the biceps and triceps.  biceps bend the elbow, triceps straighten
17:10.15coredumbtried different settings but nothing really helping ...
17:10.19chuckpAnyone ever have an issue where dahdi will just randomly unload itself? I am using a Astribank USB and a foneBridge2, when it occurs all devices stop functioning. The only logging items I see is in reference to the dropped USB port.
17:10.35coredumbbougyman: how can i verify that? isn't it supposed to be selected automatically?
17:10.37KattyNaikrovek: sweet. that's good to know
17:10.46bougymanyou'd see it in the debug scroll.
17:11.06NaikrovekKatty: i remember some of my muscular anatomy class.  ping me anytime for questions
17:11.12KattyNaikrovek: that also exlains why my biceps are fine, and the triceps need help.
17:11.19Naikrovekyeah
17:11.23*** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net)
17:11.28coredumbbougyman: -- Remotely bridging SIP/2000-0000000c and SIP/2001-0000000d
17:11.33coredumbonly get that
17:11.39KattyNaikrovek: throwing punches may not help then
17:11.42Naikrovekpulling = bicep, push = tricep.  push some more dudes away from you elaine styel on seinfeld "get OUT"
17:11.44KattyNaikrovek: i dont' do a lot of arm straightening
17:11.57Kattywait nevermind
17:11.58Kattyyes they do
17:12.07Naikrovekyou just let gravity do it probably
17:12.29tzafrirchuckp, what version of dahdi?
17:12.30Kattyi mean the force of it just straightens the arm
17:12.35Naikrovekyeah
17:12.42Kattydoes that still work the tricep?
17:13.11Naikrovekany EFFORT on your part to straigten your elbows will work the triceps
17:13.16Kattyk
17:13.52Naikrovekif you want to build muscle, find a weight or exersize that you can do more than three times in a row, but less than 12.  if you can do more than 12, find more weight or more resistive loads
17:14.26Naikrovekif you want to tone, i think the rule is between 12 and 24
17:14.40Naikrovekor maybe just 12 and up
17:15.04Naikrovekso say you're laying on your bed, with your dumbells up in the air, your arms straight
17:15.21Naikroveklet them down by bending your wlbows much more slowly than you push them up
17:15.30*** join/#asterisk timahvo1 (~rogue@41.223.57.82)
17:15.32Naikrovekpush them up in like 2 seconds, let them down over 4 seconds
17:15.37Naikrovekthis is how you get strong
17:15.52Naikrovekhaving testosterone really helps, so it'll take you a bit longer :/
17:16.20chuckpAnyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182
17:17.13*** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com)
17:17.29nnyanyone know of any linux tools to convert g729 based MCF files?
17:17.38Naikrovekmcf?
17:17.44Naikroveksox or ffmpeg won't do it?
17:18.25nnynot sure
17:18.37bougymantranscode
17:18.48nnyhave a broken implementation of oreka I am replacing with native asterisk monitor + script/ php/ mysql magic
17:19.25nnybougyman: http://www.transcoding.org/ ?
17:19.43bougymanyes
17:19.52nnybougyman: k thanks
17:19.53bougymani've never seen anything it can't convert.
17:20.45nnybougyman: cool
17:21.04nnydoes anyone have a favorite opinion on solutions for ~ 100 concurrent channel recording?
17:21.16nnyi have been told roll your own + asterisk monitor is the best way
17:21.24nnyjust curious what other people think
17:21.46bougymannny: i do 400+ concurrent channel recording
17:21.54bougymanit has more to do with architecture.
17:21.55*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)
17:22.17bougymanwe use 3ware or highpoint raid cards with write-ahead cache and BBUs, getting 6500 iops/sec
17:22.27bougymanplenty for 600 calls at ulaw
17:23.19[TK]D-FenderThese days Even RAID for that seems wasteful....
17:23.29bougyman[TK]D-Fender: what do you mean?
17:23.37[TK]D-FenderSingle SSD, move the files off upon completion.
17:23.39nnybougyman: currently the server is a http://en.gentoo-wiki.com/wiki/HP_ProLiant_DL380_G5
17:23.47[TK]D-FenderSSD annihilates RAID
17:24.11Qwellfor writes?
17:24.13bougymansure but you still pay the price to move to real storage at some point.
17:24.17bougymanramfs > ssd
17:24.18QwellI thought SSD was still ridiculously slow on write
17:24.27nnyyeah ramfs was one of the examples I saw used
17:24.29[TK]D-FenderQwell: > 250MB/s
17:24.30bougymanQwell: i've seen 15000 iops with ssd
17:24.32bougymanso it's faster.
17:24.42bougymanthough I can get 1GB/s writes.
17:24.48bougymanso I'd hope ssd could get that, too.
17:25.55[TK]D-Fenderbougyman: Technically I don't see why you couldn't just use a RAMFS mounted folder and move them from there....
17:26.07[TK]D-FenderAt which point array speed is really moot
17:26.17[TK]D-FenderYeah, almost no reason for high load :)
17:26.23nnyI like the SSD route, cheap (and cheaper than RAMfs) and straight forward. We are considering a daily move to another storage device
17:26.33bougymanno, it's not really moot
17:26.41bougymanyou still get the draw offloading them.
17:27.08[TK]D-Fenderbougyman: Sequential draw at the end of a call as opposed to 400 continuously thrashing writes?
17:27.15Naikrovekyeah
17:27.17Naikrovekwhat he said
17:27.20nnyso a 64 GB SSD could act as a staging platform. There is ample after hour down time to move the files
17:27.26[TK]D-Fenderbougyman: I don't thinkk the comparison even belongs on the same page.
17:27.29bougymandepends.
17:27.38bougymanwe fill up 64G in about 11 minutes.
17:27.43Naikrovekwhat
17:27.45Naikrovekwow
17:27.49[TK]D-Fendernny: Yes, "staging" would be a good term for this aproach.
17:28.16nny[TK]D-Fender: yeah this isn't as high volume, probably max 100 concurrent
17:28.21[TK]D-Fenderbougyman: That is a scary number I'd like to see a breakdown for...
17:28.38bougyman[TK]D-Fender: a bunch of call centers doing collections.
17:28.47[TK]D-Fenderbougyman: on a single server?
17:29.05bougyman2000-5000 calls per server.
17:29.13Naikrovekreally
17:29.16Naikrovekwhat are those server specs
17:29.19Kattyi read that as 2000-5000 cals per serving
17:29.27NaikrovekKatty: hungry?
17:29.28QwellKatty: part of a complete breakfast
17:29.29bougymanxeon 8-way mostly, a couple 24 core opterons.
17:29.34KattyNaikrovek: just ate
17:29.43Naikrovekbougyman: dual quad xeons?
17:29.48[TK]D-FenderKatty: I highly recommend the "Stop Eating You Fat Bastard Diet".  I lost 70 pounds in 5 months :)
17:29.50bougymanNaikrovek: yes.
17:29.51doolittleworkhow does the 3cx phone rate?
17:29.56Naikrovekbougyman: nice
17:30.07bougymandoolittlework: our users love it, but I think it's because it looks like an iPhone.
17:30.12nnyactually
17:30.12Katty[TK]D-Fender: that doesn't really work well when you're anemic
17:30.21nnythis server is dual quad xeons, 3.8
17:30.23doolittleworkis it a soft or hardphone
17:30.25Katty[TK]D-Fender: which i'm still having problems with, but i had some ham on my baked potato
17:30.26[TK]D-FenderKatty: Like I told you before... CORN FLAKES
17:30.33Kattyis eating corn flakes
17:30.38KattyRIGHT NOW
17:30.40Kattyas dessert. tyvm
17:30.41[TK]D-FenderKatty: And that baked potatoe = SIN
17:30.46nnyI am going to setup testing, but I assume it should be able to handle 100 concurrent recorded calls
17:30.52Katty[TK]D-Fender: then i'm a sinner.
17:30.54[TK]D-FenderKatty: Anti-progress
17:31.11bougymanbtw, the new opteron 4x1U servers from supermicro are King.
17:31.19Naikrovek[TK]D-Fender: my wife had surgery and has been losing 10lbs a week for 4 weeks.  it's freaky to watch
17:31.28Kattyyikes.
17:31.31[TK]D-FenderNaikrovek: What kind?
17:31.31bougymanthat's 4 24 core serversin a 1U
17:31.42Naikrovek[TK]D-Fender: sleeve, it's called.  not gastric bypass
17:31.49Kattyis she taking vitamins?
17:31.53Kattyher hair is going to start falling out )=
17:32.00Naikrovekthey basically shrink one's stomach down to about 1/10th its normal size
17:32.13Kattythe band thingy?
17:32.13[TK]D-FenderNaikrovek: is that where they use a wrapper to impede the expansion of the stomch?
17:32.21Naikrovekno, but it works in the same way i think
17:32.29Kattysounds scary
17:32.35bougymansorry, in a 2U
17:32.38[TK]D-FenderNaikrovek: sounds like a nasty choice, much like staples, etc
17:32.38Kattyi'd rather just work out and eat slightly healthier
17:32.42bougymanhttp://www.neqx.com/config.asp?config_id=SMA2U12C
17:32.59Katty[TK]D-Fender: bp is back down to 90/60ish
17:32.59Naikrovekthey just sew the stomach from the size of a stomach down to the size of a small banana
17:33.19Naikrovekthen they take out what can't be used anymore
17:33.21KattyNaikrovek: yeah i think i'd just avoid surgery all together.
17:33.25Naikrovekyeah
17:33.36Kattybut that's just me. i have the willpower to do it
17:33.38Naikrovekshe can only eat like 3 bites before getting full
17:33.44Katty[TK]D-Fender: and my bmi is down 5%
17:33.47*** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com)
17:33.48Naikrovekif she eats a 4th, it's vomit time
17:33.53nnycurrent spinny disk speeds = Timing buffered disk reads:  376 MB in  3.00 seconds = 125.27 MB/sec
17:34.14nnywonder what 1 second of monitor file size is
17:34.22Naikrovek64kbit/s
17:34.32Naikrovekso 8kb
17:34.43Naikrovekper call per second
17:35.05Naikrovek(for g711)
17:35.06coredumboh gosh can't get sound out of that damn handset :'(
17:35.16nnywhat about g729
17:35.17nny?
17:35.18Naikrovekheadset problem after all this?
17:35.19nnyless?
17:35.26Naikrovekg729 is much less.  8kbit/s
17:35.32nnyor 1 kb
17:35.35Naikrovekyes
17:35.38nnythanks!
17:35.46Naikrovekpleasure!
17:37.09*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
17:38.17nnyNaikrovek: are those figures the same for monitor if the output format is wav?
17:38.45nnyand does monitor use WAV or WAV49?
17:38.54*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
17:38.54*** mode/#asterisk [+o leifmadsen] by ChanServ
17:40.21Naikrovekthose are WAV i believe
17:40.21*** join/#asterisk ahowlader (howlader@180.234.33.169)
17:40.41nnyuppercase? So wav49 (native iirc for voicemail files)
17:40.47*** join/#asterisk _structz (~structz@187.37.203.184)
17:40.50_structzHello all!!
17:41.51Naikrovekhello.
17:42.30_structzMy asterisk seems to have a bug
17:42.38_structzasterisk v 1.6.2.11 on debian x32
17:42.53_structzfrom time to time i get theses messages and the asterisk craches  > http://pastebin.com/rW3SyTtu
17:43.46_structzany thoughts?
17:44.22russyou want to get the core
17:44.32russand see the backtrace
17:45.07_structzwhere i find the core dumped?
17:45.08_structzfile
17:45.36russcore dumping is probably disabled on your distro by default
17:45.59_structzdunno
17:46.07_structzits a debian lenny btw
17:48.53*** join/#asterisk Tim_Toady (~moi@77.49.12.31.dsl.dyn.forthnet.gr)
17:52.02_structz??
17:52.56coredumbbougyman: Qwell this is coming from the handset btw, can't get any sound from it when calling outside numbers - works flawlessly with sipdroid - no error in asterisk output... Gigaset S685IP any known problem with that phone?
17:53.04chuckpAnyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182
17:54.23tzafrirchuckp, what versions of dahdi and of asterisk?
17:54.28[TK]D-Fendercoredumb: No.  You may want to actually LOOK at the call....
17:54.59Diffen2hello, are there any variable where i can pickup the b-number från calleg 1 and use it on calleg 2 that are sent out to the user?
17:55.45[TK]D-FenderDiffen2: HuhÉ
17:56.15tzafrirruss, core dumps are not "disabled by distros". Distros provide a separate debug information package
17:56.15coredumb[TK]D-Fender: how am i supposed to do that ?
17:56.18Diffen2<PROTECTED>
17:56.39tzafrirEvent without it you can use gdb to get a trace, but it's rather meaningless
17:56.52Diffen2exten => s,n,SIPAddHeader(Diversion:\"TEST\"<sip:${DID from calleg 1}@domain\;user=phone>) think it should look something like this but i cant manage to get the DID from the first calleg into the diversion header of the second calleg
17:57.15[TK]D-Fendercoredumb: SIP DEBUG.  And provide actual details about where your phone and * are relative to each other networking-wise
17:57.46[TK]D-FenderDiffen2: Show us your attempt
17:58.19Diffen2ok hold on
18:00.18*** join/#asterisk Benwa (~Benwa@unaffiliated/benwa)
18:00.53coredumb[TK]D-Fender: by * you mean all other stuff? phone/asterisk/sipdroid on same subnet sipdroid trough wifi, phone ans asterisk rj45 on same router. sipdroid <> phone calling ok but no sound in phone earing - sipdroid > outside everything perfect - phone > outside ringing but no sound on the other side
18:02.15[TK]D-Fendercoredumb: * = ASTERISk
18:02.25Diffen2~pb
18:02.25infobot[~pb] A "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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:02.29coredumbahah lol
18:02.56[TK]D-Fendercoredumb: and " phone > outside" <- what is "outside"
18:03.03[TK]D-Fendercoredumb: proper description please
18:03.38coredumbgsm phone passing by external provider
18:03.48coredumbor pstn number
18:04.02coredumbexternal provider registered trough asterisk
18:05.09[TK]D-Fendercoredumb: What is this external phone's relationship to *?  Please draw aq direct line here...
18:05.35*** join/#asterisk bsaxon (~bsaxon@12.107.149.61)
18:05.47[TK]D-Fendercoredumb: ie : Siemens ---> LAN -----> Asterisk ---> NAT router ---> internet ------> ITSP
18:05.55[TK]D-Fendercoredumb: for example
18:06.36coredumbsounds like correct to me
18:06.46[TK]D-Fendercoredumb: Because random insertions like "phone" and "GSM" really start confusing things.
18:07.20[TK]D-Fendercoredumb: You sure on that?
18:07.51*** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com)
18:09.01coredumbyep should be right NAT router ---> internet ------> VOIP provider --------> ITSP
18:09.41[TK]D-Fendercoredumb: ITSP = VoIP provider
18:09.57[TK]D-Fendercoredumb: pastebin yuor sip.conf masking only passwords
18:10.06[TK]D-Fender~pb
18:10.06infobot[~pb] A "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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
18:10.08[TK]D-Fender^^^
18:12.53Diffen2ok tk d-fender here we go. http://pastebin.com/khQzjKPc
18:13.00coredumbhttp://pastebin.com/55P1LxbK << 2000 is sipdroid and 2001 siemens
18:14.35nnybtw for those that were discussing with me the ability to write recordings to a local disk as a stage, I came up with this. If you want to check it and make sure I am not crazy please do. This assumes only the disk write speed, no CPU overhead etc.
18:14.35nny10 minute conversation ~ 9,543,680 bytes/9544 KB (15.91 KB/ second)
18:14.35nnyDisk write speed = 150 MB/s, total concurrent write theoretically available = 9414
18:14.47nnyi tested the disk write speed with dd
18:14.51*** join/#asterisk tiny_D (~tiny_D@e178194249.adsl.alicedsl.de)
18:15.09nnysince I only want ~100 to 200 concurrent calls recorded, I am theoretically assuming the disk write speed is sufficient
18:15.22[TK]D-Fendercoredumb: You are missing a LOT of required settings for * to work from behind NAT.  Follow this guide
18:15.24[TK]D-Fender~sipnat
18:15.24infobotextra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
18:15.26[TK]D-Fender^^^^
18:16.06[TK]D-FenderDiffen2: That is not what I asked for.  Show me the complete CALL.
18:16.26coredumb[TK]D-Fender: i mean i tried on the LAN only before trying outside calls, and result is the same no sound outputing from the handset
18:17.15*** join/#asterisk justnulling2 (~jnull@ool-4b7fd02a.static.optonline.net)
18:17.24[TK]D-Fendercoredumb: What's on the other side of that call?  So far all I have is "Seimens".  Please draw a NEW sdriect line like I showed and provide some real details.  Yuo are leaving things out again.
18:17.48Diffen2tk d-fender from asterisk? if so are there any way to filter the notify message and stuff like that?
18:18.10*** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net)
18:18.22nnyDiffen2: try [T (tab)
18:18.29coredumb[TK]D-Fender: siemens ------> LAN -------> Asterisk ---------> LAN -------> SIPdroid
18:18.31nnyit will autocomplete names
18:18.45*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
18:18.54[TK]D-FenderDiffen2: Addheader is only for a NEW INVITE.  You cannot add odd fields to different kinds of messages
18:19.06nnycoredumb: are you sure SIP droid is using wifi?
18:19.08*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
18:19.23coredumbnny: definitely
18:19.30Diffen2ahh ok tk d-fender :(
18:20.02nnycoredumb: I would follow [TK]D-Fender's advice and do a SIP debug
18:20.07nnycoredumb: will tell you everything
18:20.16[TK]D-Fendercoredumb: pastebin the complete call with SIP DEBUG enabeld
18:25.23*** join/#asterisk Cain (~Geek@unaffiliated/cain)
18:28.28KavanSsipdroid is pretty cool...
18:28.39KavanSI've been using siphon on the iphone - great for outgoing calls
18:30.52coredumb[TK]D-Fender nny http://pastebin.com/HaDsp8Dx
18:32.03coredumbi was calling from the phone to the sipdroid
18:32.12*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
18:32.46tzafrirchuckp, ping
18:33.59[TK]D-Fendercoredumb:  All looks local... ok, "iptables --list" on your server now.
18:34.26coredumbno iptables rules
18:36.38coredumbi did try different settings on the handset side but didn't give me better results
18:37.23*** join/#asterisk Iaxy (~bevins@modemcable005.163-22-96.mc.videotron.ca)
18:37.28Iaxyexit
18:37.32*** part/#asterisk Iaxy (~bevins@modemcable005.163-22-96.mc.videotron.ca)
18:41.13*** join/#asterisk timeshell_ (~timeshell@gw.lusi.on.ca)
18:43.38*** join/#asterisk _structz (~structz@187.37.203.184)
18:44.18coredumb[TK]D-Fender: any idea?
18:44.27nnycoredumb: is this only with sipdroid?
18:44.37nnycoredumb: have you tried 2 ip phones directly?
18:45.00coredumbnny: i don't have two ip phones
18:45.17coredumband it reproducible when calling outside trough asterisk
18:45.18nnycoredumb: hmm maybe try xlite
18:45.31coredumbalso tried linphone
18:45.44nnyoutside via SIP?
18:45.47coredumbyes
18:45.54nnylinphone had no audio either?
18:46.08coredumbnope sipdroid/linphone worked flawlessly
18:46.20coredumbsiemens no audio
18:46.22nnyoh
18:46.28coredumbyep
18:46.40nnyso the siemens has no audio.. any chance the phone itself is the issue?
18:47.06coredumbnope it works when calling it on landline and it worked when registered directly on my ITSP
18:47.27nnycould be a setting still
18:48.01coredumbi tried all that looked like "a setting that could be" no luck, i even upgraded firmware lol
18:48.17nnyso linphone and sipdroid can call out or each other fine
18:48.18nny?
18:48.34coredumbi can call out all audio fine
18:48.51coredumbi can call siemens fine but i only get audio from siemens speaker
18:49.03nnybut not from handset?
18:49.44coredumbs/speaker/mic/
18:49.56coredumbyep sorry
18:50.21coredumbsame behaviour when calling outside with siemens ppl hear me talking but i can't hear them
18:51.10[TK]D-Fendercoredumb: You seem to confirm 1-way audio for 2 LAN device.  I don't see that firewall dump.  Ned you should also test each side to * DIRECTLY
18:51.19[TK]D-Fendercoredumb: IE Record + Playback
18:51.36[TK]D-Fendercoredumb: Isolate if one specific end is what is failing
18:52.29coredumb[TK]D-Fender: yeah but it's also one way audio from siemens > outside while it's two way with sipdroid > outside
18:53.41[TK]D-Fendercoredumb: you just showed Droid > seimens, STOP CHANGING THE SUBJECT
18:53.50[TK]D-Fendercoredumb: so far I don't trust your DEVICES
18:54.05[TK]D-Fendercoredumb: Test them INDEPENDENTLY and as I said your outside stuff should not work period
18:54.06coredumbi don't trust my siemens that i'm sure
18:54.17[TK]D-Fendercoredumb: because of your lack of NAT config
18:54.32coredumbwell it does work with droid anyway
18:54.57coredumbhow am i supposed to test siemens independently ?
18:55.17[TK]D-Fender[14:51]<[TK]D-Fender>coredumb: IE Record + Playback
18:55.46coredumbyeah don't know how to do that :S
18:58.42coredumbi can even call from the droid to the siemens trought landline it works ok
18:58.46coredumb-_-
19:00.10[TK]D-Fendercoredumb: Less story, more test.
19:02.45*** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2)
19:03.27coredumb<PROTECTED>
19:07.19[TK]D-Fendercoredumb: Not what I told you to test
19:08.44coredumbyep done record + Playback it works ok
19:09.19coredumbfrom siemens
19:15.58*** join/#asterisk ManxPower (~manxpower@24.236.78.136)
19:16.21ManxPowerremind me what the correct dial via sip without sip.conf entry?  Dial(SIP/username:secret@host)?
19:17.32coredumb[TK]D-Fender: i did record + playback from both devices it works ok
19:17.33jdoehah...
19:17.48[TK]D-Fendercoredumb: I see nothing....
19:17.50jdoerolling my own system with asterisk cut my workplace's phone bill by an order of magnitude.
19:17.56jdoesad.
19:18.20coredumb[TK]D-Fender: oh you wanted a sip debug of that?
19:19.03[TK]D-Fendercoredumb: For those working calls, just basic CLI.
19:19.10coredumbok
19:20.15coredumbhttp://pastebin.com/CdnKLDjE there it is
19:20.46ManxPowerwhy do I even bother?
19:20.48*** part/#asterisk ManxPower (~manxpower@24.236.78.136)
19:21.08Naikrovek...
19:21.15Naikrovek:(
19:21.24[TK]D-Fendercoredumb: [Aug 30 21:13:56] NOTICE[2430]: channel.c:3980 __ast_read: Dropping incompatible voice frame on SIP/2000-00000001 of format ulaw since our native format has changed to 0x8 (alaw) <---
19:21.35[TK]D-Fendercoredumb: Specify the SINGLE codec to use for your peers
19:21.45[TK]D-Fendercoredumb: And each worked for itself?
19:22.05coredumbyes each worked recording playing back after pressing #
19:22.51[TK]D-Fendercoredumb: dump your server's firewall as I told you to earlier
19:23.03coredumbthere's no firewall there
19:24.47*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
19:25.10coredumbAh nice now it's working between siemens and droid :)
19:25.50[TK]D-FenderIt's a MIRACLE
19:26.30Naikroveklol
19:28.51coredumbThx pointing that out
19:30.12*** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun)
19:32.33*** join/#asterisk justdave (~dave@unaffiliated/justdave)
19:38.02*** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1)
19:47.36NEEDINGHELP123do we have any asterisk developers here? looking for a quick project to be done for a high price in a short time ...
19:47.46*** join/#asterisk uqlev (~yuriy@91.184.221.31)
19:48.26russellbi'll do it for 1 million.
19:48.39NEEDINGHELP123man no need for those comments please
19:48.42russellbk!
19:49.00NEEDINGHELP123it is $1k - $3k
19:49.02Qwellrussellb: yeah, man!  can't you see he's needinghelp?
19:49.08NEEDINGHELP123looool
19:49.24Qwell$1-3k is not exactly a "high price"
19:49.47NEEDINGHELP123yeah i guesy our right
19:49.48NEEDINGHELP123but still
19:49.53NEEDINGHELP123it's not exactly a huge project
19:49.56NEEDINGHELP123well it's pretty huge
19:49.57NEEDINGHELP123;)
19:50.07*** join/#asterisk titter (~titter@c-98-208-152-139.hsd1.fl.comcast.net)
19:50.19[TK]D-FenderClearly its worth $5000 then :)
19:50.32NEEDINGHELP123if it could be done today
19:50.35NEEDINGHELP123i would be willing to move nearer
19:50.38NEEDINGHELP123it is all about time
19:50.48titterSay all Dahdi channels are in use, would dial status report chanunavail?
19:51.20tzafrirtitter, what do you mean by "in use"?
19:51.24[TK]D-Fendertitter: "chanunavail" is an application, not a dialstatus
19:51.39[TK]D-Fendertitter: "congestion" would be the dialstatus normally
19:52.05titter[TK]D-Fender: Thanks, I was going by this http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
19:52.40*** join/#asterisk geneticx (~geneticx@host-208-88-126-198.biznesshosting.net)
19:52.42justnulling2hi, i was asking yesterday how to fix oneway audio with Packet2Packet or disable p2p altogether
19:52.56[TK]D-Fendertitter: Considered looking at the call?
19:54.39titter[TK]D-Fender: I am about to pull logs, I just figured I would ask first.
19:57.26leifmadsenNEEDINGHELP123: what is the project?
19:58.56NEEDINGHELP123a mysql central database that feedsremote asterisk's  with live peer information, a little more detailed
19:59.05NEEDINGHELP123i have the structure, etc already
19:59.09NEEDINGHELP123and the full write up
19:59.16*** join/#asterisk eric_hill (eric_hill@204.94.175.11)
19:59.24NEEDINGHELP123pretty much, load alancing
19:59.57leifmadsenyour first sentence does not lead me to think "load balancing" in my mind
20:00.08eric_hillCan someone call IBM's technical support number (800-426-7378) and verify that they aren't "answering" the call, just playing pre-audio?
20:00.17Qwelleric_hill: most 800 numbers do
20:00.18*** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk)
20:00.21Qwellwell, many.
20:00.40NEEDINGHELP123leifmadsen well, they have to be in the asterisk and then dialed randomly or so to perform some kind of load balancing with call limits inplace
20:00.42eric_hill@Qwell: It's super annoying.  We hang up after 30 seconds of no-answer.
20:00.52Qwellmmhmm
20:01.02NEEDINGHELP123to be honest if you know what your doing, it could be explained alot quicker and more descriptive at the same time
20:01.08leifmadsenah, you want a multi-system dialer
20:01.11NEEDINGHELP123and it probably would not take a long time, but i've had enough headache
20:01.13NEEDINGHELP123no,
20:01.14Qwelleric_hill: fwiw, it saved them about a penny.
20:01.17NEEDINGHELP123i need something to take my traffic from my clients
20:01.26NEEDINGHELP123and read from a central DB of provider's
20:01.37NEEDINGHELP123like a ,,, arbinet kind of system but tiny
20:01.49leifmadsenyou mean call distribution then
20:01.51NEEDINGHELP123and equally or ranodmly distributes that traffic
20:02.00NEEDINGHELP123well yes, but reading from an alread yexisting structure live
20:02.06leifmadsenwell obviously
20:02.10NEEDINGHELP123and noticing states held in a database, ie: 1 or 2 dead or alive
20:02.13leifmadsenthat still doesn't sound like load balancing to me
20:02.19NEEDINGHELP123well then it's not load balancing
20:02.25NEEDINGHELP123as i said my wording is probably confusing in itself
20:02.34leifmadsenanyways, that is a much more expensive project to implement than you think
20:02.41*** join/#asterisk KavanS (~KavanS@unaffiliated/kavans)
20:02.42NEEDINGHELP123how do you mean?
20:03.19leifmadsenif you all the scripts need to be written, call traffic tested and distributed amongst multiple boxes, etc... then I wouldn't do it for $3000
20:03.27leifmadsenit would probably be closer to $10k
20:03.33leifmadsenbut that's just me
20:03.39NEEDINGHELP123are you serious?
20:03.40geneticxHi everyone, I started having this weird problem calling extensions on one of our branches that run Cisco's CallManager. We have a dedicated VPN tunnel just for our sip trunks and everytime we try to call any of their extensions the call drops after about 20-40 seconds into the call.. VPN does not drop during this time. Can someone please take a look at my pastebin, any help is appreciated: http://pastebin.com/9Tx5VHga
20:03.42leifmadsenthen again, I know what I'm doing (sometimes)
20:03.48*** join/#asterisk russ (~russ@206.29.188.232)
20:03.48NEEDINGHELP123can i PM you please?
20:03.50leifmadsenNEEDINGHELP123: I'm never serioues
20:03.52leifmadsenserious*
20:04.11leifmadsenNEEDINGHELP123: I was just curious what you were trying to do -- I'm heading back to the basement to finish building this VM server
20:04.21leifmadsenI'm not looking for other projects at the current time
20:04.36leifmadsenNEEDINGHELP123: I'm not too sure where I acknowledged a PM :)
20:04.53NEEDINGHELP123no issues
20:05.08NEEDINGHELP123goodluck with your VM server bro
20:06.16*** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net)
20:06.55t_dot_zillaif a codec is listed when you type "core show codec" does that mean its installed?
20:07.09t_dot_zilla"core show codecs"
20:07.45t_dot_zillawe have a licensed copy of g729 but i can't tell if it's installed
20:07.58*** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com)
20:08.00[TK]D-Fendert_dot_zilla: "help g729"
20:08.25t_dot_zillano such command
20:09.24t_dot_zillaNo such command 'g729'.
20:09.47t_dot_zilladoes that mean it's not installed?
20:10.33[TK]D-Fendert_dot_zilla: Likely.  try loading the module directly
20:14.03t_dot_zillawith "module load format_g729.so" ?
20:14.15t_dot_zillai get "Module 'format_g729.so' already exists."
20:15.39paulc@leifmadsen: got a sec for a quick/easy question (that is nothing technical)?
20:16.13[TK]D-FenderFormat != codec
20:16.37*** join/#asterisk bcrisp (~bcrisp@70.102.242.138)
20:17.00bcrisphey... anyone heard of an outbound sip provider blocking calls to 1-800 / 1-866 numbers?
20:17.15*** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net)
20:17.41citywokin *.1.6.2.11 using RealTime for SIPPEERS a sip reload does not cause asterisk to re-read the database for updated changes to peers (i changed the codec on a peer, asterisk won't pick it up)
20:17.59bcrispdid you commit your chagne?
20:18.02bcrispchange
20:18.14citywokin mysql, yea.  i can view the row and it's updated.
20:18.50t_dot_zilla[TK]D-Fender: i don't see any g729 codec module
20:19.22bcrispmaybe check the bug db.. not sure on that one
20:19.37[TK]D-Fendert_dot_zilla: If you'er a bit lost I'd check in with Digium support.
20:21.44geneticxanyone?
20:22.08citywokbcrisp: https://issues.asterisk.org/view.php?id=16112 -- looks like it's possibly related to the table strucutre.  i will investigate.
20:22.18bcrispcitywok: cool
20:22.43bcrispive noticed errors showing up relating to missing columns all of a sudden with cdr
20:23.08NEEDINGHELP123ok, i need an asterisk dev / programmer / consultant. i need it now, anyone available? i will offer a ridiculous price, just need the project done ASAP. let me know, thanks.
20:24.14keith4NEEDINGHELP123: what sort of project?
20:24.49paulcdefine ridiculous :)
20:25.32*** join/#asterisk oryxtec (oryxtec@119.152.112.206)
20:25.56seanbrightdeserving of ridicule; foolish; absurd
20:26.13[TK]D-Fender:)
20:27.31[TK]D-Fendercheckout time, BBIAB
20:27.32oryxtec2.8GHz Dual Core, 2 x 250GB SATA Hard Drive, Raid 1. 4GB RAM --- please can any one tell me how many calls this server can handle?
20:28.49keith4oryxtec: lots. google "asterisk dimensioning"
20:28.51russellbat least 2
20:29.19Deeewayne:q
20:29.26Deeewayneoops
20:29.38paulcoh come on, russelb, can't we go with "at least a handful"? ;)
20:29.48seanbrightrussellb: i have stability problems with 1+ calls
20:29.52oryxtec@Deeewayne: at least 2
20:30.02citywokbcrisp: that was easy.  i was missing the mask field, even though i'm using permit/deny.  now to figure out what exact mask does when you already have permit/deny.
20:30.09oryxtec??
20:30.10russellbseanbright: i'm sorry.
20:30.11seanbrightrussellb: i've found that "around 0" is the sweet spot for my installation
20:30.15seanbright:P
20:31.15citywokanybody know what mask does in RealTime for peers?
20:38.49t_dot_zillai'm having trouble installing the g729 codec, i bought a license from digium and "codec_g729a.so" is in /usr/lib/asterisk/modules, but when i run asterisk -rx "reload codec_g729a.so" i get "No such module 'codec_g729a.so'"
20:39.13citywokIIRC you don't need the .so on the end
20:39.44leifmadsenyou do
20:39.56leifmadsenmodule load codec_g729a.so  <-- if not already loaded
20:40.01t_dot_zillaeither way i get No such module 'codec_g729a' or No such module 'codec_g729a.so'"
20:40.14KattyOHAI
20:40.23citywokoh heh, my bad!
20:40.49t_dot_zillaleifmadsen: i got "Module 'codec_g729a.so' does not provide a license key." --i'll run the register script again
20:41.05leifmadsent_dot_zilla: ya, sounds like the license isn't installed
20:41.23citywokleifmadsen: you may know, what does mask do for permit/deny in RealTime.  without it realtime does't work properly, but i cant seem to find any documention on what it is supposed to be set to. i set it to 0.0.0.0/0.0.0.0 and realtime works properly and it follows the permit/deny settings.
20:41.50leifmadsencitywok: my guess would be it's the network mask for a permit/deny statement?
20:41.56leifmadsenI've never seen that field name
20:42.07leifmadsenI'd suggest it either needs to contain 24 or 255.255.255.0 (for example)
20:42.31citywokYea, me either.  but you have to add the mask to the permit/deny statement already.  they are set to 192.168.0.0/255.255.0.0 and 0.0.0.0/0.0.0.0 -- this is what made it functional.
20:43.09citywoka single mask that applied to both permit & deny doesn't make sense, because you can permit & deny at different masks, 24/16/8
20:43.31citywokit works with mask set to 0.0.0.0 but it's confusing that it is required, not documented, and doesn't really appear to do anything.
20:44.04citywokI just posted it on this bug: https://issues.asterisk.org/view.php?id=16112
20:45.51t_dot_zillaleifmadsen: i just ran registration script and still getting "Module 'codec_g729a.so' does not provide a license key."
20:46.11leifmadsent_dot_zilla: it's a commercial product, so I'd just call Digium
20:46.23citywokg729 show licenses
20:46.25leifmadsencitywok: I still don't know the answer
20:46.29citywokok, thanks anyways!
20:46.38*** join/#asterisk abel408 (429863d2@gateway/web/freenode/ip.66.152.99.210)
20:48.05_structzHello can somebody help me out? from time to time i get theses messages and the asterisk crashes  > http://pastebin.com/rW3SyTtu  (Ast 1.6.2.11 debian 5 lenny x32)
20:48.06t_dot_zillacitywok: No such command 'g729 show licenses'
20:48.48citywokah, so the module doesn't load at all.  make sure you followed all of hte installation instructions properly, and that you keyed in your license ID values properly.
20:50.02Qwellcitywok: what version of Asterisk is this?
20:50.17citywok1.6.2.11
20:50.24citywokbuilt and compiled last week
20:50.52QwellThere is no such "mask" field.
20:51.40citywokyea, but without adding it to my mysql structure, asterisk will never update the peer info.  check bug id 16112.  i had the exact same problem until i added the mask field.
20:51.42QwellWhat makes you think it's needed?
20:51.55t_dot_zillacitywok: i followed instructions from http://kb.digium.com/entry/5/
20:52.16citywokt_dot_zilla: then as leifmadsen suggested i'd call digium support
20:52.17Qwellcitywok: what else did you change at the same time?
20:52.23citywoknothing
20:52.35Qwellwhere did you get "mask" from?
20:52.59citywokhttps://issues.asterisk.org/file_download.php?file_id=24435&type=bug
20:53.14citywokin the file attachment with the working table definition, there is a mask field that says #required in 1.6
20:53.16*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:53.31citywoki've also seen the same comments on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip
20:54.26*** join/#asterisk talntid (~talntid@c-76-104-157-191.hsd1.wa.comcast.net)
20:56.37_structzHello anyone?!
20:57.11talntidHi.
20:57.20citywok_structz: without your dialplan it's hard to help
20:57.34citywokQwell: any idea what the heck mask is/does? lol
20:58.40Qwellcitywok: No, no such field appears to exist.
21:00.37citywoklol.  alrighty then. i'll just leave it be and set it to 0.0.0.0/ dpj
21:00.39citywokdoh*
21:06.25*** join/#asterisk zircote_ (~zircote@64.74.105.240)
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21:12.39*** join/#asterisk NiugeS (~NiugeS32@5e0c4f5a.bb.sky.com)
21:14.22NiugeShi all.. anyone know of a program that will allow me to click and dial a tel number on a website in Internet Exploer? It needs to be IE.. i have a SIP Tapi that works via outlook and the asterisk box but need something for web sites.. .any suggestions appreciated
21:16.33*** join/#asterisk coredumb (~coredumb@cust.static.213-200-235-213.cybernet.ch)
21:17.12*** join/#asterisk russ (~russ@206.29.188.184)
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21:28.04antiwireyo dudes
21:28.09antiwirecheck out this http://i37.tinypic.com/k1s0b4.jpg
21:28.35Chainsawhttp://photo1.bababian.com/usr651933/upload11/20080619/syC5yBplS1PNBJh3eahKqNrv2UXMAZGvf130PZA7C+3K2lB8PX3lKTQ==.jpg
21:31.28carrarpeople still use windows?
21:33.24mysterwas that a windows system with 86 days uptime?
21:34.37p3nguinAmazing, isn't it?
21:34.38antiwireThat is the windows administration interface tool for a Zultys MX250 IP PBX
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21:38.32keith4is there an equivalent to zap's "immediate" option (e.g., the red phone) in SIP?
21:38.50Qwellkeith4: ask your phone manuf
21:38.52*** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca)
21:40.00keith4Qwell: ah... right. the phone would just have to "dial" immediately then
21:40.25keith4thanks. will do
21:44.12p3nguinbat phone?
21:44.21keith4yah
21:52.47*** join/#asterisk Cain` (~Geek@unaffiliated/cain)
21:57.24drmessano86 days... pfft
21:57.57drmessanoI've had Windows boxes with 300 days uptime
21:58.32QwellI had a Windows machine with that much uptime once.  It was a laptop that was never used physically.  It was our source code repository.
21:58.40QwellStayed up until somebody stole it...
21:58.48drmessanolol
21:59.07Qwelltrue story too ;/
21:59.11*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
21:59.11*** mode/#asterisk [+o malcolmd] by ChanServ
21:59.17drmessanoThat's messed up
21:59.45*** join/#asterisk russ (~russ@206.29.188.232)
22:00.05Qwellyeah.  a day or two later, somebody that worked there asked us if we could help him reimage his personal laptop.  Just so happened to be the same model.
22:00.30talntidconvienient.
22:00.47talntidbut could he be so ignorant?
22:00.50Qwellyes.
22:01.50drmessanoI had a guy ask me about a workstation I had sitting in a closet.. I told him I wasn't trashing it.. and he asked me "If you do, can I have it"  "yeah, but it's not going in the dumpster".. Two days later it disappeared
22:01.59Qwelllol..
22:02.13Qwellso yeah, people can be that ignorant.
22:02.46*** join/#asterisk zircote (~zircote@64.74.105.240)
22:02.54drmessanoSo next time I saw him, I told him "I guess you wont be getting that machine.. someone stole it.. "  "Oh yeah?"  "Yeah, but it has an IP location tracking device onboard.  I'll have it back from the cops in like no time
22:03.09drmessanoHis mood changed a bit and he went on his way
22:03.41drmessanoOf course, I dont have a happy "The next day someone returned it" ending, but hopefully he crapped his pants, just a little
22:03.57drmessanoor freaked out and threw it in a dumpster.. which is fine.. I needed someone to help me carry it there
22:05.50adynI had someone steal a laptop from a car after someone had just picked it up from servicing it. Then bring it into the store 2hrs later for "wiping the hd and restorin the os" service. Just happened the police were there filling out the report and we always take a picture of the S/N when we serviced it. Go straight to jail... do not collect $200...
22:07.50p3nguinI don't take pictures, but I always note the model and service tag on the invoice.
22:09.25adynat that place we took a snapshot of all edges. It wasn't uncommon for people to ship laptops in for service and then get them back and complain of scratches. So we always took incoming and outgoing pictures just to CYA.
22:11.42geneticxHi everyone, Can someone please shed some light..been trying to solve this issue but no luck. We have a sip trunk from our asterisk to a cisco call manager over a vpn connection but when I call any extension, call drops 20 seconds into the call with this error: SIP/2.0 481 Call leg/transaction does not exist , this problem does not happen if they try to call us, it's only outgoing. This was working fine I don't know what could'v
22:14.52drmessanoThanks for waiting
22:15.06*** join/#asterisk geneticx (~geneticx@host-208-88-126-198.biznesshosting.net)
22:17.14eric_hilllooks like geneticx has the same problem with his IRC connection as his phone calls...
22:21.25geneticxeric_hill: no, actually read somewhere that someone had the same problem and cleared the NAT translations on the terminating the VPN tunnel, so I tried doing this..obiously lost conn for a bit.
22:21.35geneticxrouter*
22:22.16geneticxtrying everything at this point =(
22:26.28*** join/#asterisk xibalba (~reza@216.105.40.7)
22:26.33xibalbahello asterisk users
22:26.50xibalbai was wondering if there was anyone proficient with polycom phones, and if they knew of a way to get the config file info via the web interface
22:33.42*** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger)
22:35.15*** part/#asterisk bsaxon (~bsaxon@12.107.149.61)
22:36.11*** join/#asterisk citrus2 (~citrus2@72.215.183.28)
22:36.28[TK]D-Fenderxibalba: The web data wil reflect the provisioning.
22:36.50[TK]D-Fenderxibalba: But you cannot "extract" full configs via it
22:37.19citrus2hey all,  if anyone can help me for a simple question.             i am trying to use the command SetTransferCapabilty on my asterisk 1.6 system    however it looks like it was last used in asterisk 1.4   was it replaced by something else?
22:37.38russellbUPGRADE*.txt files are your friend
22:39.54xibalba[TK]D-Fender , we had some frauds htis morning. i'm trying to track down how they did it, what i found was the polycom web ui was open to the internet
22:40.18xibalbado you think that's is a possibility ?
22:40.50[TK]D-Fenderxibalba: Well you seem to have jumped on the phone itself being hacked... it IS a possibilty but more commonly server mistakes are teh cause.
22:41.02citrus2russellb:  i searched for that command in upgrade-1.6.txt  but nothign was found
22:41.09*** join/#asterisk grolloj (~chatzilla@209.212.73.50)
22:41.31*** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk)
22:44.23xibalbayeah one big friggin server mistake is that our CFG files are published on one of our servers over HTTP for config
22:44.35xibalbaso anyone who brute forces that box w/MAC addresses.CFG will gt the passwords
22:44.44*** join/#asterisk RypPn (~TuMbL@rosscom.co.uk)
22:45.43[TK]D-Fenderxibalba: Go change that pronto
22:46.17xibalba[TK]D-Fender , i told our CTO. He makes the decision
22:46.39xibalbaapparently it never occured to anyone
22:47.39*** join/#asterisk rayk_sland (~rklassen@mail.mccscs.com)
22:48.33rayk_slandthe asterisknow package doesn't seem to notify users of the presence of voice mail -- any way to re-enable?
22:49.33[TK]D-Fenderrayk_sland: Which "package"?
22:49.48*** join/#asterisk bijit (~bijit@186.4.3.18)
22:50.02rayk_slandthe one from digium with the freepbx built in.
22:50.13[TK]D-Fenderrayk_sland: Show us your configs and voicemail status and the attemp of leaving a VM with debug to show if anything gets sent
22:50.22[TK]D-Fender~pb
22:50.22infobot[~pb] A "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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
22:50.23[TK]D-Fender^^^
22:51.16rayk_slandokey dokey.
22:56.19chuckpAnyone ever see an issue with Dahdi deadlocking like this: http://www.pastebin.ca/1929182
22:56.30chuckpIt just drops randomly, no Kernel Panic or anything
22:57.18chuckpI have to unload the dahdi modules and and load them again (as in rmmod) to get them to reset
22:57.59rayk_slandhttp://pastebin.com/904BTURM or do you need more data...
22:58.46chuckprayk_sland: he said config files too
22:59.46rayk_slandI just thought the block vm flag thing was significant.l I've got lots of configs... you want them all?
23:02.56*** join/#asterisk russ (~russ@206.29.188.232)
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23:04.55[TK]D-Fenderrayk_sland: I do not see a VM being left at all in there
23:05.14titterHow do you guys handle SIP dictionary attacks?
23:05.22[TK]D-Fenderrayk_sland: No configs.  No debug for however you expected the VM to be sent (SIP DEBUG if it is a SIP phone, etc)
23:05.28titterShort of not letting SIP be open to the world
23:05.38[TK]D-Fendertitter: Don't pick words out of a dictionary
23:06.16titterI don't obviously, but I would like to stop them from flooding my messages ... I log registration and drops for a reason and this annoys me
23:06.23tehrabbitt-1p3nguin: hey man
23:08.01rayk_slandthe vm is sent fine. It's accessible if you check it. But the phone doesn't indicate there's voicemail. no flashing light like it had before the upgrade and no broken dialtone...
23:08.50[TK]D-Fenderrayk_sland: I see nothing.  No configs, no debug of the VM being left and from which point it should send the notice.  No "voicemail show users" before & after.
23:08.56*** join/#asterisk Ikarus (~ikarus@dhcp-077-250-195-114.chello.nl)
23:10.46chuckpAnyone ever use Thirdlane here?
23:11.20chuckpNot looking for support just wondering impressions
23:11.40*** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn219.78-98-89.t-com.sk)
23:14.11rayk_slandtry http://pastebin.com/jMFktWTz (sip set debug -- my phone's ip)
23:15.54eric_hilltitter: Use a firewall that can rate-limit SIP REGISTER messages to a reasonable rate (2-3 per second?)
23:16.20tittereric_hill: I found this to be interesting http://mikeoverip.wordpress.com/2009/01/13/asterisk-and-sip-attack-and-monitoring-via-event-correlation-by-using-sipp-and-sec/
23:17.03Ikaruswe have an interesting conferencing setup needed, we need to take 5 participants -> conference (plain old), then having 5 additional clients, which each are listen only and only hear a single conference participant and a another 2 participants, which are joined to the conference and can either talk to the whole conference or select a specific client of the first group to talk to
23:17.33rayk_slandvoicemail show users shows a new voicemail.
23:18.17[TK]D-Fenderrayk_sland: I'm not seeing configs...
23:18.29[TK]D-Fenderrayk_sland: For the phone who should be getting notifications...
23:19.27eric_hilltitter: tx for the SEC link.  Neat tool I hadn't seen before.
23:20.31Ikarusis there a chance of doing this in Asterisk or do I need to write an external app ?
23:20.31rayk_slandlook now. This shouldn't be necessary. these are all stock extensions generated by freepbx.
23:21.35rayk_slandjust wanted to know if anything was 'special' about asterisknow 1.7
23:23.42[TK]D-Fenderrayk_sland: No, there shouldn't.  "stock" proves little, and doesn't help in debugging.  You're either looking at your problem or you aren't.
23:24.01[TK]D-Fenderrayk_sland: and saying a distro release doesn't say much at all
23:26.44Ikarusso uh, anyone actually getting what I try to do ? :)
23:26.58*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
23:29.14rayk_slandanyways what parameter would turn that kind of feature on or off?
23:30.24Ikarusis it possible to have a conference participant only hearing a specific (set of) conference participant
23:30.36rayk_slandThere are two installs that I know of that now don't have an alert at their phones with this version of asterisknow (i know i know it's irrelevant)
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23:52.27Ikarusis chanspy audio supposed to suck this badly, it seems to be highly choppy
23:54.48*** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)

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