IRC log for #asterisk on 20100214

00:03.14*** join/#asterisk jks (jks@193.189.93.254)
00:03.22russellbcrochat: that is correct.  chan_misdn is only for misdn v1
00:04.08crochatrussellb: Is chan_lcr as good as chan_misdn ?
00:05.18russellbi have never heard of chan_lcr
00:08.16*** join/#asterisk ManxPower-work (~EWieling@216.186.151.147)
00:10.11crochatrussellb: www.linux-call-router.de
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00:42.02jayteerussellb, hahahaa, poodle with tin foil hat and Glenn Beck!! Priceless!
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00:43.55russellb:-)
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00:54.31renvorakHi all-  I have spent the majority of a couple days trying to solve a problem and am hoping someone here can help.   I installed AsteriskNow 1.5 and asterisk-addons about six months ago was able to get MySQL working, but not the uniqueid field which is required to link the files to play in ARI.  I understand that I need to pass some variables when recompiling asterisk-addons-1.4.10, but for the life of me, I cannot get it to see asterisk.h despite using
00:59.20KattyHELLO SUNSHINES
01:00.05*** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net)
01:01.22jayteehi Katty
01:01.27Kattyhugs jaytee
01:01.38jayteethanks, I need that
01:01.47Kattyyou alright? (=
01:01.53jayteenope
01:01.57Katty:<
01:02.02Kattyhugs jaytee for good measure.
01:02.04Kattywhatsthematter?
01:02.15jayteehad an EEG yesterday and have to have an EKG next friday
01:02.36Kattyi'm not sure i know what those are for.
01:03.01jayteeelectro-encephalograph and echo-kardio-graph
01:03.16jayteethe latter is like an ultrasound of the heart
01:03.36Kattydo you think something is wrong with your heart?
01:03.36jayteeand the former is just a graphed output of heart rhythms
01:03.58jayteethe EEG had an irregular rhythm
01:04.05Kattynods
01:04.11Kattynot a bad idea to get it checked out.
01:04.25Kattyi hope it's nothing, but i'll keep you my thoughts just the same (=
01:04.28jayteeI've been having dizzy spells every 10 to 15 minutes since wednesday night
01:04.40jayteeso I went to the doctor friday
01:05.04*** part/#asterisk ruben23 (~AGENT@122.55.48.243)
01:05.08Kattynods
01:05.28Kattyhow is your blood pressure?
01:05.46renvorak--First time IRC user.  Can anyone see me or do I have to select all on the right hand side? thanks.
01:06.02jayteemy BP was normal when they checked it yesterday
01:06.13jayteerenvorak, I see you!
01:06.14Kattyrenvorak: welcome to a long-time tradition of ircing (=
01:06.20Kattyrenvorak: we can see you, just keep on typing!
01:06.29Kattyjaytee: good to hear you BP is okay.
01:06.43Kattyjaytee: are EKGs very specific at all?
01:06.43renvorakThanks!
01:06.56jayteeyw
01:08.05jayteeKatty, the one they're going to give me will include an amphetamine type drug that stresses the heart so they can see how it's functioning normally and when under stress
01:08.07Kattyjaytee: or is just kind of a generalized test
01:08.41jayteeEKGs are very specific
01:10.22Kattyamphetamines scare me :<
01:10.45jayteesame here
01:10.58Kattythat stuff is also addictive.
01:11.01Kattyso be careful
01:11.31Kattyi know it's for a good cause.
01:11.51Kattyjust af ew months ago i had a series of dizzy spells.
01:11.58Kattybut it turned out to be a severe sinus infection
01:12.07jayteewell, I doubt if I'll become addicted from one dose and plus I don't know anyone who deals drugs so I'm fairly safe
01:12.26Kattynods
01:12.33Kattyone dose is still enough to do something.
01:13.02Kattykeep me posted (=
01:13.05Kattyi want to know everything
01:13.21jayteeI'm more concerned with the drug giving me a heart attack than getting addicted
01:13.42Kattyi don't think it'll give you a heart attack
01:13.46Kattythe doctors wouldn't take a risk of that
01:13.54jayteesince I haven't been exercising much I don't think the old ticker can take much stress
01:14.00Kattythey'd give you a low dosage just to see how your heart responds.
01:14.24jayteeyeah, and being in a hospital that's rated very high up there in cardiac care helps too
01:14.32Kattymost definately
01:14.35Kattythey know what they are doing
01:14.57Kattythey'll have you right as rain in no time (=
01:15.26jayteethat last line made me think of The Matrix
01:17.08jayteeyay, he's quit!
01:21.22jayteeI love the new fan thing on Facebook: Can this poodle wearing a tinfoil hat get more fans than Glenn Beck?
01:22.43*** join/#asterisk jmcdowell (~nooe@173-112-156-93.pools.spcsdns.net)
01:52.30carrarWhats facebook
01:54.40carrarSounds like some place where you can buy books about cosmetics
01:55.08carraror as about as usefull!
01:59.16*** join/#asterisk rue_house (~rue@24.207.119.38)
01:59.22rue_housesdf
01:59.38rue_houseis there a driver for doing ulaw over a serial port?
02:05.28Kattycarrar: :P
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02:06.09Kattythe dogged help me put away a bag of popcorn. so good of him to save me some calories.
02:08.30rue_housemmm popcorn, you just worked out my supper problem
02:08.43rue_houseI want a serial port to do ulaw for me
02:13.20renvorakanyone had experience compliling asterisk-addons in a asterisknow 1.5 environment?  craps out when it can't find asterisk.h....
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02:45.48ManxPower-work~asterisknow
02:45.48infobotsomebody said asterisknow was based on Asterisk, but is difficult to support in #asterisk for a number of reasons.  Please seek support in #asterisknow instead.
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03:13.41KattyVS just sent me a catalog.
03:13.48Kattyi wonder if someone's trying to tell me something
03:14.25ChannelZyou're not friends with John Kerry, are you?
03:14.34ChannelZdamnit!  Edwards.  I blew that one
03:15.41Kattybaroo? ^_-
03:15.47Kattyi think that went over my head.
03:16.13*** join/#asterisk smooth_penguin (~smoove@59.95.15.112)
03:16.51ChannelZJohn Edwards.  Adulteror
03:16.55ChannelZMaybe I should have said Tiger Woods.
03:17.06ChannelZ(You're white, right?)
03:17.19Kattyno, i'm orange.
03:17.53smooth_penguinIm black and white
03:18.07Katty^- Zebra
03:18.23renvorak@infobot: thanks for the quick reply.  I have posted it in asterisknow as well, but was hoping to see if anyone else had experience as the problem is more with asterisk-addons...  Not to mention #asterisk has 20x the users.   Thanks again.
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03:19.19smooth_penguinno Katty - http://www.ccpo.odu.edu/Research/globec/main_cruises02/lmg0205/pod/gentoo.jpg
03:20.03Kattyyou missed it :<
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03:25.27Kattyhttp://www.natureandwildlife.it/foto/antarctica/11-Antarctica-Emperor-Penguins-Family.JPG <- smooth_penguin came from a happy home.
03:25.51smooth_penguin:>
03:29.01Kattybored.
03:29.47smooth_penguinis going back to bed
03:30.09Kattyninite
03:33.51smooth_penguinKatty, hows the critters
03:35.56Kattywell, the ferrets are fine.
03:36.01Kattyriddick had to go to the vet today.
03:36.07Kattybut he is okay
03:37.09smooth_penguinkk
03:38.00Kattyear infection :<
03:38.26smooth_penguinugh, how did he get that
03:38.34Kattyshe thinks he got water is his ear.
03:38.57smooth_penguin:s
03:40.24smooth_penguinbye Katty, still need to finish up sleep :s
03:51.53rue_houseanyone know how I could make a chan_rs232 driver?
03:53.53ChannelZto what end?  But look at the source for one of the channel drivers and have at it..
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03:57.03Kattyreturns with peanutbutter sammich
03:57.24KattyDear Universe, thank you for peanut butter and it's amazingness. Love, Katty
03:57.35Kattyits.
03:57.40Kattyit's
03:57.44Kattyits'
03:57.46Kattyone of those.
04:02.50carrarand
04:03.00carrarfor peanut butter & chocolate ice cream
04:03.30Kattyyesh :>>>
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04:26.52jmcdowellSo, I am looking for a GOOD asterisk from scratch howto
04:26.57jmcdowelldoes anyone have any suggestions ?
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04:48.39jcarter1hello - i need some help with setting up a connection from x-lite on my windows machine to the asterisk server running on the same laptop on ubuntu installed on virtual box
04:48.57jcarter1i dont kno which ip adress to use to register with the client
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05:15.03jcarter1anyone?
05:16.02ChannelZask virtualbox
05:16.33ChannelZThere's a whole bunch of modes it runs in for network access
05:16.48jcarter1oh
05:17.01jcarter1i will give it a try
05:17.33ChannelZin NAT mode I have no idea how you get connections into the virtual machine
05:19.01jcarter1for what i want to do, it is already enough if my client inside the vm can connect to the server - but virtual box seems to have big trouble with microphone detection, so i cant use my microphone. Now i installed a client in windows, but got no idea how to access the server -_-
05:22.01ChannelZI think you have to use Bridging mode in Virtual Box to make a fake interface on the host machine that you can use to connect to the virtual one
05:24.38jcarter1you are right
05:29.36ChannelZNAT mode (which is is probably default) is only really useful for outgoing traffic out of the virtual machine
05:30.34jcarter1#vbox helped me
05:30.36jcarter1it works now
05:30.44jcarter1i had to change to bridging mode as you said
05:31.45pentanolhey, anybody use auth over radius?
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07:50.24Whtsuphello
07:50.26Whtsupi want to ask
07:50.41Whtsuphow can i add mutiple ip address of host in sip.conf
07:52.02tzafrir_laptopWhtsup, no
07:52.12tzafrir_laptopWhy do you need that?
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08:07.09Whtsuphm
08:07.18Whtsupfor a2billing
08:07.30Whtsupmy customer told me add multiple ip
08:07.33Whtsuphow can i add
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08:10.30carrarSUM
08:10.31carrar+
08:10.41Whtsuphmmmmmm
08:10.45carrarA + C = C
08:10.52Whtsup?
08:10.57carrarA + C = B
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08:26.02tzafrir_laptopWhtsup, sure. an IP address is merely a 32bit number.
08:27.22tzafrir_laptopWhtsup, seriously, though, what do you want to do with those multiple IP addresses?
08:27.41tzafrir_laptopUse a random one?
08:27.54tzafrir_laptopUse one as a default and another one as a backup?
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08:56.19Godfather_o/
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11:03.02scurbAnyone know if MixMonitor command is notified in the manager API events?
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12:12.47hyphenexWhat's a good VoIP phone I should get? :P
12:13.11benngard~phones
12:13.11infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else.  Do not consider Grandstream phones.  Ever.
12:13.47hyphenexThanks
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12:41.49couponjuananyone want to work on a project together?  I do html css php mysql and the flash/photoshop.   I also have a little bit of a vision too of what I need this for.
12:42.20aceking5uh wrong channel?
12:42.23couponjuanno
12:42.30couponjuandynamic .call file
12:42.50couponjuancan be created and put in the outbound directory right?
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13:01.57hyphenexhey, what do you think of this one? http://www.ht.com.au/part/V2913-Cisco-Small-Business-Pro-SPA942-4-line-IP-Phone-with-2-port-Switch-VoIP-phone-SIP-v2/detail.hts
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13:30.41puzzledhyphenex: imo cisco sip firmware sucks, you even need to pay to get new firmware and now that linksys == cisco it's probably only time before they become just as bad
13:31.07puzzledget a polycom, aastra or snom
13:40.46jmcdowellPolycom has outstanding quality, but their configs are a litte hard to figure out at first due to 1rst grade documentation.  Once you get it, it's easy..
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13:43.07jmcdowellhas installed CentOS 5 and asterisk 1.6 and is currently working out the kinks.
13:43.20jmcdowellstill can't dial
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13:50.27tzafrir_laptopcouponjuan, if you ask for "a dynamic call file", you probably want to use "Originate" from the manager interface
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14:14.49krefikanyone know is asterisk-gui can substitute administrative tools like freepbx?
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14:18.13mike3I just setup Asterisknow. I'm having some issues where the voice prompts are a little glitchy sounding. How can I fix this?
14:19.51benngardMessage: Thanks for all the fish <- fun logoff message from ami :)
14:20.40n3glvqos?
14:21.14mike3I'm calling on the internal network
14:21.19mike3not much activity at al
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15:28.22niekvlesserthello, i'm trying to get chan_misdn working
15:28.34niekvlessertanyone with experience on debian etch with that?
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15:41.17puzzledniekvlessert: nope but if you are not too hung up on debian, just install centos 5.4 on the box/vm and follow http://www.asterisk.org/downloads/yum to get everything including mISDN
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15:53.59IgramulHi, is it possible to get the dialed number in a h-extension? (e.g. to write to a fax report the number the fax was sent to)
15:57.33n3glvyes
15:57.38n3glvif you had not left
15:57.39n3glv;->
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16:20.18mike3I just setup Asterisknow. I'm having some issues with audio. It seems to be glitchy. Any input?
16:24.10niekvlesserttype of phone/connection/codecs plz
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16:27.57danj1980Hi. I just had 7 hours of downtime on asterisk because the server power supply failed and needed replacing. I've been googling "asterisk redundancy" and have found loads of questions, but not many answered.
16:28.41danj1980Has anyone see a whitepaper on creating a more stable environment?
16:28.47n3glvdanj1980: I think you want high-availability and that's a hard row to hoe
16:29.12n3glvdual  powersupply and raid1 or better is easy
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16:29.19n3glvmirrored servers etc is a mess
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16:29.58danj1980The polycom phones we use can register to two servers, so that makes my job a bit easier.
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16:30.47danj1980Is it then simply a case of setting up a second server, syncing the config files and mysql?
16:31.43danj1980However, internal calls get messy because if extension 101 is on server A, and extension 102 has decided to choose server B, then they cant find each other.
16:33.21danj1980Alternatively, we can put Openser in front of both servers, but then if openser went down, it would be the same problem.
16:34.01*** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl)
16:35.01danj1980By the way, its a purely VOIP based platform. No analog or ISDN lines.
16:37.46*** join/#asterisk DrCron (rszasz@saxonco.com)
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17:01.26mike3niekvlessert, sorry
17:01.29mike3um.. let me look
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17:46.56ManxPower-work~answers
17:46.57infobothmm... answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
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18:18.28*** join/#asterisk Artur__ (~chatzilla@212.59.229.155)
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18:19.13Artur__hi everyone and sorry for my poor english
18:19.57Artur__I would like to set my basic dialplan but i have some problem with it . Can anyone help me?
18:21.11fenruswhat do you want your dialplan to do ?
18:21.17*** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net)
18:21.20fenrushave you read any examples?
18:21.42fenrusyou need to be a bit more specific
18:21.57Artur__yes i read and some steps from book of asterisk but i stil have litle problem whit it
18:22.15Artur__ok can you wait i put hire some of my config
18:24.09Artur__[ipfon]
18:24.10Artur__exten => 327218816,1,Dial(SIP/artur&SIP/test)
18:24.12Artur__[sipdiscount]
18:24.14Artur__exten => 0048327218816,1,Dial (SIP/artur&SIP/test)
18:24.15Artur__[incoming]
18:24.17Artur__exten => 327218816,1,Dial(SIP/artur,20)
18:24.18Artur__exten => 327218816,1,Answer()
18:24.20Artur__exten => 327218816,n,Background(main-menu)
18:24.21Artur__[test]
18:24.23Artur__exten => 101,1,Dial(SIP/artur,20)
18:24.24Artur__exten => 101,n,Playback(boston)
18:24.26Artur__;exten => 102,1,Dial(SIP/test,20,m)
18:24.28Artur__;exten => 102,n,Playback(boston)
18:24.29Artur__exten => 102,1,Answer()
18:24.30Artur__exten => 102,n,Background(main-menu)
18:24.32Artur__exten => 102,n,WaitExten()
18:24.32*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
18:24.33neurosysuh oh
18:24.33Artur__exten => 1,1,Playback(digits/1)
18:24.35Artur__exten => 2,1,Playback(digits/2)
18:24.37Artur__;exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdiscount,10)
18:24.37*** kick/#asterisk [Artur__!~chatzilla@64.235.218.194] by [TK]D-Fender (Artur__)
18:24.55x86lol
18:25.00trelaneoops
18:25.04*** join/#asterisk Artur__ (~chatzilla@212.59.229.155)
18:25.04x86[TK]D-Fender++
18:25.13trelaneArtur__, www.pastebin.com
18:25.24[TK]D-FenderArtur__: Do NOT flood in here again
18:25.26x86pastebin.ca > pastebin.com
18:25.27[TK]D-Fender~pb
18:25.28infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
18:25.29[TK]D-Fender^^^
18:25.52x86pastebinit? ooo
18:26.30Artur__i past config to one of site
18:26.46[TK]D-FenderArtur__: WHERE'S THE LINK TO IT?
18:26.57[TK]D-FenderArtur__: Are we supposed to GUESS?
18:27.35Artur__http://paste.lisp.org/+219C.
18:27.36trelane[TK]D-Fender, of course!
18:27.45Artur__this is my exten conf
18:27.52Artur__extension.conf
18:28.18Artur__i would like to set incoing and outgoing first incoming for ipfon
18:28.31neurosyswell his [incoming] context has 2 "1" priorities...
18:28.38Artur__i can change test to incoming
18:28.38smooth_penguintrelane !
18:28.45trelanesmooth_penguin, I deny everything :)
18:28.53smooth_penguinheh ;p
18:29.13Artur__i know I'm noob with asterisk
18:29.25Artur__ok i told what i need
18:29.55smooth_penguinArtur__, well its basic IRC ethics
18:30.00smooth_penguinuse Pastebin
18:30.15Artur__i want to set exten to transfer incoming call to my mobile phone when i not answer the local phone number afret 20 sec
18:30.53Artur__pestbin is basic asterisk  Qest..?
18:30.58Artur__channel
18:31.14Artur__?
18:31.53Artur__sorry
18:32.22Artur__paste.lips is worst then pastebin?
18:32.43Artur__i paste my config to paste.lips and put url
18:34.13smooth_penguinyah
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18:35.11Artur__i know there is litle mess some ext I coment becouse i test this
18:35.22*** join/#asterisk rue_house (~rue@24.207.119.38)
18:35.33rue_houseis the D channel part of a T1's 24 channels?
18:36.00x86only on a T1/PRI
18:36.24rue_houseyou remember how many channels are in an E1?
18:36.29rue_housenot 32 is it?
18:36.30x86a T1/CAS has (24) 64kbps channels, none of them B or D channels, but you can think of them all as B channels
18:36.38Artur__anyone looking for my extension.conf?
18:36.40x8632 channels E1 yes
18:36.49rue_househuh
18:37.03x8630B 2D usually on an E1 PRI, iirc
18:37.20rue_housethe ulaw converter chips use a 2.048Mhz clock rate and recieve 8 bits for every 256 clock cycles = 32 channels
18:37.39[TK]D-Fenderrue_house: thats E1, not T1
18:37.44rue_houseyea
18:37.46Artur__can anyone help me in private chat?
18:37.57rue_housethis is just the audio codec chip
18:38.33rue_houseits REALLY easy to make a T1->analog channelbank
18:38.35rue_houseyou just need to do frame clocking
18:38.53rue_housefeed the chips the raw t1 stream otherwise
18:39.11rue_house(chips = MC14LC5480)
18:39.17smooth_penguinArtur__, whats currently not working?
18:40.05Artur__if someone cal to me and i no't answer call should transfer to my mobile phone
18:40.15Artur__but nothing hepen
18:40.18rue_houseswear its almost too easy just with discrete logic chips
18:40.31rue_houseso why are they so expensive...
18:41.03niekvlesserthey guys, it seems there's a xorcom sources.list
18:41.09niekvlessertcan anyone provide it?
18:41.21tzafrir_laptopI guess I can.
18:41.32tzafrir_laptopBut what do you mean by "xorcom"
18:41.45coppicerue_house: you have identified your route to unknown riches
18:42.05rue_houseits a 15 year old path!
18:42.17niekvlessertwell i heard there's a way to install a debian and then add some xorcom stuff to your sources.list and then install asterisk with all the crazy isdn stuff in 1 run
18:42.18rue_housethis stuff has been around forever...
18:42.31coppiceruw_house: so you are rich?
18:42.40rue_houseno
18:42.46Artur__smooth i don't understend how to write right exten to make it what should dialplan d when i don't answer after about 10 to 20 sec
18:43.02rue_housethese old chips sure make telephony interface easy
18:43.11tzafrir_laptopniekvlessert, stuff under http://updates.xorcom.com/rapid and http://updates.xorcom.com/pkg-voip is basically built off the Debian pkg-voip repo
18:43.29rue_houseI want to make a chan_rs232, Im pretty sure I could have an avr bridge it to the codec chip
18:43.40tzafrir_laptopthe pkg-voip one has quite untested packages. The /rapid one has only signed and tested packages
18:44.08rue_housedosn't mean I will
18:44.28tzafrir_laptopniekvlessert, Originally there was bristuff in Debian. nowadays (1.6.2) it's gone
18:44.53rue_houseany easy falls apart trying to do voip (cause of echo cancelation)
18:44.59niekvlesserttzarfrir_laptop: ok, nice, so this is 1.6?
18:45.04tzafrir_laptopThere's only a minor "bri" patch (to replace the "sux" message with something marginally more useful)
18:45.36niekvlessertyes it is
18:45.40tzafrir_laptopniekvlessert, yes, 1.6.2
18:45.51niekvlesserthmmmm
18:46.09Artur__smooth can you help me ?
18:47.05rue_housewhere would I start to write a rs232 channel driver?
18:47.57rue_houseulaw only, raw to serial port
18:48.20niekvlessertwhy is it so hard to get chan_misn running?!?
18:48.25niekvlessertchan_misdn
18:48.38tzafrir_laptopmisdn? it seems quite different
18:48.49tzafrir_laptoprue_house, how will the signalling be handled?
18:49.03smooth_penguinArtur__, Im not very sure
18:49.04rue_houseit wont! :)
18:49.12tzafrir_laptopHow do you dial? hang up?
18:49.14rue_houseyour always listening and alwasy transmisting
18:49.22rue_housecall it rev 1 :)
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18:49.34rue_houseinband dtmf
18:49.36rue_house:)
18:49.38Artur__smooth heh
18:49.44tzafrir_laptoprue_house, then look at the "console" channels, I guess
18:49.50rue_houseok
18:49.52niekvlesserttzafrir_laptop: trying to get it running with 1.4.29, i can compile misdn but asterisk won't find it so i can't compile chan_misdn
18:50.23tzafrir_laptopniekvlessert, what is "it"?
18:50.51niekvlesserttzafrir_laptop: not sure yes, suppserv for sure though
18:50.53*** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net)
18:50.56niekvlessertalthough i did compile it
18:50.59rue_housekirby dance!?
18:51.01rue_housesigh...
18:51.27Artur__:(
18:51.41niekvlessertit can't find misdn either!
18:51.49smooth_penguinArtur__, well look up on the voip-info website
18:52.11rue_housetzafrir_laptop, would you say the console is the must cutdown driver there is?
18:52.14niekvlesserti change menuselect-tree to see what it can & can't find
18:52.37niekvlessertso now suppserv & misdn is a problem
18:52.40tzafrir_laptoprue_house, in-band signalling is not nice
18:52.41niekvlessertnice
18:52.45smooth_penguinArtur__, http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
18:52.51tzafrir_laptoprue_house, are you sure you want to go there?
18:53.13rue_houseI'm sure I can ignore that bit for hte process of having fun
18:53.37Artur__thx i read this now
18:53.42rue_houseI had this interesting idea recently, creating parity errors on bytes to do serial sync
18:53.54rue_houseaka, bytes with a parity error are control
18:53.57rue_house:)
18:54.35rue_houseI could make every 1001'st byte control
18:54.54tzafrir_laptoprue_house, what speed is your serial channel?
18:55.18rue_housewell, 8Khz 8bit, 64Kbits... so thats.... higher than 2400...
18:55.47tzafrir_laptopif it's fast enough (115200) you can use some "extra bandwidth" for signalling
18:55.49rue_house4800...9600...  19200.... 38400...
18:56.00tzafrir_laptop(hmm... sounds like TDM)
18:56.16rue_housethe chip takes in ulaw
18:56.41rue_househmm have to be 115K too
18:57.04tzafrir_laptop1 B channel and 1 D channel :-)
18:57.20rue_houseor slightly less
18:57.32rue_house1/2 rate D channel
18:59.38rue_houseactaully, this is alot like th problem on the hexapod I'm building, muxing control data
19:00.42rue_houseconsole uses the sound card right? its not audioless?
19:01.31rue_houseARNING, "Failed to initialize audio system - (%d) %s\n"
19:01.33rue_houseah good good
19:09.15*** join/#asterisk cuco (~Diego@95.35.58.91)
19:09.47fibresCan anyone help me? I am having authentication issues with incoming calls on Ast 1.6 I tested the config im using on 1.2 and works ok on 1.6 authentication fails
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19:24.14niekvlessert./configure can't find mISDN, checked install.log, /usr/bin/ld: cannot find -lmISDN
19:24.21niekvlessertwhat does that mean?
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19:38.08*** join/#asterisk Bullterd (~IceChat7@62.249.223.152)
19:38.11BullterdEvening All :)
19:38.21BullterdCan anyone advise me on what software to use to setup an outbound proxy?
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19:53.22niekvlessertdefine proxy
20:01.03*** join/#asterisk Da-Geek (~Da-Geek@gateway.rhev.dageek.co.uk)
20:01.12ManxPower-workBullterd: A proxy or a B2BUA?
20:02.16BullterdManxPower-work: A whatnow?
20:02.50ManxPower-workBullterd: OpenSER is a SIP Proxy.  Asterisk is a SIP Back2BackUserAgent
20:02.50*** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl)
20:03.02BullterdAh.
20:03.20ManxPower-workWhich means Asterisk basically treats both legs of the calls as different calls and so does not pass SIP stuff directly between the two ends of the call.
20:03.37BullterdManxPower-work: All I know is this solved a big problem for me http://free.sipout.com/
20:03.48BullterdManxPower-work: its a free service, I dont want to rely on that for buisness, so want to set my own up
20:04.37ManxPower-workBullterd: Asterisk is not suitable for that application.
20:04.38[TK]D-FenderBullterd: "a big problem"?  You haven't describe what problem you even HAVE.
20:04.54p3nguinbullterd: Take a look at OpenSIPS and see if you think it can help you.
20:04.58ManxPower-work[TK]D-Fender: ISPs blocking 5060/UDP
20:05.12BullterdManxPower-work: No, Just a general pain with my customers and NAT
20:06.11ManxPower-workBullterd: Then your question makes no sense.
20:06.14BullterdI wasnt specifically looking for an asterisk related solution, Just a general linux one. I guessed that since it was SIP related thing, here'd be the best place to ask
20:06.21ManxPower-workVirtually all VoIP PBXs support NA
20:06.31ManxPower-workT'd clients without problem
20:06.48BullterdThanks for your help
20:08.07niekvlesserthttp://pastebin.com/d44a3cdb
20:08.19niekvlessertmisdn trouble still going on
20:08.29niekvlessertinstalled 3rd party stuff from svn.digium.com
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21:16.02*** join/#asterisk doug (doug@breakout.telerama.com)
21:16.21doughulo!  anyone have a nice way of getting text entry from a phonepad?
21:16.28dougor does that already exist in asterisk?
21:16.41dougi want to capture the text name of my destinations...
21:16.55dougin my system, if i'm calling target, it'll say "I don't know this destination, please say the name"
21:16.57*** join/#asterisk russellb_ (~russellb_@asterisk/digium-open-source-team-lead/russellb)
21:16.57*** mode/#asterisk [+o russellb_] by ChanServ
21:17.01dougthen it'll record me saying "TARGET"
21:17.12dougafterwards, i want it to say "spell the name of the destination you just spoke"
21:17.17dougand have it take a spelling via dialpad
21:17.42dougdunno what mechanism, maybe 21 for A, 22 for B, 23 for C, 31 for D etc
21:18.09dougbut somehow have it capture the spelling in a machine-readable way
21:18.15dougso it can be automatically added to the directory
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21:35.03ManxPower-workdoug: "core show applications"
21:44.57dougsure; know which one might do me?
21:45.25dougor if there's an app that's not normally compiled in...?
21:46.55carrarSo you are asking for speech recognition
21:47.14carrarlook at lumenvox
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21:56.41doughm, not speech recognition
21:56.44dougthis would be from the keypad
21:57.26dougso i'd say the name for the recording, and punch in some numbers for the spelling.
21:59.43ManxPower-workin 1.6.mumble there should be an application that allows 222=C type of entry
22:00.16dougknow the name?
22:00.25doug<PROTECTED>
22:00.42dougif not the name, something i can feed into google to get close...
22:02.05ManxPower-workdoug: I guess I could go track down a 1.6 box and find out for you, but I'm not going to.  That sort of stuff is usually left as an exercise to the reader.
22:03.29carrarYou mean people can help themselves?
22:03.33carrarThats just crazy talk!
22:04.41dougi got one, and i just went through the apps
22:04.46dougwhich is why i ask.
22:06.49*** join/#asterisk Godfather_ (~Godfather@237.Red-88-8-10.dynamicIP.rima-tde.net)
22:08.17dougalthough now i'm wondering if the sms app is also in 1.4.28
22:09.04*** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr)
22:09.14douglooks like it.
22:09.32dougno "alphabet mode" dtmf app tho, or anything close to it.
22:10.18doug"Works to ETSI ES 201 912 compatible with BT SMS PSTN service in UK."  i wonder if that implies no us easy smsing.
22:11.02ManxPower-workThe SMS app only works with European carriers.
22:11.51dougi'd give my left nut to send/receive sms via asterisk to us phones.
22:12.08*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
22:13.05dougcurse the yanks and their monopolies.
22:13.16dougor duopoloies, i guess.
22:13.23doug-o
22:13.33ManxPower-workI was wrong.  http://www.voip-info.org/wiki/view/Asterisk+cmd+DTMFToText
22:13.50*** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl)
22:14.48carrardoug, you could setup a SMS gateway
22:15.16ManxPower-workcarrar: Somehow I think that's a tad complex for his needs.
22:15.23carrarheh
22:15.50ManxPower-workHe would either always need Asterisk at both ends of the call or he would have to import SMS analog sets from Europe.
22:16.17dougsms gateway; is that something with an ss7 interface?
22:16.21ManxPower-workAsterisk's SMS uses MODEM TONES to transport the text.
22:16.32carrarToday youcan however send SMS Mesasges via Email to any phone via asterisk though
22:16.40ManxPower-worki.e PSTN SMS, not cellular SMS
22:17.08dougmodem tones?  that's some kind of asterisk-only texting hack?
22:17.35ManxPower-workdoug: not at all, it's the SMS standard for SMS over PSTN
22:17.58dougwow, that's twisted.  is that the meat of the ETSI ES 201 912 protocol?
22:18.08ManxPower-workI'm sure you've never heard of it because nobody uses it in the USA.
22:18.20ManxPower-workdoug: I'd have to look it up, but that sounds correct.
22:18.34dougpstn sms is common in eu?
22:18.40ManxPower-workdoug: yes.
22:19.03doughow does one receive texts?
22:19.16ManxPower-workMost european carriers seem to have PSTN SMS <=> Cellular SMS gateways set up.
22:19.18dougi'm not so good at recognizing modem tones...
22:19.41carrardoug, the easyest thing to do is buy one of those GMS Modems kits, buy a SIM card and conenct it via Serial to your asterisk box and send SMS commands to your GSM Modem which then can send your SMS message to the SMSC
22:19.47ManxPower-workdoug: you receive texts by receiving a call with an FSK SMS text burst to a PSTN SMS terminal
22:20.30ManxPower-workI wish they would rename the app to something like EUSMS
22:20.40ManxPower-workor SMSBUTNOTINUSA
22:20.56dougoh, needs a terminal.
22:21.06dougthanks guys, be back later.
22:21.15carrarhttp://www.sparkfun.com/commerce/product_info.php?products_id=478
22:21.25dougdtmftotext, exactly what i needed (i think).
22:21.26dougthanks manx
22:25.09*** join/#asterisk ttwhy (~tekkno@p4FECEEC4.dip.t-dialin.net)
22:35.18*** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl)
22:46.42*** join/#asterisk Reemo (~Reemo@p5B3EED01.dip.t-dialin.net)
22:50.42*** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
23:02.03*** join/#asterisk pawz (~pawz@ppp118-208-172-238.lns20.bne4.internode.on.net)
23:02.21*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
23:07.21*** join/#asterisk ruied (~ruied@188.140.25.148)
23:16.26*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
23:16.38*** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
23:28.50*** join/#asterisk a_bug (~nothingne@118.94.17.27)
23:32.27*** join/#asterisk Rix (~Rixi``@77-85-21-9.btc-net.bg)
23:33.00*** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker)
23:44.15*** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net)
23:56.59*** join/#asterisk pentanol (~Unknown@91.195.60.231)
23:57.22stallionebroke my working asterisk install trying to upgrade freepbx.. :(
23:57.37stallioneAsterisk doesn't start anymore

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