IRC log for #asterisk on 20090106

00:12.14SlicerDicerPSTN with SPA3102, I cannot get my callwaiting to work.. I call my PSTN, then have another call on PSTN I cannot flash to it? Anybody know why this would be? I have tried the double hook flash and it does not seem to work either?
00:12.26*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
00:15.27[TK]D-FenderSlicerDicer: might be no way of doing this
00:19.21SlicerDicerwhat would I need to make it work?
00:20.49[TK]D-FenderSlicerDicer: Not good at understanding "no" I see
00:21.10SlicerDicer"might" was my hope ;-)
00:24.16*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
00:27.47*** join/#asterisk lanning (n=lanning@66.151.128.195)
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00:43.46drmessano^Ok, everyone, I have some bad news
00:43.52drmessano^They are recalling BACON
00:43.56drmessano^DONT PANIC
00:43.59drmessano^Wait
00:44.02drmessano^EVERYBODY PANIC
00:44.52denondrmessano^: bacon? http://twitpic.com/zyoo
00:46.32drmessano^The Patrick Cudahy firm, of Cudahy, Wisconsin is recalling more than 3,500 pounds of bacon bit products that may be contaiminated with Listeria monocytogenes.
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00:46.45drmessano^They're recalling teh bacons
00:48.21jaytee3500 lbs? that's nothing
00:48.35drmessano^3500 of bacon BITS
00:48.42drmessano^Thats like.. a weeks supply
00:49.02*** join/#asterisk Borai (i=Bora@S0106001c109e98db.no.shawcable.net)
00:49.03BoraiHi
00:49.03jayteeI don't use bacon bits, I use fresh cooked bacon
00:49.05drmessano^I wonder what that is in uncooked bacon
00:49.23drmessano^What the hell am I supposed to put on my ice cream?
00:49.25jayteeyeah, the shrinkage factor must be pretty high
00:49.33jayteechocolate jimmies?
00:49.42drmessano^Do they have bacon in them?  NO
00:49.43drmessano^FAIL
00:49.50jayteelol
00:50.37jayteedrmessano^, did you see the question above from SlicerDicer about using a 3102 with call waiting?
00:50.57drmessano^Im eating a hot dog right now.. and as much as I love hot dogs, it feels like dating the hot chick at schools "sorta hot from the side" sister
00:51.14jayteehaaahhaaaaaaa
00:51.22drmessano^jaytee: Trying to flash the call waiting on the PSTN?
00:51.29jayteeyeah
00:52.02drmessano^I cant be bothered with such shenanigans
00:52.13drmessano^Who the hell uses Call waiting on a PBX
00:52.17drmessano^Seriously..
00:52.36denondrmessano^: actually, Ive used SendFlash to do that beforfe
00:52.49denonnot for call waiting .. to remote transfer to get calls off pstn trunks
00:53.39drmessano^"What if my cat gets out and needs to call home.  Mr Pimpernoodle will be stranded because I cant answer my beeps"
00:53.45drmessano^Answer: Buy him a cell phone
00:55.35Borai1.6.0.3-rc1 or 1.4.22 ?
00:55.55[TK]D-FenderBorai: YES
00:55.55drmessano^Cheese or wine?
00:56.02drmessano^[TK]D-Fender: WIN
00:56.10jaytee[TK]D-Fender, heheehee
00:56.10[TK]D-Fender\o/
00:56.18drmessano^"Would you like the steak or the fish?"  "Yes"
00:56.34drmessano^Someone had to
00:56.37Boraiwhich one should I upgrade to?
00:56.55[TK]D-FenderBorai: From?
00:57.02drmessano^Windows 98
00:57.07jayteeBorai, I suppose that all depends on whether your a bottom or a top. If your a bottom I'd go with 1.6
00:57.29Borai1.4.20.1
00:57.46drmessano^jaytee: Fail.  Bottoms use "trunk"
00:57.48Boraiwell I want to start from the beggining this time I dont want to make the mistakes I did in the past
00:58.08drmessano^Borai: Bear us your soul
00:58.11jayteedrmessano^, I completely forgot. yep, that would be the "max pain" avenue
00:58.32[TK]D-FenderBorai: 1.6.0.3-rc1 then, becuase you clearly can't upgrade to soemthing with FEWER decimal version positions, can't you?
00:58.37drmessano^I think Borat wants to reveal his innermost child to us
00:59.25drmessano^Go with the Really Cool (RC) one.. you know you want to.. All the other kids are doing it..
00:59.25jayteeI couldn't get 1.4.22 to compile with zaptel instead of dahdi. thought it was still an option, dropped down to 1.4.21 instead
01:00.02drmessano^I'm holding out for 1.6.2
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01:00.47drmessano^AKA the "GTFO Dahdi, you've mixed me up long enough" release
01:00.53jayteethink they'll finally have SIP TCP working right by 1.8.0?
01:00.55NovceGuruI wonder if asterisk will compile/run on a RMI 1250 RISC Processor
01:01.34drmessano^jaytee: No, but i'm sure something useless like SIP IPX will be implemented
01:01.41jayteesome guy was trying to get it running on an ARM processor a few days ago with no success
01:02.02NovceGuruSomebody has ported it to the gumstix which is the intel pxa270 arm
01:02.07jayteehahaha, IPX, wow that brings back memories...........bad ones
01:02.14drmessano^SIPBeui
01:02.17drmessano^There u go
01:02.20jayteehahahaaha
01:02.37NovceGuruI'm using a http://www.chippc.com/thin-clients/linux/?p=linux somewhere where I could use the audio out for a PA system
01:02.43jayteeI'm sorry but you can't call that person, they're on another segment!
01:03.19drmessano^You know, that is the ultimate in network security
01:03.26drmessano^Who the hell would see you running NetBeui
01:03.44jayteeSIP SNA/LU6.2 over Token Ring
01:04.03drmessano^grabs some frozen yellow hose and small mallet
01:04.29drmessano^Speaking of "What new again"
01:04.35drmessano^I saw the coolest thing today
01:04.56drmessano^Netgear has an Ethernet over Coax adapter for using your home CATV cable for networking
01:05.19drmessano^Has an in/out to so you can split your TV connection off it, and the RJ45 for the ethernet
01:05.30drmessano^270Mbps
01:05.41NovceGuruI'm waiting for hdmi/component baluns over a single coax
01:05.47NovceGuruTHAT will be the day
01:06.22drmessano^I was thinking for a 300 foot run at 270Mbps.. You can bury some RG-6 between two buildings
01:06.29*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)
01:06.34drmessano^Beats the crap out of fiber
01:06.44*** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
01:07.24NovceGurudirect burial cat5 isn't THAT bad
01:07.42drmessano^Yuck
01:07.43NovceGuru~$120 for 1000ft
01:07.45drmessano^Its still cat5
01:07.50NovceGurugigabit!
01:07.54drmessano^So?
01:08.00NovceGuruyeah
01:08.19drmessano^I would want something less fragile buried
01:08.27drmessano^Unless they encase it in a garden hose
01:08.28jayteeunderground CAT5 for 120 bucks a 1000'? where you shopping? WalMart?
01:08.43NovceGurumonoprice, it's probably junk
01:08.49drmessano^Standard Cat5 is 90 a roll
01:08.50NovceGurulowes has some with a pretty thick jacket
01:08.56drmessano^lol
01:09.03jayteeI can get standard for 79 bucks a 1000
01:09.05drmessano^"Pretty thick" does not make direct burial
01:09.06NovceGuruI get it for 65 at the electric store
01:09.11drmessano^and I dont buy cable at LOWES
01:09.14drmessano^Or Home Depot
01:09.18drmessano^Or Rickels Lumber
01:09.34drmessano^NovceGuru: Not Belden
01:09.34NovceGuruI do when I can't wait for shipping
01:09.38jayteenope, neoprene jacket with foil shield for underground and that gets pricey. like 180 a 1000
01:09.39NovceGuruthey have the WORST cat5
01:10.08drmessano^Belden is worse than Lowes brand?
01:10.19drmessano^Lay off the crack
01:10.27NovceGurunever used belden
01:10.47NovceGurufortunately, it sounds like :)
01:11.11NovceGuruI did get some krap that was pre lubed and didn't have the nylon tension cord thingy in it
01:11.15drmessano^If you buy a roll of Cat5 for $65 at todays prices, you can't possibly be getting decent cable
01:11.31jayteeI like Belden myself, I've never had a problem with it
01:11.37NovceGurucopper has came down, btw
01:11.40coppiceBelden have been very good at getting crazy high prices out of defence customers :-)
01:11.41drmessano^Belden is pretty much the best
01:11.46drmessano^Not that much it hasnt
01:11.54Boraiok so I installed dahdi-linux-2.1.0 as well as dahdi-tools now configuring and installing (Really Cool)-1
01:12.10*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5446245918ccf1d7)
01:13.48NovceGuruhttp://www.kitcometals.com/charts/copper_historical_large.html#6months
01:13.50drmessano^Armored Cat5 for the win
01:14.13*** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net)
01:14.18NovceGuruhttp://www.pcconnection.com/IPA/Shop/Product/Detail.htm?sku=9136254&oext=1038A&ci_src=14110944&ci_sku=9136254 YES
01:14.23NovceGuru$800! cheap!
01:16.28jaytee80 cents a foot isn't cheap
01:17.15coppiceit is for shoes
01:17.16drmessano^Cat5 will never be desirable for outdoor use unless its fully dipped
01:17.25jayteelol
01:17.51drmessano^Thats why RG6 is a great idea
01:18.01jayteedrmessano^, that was gel filled, foil shielded and polyethylene outside jacket.
01:20.07jayteehere ya go, 139 bucks for gel filled outdoor, just as good as that overpriced BlackBox cable at a way better price: http://www.jack2rack.com/index.php/cPath/97_73_197?gclid=CJz77Yrj-JcCFQ89awodtindEA
01:20.11coppice2 layers are stainless steel armouring, and shark repellant
01:20.42jayteeyeah, those land sharks swimming underground love to bite into that cable
01:20.48drmessano^Id be sadistic enough to use RG11
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01:21.12jayteemoles, voles and friggin gophers
01:21.18drmessano^HA
01:21.22drmessano^Black box cable
01:21.41jayteethat was the 800 bucks for a 1000 feet link he posted
01:22.11drmessano^It could be a run of Belden that didnt make the grade, or it could be a run of cable Kelloggs made in one of their offshore subsidiaries.. who knows
01:22.22drmessano^I secretly suspect Callweaver is owned by Cisco
01:22.26drmessano^But thats just me
01:22.27coppiceblack box are good for finding unusual things, but bad by any other measure
01:23.34*** join/#asterisk sosoriri (n=chatzill@222.47.180.130)
01:24.14sosoririhello, anybody have seen the soudn delay problem?
01:25.15jaytee[TK]D-Fender, your friend is back :-)
01:25.46sosoririfor example, when one user join in the conference, he must hear "Please enter your conference number followed by the pound key.", but he heard"...your conference number followed by the pound key."
01:26.13sosoriribut i can't catch it again.
01:27.21sosoririit occurs one or two times in a month or not...
01:28.24drmessano^Its moon phases
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01:28.33jayteegravity waves
01:28.42[TK]D-Fenderdrmessano^: Sheer lunacy I say...
01:28.57jayteegroans
01:29.38drmessano^That was enlightening
01:33.35Boraiok
01:34.10mchouanyone have a spa-3000 or 3102?
01:34.22Boraisuccess
01:34.32Boraiclean insall of 1.6.0.3-rc1
01:35.20*** join/#asterisk DigitalIrony (n=eric@nat/digium/x-75ef67f885360914)
01:35.27Borainow i got a polycom IP 550 unpacked sitting next to me
01:36.11jayteenice phones
01:36.14jayteeI have a few
01:36.44Boraiis there a guide on how to configure this sweet phone?
01:37.46jayteethere's a section in the book on how to setup FTP provisioning and if you use that, the Polycom whitepaper on provisioning and the SIP Admin guide for your version of the SIP firmware you should be able to do it. I did.
01:38.38jayteeor you could take the lame shortcut route and use the web interface
01:38.55Boraito access those guides
01:39.01Boraiyou have to have a polycom account right?
01:39.05jayteenope
01:39.10BoraiI couldnt register at their site their server has an error
01:39.12Boraiok
01:39.15Boraithe book I have that
01:40.05jayteehttp://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip550.html
01:40.51NovceGurutheir config files are crazy
01:41.24jayteeon that page you need to download the SIP Admin guide for YOUR phone's current version of the SIP firmware and under the section titled Other Technical Documents get the White Paper titled Configuration File Management on Soundpoint IP Phones.
01:41.25Sargun_screenjaytee: I have one of those, I like it.
01:41.51jayteeNovceGuru, with great power comes great responsibility
01:41.55Boraiok dont you need to download the firmware?
01:42.08NovceGurujaytee: yeah
01:42.34NovceGuruI'm getting an IP670 in next week, color screen!(woopdeedoo?!?)
01:42.45Boraifirmware is bootrom: in the settings of the phone right?
01:42.47jayteeyes, you can download the same version you have plus the bootrom files for what's on there now or you can use a new version. I prefer to start with whatever's on the phone or a minor version or two higher.
01:43.41jayteebootrom boots the phone. Having a copy on the FTP server lets the phone boot from it and search there for it's configs to download. read the whitepaper, it'll all come clear to you.
01:44.11bkw_has two 670's
01:44.20bkw_two 650's, one 550, and two 330's
01:44.38jayteeI have 53 330's and 6 550's.
01:44.47bkw_the 670 isn't that grand of an upgrade
01:45.05jayteebut it's got that larger screen and HD
01:45.15bkw_so does the 650 and 550
01:45.25NovceGuruI think the 650 and 550 are the same except for color screen and gige?
01:45.33bkw_right
01:45.41bkw_I'm also getting a Cisco 7975G
01:45.47jayteeyeah, 650 has color and gig
01:45.49NovceGuruI dont understand the oogling over gige....as is networks wont be backward compatable for the next 10 years?
01:45.52NovceGuruas if*
01:46.00bkw_NovceGuru: never know
01:46.20NovceGuruand let me know when sip needs > 100mbit !!
01:46.24NovceGuruheh :P
01:46.47bkw_it does if you do more than 1200 calls with ulaw
01:46.50jayteefor when your phone also has HDTV videoconferencing built in I imagine
01:46.54bkw_NovceGuru: its not for the phone
01:46.58bkw_its for the PC sitting on the otherside
01:47.02bkw_on the switch port s
01:47.08NovceGuruno it's obviously clear
01:47.18NovceGurunow*
01:47.37jayteegigabit on the backbone or on a segment's one thing, nowadays on a station or phone uplink to the switch it's a waste of bandwidth
01:47.38bkw_I can't wait till the 550, 650 and 670 support G722.1 and G722.1C
01:48.10jayteewhy, cuz you wrote it?
01:48.21bkw_No I wrote the module for FreeSWITCH :P
01:48.33bkw_but they'll have it sometime in 09
01:48.38bkw_for the other phones i'm told
01:48.49bkw_lower bandwidth usage
01:48.58jayteewell, last time I looked at my calendar it was ...... hmmmm, 09!
01:49.04bkw_try Q3 09
01:49.30jayteeI'm patient. I can wait till Q3.......of 2011 or 2012
01:49.46NovceGuruI never did follow asterisk and it's g722 issues
01:49.53BBHossbkw_: how is g722 different from .1?
01:49.58NovceGurueven with ulaw the polycoms sounds great
01:50.08bkw_G722.1 can run better on non-dsp hardware ie PC's
01:50.11jayteeBBHoss, a period and an extra number?
01:50.16NovceGuru2-3x bandwidth
01:50.21bkw_NovceGuru: Nope
01:50.25bkw_G722 runs 64k just like ulaw
01:50.28BBHosshere we go again
01:50.43NovceGurubkw_: I mean then vs .1/c?
01:51.00jayteeyep, down the old "yadda, yadda, yadda, 64k, yadda, DSP, yadda." yawn
01:51.19BBHossbkw_: so it takes less time to transcode or w/e, lower translation time?
01:51.56bkw_BBHoss: it could.. I have never really calculated that
01:52.13BBHossso its just easier to implement then
01:52.14bkw_G722.1 can run at 16 and 32kbit, while G722.1C can run at 24, 32 and 48kbit
01:52.20bkw_BBHoss: resource wise it is
01:52.50BBHossnot patent encumbered either, correct?
01:53.05bkw_you still have to get a royalty free license from polycom to use it commercially
01:53.12bkw_but its free
01:53.26jayteeif it wasn't for the fact that computerized engine analyzers cost a fortune most of today's geeks would be motorheads instead.
01:53.26BBHossyou mean to like put it in your own product?
01:53.34bkw_<PROTECTED>
01:53.34bkw_<PROTECTED>
01:53.35bkw_right
01:54.32jayteeso it's already available then
01:54.49BBHosswhere does it talk about royalties?
01:55.16bkw_BBHoss: you have to contact polycom for the contract
01:55.23BBHossahh
01:55.45BBHossthey should really make it 100% free if they want widespread adoption
01:55.51bkw_thats what they are working on
01:56.00bkw_thats why they let us put it all in FreeSWITCH codec lib and all
01:56.31BBHossoh that was nice of them
01:56.59BBHossnow they just need to get it in all of the media gateways
01:57.43bkw_pstn isn't wideband
01:57.45bkw_so why bother
01:57.49bkw_its more for sip to sip usage
01:58.38BBHosswell if every itsp would put thier numbers in enum or dundi or similar
01:58.45bkw_true
01:58.52BBHossor if you called someone on the same service
01:59.20bkw_thats where it will shine
01:59.24BBHossits almost as low-rate as g729, higher quality, and cheaper/free
01:59.38bkw_I think G729E is wideband
01:59.48BBHossyeah but who supports it
02:00.00bkw_G722.2 is good also... AMR-WB if I remember my numbers correctly
02:00.25bkw_G719 is Siren22
02:00.28bkw_Stereo baby
02:00.32BBHossbkw_: does freeswitch have any kind of dundi type network, or just enum?
02:00.37bkw_just enum
02:00.40bkw_and ISN
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02:01.20BBHossisn?
02:01.24bkw_www.freenum.org
02:01.36bkw_the stuff John Todd did
02:01.52NovceGuruahhh so much krap I know nothing about
02:02.43bkw_BBHoss: FreeSWITCH 1.0.2 is the first to include any of the Siren codec stuff...
02:02.55BBHossbkw_: i run trunk anyways :)
02:03.01bkw_hehe
02:03.07jayteewow, let's all dump * and run that then :-)
02:03.25bkw_jaytee: if Asterisk fits your needs you don't really have a reason to switch
02:03.48jayteei know, i was being facetious
02:04.03bkw_I think BBHoss runs both don't you?
02:04.09BBHossyeah
02:04.14NovceGuruI used to toy with sipxecs
02:04.22bkw_PAIN
02:04.35jayteei still run sipX as a upd/tcp proxy to Exchange UM
02:04.47bkw_jaytee: people use FreeSWITCH for that too
02:04.59jayteeuntil I can get 1.6 working right with TCP
02:05.15BBHossi want to use freeswitch for my application, but it seems the guy who wrote the Telegraph library for it is MIA, and the Asterisk Telegraph library is a good deal better
02:05.21NovceGuruI like the sipx endpoint manager
02:05.25bkw_BBHoss: I know him
02:05.34bkw_BBHoss: you could use liverpie
02:05.43bkw_http://www.liverpie.com/
02:05.46bkw_no clue where they got the name
02:05.47Boraiso i need to have an ftp account that has the  2345-12500-001.bootrom.ld and   2345-12500-001.sip.ld, as well as a sip.cfg?
02:05.59bkw_Borai: yes
02:06.10BBHossbkw_: oh yeah i remember seeing that
02:06.21Boraianything else
02:06.22bkw_they started with Telegraph and did that if I recall
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02:06.36bkw_Borai: I just unzip the whole file from polycom into the dir
02:06.41bkw_and leave it at that
02:06.53Boraiok
02:07.02BBHossyou should see the fun shit i wrote today for a voice broadcasting system
02:07.04jayteeleave it? that'll fubar it
02:07.21jayteeyou need to more than just unzip it. read the damn whitepaper
02:07.21bkw_what?
02:07.24BBHossthey should put it on github
02:07.38bkw_jaytee: I always unzip it and reboot the phone
02:07.43bkw_never once have i had problem
02:07.47bkw_I started at 2.0
02:08.20jayteewithout creating a master config file to match the mac address of the phone? or a phone config file with the unique parameters for that phone?
02:08.57BBHosswow, looks like its being semi-actively developed
02:08.58bkw_jaytee: thats already done
02:09.15bkw_BBHoss: also check out esl.. if you haven't already
02:09.16jayteebkw_, how?
02:09.25bkw_jaytee: I never modify the .cfg files from the tarball
02:09.33bkw_I just include the sip.cfg and phone.cfg
02:09.38jayteeI use them as templates
02:09.39bkw_and modify them locally in my per phone config
02:10.02BBHossi was looking for some tips on rails and freeswitch, so i googled "rails freeswitch" and MY name was the first to come up :)
02:10.09bkw_BBHoss: hehe
02:10.20Boraiok i just have 1 phone at this point, only one polycom) and i am downloading both zip files on to the server and extracting them
02:10.22bkw_I'm not sold on the rails stuff for voip yet
02:10.31Boraiand yes the whitepaper tells me i have to create a config file
02:10.50BBHossi'm just integrating it into my web app, not using rails for 100% voice
02:11.02bkw_BBHoss: thats kewl
02:11.23BBHossright now its just a voice broadcasting system for emas
02:11.26BBHosslike r-911
02:11.46Boraibut the default phone1.cfg that comes in the zip is totally confusing
02:11.46BBHossbut eventually i want to do TTS with it, so they dont have to record a message
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02:14.36sdanielshow do i remove a digit from a dialed number? I understand how to add one ex: exten => _NXXNXXXXXX,n,Dial(SIP/+1${EXTEN}@whatever) but what if i want to have _81NXXNXXXXXX and remove the 8?
02:14.51bkw_sdaniels: ${EXTEN:X:
02:14.54bkw_er
02:14.59bkw_${EXTEN:1}
02:15.12bkw_that'll remove the 8
02:15.33bkw_exten => _8NXXNXXXXXX,n,Dial(SIP/+1${EXTEN:1}@whatever)
02:16.14Boraiok now both zips are in my ftp
02:16.28bkw_unzip them both
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02:16.52*** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my)
02:17.00LemensTSwill stop gracefully kick me out of the cli when its done?
02:17.14Boraiboth unzipped
02:17.30Boraioh wait they have to be in the main directory right?
02:17.35Boraiso I cant have it under a sub directory
02:17.51bkw_Borai: you can if you feed it the full path ot the files in the phone on the ftp
02:17.56bkw_I use http://ip/polycom/
02:18.35Boraiok
02:18.43LemensTSSeems like it is sitting here a while, and i tried show channels and that didnt do anything...
02:18.52Boraithats what i did too ip/polycom/
02:18.54bkw_LemensTS: sounds like something locked up
02:19.07Boraibut in the phone's settings menu i didnt see a path
02:19.21bkw_you set it at the end of the IP
02:19.24bkw_once you set the type to FTP
02:19.25bkw_or HTTP
02:20.20Boraiok im under server menu i changed server type to http
02:20.24jayteebetter to setup DHCP to pass Option 66
02:20.31*** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com)
02:20.47bkw_he has one phone
02:20.53bkw_option 66 is over kill right now
02:21.01jayteeoh, ok. agreed
02:21.11Boraiwell dhcp i cant
02:21.21Boraii dont have dhcp at this place and the asterisk server is on a remote location
02:21.33LemensTSbkw_: exited out of the cli and went back in and it lets me do cmds now.
02:21.41[TK]D-FenderBorai: How do you NOT have DHCP?
02:21.57bkw_LemensTS: you did stop gracefully
02:21.57jayteehell, unless I wanted to configure special features I'd have just used the web interface if all I wanted was to get the phone working.
02:21.58BoraiI dont use DHCP
02:21.58bkw_if I recall correctly asterisk shouldn't be running at all once that command finishes
02:22.04[TK]D-FenderBorai: In a local LAN?
02:22.10Boraiyes
02:22.25Boraistatic ip's manual entered on all the pcs
02:22.38Boraihave 3 pc's here
02:22.43[TK]D-FenderBorai: Fine, just start the phone, go into the bootrom and set the parms manually as well as the server for provisioning
02:22.58[TK]D-FenderBorai: And for internet?
02:23.14LemensTSbkw_: there are 2 active calls still
02:23.17Boraiok cable modem connected to a linksys wrt54gl
02:23.22Boraithat is connected to 550
02:23.27Boraia server
02:23.29[TK]D-FenderBorai: and that doesn't ser DHCP?
02:23.32Boraiand to another switch
02:23.32LemensTSbkw_: think it has to wait till those are done
02:23.40Boraiit does but I dont have dhcp enabled
02:23.42[TK]D-Fenderserve*
02:23.50[TK]D-FenderBorai: CRAZY
02:23.53Boraiit does I just dont use dhcp
02:23.56Boraiwhy?
02:24.06jayteeLemensTS, until all calls are complete it won't exit
02:24.10[TK]D-FenderBorai: so you don't have to do the boring stuff...
02:24.26Boraiboring stuff?
02:24.39[TK]D-FenderBorai: assigning IP's to devices
02:24.43LemensTSit wouldnt hurt to do a stop now after you did a stop gracefully eh?
02:24.49bkw_you guy see this phone http://blog.voipsupply.com/new-products/first-look-cisco-spa-525g-desktop-ip-phone-with-wifi-bluetooth-and-more
02:24.56bkw_I think they failed cuz they didn't include wideband
02:25.53NovceGurudo ciscos still suck ass for SIP?
02:25.54jayteeanyone see the video of the kid getting shot in the head at the Fruitvale BART station while two other cops were holding him down?
02:26.07NovceGuruI remember wrestling with some 7940s < then a year ago...pita
02:26.14Boraiso server address should be myftpip/foldername right?
02:26.20NovceGurugranted 7940 is a bit older then this spa525g
02:26.56bkw_jaytee: yes I seen it
02:27.17jayteedisgusting
02:27.27bkw_someone better go away for that one
02:27.33bkw_or hell will break loose
02:27.35jayteeindeed
02:27.46bkw_is happy he lives in Oklahoma
02:27.47jayteehell's already broken loose
02:27.54*** part/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com)
02:27.56jayteean Okie? OKC?
02:28.03bkw_McAlester
02:28.12bkw_southeastern oklahoma
02:28.19jayteeah, I was stationed at Tinker during the late 70's
02:28.29bkw_I was born in the late 70's
02:28.30bkw_:P
02:28.40bkw_I like it here.. its calm
02:28.41jayteeI was around when they invented dirt
02:28.42*** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com)
02:28.42bkw_and I can carry a gun
02:29.02jayteeI liked the people but the landscape's a bit dreary
02:30.04NovceGuruahha that cisco has a "Kensington security slot support"
02:31.04*** join/#asterisk jtodd (n=jtodd@117.sub-70-214-1.myvzw.com)
02:31.32Boraierm
02:32.20bkw_the snom 820 has wifi too... I have one on the way
02:33.13BBHossbkw_: what hardware did you use to test CELT?
02:33.21bkw_BBHoss: my Mac
02:33.23bkw_and FS
02:33.30bkw_but Ekiga has support for CELT now in the latest builds
02:33.35bkw_stkn added support for it
02:33.36BBHossoh ok cool
02:33.51bkw_I have an XM radio celt stream up for people to call if they want
02:34.02BBHosswhere
02:34.09bkw_sip:886@taz.bkw.org:5080
02:34.13Boraiok
02:34.28Boraiit took like 5 minutes on the initializing screen but now its downloading the new bootrom
02:34.54BBHossbkw_: so whats the easiest way to test it on a mac?
02:34.55bkw_Borai: give it time
02:35.01bkw_BBHoss: portaudio + freeswitch
02:35.05bkw_BBHoss: thats hwo I did it
02:35.23Boraithe download was fast tho
02:35.25BBHossguess i'll do it another day then :)
02:35.35bkw_BBHoss: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-December/009746.html
02:35.43Boraiformatting fs please wait.
02:35.55bkw_Borai: just chillax you'll be ok
02:36.14BBHossits fun when its says ERROR 10 or w/e
02:36.23Boraidownloading bootrom again
02:36.30bkw_haha
02:36.36Boraiim excited what a phone can do
02:36.42Boraiits interesting
02:36.54*** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
02:37.45bkw_Borai: you have caught the voip bug I take it?
02:38.03Boraivoip bug?
02:38.14Boraiformatting again
02:38.26Boraiendless loop?
02:38.48Boraioh no nevermind now its downloading new application
02:38.53BBHossBorai: no it has to download all of them up to the latest version i think
02:38.54bkw_see told you to chillax
02:39.31Boraioh wait so you cant just go from ex. 1.1 to 1.7 you have to do 1.1, 1.2, 1.3, ...?
02:40.29BBHossBorai: dunno, but its grabs the file like 10 times
02:40.50BBHossyou can tail -f the tftp log or the log it uploads and see what its doing
02:44.00*** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net)
02:45.15Boraiok booted
02:45.50*** join/#asterisk fatnasty1 (n=fatnasty@cpe-72-190-76-209.tx.res.rr.com)
02:48.22fatnasty1Hello?
02:49.14Boraigoodbye?
02:49.18fatnasty1Cool it works
02:49.36jayteeno it doesn't
02:51.28Borailol
02:52.02Boraiok now i changed to ftp so that the phone can save the config when i change something
02:52.07Boraiwhat about tag sn to ua?
02:59.16Boraiis there an easy way to create a config file for the polycom
03:00.25jayteei use a bash script to copy template files and edit them with sed
03:00.57jayteeso i just type ./prepphone.sh macaddress XXXX and i'm done
03:01.17Boraiok nice but the problem is
03:01.23jayteeor in the case of multiple lines on a 550 I have another script to do that.
03:01.31Boraia) I have never configured a polycom
03:01.47carraryou get to learn!
03:01.50Boraibefore, b) I dont understand anything in the example file phone1.cfg
03:02.13carrarhelps to understand XML
03:02.31carrarand helps to understand how the phone works
03:02.43carrarso need to learn all that 1st
03:02.47jayteethe configuration parameters are documented in the SIP Admin guide
03:02.50carrarstart with the polycom admin guide
03:03.40carrarthen you can write scripts to make the configs for you
03:03.46carraronce you know what you need to change
03:04.30BoraiI dont need to create a script I am not that lazy I can hand edit it but
03:04.36carrarhaha
03:04.48carrareven I wouln't want to hand edit 5 polycom phones
03:04.59Borai:)
03:05.03[TK]D-Fendercarrar: I do mine by hand all the time
03:05.07carrarwhen spending that time to write the script can make life so simple
03:05.08Boraithey are documented in the admin guide?
03:05.13carrarhahha
03:05.14jayteewhat's lazy about using a script? want to spend 3 hours or 10 minutes configuring 30 phones?
03:05.17jayteeduh!
03:05.35Boraiwell i am not going to configure 30 phones
03:05.42Boraiif i was i would write a script or even hire someone
03:05.42jayteeever?
03:05.43carrarwho wants to edit dozens of phone polycom config sby hand
03:05.44Boraithat would do it
03:05.54[TK]D-Fenderjaytee: 10 minutes?  You saw how much time I've wasted to toto retards here?  10 minutes is a FIELD DAY!
03:06.01[TK]D-Fendertotal*
03:06.22carrar1 phone is easy by hand once you know what to change
03:06.48jaytee[TK]D-Fender, what about my 10 minutes?
03:07.06jaytee~IWMWB
03:07.07jbotI WANT MY WEEKEND BACK!
03:07.16carrarI took that 10 mins and wrote a script, so now going forward it takes 3 seconds
03:08.10*** join/#asterisk fatnasty1 (n=fatnasty@cpe-72-190-76-209.tx.res.rr.com)
03:08.45jayteeyeah, I exaggerated. If I write a list of all the macs for the phones I can crank out 30 configs in about 4 minutes probably but I'm a fast typist.
03:09.22carrarlist of mac's and a script, 4 seconds
03:09.22jayteetoughest part was learning to use SED
03:10.14jayteethat's cuz you're reading in the list into the script to batch process it. I hadn't taken it to the next level.......yet
03:10.25carrarI use perl
03:10.32jayteeI'm just learning perl
03:10.42carrarbest way to learn, needing to write something
03:10.43jayteegot my llama book right here :-)
03:11.44carrarthat same script can also generate your sip.conf config for the phone as well
03:12.00carrarmaking your time faster to complete that phone
03:13.10carraror anything else uniqe to that extension/phone
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03:23.33*** join/#asterisk Daejeo (n=chatzill@118.219.208.133)
03:23.45Daejeogreetings :)
03:24.50Daejeoanyone knows path to documentroot on switchvox ?
03:25.10[TK]D-FenderDaejeo: I'm sure www.digium.com does
03:25.19Daejeo"/var/www/html"
03:25.26carrardocumentroot?
03:25.52carrarYou are not suppose to have access to the switchvox OS :)
03:26.19Borairfc2543 hold yes or no?
03:26.31[TK]D-FenderBorai: leave alone
03:27.31*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
03:28.07Daejeo[TK]D-Fender: did you play with switchvox ?
03:28.22[TK]D-FenderDaejeo: No
03:29.30carrarDaejeo, it's all javascript anyways
03:29.41carrarif you know that you will find it easy
03:34.22Boraihow can you save the config file in the phone to the boot server?
03:34.53jayteecan't
03:34.56Bad_Robot-boot your live linux cd ;)
03:35.22jayteeit'll upload log files but not configs
03:35.31carrarit will uplaod changes
03:35.35Boraiok
03:36.03carrarif you have the 'overrides' directory setup
03:36.30carrar'limited changes'
03:37.21Boraiok
03:37.38Boraiwell that means i will have to create a config file
03:37.43carraryes
03:37.50Boraii just configured the phone using the menu on it
03:37.59[TK]D-FenderIn Soviet Russia, phone configures YOU
03:38.21[TK]D-FenderBorai: Only thing you should program on the phone itself is IP parms & the IP & type of boot server
03:38.44carrarnot even that!! use DHCP to do that for you
03:38.59carrarmust be lazier!
03:40.35[TK]D-Fendercarrar: No, you seem to have missed that... he doesn't even HAVE DHCP
03:40.41carraroh
03:40.45carrarthat has to suck
03:41.08carrarput that on your switchvox box
03:41.10carrar:)
03:41.28carrarI'm assuming he's running switchvox
03:41.55carrarwrong person
03:42.00carrarman
03:42.07carrarI'm just not paying enough attention
03:42.28jayteeno, it's that Daejeo guy that keeps asking questions about hacking switchvox
03:43.18carrarBorai, any reason why you can't run a DHCP service?
03:43.35jayteehe has ALS
03:43.44carrarALS?
03:43.52jayteeAcute Laziness Syndrome
03:43.55carrarhaha
03:44.14carrarI think DHCP is easier then configuring a handfull of phones via the menus
03:44.22jayteeya think?
03:44.23[TK]D-Fenderjaytee: My mom told her co-workers she hard SARS (Serios Anal Retentive Secretary)
03:44.31[TK]D-Fenderhad*
03:44.32jayteelol
03:44.39carrarheh
03:45.57carrarI could even be talked into writting his dhcp.conf
03:46.03*** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-7b6abd2a67c06443)
03:46.38[TK]D-Fendercarrar: I mean NO DHCP.  He's running a Linksys router and disabled it.  Its not for LACK of it...
03:47.07carrarWhats Asterisk running on then?
03:47.19carrarsome remote hop someplace else?
03:47.35carrarmust be
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04:15.14SlicerDiceranybody know about Aastra 480i models?
04:15.19SlicerDicerand thoughts perhaps?
04:15.34*** part/#asterisk Lord_Drachenblut (n=drachenb@12.197.74.66)
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04:18.07[TK]D-FenderSlicerDicer: Largely decent, but rare to advise
04:18.28SlicerDicerwhat there is better models or what?
04:19.20[TK]D-FenderPolycom > All
04:19.38SlicerDiceruglier than all too
04:20.13SlicerDicerif I were to vomit.. thats what a polycom would look like.. but in all seriousness what features makes them better?
04:20.15[TK]D-FenderSlicerDicer: Far from
04:20.48[TK]D-FenderSlicerDicer: Solid phones, highly configurable, superior call handling, physical & audio quality
04:21.20SlicerDicerfeatures please...
04:21.47SlicerDicerthose are just thoughts of why they are better IMO
04:21.53SlicerDicerfeature for feature what makes them better
04:22.42[TK]D-FenderSlicerDicer: And a piece of shit tah echoes on all your calls and feels like a flimsy piece of shit with an unreadable display must be AWESOME
04:22.57SlicerDicerthat makes no sense
04:23.04SlicerDicerI just asked feature for feature what is better
04:23.23SlicerDiceryou come back with some garbage..
04:23.31maddog01what phones are you comparing
04:23.44SlicerDicerobviously you are biased with polycom if you wont answer a simple question
04:23.48[TK]D-FenderSlicerDicer: wXML microbrowsers, call join/ split, drop-conferencing, etc
04:23.52SlicerDicermaddog01: I am asking about the Aastra 480i
04:24.02maddog01good phones
04:24.06maddog01i use them
04:24.09SlicerDicergood deal :)
04:24.22SlicerDicerI have the Aastra 480 analogs and such :) thats why I am considering them for sip
04:24.27maddog01but i would go for the new 57i
04:24.35[TK]D-FenderSlicerDicer: What was that about BIAS?
04:24.56SlicerDicer[TK]D-Fender: if you are aprehensive and get so angry obviously there is something in your line of thinking
04:24.58[TK]D-Fenderand I had a 57i CT... made me wish for my old bedside Polycom IP 301 <-
04:24.59SlicerDicerjust sayin
04:25.27[TK]D-FenderSlicerDicer: Aastra took a bad turn on the 5i seriesYes...
04:25.34SlicerDicer57i yeah I was looking at that one
04:25.50maddog01i bought five different phones when i converted to voip
04:25.52SlicerDicerhowever I am able to snag some 480i for 75$
04:26.12SlicerDicer480i CT for 135
04:26.18SlicerDicerso thats why I am considering :)
04:26.22[TK]D-FenderSlicerDicer: that is a great deal...
04:27.39maddog01good price i perfered the look of the 57i
04:27.54SlicerDicerunderstood :)
04:28.06[TK]D-FenderSlicerDicer: 480i was better than the 5i series... their production standard went crappy...
04:28.55SlicerDicerwhat happened that made them go crappy?
04:29.29maddog01i don't think so 57i supports alot more feature though
04:29.44[TK]D-FenderQuick summary of 57i CT = Base & handset have NO wieght, shifted on the table... rubber feet = BLEH.  Backlit LCD... with NO VIEWING ANGLE!  pixel disply with retarded char-martix DRIVERS (DIE!).  Crashed somewhat regularly.  shit-for-all rubbery buttons.
04:29.47maddog01newer firmware and hardware
04:29.52[TK]D-Fender^^^^^^^^
04:30.10[TK]D-Fender5i series buttons = STAB
04:30.22maddog01the viewing angles sucks. hes right
04:30.26[TK]D-Fendergrabs a big knife
04:30.33stintelagrees with the viewing angle ツ
04:30.36SlicerDicer:/
04:30.57maddog01the buttons are okay
04:31.04[TK]D-FenderH8
04:31.14SlicerDicermaddog01: is it lightweight?
04:31.16maddog01they actually feel better then the 3com
04:31.18SlicerDicercause I mean my 480's are bricks
04:31.19SlicerDicerlol
04:31.31maddog01i used at another place
04:31.51[TK]D-Fender480 is a very decent phone and at 75$ its a no-brainer.
04:31.51SlicerDicerahh
04:32.03SlicerDicer[TK]D-Fender: thats what I was thinking
04:32.04maddog01well they are light but that dosn't mean anything
04:32.17[TK]D-FenderSlicerDicer: now, NEW would be another matter, but would I suggest you spend more and pass up that deal?  Hell no.
04:32.28maddog01i havent had anyone complain about the weight
04:32.35SlicerDicer:)
04:33.01[TK]D-Fendermaddog01: Mine dragged across the desk and the PITA of resetting it after calls because of the SHIT FOV... forget it
04:33.14SlicerDicer[TK]D-Fender: I was thinking of getting 2 of them heh
04:33.37maddog01[TK]D-Fender: get a longer cord
04:33.41[TK]D-FenderSlicerDicer: I'm not sure, but they might be PoE only... be warned
04:33.49maddog01no
04:33.52maddog01there both
04:34.05[TK]D-Fendermaddog01: 280i's?  You sure his have bricks?
04:34.09[TK]D-Fender480i*
04:34.30maddog01?
04:34.43[TK]D-Fendermaddog01: Both what?
04:34.51SlicerDicer[TK]D-Fender: PoE?
04:34.53maddog01oh poe
04:35.04[TK]D-Fendermaddog01: Yes.. welcome to the conversation..
04:35.07[TK]D-Fender~poe
04:35.08jbotmethinks poe is Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet
04:35.09[TK]D-Fender^^^
04:35.11maddog01power pack or poe
04:35.40maddog01the 480i don't have PoE
04:35.55[TK]D-FenderSlicerDicer: Most decent SIP phones support PoE for power and some that do don't have wall-plugs.
04:35.57SlicerDicerI dont think any of the 480i have PoE thats why I was wodnering...
04:36.01[TK]D-Fendermaddog01: Yes they do...
04:36.16SlicerDicerwell fricken A
04:36.21SlicerDicerthay have it?
04:36.27SlicerDicerbut also all of them have power adapter
04:36.37[TK]D-FenderSlicerDicer: If they do then you're covered
04:36.39maddog01built in or they have a clip on spliter
04:36.47maddog01???
04:38.02jayteeand be careful what you pick to use PoE. Cisco's poe switches and injectors only work with their phones, not with Polycoms or any other 802.3af compliant phones
04:38.24SlicerDicerI will make sure they are not PoE
04:38.29SlicerDicerthat should solve that eh? ;-)
04:38.46jayteelong as you have an outlet nearby
04:39.16[TK]D-FenderSlicerDicer: they ARE Poe.  That is a FACT.  I'm not sure if they are Poe *ONLY*
04:39.18drmessano^I love 802.11ruok
04:39.25SlicerDiceryeah thats what I mean fender
04:39.44[TK]D-FenderSlicerDicer: If they aren't and you have bricks for them, great.  If they come with injectors, equally great.
04:39.53SlicerDicerjaytee: there is Aastra phones there now currently :)
04:39.55jayteeonly downside to that is power failures. If you have a 802.3af poe switch or midspan hub plugged into a UPS to power phones they'll stay up when the power blips. If you don't then the phone has to reboot
04:40.00maddog01i just checked aastras site
04:40.15SlicerDicerjaytee: they are just analog not voip
04:40.26maddog01they don't say anything about PoE
04:40.43[TK]D-FenderSlicerDicer: 280i no voip?
04:40.49[TK]D-Fender480*
04:40.50jayteemaddog01, what phone?
04:40.54[TK]D-FenderWTF!?
04:41.18[TK]D-Fender[23:25]<SlicerDicer>however I am able to snag some 480i for 75$
04:41.26[TK]D-FenderSliWTF are are you talking ANALOG for here?
04:41.55SlicerDiceryep [TK]D-Fender
04:42.03jayteefuck sip phones, everyone who's anyone knows that FXS is the new black!
04:42.08SlicerDicerthey are 480 non voip they are old
04:42.38[TK]D-FenderSlicerDicer: You said 480i there.  Multiple times
04:42.39SlicerDicerremember earlier talking about fxs module stuff ADSI etc etc?
04:42.46drmessano^I had a 480i analog
04:42.47[TK]D-FenderSlicerDicer: 480i = SIP <---
04:42.57drmessano^I gave it to goodwill
04:42.59SlicerDicerI said there is aastra phones there
04:43.08[TK]D-FenderSlicerDicer: http://www.aastra.com/cps/rde/xchg/SID-3D8CCB6A-07999B78/04/hs.xsl/19493.htm
04:43.08SlicerDicerthey are analog no voip
04:43.12[TK]D-FenderSli480e = analog
04:43.23[TK]D-FenderSlicerDicer: COMPLETELY different
04:43.32SlicerDicerI know
04:43.39SlicerDicerbut they look almost identical thats what I was driving at
04:43.43SlicerDicersame footprint etc etc
04:43.48[TK]D-FenderSlicerDicer: And you would be a complete retard to pay $75 for a &#^$%ing analog phone
04:43.51SlicerDicerthus no problem delete/replace
04:43.52drmessano^LOL
04:43.55drmessano^Yeah
04:43.59*** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
04:44.01SlicerDicer[TK]D-Fender: JESUS H CHRIST!!!
04:44.14maddog01sorry im back
04:44.31drmessano^Is it "Let me do the worst thing I can do with VoIP" day?
04:44.32SlicerDicerI have analog phones NOW!!! I am going to buy VOIP!!! readzor please
04:44.32[TK]D-FenderSlicerDicer: You want to compare FEATURES?!  You can't hold a friggen call on those!  No MoH!
04:44.37drmessano^I thought that was last week?
04:44.40drmessano^Shit.. I am off
04:44.52[TK]D-FenderSlicerDicer: You just said you're goign to go buy the analog 480's
04:45.05SlicerDicer[TK]D-Fender: as in I am taking your advice and doing away with my analog phones that you said earlier
04:45.11drmessano^HES CONFUSED BY THE "PHONE" PART
04:45.16drmessano^LEAVEM A LOAN
04:45.21maddog01the 480i» Compatible with IEEE 802.3af inline power device
04:45.21maddog01» Optional Power Over Ethernet (POE) injector available, but not provider
04:45.38maddog01provided
04:45.45SlicerDicer[TK]D-Fender: where di I say I was going to buy "analog"
04:45.48drmessano^How does 802.3af work with analog?
04:45.51maddog01straight from aastra
04:45.53drmessano^Ping 10 and 14?
04:45.57drmessano^Pin
04:46.02maddog01the 57i is built in
04:46.17SlicerDicer[21:40:42] <SlicerDicer> jaytee: there is Aastra phones there now currently :)
04:46.17SlicerDicerSlicerDicer> jaytee: they are just analog not voip
04:46.27SlicerDicer[TK]D-Fender: does that clear it up?
04:46.42SlicerDicerI said nothing about buying analog phones
04:46.49[TK]D-FenderSlicerDicer: I hope to hell not.
04:47.10[TK]D-FenderSlicerDicer: But your broken prasing wasn't helping
04:47.14[TK]D-Fenderphrasing*
04:47.20SlicerDicerlol
04:47.22[TK]D-Fenderdang, typing skills fading fast...
04:47.32SlicerDicersorry my brain works
04:47.34SlicerDicerin clusters!
04:47.36SlicerDicer;-)
04:47.50SlicerDicerI will try to complete my thoughts better :)
04:47.50maddog01SlicerDicer: i would buy 5xi series they have alot more feature and faster hardware but if your looking from the cheapes price go with the 480i
04:48.27SlicerDicermaddog01: basicly what I wanted to do was get my house fitted out with voip phones to start with
04:48.36SlicerDiceronce that transition is complete I can work on going "nicer"
04:48.48SlicerDicerthen pawn the other phones down the food chain to family and what not maybe :)
04:48.56SlicerDicerpawn = give
04:49.29maddog01lol
04:50.08maddog01you don't need aastra phones for your house
04:50.20maddog01get some thing cheap
04:50.25maddog01and wireless
04:50.47maddog01vtech or panasonic sip phones with dect
04:51.00SlicerDicerhmm
04:51.14SlicerDicerI dont like "cheap" cause that leads to shit quality
04:51.15maddog01these are business phones
04:51.21SlicerDicerleast I know the aastra had good quality :)
04:52.18drmessano^I've used ATA's and cheap phones at home.. Now I have a mix of things
04:53.20drmessano^VoIP phones in the office, next to the bed.. ATA's with $25 5.8GHZ cordless on them for roaming around.. Then a couple $5 phones on ATA's that are wired into the house wiring for filling in the gaps
04:53.28SlicerDicerhttp://idisk.mac.com/slicerdicer/Public/phone.JPG blurry yes but thats what I currently use maddog01
04:53.59SlicerDicerI actually have 3 of them all together
04:54.10drmessano^I've been told the $18 vtech 5.8GHZ at walmart is an awesome phone
04:54.27SlicerDicerI gave my 5.8ghz vtech away
04:54.39denonbut does it have a sepaeate flash button!
04:54.44denonseparate
04:54.47denonthat's the question
04:54.52maddog01slicerdicer: check this out http://www.canadianvoipstore.com/product_info.php?products_id=3708
04:54.54drmessano^lol
04:55.14maddog01thats more then enought
04:55.23SlicerDicerCAD?
04:55.30SlicerDicerthats like what 20$ USD? ;-)
04:55.37drmessano^I'm done listening to SlicerDicer anyway.. $75 analog Aastras.. WTF
04:55.50SlicerDicerdrmessano^: errrm... damn it
04:56.01SlicerDicerdrmessano^: they are voip aastras...
04:56.04maddog01SlicerDicer: shit wrong phone and whats wrong with canada
04:56.07SlicerDicerthe one I linked the picture of is what I have
04:56.12maddog01one csec
04:56.14maddog01sec
04:56.20SlicerDicermaddog01: nothing is wrong with canada :)
04:56.21drmessano^maddog01: Damn canucks live there
04:56.40maddog01lol
04:56.40drmessano^maddog01: Three words "Canadian bacon, eh"
04:56.56maddog01i'm having some right now. lmao
04:56.58SlicerDicerlol
04:56.59drmessano^and it's "ABOUT", not "ABOOT"
04:57.14SlicerDicerABOOT is a car trunk
04:57.27SlicerDiceror for ABOOT up your ass ;-)
04:57.48drmessano^jaytee: Did you LOL?
04:57.51maddog01now now lets not start a international insident
04:58.18jayteeLOL
04:58.32SlicerDicerlol
04:58.41drmessano^Go play some HACKEY, TOOK-face!
04:58.45SlicerDicermaddog01: ahh I am far enough away
04:58.56SlicerDicerI can flee to mexico before canada makes it to New Mexico ;-)
04:59.12maddog01lol
04:59.25drmessano^BYE,EH is NOT A SIP HEADER
04:59.30maddog01but yet you want to buy a canadian made phone
04:59.50SlicerDicermaddog01: your argument is invalid as I like canada :)
04:59.56drmessano^maddog01: An analog one, to BOOT
04:59.59SlicerDicerI just have some fun from time to time :)
05:00.08maddog01aastra the phones you all love and hate
05:00.21SlicerDicerI mean hell I am in New Mexico.. think of all the shit you could come up with for me... LOL
05:00.22drmessano^Oh god, he puts smileys at the ends of sentences to convey non-threatingness
05:00.41drmessano^Oh god, he puts smileys at the ends of sentences to convey non-threatingness :)
05:00.45SlicerDicerlol
05:00.58drmessano^Next it'll be "hah lol j/k"
05:00.59SlicerDiceronly thing bad about canada?
05:01.05SlicerDicerfucking cold...
05:01.10SlicerDicernuff said on that
05:01.11SlicerDicerlol
05:01.24SlicerDicerhay guise! does that work drfreeze
05:01.29SlicerDicerbah tab completion ftl
05:01.31drmessano^maddog01: I do have one nice thing to say about canada
05:01.56*** join/#asterisk zeeesh (n=zeeesh@203.215.179.43)
05:02.04drmessano^maddog01: I have proof that contrary to popular comedic belief, all the assholes are NOT in canada
05:02.15drmessano^maddog01: They migrate south
05:02.17jayteespeaking of Canadian, a friend of mine in another chat channel was just talking about Mafia Wars on Facebook, she said: "When I'm feeling silly I just go to the hit list and sucker punch someone at random, heh"
05:02.18SlicerDicerlol...
05:02.24*** join/#asterisk rob0 (n=rob0@cardinal.lizella.net)
05:02.33maddog01yah there just south of canada
05:02.37maddog01lmao
05:02.53drmessano^maddog01: Some even closer to mexico
05:03.00jayteeI'm a Masshole by birth but I live in Indiana now
05:03.11maddog01lol
05:03.12SlicerDicerhey now
05:03.14SlicerDicerI resemble that remark
05:03.17drmessano^maddog01: Where $75 analog phones roam free like el chupacabra
05:03.24jayteewhy, you from Mass?
05:03.38maddog01lmao
05:03.41SlicerDicerI am from New Mexico... thats close to Mexico
05:03.46jayteeoh
05:04.10drmessano^maddog01: Apparently the legend of El Cheapacabra is true
05:04.13jayteesad about Richardson having to bow out
05:04.19SlicerDicerbut we dont use Pesos here just to clear that up
05:04.22SlicerDiceryeah jaytee
05:04.29*** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net)
05:04.31SlicerDicerI hope he has done nothing wrong for the sake of new mexico..
05:04.33maddog01drmessano^: is on a roll
05:04.35rob0It's newer than Mexico, obviously!
05:04.36SlicerDicerwe dont need another scandal...
05:05.06jayteeLand of Enchantment my ass! how enchanting can several hundred thousand square miles of desert be anyways?
05:05.17SlicerDicerjaytee: well it keeps the unbelievers out
05:05.23SlicerDicerthey rot at the borders
05:05.27rob0I think they're talking about upgrading the name ... New and Improved Mexico
05:05.31jayteeleaving just you tin foil hat folk
05:05.35*** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com)
05:05.49maddog01newer Mexico
05:05.52maddog01lmao
05:05.54SlicerDicerlol
05:06.20jayteehow about changing the motto to "Yeah, we got Roswell and aliens"
05:06.40SlicerDicernot kosher with Groome Lake jaytee
05:06.47Zippomanhey can someone help me out with some asterisk issues?
05:06.55drmessano^~elcheapacabra
05:06.55jbotWatch out, EL CHEAPACABRA will suck your blood, leaving a lifeless corpse, all for the thrill of putting cheap analog phones on barely working Trixbox PII-300s with less RAM than an iPhone.  BEWARE!
05:07.11jayteewhat does Groome Lake have to do with Hassidic dietary laws?
05:07.19maddog01slicedicer: okay i found a good site with a bunch of phones: http://www.cordless-phones.uk.com/voip-phones/voip-dect-phones/
05:07.30drmessano^VoIP DECT PHONES
05:07.33drmessano^WHY O WHY
05:08.02SlicerDiceryeah but thats UK?
05:08.02jayteebecause we can!
05:08.07drmessano^ATA + Some Analog Deal of the Weak DECT/non-Dect phone
05:08.12drmessano^oh lord
05:08.18mchouhell no to VoIP DECT
05:08.18maddog01cordless voip phones for the house
05:08.22Zippomanwhat does it mean if i call my number and i get a long silence
05:08.30mchouagrees with drmessano^
05:08.39drmessano^maddog01: STOP POSTING LINKS WITH INFO THAT HAVE FOREIGN DOMAINS.. YOURE CONFUSING HIM WITH TLDs
05:08.39maddog01drmessano: who still uses analog
05:08.46drmessano^maddog01: Thankya
05:08.56SlicerDicerdrmessano^: no.. I just dont usually order international if I can help it
05:09.06SlicerDicerI dont like customs peering in my shit
05:09.07SlicerDicerlol
05:09.14drmessano^maddog01: I do.. ATA + Cordless analog is a WAYYY better idea than some VoIP DECT phone..
05:09.26maddog01it's a site for information
05:09.32SlicerDicerohh
05:09.47maddog01you have to find the phones locally
05:09.54drmessano^At least with an ATA you have a known working SIP client and can put whatever Cordless you want on it
05:10.13Zippomancan any of you help me?
05:10.25drmessano^Otherwise youre stuck with a niche product that is gonna have less QC in the field
05:10.29mchouyup.  and cordless analog phone are improve very quickly and are inexpensive
05:10.42maddog01drmessano: your right but i stopped using analog in 2002
05:10.56maddog01drmessano: i dont want to go back
05:11.05drmessano^maddog01: What's wrong with an ATA?
05:11.14drmessano^I didnt say "USE ANALOG"
05:11.19SlicerDicermchou: vtech are nice no ;-)
05:11.20drmessano^I said USE AN ANALOG PHONE
05:11.33maddog01i still have to used one for my 10K fax/printer thats what
05:11.46mchouSlicerDicer: next time get panasonic :)
05:11.51SlicerDicerhaha
05:11.56drmessano^Christ, an analog phone on an ATA is hardly going back 20 years in time.. What the hell do you think is inside your dect VoIP phone?
05:12.03drmessano^Cordless bits + ATA
05:12.10drmessano^Just not in 2 boxes
05:12.30maddog01depends on the cordless phone
05:12.49maddog01some are pure digital some are hybrids
05:12.58rob0Folks, I find myself in the very sad situation of having to block an abusive caller, by caller ID. I know it can be done, just not sure about the "best" way. Suggestions please? Thanks.
05:12.58maddog01your right
05:13.03drmessano^Well, I dont need a multiline Dect VOIP system.. thats why I have a PBX
05:13.31maddog01call then and tell them to stop calling you. lol
05:13.32rob0and no, it is not a creditor ;)
05:13.33mchouactually, are there consumer corldeless analog phones with full duplex speaker?
05:13.48mchoucordless*
05:13.53SlicerDicerhttp://www.cordless-phones.uk.com/img/watermark/008189.jpg thats just lunacy maddog01
05:14.05*** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:5d00:b8cf:9894:f1a5)
05:14.05maddog01a pbx for a house is over kill
05:14.29drmessano^So is a multiline VoIP Dect system
05:14.40drmessano^I see no justification for it
05:14.55mchourob0: google astdb blacklist
05:15.27mchourob0: is it an ex-girlfriend? :)
05:15.31maddog01SlicerDicer: what phone was that
05:15.38SlicerDicerhttp://www.cordless-phones.uk.com/voip-phones/voip-dect-phones/siemens-a580-ip-phone
05:15.44drmessano^HA
05:15.46maddog01ah
05:15.46Zippomancan any of you help me configure my asterisk for certain tasks...id be happy to pay for services
05:16.05maddog01you can do up to 8 handsets
05:16.05mchouZippoman: depends on the task
05:16.07drmessano^mchou: Im staying out of this one.. Maybe Panasonic will sell a VoIP dect phone after all
05:16.08maddog01nice
05:16.30mchoudrmessano^: panasonic sells consumer voip phones
05:16.39rob0mchou: it is a very sad story. I failed as a parent. It is my daughter. :(
05:16.45drmessanomchou: I know.. scary
05:16.51maddog01all: it's on the link i posted
05:16.51rob0thanks for the tip BTW
05:16.55mchourob0: you serious?
05:17.14rob0unfortunately yes, phone is ringing even now.
05:17.20rob0ringers turned off
05:17.21mchoulordy
05:17.38maddog01rob0: get a new number
05:17.44mchouwtf is going on w/ society these days
05:17.51Zippomanlmao
05:17.58maddog01lmao
05:18.20drmessanoYou're blocking your daughter?
05:19.12Zippomananyone think they can help me out
05:19.18drmessanoputs on his drphil mask
05:19.20denonDr Phil
05:19.26drmessanoSo, tell me where the problem started
05:19.28maddog01lmao
05:19.33denonno .. not drmessano as dr phil .. the real dr phil
05:19.35drmessanoWhat brought things to this point
05:19.35denondrmessano is trouble
05:19.36denon:)
05:20.00mchoulol
05:20.21mchoudrmessano is not a REAL doctor
05:20.22drmessanoAt what point did things get so bad, that you needed to Asterisk blacklist your own daughter?
05:20.27drmessanoWas it the iPhone?
05:20.35Zippomanhahaha
05:21.05maddog01l don't think i have ever laughted this hard
05:21.17maddog01LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO LMFAO
05:21.24drmessanoWhere are the hugs?  Where is the love?  Where is "I love you, even if you did stab my cat?"
05:21.28mchoudude, this is no laughing matter
05:21.29drmessanoCan we get a hug here?
05:21.46Zippomanthis sucks for that guy
05:21.48drmessanoWhen did you know she had a drug problem?
05:21.58drmessanoerrr
05:22.05rob0Thanks mchou, and guys, we really are broken up about it. She's 19, just married a 52-y-o loser.
05:22.13drmessanoOuch
05:22.22rob0a total wacko
05:22.23drmessanoreadjusts his hat
05:22.26maddog01sugar daddy
05:22.30Zippomanim sorry dude
05:22.30rob0you think I'm bad, boy!
05:22.34drmessanorob0: Can we get him on the show
05:22.52rob0heck no, he's a total loser, has NEVER had a real job that I can tell
05:22.53drmessanorob0: I want to know.. what is going through this mans head.  She is just a chid
05:22.56drmessanochild*
05:23.11rob0I can tell you. He wants someone to be his meal ticket.
05:23.43drmessanorob0: We don't extend this offer very often, but the Dr Phil show has a team of hired assassins that keep us from having followup shows in tough cases
05:23.48drmessanorob0: How can WE help YOU?
05:23.58rob0haha :) that might help
05:24.29*** join/#asterisk Maliuta_CA (n=biteme@206.47.36.150)
05:24.49*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-5393d80c55968350)
05:24.56maddog01some one should rename this channel the Comedy Channel
05:24.58maddog01lol
05:25.15drmessanoYou know.. Im not against the whole "marrying someone half your age" thing, but when its because her job at Burger Barn is your way out of the trailer park, TIME FOR A LIFE READJUSTMENT
05:26.25Zippomanhey guys someone please help me out with this...i call my number and i just get a long silence...im using voicepulse...i think i have a problem with my extensions.conf...i basically want to be able to call my number and then me prompted to enter another number...any help greatly appreciated
05:26.26rob0I hate to get anyone's interest piqued, but all I can really say is that it's far worse than it sounds.
05:26.51drmessanorob0: Perhaps rather than helping block your daughter, maybe we can mend this problem by helping you set up a dialplan to give him a cardiac at 3am on some random Thursday.. I am checking "show applications" now..
05:26.52rob0Zip, verbosity in the console? Are the calls arriving?
05:27.38*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
05:29.18drmessanorob0: While it's not my daughter, I have a sister, my only sibling, who has two children by some hippie druggie loser who has beat her to the point of being put in the hospital twice, which doesnt count the daily beatings that dont warrant emergency care.  I almost never want to answer the phone when she calls
05:29.42drmessanoPeople are sick
05:30.11maddog01i there is more serious matters but im watching Angelina Jolie get her ass handed to her in Mr & Mrs Smith and she is so hot
05:30.12rob0Indeed, a screwed up society. At least my other kids are doing better.
05:31.37drmessanorob0: Her kids have already learned profanity that I have yet to pick up in my day-to-day.. and I work in *IT*.. oh, they are 4 and 2
05:32.14maddog01thats suck drmessano
05:32.16mchourob0: I'm curious.  did you use asterisk before all this happened?  Or are you setting asterisk up just for this?
05:32.30maddog01that sucks drmessano
05:32.59rob0I've been using * for several years, dating back to pre-1.0 and pulling from cvs.
05:33.17rob02004 maybe
05:35.30maddog01rob0: just setup a inbound route for that Caller ID and tell it to ring busy or hang up or go to voice mail.
05:35.45rob0yeah I want to do the VM
05:35.48drmessanoI was JUST gonna say that
05:36.52rob0it will take her some time to figure out that she's blocked, and we'll be able to identify those calls by area code.
05:37.05rob0(those calls==calls from other numbers)
05:37.29maddog01ah
05:37.54*** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net)
05:38.32drmessanoYou know, I always store paint thinner in Wild Turkey bottles
05:38.33maddog01has anyone setup fax to email on a trixbox install
05:38.37drmessanoIm just sayin
05:39.09drmessanoOh, world.. ---> /clear
05:39.24maddog01the faxes come in and get stored to the tmp folder but don't send
05:39.28maddog01any idea
05:39.56drmessanoYou said trixbox.. I heard noises and fuzzy light from there
05:40.10maddog01they get saved as tif but not converted to pdf
05:40.24maddog01lol
05:40.58drmessanoKerry has a hitman looking for me, I would think
05:41.07maddog01i know but i liked the trixbox asterisk scripts
05:41.11drmessanoGPLassassin
05:41.56maddog01i ment trixbox aastra scripts
05:42.06drmessanoTrixbox blows harder than an air pump for the Charlie Brown float at the Macy's Thanksgiving Day Parade
05:43.14ZippomanOk so I need help...I have an asterisk box and I am going to use this line so my cellphones can dial in to the asterisks DID and then have a voice prompt on it asking for a telephone number to call next...I looked into this one extension type thing that had the same idea but it was for entering caller id so you could spoof.... then asking for the telephone number to dial out...this does not work...when i call the phone I just get t
05:43.15SlicerDicertrixbox is ET
05:44.04maddog01whats ET??
05:44.26*** join/#asterisk sah-work (n=Bawbatos@adsl-76-236-67-190.dsl.pltn13.sbcglobal.net)
05:44.30mchouEntertainment tonight?
05:44.44maddog01us canadian arent down with the mexican lingo
05:44.46maddog01lol
05:45.38maddog01Zippoman: look into inward system access
05:46.20SlicerDicermaddog01: ET, Phone Home!
05:46.26maddog01lol
05:46.30SlicerDicerlol
05:46.55maddog01SlicerDicer: did you find anyphones you like
05:47.07SlicerDicerwhat at the site you linked?
05:47.16SlicerDicerthose dect phones are out of control man.. thats all I can say
05:47.16maddog01yes
05:47.24maddog01lol
05:47.31SlicerDicerwtf is with USA not having phones like that?
05:47.45SlicerDicerI mean the one that vtech offers for USA is like a FORD...
05:47.45mchoucause USA is sane
05:47.56SlicerDicerfucker only rolls downhill ;-)
05:48.03maddog01North america is always behind
05:48.05SlicerDicerlike the Taurus sorta that garbage car
05:48.06Zippomani looked into the inward system access didnt have much info and all pretty much has sip.conf
05:48.34jayteepeace out!
05:49.00drmessanomchou: Indeed
05:49.04SlicerDicermchou: the dect phones would not be so bad I guess if they were as cheap as normal models
05:49.22mchouSlicerDicer: what happens when your phone batteries no longer take charge?
05:49.25SlicerDicerlike the set I got from costco if they were cheap like that for all the handsets and shit. I could justify it
05:49.37SlicerDicermchou: well with vtech you can buy new batteries
05:49.39SlicerDicerthey are not expensive
05:49.44drmessanomchou: You need to find a battery for an uncommon niche cordless phone, DUH
05:49.56SlicerDicerif they are NMIH beat them with a hammer
05:50.01SlicerDicershort the connections
05:50.05SlicerDicerwill give you a bit more life
05:50.09SlicerDicerI keep doing that with my drill lol
05:50.13maddog01Direct Inward System Access: its part of free pbx
05:50.14mchouyou get a regular dect phone, when batteries die you buy new phone (cause batteries alone cost same as new phone)
05:50.23SlicerDiceryeah
05:50.26drmessanomchou: Exactly
05:50.30SlicerDicermchou: I was saying if they were sane
05:50.40SlicerDicerand offered stuff like the analog variants
05:50.43drmessanoSlicerDicer: THATS THE WHOLE POINT.. ITS A DUMB IDEA
05:50.54SlicerDicerwell it came from europe
05:50.56SlicerDicerwhat do you expect
05:51.23SlicerDicer(formerly Digital European Cordless Telephone)
05:51.35drmessanoMy 5.8GHZ $25 GE Wal Mart phone on a PAP2 works great when I need cordless
05:51.45maddog01Zippoman: DISA for short
05:51.52drmessanoSlicerDicer: I dont mean DECT
05:52.12SlicerDicerdrmessano: thats what I was talking about..
05:53.09maddog01vtech is the lower form of consumer phone there is. what do you want you get what you pay for.
05:53.19drmessanoSlicerDicer: We are talking about this stupid idea of buying a VoIP DECT phone that not only has a SIP UA that will see little real world testing and likely never a firmware update, but a handset that will be nearly impossible to find batteries for, and you're locked into a $100 cordless phone long after than generations tech has died and theres something newer/better that all the other kids are using on now $18 phones
05:54.21mchouin fact almost all consumer cordless VoiP phones have died on the vine
05:54.53mchoujust too many issues
05:55.09drmessanoToo much lock-in as well.. Which will always keep that market small
05:56.09mchouplus most consumer phones arent easily hackable :)
05:56.40mchoufirmware hacking is fun
05:56.51maddog01the avg. joe is not ready for cordless voip phone but for someone with a tech savy why not
05:57.03drmessanomaddog01: For all the mentioned reasons
05:57.07drmessanoIts a BAD idea
05:57.34maddog01that's becuse your hung up on analog
05:57.38maddog01let it go
05:57.40drmessanoHas nothing to do with it
05:57.44drmessanoYoure not reading
05:57.47drmessanoIm not hung on analog
05:57.58maddog01just like tv signals in feb 19.
05:58.06maddog01analog is dying
05:58.06drmessanoIm hung on not putting all my eggs in one basket like a fucking moron
05:58.14mchoumaddog01: how do you overcome the battery and firmware issues? :)
05:58.21drmessanoBuying some $120 phone with a SIP agent that will never see a firmware update
05:58.30drmessanoThat less than 1000 people are gonna buy
05:58.44drmessanothat I will get HALF the life out of than I should due to quality issues
05:58.58maddog01buy a phone from a company like seimens or panasonic
05:58.59drmessanoGo buy a TV/DVD combo if you like all-in-one so much
05:59.13drmessanoBTW, when the DVD player starts skipping, enjoy the TV
05:59.25drmessanoSeimens or panasonic doesnt mean shit to VoIP
05:59.26maddog01lol
05:59.42maddog01means more then vtech
06:00.08drmessanoYoure a trixbox user.. why is this convo even relevant?
06:00.27drmessanoSeriously.. youre arguing about Siemens and Panasonic phones and clutching to Trixbox
06:00.51drmessanoMake me stop LOLing
06:01.23drmessanoBuying a DECT VoIP phone is about as lame as buying a Skype phone
06:01.35maddog01not clutching im in the process of installing asterisk from scrach on cent os
06:01.53maddog01trixbox was a start point
06:03.34drmessanoVoIP DECT phones make no sense economically.. An ATA with your choice of Cordless gives you the ability to swap the phone out whenever you want/need, gives you a PROVEN SIP User Agent that actually HAS firmware updates, that is supported on every platform out there, and is also much cheaper
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06:03.39drmessanoIts a win, win, win
06:04.08orkidwell, where's that >5.1.7 update for spa-3102!
06:04.09orkidarg
06:04.15orkid(for the double hook flash problem :P
06:04.20mchoulol
06:04.34drmessano"How can I make my Siemens FUFJGN-8078475 work on Asterisk?  Help me config?"
06:04.38drmessano^^^^ LAFF
06:05.23mchouorkid: gotta hack the firmware for that
06:05.27drmessano"Oh, theres a bug in that firmware thats been well documents on Voxilla.  The new Siemens fixed it, but they dont do firmware updates"
06:05.35drmessanodoucmented
06:05.38mchouorkid: or rever to 3.x
06:05.47mchourevert*
06:06.02drmessanoThats what I want.. Lemme go buy a new VoIP DECT phone for a bug fix
06:06.12orkidmchou: is there a webpage that describes how to hack it up right?
06:06.41mchouorkid: please.  firmware hacking is fine art
06:06.59mchouorkid: nobody gonna put it up on web page :)
06:07.05drmessanomchou: So youre saying I need Word instead of Notepad?
06:07.05orkidwhy not :P
06:07.18orkidmchou: are there hacked up firmwares available for the 3102?
06:07.33drmessanoheh
06:07.41mchouorkid: disassemble
06:08.16orkid...
06:08.19drmessanomchou: Just a link to the hacked firmware please.. I cant code, but I wanna tell people I am using hacked firmware and be cool
06:08.28drmessanomchou: KTHX
06:08.32denonata+cordless is always going to have a crappy interface
06:08.41denonand you'll be jackin around with flash and #
06:08.46drmessanoSo are most DECT VoIP phones
06:08.57orkiddrmessano: not everyone is as 'cool' as you
06:08.58denonwell, I guess I was thinking of a native sip phone
06:08.58mchoudrmessano: spa-3102 has real double hook flash bug
06:09.00drmessanoThey're barely more than that
06:09.08denonwifi sip
06:09.20mchoulol
06:09.21denonof course, they're lacking in other ways ..
06:09.23drmessanoThis isnt about Wifi SIP phones
06:09.31denonbut they have a consistent interface with other sip phones people are used to
06:09.33mchouthat's even eorse than VOIP DECT
06:09.39mchouworse*
06:10.12orkidmchou: have you ever heard of anyone successfully hacking in double-flash into 5.1.7? which is the last firmware that works with double-flash, do you know?
06:10.43mchouorkid: 3.x, cant remember exact version
06:10.59mchounot posted on linksys official web site
06:11.13drmessanoOther than IP configuration, my $25 cordless GE phone as a single line phone is not missing one feature my desktop VoIP phones have interface-wise.. except for a VM specific button, but stored memory 1 does that too
06:11.28orkidonly one in the 3.1.x i know of is 3.1.2
06:11.45mchouorkid: then that must be it
06:12.04denondrmessano: attended xfer, blind xfer, sip hold ...
06:12.19mchoudenon: that's ALL supported
06:12.23orkid... i was actually going to get a VOIP DECT.. siemens A580IP.. what are you guys' issues with VOIP DECT?
06:12.24drmessanodenon: I use the same feature codes
06:12.35drmessanodenon: No loss there
06:12.40denonbut no dedicated button for em
06:12.52drmessanoMy desktop phones dont have it
06:13.02denonoh, then you have crappy desktop phones :)
06:13.11drmessanoPolycoms are shit, I agree
06:13.15drmessanoAs are the Linksys
06:13.16mchoulol
06:13.22drmessanoYep, youre right
06:13.30mchouthis ia great
06:13.31drmessanoWhat a dumb fucking conversation
06:13.34drmessanoSeriously
06:13.42denonI dont get it .. since when don't your linksys phones have a xfer button?
06:13.50denonsoft buttons count as buttons
06:14.38drmessanoGenerally, I dont use the soft buttons
06:14.58denonwell, ok, Hold then ..
06:15.06denonyour linksys has a big huge honkin hold button
06:15.17drmessanoOk
06:15.25mchouso does my analog handset
06:15.37mchou(cordeless)
06:15.40denonmchou: a hold button that will trigger pbx music .. not just beep-beep
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06:16.03orkiddenon is pro VOIP DECT, mcho/drmessano anti?
06:16.08drmessanoWell I give up.. I guess a Diahatsu VoIP DECT phone is better
06:16.09denonusers always get confused by flash for hold/xfer/etc .. they always lost more calls
06:16.15drmessanoWell worth it
06:16.38orkidhello?
06:16.46mchouorkid: I have Desktop sip phones.  thaes up way too much desk space
06:16.46drmessanoAs is a 2.4GHZ VoIP cordless
06:16.47denonorkid: I'm not pro anything, I'm just mentioning the fact that native sip phones do have their advantages, such as voip sip, not necessarily dect, unless they extend the sip featureset to the handset
06:16.53mchoutakes*
06:16.55drmessanoWhich is still useful 2 years alter
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06:17.04drmessanoYep..
06:17.29drmessanoI bet you can get a good deal on used 2.4GHZ VoIP Cordless phones on eBay
06:17.33orkiddenon: do you know of any 'decent' ones w/o major bugs? or are most still buggy
06:17.38mchouorkid: the only strike against analog handests are crappy speakerphones
06:17.43drmessanoReal cheap.. Paid $250, probably get em for $125
06:17.47mchouhandsets*
06:18.04denonorkid: that seems to vary a lot on user
06:18.30orkidthere are lots of crappy voip 2.4ghz phoens on ebay.. they all strike me as real junky, not nice buttons, numbers that ruboff.. etc. i was thinking about siemens gigaset, but i never had one really. only used a few.. they aren't sold in canada
06:18.46drmessanoorkid: The point went WAY over your head
06:18.53drmessanoLet me use smaller thoughts
06:19.00orkidmaybe.... is it your usual sarcasm?
06:19.04drmessanoNo
06:19.25drmessanoLets go back two years and have this convo about some great 2.4GHZ Siemens DECT VoIP phones
06:19.37orkid...
06:19.41drmessanoNow lets go forward two years and look back at my awesome $175 purchase and how its doing
06:19.46drmessanoerrr
06:19.49orkidwhy not just get to the point?
06:19.52drmessanoLets go back two years and have this convo about some great 2.4GHZ Siemens Cordles VoIP phones
06:19.55drmessanoI AM
06:19.57drmessanoFUCKING LISTEN
06:20.02orkidwhoa
06:20.09orkidok
06:20.10orkid...
06:20.26drmessanoLets go back two years and have this convo about some great 2.4GHZ Siemens Cordless VoIP phones and flash forward to today about how useful they are now
06:20.36drmessanoand how much my $175 purchase was worth
06:21.12drmessano^^^^^^ Obsolete waste of money thanks to two years of technological advances
06:21.34drmessanoWhat made it obsolete?
06:21.37drmessanoNot the SIP agent
06:21.45drmessanoThe RF technology
06:22.06orkidi think i still use a 900MHz analog cordless, we've had it for almost 10 years?
06:22.10orkidand it's great.
06:22.28drmessanoYou're throwing away the $50 part of the $175 phone because a 2.4GHZ phone is useless
06:22.28denon900's only great now that everyone is off it ;)
06:22.43drmessanoPoint >>> **** You're throwing away the $50 part of the $175 phone because a 2.4GHZ phone is useless ****
06:22.49orkidi actually was going to think that SIP/VOIP is too mocing too fast/much for the voip dects to keep up.. but i guess that's not what you were trying to say
06:23.04drmessanoIT IS MY POINT
06:23.05orkidwhat's wrong with DECT though? are there too many users?
06:23.08drmessanoNo
06:23.43orkidso?
06:23.45drmessanoBut in two years, when DECT 8.0 is out and you want that 2 mile coverage and ePenis bragging rights, you gotta toss all that out the door
06:24.11orkidoh... i'm ok with a phone that does sip/voip well, whatever the rf and distance..
06:24.11drmessanoAll because the $50 part of the device is outdated
06:24.21drmessano....
06:24.24drmessanoForget it
06:24.27denondrmessano: you're forgetting that to some business users, they'd be replacing in 2 years anyway
06:24.28orkidthe 900 is good enough really, but the double-hook-flash doesn't work, and having multi-handsets would be nice
06:24.31denonconstantly rotating
06:24.38denonie: warehouse staff are hard on hardware
06:24.42denonbut need lots of toys, often
06:25.01drmessanodenon: Bullshit.. Do you have any idea how long people keep phones?  YEARS
06:25.08orkid10 years here :)
06:25.13denonpersonal phones, yes
06:25.16drmessanoNo
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06:25.19drmessanoBUSINESS PHONES
06:25.41drmessanoPeople dont buy new PBX's and handsets every two years.. and often those purchases go hand in hand
06:25.41orkidso... any thoughts on 'good' voip/dect phones available now? is gigaset normally good? :)
06:25.41denondrmessano: we've all been in this business a long time, each vertical is different
06:25.50drmessanoPeople go 6, 8, 10, 15 years
06:26.02orkid.... for home use
06:26.15denonportable phones get replaced much more often in ruggid environments
06:26.16orkidor is that ccc dect 'crack' really bad
06:27.01denonanyway, whatever .. I'm not making a holy war out of this, and at this hour, you usually get into that mood
06:27.08denonso I think I'll sign out
06:27.17drmessanodenon: All the more reason not to waste $150 on a $25 quality phone because its got a SIP UA built in
06:27.39drmessanoOh.. Classic
06:27.50orkiddenon: any thoughts before you go?
06:27.55orkid:)
06:28.14drmessanoIf you dont want to get into a holy war, dont be an IRC douchebag and make a personal attack like "Well, we all know how you are" and run off
06:28.25drmessanoI didnt detect any tone here, but whatever
06:28.54drmessanoIf you cant take a counterpoint without being a drama queen, dont discuss
06:29.00drmessanoAnwyay
06:31.04orkiddude really, why not just talk about it w/o any emo/drama getting in the way
06:31.11drmessanoThere wasnt any
06:31.50drmessano"I'm not making a holy war out of this, and at this hour, you usually get into that mood"  <-- I didnt make any of this personal.. But denon always gets to a point where if you disagree with him enough, he does
06:31.52drmessanoOh wait..
06:31.55*** join/#asterisk thx2000 (n=bob@ip68-101-126-92.oc.oc.cox.net)
06:32.02drmessanoBut again.. moving on
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06:32.53orkidjust another way of saying "i dont think this is going anywhere, we're going in circles" i think, to be more polite
06:32.59drmessanoHA
06:33.08orkidanyway, you have a voip dect phone?
06:33.19drmessanoNo, it was quite clear what he said.. as I expect from him..
06:33.21orkidwhich ones from your experience are buggy, and which ones are not? any comments on the gigaset?
06:33.41drmessano"You usually get in that mood" is definitely not an amicable statement of mutual disagreement
06:33.44drmessanoBut whatever
06:33.45[TK]D-Fenderorkid: General word is that Seimens DECT is pretty decent
06:34.41maddog01it;s way to late for another round of this shit. good night.
06:35.13orkid[TK]D-Fender: thanks. i'll look into it once more, and maybe get one
06:35.39drmessanoI hear Linksys makes some nice SIP Wifi phones too
06:36.20orkiddo they allow for non VOIP handsets to still use the base station through some compatibility mode? since i'm guessing they might not have all the features (i think some of the siemens handsets have a way to choose voip or pstn directly)
06:36.39orkidi actually heard good things about dlink wifi phones
06:37.03[TK]D-Fenderorkid: WiFi = shit.  Avoid unless you have little other practical choice
06:37.13orkidbut am weary of the pwer consumption/emission.. even though i dont know much about the differences between dect/wifi emissions, but i think dect is smaller
06:37.22[TK]D-Fenderorkid: If you are dealing with single site users, never touch Wifi
06:37.35[TK]D-Fenderorkid: Wifi batter life is garbage.
06:37.37drmessanoFor $250?  HA
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06:40.19drmessanoTHose Dlinks are pretty, but I am not a Dlink person.. and the price is nuts
06:40.30[TK]D-Fenderok, checkout time.
06:40.30drmessanoNevermind SIP Wifi in general being suck
06:40.31[TK]D-Fenderlater all
06:40.35drmessanolater
06:41.11drmessanoIm gonna go too before someone accuses me of being a bastard IRC asshole troll because I dont like Green M&M's
06:41.19drmessano**poof**
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06:46.13orkidlets stick to what we know :)
06:46.34RugIs there a FAQ I can read to find some basic answers?  I've search voip-info.org with no success.
06:47.39RugOr, could anybody tell me if an X100P card will work with basci analog phones?  I want asterisk to act as a basic answering machine, nothing fancy, no VOIP
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06:48.03rob0~x100p
06:48.04jbotwell, x100p is an obsolete card.  You don't want to bother trying to make it (or any of the "digium compatible" clones) work.  Get a TDM01B, and you will save your sanity, your hair, and countless other things.
06:48.20Rugcool, thanks
06:49.12RugWill the above card work or shall I research it first?
06:49.26rob0It can be done, but it really is a piece of junk.
06:49.47RugThe TDM01B is junk?
06:50.02rob0nono I thought you meant the x100p
06:50.34Rugyou convinced me to ignore the X100P.  Moving on, now
06:50.37Rug=)
06:50.40orkidthere are also some clones, like openvox
06:51.43RugI don't really care about cost, I am doing the contracting on this project.
06:53.05rob0oh. hmmm ... I don't know what to recommend other than to avoid the x100p
06:53.16Rugthanks, that is good news.
06:53.29rob0I think there's a newer Digium card, newer than the TDM cards.
06:54.23RugIt looks like the TDM01B requires an FXO digital phone, is that correct?
06:54.39orkidfxo is analog
06:54.42orkidfxs is analog
06:54.47orkidt1/e1 is digital
06:54.47Rugack, sorry
06:55.47RugPOTS -> TDM01B -> Analog desk phone (right?)
06:56.11Rugif I need more lines, just get a card with more FXO ports?
06:56.30Rug** Scratch that
06:56.48rob0TDMxyB .... x is number of FXS, y is number of FXO ports
06:56.49RugIf I need more desk-extensions, get a card with more FXO ports?
06:57.04rob0nope, your extensions will be on FXS.
06:57.35rob0~fxo
06:57.36jbotforeign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo
06:57.39rob0~fxs
06:57.40jbotfrom memory, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx
06:59.04RugIf the customer wants to upgrade in the future, can I add a second card, or should I get a card with N+1 ports?
07:00.21RugWould you suggest Debian or Ubuntu for the server OS?
07:01.08rob0I would suggest that you choose the distro you are most comfortable with.
07:01.54RugAre the debian dependancies 'new-enough' ?
07:02.07rob0<== not a Debian user
07:02.13Rugahh
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07:04.12RugIf the customer wants to upgrade in the future, can I add a second card, or should I get a card with N+1 ports?
07:04.57clone_junoi have a problem
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07:06.09clone_junoNow ds3000p is being produced ?
07:07.06*** join/#asterisk Assid (n=assid@unaffiliated/assid)
07:07.21Assidyo
07:07.33Assidanyone here tinkered with the likes of cisco 7960
07:07.42Assidi cant seem to get it to pick up the sip firmware
07:08.07rob0The cards are modular, and you can indeed have more than one, but I'm not sure that's going to be cost-effective. At some point you might want to switch from analog to digital, with IP phones.
07:09.02RugRight now there is 1 line and 1 phone.   Sometime in the future it might grow?
07:09.57RugIt's a Doctors office.   He _just_ wants a fancy answering machine with "unlimited" storage.
07:10.02clone_junosorry, ds300p, is that now continue producing?
07:10.51rob0Cheapest (not necessarily best, but it works) way to grow is to add ATA's for additional extensions.
07:10.58rob0~ata
07:10.58jbotata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
07:11.08Rugok thanks
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07:15.28clone_junohow many signalling link  support on asterisk+libss7?
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07:36.44rjaredanybody here set up iax? im trying to connect t two asterisk boxes but keep getting Rejected connect attempt from xxx
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07:42.59DaPrivateerztdummy on FreeBSD... anyone tell me why I can't seem to find how to install it / make it work?
07:44.45drmessanohttp://tfot.leifmadsen.com/ch03s04.html
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07:45.02Corydon76-digDaPrivateer: are you using the FreeBSD port?
07:45.09DaPrivateeraffirmative
07:45.41Corydon76-digDaPrivateer: http://www.mercenary.ca/articles/zaptel_asterisk.php
07:46.02DaPrivateerhrm, ill give that a shot -- thanks
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09:29.34KOCATEPEhi all
09:29.43KOCATEPEanybody from Turkey ??
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09:31.27KOCATEPEanybody from Turkey ??
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09:50.41Daejeodoes switchvox Encrypt the  Partitions during Installation?
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10:03.03kinnazhas anyone installed linux on audiocodes mediant 1000 ?
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10:10.28ibm2I want to know how to  retrieve the status of a call
10:14.49DigitalIronyAsteriskdocs.org
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10:36.51Daejeohello DI
10:37.03Daejeoare you still awake?
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10:39.42aiksa[LV]tzafrir_laptop: hay man, found the problem - i had added the option to configuration sections
10:40.02aiksa[LV]to the wrong configuration section* . my bad
10:40.10DigitalIronyDaejeo sup
10:40.14tzafrir_laptop:-)
10:40.56Daejeodoes switchvox Encrypt the Partitions during Installation?
10:41.34DigitalIronyDaejeo: I am not sure. I don't really work with gui's much
10:41.52DigitalIronyDaejeo: I would guess not, but Im probably wrong :P
10:45.44aiksa[LV]btw, I was wondering why even there is jitterbuffer option in dahdi configuration?
10:46.28aiksa[LV]PRIs shouldnt have any jitter at all. and it is hard to imagine where would jitter show up on analog interfaces
10:46.36aiksa[LV]or am I missing something here?
10:47.30angryuseraiksa maybe during transcoding to sip, imagine server overload
10:47.50tzafrir_laptopthe jitter is on voip
10:47.50angryuserto another codec*
10:48.04aiksa[LV]tzafrir_laptop: i do understand that
10:48.10tzafrir_laptopand hence you may need a jitter buffer on the pstn/voip border
10:48.22aiksa[LV]thats why I was wondering why receiving side of sahdi should have jitterbuffer at all
10:48.31aiksa[LV]tzafrir_laptop: oh i see
10:48.46aiksa[LV]if the far side had jitter due to transcoding, network, etc.
10:49.20aiksa[LV]then to be able to succesfully deal with that, jb should be used.
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11:59.05SHZOLhi
11:59.44SHZOLhow to implmnt Hot-desking
11:59.48SHZOLany idea ?
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12:01.44Assidso anyone here managed to get cisco 7960 working? with sip?
12:01.55Assidi cant seem to get the firmware to upgrade.. im just lost
12:03.00mort_gibAssid: Polycom and Snom are nice phones to :-)
12:03.20Assidmort_gib: i know i mostly deploy polycoms
12:03.40*** join/#asterisk brian (n=brian@unaffiliated/brian)
12:03.41Assidbut my stupid friend has a cisco 7960 and asked me to get it up on  my system
12:04.02mort_gibYou need the SIP firmware from Cisco don't you??
12:04.47Assidi got that.. i just cant get this phone to get it
12:06.19mort_gibtftp server setup and all??
12:06.25Assidyes
12:06.39mort_gibStrange....
12:06.51Assidmaybe i didnt configure it right.. the files
12:07.23mort_gibWell you need to load the right firmware first, but I'm not a Cisco phone wizard...
12:07.53Assidim trying even 6.0->6.3
12:07.57Assiddoesnt wanna work :(
12:10.27mort_gibWhy is that we don't turn around and tell our "clients" that it's unsupported??
12:11.04mort_gibCommercial support is all too prepared to do so!
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12:16.55mort_gibI have an issue with caller id on a Sangoma A200 card... I get : callerid.c: No start bit found in fsk data.
12:17.05mort_gibin /var/log/asterisk/messages
12:17.37mort_gibI have usecallerid=yes in zaptel.conf
12:19.30*** join/#asterisk Assid (n=assid@unaffiliated/assid)
12:19.55Assidmort_gib: cause we're idiots
12:20.05Assidand they keep saying hey its supported.. its cisco.. yadda yadda
12:20.13Assidand we want the business
12:20.26SHZOLhi
12:20.30SHZOLi want to do  hot-desking
12:20.31SHZOLhow
12:20.31mort_gibTrue...
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12:20.42mort_gibSHZOL: Budget??
12:20.55SHZOLwat u mean ?
12:22.09mort_gibYou are talking about call center solutions -right??
12:23.11mort_gibSHZOL: you want to have a look at "agents" and call queues
12:23.44mort_gib-But there are some commercial solutions that builds on Asterisk that are good value for money....
12:24.01mort_gib-Mind you, it ALL depends on the size and type of your install
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12:40.53aiksa[LV]hmm, i sometimes see the CPU % usage in top for asterisk to jump up to 9999% what could cause it?
12:41.05aiksa[LV]nothing special happens at that time
12:41.12aiksa[LV]just normal call flow
12:41.17aiksa[LV]nothing extraordinary
12:42.00aiksa[LV]aversion 1.4.22
12:43.33SHZOLi want to do  hot-desking
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12:56.16orTixGot this problem, i got an cisco ip phone 7940 here (same was 7980) I got some new firmware from cisco, however the phone loads the firmware correctly from my tftp server butt it keeps reloading itself all the time.. ? anyone know howto solve this problem :X
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13:00.14phpboySHZOL: Hot agents?
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13:07.39*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
13:09.57SHZOL+phpboy: what u mean by hot agent
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13:34.05SHZOLi want to do  hot-desking
13:34.18SHZOLheloo
13:34.56coppicethe hot agents usually find better paying jobs
13:35.27SHZOLcoppice: what u mean
13:36.00beekSHZOL: Read the book in the section under FUNC_ODBC.   There's a whole application written to do hot desking.
13:36.02beek~book
13:36.03jboti heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
13:36.13beekSHZOL: ^^^^^^^^^^^^^^^^^^^^^^
13:37.04SHZOLcoppice: thanks
13:37.14beekSHZOL: you're welcome.
13:39.05Rico29hi
13:39.19Rico29I've got a problem with a perl AGI that I really don't understand :
13:39.31SHZOLcoppice: thereis no sample and eyes way of doing it ?
13:39.42orTixGot this problem, i got an cisco ip phone 7940 here (same was 7980) I got some new firmware from cisco, however the phone loads the firmware correctly from my tftp server butt it keeps reloading itself all the time.. ? anyone know howto solve this problem :X
13:40.05[TK]D-FenderSHZOL: Maybe somebody blogged some sample of this, but good luck finding it.
13:40.14Rico29this line works : $AGI->stream_file($country . "/custom-enter_pwd");
13:40.14Rico29but this one doesn't : my $SIP_passwd =$AGI->get_data($country . "/custom-enter_pwd");
13:40.26Rico29does anybody knows why ?
13:40.29[TK]D-FenderSHZOL: And you need to be very clear about your definition of "hot desking" and how it pertains to your dialplan
13:40.31SHZOL[TK]D-Fender: can u help me doing this project
13:41.25SHZOL[TK]D-Fender: i have more then 300 ip phone and 300 emp in our office, i want any emp can login from any ip phone, thats what i need.
13:42.04SHZOL[TK]D-Fender: also he can ably to make outgoing call and incoming calls.
13:43.30[TK]D-FenderSHZOL: You'll need to make an exten to login toa phone that will set a value based on the emp that is logging in, and to associate to that phone device.  Then in every exten that they can diaal out through it will first check who is logged into that device and act accordingly.
13:43.55[TK]D-FenderSHZOL: And then of course an exten so they can "log out.
13:45.13SHZOL[TK]D-Fender: i found what i want in this web site, but is not full, ducomantions, http://etel.wiki.oreilly.com/wiki/index.php?title=Simple_Hot-desking&redirect=no
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13:47.19telnettechgood morning. I have another question. I have zaptel workng on a 64bit Linus server. with asterisk 1.2.28. It is requiring me to have ztdummy turned on for sound files to work. But i need to have ztdummy off so that my redfone device works properly.....any suggestions?
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13:47.46[TK]D-FenderSHZOL: tahts a pretty good sample
13:48.11[TK]D-FenderSHZOL: You need to master the dialplan.  Keep reading and keep trying stuff.
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13:51.08aiksa[LV]is there a way to catch transfer event in pre 1.6 asterisk?
13:51.57aiksa[LV]any transfer - through SIP messages as well as feature codes
13:52.08aiksa[LV]through AMI of course
13:52.41Rico29anyone for my probleme ? please
13:53.32aiksa[LV]Rico29: sorry, perl is not my field
13:53.43Rico29ok
13:54.02phpboyLovely, asterisk crashes when I introduce a lot of calls via IAX2 :(
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13:54.58BBHossphpboy: with what error?
13:55.13[TK]D-FenderRico29: You don't seem to be showing us the error
13:56.00Rico29ok, 2sec [TK]D-Fender , i'm pastebining it all
13:56.30SHZOL[TK]D-Fender: u sow the link i give it you, how many database is there ?
13:56.56[TK]D-FenderSHZOL: HUH!?
13:57.04BBHossheh
13:57.14aiksa[LV]Is there a way I could take a closer look at the internals of the asterisk - its the same old CPU%9999 issue i am having. Is there a way to have a more detailed split by modules?
13:57.20timeshellAnyone know the current build number of asterisk-gui?
13:57.56timeshellAnd whether it works better with DAHDI hardware detection?
13:58.02aiksa[LV]but without introducing profiler to the whole scene :P
13:58.04*** join/#asterisk brian (n=brian@unaffiliated/brian)
13:58.44SHZOL[TK]D-Fender: how many databases in this sample http://etel.wiki.oreilly.com/wiki/index.php?title=Simple_Hot-desking&redirect=no
13:58.54SHZOL[TK]D-Fender: can u tell me pls
13:59.10[TK]D-FenderSHZOL: What the hell does "how many database" mean?
13:59.23[TK]D-FenderSHZOL: *1*
13:59.32[TK]D-FenderSHZOL: This just uses AstDB
13:59.49[TK]D-FenderSHZOL: "core show function DB" <- go read up on this
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14:02.16SHZOL[TK]D-Fender: ok
14:05.52SHZOL[TK]D-Fender: can u give brief introduction about it pls
14:06.25fileDB is a dialplan function which gives access to the underlying Asterisk database, a berkeley database
14:06.29fileyou can store and retrieve values
14:09.38aiksa[LV]hmm, htop is nice for monitoring processes
14:09.46aiksa[LV]somehow I had not noticed it before
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14:18.34Rico29[TK]D-Fender> http://debian.pastebin.com/m1e58703f
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14:27.35gambler1is it possible for dial application to react on sip messages? ie when I get "address incomplete" which dial translates to busy
14:27.37Rico29[TK]D-Fender> any idea ?
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14:29.06[TK]D-FenderSHZOL: Go read the WIKI and the instructions for DB like I showed you.
14:29.31Boraimorning
14:29.48[TK]D-FenderRico29: Seems to play.  Whats the issue?
14:29.57Rico29doesn't play...
14:30.12Rico29it seems, but it doesn't
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14:30.56BoraiI got the ftp server and all the configs ready now my phone boots from the server.
14:31.03[TK]D-FenderRico29: is that a complete call from beginning to end?
14:31.13Rico29this is one
14:31.18[TK]D-FenderRico29: I don't see CLI output there....
14:31.29Rico29i can pastebin the agi debug
14:31.49[TK]D-FenderBRB
14:31.51Rico29http://debian.pastebin.com/m1e58703f the 5/6th first lines
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14:32.09Rico29what is "BRB" ?
14:32.13Rico29ah ok
14:32.15Rico29be right back
14:32.17Rico29:p
14:32.24Borailol
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14:32.48tzafrir_laptopwaits for someone to ask what lol is
14:33.19DaejeoHow many user extensions can I create in Switchvox Free Edition ?
14:34.22BoraiOMG!
14:34.50*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
14:34.51Borai*is
14:36.04Boraihowever my polycom tells me line 1 (not registered), where sip show peers tells me the peer is online
14:36.13Rico29tzafrir_laptop> huhu
14:36.44troubledanyway to test that wideband codecs are working in my 1.6? Know of any good clients that would work?
14:37.16troubledive tried 2 clients so far with multiple codes that do 16khz, but * doesnt seem to have them, or they arent being used, like speex
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14:39.13Rico29[TK]D-Fender> did you see the pastebin ?
14:39.15coppicemost things supporting wideband support G.722, and * has that
14:39.24Rico29the CLI log is in the 5th first lines
14:39.27[TK]D-FenderRico29: do you see when I came BACK?
14:39.48Rico29euh... yes ?
14:40.00SHZOL[TK]D-Fender: already read, but not understanding, pls give simple world defnations, what is it so i can fllow up with my priject
14:40.01[TK]D-FenderRico29: No PB since then <-
14:40.33[TK]D-FenderSHZOL: read that apps INSTRUCTIONS.  AstDB is fairly well described on the WIKI.
14:40.34[TK]D-Fender~wikis
14:40.35jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
14:40.36[TK]D-Fender^^^^^^^^^
14:40.40Rico29mh, I don't understand everything
14:40.58troubledcoppice: any recommended clients that do 722?
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14:41.23coppicehave you found a wideband client that *doesn't* do G.722?
14:41.38SHZOL[TK]D-Fender: thanks you.
14:41.39troubledwell, i have xlite and wengo atm
14:41.51coppicethey do G.722
14:42.08Rico29ahahaha
14:42.37Rico29$agi->exec(Background...) works, and $agi->get_data(sound) doesn't...... WTF ?!
14:42.44troubledcoppice: well, wengo only has speex/16000 and AMR-WB/1600 listed
14:43.07Boraidoes the polycom extension have to be peer or friend?
14:44.13[TK]D-FenderBorai: peer
14:44.31SHZOL[TK]D-Fender: got it man, Berkeley DB which works very much like the Windows registry. thats it, Right ??
14:44.46[TK]D-FenderSHZOL: a little, yes
14:45.23eppigyTRABAJO
14:45.28troubledcoppice: well, trying xlite, and I noticed that the sip channel said it was using the .slin sound file, but I can only tell its using speex atm
14:46.02coppicewell, wengo became qutecom, and that certainly supports G.722
14:46.09[TK]D-Fendereppigy: YOU go work.
14:46.11Borai[Jan  6 06:45:39] WARNING[17785]: chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission bb467dac-17c92445-6e9a77fe@192.168.1.3 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt.
14:46.21Boraithis is what i get all the time
14:46.34Boraiwhen i try to dial some extension from the phone
14:46.40eppigy8[]
14:47.18troubledcoppice: hmm, im using wengo 2.1.2, guess I grabbed an old fork or something
14:47.39[TK]D-FenderBorai: last we checked you were running a phone behind a remote NAT.  Naturally I don't trust that you set any of this up right.
14:48.04[TK]D-Fender~sipnat
14:48.05jbot[~sipnat] Quick guide on configuring * + 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
14:48.11telnettechok guys i really need some help....I have an issue where my zap channels are not loading and cant figure out what it is trying to tell me......i have included both 'ztcfg -vv' and 'load chan_zap.so' outputs on the pastebin.....please help
14:48.16telnettechhttp://pastebin.com/d7dc2054d
14:48.27coppicetrouble: oh, yeah, xlite is about the one thing which does not support G.722
14:49.01troubledcoppice: i noticed it does wideband speex, but it seems asterisk doesnt?
14:49.02BoraiI run the asterisk server on a public ip
14:49.42[TK]D-Fendertelnettech: Sure looks like you've mixed up settings between T1 & E1
14:49.42Nuggettelnet is eeeeeeevil!
14:49.44troubledcoppice: so is there a way to tell the freq/fidelity of the sip call to verify?
14:49.54[TK]D-FenderBorai: but your PHONE is behind NAT
14:50.09[TK]D-FenderBorai: and of course I still wouldn't trust your firewalling
14:50.13[TK]D-FenderBorai: go read <-
14:50.27telnettechTK: it is supposed to be an ISDN 30.....i have tried to find everything i can but it just doesnt make sense to me
14:50.28Boraiok but shouldn't technically DMZ avoid NAT?
14:50.40coppicetroubled: * doesn't really support wideband right now. it just has G.722 fudged in
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14:51.12troubledcoppice: not even in 1.6?
14:51.31Boraiand im assuming that in my case it is Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk
14:51.37Boraiam i right?
14:52.33[TK]D-FenderBorai: Your peer should have "nat=yes", "canreinvite=no", "qualify=yes", "host=dynamic"
14:52.39troubledcoppice: seems that even the latest qutecom client only does 8000 g722
14:53.14coppicethat is actually wideband. the 8000 is because everyone has to follow a typo in an RFC
14:53.14jaytee[TK]D-Fender, good morning! feelin any better?
14:53.30[TK]D-Fenderjaytee: Yea, I've had this beat since about Sat night
14:53.36troubledcoppice: the only 16000 listed codec is speex
14:53.47jaytee[TK]D-Fender, good to hear. :-)
14:53.50beekGood morning [TK]D-Fender and jackson__
14:53.55troubledcoppice: oh
14:53.59beekGood morning jaytee
14:54.01Boraii did not have qualify=yes let me reload
14:54.07jayteegood morning beek
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14:54.33jayteeeverytime he types something I see Sasha Boren-Cohen's face :-)
14:54.47telnettechTK do you have a good website that explains how to setup an ISDN30
14:55.08[TK]D-Fendertelnettech: www.google.com
14:55.15Borai[TK]D-Fender: thank you very much adding qualify=yes fixed my problem.
14:55.17jayteetelnettech, morning
14:55.20troubledcoppice: looks like it only wants to use gsm *shrug* thanks for trying
14:55.26telnettechmorning jaytee
14:55.42coppicewhat only wants to use GSM?
14:55.58jayteeusually about this time Nugget chimes in
14:56.05troubledcoppice: when i made a call, the "sip show channels" only shows the call as gsm
14:56.41[TK]D-Fendertroubled: that its what it wanted... thats what it SETTLED ON
14:56.48coppicehave you listed G.722 as your preferred codec?
14:56.57telnettechi get about 3000 hits there TK.....i have looked at about 20 or so.....i cant seem to find any info at all.....alot of the hits have to do with issues with asteirsk and BT ISDN30
14:56.58[TK]D-Fendertroubled: Look at SIP debug of the call attempt or you're wasting your time
14:57.22[TK]D-Fendertelnettech: I don't see your configs, do I?
14:57.40telnettechTK: no but i can show you
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14:58.17troubledcoppice: its just a priority list, i tried with 722 at the top and bottom just in case. still using 722 in asterisk either way
14:58.54coppicewell, if you only allow G.722 in *, that should force it :-)
14:59.07troubledya, just gonna try that now. let you know in a sec
14:59.43telnettechTK: here are the config for zaptel and zapata     http://pastebin.com/d909e80d
15:00.26troubledcoppice: ooh, that played the demo-echotest.g722 file, but it was choppy
15:00.40kannanhello , i am editing the SIP<MAC>.cnf for a cisco 7960 to change the phone extension number, the tftp server is also the same as *. After restart the tftpd and the phone, it is still not updated . any ideas what i am missing
15:02.06troubledcoppice: probably just the client cpu lag
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15:05.05[TK]D-Fendertelnettech: And your fonebridge dies when it hits 24 (T1 D-Chan).  Check ITS config
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15:06.06telnettechthe fonebridge programming is fine.....it is the zaptel and zapata......the redfone is just an external T-1 card basically using zaptel to interface with asterisk
15:06.50NoxIn-Hello all, is it possible to call-limit only calls comming from a peer to asterisk but not the calls from asterisk to the peers ?  since the call-limit apply to both, I envisaged to create a type=peer for outgoing calls and a type friend for incoming calls, but type=friend is deprecated so anyone have a clue ?
15:06.53telnettechI verified the fonebridge programming by putting a loopback jack in the rj45 port and it goes green...that tells me the red fone is working fine
15:07.27[TK]D-Fendertelnettech: Doesn't mean it AGREES with *
15:07.37[TK]D-Fendertelnettech: And doesnt' tell me what it thinks it should be doing.
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15:07.49[TK]D-Fendertelnettech: You have reassured NOTHING.
15:07.52telnettechright but that is where the zaptel comes into play
15:08.06telnettechit uses zaptel to communicate
15:08.12telnettechwith asterisk
15:08.33[TK]D-Fendertelnettech: fonebridge needs to be set up for the proper signalling in its OWN config.
15:08.51telnettechok so you want to see the redfone config
15:09.01[TK]D-Fendertelnettech: I have to ask?
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15:11.20telnettechTk no....i just thought that i would tell you that the progrmming is correct cause im geting signalling verified thru the loopback.....here is the pastebin for the redfone.conf
15:11.24telnettechhttp://pastebin.com/d6bd72898
15:11.28doolphhi there
15:11.57[TK]D-Fendertelnettech: Loopback does not tell me it AGREES WITH ASTERISK
15:13.08[TK]D-Fendertelnettech: Ans since when does E1 use ESF, B8ZS?
15:13.24[TK]D-Fendertelnettech: those are T1 typical
15:13.42telnettechspan 2 is not used but redfone requires that you program both all spans even if your not using it
15:13.52telnettechspan 1 is supposed to be the isdn30
15:14.28[TK]D-Fendertelnettech: Go ask Redfone for support with their device....
15:14.46telnettechso you are thinking redfone problem?
15:15.24[TK]D-Fendertelnettech: So far, yes
15:16.37telnettechso the zaptel and zapata are correct for an isdn30
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15:18.47kannancan we config the cisco ip phone thru the TFTP server, or does the .cnf in tftpboot directory get written by the phone?
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15:20.22orTixMy cisco ip phone (7940) keeps loading the new firmware thought TFTP server, does anyone know the problem (google is empty :p)
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15:40.58Nuggeteyes jaytee
15:41.12jayteeyou missed your queue
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15:41.21Nuggetdang!
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15:41.22[TK]D-Fenderhands Nugget a spoon
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15:53.45eppigyhello
15:53.48eppigyi am dave
15:54.42Rico29hello, I am rico :)
15:56.06orTixnobody can awnser my question?:X
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16:00.17telnettechTK: thanks....you wer correct...there were a couple of the settings incorrect on the redfone config
16:00.50telnettechi apologize if i ruffled your feathers there
16:04.07jayteehe doesn't have feathers, he has scales and talons and breathes fire
16:04.58jayteeand to him, you're crunchy and taste good with ketchup
16:04.58jayteeso beware
16:05.06jayteefile, you around? gotta real quick question
16:08.12*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
16:08.37[TK]D-Fenderjaytee: If humans weren't meant to be eaten then why are they made of meat & treasure? :D
16:08.59jaytee[TK]D-Fender, good point!
16:09.41filejaytee: moo
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16:10.50jayteefile, do you think it's a good idea to include the h extension in each subsection of an IVR tree that calls SpeechDestroy()?
16:11.10fileit doesn't really matter
16:11.27Assidgeta
16:11.29Assidheya
16:11.30filethe speech structure itself is associated with the channel and when the channel is hung up it also gets destroyed
16:11.38jayteeah, excellent
16:11.39Assidanyone here worked with a cisco 7960
16:11.47jayteefile thanks! one more?
16:11.51filego ahead
16:12.04Assidi cant seem to get the firmware shifted to sip
16:12.51jayteecan I have one tree handling spanish and one handling english? as long as I'm not trying to activate two different languages in the grammars for one section?
16:13.23fileyou can
16:14.37jayteeok, that's what I thought but wanted some confirmation. I read the docs and thought that's what they meant. I can't have both languages in one single instance though. Cool. Thanks for the help!!!
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16:18.22kannansorry to repeat, howto effect the changes made in SIP<mac>.cnf in root tftpd dir file. The phone's old number still shows up , even after a restart of the Cisco 7960 phone?
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16:18.52kannani have followed the Cisco Pgone Admin guide
16:19.23Assidkannan: you got it to do that? mine looks for SEP<mac>.cnf.xml
16:20.20kannanAssid: the current extensions are working fine
16:20.28kannanSIP<mac>.cnf
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16:21.49Assidnah.. mine looks for sep.. and i cant get it to load the new firmware
16:23.15engagedif i need 10 channels for CSR queue... what is a good VOIP provider?
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16:26.44[TK]D-Fender~itsplist-us
16:26.45jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
16:27.06[TK]D-Fenderengaged: les.net & vitelity seem to be pretty decent
16:27.17[TK]D-Fenderengaged: as I mentioned we nned to rate shop.
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16:31.16nerdygirl_ellieHello All.  Is it possible to accept a pound/hash key (#) with the Asterisk Read() command in extensions.conf?  If so, how do I match it with a GotoIf??
16:31.24nerdygirl_ellieThanks in Advance, Ellie
16:31.49nerdygirl_ellie(Sorry about the Double-??).
16:33.05[TK]D-Fendernerdygirl_ellie: No
16:33.33[TK]D-Fendernerdygirl_ellie: You'll have to make a custom IVR to collect digits
16:34.20nerdygirl_ellieThat is unfortunate.
16:34.27jameswfwhooop there i is
16:35.04nerdygirl_ellieDoes that mean a separate application outside of extensions.conf, or...?  Any pointers would be appreciated.
16:35.45nerdygirl_ellie(I think the lazy solution is to re-record the greeting and use 9 instead of #.. :) )
16:37.01nerdygirl_ellieThank you for the help.
16:37.07[TK]D-Fendernerdygirl_ellie: No, as I said, a special IVR to collect your input.  All dialplan.
16:38.08anonymouz666nerdygirl_ellie: better yet, try to use read() and at each digit you read you repeat to the user. ;)
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16:40.57Assidokay cisco just made it to my list of DO NOT buy ip phones
16:41.12Assidi cant configure this thing.. and every guide up there says do this.. do that..
16:41.13dmzwhy?
16:41.28dmzah yeah configuring is a pain, but they do work when setup is done :)
16:41.48Assidso do polycoms.. and they werent that hard to configure
16:42.03Assidhas to thank [TK]D-Fender for suggesting polys..
16:42.12jameswfI like AASTRA
16:42.24Assidi wanna make this stupid thing move to sip..
16:42.34[TK]D-FenderAssid: Funny you say that after going against advices and hitting brick walls..
16:42.43[TK]D-FenderAssid: Hind-sight is 20/20
16:43.00Assidgoing against advices? me ?
16:43.36Assidall my other phones are either ATA's or polycoms
16:43.55Rico29i think I really have a problem with asterisk::agi
16:44.08Rico29I've installed an all clean Asterisk
16:44.19Assidbut i need to get a friend onto my pbx.. so.. trying to whack my brains on this
16:44.21Rico29i'v made the simpliest agi I can
16:44.26Rico29and it's still not working
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16:44.36RypPnAssid you need help getting sccp going?
16:44.57AssidRypPn: i wanna move it to sip.. i like sip.. i want it to run on that instead of sccp if possible
16:45.03dmzi wish i could try a polycom, i got my cisco for free so i "had" to get it working :)
16:45.10Assidgiven the option" exists"
16:45.30RypPnAssid I use sccp with mine, If you change your mind let me know :)
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16:45.46Rico29I erlly need help
16:45.48Rico29really
16:45.56Assidhrmm i wouldnt mind learning... nevertheless
16:46.11Assidso sure
16:46.23Assidone sec
16:46.25Assidphone
16:46.50RypPnpm me if you like and we can look at it :)
16:47.40Rico29if someone can take a look : http://debian.pastebin.com/m558c0ca3
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16:49.02Linuturkif I call dialplan reload, will that drop any existing calls?
16:49.16[TK]D-FenderRico29: ANSWER the damn call first
16:49.17Rico29Linuturk> no
16:49.27Rico29TK > ok
16:50.06Rico29TK > Doesn't change anything
16:50.13kannandmz, do you config the phones sip params by tftp?
16:50.18kannanor on the phone itself
16:50.21dmzyeah tftp
16:50.28Linuturko, well that command doesn't seem to exist in 1.2x     thanks anyway Rico29
16:50.28Rico29I've found a solution, but it looks stupid for me
16:50.30Linuturk:)
16:50.35kannani will be glad to learn how to modify a phone setting .cnf in tftp
16:50.46Rico29I have to put a "sleep" after the stream_file
16:50.48kannani edited the SIP<mac> and thats it?
16:50.53Rico29then I can hear the sound
16:50.58kannanon restart the phone shld take the new params?
16:51.17Rico29kannan> depends on the phone
16:51.21Rico29of
16:51.31kannancisco 7960
16:51.42kannani have 10 working
16:51.48kannanbut cant modify the setting
16:51.59kannani want to change the exten numbers
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16:52.15dmzi've only setup the lineN_name, lineN_displayname, lineN_shortname, lineN_authname, lineN_password
16:52.20dmzwhat other settings do you want to change?
16:52.22kannanme ttoo
16:52.39kannanonly the lineX_name and passwd
16:52.41dmzinbound or outbound exten numbers?
16:52.42kannanauthname
16:53.16dmzsetup different sip names/details if you want different outbound (or inbound depending on dialplan config)
16:53.21kannandmz, i dont get that? these are phones registering to *, and can do both. the sip.conf handles the registers
16:53.23[TK]D-FenderRico29: Show me
16:53.54kannanif i change the .cnf in the tftp, the phone's extension number like 101, 102 etc doesnt update
16:53.57dmzkannan, you have 6 line buttons on the right side of the phone, do you want to assign a different extension to each?
16:54.03Rico29what do you want to see [TK]D-Fender ?
16:54.04[TK]D-FenderRico29: In AGI I suppose you can do things while the audio is playing
16:54.19[TK]D-FenderRico29: Therefor you would ahve to wait until end of stream
16:54.20dmzkannan, the extension info is on how you have your dialplan send calls to the phone
16:54.20kannanno, i want to change the params for line 1 in the .cnf files
16:54.25[TK]D-FenderRico29: Makes sense
16:54.30kannandmz, thats not what i meant
16:54.40kannansorry i said extension
16:54.45kannancisco's phone number
16:55.04kannanthe phone is SIP/101
16:55.11kannani want to change to SIP/102
16:55.20kannanfor a particul;ar mac
16:55.23Kobaz102? that's crazy talk
16:55.23Rico29TK > but with get_data (= Background), it seems right to you to dont stream the audio message ?
16:55.33eppigyD:
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16:56.01kannandmz, hope i am clearer now
16:56.07dmzI just have multiple defined extensions in my dialplan, and each of those has: Dial, SIP/dmz7960
16:56.37dmzthen i've setup a findme function to either ring all of them or have escallation based on if they are busy for inbound calls going to different lines; not sure what your trying to do
16:56.46kannandmz, :(
16:57.15kannandmz, i have defines the sip params for a Cisco phone like SIP<mac>.cnf in the tftp root dior
16:57.36kannani want to change those params and so the Cisco phone gets a new number and , authname and passwd
16:58.32kannanbut editing the .cnf and restarting the phone doesnt do it
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16:59.02kannandmz, hope i communicate correctly now
16:59.22dmzhmm, are you sure it's realoading file?
16:59.37dmzi have same config for each line, i've not tried different ones, just assumed it worked
16:59.38kannanthe phone or * or the tftp?
16:59.41SlicerDicerwell I should have my Aastra 480i in 2 days :)
16:59.45SlicerDicerx2
16:59.55dmzlook at your syslog output when booting to see if it finds/loads file
16:59.57Kobazpolycom!
17:00.44SlicerDicerKobaz: they were 1000 pesos or 75$ maddog01 ;-)
17:00.55kannanwhen rebooting it takes only the old values
17:00.56Kobazhmm
17:00.57Kobaznot bad
17:01.01SlicerDicernot at all
17:02.32SlicerDicerKobaz: sorry for injecting the pesos was messing with maddog01 as I am in "New" Mexico heh
17:03.33Talkradiodo you see any of the violence happening near the border
17:04.10Talkradiomy girlfriend goes to mexico every other weekend and said it's getting really really bad
17:04.34mogshe a mule?
17:04.41mog^_^
17:05.13Talkradioi wish then i wouldn't have to support her lol
17:05.54Talkradiohmm is she a mule? she can take a big load and keep on trucking so maybe heheh
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17:09.05jameswfyou know you have spent to much time on the internet when you see a post titled "Two People One IP Phone" and your mind goes directly to the gutter
17:09.47jjshoehaha
17:10.29Talkradiohaha
17:10.57Rico29[TK]D-Fender> stream_file : "Returns: -1 on error or hangup, 0 if playback completes without a digit being pressed, or the ASCII numerical value of the digit if a digit was pressed"
17:11.01Rico29mine returns 1...
17:11.07jameswfanyone who caught the reference should log off too
17:13.25Kobazterm should be rxvt
17:13.26Kobazer
17:13.30SlicerDicerTalkradio: from what I hear the shit has hit the fan and _is_ spilling across the border
17:13.45SlicerDicerits not borderline or anything civil war... its 100% civil war down there
17:14.19SlicerDicerdrug lords, corrupt cops, bad government, angry citizens all mad...
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17:14.28Kobazfun fun
17:14.32SlicerDicerI dont think the druglords are as much to blame as the USA imo...
17:14.39SlicerDicerit is our fault for inviting them
17:14.42Talkradioit's scary
17:14.46SlicerDicerindeed
17:14.48russellblet's play a game called "talk about Asterisk"
17:14.51russellbit's really fun
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17:14.56Talkradio:D
17:14.57jjshoerussellb agreed
17:15.03SlicerDicersorry russellb he asked a question :)
17:15.21russellbnp, not directing it at any specific person ;)
17:22.08angryuseranyone using res_snmp.conf with nagios/centreon ?
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17:26.30Get_The_FishAnyone running 1.6 in here?  Whats the verdict- is it ready for "prime time"?
17:31.20jayteefor most uses in testing I've found 1.6 to be solid. I understand there are a few minor bugs with SIP TCP though but that'll be worked out quickly enough.
17:31.34Get_The_Fishwell thats good to hear
17:32.25Get_The_FishI dont use SIP TCP currently anyways
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17:47.35DarkRiftSimple question, when I want to make a .call file, let's say I want it to come from a "service" to monitor service status, for the channel, can I provide a non-existant one like if there is a built-in one that asterisk provides that would lead nowhere but in memory, or do I need to create a static user for it ? This call doesn't need to have a real connection on the other part cause it will only play a sound/text file
17:49.16DarkRiftLet's say I have a cron job that monitors my apache service, and this cron job creates a .call file to warn a tech that it failed, is there a pseudo-channel I can use for that or I absolutely need to create a static user ?
17:50.18filea call file calls one destination... and then executes dialplan logic OR executes an application...
17:50.44fileso what do you mean by "create a static user"?
17:51.36DarkRiftThe caller that 'originates the call'
17:51.50fileAsterisk is originating the call
17:52.13DarkRiftIc so Channel refers to the callee and not the caller, I think that was what I had wrong
17:52.19fileyes
17:52.24DarkRiftAlright, thank you
17:53.16filein this context at least... if you were using a call file for something that called a destination and then called elsewhere then it sort of would be the caller...
17:53.28phpboyTime for a fresh re install
17:53.30phpboyYAY! :D
17:53.52DarkRiftyeah I understand that
17:54.41DarkRiftI just thought the Channel initialy referd to the "real" caller that originated the call, thank you for this explanation
17:54.46*** join/#asterisk sah-work (n=Bawbatos@adsl-76-236-67-190.dsl.pltn13.sbcglobal.net)
17:59.22markgreeneis anyone in here aware of an RTP timestamp issue in the current branch that affects polycom phones?
18:04.18*** join/#asterisk dieguito84 (n=diego@host118-190-dynamic.12-79-r.retail.telecomitalia.it)
18:09.46*** join/#asterisk l2trace99 (n=jr@p1-bh-mco-1.prismone.net)
18:11.05*** join/#asterisk fukz (n=basti@pD9543FDE.dip.t-dialin.net)
18:12.34*** join/#asterisk lanning (n=lanning@66.151.128.195)
18:14.33Get_The_Fishmarkgreene: no, is this something that you are experiencing?
18:14.40*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:15.27*** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com)
18:17.17QwellManxPower: interesting hostname
18:18.09*** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar)
18:18.34*** join/#asterisk Sargun (n=Sargun@66.151.148.225)
18:18.36markgreeneGet_The_Fish,  I've got a polycom that has no audio for the first ~4 seconds of conversation
18:19.13Get_The_Fishinteresting.  What model and firmware rev?
18:20.54markgreeneGet_The_Fish, asterisk 1.4.16.2 and the model ip650 or ip550, etc.
18:21.31*** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr)
18:22.23russellbthere have been 1156 changes to asterisk 1.4 since 1.4.16 btw ;)
18:24.04*** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr)
18:25.05markgreenerussellb, that helps I guess
18:25.15Get_The_Fishmarkgreene, which rev of the polycom SIP application though?
18:25.49*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
18:26.51markgreeneGet_The_Fish, I don't have that on hand. The phone is offsite. Are there known issues with the old application?
18:27.30*** part/#asterisk beek (n=klinebl@65.211.106.242)
18:28.08Get_The_Fishnot that I know of, but I was going to look.  Have you contacted the vendor?  They have access to polycom's support channel, which is pretty good.  It's the best place to start if this is only affecting polycom phones.  I assume that you dont see the same behavior on a softphone, right?
18:30.23phpboyIs it safe to install asterisk and dundi off a fresh CentOS 5.2 installation?
18:30.29Get_The_Fishyes
18:30.32phpboyI.e. Would i require any extras?
18:30.46[TK]D-Fenderphpboy: no
18:30.59phpboy[TK]D-Fender: no to which question?
18:31.00Get_The_Fishphpboy, it depends on what options you chose on the centos install, but using defaults no
18:31.09Get_The_Fishno extras
18:31.11[TK]D-Fenderphpboy: Install CentOS, install *, done.
18:31.17phpboyperfect
18:31.31Get_The_Fishsi, did it myself, and it installs flawlessly
18:31.36phpboyI always went through the process of upgrading everything
18:31.45phpboyIt was very timeconsuming
18:31.59Get_The_Fishnormally a good idea phpboy, but not really necessary
18:32.27phpboyWell, this server will be replacing our main server which has been giving me hell of late
18:32.43phpboyI've also been doing live testing and dev on it which has also caused some headaches
18:33.00Get_The_FishI have had very very good luck on centos 5.2 and asterisk 1.4.22
18:33.14phpboySo this is going to be a 'clean' install and I'm going to reconfigure it from scratch
18:33.24phpboyGet_The_Fish: Zaptel or Dahdi also?
18:33.32Get_The_Fishno, SIP trunking only...
18:33.44phpboyah, I'm having problems on the PSTN side
18:33.48phpboyload related issues
18:34.02Get_The_Fishbut I know of several installations using centos 5.2 with zaptel on both sangoma and digium hardware
18:34.16Get_The_Fishload issues?  Such as?
18:35.32phpboyGet_The_Fish: Well, this box is VERY VERY busy, To give you an idea, we have 110 outbound/inbound IAX2 channels, 2 spans Outbound only E1, and 2 spans inbound and some outbound E1
18:35.46phpboyso you can imagine how busy the box must be to justify all that hardware
18:35.53phpboyor rather all the channels
18:36.06phpboyanyhoo, at night when it's quite, everything runs just fine
18:36.28ManxPowerWhy not just split the one box into 2 boxes?
18:36.34Get_The_Fishis the the rest of the hardware underpowered- cpu, RAM, etc?
18:36.54phpboyoffice hours, after 10 to 20 or so calls I get timing errors, get yellow alarms on my inbound calls. It's a mission
18:37.07phpboyManxPower: I'm not sure how I'd lay that out?
18:37.14phpboyconsidering how the call centers work, etc
18:37.14ManxPowerdo you also get HDLC abort errors?
18:37.22phpboyManxPower: yes
18:37.32ManxPoweryou must have a very old card and old zaptel
18:37.36Get_The_Fishglare?
18:37.44Get_The_Fishwhat protocol are the T's using?
18:37.49ManxPowerGet_The_Fish: Glare does NOT call yellow alarms
18:38.25phpboyManxPower: Zaptel zaptel-1.4.12.1 ?
18:38.40ManxPowerphpboy: you must have an old card then
18:39.00phpboyIt's not THAT old, it was the newest card at the time back in 01-04-2008
18:39.22phpboyI've got 2 new TE420's on PCIx in this new box
18:39.33ManxPoweryour problem is that the audio data is coming from the card too fast for the computer to process, usually these issues are caused by things like SATA or GigE on the motherboard locking interrupts for a long time.
18:39.35phpboyI believe in my heart that this will tend to my problems
18:39.39*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
18:40.01ManxPowerthe new cards are much better at working around interrupt latency issues.
18:40.05phpboyManxPower: I read something about that, I'm using SATA2 based RAID 5?
18:40.18anonymouz666phpboy: what card?
18:40.25anonymouz666TE110P?
18:40.31phpboyyes
18:40.34anonymouz666RMA it.
18:40.39phpboyRMA it?
18:40.50ManxPoweralso why did you put TWO cards in the box?  You are generating twice as many interrrupts as you need to.
18:41.05ManxPowereach card generates 1,000 interrupts/second if I recall correctly.
18:41.21phpboyManxPower: the thing is, we have more lines that we're currently not using
18:41.30*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
18:41.34phpboyalthough, I will initially only be running one TE420 in the new box
18:41.37ManxPowerphpboy: how does that apply to using one card instead of two cards?
18:42.12ManxPoweror do you mean you need 8 ports?  I don't keep up with the way Digium randomly names their cards these days.
18:42.13phpboySee, We had trouble with this hardware, specifically with the IRQ's
18:42.25phpboyManxPower: I need 8 ports
18:42.30phpboy2 x 4 port cards
18:42.41ManxPowerThen forget what I said about 2 cards.
18:42.50phpboyanonymouz666: What's your thoughts on this concidering the TE410P ?
18:42.58phpboyManxPower: ok
18:43.00[TK]D-Fenderphpboy: ANCIENT
18:43.06phpboy:(
18:43.08telnettechi want to 1st thank everyone on here for the help that they have given me over the last 2 months....i dont think i could have gotten as far as I have with this install. I hve been working on Asterisk for only 4 months and have learned alot from here, the Jason Smith/Leif Madsen book and voip-info.org
18:43.11*** join/#asterisk beek (n=klinebl@65.211.106.242)
18:43.13phpboy[TK]D-Fender: you think this could be my problem?
18:43.17[TK]D-Fenderphpboy: And ditched my 2 fast.
18:43.41ManxPowerphpboy: those old cards have massive issues with interrupts and lost data on many different motherboards.
18:43.53telnettechwith that said, I am needing some more help....I can see on the CLI that sound files are playing but I dont hear them on the phones....can omeone point me in the right direction?
18:44.07ManxPowerChances are you could have just replaced the cards in your existing system to fix the problem
18:44.15*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
18:44.39phpboyManxPower: I can give that a shot.
18:44.56ManxPowertelnettech: either you have an unconfigured zaptel card in the system or you have a system with a kernel HZ not equal to 1000 or you have borked RTC support in your kernel
18:46.04phpboyManxPower: I'm going to put one of the new cards into the current server and test it tomorrow durin office hours.
18:46.20ManxPowerremember some of the new cards use new drivers
18:46.53phpboyManxPower: this would mean I'd have to switch to dahdi :/
18:46.56telnettechManxP: i have no zaptel cards in my system. we are using a redfone device for the Telco connection. what is kernel HZ and what is borked RTC support...i am using Red Hat Linux 5.1 OS as a 64bit system
18:47.05ManxPowerphpboy: I did not say that!
18:47.25phpboyManxPower: Well, I've got the latest drivers and the claim to support this specific card, well kinda
18:47.28ManxPowertelnettech: just make sure ztdummy is not loaded (lsmod)
18:47.33phpboyTE4XX
18:48.20*** join/#asterisk crevetor (n=crevetor@bureau.ubity.com)
18:48.24phpboyManxPower: I shouldn't assume that if it works now (afterhours) with one of the new cards that the drivers are fine?
18:48.29crevetorhi
18:48.37ManxPowerphpboy: yes
18:49.02telnettechManxP: I dont have ztdummy running.....funny thing is that if i do, the sound files play fine but my redfone device doesnt work
18:49.12crevetorquick question : how bad is it if I have my sip peers in a mysql DB using dynamic realtime but I don't allow asterisk to write to the DB
18:49.36crevetorfrom what I see in the debug it only tries to set empty values when my user registers
18:49.46ManxPowertelnettech: TDMoE is not well supported in Asterisk since IAX2 happened.  Best of luck with that.
18:49.48telnettechi have been working on this since yesterday afternoon
18:50.01crevetor<PROTECTED>
18:50.09ManxPowercrevetor: I don't think anyone has been stupid enough to try using a read only Realtime DB
18:50.10*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)
18:50.24crevetorManxPower: Well, I am :)
18:50.54phpboyOk, let's give it a shot
18:50.57crevetorManxPower: and I'd like to know in which way it's so stupid
18:51.01ManxPowertelnettech: Not suprizing.  very few people ue TDMoE
18:51.02telnettechManxP: this is Asterisk 1.2.28.....we have the same setup at another location that is working...the only difference between the 2 sites is 1 is a 32bit OS and this is a 64bit OS
18:54.17*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
18:55.04*** join/#asterisk Assid (n=assid@unaffiliated/assid)
18:55.10Assidstupid stupid net connection
18:58.14telnettechManxP: any other suggestions?
19:00.43ManxPowertelnettech: nope.  never used redfone, never used Asterisk on a 64 bit OS.
19:00.54ManxPowerCan't imagine why anyone would want to do either of those things.
19:01.12phpboyManxPower: I'm running my asterisk on a 64bit OS
19:01.23telnettechim not the developer....im just a technician told to make it work
19:01.38phpboyManxPower: Just put in the new card still getting Yellow alarms every now and then :(
19:01.52ManxPowerphpboy: but are you getting HDLC aborts?
19:02.08phpboyManxPower: no
19:02.34phpboyI can see no reason for these yellow alarms :/
19:03.31phpboy[Jan  6 21:01:04] WARNING[6967] chan_zap.c: Received NOTIFY on unconfigured channel 255/255 span 3 <--- now that's new
19:04.30ManxPowerphpboy: especially because span 3 is channels 49 - 73
19:05.03phpboyyes and 255 is not in zaptel.conf or zapata.conf
19:05.21phpboyIt seems this new card is dropping the calls even more then the old card :(
19:05.43ManxPowerphpboy: what verison of zaptel did you say you are using?
19:06.27phpboyI just put it onto another span and this lovely error poped up
19:06.31phpboy[Jan  6 21:03:58] NOTICE[8699] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 <-----
19:07.04ManxPowerphpboy: What version of Zaptel did you say you are using?
19:07.05phpboyZaptel Version: 1.4.12.1
19:07.24ManxPowerat least you have the latest.
19:07.43ManxPowerfor those errors you can contact Digium support, it's covered under the card support.
19:07.51ManxPowersince there is nothing you can do in ASTERISK to fix them.
19:08.04ManxPowerIT's all zaptel/dahdi, the card, the cables, or the telco
19:08.13telnettechManxP: thanks.....i convinced the devlopment lead that the trouble is that the 64bit is not the correct version..they are going to do more development but I have been told to get customer working
19:08.30*** join/#asterisk dandate (n=dandate2@adsl-99-55-171-193.dsl.pltn13.sbcglobal.net)
19:08.31ManxPower~mailinglist
19:08.32jbot[~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search.
19:09.39dandatemy asterisk machine cannot connect to the internet, do i just need to plug the cable modem directly to the machine or does it have to be connected through a router?
19:09.45phpboyManxPower: This is even stranger, I plug in another E1 and the errors become a lot more
19:09.59*** join/#asterisk voxter (n=voxter@76.77.95.2)
19:10.08phpboyif it makes any diffs, the one E1 gets here through fiber and the other through copper from the telco
19:11.20dandateim trying to install from flashpbx, it wants to download an update but it cannot connect to the internet with this default centos install
19:17.06riddleboxdandate: plug the machine into the switch
19:17.13riddleboxor router
19:18.13*** join/#asterisk jeffgus (n=jeffgus@static-173-51-181-4.lsanca.ftas.verizon.net)
19:19.15phpboyManxPower: to further your theory on my problems, it is posible that this would only affect one PRI?
19:19.31phpboyAs the other two are running just fine on the same card
19:23.11*** join/#asterisk LemensTS (n=customgt@70.238.154.243)
19:24.00LemensTS<PROTECTED>
19:24.34rob0Try "password".
19:25.00QwellLemensTS: Asterisk does not create an 'asterisk' user.
19:25.01LemensTSnope
19:26.05rob0(I wasn't being serious.)
19:26.46*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
19:27.14rob0Ask whoever has root on that box to reset it for you. If that's you, see "man passwd" and your distro docs.
19:27.19SlicerDicerhow do I make sure asterisk is acting as a media proxy?
19:27.57SlicerDicerrob0: take a lesson from macrumors eh ;-)
19:28.07[TK]D-FenderSlicerDicer: "canreinvite=no" <- under [general] & all peers
19:29.25SlicerDicerok [TK]D-Fender thanks
19:29.49*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
19:30.05*** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za)
19:30.12*** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za)
19:30.44*** join/#asterisk kannan (n=kannan@121.246.242.95)
19:37.06*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
19:37.13Assidokay, until i figure out how to move this phone to sip, im gonna try SCCP
19:37.26Assidso, i got sccp up, but i cant manage to make my calls
19:37.34Assidhttp://assid.pastebin.com/d39f37288 -- a log of when i try to make it
19:38.28*** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca)
19:40.42Assidoh it takes time
19:41.38Assidokay voice from cisco reaches polycom.. not the other way around
19:42.54phpboylibpri is obviously a requirement when using E1 on dahdi?
19:43.12eppigyYEAH SON
19:43.49*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
19:54.22Assidokay im having 1 way audio for a sccp <-> sip call
19:58.47mmlj4are you behind NAT?
20:00.42phpboyThere's actually quite a bit you've got to do before you can install asterisk/MySQL/dahdi/Libpri
20:01.49*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
20:03.05telnettechhello
20:03.32jayteehi
20:03.46telnettechi was checking to see if i was still connected
20:04.20jayteenope, you're offline
20:04.28jayteecompletely out of touch
20:04.30*** join/#asterisk mindCrime (n=chatzill@74.213.159.129)
20:05.05jayteeadrift in limbo
20:05.08*** join/#asterisk jtodd (n=jtodd@nat/digium/x-854a1f4b87c7e729)
20:06.36telnettechi wish i was adrift with either a pina colada or tequila sunrise on the beach instead of this hot telephone room
20:09.08denonif the telephone room is hot, you'll probably be spending even more time in there
20:09.24*** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net)
20:09.25denonlikes ~60 degrees F
20:09.27*** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net)
20:09.51*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
20:10.30telnettechthey have AC but i have to door open.....it has a small Mitel but the humidity is bad here in Aruba today
20:11.00telnettechnot too hot for the equipent.....yet
20:11.39telnettechbut im sweating and would rather be sweating sitting on the beach looking and chasing women
20:14.21Assidmmlj4: yes
20:14.45phpboyThis new server is going to be BEAUTIFUL when I'm done with it
20:14.48Assidmmlj4: sorry for the delay, trying to figure this out
20:15.39*** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net)
20:17.17*** join/#asterisk scottgutman (n=scottgut@c-76-18-0-243.hsd1.fl.comcast.net)
20:17.31*** part/#asterisk timothy_jones (n=chatzill@207.90.4.3)
20:17.35*** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com)
20:17.47Assidcant even get it to play back voicemailmain
20:18.14scottgutmanhello all
20:19.01phpboyhi
20:20.15scottgutmani was hoping i could find some help.  i am a noob at asterisk
20:20.42phpboyscottgutman: We cannot help if we don't know what the problem is?
20:24.24scottgutmani am using a kiax soft phone, asterisk 1.2.24/slackware 12, connecting to binfone IAX termination.  When i make a call the Connection is not transferring correctly and route through  asterisk instead of making a direct connection.  I have transfer=yes in the iax.conf
20:25.59edoceohow do I get a local number in Thailand to ring my server in Seattle?
20:26.27phpboyedoceo: Over the internet?
20:30.42*** join/#asterisk c0ldk1ll3r (i=be348af8@gateway/web/ajax/mibbit.com/x-a866f85c853cf93f)
20:31.00SuPrSluGscottgutman: if your a noob. why use asterisk 1.2.x they're about to release 1.6 already.
20:32.06scottgutmani followed a step by step guide, and i did not want version conflicts.  after this i plan to install freepbx and vicidial
20:32.11SuPrSluGanyhow are the phone and your phones registered
20:32.52SuPrSluGerr provider
20:33.36scottgutmanshould i paste from the iax.conf?
20:34.28edoceophpboy: right - but do I need a server in Thailand to have the local number terminate to?
20:35.00SuPrSluGgo to the cli > iax2 show registry
20:35.49scottgutmanHost                  Username    Perceived             Refresh  State
20:35.49scottgutman208.72.186.132:4569   103067      76.18.0.243:4569           60  Registered
20:37.28SuPrSluGuse pastebin when doing that
20:37.30phpboydahdi_genconf automatically generates configs?
20:37.34SuPrSluG~pb
20:37.35jbot[~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://www.rafb.net/paste, or , http://bin.cakephp.org/
20:37.51scottgutmangot it
20:38.11phpboyedoceo: yes, you would need a server in thailand
20:38.32SuPrSluGcouple lines ok. more than that some folks get ornery
20:38.49scottgutmanmakes sense
20:40.22SuPrSluGso the phone and provider are connected. what happens when you make a call? do a cli> set verbose 7    and make the same call
20:41.04SuPrSluGlook for errors
20:42.35scottgutmani have a message log open and verbose is set to 88. here is the last call. http://pastebin.com/d6cc00ec1
20:43.57jayteebeek, are  you here?
20:44.13beekjaytee: yes sir
20:44.24phpboyThis sucks
20:44.35jayteebeek, embedded silence in a text source? ever done it?
20:44.37phpboydahdi_genconf doesn't wanna auto generate config files :(
20:45.00beekjaytee: no.  I could try it though.   Are you having issues with it?
20:45.51jayteewas just wondering if there was a special character set for Cepstral to recognize. Didn't find anything on their site yet. I'll keep searchin and googling.
20:46.11beekjaytee:IIRC, you can get swift to embed other audio files so you should be able to do it using the wav files from the Asterisk distribution.
20:46.13jayteeif all else fails I'll just pad silence with audacity
20:46.54beekjaytee: Look in their docs for the link to the embedded TTS codes from another web site.  It had that which you seek.  Let me see if I can find it again.
20:47.33jayteebeek, never mind, I'll do the diggin. just thought I'd ask if you knew off the top of your head.
20:47.36riddleboxKatty: you around?
20:48.06beekjaytee: start here http://www.cepstral.com/cgi-bin/support?page=faq&type=ssml
20:48.16jayteebeek, thanks. will do
20:48.30beekjaytee: Then here: http://www.w3.org/TR/speech-synthesis/
20:49.10*** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com)
20:50.51scottgutmanSuPrSluG: did i put too much?
20:51.29jayteebeek, yup! found it. <break time='4500ms' /> inserting that into the text where I want silence gives me a 4.5 second pause.
20:52.13beekjaytee: cool.
20:55.15*** join/#asterisk joesuffceren2 (n=chatzill@srv.fgp.com)
20:55.35joesuffceren2can anyone recommend an asterisk compatible tapi driver that will work on a 64-bit terminal server? I really like activatsp (installed on my workstation),but they don't have a 64-bit version, and I don't have visual studio to compile my own
20:57.01Assidanyone know why my cisco phone cant hear anything? i mean outgoing audio works, incoming nope
20:57.10Assidthe box is remote, phone is behind a nat
20:57.39phpboyI wish I could find a nice doc on how to configure dahdi properly for E1 :/
20:57.53Assidcalls between cisco and polycom polycom can hear cisco, but not the other way around
20:58.11*** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net)
20:59.32Qwell<Assid> the box is remote, phone is behind a nat
20:59.37QwellAssid: sounds like you answered your own question
20:59.52jjshoesipnat~
20:59.55jjshoeor something like that
20:59.59jjshoehowever that works in here
21:00.01Assidccspnat
21:00.04[TK]D-FenderQwell: Worse yet... he's using SCCP
21:00.12Assidsccp even
21:00.14jjshoe~sipnat
21:00.14jbot[~sipnat] Quick guide on configuring * + 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
21:00.31Assidmy other sip devices are able to talk to each other
21:00.36Assidits only this stupid thing
21:01.28Kobazis nat=yes
21:01.31Assidmainly cause i cant figure out how to push this to sip, and if i get it working, theres a chance i will have a job/project i can work with
21:01.37AssidKobaz: on sccp?
21:01.45Kobazyou said sip
21:01.51Kobazoh, other sip devices
21:02.21Assidand i normally use nat=route :P
21:04.13SuPrSluGscottgutman: host=dynamic ?
21:04.56scottgutmanyes
21:06.49SuPrSluGhave you tried using sip termination? this is an odd problem.
21:07.18DaPrivateertrying to use Page() on 1.4.22 w/ FreeBSD 6.2 and getting error app_meetme.c:1620 conf_run: Unable to set flags: Inappropriate ioctl for device -- anyone have any suggestions?
21:07.37phpboyDahdi is a mission to get working :/
21:07.38SuPrSluGthat or try newer verison of asterisk
21:07.49*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
21:07.58scottgutmanthe sections from the iax.conf: http://pastebin.com/m1ca33f
21:08.56scottgutmando you know about vicidial? do you think there will be a conflict?
21:09.47SuPrSluGyes I played around with it a while back. they too are using 1.4   now
21:09.48*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-221-124.phlapa.east.verizon.net)
21:10.07SuPrSluG<PROTECTED>
21:11.00SuPrSluGi had a 3 man call center with it. works as advertised
21:11.06scottgutmanto upgrade, just download the tarballs, make, make install and poof, upgraded?
21:11.18SuPrSluGyeah
21:11.31scottgutmanshow i do 1.6 or 1.4
21:12.18SuPrSluG1.4 if your going towards vicidial they'll be months behind when 1.6 is released.
21:12.39*** join/#asterisk thsyrus (n=thsyrus@cpc3-darl3-0-0-cust1018.midd.cable.ntl.com)
21:12.51scottgutmanany particular ver of 1.4 or the lastest one will do?
21:13.05SuPrSluGthere's a lot to test and rewrite
21:13.38*** join/#asterisk qdk_ (n=qdk@79.138.251.161.bredband.3.dk)
21:13.46SuPrSluGcheck what they're using in their forums i'd guess 1.4.22 should be good
21:15.44scottgutmanthanks for your help
21:15.52scottgutmanand advice
21:16.12Assidokay i wnna shift this to sip
21:16.16Assidim tired of sccp
21:16.24Assidi want to make calls and be able to hear people
21:17.10srdif you want to hear people you'll have to buy the hearing people package
21:19.54kannanAssis, try to change rtp.conf to match the cisco's RTP map, its some like 16000+ to 3000o i think
21:20.04kannanAssid
21:20.19kannanit may not be the NAT issue
21:21.21Assidhrmm
21:21.21Assidk
21:23.54Assidyeah it came up once
21:26.00Assidnow i cant get audio again
21:26.24*** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net)
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21:27.05[TK]D-FenderCheckout time, later all
21:27.07*** part/#asterisk scottgutman (n=scottgut@c-76-18-0-243.hsd1.fl.comcast.net)
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21:29.34*** mode/#asterisk [+o bkruse] by ChanServ
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21:30.09*** mode/#asterisk [+o bkruse] by ChanServ
21:37.38NovceGuruAnybody know anything about PTCRB certs? If a gsm/cdma module is certified, does the final product using that module have to be certified?
21:37.51*** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net)
21:38.23khronos<PROTECTED>
21:38.31NovceGuruoh?
21:43.10phpboyDAMNIT!
21:43.23phpboyNow I get seg faults when using IAX2 :/
21:48.10*** join/#asterisk ScarEye (n=scareye@12.27.87.66)
21:52.35gambler1Hi, is there any method that we can handle sip messages in dialplan? (like addres incomplete)
21:53.07gambler1ax.. my fingers... is there any method that we can use to handle sip messages in dialplan? (like addres incomplete)
21:53.46*** join/#asterisk dandate (n=dandate2@adsl-99-49-15-194.dsl.pltn13.sbcglobal.net)
21:54.12dandateI'm trying to install flash pbx and it has been stuck at *now compiling zaptel for over an hour, is this usual?
21:54.39dandatei'm sorry thats pbx in a flashj
21:56.24*** join/#asterisk Winkie (n=urmom@ur.fa.gs)
21:58.01*** join/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu)
21:58.32jantmancan anyone recommend a good forum for general VoIP questions (perhaps asterisk-specific, but more general advice than tech stuff)?
22:00.09*** part/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu)
22:02.16x86hey guys... trying to think of the best way to do night mode...
22:02.49x86was thinking about setting up an extension that would simply touch a file, and if the file exists, the main IVR would know it was in night mode
22:03.01x86and then just dial that extension again to have it remove that file
22:03.22x86is that a decent idea, or is there a better practice
22:03.59*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
22:05.27carrarx86, why not use a db?
22:05.32carrarthats what I use
22:05.51carrarthen they can do it via a call or via the web
22:06.16*** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk)
22:07.40thedonvaughnIf I assign a mailbox to a sip user in users.conf via mailbox = , where is the vmpass and other info stored?  It's not writing anything to /etc/asterisk/voicemail.conf and I do not see anything stored in astdb.  However, my voicemail is working and my pin is set.   I'm just asking because I forsee someone forgetting their vm pin sometime in the future after setting it through the menu and i just want to know how i can a) reset it or b) retriev
22:09.01[TK]D-Fenderthedonvaughn: voicemail.conf <-
22:09.17thedonvaughnnone of my voicemails are there
22:09.49[TK]D-Fenderthedonvaughn: thats where the PIN is.  VM's are under /var/spool/asterisk/voicemail normally
22:10.20thedonvaughn[TK]D-Fender: no i mean no entries for any of my voicemail boxes in /etc/asterisk/voicemail.conf.   That's why i'm asking :)
22:10.30*** join/#asterisk sevard (n=sev@multimedia.dvc.edu)
22:10.36jtoddx86: If you're looking for bleeding-edge, you could also look at the calendar stuff that twilson is working on.  Then just create a calendar with events that span "night" times (or holidays, or weekends, or whatever.)
22:10.38thedonvaughnI only set them up in users.conf, and voicemail.conf is never written to.   I can even do voicemail show users in console and see all my users.
22:10.52thedonvaughneveryone's working fine, they can change their pins via the menu system and all is good.  Just need to know _WHERE_ that info is stored.
22:10.56thedonvaughnit's definitely not in voicemail.conf for me
22:11.14[TK]D-Fenderthedonvaughn: Check the VM folders then
22:11.23thedonvaughnyah let me start there
22:11.24thedonvaughnthanks
22:11.24[TK]D-Fenderthedonvaughn: perhaps it is in users.conf
22:11.31thedonvaughn[TK]D-Fender: yah thought that to, it's not heh
22:11.39[TK]D-Fender~users.conf
22:11.39jbotusers.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system.
22:11.41[TK]D-Fender^^^^^^^^^^
22:12.05thedonvaughnyah i'm starting to realize this
22:12.50DaPrivateertrying to use Page() on 1.4.22 w/ FreeBSD 6.2 and getting error app_meetme.c:1620 conf_run: Unable to set flags: Inappropriate ioctl for device -- anyone have any suggestions?
22:14.35[TK]D-FenderDaPrivateer: http://groups.google.com/group/mailing.freebsd.ports-bugs/browse_thread/thread/0b6d0fbfa8d85255
22:15.21DaPrivateeryes, i saw this, but commenting out an fdset slightly worries me
22:15.59*** join/#asterisk dandate2 (n=dandate2@adsl-99-183-242-53.dsl.pltn13.sbcglobal.net)
22:16.11dandate2I'm trying to install pbx in a flash and it has been stuck at *now compiling zaptel for all time, is this usual?
22:16.31voxterHey FYI whoever out there using polycoms, and potentially has left the polycom default username and password for FTP login for provisioning purposes (PlcmSpIp/PlcmSpIp) - There are bots going around lately trying to log into servers with those credentials and exploit them to join botnets.
22:19.37[TK]D-Fendervoxter: Anyone leaving defaults like that deserve what they get.  Just think of all the Trixbox idiots getting pwned out there ;)
22:19.50*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
22:20.56sevard[TK]D-Fender: I agree.  Leave your linksys wrt54g open and your neighbor will leech and not even invite you over for his bbq.
22:21.09voxter[TK]D-Fender: I've posted stuff about trixbox on their forums like this before too. Just saw it happen to a couple people who thought "nobody will ever guess that anyways" - thought it'd be useful to point out to people that yes, even though its obscure, it does happen to people.
22:21.49[TK]D-Fendervoxter: 2 words : NATURAL SELECTION
22:22.14[TK]D-Fendervoxter: Let'em whine
22:22.42beeksevard: http://ars.userfriendly.org/cartoons/?id=20090105
22:23.05sevardbeek: hahahaha.
22:26.02*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
22:32.26Kattyblergh
22:34.03[TK]D-FenderKatty: Mew.
22:34.44Kattyhey
22:34.47Kattyjbot: who is peter grace?
22:34.49jbotI think you lost me on that one, Katty
22:35.19[TK]D-Fenderjbot: Who killed J.R.?
22:35.35[TK]D-Fenderjbot: USELESS
22:35.36jbotACTION starts crying and hides from [tk]d-fender in the darkest corner of the room. :(
22:35.46[TK]D-Fender:D
22:35.52Corydon76-dig~drumkilla
22:35.52jbotmethinks drumkilla is Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>, or someone who should be ph33r3d, or russellb
22:35.56Kattywonders how these people know her ^_-
22:36.03Corydon76-dig~katty
22:36.04jbothmm... katty is the only girl in the channel, so be nice to her
22:36.14seanbrightpffft
22:36.28Kattyseanbright: who is peter grace? :<
22:36.28Corydon76-digThat's right; seanbright is also a girl...
22:36.33Corydon76-digWell, a girly-man
22:36.39seanbrightKatty: no clue
22:37.51Corydon76-digKatty: not sure why, but the letter 'N' comes to mind
22:37.52Kattywhat about a david mcnett
22:37.56*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
22:37.57seanbrightnope
22:38.00seanbrightnothing
22:38.11Kattyyet they all claim to know me
22:38.13Kattythis is madness
22:38.16russellbo.O
22:38.25russellbwonders how he got brought into this
22:38.25Corydon76-digI remember Peter Grace, but I don't remember his nick
22:38.39seanbrighti remember big league chew
22:38.44Corydon76-digrussellb: d-fender made the bot cry
22:38.54russellbbad [TK]D-Fender !
22:38.59[TK]D-FenderKatty: If you believe that.... then you may already have won the Publisher's Clearing House Sweepstakes!!!!!!
22:39.34[TK]D-Fender~jbot
22:39.34jbotit has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass
22:39.36Katty[TK]D-Fender: yeah, but they have 'mutual' friends which i recognize :/
22:39.49[TK]D-FenderCorydon76-dig: MY bitch... get your own! :p
22:40.17[TK]D-FenderunfUNFunfUNFunfUNFunfUNFunfUNFunfUNF
22:40.53Corydon76-digKatty: remember "km-" on here?
22:42.33KattyCorydon76-dig: sound familiar
22:42.57Corydon76-digKatty: that's Peter Grace
22:43.15KattyOoooo
22:43.52Kattywow.
22:43.56Katty120 friends down to 21
22:44.06Kattyhow about that for almost-spring cleaning
22:44.08Corydon76-digFacebook?
22:44.14*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)
22:44.21Kattynods
22:44.27russellbI bet you unfriended me, didn't you!
22:44.27Corydon76-digKatty: add me!
22:46.05fileKatty: pssssssst, David McNett is Nugget over there
22:57.31*** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
23:01.54jetsHas any one found a good way of demonstrating G722 inside of asterisk trunk?
23:02.28jetsTwo phones that call one another sound great but I was hoping to have an extension with a gsm audo file played
23:02.36jetsand an extension with an mp3 played to show the call quality difference
23:03.21drmessanoSet up a conference
23:03.35drmessanoPut the 2 G722 phones, one set to GSM, maybe one set to G711 in it
23:03.38[TK]D-Fenderdrmessano: Meetme still mixes to 8khz IIRC
23:03.54drmessanohmm
23:03.58drmessanoDidnt realize that
23:04.12drmessanoOk
23:04.12jetsIt seems like everything inside asterisk does :(
23:04.17drmessano3 way calling
23:04.22[TK]D-Fenderdrmessano: thats a point FreeSWITCH pimps is that their core is higher
23:04.38drmessanoGet a GSM and a G722 phone, call the G722 phone on call waiting
23:04.41drmessanoClick back and forth
23:04.58[TK]D-Fenderok, off to martial arts, BBIAB
23:05.15jetsI need something automated as if i were in a sound booth.  I was hoping I could set extension 100 to be a high bit rate g722 audio file or prompt.
23:05.20*** join/#asterisk murdock_ut (n=chatzill@64-42-64-98.atgi.net)
23:05.22jetsand 101 to be a g711/gsm prompt played
23:06.03_ShrikEjets: this demo does pretty much exactly what you are asking for.  sip uri:  wbdemo@conf.zipdx.com
23:06.07murdock_utI've ran into a bit of a problem.  Using 1.6, when I use one touch parking and I pick back up the parked call before it rings back I cannot repark the call.
23:06.24_ShrikEassuming it is still up.
23:07.38murdock_utIf the parked call rings back I can repark it just fine.
23:11.23jetsAny tips to calling this sip uri from asterisk?
23:11.30jetsIt keeps telling me I am called in at narrow band :(
23:12.41*** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il)
23:13.02rob0"core show application dial"?
23:20.28Nuggetsits down next to Katty
23:20.43_ShrikEjets: You may need to create a peer and only allow g722
23:21.06jets_ShrikE: I did that and a show on the channel showed g722, but ulaw being used
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23:49.44*** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com)
23:49.47ZippomanHey guys
23:49.51*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
23:50.01Dovidhi Zippoman
23:50.39Zippomansmall issue...when i dial out from asterisk it always says my number is blocked when right before it i set the callerid
23:51.00*** join/#asterisk `paul (n=temp_acc@122.55.36.3)
23:51.17*** join/#asterisk jmacz (n=jmacz@190.159.100.152)
23:51.23Dovid~pb
23:51.23jbot[~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://www.rafb.net/paste, or , http://bin.cakephp.org/
23:51.28`paulwill dhadi work on older versions of asterisk???
23:51.31Dovidpost ur extensions.comf
23:51.47Dovid`paul: What version ? I dont think it will for real old versions
23:52.09`paul1.4....
23:52.10beek`paul: 1.4.xx (where xx = I'm not quite sure) and above
23:52.19Dovidextensions.conf*
23:52.31`paulcause i cant seem to get ztdummy working for meeme
23:52.46`paulbut i remember making it work on dhadi
23:53.07beek`paul: Is it the newest version of 1.4?
23:53.08Dovidnot sure if all of 1.4.X works
23:53.16Doviddo a make menuselect and see if dahdi is there
23:53.53*** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net)
23:54.29jayteedahdi will only work on 1.4.22 or any higher rc version, not on 1.4.21 or earlier
23:55.02`pauli did menuselect zaptel is required for meetme
23:55.11Dovidthen ther is ur answer ;)
23:55.19Dovidu can optomize the kernel to work better
23:55.23`paulwell actually i got zaptel installed but i get a psuedo device warning when using meetme
23:55.54jayteeor to be more precise as I understand, 1.4.22 was the first release version with dahdi, you might get it working on some slightly earlier versions with some tweaking
23:56.13`paulapp_meetme.c:772 build_conf: Unable to open pseudo device  <---all i get is this when using meetme
23:56.27`paulpls help :(
23:56.47jayteedo a ps aux | grep ztdummy
23:57.07jayteesee if it's loading ztdummy
23:57.33`paulits not
23:57.50`paulactually it fails when i do a /etc/init.d/zaptel start
23:58.06`paulLoading zaptel framework:  FATAL: Module zaptel not found
23:58.14`paulWaiting for zap to come online...Error: missing /dev/zap!
23:58.23`paulthose are the error msg
23:59.09jayteewhat distro?
23:59.30`paulcentos 5
23:59.58jayteedid you do a make install when you compiled zaptel?

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