irclog2html for #asterisk on 20060323

00:00.31MacWeenielooks like SIP has some options for DTMF.. but not IAX
00:00.34*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
00:00.42justinufuzzbawl: ask them to explain themselves
00:01.31fuzzbawlI did. they said I would have to get my account rep to re-order the line as PRI
00:01.54fuzzbawlthe tech said they have to "change equipment" for us to have PRI
00:03.23*** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc)
00:05.22h3xfuzzbawl: theres a variety of reasons for that ;)
00:05.41h3xthere are different switch partitions on the big iron switches for pri and robbed bit circuits
00:06.06h3xsometimes they dont have a DACS to move your circuit to another paritition, so they need the LEC to connect it somewhere else
00:06.24fuzzbawl<PROTECTED>
00:06.33fuzzbawlannoying though
00:06.37h3xit should be possible to do it with a MACD (change order)
00:07.37cripitokrisk84: modifying the files for my info...
00:08.35*** part/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com)
00:08.47Luke-Jranyone have sellvoip working for inbound?
00:10.00cripitokrisk84: no...
00:10.51MacWeenieLuke-Jr: i just got it working
00:11.02MacWeenieLuke-Jr: but my DTMF is not wokring inbound.. calls are working though
00:11.28MacWeenieDTMF works through my softphone though...
00:13.51MacWeenieLuke-Jr: what problem are you having?  i had to play around a little to get it to work. but mainly because i don't know exactly what i'm doing
00:15.48Op3rI got this
00:15.48Op3r-- Executing AGI("SIP/334-70e5", "recordingcheck|20060323-081502|1143072902.2190") in new stack
00:15.48Op3r<PROTECTED>
00:15.49Op3r<PROTECTED>
00:15.49Op3r<PROTECTED>
00:15.59Op3rdoes that mean the call is being recorded
00:17.21fuzzbawlin zaptel.conf, it says that the specific implementation of e&m is handled by the userspace library? I assume that would be zapata.conf/
00:17.29fuzzbawlor do I set that somewhere when doing a modprobe?
00:20.20Luke-JrMacWeenie: after the call is answered, nobody hears ea other and hangups don't work
00:20.36*** join/#asterisk decanus (n=deano@host86-143-129-183.range86-143.btcentralplus.com)
00:20.37MacWeenieare you using conf?
00:20.47Luke-Jrno
00:20.51decanusdoes anyone know if its possible to set a DDI to go direct to voicemail i.e. the mailbox to check your mail.
00:21.15MacWeenieLuke-Jr: do you have it setup as a peer, friend or user?
00:21.23Luke-Jrfriend
00:21.43decanusfriend
00:21.44MacWeeniedoes it work with a regular softphone?
00:21.47Luke-Jrit's the same problem for both SIP and IAX2
00:21.48*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
00:21.50Luke-JrI haven't tried
00:22.19MacWeenieLuke-Jr: how did you switch between SIP and IAX2?
00:22.34MacWeenieis it just the same credentials?
00:22.43*** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com)
00:23.10MacWeeniei had 2 inbound calls using meetme, and we could talk to each other
00:23.50Luke-JrMacWeenie: almost-- one digit diff in username
00:23.58MacWeeniemy only problem now is not getting DTMF tones when i use the DID.. i do get dtmf tones from the softphone.  i would like to try out SIP because i think it sends DTMF separately or something
00:24.25decanuscan i have an incoming line go to asterisk mail?
00:26.03decanusshould i take that as a no?
00:26.22*** part/#asterisk Peaceful (n=Peaceful@70.98.162.62)
00:28.25Luke-JrMacWeenie: can you pastebin your context?
00:29.32Qwell[]decanus: explain
00:29.33MacWeenieLuke-Jr: sure, (i've never actually used pastebin)
00:29.40Qwell[]just when somebody calls, automatically go to voicemail?
00:30.12MacWeenieLuke-Jr: btw, i talked to the guy at sellvoip just now, and he was really helpful.. he said there is a known issue with some of teh DID providers with DTMF that may need some tweaking their end.  he's looking into it now
00:31.18MacWeenieLuke-Jr: which file do you want me to paste from?  iax.conf ?
00:33.04*** join/#asterisk iq|mobile (n=iq@71-214-5-20.omah.qwest.net)
00:34.06MacWeeniehttp://pastebin.com/617205
00:38.48Lino`just called cisco systems
00:38.58fuzzbawlFYI, Broadwing does a 50ms wink timeout
00:39.11Lino`goddamnit thats a buerocratcy
00:39.16Lino`cracy
00:39.21Lino`buerocracy
00:39.28Lino`one that makes you sick and crazy
00:39.51Lino`"i want to buy a support contract", "well call your european rep"
00:40.07*** join/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com)
00:40.29Lino`"well let me forward you to someone who may know what you need" *gets forwarded* [...] "didnt we just talk 5 minutes ago?" *lol*
00:42.58*** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
00:44.42TUplinkany ideas on how to make h323 work with nat?
00:45.32TUplinkQwell you there?
00:46.14*** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com)
00:47.03Luke-Jrhrmm
00:47.51TUplinkLuke whats up?
00:48.39Qwell[]no
00:49.51TUplinkhey Qwell... you said you got sip to work with double nat?
00:50.05Luke-JrTUplink: can't seem to get sellvoip to work
00:50.22TUplinkwhats sellvoip?
00:51.01MacWeenieLuke-Jr: might be worth it to setup 2 softphones takling to each other first
00:51.32MacWeenieto exclude a configuration problem.. helped me determine my dtmf problem was on sellvoip side
00:51.36austinnichols101TUplink: running double nat here
00:51.50TUplinkwant to explain it to me...
00:51.55austinnichols101it was painful
00:52.06austinnichols101describe your setup
00:52.08TUplinkyes.. .so i know
00:52.36TUplinkserver<-->nat<-->inet<-->nat<-->client
00:52.57austinnichols101firewall has it's own public/private IP pair?
00:53.06austinnichols101sorry server
00:53.15TUplinki think if i can get sip to work maybe ill be easer to get h323 to work...
00:53.22austinnichols101probably
00:53.30r_evolutionhey Mac
00:53.31TUplinkyea... 172.17/255.255.0.0 and a class c inet
00:53.36_Simoncould anyone help me out please? I keep getting this when trying to access voicemail app_voicemail.c:5064 vm_authenticate: Unable to read password
00:53.37austinnichols101k
00:53.45_SimonI dunno whats causing it
00:53.46r_evolutioni told you earlier that sometimes the provider has trouble passing DTMF properly :)
00:53.47austinnichols101what firewall (may not make a difference)
00:54.06TUplinkLinksys routers on both nats
00:54.07*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:54.11austinnichols101k
00:54.29TUplinkits all in sip.conf aint it?
00:54.42MacWeenier_evolution: crap, i must have missed your message
00:54.51austinnichols101asterisk needs to have externip and localnet settings
00:55.18TUplinkexternip...... dose a hostname work?
00:56.17_Simoncan anyone help me out please?
00:56.19MacWeenier_evolution: what do you usually do in that case?
00:56.35[hC]Anyone here using an SPA-941 (or any sipura) found problems with echo? (and potentially a solution?)
00:59.34*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
01:01.05TheCopsSomeone tried Asterisk with more then 50 IP Phone ?
01:01.11Qwell[]TheCops: no, never
01:01.20TheCops:/
01:01.23Luke-Jrare there any VoIP service providers that allow you to manipulate your own extentions.conf stuff directly? ;)
01:01.34Qwell[]Luke-Jr: any that let you use asterisk
01:01.34Lino`hm?
01:01.38Luke-JrTheCops: it's sarcasm
01:01.56TheCopshehe
01:01.56Luke-JrQwell: eh, I mean like origination and termination providers, not PBX hosts
01:02.05Lino`hmmm
01:02.15Qwell[]Luke-Jr: yeah...just run asterisk, and connect to them - and you can use your own extensions.conf
01:02.28[av]baniblah
01:02.31Luke-JrQwell: I'm talking about extensions.conf on their end
01:02.32[av]banipolycom bugs
01:02.37TheCopsQwell[], What are phones you was using ?
01:02.46Qwell[]Luke-Jr: Why?  All calls would go direct to you...you can do whatever you like
01:02.49Luke-JrQwell: eg, to setup fallback mechanisms and such
01:02.52Qwell[]TheCops: cisco
01:03.02Qwell[]Luke-Jr: asterlink lets you do fallback stuff - so does nufone now, I believe
01:03.05Luke-JrQwell: and to debug problems when customer support isn't available
01:03.37Luke-JrI'd rather direct extension editing that messing with some fancy UI
01:03.49Luke-Jrof course, I'
01:03.53Luke-JrI'd prefer AEL too, but...
01:04.36Qwell[]Luke-Jr: asterisk read it in, and it gets put into the asme structure as extensions.conf
01:04.37*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
01:04.46Qwell[]so...shouldn't be hard, if you use *
01:05.16litagehow does openpbx.org compare with Asterisk?
01:05.31Qwell[]litage: asterisk actually has developers working on it
01:05.44Qwell[]I don't think openpbx can say the same
01:06.08litageQwell[]: why are * developers working on openpbx.org?
01:06.13*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
01:06.24Qwell[]litage: try reading that again
01:06.44litageoh hahah
01:06.47litage=P
01:07.10TheCopsQwell[], Do you have some bug in the network and stuff like that ?
01:07.19litageQwell[]: i've read openpbx.org's pages and the wiki's openpbx.org page, but none of that gives more details other than "openpbx.org is a free software PBX"
01:07.27Qwell[]litage: exactly
01:07.32litagewhich is a broad, fairly useless, statement
01:07.43Qwell[]I'd be surprised if it even compiles
01:07.49litageah
01:08.05fuzzbawli'm off to home. later all, thanks for the help
01:09.02Luke-JrQwell: yes, I know. I was pondering translating outside of Asterisk
01:10.31Luke-Jrfwiw, I'm discouraged from doing any coding on Asterisk due to the licensing issues-- so a fork that doesn't break compatibility (as I've heard OpenPBX does) would be welcome to me =p
01:10.48Qwell[]"licensing issues"?
01:11.13mog_workopenpbx broke compatibility right away ^_^
01:11.19mog_workchanged ast to opbx
01:11.23mog_workall over code
01:11.34*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
01:11.36mog_workbut i agree with Qwell licensing issues?
01:11.37Qwell[]heh
01:11.58Qwell[]mog_work: We still keep copyright on the long disclaimer, right?
01:12.07Luke-JrQwell: I don't want Digium or anyone to use my code in non-open source programs
01:12.29mog_workyup Qwell
01:12.35Qwell[]only Digium would be able to
01:12.42Luke-JrI don't want Digium to
01:12.51Qwell[]and...only to specific programs, no?
01:14.04TheCopsAsterisk and Cisco IP phone are completly compatible between us ?
01:14.10TheCopseach other sorry
01:14.26mog_workdepends
01:14.30mog_worksip you are fine
01:14.37mog_workskinny you should talk to Qwell
01:14.46Qwell[]Luke-Jr: let's put it this way...by not contributing, nobody gets to use it
01:14.58Qwell[]yeah, so, Digium makes a buck or two...big deal
01:15.07Luke-JrQwell: with a compatible GPL-only fork, that is solved
01:15.22Qwell[]never gonna happen :)
01:15.32Luke-JrI have no problem with Digium making a buck or two-- or a few hundred-- but not using immoral means
01:15.38Qwell[]immoral?
01:15.48Luke-Jrproprietary licensing
01:15.51Qwell[]mog_work: Have you been killing kittens agin?
01:15.53Qwell[]again*
01:15.58mog_worki kill no kittens
01:16.06mog_worki do hit my dog some times
01:16.09TheCopsmog_work, I have a project of 50 IP phone and asterisk, I'm now searching for hardware...hehe
01:16.20mog_workget polycom 601
01:16.23mog_workor the other one
01:16.26mog_workthey rock
01:16.35TheCopsI have a 501 here and it rock yeah!
01:16.42TheCopsbetter then Cisco ?
01:16.46TheCopsOr just a money history ?
01:16.46mog_workwell
01:16.53mog_worki like ergonomics better on cisco
01:17.02mog_workbut interface is better on polycom at least in sip mode
01:17.10mog_worki dont have any sccp devices
01:17.14TheCopsWhat's about the sound ?
01:17.19mog_workgreat sound
01:17.23TheCopsOn both ?
01:17.23mog_workesp speaker mode
01:17.26mog_workyeah
01:17.26Qwell[]same sound, no?
01:17.36mog_workcisco uses polycoms stuff for speaker
01:17.38mog_workits the same
01:17.39Luke-JrMacWeenie: around?
01:17.46MacWeenieLuke-Jr: yep
01:17.48TheCopsmog_work, does polycom have again the restriction on buddies list ?
01:17.53TheCopsDo you know ?
01:17.56Qwell[]Luke-Jr: Well, meanwhile, nobody gets to benefit from your "kindness". :)
01:17.57mog_workyeah 7 users currently
01:18.00TheCopsduh
01:18.02mog_workat some point that will be fixed
01:18.04Luke-JrMacWeenie: can you pastebin how your access number is configured? in the ya interface
01:18.08TheCopsI guess
01:18.13mog_workwell soonish
01:18.20mog_workhow soon i dont know
01:18.22Luke-JrQwell: in theory, I could release a GPL-only patch
01:18.28Qwell[]Luke-Jr: sure, you could
01:18.31mog_workthat no one will use
01:18.34Qwell[]but you'd have to update it all the time
01:18.36Qwell[]look at chan_sccp
01:18.40*** part/#asterisk jasonpr (n=jasonpr@c-24-10-236-54.hsd1.ut.comcast.net)
01:18.42Luke-JrQwell: only if the code around changes
01:18.45Qwell[]AND make it backward compat
01:18.47mog_workpreach it Qwell
01:18.49Qwell[]Luke-Jr: yeah...it does...daily
01:18.56TheCopsmog_work, I think it have a queues XML interface for the 601
01:19.02Qwell[]Luke-Jr: look at chan_sccp sometime - you'll cry
01:19.20Qwell[]if he were to disclaim it all...none of that would be problematic anymore
01:19.23Luke-JrQwell: or Digium could stop using immoral licensing just for profit
01:19.26mog_workooh thats hot TheCops
01:19.40TheCopsDoes the hardware echo cancellation on Analog FXO board is better ?
01:19.41Qwell[]Luke-Jr: and they could go out of business, and stop producing asterisk...yeah, that sounds fair :P
01:19.44mog_worki wish everyone could come down to digium for a day
01:19.45MacWeenieLuke-Jr: I just have default settings
01:19.51TheCopsI  mean, for the price
01:19.57TUplinkhow do i setup a oh323 client... so that i can call it
01:19.59Luke-JrMacWeenie: well your defaults work =p
01:20.04mog_workbecause you all know i come to work in a suit and smoke a cigar and think of ways to be evil
01:20.11Qwell[]haha
01:20.17Luke-JrQwell: or they could run the business without immoral means
01:20.25TUplinkhaha to who?
01:20.34mog_workthe asterisk community of course
01:20.41denonLuke-Jr: if you dont like asterisk, go buy a nortel
01:20.44Qwell[]Luke-Jr: tell me, oh wise one...if Digium couldn't sell asterisk - why would they continue to write it?
01:20.56Luke-JrQwell: I never said they couldn't sell Asterisk
01:21.00mog_workwhy build a community and a whole ecosystem around product you release if not to be evil to them
01:21.11denonif you dont like digium, then you dont like asterisk .. like I said, go buy a nortel
01:21.33Luke-Jrdenon: nonsense
01:21.37Qwell[]no more GPL code for you!
01:21.41Qwell[]How's that for immoral? :P
01:21.52*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:22.02denonbaha
01:22.05Qwell[]mog_work: eww, your x is in color?
01:22.13mog_workwell 2 Qwell
01:22.13Qwell[]need to fix that
01:22.15mog_workblack and whit
01:22.16denonmog_work: quit trying to run X on that 286 digium gave you :)
01:22.16mog_workee
01:22.19Luke-Jrdenon: sure, but Asterisk is available as either black or white
01:22.20Qwell[]ahh
01:22.22mog_workhehe
01:22.40Luke-Jrdenon: but Digium insists all modifications are also both black and white, and white-only is not enough
01:23.04Luke-Jrthough I'm guessing they have black-only modifications
01:23.07denonLuke-Jr: I dont understand the problem .. if you like the confines of the product, you use it ... if you dont agree with them, you use something else
01:23.26mog_worklol there has been so many patches that made it into oss from be
01:23.32mog_worknot many went other way
01:24.03mog_worki dont have one in my collection
01:24.05Qwell[]Luke-Jr: You still have yet to explain how selling* is immoral
01:24.07denonEpson Equity II+
01:24.09mog_workyou run linux on it?
01:24.15denon286 8mhz or 12mhz (has a switch)
01:24.20denonnope, dos 3.3
01:24.22mog_workyou could probably get debian on it
01:24.23Luke-Jrdenon: Any change to Asterisk is required to allow Digium to close the source. I only want my changes GPL'd.
01:24.25Qwell[]Luke-Jr: It's not like they steal your copyright.  You can use the code whereever the hell you like
01:24.27mog_workwithout too much trouble
01:24.28denonthough it runs dos 5 just fine
01:24.36Luke-JrQwell: I have nothing against selling, just immoral licensing
01:24.40Qwell[]Luke-Jr: Asterisk will *ALWAYS* be GPL
01:24.44Qwell[]*ALWAYS*
01:24.53Qwell[]Nobody can EVER take it out of GPL - not even Digium
01:25.03Luke-JrQwell: then why does Digium demand non-GPL licenses?
01:25.06*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
01:25.09Qwell[]so that they can sell it
01:25.15Luke-Jryou can sell it with GPL
01:25.29ManxPowerI just ran SIP/RTP over WiFi for the first time.  Gawd, what lag.
01:25.29Qwell[]Luke-Jr: There are some things that, no, they most certainly cannot
01:25.32mog_workLuke-Jr, there are also cases where we would need to have it not be gpl
01:25.35mog_workfor example
01:25.38Qwell[]^
01:25.40mog_workwe sell be with a waranty
01:25.48Luke-JrGPL does not forbid a warranty
01:26.15mog_workor there are certain software parts that can not be gpl do to the other party
01:26.24Qwell[]such as g729, no?
01:26.26Luke-Jrthen such parts cannot be used
01:26.26mog_workneedless to say
01:26.38Luke-Jror distributed linked, anyway
01:26.56mog_workthe big reason why people have been fine with digium's control as we have been benevolent at least in the minds of 99% of the community
01:27.07denon[brainwashing]
01:27.09*** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com)
01:27.15mog_workand a large amount of development has come from our doors
01:27.22mog_workand continues to do so
01:27.27denon[propaganda]
01:27.36Lino`lol
01:27.56denonhehe
01:28.02TheCopsmog_work, a network with 50 Polycom 601 and a dual xeon 3ghz 2gb of ram, 6 hard drive 73gb raid 5, dual power supply, with a beautiful HP Procurve PoE switch, do you think it will be stable ? hehe
01:28.07*** join/#asterisk jasonpr (n=jasonpr@c-24-10-236-54.hsd1.ut.comcast.net)
01:28.07mog_workbut if you dont feel that way fine
01:28.09TheCopsCan be cool
01:28.11jasonprhey does anyone know how to do a Meetme through the Manager Interface (socket)?
01:28.13mog_workwe arent forcing anyone
01:28.20Luke-Jranyway, those decisions are why I simply work around bugs instead of fixing them
01:28.21mog_worksure
01:28.29mog_workgood for you
01:28.41TheCopsI guess it will be stable and a good network
01:28.42mog_workits free software you have that right
01:28.44mog_workor any right
01:29.24Lino`:D
01:29.29Lino`who is anti community?
01:29.30Luke-JrI don't care if they profit either
01:29.41Luke-Jrso long as the profit comes morally
01:29.43Lino`XD
01:29.50Luke-Jrproprietary licensing is anti-community
01:29.51mog_workso you are typing on a computer that is closed source how is that moral?
01:29.55alephcomIsn't all profit moral?
01:29.58mswLuke-Jr: nah
01:30.09Lino`well i do produce closed source software, is that moral?
01:30.13Luke-Jrno
01:30.17mog_workyou are connecting to us over prop. networks
01:30.23mswLuke-Jr: it's a different path for people (buysers) with different needs
01:30.24mog_worktalking to people on prop. systems
01:30.24mog_worketc
01:30.38mswbuyers
01:30.49Lino`see, theres closed source everywhere, i do open source, i do closed source. thats the way it is.
01:30.53Luke-Jrmsw: Nobody needs a lack of code or rights
01:30.54TheCopsdigium found the right way to get profit with the hardware...
01:31.07mog_workheh
01:31.59Qwell[]Luke-Jr: Regardless of what you think, there are legal reasons why Digium can't release the source to some things.  And in order to link to those things, they need to be able to close the source
01:32.03[av]baniblah
01:32.10Luke-JrQwell: so don't link to them
01:32.14[av]banipolycom keeps ringing after queue has hung up... anyone know why?
01:32.23mog_workshow channels?
01:32.26[av]baniall other phones in the queue (grandstream ,cisco) dont have a problem
01:32.27mog_workare any up?
01:32.30[av]banishow channels shows none
01:32.32[av]baniempty
01:32.33[av]baninada
01:32.33[av]banizip
01:32.34[av]banizero
01:32.39Qwell[]Luke-Jr: So naive :)
01:32.50[av]baniall the other phones in the queue have NO problems
01:32.55[av]banionly the polycom keeps ringing like a moron
01:33.40Luke-JrQwell: also note that technically, the GPL allows exceptions for the GPL'd program to link to immoral libs
01:33.52Luke-Jrwhich isn't *much* better, but terms I would accept
01:34.09mswLuke-Jr: only if the "immoral" libs are "system" libs
01:34.18Luke-Jrmsw: no, any libs
01:34.32Luke-Jrit just needs to be an explicit exception for non-system ones
01:34.38mswLuke-Jr: otherwise the licensor has to explicitly allow linking to GPL-incompatbile (a.k.a "immoral")
01:35.17[av]banii can call it directly (eg Dial(SIP/bla)) and it works perfectly
01:35.22[av]banionly queue calls cause the problem
01:35.23Luke-Jrmsw: which is better than requiring the licensor to allow immoral distribution altogether
01:35.57mswLuke-Jr: linking isn't the only reason to have people disclaim, and for dual licensing
01:36.10Qwell[]$20 says Luke-Jr uses sun java
01:36.13Qwell[]any takers?
01:36.17Luke-Jrmsw: dual licensing is exactly why I refuse to contribute to it
01:36.20Luke-JrQwell: nope
01:36.32mog_workLuke-Jr, as much as i wish we lived in richard stallman's world we don't.  Without compromising from time to time you cant win, you just  come off as crazy.
01:36.51Luke-Jrmog_work: you can't win if you compromise
01:37.02mog_worki disagree
01:37.04Luke-Jr... too much, anyway
01:37.05mog_workthats how you win
01:37.22mog_worki admit sometimes you have to fight
01:37.28Luke-Jras I said, I would accept terms allowing linking exceptions
01:37.44Luke-Jrbut not terms that put the code *I* write under a closed license
01:37.58Luke-JrMacWeenie:  :\
01:38.21mswLuke-Jr: you're also not going to contribute to OpenOffice.org or mozilla?
01:38.38r_evolutionhaha im such a fucking geek... you know... i sit here... and i just watch the CLI output run...
01:38.38Qwell[]it's possible for your code to be put under commercial licenses, even if you release it as GPL, isn't it?
01:38.45r_evolutionover and over and over
01:38.48*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
01:39.00Qwell[]BSD is GPL compat, right?
01:39.09mog_workits one way Qwell
01:39.13Luke-Jrmsw: nope, neither project would be worth it even if GPL'd only
01:39.38TUplinkis a stun server hard to setup?
01:39.40Luke-Jrmsw: they're both slow and bloated; KOffice suits my office-needs and Konqueror is superior to FireFox
01:39.47Qwell[]I suppose I went the wrong way, heh
01:40.14Luke-JrQwell: The GPL is a commercial license
01:40.18mswLuke-Jr: wait...
01:40.25mswLuke-Jr: Qt is a core component
01:40.37mswLuke-Jr: and trolltech dual licenses, do they not?
01:40.45Qwell[]msw: They most certainly do
01:40.47Luke-Jrmsw: indeed, but I don't contribute to Qt either
01:41.03Luke-JrKOffice and Konqueror are GPL-only
01:41.31Qwell[]and you don't play mp3's or dvds either, I assume?
01:42.03Luke-Jronly with GPL'd software
01:42.26mog_workare you in states Luke-Jr ?
01:42.33Qwell[]funny, because that's immoral
01:42.44Qwell[]unless you pay patent royalties
01:42.46Luke-Jrmog_work: no, I don't care if it's illegal or not
01:42.47mog_workonly stateside and in most of civ. world
01:42.49Luke-Jrpatents are immoral
01:43.05Qwell[]and patent infringement is perfectly moral...gotcha
01:43.09r_evolutionwait
01:43.15Luke-Jrviolating immoral laws is
01:43.34mswLuke-Jr: but you shouldn't distribute a GPLed program if you can't give people the IP rights to use/run it
01:43.42mog_workwell by my moral values kicking you would be fine
01:43.45Luke-Jrmsw: "IP" is fiction
01:43.47mog_workim sure you would think otherwise
01:43.53r_evolutionyeah im just going to totally ignore that one... you're not REALLY arguing that any sort of patent of any nature is 'immoral'
01:43.56r_evolutioni know im coming in like...
01:43.59mog_worklol ip is fiction
01:44.00mswLuke-Jr: it's "immoral" to even *write* a gPLed program to play DVDs or MP3s
01:44.02r_evolutiongod only knows how far into this one
01:44.03Qwell[]...
01:44.11Qwell[]mog_work: Wasn't IP the whole BASIS of his argument?
01:44.22mswCopyright is IP
01:44.22mog_workwithout ip there is no hardware
01:44.24r_evolutionbut seriously... you're not really trying to say that a patent or license is a wrong thing
01:44.28mog_workwithout hardware there is no software
01:44.30mswand the entire GPL is built on IP
01:44.37mswerm
01:44.39mswcopyright
01:44.45Luke-JrThe GPL is built to reverse copyright onto itself
01:44.46mog_workyou are typing on a computer that is all prop "immoral" in your own words
01:44.53mog_workno Luke-Jr
01:44.55mog_workits not
01:45.01mog_workgpl is ip mangment
01:45.06Qwell[]GPL *RELIES* on copyright
01:45.14Qwell[]Without copyright - there is no GPL
01:45.14Luke-JrGPL enforces what should be in law
01:45.18Luke-Jrnothing more
01:45.24mog_workexactly Qwell
01:45.47Qwell[]oh fuck that
01:45.54Qwell[]Dell bought Alienware :(:(:(
01:45.58Luke-Jrcopyright should be abolished and anti-plagerism laws should be defined to require source release for protection
01:45.59mswLuke-Jr: no, that's backards
01:46.03mswbackwards
01:46.07mog_workeh Qwell
01:46.10Qwell[]mog_work: /. :(
01:46.17Qwell[]That makes me very sad
01:46.29r_evolutionWould if help if I bought a prostitute for you?
01:46.32Qwell[]umm
01:46.37Qwell[]Luke-Jr: Without copyright...
01:46.41mog_workim sure Qwell's wife would love that
01:46.42Qwell[]people can like...
01:46.48Qwell[]take your code...and like...
01:46.48r_evolutionok... fine... a stripper
01:46.52Qwell[]use it "immorally"
01:46.54*** join/#asterisk Kumbang (n=kumbang@167.205.24.5)
01:47.05Qwell[]please make up your mind
01:47.06Luke-JrQwell: how so?
01:47.13alephcomNo kidding.  Without copyright you can get that I wouldn't be wasting my time pounding my keyboard tonight.
01:47.14Qwell[]because NOTHING would stop them from copying it
01:47.20mog_workLuke-Jr, do you use immoral hardware? drive immoral car? sleep in an immoral bed?
01:47.24[av]baniwell this sucks. cant tell if its a polycom bug or an app_queue bug.
01:48.00Luke-JrQwell: anti-plagerism laws prevent them from copying it immorally
01:48.10Qwell[]umm
01:48.11alephcom[av]bani: Blame it on Polycom.  That's easier and it's harder for people to say "Fix it yourself" :-)
01:48.11mswLuke-Jr: sounds like copyright
01:48.12mog_workdid Luke-Jr just /ignore me?
01:48.34Luke-Jrmog_work: no, I just ignore ridiculous comments ATM
01:48.52mog_workhow is comparing software to any other copyright object ridiculous?
01:48.53Luke-Jras I am busy outside IRC
01:49.08Luke-Jr'copyright' attempts to turn information into property
01:49.10Luke-Jrwhich it is not
01:49.10mswLuke-Jr: I think it's immoral to tell people what to do with their inventions -- their very _thoughts_
01:49.23Luke-Jrmsw: exactly
01:49.45Luke-Jrmsw: copyright aims to do just that
01:49.59Luke-Jrafter all, once you tell me your thought, it is my thought now also
01:50.03mswLuke-Jr: you're saying that everyone should make everything they think or create available to everyone, for the common good of mankind (which is very utopian)
01:50.25mswLuke-Jr: but you can not claim it as your own
01:50.35Luke-JrI'm saying if someone receives something, they can do what they want with it
01:50.38mswLuke-Jr: you can enhance it, build on it, but my thought is still a part of that
01:50.46Luke-Jrmsw: claiming it as your own would be plagerism
01:50.57Luke-Jras I have said, we need laws against that
01:51.39Qwell[]as fun as this little chat is, I've got better things to do...
01:51.41Qwell[]bbl
01:51.47alephcomaka copyright
01:52.01mswI think that this kind of philosophy stifles creative expression and invention
01:52.15mog_workbye Qwell
01:52.40alephcomlol, I hope it's good
01:52.46r_evolutionchange your name to immoral_mog_work
01:52.50mog_workyeah but they wont tell me how to make it myself
01:52.55Zodiacalanyone know why my ip communicator's status says "no CTL installed" ? whats CTL?
01:53.06mog_worki am closed source r_evolution getting recipe from my parents would be quite a trick
01:53.11Zodiacalanyone goten ip communicator to work with chan_sccp
01:53.14mog_workand require several ndas
01:53.37r_evolutiondamn you and the immorality of your family
01:53.39mog_workyou should bug Qwell i guss Zodiacal as he is only chan_sccp person i know
01:53.41mog_workbut he is gone
01:53.54mog_worki know we have a long history of "dont kiss and tell"
01:54.02Primerso there is absolutely no way to change the bitmap on a 7960 dynamically?
01:54.03r_evolutionyour immoral computer... your immortal connection... your entirely immortal clothes
01:54.29r_evolutioni'd think it was more kiss and don't tell
01:54.56Luke-JrMacWeenie: poke
01:55.21mog_worklol r_evolution
01:55.23mog_workwell im off
01:55.25mog_workpeace
01:55.31file[laptop]ttyl
01:55.47r_evolutiondisappearfs?
01:55.55Zodiacalhrmm....
01:55.59r_evolutionit shouldnt be zodiacal
01:56.13r_evolutionbut i dont have time to help you tonight :)
01:56.16r_evolutioni am leaving in.....
01:56.17r_evolution4 minutes
01:56.22r_evolutionto pawn this girl off on my homeboy
01:56.24r_evolutionso she'll leave me alone :-D
01:56.29r_evolutionshe's all aasking about him and shit now...
01:56.31r_evolutionRAWKKK
01:57.46Zodiacalr_evolution can i take 1 of those mins for you to look at my sccp.conf http://pastebin.com/617313
01:58.02Zodiacalwhen adding a second sccp device do i just append the entries like that?
01:58.16Zodiacalor is there a seperate context [] i need for each device, line?
01:58.48Zodiacaldevices] = [devices]
01:58.49r_evolutionoh... dude i dont mess with sccp at ALL
01:58.55Zodiacaloic
01:59.55r_evolutionhttp://www.voip-info.org/tiki-index.php?page=Asterisk+SCCP+channels
01:59.56r_evolution?
02:00.12r_evolutionK!
02:00.13r_evolutionOUT!
02:00.39SedoroxI need to play with sccp.. need to get asterisk to talk to ccm :/
02:01.00[av]baniick
02:01.56Zodiacalsccp works fine on my hardphone
02:02.04Zodiacali set it up the same on the softphone but it doesnt werkie
02:02.29ZodiacalCTL is some kind of cisco certificate or somthing :/
02:02.34Zodiacalper gugle
02:08.51*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
02:09.06*** join/#asterisk juice (i=1000@209.33.109.224)
02:10.13*** join/#asterisk bjohnson (n=bjohnson@i216-58-43-154.cybersurf.com)
02:10.24harryvvanyone here have the ip500 and have set it up to flash over to another incomming call?
02:11.26*** part/#asterisk _Simon (n=IRC@i216-58-40-193.cybersurf.com)
02:22.01Qwellmog_work: immoral dinner?  haha
02:22.22tzangerblitzrage: quick dumb question... when carving up a variable, how do I take the first 3 digits?  I thought ${VAR::1} would chop off the last digit (keeping all but the last digit)
02:22.55Qwelltzanger: ${VAR::3} should get the first three
02:23.20*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
02:24.05tzangerQwell: nope
02:24.20tzangerSet(VAR=1234), ${VAR::3} is 1234
02:24.47Qwellmaybe you need the 0
02:24.50Qwell${VAR:0:3}
02:25.17Qwellw00t, new antenna
02:26.09tzangerthat did it Qwell, thank you
02:26.27astra^^Mar 22 20:18:10 WARNING[2467]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 212299640@xxx.xxx.xxx  for seqno 1 (Critical Response)
02:28.01astra^^any one please help me.. what does this message signify . why am i gettin this..
02:30.11astra^^helloo ... :/
02:36.53tzangerMar 22 21:36:20 WARNING[22374]: app_voicemail.c:5677 vm_box_exists: VM box 201@darlene_wayne exists, but extension 2015, priority 104 doesn't exist
02:37.00*** join/#asterisk tengulre (n=tengulre@222.90.66.4)
02:37.00tzangerI thought priorityjumping was disabled by default
02:37.01tzangerwtf
02:38.19astra^^anyone in the room...  :'
02:39.39blitzragetzanger: uhhh -- you need to add a 0 :)
02:39.53blitzragetzanger: there's been a lot of "stupid/dumb" questions flying around lately :)
02:40.08tzangerblitzrage: heh
02:40.12blitzragetzanger: nope -- its on by default in 1.2
02:40.37tzangerblitzrage: sucks.  ok I'll just deal withthe warning for now since there are some jumping things in other contexts this person has
02:41.02blitzragetzanger: ahhh gotcha -- you can disable it obviously if you needed
02:41.09*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
02:41.18blitzragetzanger: better -- disable it, then just add the 'j' flag where you need to still use it
02:41.36tzangerblitzrage: yeah that's the eventual plan, just not tonight :-)
02:41.45blitzragethat'd get rid of your warnings (although, if your DP is complicated, at the risk of screwing something up :))
02:42.01blitzragetzanger: I hear yah -- I'll be up for a few more hours working on this training material
02:44.26tzangerblitzrage: I have a contract here for a voicemail system for about 700 mailboxes
02:44.39tzangerand mailboxexists caches the voicemail.conf file  :-(
02:50.07QwellFuriousGeorge: I don't foresee that being a hit single.  Sorry...
02:50.15FuriousGeorge:(
02:50.30FuriousGeorgeits been in my head for like a month already
02:51.09astra^^Mar 22 20:18:10 WARNING[2467]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 212299640@xxx.xxx.xxx  for seqno 1 (Critical Response)
02:52.14FuriousGeorgei even got a verse about:  it treats me very nicely, with  S-I-P device-eys
02:52.40*** join/#asterisk w32 (n=David@adsl-70-227-190-108.dsl.sbndin.sbcglobal.net)
02:53.40w32Hi, for an AAH server to function properly should I have a static IP address ?
02:54.01FuriousGeorgeit helps
02:54.05w32*static wan IP address I mean
02:54.12FuriousGeorgei have pretty good luck lately with *.dynu.com addresses
02:54.14FuriousGeorgeor dyndns
02:54.35w32what kind of issues can I expect using dynaic dns services ?
02:54.52w32*dynamic...ok I cant spell
02:55.24FuriousGeorgei dunno if its related but sometimes i get a peer that becomes unreachable and i gotta start reloading iax2
02:55.50FuriousGeorgethis is between two asterisk servers talking IAX
02:56.11FuriousGeorgei set them up as host=asterisk1.dyndnu.com etc
02:56.29w32Well, I'm planning on using broadvoice and voipjet these are sip right ?
02:56.33*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
02:57.15FuriousGeorgewith sip i set externip to my dyndns provider and i get no problems, but i just play with a free gizmo account for testing
02:57.18xachenvoipjet is iax
02:57.23xachenbroadvoice is sip
02:57.50w32The thing is I don't want it to become a support/availability  headache....Ahhhh, I didn't realize voipjet was iax
02:58.17w32I'll bbl damnit.....
02:58.48FuriousGeorgeif ur * server is the client side i wouldnt worry about it too much.  it is only a minor issue when you need to connect a peer to * remotely
02:58.57FuriousGeorgeeven then its doable without much extra effort
03:01.36xachenIAX is evil
03:01.42blitzrageFuriousGeorge: LOL!!!!!!!!!!!!!!!
03:02.30FuriousGeorgeblitzrage: you like my lovely asterisk trunk? (check- it-out)
03:03.35tzangerasterisk cdrs suck :-(
03:05.17blitzrageuse SER :D
03:05.24blitzrageFuriousGeorge: yah -- pretty funny :)
03:05.41FuriousGeorgeblitzrage: maybe ill post it on voip-info :)
03:05.43FuriousGeorgebbl
03:06.40*** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net)
03:08.09blitzragehrmmm... having a tough time definding what a dialplan function is -- any suggestions?
03:08.24blitzrageI know WHAT they are -- just can't seem to come up with a good way of explaining it succinctly
03:15.05tzangerblitzrage: I've never been able to tell why some dialplan magic is functions and some oare applications
03:15.18tzangerI *think* functions can be used as variables, and applications can't
03:17.16blitzrageyah, basically -- you can have Set(CALLERID(num)=8885551212), or, you can do something like NoOp(${CALLERID(num)})
03:17.26*** join/#asterisk walalang (n=sample@203.177.13.60)
03:17.36blitzragewhich basically executes an "application within an application"
03:17.38blitzragekind of idea
03:18.14walalangto what platform is asterisk do well? linux? bsd?
03:20.29x86anyone from san jose area?
03:20.35x86specifically, mountain view?
03:23.05blitzragewalalang: use Linux
03:23.54walalangblitzrage: if linux, which disto? redhat, etc?
03:24.03x86gentoo ;)
03:24.05walalangi heard its going great with centos
03:24.13Qwellcentos works
03:24.17Luke-JrHow does Asterisk associate a registration with an IAX2 context?
03:24.44blitzragewalalang: what distro do you like? Whatever one you're going to name is perfect
03:24.51walalangwhat if boost in perf?
03:25.30walalangblitzrage: curently i have RH EL4
03:25.31blitzragewalalang: personally -- I like CentOS -- although I've heard good things about FC5 (I didn't like the previous FC releases)
03:25.32Luke-Jror rather, how can I specify the IAX2 context a registration should use?
03:25.37blitzragewalalang: use that then
03:26.03walalanghmmmm great
03:26.07blitzrageLuke-Jr: registrations are only used to determine where the far end is -- you still need a user (for incoming)
03:26.45Luke-Jrblitzrage: I know, but how do I specify which user the server contacts?
03:26.58QwellLuke-Jr: in your register line
03:27.03walalangnow is there a preparation docs available, meaning requirements before implementing asterisk? checklist or the like?
03:27.10*** join/#asterisk Kumbang (n=kumbang@167.205.24.5)
03:27.18blitzragewalalang: www.oreilly.com/catalog/asterisk
03:27.29blitzragewalalang: if you're cheap -- www.asteriskdocs.org
03:27.57blitzrageregister => user:pass@server
03:28.02Qwellblitzrage: http://svn.digium.com/viewsvn/asterisk/trunk/ :P
03:28.11blitzrageQwell: blah! :)
03:28.25blitzragedamnit -- why are dialplan functions so hard to define :)
03:28.25QwellIt's just view, actually
03:28.36Luke-Jrblitzrage: isn't that the user on the remote server?
03:28.43Qwellblitzrage: They can be set and read.  It's like a variable and an app
03:28.57Luke-JrAstX is registering with AstY. How does AstY know what user on AstX to auth as?
03:29.20*** part/#asterisk brockj49464 (n=brockj49@63.87.56.236)
03:29.28blitzrageLuke-Jr: not user, peer -- and that tells the remote system where you are -- then you need a matching user for incoming calls from that system
03:29.55blitzrageLuke-Jr: because AstX tells AstY which user it is registering with in the register line
03:30.16Luke-Jrblitzrage: the register line specifies the user on AstY, not AstX...
03:30.21Luke-Jror not?
03:30.22blitzrageLuke-Jr: and -- in your question, it doesn't -- AstY needs to know what username to use to send a call to (and to auth with)
03:30.35walalanghow much an E1 card cost?
03:30.40blitzrageLuke-Jr: no -- all the register line does is tell the far end what IP address you're at
03:30.40Qwellwalalang: 1 port?
03:30.47blitzragewalalang: www.digium.com to find out
03:30.47*** join/#asterisk inv_Arp (n=junya@fiudial2-55.fiu.edu)
03:30.47walalang4
03:30.52QwellWhat he said
03:30.53blitzrageabout $2000 US
03:30.59Qwellblitzrage: bit less..
03:31.07blitzrage$1995 :)
03:31.08Luke-Jrblitzrage: so how does the far end know whose IP that is?
03:31.08*** join/#asterisk Kumbang (n=kumbang@167.205.24.5)
03:31.11walalanghmmmmm
03:31.14Qwelllike, $1595?
03:31.30blitzrageLuke-Jr: because you will have a matching peer definition with host=dynamic
03:31.43blitzrageLuke-Jr: www.asteriskdocs.org <-- Read the book online
03:31.50Qwellheh, much less
03:31.56Qwellvoipsupply claims $1345
03:32.06Luke-Jrblitzrage: so both AstX and AstY need to use the same usernames for each other?
03:32.14walalangany other good docs aside from online help and oreilly?
03:32.30Qwellwalalang: buy the o'reilly book
03:32.35blitzragewalalang: www.sokol-associates.com and come to a training class
03:32.59blitzragethe O'Reilly book is the best Asterisk book out so far :D
03:33.08walalangok
03:33.24Qwellblitzrage: That wouldn't maybe be a biased opinion, would it? ;)
03:33.30blitzrageQwell: no way
03:33.36Luke-Jrblitzrage: I see no way to read it online. Only download links.
03:33.40blitzrageQwell: totally impoartial review
03:33.48blitzrageLuke-Jr: ummmm... so download it? :)
03:33.59Qwellgoogle it, let it convert to html
03:34.05blitzrageQwell: good idea
03:34.14blitzrageQwell: I think its tar'd or zip'd though
03:34.19Qwelloh
03:34.29Luke-Jrblitzrage: maybe reading a book is way more than anyone should need to do for something as simple as specifying a username?
03:34.43blitzrageLuke-Jr: then why are you asking a question if its so eas?
03:34.46QwellLuke-Jr: or you could pay somebody to do it for you
03:35.08blitzrageLuke-Jr: if you don't want to learn to read to figure something out in Asterisk, you have little hope in making it
03:35.11Luke-Jrblitzrage: because the example config file is lacking?
03:35.19blitzrageLuke-Jr: then write a better one
03:35.28Luke-Jrmaybe I'll just use SIP, that seems to work
03:35.35blitzragewhen I started, I didn't even have the luxury of documentation
03:35.52Qwellblitzrage: pfft, you had the code! :P
03:36.07blitzrageQwell: about as good to me as a Korean bible :)
03:37.10QwellLuke-Jr: http://linuxhelp.tv/articles/2006/03/21/joel-jjshoe-gets-a-new-desktop
03:37.17QwellLuke-Jr: Yes, go to SIP.  It's easy.
03:38.25Luke-JrQwell: was that a serious comment or sarcasm?
03:38.28blitzragethe configs for a simple SIP config should pretty much be interchangable with an IAX config
03:38.33QwellLuke-Jr: see the above link
03:39.08Luke-Jrblitzrage: SIP doesn't have the same concepts of authentication, it seems
03:39.30Qwellyeah, SIP is an authenticationless protocol
03:39.31Luke-Jrblitzrage: and SIP's config allows specifying a context for registration
03:44.55*** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net)
03:45.25gandhijeeanyone know what type of encoding a CDMA cell phone call uses?
03:46.21Igg-manWhy would my other party only hear me for a few seconds, but I can hear them just fine?
03:47.41gandhijee?
03:47.48gandhijeeover zap, IAX, sip??
03:47.53Igg-mansip
03:48.43gandhijeeyou behind a firewall/NAT?
03:48.47Igg-manYep
03:48.51Igg-manso is the other end
03:48.53gandhijeeyou open all the ports?
03:49.09Igg-manWhat should I open?
03:49.14gandhijee5060
03:49.15Igg-man5060?
03:49.16Igg-manok
03:49.19gandhijeeand 10k to 20k
03:49.28Igg-manActually, I think that 5061 in my case
03:49.40blitzrageLuke-Jr: show me
03:49.40gandhijeeif u edit the rtp.conf u can drop to 15k
03:49.48Igg-manI don't think my router lets me forward by port range
03:49.53gandhijeeIgg: did you set the listen port?
03:50.07gandhijeewell you need to foward 10k to 20k to your asterisk server for RTP
03:50.08Igg-manIs that the 5060 port?
03:50.11gandhijeeand so does the other end
03:50.23Igg-manHmm... not running asterisk
03:50.36Igg-manwell, not here anyway.  I'm connecting directly to FWD
03:50.47gandhijeeexplain your setup to me
03:50.48blitzrageI guess if you dont' specify a user on the system which is registering, you may be able to direct the call to the default context....
03:50.56blitzrageI'm not even sure if that'll work though
03:50.59Igg-manI'm connecting directly to FWD using a linksys pap2
03:51.04Igg-man(on the 2nd port)
03:51.18*** join/#asterisk bmg505 (n=leon@dsl-146-27-132.telkomadsl.co.za)
03:51.24Igg-manthe other end is using a softphone
03:52.07gandhijeeigg: so there is no asterisk server on your end i assume
03:52.09Igg-manHe is going to try another softphone
03:52.24Igg-manThere is, but it's outside the NAT on its own public IP
03:52.34gandhijeeyour asterisk server is?
03:52.35Igg-manI'm just playing with stuff now
03:52.39Igg-manyep, it is
03:52.41gandhijeeI C
03:52.54gandhijeedid you set the option for behind nat?
03:52.55gandhijeei don
03:53.03gandhijeet remember what it is.
03:53.18Igg-manBTW, that's annoying... you have your choice of running with the NAT support and talking to the asterisk server, or disabling the nat support, and talking internally
03:53.21gandhijeeare you both peering to the same asterisk server?
03:53.33Igg-mannope, using fwd
03:53.40gandhijeei am confused.
03:53.46Igg-manS'okay
03:54.15gandhijeeso you are peering to FWD via SIP w/ the PAP2
03:54.27gandhijeeand your buddy is peering to FWD w/ a softphone?
03:54.29gandhijeecorrect?
03:54.33Igg-manyep
03:54.38Igg-manMythPhone, actually
03:54.50Igg-mannow he's using the client that fwd has, I think its working better
03:54.50gandhijeephone w/ MythTV?
03:54.54Igg-manYep
03:54.57gandhijeeok
03:55.07Igg-manI think it must be mythphone, we are still connected using the windows softphone
03:55.23gandhijeeso why are u asking this question in the asterisk room if no party is using asterisk?
03:55.40Igg-manSmart people?
03:55.52Igg-manIt was the channel the IRC client opened up to
03:55.53Igg-man:-)
03:55.59gandhijeeO
03:56.05Igg-manI figured it would be a common problem
03:56.28gandhijeei always had problems gettin my parents phone to reg to my asterisk server when it was sip based
03:56.36gandhijeeand my server was behind the firewall
03:56.42Igg-manso, you switched to IAX?
03:56.47gandhijeeso i said F it and made it IAX
03:57.00Igg-manWhat are your parents using for softphones?
03:57.04Igg-manor... hardphoens?
03:57.06gandhijeehardphone
03:57.09Igg-manCool
03:57.18Igg-manAnalog adapter?
03:57.23gandhijeenope
03:57.26gandhijeeIAX hardphone
03:57.32Igg-mansweet
03:57.33Igg-manwhat is it?
03:57.36gandhijeepeers directly to my asterisk server
03:57.49gandhijeei got it from IareaPhone
03:57.59gandhijeeit uses the PA16xx chipset
03:58.02Igg-manNow, if I remember reading the docs right, the IAX does NAT much better, right?
03:58.12gandhijeeseems to
03:58.16gandhijeeonly 1 port to open
03:58.18gandhijee4569
03:58.25Igg-manTCP or UDP?
03:58.31gandhijeeTCP i think
03:58.45Igg-manthat sounds more secure too, or much easier to firewall
03:58.51gandhijeeyeah
03:59.04gandhijeei think there is encryption on the IAX link too
03:59.24gandhijeeor else u can use some shit to play back the RTP stream
03:59.28gandhijeeif its captured
04:00.26gandhijeeif it is SIPs based
04:00.27Igg-manI'd bet
04:00.30Igg-manCool
04:00.34Igg-manWas the phone cheap?
04:00.47gandhijeelike 60 bux
04:00.56gandhijeefwd does IAX peering too
04:01.56Igg-manOh, does it?
04:02.04Igg-manHow does that work if the other end doesn't support IAX?
04:02.34justnulling2does anyone use freepbx?
04:02.41gandhijeeFWD handles all that
04:03.09gandhijeelike a IAX to SIP bridge
04:03.32gandhijeei have SIP phone internally that asterisk's can get a hold of
04:03.37gandhijeewhen my parents call
04:03.52gandhijeeit just takes the IAX and "moves" it to SIP
04:04.58*** part/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
04:08.16Luke-Jrblitzrage: hmm?
04:08.24Luke-Jrblitzrage: register => username@context
04:09.38*** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com)
04:09.55Luke-Jrblitzrage: I think it's still somewhat of a hack-- would be nice to be able to drop username from the register line too
04:10.41Luke-Jrbtw, fixed my problem by adding 'accept=...' to my context. Is the default to reject all?
04:11.04*** join/#asterisk I-MOD (n=I-MOD@68.62.165.168)
04:19.42Nuggetcan anyone recommend a vendor where I can buy a cisco phone and get the service agreement and everything all in one purchase?  I don't want to go through the ordeal I had when I bought my 7960s.
04:21.22QwellNugget: cdw?
04:21.55Nuggetthat's where I had my previous ordeal, and searching for "7970" turns up nothing there anyway
04:22.06Nuggetmaybe insight direct can do it.
04:22.09Qwellheh
04:22.32Nuggetyou've convinced me to play with sccp.  :)
04:22.37Qwell;]
04:24.20*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
04:25.00astra^^do i have to follow the whole process of downloading the .do file and others while i upgrade from a single g729 to 2
04:25.55astra^^or can i register directly wid the licence number
04:28.17*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
04:28.42Beirdohmmm
04:29.09astra^^?
04:29.10Beirdois there a way to reduce the number of threads asterisk uses on a particular system?
04:29.25*** join/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
04:33.18blitzragerm -f /usr/bin/asterisk
04:33.25Beirdoheh
04:33.28blitzrageactually.. I think its sbin :)
04:33.31Beirdothat's not what I meant :)
04:33.34blitzrage:D
04:33.41blitzrageI don't have a *good* answer for you, sorry :)
04:34.08Beirdoon my linode, asterisk is running like 18 or 19 threads
04:34.19*** part/#asterisk io_error (n=error@mdsnwikwbas08-pool7-a82.mdsnwikw.tds.net)
04:34.19Beirdojust seems a little excessive
04:34.31astra^^do i have to follow the whole process of downloading the .do file and others while i upgrade from a single g729 to 2
04:34.40astra^^*.so
04:35.21*** join/#asterisk tainted_ (n=identd@ppp-71-134-51-75.dsl.irvnca.pacbell.net)
04:36.33*** join/#asterisk jpeeler (n=jpeeler@c-71-226-105-137.hsd1.sc.comcast.net)
04:40.07Nuggetit
04:40.15Nuggetit's in /usr/local/ on freebsd.  :)
04:44.06blitzrage:)
04:44.22blitzragewoohoo -- I got to mention Choose Your Own Adventure in a presentation
04:46.43Beirdoblitzrage, found my own answer experimentally
04:46.57Beirdodisable loading of useless chan_*.so
04:46.59astra^^can i get some help out her please
04:47.04Beirdolike all the modems, skinny, etc
04:47.14Beirdoseems to be a thread per channel type
04:47.20blitzrageBeirdo: oh nice -- seems to kinda make sense
04:47.53blitzragedamn it just got cold in my room
04:48.16Beirdomy goal is to use less memory on that linode :)
04:48.27Beirdoit took a swapping shitfit earlier this evening
04:48.49Beirdorestarted apache, asterisk, mysqld, and went from 250MB in swap to 0MB
04:49.01blitzragewow
04:49.03BeirdoI'll tweak the other two AGAIN tomorrow :)
04:49.17blitzrageI upgraded from 1.2.1 to 1.2.5 on a box and went from 1.38 to .38 load avg :)
04:49.28Beirdonice
04:49.33BeirdoI should upgrade sometime
04:49.47blitzrageyah -- mem leak in pre-1.2.4 I think
04:49.50Beirdothe linode has 1.0.6
04:49.52blitzragelol
04:50.01*** join/#asterisk bmg505 (n=leon@165.146.27.132)
04:50.03blitzrageI guess just use whatever works right?
04:50.11Beirdopretty much
04:50.16Beirdogot it working, left it alone
04:50.26blitzrageexactly -- don't fix what ain't broke
04:50.38Beirdobut now I'm tweaking it a bit :)
04:50.39Beirdoheh
04:50.44blitzrage:)
04:52.01BeirdoI don't understand why chan_sip.so REQUIRES res_musiconhold.so
04:52.03Beirdoheh
04:52.11blitzrageI never got that either
04:52.26Beirdosomething's requiring adsi too
04:55.20Beirdodown to 14 threads
04:55.40*** join/#asterisk |omni| (i=cathode@c-67-185-96-86.hsd1.wa.comcast.net)
04:57.36blitzragefrom how many?
04:57.40Beirdo20
04:59.10Beirdodown to 12 :)
04:59.27Beirdogetting better and better
05:01.38*** join/#asterisk dextro (n=dextro@cpe-70-116-14-87.austin.res.rr.com)
05:01.48blitzragenice!
05:01.50blitzragealmost half
05:02.15Beirdoyeah, much better, still trying more :)
05:03.47*** join/#asterisk harlequin516 (n=sham@65.39.84.194)
05:04.02harlequin516What is the preferred lang for calling AGI?
05:04.14Qwellharlequin516: Whatever you can write
05:04.37harlequin516Is there a technology advantage for say using php over Java ?
05:04.55Qwellharlequin516: not really
05:05.29harlequin516What's most common..  I want to use that for which I will most likely be able to find help.
05:06.24harlequin516What's the best supported AGI environment?
05:06.54*** join/#asterisk hfb (n=hfb@adsl-69-231-50-224.dsl.irvnca.pacbell.net)
05:08.12JunK-Yharlequin516: its really up to you, but many users choose perl
05:08.14*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
05:08.32harlequin516Eeuuw.
05:08.39blitzrageI like PHP, some use Perl, others use C (although if you're going to write C, you might as well just make an app)
05:08.51blitzrageYou could use python too
05:08.58Qwellruby
05:09.06blitzrageprobably not a bad idea
05:09.17blitzrageI'd say pick something that doesn't need to load a very big interpreter
05:09.22QwellI know somebody who will abuse the hell out of RAGI
05:09.27blitzragesince each AGI spawn will need to load it
05:09.41blitzragewonder what Ruby looks like
05:09.41Qwellfastagi?
05:09.47Qwellblitzrage: crazy small, heh
05:09.49blitzragecould use that too :)
05:10.01Qwellyou can write an irc client in like...4 lines :P
05:10.05blitzragesomeone apparently did a comparison between Ruby and PHP, and he ended up choosing PHP for some reason :)
05:10.08blitzragelol
05:10.16Qwellhe was a newb
05:10.19Qwellruby > php
05:10.20blitzragelol
05:10.26blitzrageblitzrage > Qwell
05:10.36harlequin516Hahah I better read up on ruby
05:10.36Qwellblitzrage >= Qwell
05:10.50blitzrageblitzrage >~= Qwell
05:10.59Qwellumm...ok
05:11.01blitzragelol
05:11.06Beirdo10 threads
05:11.11blitzrageBeirdo: nice moves!
05:11.18blitzrage1 thread
05:11.23Beirdoheh
05:11.26Beirdonot likely to happen
05:11.31blitzrage:)
05:11.34blitzrageSLIDES!!!
05:11.45blitzrage(I'm trying to motivate myself to make a few more :)
05:11.46BeirdoI use SIP and IAX2, and there has got to be a central thread
05:12.12Beirdoooh, app_meetme.so can get bent
05:12.17blitzragelol
05:12.27Beirdoit won't work anyways, no timing source, no kernel modules
05:13.59harlequin516Anyone able to use fwdOUT?
05:14.11harlequin516I tried my damndest and I can't
05:14.35Beirdoblitzrage, any idea what chan_local is for?
05:14.55BeirdoLocal Proxy Channel doesn't really tell me if it's necessary for "normal" use :)
05:14.56walalangblitzrage: very nice book u suggested
05:15.10harlequin516ruby looks awfull....  I prefer the clean easily understaood C-like lang syntaxes.
05:15.37blitzrageBeirdo: yah, its only required if you're using the Local channel --which is basically like a channel that can perform dialplan logic -- if you're not using it, you probably don't need to load it
05:15.45Beirdocoool
05:15.52Beirdoanother one bites the dust
05:15.52blitzragewalalang: thank you :)
05:16.03blitzragewalalang: glad you like it
05:16.33Beirdobah, it stayed at 10
05:16.46blitzragedoh
05:17.27*** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
05:18.05walalangblitzrage: is it ok to pm u?
05:18.27blitzragewalalang: sure, for a minute -- I'm rediculously busy, so I might not reply real quick :)
05:20.30walalangdigium cards are only needed if you want integrate with existing PBX or connect to local PSTN.. right?
05:21.13blitzrageonly if you need to interface with a physical componant of some sort -- else -- just use VoIP
05:23.33walalangfor soft-phone, what is recommended? or basically commonly used?
05:23.52blitzrageI like X-Lite
05:23.56blitzrageworks on both Linux and Windows
05:24.00inv_Arpskype ....
05:24.01blitzrageand OSX I think
05:24.13blitzrageinv_Arp: blasphemy!
05:24.37inv_Arp;]
05:24.50walalangskype? hmmmm
05:25.06blitzrageyou can't use skype with Asterisk
05:25.22walalangthat's what i know also
05:25.59BeirdoMar 23 00:25:44 WARNING[9607]: Unable to open IAX timing interface: No such file
05:26.03Beirdo<PROTECTED>
05:26.05Beirdooh that's pretty
05:26.22Beirdowonder what I removed that I shouldn't
05:29.30blitzragechan_zap.so ?
05:29.51Beirdoit still said that with that loaded
05:30.04blitzragehrmmmm...
05:30.54BeirdoMar 22 23:37:18 WARNING[9024]: Unable to open pseudo channel for timing...  Soun
05:30.58Beirdod may be choppy.
05:31.01Beirdothat too
05:31.06Beirdoit looks like it's always said it
05:31.08Beirdomeh
05:32.18blitzrageno ztdummy loaded?
05:32.23Beirdocan't
05:32.31blitzragebsd?
05:32.40Beirdothere's no kernel modules allowed on linodes
05:32.48Beirdoit's linux under UML
05:34.14*** join/#asterisk joelsolanki (i=joelsola@202.160.161.93)
05:39.32*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
05:39.45*** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au)
05:40.54DaminNo matter what you do, you will always have timing issues when running Asteisk inside of UML or any virtualization platform.
05:41.22*** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com)
05:41.29DaminIt might work for a couple of calls, most of the time.. but it won't be reliable...
05:45.23justinubierdo: uml?
05:53.47*** join/#asterisk websae (i=websae@CPE-24-167-204-30.wi.res.rr.com)
05:54.37*** join/#asterisk Trifix (i=Trifixio@c-69-181-48-164.hsd1.ca.comcast.net)
05:55.02TrifixHas anyone used a channel bank to split an OC3 into multiple T1s for use with an Asterisk box?
05:55.03asterboyWhen I forward my cell phone, I'me getting a different Callid for the same numbers.  Rather interesting to say the least.
05:55.22asterboyAnyone else experience this?
05:55.29Trifixasterboy - be more specific?
05:55.54asterboyI have a number, say, "555-1234".
05:56.09asterboyEvery number has a Call ID associated with it.
05:56.12TrifixLet me rephrase my question to the channel: Does anyone even know an example of a product name/number I would use to split an OC3 into 28 T1s?
05:56.17FuriousGeorgemy callerid used to say "maria" "103" on inbound calls sometimes, but i put a catch for that in my dialplan, and now it says "asterisk" like all the other ones
05:56.33FuriousGeorgeeven though i had the catch change it to "unknown"
05:56.34asterboyI have say "WANG Y" = "555-1234"
05:56.48Trifixasterboy - what are you talking about? in extensions.conf?
05:57.03asterboyAND "DATAMARK SYSTEM" also for "555-1234"
05:57.14asterboyNo, on my call logs.
05:57.33asterboyCalls I forward from my cell phone to an FXO port on my * box.
05:57.52Trifixok you're pissed because the CID in your call logs is wrong?
05:57.55Trifixthe source CID?
05:57.57Trifixor the dest?
05:58.01asterboyThe FXO is coming from a VOIP provider. i.e. Vonage.
05:58.19asterboySource.
05:58.29asterboyNot pissed...fascinated.
05:58.36Trifixyou're still not being clear.
05:58.49Trifixgo through exactly what happens, and what you want to happen.
05:59.03asterboyI have a cell.
05:59.35asterboyI forward the cell phone to my * box.
05:59.41Trifixok stop right there.
05:59.43Trifixhow do you forward it?
05:59.50Trifixusing some setting on your cellphone?
05:59.50asterboyThe * box has an FXO attached to a VOIP provider.
05:59.57asterboyyes
06:00.06Trifixok so you forward it to a VOnage #?
06:00.25asterboyyes, but it's not Vonage...not that it matters.
06:00.36Trifixok ok
06:00.57asterboyEach # had a CallID that looks correct.
06:01.46asterboywhen I look through the actual logs, the CallID becomes different for each instance where the number is the same.
06:01.56Trifixok that last sentence made no sense.
06:02.02asterboylol
06:02.25asterboysay "555-1234" has CALLID "DATAMARK SYSTEM"
06:02.48asterboythere is also another entry, "555-1234".
06:03.00asterboyThis time the callid is "WANG Y"
06:03.10Trifixok so just the name part is wrong.
06:03.14Trifixis it ever right?
06:03.21asterboydon't now.
06:03.26Trifixit's always different?
06:03.36Trifixok actually i can answer your question.
06:03.43asterboyyes
06:03.44Trifixthe way callerid works is that the number and name are matched using this big database.
06:04.00Trifixprobably somewhere along the forward they're using some random phone number for cid, and it happens that that's when it does the name match.
06:04.13Trifixit's interesting, actually, if you get a T1 line and set the outbound CID and call a landline with it,
06:04.17Trifixthe name will get set automatically.
06:04.22Trifixso you can sort of find out who has a given phone number that way.
06:04.37Trifixanyway, there's likely no fix. it's either your cell carrier is using a random CID or more likely your voip carrier.
06:04.49asterboythat is what I thought.
06:04.56asterboydam bizzare to watch in the logs.
06:04.58Trifixsorry :(
06:05.13asterboyI'm thinking, where are these people/companies comeing from?
06:05.26asterboyI kinda like it.
06:05.40asterboyBut then thats just my sick fascination with caos.
06:05.44Trifixyeah - the name part of caller id is sort of worthless.
06:06.06asterboyI'm going to watch for patterns none the less.
06:06.07x86what is the cheapest of the cheap as far as ATA's go?
06:06.20Trifixx86 - sipura 3000 is pretty cheap and good.
06:06.24Trifixspa3000
06:06.27x86i dont care about good
06:06.28asterboyunlocked PAP2 you cheap paki.
06:06.35asterboy:P
06:06.40Trifixyeah buy one of the vonage ones and unlock it.
06:06.40x86i want absolute cheapest possible ;)
06:06.47x86you can do that?
06:06.49Trifixthe locked vonage ones are like $5 on ebay.
06:06.54Trifixi dont know how, but i know it's doable.
06:07.29asterboybut they will call home or already have the latest firmware, which is as far as I know, still unlockable.
06:07.52asterboyyou need to do homework.
06:09.13x86so besides that, whats the cheapest option?
06:09.27Trifixdude that question is kind of annoying.
06:09.37Trifixit's like, you're wasting our time to save you money.
06:09.39asterboyotherwise you get one of these:
06:09.41asterboyhttp://www.dslreports.com/forum/remark,14450684~days=9999~start=969
06:10.52asterboyjust about any ata like a sipura or grandstream
06:11.32x86Trifix: what are you doing right now anyway? :P
06:11.54x86Trifix: obviously you have enough time to sit and bullshit on IRC, so why not offer me some advice? :P
06:11.57Trifixworking of course!
06:12.02Trifixsuck it
06:12.12asterboylol
06:12.24blitzrageadvice is for suckers
06:12.46asterboyIf I was Digium, I'd pay someone to support IRC hardware I was selling.
06:12.48Trifixactually i'm on here because i want someone to tell me the name of a channel bank i can use to split an OC3 into 28 T1s and i dont want to have to pay a consultant for that info.
06:12.49Qwell/nick Qwell[sucker]
06:13.05QwellTrifix: DS3?
06:13.10Trifixsorry
06:13.11TrifixDS3
06:13.12Trifixmy bad
06:13.20Qwellthere is cisco gear that can do it
06:13.20asterboyya, just go to any search engine.
06:13.21blitzragejeezus -- OC3 would be a bunch :)
06:13.24Qwellblitzrage: yeah :p
06:13.30Trifixright but do you know a product name/number?
06:13.34Trifixsearching that stuff is a disaster.
06:13.38asterboyjust 28 T1s?
06:13.39QwellTrifix: it's easy...
06:13.42Trifixhaha
06:13.54Qwellhttp://www.google.com/search?hl=en&lr=&safe=off&q=ds3+demodulation+cisco&btnG=Search
06:14.09Qwellthere you go - cisco as5800
06:14.14Qwell:p
06:14.29Trifixhave you ever done this?
06:14.32Qwellno
06:15.24asterboyAnd I couldn't even sell 1 telephone line today.  Wasted gas and time.
06:15.24Trifixbtw i assume that there is still no DS3 card for asterisk (not that a server could keep up with 672 phone lines anyway)
06:15.24blitzrageTrifix: not yet afaik
06:15.24asterboy672 phone lines?  That's it?
06:15.36Qwell24 * 28
06:15.38Trifixright so i'm looking at maybe 7 servers each with an A104.
06:15.41blitzragethinking you'd have to go to quad xeon to handle that kind of load
06:15.51blitzrageQwell: 23 * 28 ?
06:15.51Qwellquad dualcore opteron!
06:15.53Trifixi'd rather have multiple servers anyway.
06:15.57blitzrageQwell: mmmm... tasty
06:15.58Qwellblitzrage: 24 * 28 = 672
06:16.02asterboyquad xeon? Just 4?
06:16.11blitzrageQwell: what is they were PRI though? :)
06:16.22Qwellthen, you use NFAS :p
06:16.25Trifixyeah they might be PRIs.
06:16.43asterboyI gotta start climbing some ladders.
06:16.46Trifixall i know is that pricingwise once you pass 7 T1s, a DS3 is cheaper.
06:16.47blitzrageI'd think there would be a better signalling protocol for that density though
06:16.56Qwellblitzrage: You'd think
06:17.08blitzrageQwell: I've not dealt with that stuff, so I'm not sure what it uses
06:17.17blitzrageand I can't remember from college :)
06:17.24QwellI think DS3 does PRI too
06:17.28Trifixi dont think the AS5800 is the right product.
06:17.38QwellTrifix: yeah, maybe not.  Call cisco and talk to a sales guy
06:17.44Trifixnooooooo!
06:17.55Qwellwhy not?  They're quite friendly.
06:18.00QwellI was approached by one at VON...heh
06:18.15Qwellblitzrage: Did you catch that?  He was harrassing me about asterisk.
06:18.47Trifixqwell - is "demodulation" the right word for what i want?
06:18.54QwellTrifix: or something
06:19.12Trifixi thought it was "channel bank"
06:19.19Qwellno
06:19.20Trifixbut that's the name of the thingy that goes t1->24 fxo ports
06:19.28*** join/#asterisk BugKham (n=lamer@202.8.86.163)
06:20.11*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:20.14BugKhamhow to forward callerid from one PBX to another using IAX?
06:20.17Qwella DS3 channelbank would be...excessive
06:20.32Trifixyou mean one with 672 fxo ports?
06:20.39BugKhamI tried sendani=yes but it doesnt seem to work
06:20.39Qwellfxo/fxs
06:20.53TrifixBugKham - it happens automatically.
06:21.12Trifixit's like this in extensions.conf:
06:21.28TrifixSet(CALLERID(ANI)=2125551212)
06:21.35BugKhamTrifix: my cdr shows "Guest IAX User" as a callerid
06:21.37TrifixSet(CALLERID(NUMBER)=2125551212)
06:21.46TrifixSet(CALLERID(NAME)=BugKham)
06:22.09astra^^Mar 23 00:21:34 WARNING[3767]: chan_sip.c:1210 retrans_pkt: Maximum retries exceeded on transmission 226904875@
06:22.21astra^^i am getting this mesage frequently
06:22.28astra^^what might be the problem
06:22.37Trifixhere let me try calling you.
06:22.50Trifixhaha
06:23.17TrifixQwell - i'm finding something called a "multiplexer"
06:23.32Qwellsure
06:23.52QwellI was actually thinking demux, not demodulate
06:24.11Qwellhttp://www.google.com/search?hl=en&lr=&safe=off&q=cisco+ds3+demux&btnG=Search
06:25.05blitzrageastra^^: I see those too -- I usually ignore it safely. Its probably the NOTIFY or OPTIONS packets (I can't remember which it uses for qualify) timing out -- if I remember right I sometimes see it at the end of calls
06:25.18blitzragehowever its late, and I'm really just guessing (bullshitting)
06:25.30Trifixqwell- what do you think about this? http://www.firsttechcommunications.com/specials/cac/CarrierAccessABI.html
06:25.50astra^^blitzrage:so it has got nothin to do with the calls droping.. my asr is droping
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06:26.46blitzrageastra^^: I don't think so -- but I haven't read the code -- thats really the only way to really know :)
06:26.53QwellTrifix: I'm obviously not the right person to ask about this
06:26.59Trifixhaha me neither!
06:27.24blitzrageTrifix: send a message to the mailing list -- you'll have a larger amount of sampling data
06:27.43blitzrageTrifix: statistically you'll get a better quality answer
06:27.49tainted_what does "Call rejected by xxx.xxx.xxx.xxx: : No such context/extension" usually mean
06:27.50blitzrageTrifix: now define quality
06:27.58Trifixhaha
06:28.02tainted_something on my local box misconfigured on on the remote box
06:28.13astra^^tainted_: u dont have an extension defined
06:28.13blitzragetainted_: means the context you're trying to drop the call into does not exist
06:28.28tainted_but this is outgoing call
06:28.32Trifixhas anyone here tried actually using the thing that sends Voicemails into an ODBC server? it crashes * for me.
06:28.45blitzragetainted_: then you must be requesting a context in your IAX2 dial string?
06:28.53QwellTrifix: works fine here...mostly
06:28.55tainted_astra^^ i get that error when i call out
06:29.14blitzragealthough it says extension as well -- so the far end could be rejecting your call due to a non-existant extension
06:29.21tainted_yea
06:29.22Trifixseriously? weird.
06:29.36tainted_so his diaplan doesn't know how to handle the extension i sent him right?
06:29.39Trifixi just altered the code to log the uniqueid of the call into the text file and use a cron job to jam them into the db.
06:29.47blitzragetainted_: its one of many possibilities
06:29.59tainted_it couldn't be something on my end
06:30.08tainted_iax2 show registry shows that i'm registered fine
06:30.25blitzragethat doesn't mean you can place a call there -- just means your authentication is correct
06:30.31tainted_right
06:30.38tainted_but my dialstring is so simple
06:30.46tainted_IAX2/HISBOX/${EXTEN}
06:30.47blitzragewhich your message also suggests since it says the call was rejected for a non-existant context/extension
06:31.04blitzragedoesn't mean its anything wrong on your end -- just means he's rejecting the ${EXTEN} you're requesting
06:31.43tainted_i knew it!!
06:31.55tainted_curse you seeds of doubt!
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06:33.12daebodhey
06:33.26Trifixhi daebod
06:33.31Trifixwhat's your question ... for me?
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06:35.27daebodafter a reboot, asterisk mysteriously dies on startup
06:35.29daebodhttp://rafb.net/paste/results/QklwAP82.html
06:35.54Trifixoh no!
06:35.58tainted_do u need chan_modem
06:35.59Trifixdoes it ever start?
06:36.12tainted_noload chan_modem.so d00d
06:36.31Trifixoh yeah. hilarious.
06:36.59kaldemardaebod: nothing mysterious about that. set the noload lines in your /etc/asterisk/modules.conf. you'll probably have to set more than one module.
06:37.19daebodhaha
06:37.20tainted_how do u get that debug
06:37.29daebodi remember it failed before, when i had that as 'noload'
06:37.45tainted_then u probably have to noload a couple more modules
06:37.54daebodi'm confused why it doesn't work now.. i guess i could have upgraded asterisk since the last time it was restarted
06:37.56tainted_most likely u don't use chan_modem
06:38.48kaldemardaebod: chan_modem_aopen.so, chan_modem_i4l.so and chan_modem_bestdata.so are my guess.
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06:40.10Trifixwtf is chan_modem even for anyway?
06:40.12daebodhaha that's weird. i tested it, and my cordless phone was picking up on someone else's conversation
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06:40.46kaldemarTrifix: B2 isdn, i think.
06:40.56Trifixoh. that's common.
06:41.00Trifixhehe
06:41.13tainted_that's chan_modem.so
06:41.25kaldemari've used it with hisax drivers, but it didn't work.
06:41.49tainted_it enables wideband 900MHz frequency interception
06:42.02tainted_at least mine does
06:44.34daebodbah
06:44.44daebodanyone know why i get TWO dialtones?
06:46.17Trifixum
06:46.24Trifixcmon man you're just abusing this channel
06:46.32astra^^i need some help .. anyone can spare 5 min.. please
06:46.38Trifixastra - sure
06:46.58astra^^can i pm u.
06:47.02Trifixyeah
06:47.40Trifixhye does anyone here use a KVM over IP system for managing servers?
06:51.46Trifixjust so everyone knows, astra wants you to log into his server so he can figure out how to hack your server and take you down with him.
06:52.06Trifix(cue astra saying "no that's not what i want i just need help i'm a newbie, etc.")
06:52.47astra^^hahah wat a joke
06:55.50daebodmeh
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07:32.05kmilitzerMorning everyone ...
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07:33.05Vega27hi ho
07:33.13CrashHDhola
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07:38.33CrashHDquiet in here tonight
07:38.35CrashHD*crickets*
07:41.51Vega27yeah man quiet everywhere
07:42.26CrashHDdead wednesday
07:42.46[Airwolf]Thursday here
07:42.49[Airwolf]And morning
07:42.55CrashHDheh
07:43.04CrashHDlocation?
07:44.00*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
07:46.00kmilitzerIt's not so quit here and the sun is already shining brightly ... ;)
07:46.59CrashHDGMT -8 here
07:47.01CrashHDmidnight
07:47.05CrashHDbest time
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07:47.14CrashHDpeaceful
07:47.53kmilitzerCrashHD: Where are you located? It's UTC +1 here ...
07:48.06CrashHDUS, California
07:48.55kmilitzerThat's a nine hour difference ... wow wouldn't have thouhgt that it was so much ... i'm located in germany
07:49.05CrashHDvery nice
07:49.09CrashHDhope to visit there one day
07:50.06kmilitzerI have to say the same for california ;)
07:50.21CrashHDgrass is always greener on the other side of the fence eh?
07:50.34CrashHDyou have the autobaun
07:50.41CrashHDmy kind of road
07:50.52Vega27GIVE ME GERMAN BEER!
07:51.04tsumegive me home made mead!
07:51.16tsumeand it better not be snauptz :P
07:51.30tsumeor whoever you spell the drink which tastes like puke
07:51.38kmilitzerThe Autobahn is fine, if there's not too much traffic ... but it's no fun when you crawl more then drive ;)
07:51.42Vega27meh i understood what you said thats all that matters
07:51.52Vega27LOL
07:51.59CrashHDheh
07:52.19tsumegermans steal the recipes from others and give it a german name ;)
07:53.00kmilitzertsume: Ehh yes? For example what, I would be really interested to know that ;)
07:53.06CrashHDsometimes it's not about the idea but the distribution of such
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08:16.50salviadudcommon linux question, how do i check out how much space i got left?
08:17.05tzafrirdf -h
08:17.13salviadudthank you
08:17.40salviaduddamn, i have downloaded too much pr0n...
08:17.53Frogzoosalviadud: I prefer 'df's format, but that's just me
08:18.01salviadudi like it
08:18.05salviadudi'm on slackware
08:18.15tzafrirdu -sch path/*
08:18.31tzafriror du -sc path/* | sort -n
08:18.48tzafrirto figure out where that pr0n is hiding
08:19.13tzafrirNote that due to caching, only the first run is supposed to take long time
08:19.33salviadudmmmm, i'm not getting that path thing done right
08:19.52salviadudshould i put in my hdd?
08:19.56salviadudlike /hda5?
08:20.25tzafrirthe object is always a filesystem
08:20.34tzafrirdf will show you the mounted filesystems
08:21.01tzafrirno point at starting to get the list of all files in /proc and /sys
08:21.17salviadudso, i could look up /home for example?
08:21.34tzafriror du -sc home/* | sort -n
08:21.40tzafriror du -sch home/*
08:21.44tzafrirtake your pick
08:22.07tzafrirThe former will only return while its done, but gives sorted output
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08:22.55salviadudi got like 2.2 gigs of home
08:23.11salviadudthanx man, i always learn something new with linux
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08:23.34salviadudhave you heard about 2.6.16?
08:23.50salviadudi saw the changelog... my computer isn't that sofisticated
08:23.53salviadudi don't think i'll use it
08:24.00iDunnohmm. I should upgrade to 2.6.16
08:24.07salviadudreally?
08:24.10iDunno$ uname -a
08:24.10iDunnoLinux erwin 2.6.16-rc5 #1 Tue Mar 7 01:29:20 UTC 2006 i686 GNU/Linux
08:24.33salviadudyou crazy man, you need an oracle filesystem module or something?
08:24.35iDunnoyes. 2.6.16 supports the network card in this laptop out of the box... which anything previous didn't ;)
08:24.49salviadudoh, well, that's nice
08:25.12iDunnohardware support is always the best reason for kernel upgrades ;)
08:25.18iDunno(oh, and bug fixes :)
08:25.22salviadudyep i agree
08:25.30salviadudwhat laptop do you got?
08:25.45iDunnoToshiba Portégé R200
08:25.59salviadudnice
08:26.10salviadudi got a dell inspiron 710m
08:26.14salviadudvery small, leen
08:27.16salviadudwhat distro are you running idunno?
08:27.37iDunnodebian unstable - as all sane people do :)
08:27.58Frogzooonce digium releases their new card, how many g729 chans will you get from a 24U rack?
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08:29.05salviadudgotta sleep, peace out
08:31.38austinnichols101~seen opsys
08:31.42jbotopsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 38d 2h 50m 40s ago, saying: 'betaboi" true'.
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08:34.29blitzrageFrogzoo: a lot
08:36.28CrashHDlol
08:36.48CrashHDwhy would g729 be dependent on cards?
08:36.55CrashHD*curious*
08:37.20blitzragewhat do you mean?
08:37.29blitzrageg729 requires lots of CPU power to transcode
08:38.00CrashHDFrogzoo's comment about "once digium releases their new card" followed by the question about number of 729 calsl
08:38.07CrashHDs/calsl/channels
08:38.21CrashHDwhat does one have to do with the other?
08:38.36blitzragethe cards digium is going to release soon will be able to perform transcoding on the card instead of burdoning the CPU to do it. This way, you can get more calls on a single box -- in reference to G729, the licenses for the codec will be included with the hardware
08:38.48CrashHDnice
08:39.02austinnichols101isn't that kind of anti-zapta?
08:39.03CrashHDwill the card be a voip card just for processing?
08:39.08Greek-Boyhmmm, when will digum release these?
08:39.13blitzrageso you put two transcoder cards, and two 4-port E1 cards, you won't really use any CPU at all to do it
08:39.19CrashHDahh
08:39.26CrashHDso it's a processing card only
08:39.28CrashHDpretty sweet
08:39.31austinnichols101kewl
08:39.32blitzrageyes -- pur voip -- thats what transcoding is related to
08:40.01blitzrageyah -- should be in about a month or so -- I was speaking with John at VON, and probably going to be beta testing them soon
08:40.01CrashHDdo you know how many channels each card will be able to handle? and will it also use the cpu as an "overflow"?
08:40.23blitzrageapproximately 150 channels or so -- they aren't sure exactly yet -- thast theoretical
08:40.32CrashHDwoah
08:40.34CrashHDlots of channels
08:40.36blitzrageand yes, you will still be able to use the CPU
08:40.37CrashHD150 ulaw?
08:40.48blitzrage150 ulaw->g729 transcoded
08:40.52CrashHDdamn
08:40.56Greek-Boyand what about zap calls, will that still go through CPU?
08:40.58blitzrage150 ulaw native is already easy to do
08:41.14austinnichols101I'll be able to finally press my Pentium I back into service
08:41.16blitzrageZap to Zap just stays in the card I'm pretty sure
08:41.24CrashHDthanks for the info, sounds exciting
08:41.57blitzragedamn... its like, 3:41am, I'm going ot bed :)
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08:42.22CrashHDcan someone explain to me how multiline phones work with *? and how to set them up?
08:42.30CrashHDand which phones have multiline capabilities
08:43.18Greek-Boyzap to zap stays in the card? even if u use voip phones to access the analog zap lines?
08:43.43CrashHDGreek my assumption would be that it will add to the processing pool
08:43.53CrashHDso any processing that could be offloaded to these cards would be
08:44.21CrashHDI could be wrong though
08:44.33CrashHDmay be a technical hurdle
08:46.04kmilitzer~seen Cresl1n
08:46.07jbotcresl1n <n=matt@gateway.digium.com> was last seen on IRC in channel #debian, 29d 16h 10m 2s ago, saying: ':-)'.
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09:32.46exten123guy where can I read from to know extact each column represent what infor in CDR log?
09:38.10Zeeeki think you'll find it in the sample configs?
09:39.22exten123Zeek, which sample configs?
09:40.33Zeeekof asterisk
09:41.08Zeeekjust a sec
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09:42.06exten123zeeek,okie let me see first. thanks
09:43.21exten123by the way any one broadcast in asterisk to WAN? what thing that I need to configure, like switch and asterisk configuration. any place for this kind of info?
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09:45.39Zeeek<PROTECTED>
09:46.28Zeeeklook at cdr_csv.c
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09:48.58Zeeekthis is the file to look for  /usr/src/asterisk/cdr/cdr_csv.c
09:49.46shiznatixhello, I have iaxmodem and hylafax installed on asterisk but I do not have a fax machine. I need to send a fax and have it recieved without a fax machine on either end. is this possible?
09:50.16cfhSomeone have succesfully config ISDN card Primux 4s on NT mode?
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09:58.09Angeljarodhi guys
09:58.18Angeljarodis there someone here ?
09:58.32kmilitzerI have a completely offtopic question: can someone invite me to gmail? ;)
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10:04.30Angeljaroddoes somebody know how to configure a snom phone programmable key to intercept calls in a queue ?
10:05.21exten123by the way what is billable seconds in CDR? what the diff with duration?
10:05.40Zeeekmake a tes call and see
10:06.01Zeeekkmilitzer what's your address I'll do it now
10:06.07areskiexten123, exten123 billsec use to count duration only after the answer
10:06.25areskithe real billable seconds
10:06.51Zeeekkmilitzer hurry up and give your email and I'll invite you
10:07.54Zeeekanyone else want a gmail invite while I have that WIndow open? Speak now or forever hold your peace
10:08.19kmilitzerZeeek: cool km@westend.com
10:08.41Zeeekdone
10:08.46kmilitzerZeeek: Thanks a lot
10:08.47Zeeekanyone else want a gmail invite while I have that WIndow open?
10:09.22walalangme...
10:09.55Zeeekwell?
10:09.59walalangjdelacrus@gmail.com ...........
10:10.08Zeeekinvite yourself!
10:10.11walalangjust joking LOL
10:10.14walalang:D
10:10.16BugKhamdo we need to specify "context=" in the outgoing iax2 config?
10:10.33Zeeeknot if it's only for outgoing
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10:12.24BugKhambut the other box generates this error Rejected connect attempt from 192.168.1.23, who was trying to reach '3001@'
10:12.49Zeeekthe other box?
10:13.16BugKhamyeah, i'm calling from one * to another
10:13.31Zeeekthe other box needs a context
10:13.55CrashHDiax2 works funny
10:14.04CrashHDthe auth part of things is weird
10:14.06Zeeekthat would come from this: Dial(IAX2/blahblhablha.../ThisContext)
10:14.41BugKhamZeeek: yes, I did that
10:15.29Zeeekand ThisContext exists on theother box
10:16.00BugKhamumm I put the "ThisExtension"
10:16.30BugKhamand put the context in the iax.conf of the other box
10:16.33Zeeekextension and context are two different things
10:16.41BugKhamlet me try
10:16.52Zeeekgive the dial command here just that one line
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10:17.43BugKhamDial(IAX2/${myotherbox}/${EXTEN},30,r)
10:18.04vgstercan asterisk do directed pickups yet?
10:20.14RoyKdirected?
10:20.47vgsteras in ext 123 is ringing which isnt part of my call group but i can dial *8123# to pick it up
10:22.37Zeeekwon't pickup() do that?
10:22.42vgsterdont know
10:22.50Zeeekwell look it up and see
10:22.58vgsterthats what im asking, but its a point i will follow up on
10:23.11Zeeekshow application Pickup
10:23.40vgsteri know of pickup but didnt think it could be used like that but u will investigate
10:23.43BugKhamZeeek: should thi work => Dial(IAX2/${myotherbox}/${EXTEN}@from-first-box,30,r)
10:24.14Zeeeki'm looking at my config on server2 now
10:24.44*** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) [NETSPLIT VICTIM]
10:24.44*** join/#asterisk kll (i=kll@insomnia.juniks.net) [NETSPLIT VICTIM]
10:24.48BugKhamI had this response " chan_iax2.c:6985 socket_read: Call rejected by 192.168.1.22: No authority found"
10:25.17BugKhamI will work if I enable the guest user in iax.conf
10:25.35Zeeekthe call isn't arriving in the right context
10:25.43BugKhamand the calls will go to the context of the guest user
10:25.44*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
10:26.08vgsteryes pickup will do it thanks guys
10:26.32ZeeekDial(IAX2/username:password@domain.tld/EXTEN)
10:27.10shiznatixdoes anyone know of a free software fax machine for linux?
10:27.32Zeeekthe on server2 [username] type=user context=incomingserver1 etc etc
10:28.47ZeeekBugKham so long story short, you need to create a user on box2 and call it from box1
10:29.14BugKhamZeeek: it's working this way
10:29.51Zeeekit will work if you want and have a guest user, yes
10:30.55BugKhamZeeek: I meant I following your advice by using Dial(IAX2/username:password@domain.tld/EXTEN)
10:31.00BugKhamand it worked
10:31.13Zeeekgood!
10:31.22CrashHDis there a way to use the ip as the auth portion?
10:31.24CrashHDlike sip does?
10:31.29CrashHDI ran into this the other night
10:31.31CrashHDdrove me nuts
10:31.39CrashHDmy provider isn't authing
10:31.50CrashHDso it shows as an unauthenticated incoming call
10:32.03CrashHDis there a way to auth it solely on the incoming host ip?
10:32.03*** join/#asterisk opus_ (n=opus@dahphish.org)
10:32.09BugKhamZeeek: so there is something wrong with by outgoing config
10:32.20opus_hey -- how do you disable native briding in chan Local -- /r at the end?
10:32.26ZeeekBugKham I thought you said it worked?
10:32.46BugKhamZeeek:  yes, by using Dial(IAX2/username:password@domain.tld/EXTEN)
10:32.53walalangwhat's the recommended FXO card? model?
10:33.22opus_sangoma a200
10:33.23BugKhamZeeek:  but I'm wondering why can't I use Dial(IAX2/myotherbox/EXTEN)
10:33.51BugKhamwhere I create [myotherbox] in the iax.conf
10:33.58ZeeekBugKham because there's no context given
10:34.14*** join/#asterisk backblue (n=igor@82.102.1.42)
10:34.25walalangopus: TY
10:34.28Zeeeklook at the asterisk sample files or read asteriskdocs.org
10:34.55BugKhamZeeek: okay
10:43.38*** join/#asterisk Whisk (n=a@194.130.117.202)
10:44.16cfhwhere can I find doc/howto for Primux card to work with asterisk?
10:44.48BugKhamZeeek: I added "username=" on my first box's iax.conf which is the [username] on my other box and it's working now
10:45.02Zeeekgreat
10:46.10BugKhamZeeek: thanks for your help
10:46.38*** join/#asterisk eset (n=eset@ip545186e3.direct-adsl.nl)
10:46.54esetanyone know a good sip client on osx with stun support?
10:54.41opus_eset sjphone
10:55.09ZeeekBugKham np, glad it worked out
10:58.03*** join/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl)
11:00.30*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
11:00.49*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
11:24.59*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
11:29.29esetopus_ : thanks
11:29.55esetopus_ : cant seem to get it to register with asterisk...hmmm
11:30.48Zeeekwhat does the console say?
11:32.42esetthere is no report from the console, sjphone isnt trying to reigister
11:33.01eseti must be missing something simple in the sjphone config
11:33.07Zeeekuse a sniffer to see whats what
11:33.29Zeeekor sip debug
11:33.52esetokee, ta
11:40.50Zeeekeset any luck on the sip debug?
11:41.37*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)
11:42.36fourcheezeok, I'm about to run the gauntlet with faxes
11:42.42fourcheezeanyone want to tell me not to bother
11:42.50fourcheezeor that there's a better way
11:43.24Zeeekwe're all ears!
11:43.28esethow about email ;)
11:43.37fourcheezeyeah
11:43.40fourcheezepersonally I hate faxes
11:43.48Zeeekyeah fax has no place in the 21st century
11:43.48fourcheezecustomer has requested fax support
11:44.02Sebbone question: if i have two e1-lines from different providers, and a te411p card, what do i set as timing source in zaptel.conf?
11:44.06Zeeekincoming or outgoing?
11:44.13fourcheezeboth AFAIK
11:44.30Zeeekfor outgoing you're better off with a simple fax machine
11:44.35fourcheezenow my incoming numbers are supplied via g729
11:44.37Sebbboth incoming.. i didn't try it yet, but i don't know if it makes problems
11:44.53fourcheezeI'm told that t.38 support makes this all possible
11:44.58Zeeekfor inciming, I have never been able to get spandsp and rxfax to work for 100% of incoming faxes
11:45.05fourcheezeI also see that * doesn't support t.38
11:45.23fourcheezewhat's spandsp ?
11:45.24Zeeeksee the bug tracker I think there are patches
11:45.57Zeeekspandsp is the lib that allows the apps rxfax txfax to work AFAIK
11:46.02fourcheezeok
11:46.05fourcheezehowever
11:46.13fourcheezeI don't want asterisk to do anything to the faxes
11:46.23fourcheezeat least I don't think so
11:46.31ZeeekI love the fact that it doesn't print them, it just emails them to me
11:46.41fourcheezeyeah, now that wold be nice
11:46.48fourcheezehowever it's not what they want
11:46.57RoyKfourcheeze: just use rxfax and do a little scripting...
11:46.57fourcheezeAFAIK the proposal is for a sipura with fax attached
11:47.07RoyKthen you need t.38
11:47.13Zeeekhowever, one machine of four doesn't work with the verion I use whereas the same machines send flawlessy to a windows sharware program on a modem
11:47.21fourcheezedo I need t.38 in asterisk or just in the sipura?
11:47.28RoyKboth, obviously
11:47.36fourcheezewhy is that obvious?
11:47.38RoyKdunno if sipuras have t.38, though
11:47.52RoyKfourcheeze: just as we both need to know engslish to understand oneanother
11:47.57fourcheezeok
11:48.12fourcheezeisn't there a t.38 passthrough option?
11:48.15RoyKjeg kan godt snakke norsk til deg - det blir vel omtrent som t.38
11:48.33RoyKthere's a patch for asterisk that allows t.38
11:48.38RoyKpassthrough iirc
11:48.41fourcheezeok, but we're both using irc and neither of us speaks tcp/ip
11:49.05RoyKjeg kan fortsette å snakke norsk hvis du vil :)
11:49.09dpryo:P
11:49.14fourcheezethis is already sounding cruddy
11:49.30Zeeekpurchase two cheap fax machines and get over it
11:49.45fourcheezethey are going to use fax machines
11:49.48Zeeek|| nice
11:49.49fourcheezeit's phone lines they don't have
11:50.03Zeeekin that case email is recommended :)
11:50.13*** join/#asterisk Bambr (n=Bambr@213-35-232-186-dsl.end.estpak.ee)
11:50.19RoyKfourcheeze: you need t.38
11:50.25RoyKfourcheeze: and you need t.38
11:50.31fourcheezedamn
11:50.58RoyKit is possible to do faxing without it, with good luck and a good network, REALLY low latencyt
11:52.02RoyKthat way you should be able to build a t.37-like system with an asterisk box receiving the fax with app_rxfax, transmitting the file using some protocol, and then have another asterisk server txfax it to the destionation
11:52.58RoyKbut doing fax over ip without t.38 is generally not easy. some personal gods may help
11:53.09shiznatixRoyK, can you help me with sending faxes?
11:53.26RoyKread above
11:54.02RoyK~foip
11:54.12RoyK~ping
11:54.13jbotpong
11:54.14fourcheezehmm
11:54.44fourcheezeRoyK: does the need for t.38 regardless of codec?
11:54.52fourcheezecan't it just be treated like voice?
11:55.00fourcheezeif I was using ulaw/alaw ?
11:55.45Zeeeki think it works with ulaw
11:55.48RoyKjbot_: FoIP is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
11:56.25RoyK~foip
11:56.37RoyK~lart himself
11:56.48fourcheeze~FoIP
11:56.54fourcheezehmm
11:56.59RoyK~FoIP is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
11:57.00jbotokay, RoyK
11:57.01fourcheezejbot borked
11:57.02jboti heard borked is broken in other words
11:57.12RoyK~foip
11:57.13jbotfrom memory, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
11:57.27Zeeek~Foip
11:57.28jbotfoip is probably Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
11:57.33fourcheeze~FoIP
11:57.34jboti guess foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
11:57.39RoyK~lart fourcheeze
11:57.50fourcheezeta
11:58.10kaldemarare you guys sure we all know what foip is by now? :)
11:58.13fourcheezehow about outgoing fax?
11:58.29Zeeek~voPI is Voice Over Public Internet
11:58.30jbotokay, Zeeek
11:59.04Zeeek~tfutf is Too Fucked Up To Fix
11:59.05jbotZeeek: please, watch your language.
11:59.18Zeeek~seen my language
11:59.28jboti haven't seen 'my language', Zeeek
11:59.40Zeeek~seen Your Language
11:59.42jboti haven't seen 'your language', Zeeek
11:59.52Zeeekso why are you criticizing it?
12:00.01RoyK~fubar
12:00.03jbotrumour has it, fubar is f*cked up beyond any recognition
12:00.16Zeeek~tfutf is Too Fscked Up To Fix
12:00.17jbotZeeek: okay
12:00.51RoyKmethinks rewriting four-letter-words to make them nicer is quite stupid, quite american....
12:00.57Zeeekñyuk is the trademarked laugh of Curly of the Three Stooges. Always occurs in pairs as in "nyuk, nyuk"
12:01.10RoyK:)
12:01.20Zeeek~nyuk is the trademarked laugh of Curly of the Three Stooges. Always occurs in pairs as in "nyuk, nyuk"
12:01.22jbotACTION grabs is the trademarked laugh of Curly of the Three Stooges. Always occurs in pairs as in "nyuk, nyuk" nose and hammers it with his other hand!
12:01.36Zeeeknyuk nyuk
12:02.03*** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net)
12:02.09*** join/#asterisk jeffik (n=Jeff@CPE0050babf4cd6-CM014350000760.cpe.net.cable.rogers.com)
12:02.10shiznatixI need to test faxes but I don't have a fax machine to test with, is there a software fax machine that I can use?
12:02.18Zeeekj2.com
12:02.30Zeeekefax.com
12:02.44Zeeek~google
12:02.46jbotwell, google is a search engine found at http://www.google.com/
12:02.56RoyK~google Zeeek
12:03.22[ProB]CrazyManshiznatix: you still stick to the same problem ... but your Asterisk has an PSTN connection ?
12:04.15shiznatix[ProB]CrazyMan, yes same stupid problem but we do not have a PSTN line yet, we will get it soon though so if you have info on that it would be helpful
12:05.25vgsterdo i need to do anything different between *1.0.10 and *1.2.5 on my GXP-2000s to get MWI working?
12:05.26[ProB]CrazyManshiznatix: to test faxing ... you need an analog or digital pri card ... anywhere must the fax come in
12:06.09Zeeekvgster there is an issue of adding the context in 1.2.5 I think
12:06.10shiznatixi dont have any of that stuff so basically... i can't do any testing of any kind with faxes?
12:06.22Zeeekvgster as in 2002@office
12:06.37[ProB]CrazyManvgster: yes there is an different ..
12:06.45vgsterah
12:07.02*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
12:07.06[ProB]CrazyManvgster: the extension jumping changed ..
12:07.12vgsterso i need to specify the mailbox against the account
12:07.25ScaredyCatWalalalalalalalalalalalalalalalaalalalalalalalala-plop
12:07.38ZeeekHoly Cats!
12:07.42vgsterthanks
12:07.43vgsterfixed
12:07.59esethhmm, anyone know why asterisk would say something like "couldnt make SIP/xx compatabile with IAX/xx.xx.xx" ?
12:08.05[ProB]CrazyManpriorityjumping=yes
12:08.18esetusing the xten client
12:08.18shiznatix[ProB]CrazyMan, what about SIP / IAX faxing?
12:10.20[ProB]CrazyManshiznatix: yes (i never tried this) when I tried to send fax from asterisk to the same asterisk box, it doesn't work because spandsp could not send faxes to itself. then you need two boxes, how it is with IAX/Sip i do not kno
12:11.10shiznatix[ProB]CrazyMan, I need 2 boxes with asterisk installed or just use 1 asterisk box as a gateway?
12:11.34[ProB]CrazyMan2 boxes
12:11.57shiznatixboth with asterisi
12:12.03shiznatixasterisk?*
12:12.05[ProB]CrazyManjup
12:12.34fourcheezeI saw a grandstream video phone yesterday
12:12.38shiznatixhummm
12:12.40fourcheezeactually I saw 2
12:12.44fourcheezeone had crashed
12:12.51fourcheezeand the other simply wasn't working
12:13.04shiznatix[ProB]CrazyMan, how would I go about sending a fax from the 1st box to the second?
12:13.11shiznatix<PROTECTED>
12:13.22fourcheezegood to see grandstream keeping up their quality
12:13.30[ProB]CrazyManshiznatix: but anyway .. you could by an cheap fritzcard .. and install it into an normal pc to fax ...
12:13.47ScaredyCatgrandstream? quality? eh?
12:13.51*** join/#asterisk zotz (n=zotz@24.231.32.85)
12:14.12fourcheezequality can be used in the positive and negative
12:14.14[ProB]CrazyManshiznatix:via an pri card ?
12:14.26iDunnoalternative reality? what? where?
12:14.28ZeeekScaredyCat funny only one of my three GS BT101 work now, 18 months later
12:14.41kaldemarfourcheeze: take a look at HT 488 and then we'll discuss quality. ;)
12:14.50shiznatix[ProB]CrazyMan, none of this is really a option for me because this is just a test stuff to sell to other companies who want to configure asterisk
12:15.02ScaredyCatwell you should have got ciscos then Zeeek :)
12:15.07fourcheezeis it possible to tell which clients are watching which hints?
12:15.18ZeeekI have cisco now: SPA940
12:15.21fourcheezeshow hints gives me a number
12:15.31shiznatix[ProB]CrazyMan, so we are looking for the easiest, out of the box way to send faxes
12:15.32fourcheezebut doesn't say which
12:15.34ScaredyCat940?!
12:15.41[ProB]CrazyManshiznatix: so you have to do faxing via SIP / IAX (FoIP)
12:15.43ZeeekI thought show hints was like "it's round and has three loegs"
12:15.46fourcheezethat's only pretending to be a cisco
12:15.59Zeeekit is a cisco because it says it is
12:16.13kaldemarfourcheeze: show subscriptions ?
12:16.19[ProB]CrazyManshiznatix: but then you need an provider which sends the faxes threw the pstn
12:16.25Zeeekbut seriously it's a great entry level phone with an excellent speaker
12:16.39fourcheezekaldemar: ooh thanks for that
12:16.45Zeeek<PROTECTED>
12:16.50shiznatix[ProB]CrazyMan, ok, but then how would i do all of that?
12:17.08kaldemarfourcheeze: having said that, i noticed that there might be no such command anymore.
12:17.24ScaredyCatSony SPP-A940 Cordless Phone Batteries and Accessories
12:17.28ScaredyCatpah
12:17.32fourcheezekaldemar: prefix with sip
12:17.47Zeeeknyuk nyuk
12:17.49fourcheezeZeeek: did you mean a 941 ?
12:18.01Zeeekso ScaredyCat still in the Dutch realm?
12:18.10ScaredyCatYeah Zeeek...
12:18.16ScaredyCatstill can't escape
12:18.27Greek-Boyis there any way to interface a GSM cellphone to Asterisk? perhaps via bluetooth?
12:18.37ZeeekI think I did - the model isn't written on the phone just "Cisco - Linksys - VoIP¨voice IP Phone"
12:18.49ScaredyCatYou still with Les Frenchies?
12:18.56Zeeek25 years now
12:19.00[ProB]CrazyManshiznatix: depends on how you want to solve things ... there are providers which send faxes via emailgateway .. so you dont need asterisk
12:19.12ScaredyCatMan - you are sick ;)
12:19.14*** join/#asterisk fugitivo (n=user@201.255.177.90)
12:19.16[ProB]CrazyManor you do it via FoIP
12:19.21ZeeekI'm a citizen now
12:19.26ScaredyCatYoink!
12:19.30ScaredyCatthe let you?
12:19.33ScaredyCatthey
12:19.36ZeeekI let them
12:19.39ScaredyCathehehe
12:19.57shiznatix[ProB]CrazyMan, ok so thats the way with straight e-mail but what about if
12:20.02Zeeekhey you updated your site but I still didn't paypal you and money
12:20.04shiznatix[ProB]CrazyMan, ok so thats the way with straight e-mail but what about with asterisk*
12:20.12Zeeeknot enough naked shots
12:20.21ScaredyCat29 days - tick tock
12:20.32Zeeekuntil ?
12:20.37ScaredyCatsince
12:20.42Zeeekoh
12:20.48[ProB]CrazyManshiznatix: you need an voip provider over which you could send FoIP
12:20.55Zeeekall you did was change the font and thank someone else
12:21.02fugitivoThis is cool, xchat in a nokia 770
12:21.06[ProB]CrazyManshiznatix: http://soft-switch.org/foip.html
12:21.12fourcheezefugitivo: aha, how do you rate it?
12:21.16shiznatix[ProB]CrazyMan, ok lemme give that a look
12:21.19fourcheezefugitivo: is there a sip client for it?
12:21.23Zeeekyour site
12:21.29ScaredyCatwhat about it...
12:21.30Zeeekautomation in Italy
12:21.46ScaredyCatnot in italy...
12:22.07Zeeekhey you should get in touch with our buddy François - they sell all that ham radio/home control stuff
12:22.08ScaredyCatit;s like... automated it ... as in - hey look, I automated it
12:22.22ZeeekTLD are TLD
12:22.42fugitivoFourcheeze: it's awesome, i think there's an app called minisip that should work
12:22.42ScaredyCatbut tld != country
12:22.46Zeeekdid you see the funny Nigerian post on the ML?
12:22.55Zeeekspeaking of countries
12:23.04ScaredyCatno, not read it for a while now...
12:23.08fourcheezefugitivo: is that the same minisip you can run on linux boxen?
12:23.16fugitivoYes
12:23.19ScaredyCatI have 25485 unread emails
12:23.19fourcheezehmm
12:23.27ZeeekI'd paste it here but then all the newbies will yell FLOOD! a word they learned yesterday
12:23.32ScaredyCatpm me
12:23.34fourcheezefugitivo:  don't depend on it being great then :-)
12:23.41fugitivothe nokia 770 comes with linux
12:23.46fourcheezefugitivo: sure
12:23.50fourcheezehow good is it as a phone?
12:23.50ScaredyCator use paste bun
12:23.52ScaredyCatbin
12:23.59RoyKhm. anyone who knows how to change an opereator-locked sjphone? i need to make it connect to another server :{
12:24.01Zeeekyeah looking now
12:24.24fugitivoWell, i think in the future there'll be more ports
12:25.23ZeeekScaredyCat
12:25.25Zeeek<PROTECTED>
12:25.36Zeeek#
12:25.36Zeeek[Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
12:25.51fourcheezefugitivo: oh it doesn't seem to have 3g
12:26.00Zeeek#RoyK look at the executable in a hew editor
12:26.17ZeeekPaste Bun indeed
12:26.22ZeeekI won't go there
12:26.56ScaredyCatZeeek: roflmao! WTF!
12:26.59Zeeek<PROTECTED>
12:27.00Zeeekthis really is a good read:  http://pastebin.ca/46697
12:27.06Zeeekoops
12:27.24ZeeekA few people actually started pissing on the guy
12:27.55Zeeek#
12:27.55ZeeekFirst and foremost,I apologized
12:27.55Zeeek#
12:27.55Zeeekusing this medium to reach you for a transaction/business of this magnitude,
12:27.55Zeeek#
12:27.56Zeeekbut this is due to Confidentiality and prompt access reposed on this
12:27.58Zeeek#
12:28.00Zeeekmedium. Be informed that a member of the #asterisk channel on Freenode who
12:28.02Zeeek#
12:28.04Zeeekis well familiar with you gave your enviable credentials/particulars to
12:28.06Zeeek#
12:28.08Zeeekme.
12:28.09ScaredyCatsounds pretty typical of the * list - to piss on someone asking for help...
12:28.10ZeeekSHIT
12:28.12ZeeekFLOOD
12:28.12ScaredyCaterrmm
12:28.17ScaredyCatyou just pasted it?
12:28.19ScaredyCatlol
12:28.36Zeeekno it was just one line but became many in the beautiful WIndows world
12:29.03Zeeekstill using SMS?
12:29.20ScaredyCatnot much, they put the price up from free to 25c er sms
12:29.28ScaredyCatper
12:29.47ScaredyCatit still works though.. :D
12:29.49RoyK~pb
12:29.50jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
12:29.55Zeeekdamn. I think it's 10€ cents here
12:30.06*** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it)
12:30.20ZeeekWhat is the command to run when you already flooded the channel?
12:30.29Zeeek"First and foremost,I apologized using this medium to reach you for a transaction/business of this magnitude, but this is due to Confidentiality and prompt access reposed on this medium. Be informed that a member of the #asterisk channel on Freenode who is well familiar with you gave your enviable credentials/particulars to me."
12:30.30ScaredyCat/part
12:31.18ZeeekI used the SMS thing a lot on vacation to get asterisk to call me back
12:31.29ScaredyCatI keep kicking my * box and the fxo stops working... no wonder I didn;t get any calls today!
12:31.39iDunnoheh
12:31.44iDunnodon't kick it ;)
12:31.45Zeeekin SPain for example, they don't give the phone numbers of public phones
12:31.55*** join/#asterisk Skarmeth (n=Skarmeth@200.165.81.130)
12:32.07ScaredyCatso you can't use callback?
12:32.19ZeeekI don't think the CID goes out
12:32.26ScaredyCatiDunno: yeah.. I guess I could move it from under the desk
12:32.33Zeeekso point is, at any known number I can SMS asterisk to call me back
12:32.56Zeeekand that works great since I can specify routing in the same SMS
12:33.04ScaredyCatyou could just take an ATA or softphone with you :D
12:33.15ZeeekI did that at Astricon
12:33.19Zeeekboth
12:33.32ZeeekIAX hardphone in the room and softphone on the laptop
12:33.55Zeeekno wonder there were so few hookers at a hotel where 1000 geeks were talking to laptops thru headsets!
12:34.38iDunno@)
12:34.45ScaredyCat... do't talk with your mouth full!
12:35.03esetyes they were all off the hook(er)
12:35.08*** join/#asterisk __chris (n=chris@unaffiliated/redlined)
12:35.19*** join/#asterisk |cleric| (n=dacleric@p5482944D.dip0.t-ipconnect.de)
12:35.23ScaredyCatllama llama duck
12:35.28Zeeekewwwww
12:35.34Zeeekñyuk
12:35.35esetthank u thank u
12:35.41Zeeek~nyuk
12:35.53*** join/#asterisk shiznatix (n=shiznati@213-35-232-186-dsl.end.estpak.ee)
12:36.26ScaredyCathttp://tinyurl.com/o7dl3
12:36.39*** join/#asterisk flot (n=flot@user244.hovrino.net)
12:36.48Zeeekwhat's the helium for?
12:37.06iDunnouplifting?
12:37.34Zeeek"pre-owned, mission damaged" sounds like my ex-wife!
12:37.48Zeeek[drum roll?]
12:38.38kmilitzerI am trying to get a test-this-branch to register like <sip:02418903117@212.117.82.220> at a ser, but what the asterisk does is to register with something like: <sip:ASTZVXW-2a90f99f4b11000044674e276b707822@212.117.82.220> is there a way to change this?
12:41.11ScaredyCatsucky-sip implementation
12:42.11ScaredyCatwell, I seem to have solved my spam problem
12:42.25Zeeekgood
12:43.14ScaredyCatyeah.. I just deleted all my email accounts :P
12:43.21Zeeekthat would help
12:43.42Zeeekwhat version of asterisk are you currently running in production?
12:43.58ScaredyCatspamassasin switched to low tollerance and coupled with spamhilator seems to do the trick
12:44.02ScaredyCatme?
12:44.04ScaredyCat1.0.7
12:44.06ScaredyCat:D
12:44.08Zeeekmuhahah
12:44.19Zeeekreturn to the future, eh?
12:44.24ScaredyCatand in fact on one box - pre 1
12:44.34Zeeekrun some DNSBL you'll like them
12:44.43ScaredyCatI just use what's stable...
12:44.57Zeeek<PROTECTED>
12:45.01ScaredyCatand what hasn't been fucktarded up...
12:45.19ScaredyCatDNSBL are just stupid though...
12:45.32Zeeekhey I had a funny but nice surprise a couple of days ago
12:45.45ScaredyCatit's easy to get on one, and almost impossible to get off one
12:45.50Zeeekno DNSBL when configured correctly work very well
12:46.07ScaredyCatit's onot the config that's the issue...
12:46.13Zeeeksorry you're wrong about that. Only spews was like that and a few other really marginal geek ones
12:46.14ScaredyCatit's the blacklists
12:46.24Zeeekpart of config is chooosing the lists
12:46.46Zeeekspamcop is always temporary listings and you can whitelist anyone you like
12:46.48ScaredyCatpah humbug
12:47.00ZeeekI send dozens of spams a day at spamcop
12:47.17Zeeekany problems are solved by whitelising
12:47.57Zeeekthe funny but nice surprise was that one of my SIP providers is at 3ms from my new server
12:48.09Zeeekvery odd indeed that
12:48.42ScaredyCatwell that's not so bad...
12:49.04Zeeekconsidering they're at 150ms from my office server, no 3ms isn't bad at all
12:49.36ScaredyCatwhat's ur office hosted on?
12:49.42Zeeekthe lowest lag provider Ihave is voiptalk at around 20ms
12:49.55*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
12:50.19Zeeekthe office is on a consumer-leve 3meg DSL. In a few years everyone will have access to 1GIG internet
12:50.23Eitchhau
12:50.42Zeeekhow
12:51.09ScaredyCatZeeek: move to Sweden... they have really goot net connectivity...
12:51.30ZeeekI do,'t want to have to get se citizenship tho
12:51.33ScaredyCat100mbit (SDSL) for 35 EURO
12:53.01Zeeek1000mbit for €70
12:53.08Zeeeknyan nyan na na nay
12:53.12ScaredyCatSDSL?
12:53.20ZeeekFFDSL
12:53.23austinnichols101bi-dir
12:53.26ScaredyCator 1000mbit down and 2 bit up
12:53.29Zeeeksip show peers
12:53.34Zeeeknot.
12:54.25Zeeekanyone have a number I can call in the US that has a long recording to listen to for testing ?
12:54.42Zeeek(besides American Airlines reservations)
12:54.49ScaredyCat5551212 ?
12:54.53Zeeekno good
12:54.55ScaredyCat212
12:55.19Zeeeksomeone here in da community
12:55.44ScaredyCatin da hood
12:55.45Zeeekhey they're slipping from 3 to 6ms!
12:57.15Zeeekone odd thing here is that I keep seeing "Found no files in /usr/local/share/asterisk"
12:57.30Zeeekyes I find no reference to that anywhere in any conf
12:58.36ScaredyCatwhere do you see it?
12:58.42ScaredyCatin the * cli?
12:58.50Zeeekon the console from res_musiconhold
12:59.04austinnichols101zeeek: how long of a recording do you need?
12:59.14dyerfRoyK: i mean, you can download sj-phone from sjlabs.com and registered it (registration is free for up to 5 licence)
12:59.27Zeeekbut the directory in that conf is  ~/mohmp3
12:59.36Zeeekaustinnichols101 a minute or more is fine
12:59.49*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
13:00.36austinnichols101call my main office number
13:00.40Zeeekthis is in FreeBSD: asterisk seems to try to spawn mp3 every x minutes
13:01.33*** join/#asterisk atta (n=atta@p54B6F809.dip.t-dialin.net)
13:01.51austinnichols101see separate window
13:01.54atta#moin
13:02.49attai have one quastion about ldap and asterisk cann me help one ???
13:04.30*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:08.07*** join/#asterisk fugitivo (n=ajf@201.255.177.90)
13:17.45*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)
13:17.59*** join/#asterisk viLeR (i=1000@66.128.47.232)
13:19.15*** join/#asterisk bartpbx (n=bartpbx@217.24.210.210)
13:19.16*** join/#asterisk oej (n=oej@apollo.webway.se)
13:19.43bartpbxhello, can anyone explain the output of iax2 show stats?
13:24.03*** join/#asterisk Aurs (i=aurs@hallo.aurs.info)
13:30.06cfhis possible run asterisk with primux card
13:30.07cfh?
13:30.40jsharpDoubtful.  I don't think there's drivers for.
13:37.23*** join/#asterisk saftsack (n=saftsack@p54A7D706.dip.t-dialin.net)
13:37.50*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
13:40.04*** join/#asterisk atta (n=atta@p54B6F809.dip.t-dialin.net)
13:41.58*** join/#asterisk gr0mit (n=w10277@206.41.25.138)
13:42.48gr0mithi
13:43.07gr0mitam trying to set up a T1 with robbed bit signalling
13:43.57Greek-Boyhow does VoIP sound quality compare to analog? i'm talking for local calls on PABX
13:43.58gr0miti can make calls ok but periodically i seem to get a 'ghost' incoming channel seizure, but I do not see any state transitions on the ABCD bits.
13:44.43gr0mitGreek-Boy - if you use A or mu law you will honestly not be able to tell the difference.
13:44.43jsharpGreek-Boy:  If you're running ulaw codecs on a local lan, there's virtually no difference.
13:45.55jsharpSounds like your telco's switch has a case of tourettes.
13:47.11gr0mitwell this is the first time I have ever had to use such a strange system!
13:47.23*** join/#asterisk astra^^ (n=muhajir_@59.145.104.74)
13:47.27*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
13:48.00gr0mitbut if it were a real call I would expect to see RxABCD go high to seize the trunk, right?  and I don't
13:48.02*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
13:49.03jsharpI think so.   I'm a bit rusty on RBS, though.
13:49.55gr0mithaving a hard time understanding how the US put men on the moon if they use RBS on telecoms!
13:51.40jsharpCause back then, the only people who talked RBS to the telecoms were the guys who ate & slept telephony.  They could tell you what bits were set and what framing was being used by sticking the T1 cable into their mouth.
13:51.56*** join/#asterisk eliel (n=eliel@200.123.183.89)
13:51.58jsharpAnd they walked to school in the snow and it was uphill both ways.
13:52.08gr0mitvery true.
13:52.40gr0mitIt Just Works (TM)
13:52.58fourcheezehmm
13:53.14fourcheezelooks like I'm going to be sullied with that soon
13:53.23fourcheezewhat's BRI like with asterisk ?
13:53.50gr0mitdepends where you are and what your previous experience is, fourcheeze
13:53.52jsharpI wouldn't say "It just works", but making RBS work right reliably is like painting your house with a rotten trout.
13:54.08fourcheezeUK - 0 experience of BRI
13:54.22gr0mitin the UK it just works.
13:54.33gr0mitI have several boxes with BRI
13:54.46austinnichols101don't see much bri in the us
13:54.49fourcheezedoes ordinary asterisk 1.2.5 work or does it need some patch?
13:55.02gr0mityou need the bristuff patch.
13:55.07fourcheezeok
13:55.19gr0mithow many ports of bri do you need?
13:55.32fourcheezeprobably just a couple
13:55.35fourcheezeotherwise we would do pri
13:55.44jsharpBRI in the US is ranges from OK to downright painful, depending on what telco you're talking to.
13:55.49gr0mitif just one or 2 you can use Billion ISDN cards
13:56.03gr0mitI have 3 in my box at home
13:56.08austinnichols101hardest part was always ordering, with configuring a close second
13:56.20gr0mit1 on a BRI, the other two on ISDN terminals
13:56.47fourcheezeok
13:57.14jsharpordering & configuring was never a problem for me.  Keeping it up and running and beating the telco into understanding that problems were in their infrastructure...that was the hard part.
13:57.21fourcheezeis bri the same as isdn2 ?
13:57.24gr0mitbut a better solution is a 600 euro card
13:57.27gr0mityup.
13:57.29jsharpyet, it was fun.   I love abusing the phone company.
13:57.41gr0mityou just need a BT ISDN2e line
13:57.50austinnichols101jsharp: last time I order was when they introduced the color codes for bri flavor
13:58.02austinnichols101supposedly to make ordering easier
13:58.04jsharpcolor codes?
13:58.27tzangerwhee
13:58.32austinnichols101pri 'orange' I think it was.  They (intel and the telcos) were working together to make it easier to order
13:58.39tzangercolour codes for BRI flavour?  sounds interesting
13:58.42austinnichols101so you could just say 'orange' and they would know how to configure
13:58.50jsharpI always just called SBC and said "I need an ISDN line at 123 Main Street".
13:58.57austinnichols101problem was that the actual staff at the telcos had the same response you just did
13:58.59austinnichols101:)
13:59.07tzangerI want an orange swirl PRI with moccacino sprinkles and just a hint of cream
13:59.09jsharp2 weeks later, a telco monkey with an ISDN test would show up and punch it down.
13:59.12austinnichols101this was 15 years ago
13:59.13gr0mitfourcheeze, what is your application?
13:59.26tzangerthat's what happens when you do that, you end up Starbucks'ing your PRI
13:59.27fourcheezejust as a backup if dsl fails
13:59.37tzangerand nobody, including you, knows exactly what it is you're ordering
14:00.08gr0mitdsl ? fails? never, surely!
14:00.13tzangerhaha
14:00.16fourcheezewell, my preference is for a second dsl
14:00.18gr0mitwhere in .UK are you?
14:00.19tzangerme fail english?  That's unpossible!
14:00.27fourcheezegr0mit: farnham surrey
14:00.28austinnichols101tzanger: but then you're unique and not following the herds
14:00.41fourcheezegr0mit: at least that's where the office is - I'm in Wiltshire
14:00.43gr0mitaaah. we are not far from you!
14:01.00gr0mitI am in Delightful Basingstoke!
14:01.06tzangeraustinnichols101: true
14:01.09fourcheezeach
14:01.12tzangerI am a unique and special person
14:01.37fourcheezegr0mit: poor sod
14:02.09*** join/#asterisk trelane_ (n=trelane@mail.allthingsit.com)
14:02.37gr0mitI _like_ Basingstoke....especially now the sun is shining.
14:03.48fourcheezeit just has no soul
14:04.00gr0mitif you just want a backup why don;t you just use the analogue line you are using for your ADSL?
14:04.21gr0mitISDN is too expensive and too reliable just as a backup
14:04.34tzangerhaha
14:04.36tzanger"too reliable"
14:04.58gr0mit:-)
14:04.58fourcheezewell, another idea is to use the 128k isdn connection and get about 5+ channels of g729 down there
14:05.04*** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe)
14:05.14gr0mitnah. G729 sucks
14:05.27fourcheezeg729 just is
14:05.50austinnichols101g729 r0x0rs
14:11.27cfhis possible use a hfc card in NT mode ?
14:12.34gr0mitcfh - yes it is.
14:15.06cfhgr0mit, i have a atlantis card single bri HFC cologne chipset and it works good on te mode, but when i try to switch it on NT mode asterisk dont
14:15.14cfhworks
14:15.46gr0mithave you made the crossover cable with the 100 ohm resistors?
14:16.10gr0mitand have you modprobed the card with the correct parameters to set it to NT mode?
14:16.22cfhyes but without resistor
14:16.44cfhzaphfc.o modes=1
14:16.48gr0mitah. you know a normal ethernet xover will not work, don't you?
14:17.28cfhno i make a cross isdn cable
14:17.45gr0mitok good.
14:17.59gr0mitwell you MUST put the resistors in or it wont work
14:18.01cfhlike this http://www.pro-linux.de/work/asterisk/asterisk-1.html
14:18.53*** join/#asterisk shiznatix (n=shiznati@213-35-232-186-dsl.end.estpak.ee)
14:19.03cytrakgood mornii
14:19.46cfhbut i try to start asterisk and i try to do ' pri show span 1' the pri command there isnt
14:20.13cfhand if the card is on te mode the pri command there is
14:20.29gr0mitwhat version of astertisk are you running?
14:20.32*** join/#asterisk fuzzbawl (n=fuzzbawl@69.44.167.126)
14:20.45fuzzbawlhey all
14:23.16*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:28.00*** join/#asterisk rbd (n=robbyd@cpe-066-057-011-095.nc.res.rr.com)
14:29.23*** join/#asterisk BattleRaT (n=BattleRa@oliver.altascumbres.cl)
14:29.34rbdany benchmarks/loadtesting on the use of asterisk as a conference server (meetme)? I was wondering how many sessions a midrange server box (DL360, 2x 3.2GHz, 2GB ram, etc) could handle
14:29.48nettieHi guys, I just bought a couple of polycom soundpoint ip501 to use with my remote asterisk server. All the phones are located in my lan which has a router that nats the internal private addresses to the internet. When I try to call a local phone I only get one way audio. I think this might be a NAT problem because all the sip "friends" are configured with "nat=yes" anyone can suggest me a workaround please? Of course setup a vpn is not an
14:31.08jsharprbd:  Depends on how your calls will be coming in.  Voip or PSTN?
14:31.27BattleRaThummm, I guys
14:31.34BattleRaTi have a little and simple question
14:31.44BattleRaTCan i create a " high trafic callCenter " System (Profesional predictive dialers, robust inbound y outbound system, good stats generators, etc) with Asterisk ?
14:32.37BattleRaTAsterisk work fine for me with one FXO post and SIP softphones, but...  Callcenter is a big project
14:33.15jsharpThere are people who have built good sized call centers around Asterisk with predictive dialers.
14:33.34jsharpSo it can be done.  How easy/reliable/expensive it is, I dunno.
14:33.48rbdjsharp, VOIP (SIP trunk)....using ztdummy probably
14:33.49BattleRaThumm can you help me with some " good " opensource programs to do this?
14:34.38BattleRaTa good " predictive dialer " to put in a production callcenter?
14:34.48jsharprbd: Hrm.  Now it depends on what codecs you'll be using.  That's going to beat up your CPU.
14:36.08jsharpBattleRaT: http://astguiclient.sourceforge.net/vicidial.html
14:38.36*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
14:38.48*** join/#asterisk Tili (n=Tili@82-217-236-131.cable.quicknet.nl)
14:38.51BattleRaTlet me see
14:38.59Tilihey is asterisk available for 64 bit platforms
14:39.06*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
14:39.34rbdjsharp: yes, definitely... is it possibly to have asterisk convert it out to mp3 on the fly? (if not, I can use wav and then run it though lame). When I posed this question I was thinking wav out, but if it can be converted to mp3/ogg on the fly that would be even better
14:40.35jsharprbd:  MP3 codec in a soft phone?
14:42.33rbdjsharp: nope, asterisk will be pulling in the conversations as g711  (or g729 but afaik it can't decode that?)....I'd like to save them to disk as mp3. at least looking at the record/monitor command's documentation, mp3 isn't a supported output format (would I need to add a handler)
14:42.54cfhgr0mit, asterisk 1.0.9
14:43.13*** join/#asterisk apardo (n=apardo@62-14-56-136.inversas.jazztel.es)
14:43.24cytrakfugitivo: hey did you get the siemens setup ?
14:43.25jsharpOh.  Okay. g711 won't be all that hard on your machine.  You can decode g729 with a licensed (and paid for) codec, but that *will* beat up your machine if you run lots of calls.
14:43.31rbdnevermind on that last part... http://www.voip-info.org/wiki/view/Monitor+stereo-example (mono is fine though....toll quality)
14:44.02gr0mitcfh: are you running bristuff ?  you must be, right?
14:44.30cfhyes bristuff
14:44.38gr0mitmkay
14:44.59cytrakhow can I exit the CLI without killing the app .... I started * with -cvvv
14:45.29shiznatixi want to setup my linux box as a fax device for iaxmodem to send / recieve faxes to. how do i do this without an actual fax machine?
14:45.34Zeeekcytrak you can't use safe_asterisk or asterisk -cvvvv &
14:45.49shiznatixwill i ever make it out of this fax world hell alive?
14:46.10austinnichols101alive, but not unscathed
14:46.39cytrakis there a daemon option ?
14:47.05*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
14:48.01Zeeeklook at safe_asterisk
14:48.07*** join/#asterisk cjk (n=cjk@80.92.64.103)
14:48.40*** join/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com)
14:50.03kreilmeier* is constanly ignoring to jump into the per-user-defined context, but always going into the domain-specified default context. how can i solve this issue?
14:51.25*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:52.30*** join/#asterisk cfh (n=luca@82.193.23.6)
14:53.51shiznatixis it basically futile to try to figure out how to fax things without some actual fax machines?
14:54.05RoyK~foip
14:54.06jbothmm... foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
14:54.37nDuffshiznatix: hook iaxmodem up to hylafax.
14:54.49shiznatixthen what
14:54.51nDuffshiznatix: that's what iaxmodem is *made for*.
14:55.10nDuffshiznatix: then what? Then use hylafax to send faxes, or set up a recvfax script to handle received ones.
14:55.11RoyKiax does not have support for fax AFAIK
14:55.14*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
14:55.16shiznatixok then how to do i send a fax with iaxmodem. there is no documentation on that
14:55.26nDuffRoyK: iax is a lowish-level protocol; it doesn't care what runs on top of it.
14:55.42nDuffshiznatix: you use HylaFAX to send the faxes.
14:55.53shiznatixok then how do i use HylaFAX to send a fax
14:55.54nDuffshiznatix: iaxmodem just provides an AT-command-set level interface.
14:55.57RoyKwell - you fuckup the fax timing without stuff like t.38, so it does not work
14:56.00nDuffshiznatix: RTFM.
14:56.06RoyKnDuff: RTFM
14:56.07shiznatixi did, about 543 times
14:56.21RoyKhe even ounted
14:56.22RoyKcounted
14:56.25nDuffRoyK: well, that's why iaxmodem<->asterisk is supposed to be done over a local connection or a dedicated network with nothing else on it.
14:56.32nDuffRoyK: I not only read TFM, I'm on the mailing list.
14:56.58nDuffshiznatix: the HylaFAX manual?
14:57.03nDuffshiznatix: or the iaxmodem one?
14:57.03RoyKnDuff: fax is possible on a dedicated link, but then, still, it can be hard
14:57.23tzangersip.conf can define pickupgroups just like zapata.conf right?
14:58.04shiznatixwell neither have 'this is how you send a fax'
14:58.07nDuffRoyK: the iaxmodem manual discusses which codecs to use over your dedicated link to get reliable results -- and it *does* get reliable results. The folks reporting problems are typically seeing bugs in iaxmodem's AT command set emulation or SpanDSP's DSP emulation, not issues related to the link.
14:58.23nDuffshiznatix: the hylafax man pages cover the sendfax command, so you haven't read them enough.
14:59.09nDuffshiznatix: also, make sure you're running Lee's port of HylaFAX (the one from hylafax.sourceforge.net, not www.hylafax.org)
14:59.19nDuffs/port/branch/
14:59.22RoyKstill, the way to get reliable fax, is t.38
14:59.25RoyKor t.38
14:59.46nDuffRoyK: yes, but nobody has a Free implementation yet that I know of.
14:59.51shiznatixnDuff, what if i want to send a fax to a IP and not a number?
15:00.04nettieyeah faxes without t.38 simply doesnt work
15:00.04RoyKnDuff: coppice has done quite a bit
15:00.06nettienoway
15:00.22nettieI tested gazillions of carriers
15:00.22nDuffshiznatix: Don't. Sending faxes over VoIP long-distance is just asking for trouble.
15:00.29cytrakdo I need to configure my zaptel to have incoming channels and outgoing channels ? I'm unable to get any sound when calling * extensions from a regular phone
15:00.32RoyKnDuff: although some of it is gpl and therefore bloody impossible to use in asterisk
15:00.35nettieeven short distance
15:00.38shiznatixnDuff, its inside my office and i have to be able to do it
15:00.45RoyKnDuff: t.38 works over long distance
15:00.47RoyKbeleive me
15:00.57nettienow i'm using a grandstream handytpne 286 with a t.38 carrier and it works perfectly
15:00.57shiznatixnDuff, or would this be there a PTSN comes in?
15:01.20RoyKwe have welltech sipgate with t.38, and that works like a dream
15:01.23tzangernettie: who's your t.38 carrier?
15:01.26nettiet.38 works everytime short/long/medium distance. it just work!
15:01.30nettieskypho
15:01.33nettiewww.skypho.net
15:01.50RoyK~seen zoa
15:01.58jbotzoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 7d 22h 18m 41s ago, saying: 'it looks kinda suspicious :p'.
15:01.58nettieaccount is free
15:01.58nettierates are good imho
15:02.04nettieand t.38 is rock stable
15:02.13nettielast week we sent about 300 faxes
15:02.16nettieflawless
15:02.21nettieworldwide
15:02.23rbddoes asterisk have mp3 output functionality with the monitor command (e.g. Monitor(mp3,myfilename) ) ...if not, what might be required to add this. I would need conversion to mp3 on the fly, not after the conversation ends
15:02.24*** join/#asterisk coppice (n=chatzill@34.199.17.210.dyn.pacific.net.hk)
15:02.48*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
15:03.13*** join/#asterisk adker (n=adker@70-100-227-113.br1.glv.ny.frontiernet.net)
15:03.50*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)
15:04.25*** join/#asterisk oej (n=oej@apollo.webway.se)
15:04.52oejMe back?
15:04.57*** join/#asterisk ozant (n=zet@mail.grid.com.tr)
15:05.04*** part/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl)
15:05.10RoyK:)
15:05.21oejWho am I?
15:06.04MikeJ[Laptop]some weird northerer
15:06.09MikeJ[Laptop]northerner
15:07.15coppicenortherer sounds weirder than northener
15:08.40MikeJ[Laptop]yes
15:08.59MikeJ[Laptop]but your all just alaw's anyways...
15:09.15coppicei'm a ulaw at present
15:09.31coppicethough I was born and raised an alaw
15:10.46MikeJ[Laptop]coppice, where?
15:11.03MikeJ[Laptop]US?
15:11.09coppicenope
15:11.12*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfik5.dialup.mindspring.com)
15:11.19MikeJ[Laptop]in HK?
15:11.25coppiceyep
15:11.37*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfik5.dialup.mindspring.com)
15:11.38MikeJ[Laptop]didn't know HK was ulaw...
15:11.40MikeJ[Laptop]cool
15:11.44MikeJ[Laptop]where else
15:11.50coppiceweird, huh? a British colony chose ulaw
15:12.12coppiceulaw is used in taiwan and japan too
15:12.12*** join/#asterisk Dovid (n=Dovid@89-138-33-253.bb.netvision.net.il)
15:12.13MikeJ[Laptop]yeah
15:12.25MikeJ[Laptop]really?
15:12.36MikeJ[Laptop]I thought it was jsut the silly americans
15:14.49nDuffshiznatix: why do you have to do it? There's probably another way.
15:15.26nDuffshiznatix: If you want the asterisk server and the hylafax server to be apart, for instance, you can run the AT commands over the network -- unlike the IAX stream, they aren't time-sensitive.
15:15.27Dovidanyone know how to do an announce me feature ?
15:15.43nDuffshiznatix: you can also do fax<->email or use HylaFAX's network support for remote access.
15:17.28*** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk)
15:17.29shiznatixnDuff, this stuff is way too complicated for me. I just want to be able to be like 'computer, send this fax to this users number on asterisk'
15:17.52shiznatixnDuff, and that fax either be saved in a special folder on the asterisk server or somtin else like that
15:18.15shiznatixnDuff, but I have 0 fax machines to test anything with and I dont know how to setup anything
15:18.50Dovidanyone know how to set up an anounce me feature like the person has to say thier name and the one getting the call can press 1 to accpet and 2 to go to a vm or somethig ?
15:19.26*** join/#asterisk GuruDom (n=domiplus@66-202-165-66.rev.knet.ca)
15:20.47*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) [NETSPLIT VICTIM]
15:20.48*** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM]
15:20.48*** join/#asterisk flynux (i=v8hy3c1@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM]
15:21.43shiznatix~foip
15:21.45jbotrumour has it, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject
15:22.20*** join/#asterisk wantmoore (n=wantmoor@66.20.49.205)
15:22.23shiznatixhow do i setup hylafax to use iaxmodem software module?
15:23.22shiznatixthey are both installed but wont communicate with eachother
15:23.34*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
15:23.50tmccraryI have a TE110P and I am getting a lot of distortion
15:24.10tmccraryThe ring and the voice are both distorted
15:27.58*** join/#asterisk atta (n=ansgar@213-239-206-114.clients.your-server.de)
15:28.33*** join/#asterisk brc_ (n=brian@pdpc/supporter/basic/brc)
15:29.18tmccraryokay, actually everything is really slow
15:30.43*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
15:30.59gr0mittmccrary sounds like you have an IRQ shareing problem
15:31.05tmccraryI don't think so
15:31.15tmccraryI have APIC setup and all that
15:31.24tmccraryplus, this is the only pci device
15:31.45gr0mitwell you need to check it is not sharing with a hard disk controller
15:31.55gr0mitor graphics card
15:32.52*** join/#asterisk rushowr (n=rushowr@cpe-65-189-187-159.columbus.res.rr.com)
15:33.04rushowrhello all
15:33.54*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
15:34.51rushowrhey can anyone help me with a minor question? I'm trying to perform actions after a dial, after the caller and callee have hung up
15:35.05Dovidok
15:35.05rushowrI'm using the g option to the Dial command but it's not progressing
15:35.15rushowrI get the dial and then nothing
15:35.49rushowrafter I hang up the test call, no further progression in the dialplan
15:36.35rushowrAsterisk 1.2.1 is the veresion
15:36.41rushowr(s/p) version
15:37.54rushowranyone else had issues with this
15:39.56*** part/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com)
15:41.08atta- Executing Dial("SIP/10-2a53", "SIP/03943####@sipgate|120") in new stack
15:41.08attaMar 23 16:39:15 NOTICE[16785]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
15:41.08attaI Cont call out of my networke can anyone help me??
15:42.39*** join/#asterisk redondos (n=redondos@190.48.44.119)
15:42.42Dovidatta: whats the problem
15:43.16attaDovid:  Unable to create channel of type 'SIP' (cause 3 - No route to destination)
15:43.21*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com)
15:43.29Dovidpaste ur dial plan in pastebin.ca
15:43.42Dovidatta: paste ur dial plan in pastebin.ca
15:43.48attaOK
15:44.37SkalTuraguys
15:44.49SkalTuraactually...
15:44.54SkalTurawait a minute, i might have this solved ;)
15:46.16Dovidatta: post the url to it here and i will look at it
15:46.41rushowrany thoughts about the 'g' param for Dial app not working properly?
15:46.50attasorry  pastebin.ca theams to be down :-(
15:47.08*** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net)
15:47.09jsharppastebin.com
15:47.33atta:-)
15:47.34gandhijeewhats a good language to start doing AGI with?
15:47.39jbalcombperl
15:47.43jsharpWhatever language you can write in.
15:48.01jsharpAs long as it groks STDIN and STDOUT, you're all good.
15:48.09gandhijeeO
15:48.14SkalTurayeah got it solved :)
15:48.19gandhijeecool
15:48.50SkalTuramy first asterisk setup will be for a large company here, individual boxes to each of the 3 stores, plus links to PSTN
15:50.21Dovidatta: try pastebin.com
15:50.33attayes one second
15:50.47nDuffshiznatix: setting HylaFAX up to use iaxmodem is covered in the iaxmodem documentation. See the HOWTO on the iaxmodem page, at the end where it starts with "If you application is HylaFAX".
15:50.57*** join/#asterisk kreilmeier (n=kreilmei@hq.commoveo.com)
15:50.59cytrakis it possible to set gsm on zapata.conf ?
15:51.06attaDovid: http://pastebin.com/618118
15:51.24nDuffshiznatix: that covers the only differences from the regular HylaFAX setup.
15:51.50Dovidatta: what part of the dial plan u having a problem with ?
15:52.01kreilmeierwell, i'll try again. is anyone in the mood to explain or discuss sip user/peer matching in asterisk 1.2?
15:52.05*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
15:52.21attacall sipgate form inten out too the hole world
15:52.42kreilmeieri just don't understand how asterisk matches incoming contexts.
15:52.46Dovidatta: which context ?
15:54.14gandhijeekremil: where did u find some info to start on that?
15:54.21gandhijeei need to do the same thing, but with IAX
15:54.29nDuffatta: "too" means also; you mean "to" for "in the direction of". Likewise, "hole" means an opening in something; "whole" means complete.
15:55.08kreilmeiergandhijee: is remil me?
15:55.37attathanks i have forgotten the sipgate part ;-)
15:55.54attaDovid: thanks i have forgotten the sipgate part ;-)
15:56.51Dovidkk
15:57.02kreilmeiercan it be that a defaultcontext for a domain (sip.conf) like "domain = sip.something.com, fromLocalSipSomething" always has precedence over a context defined in a user/peer/friend definition?
15:57.33rushowrI believe so
15:58.22gandhijeeKermilmeier: yea, i didn't feel like typing your hole nam
15:58.22gandhijeee
15:58.38kreilmeierOK
15:58.39*** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe)
15:59.37kreilmeierwell there is http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer and the part inoming connections at http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+channels
16:00.06kreilmeierbut all this is missing information regarding domain-specific default contexts.
16:00.35rushowrI'm guessing noone has any idea about the issue with the 'g' param to Dial() possbly not working.... I'll try some more debugging
16:01.09kreilmeieri also found some info at http://www.asterisk.org/doxygen/Config_sip.html in the "SIP DOMAIN SUPPORT" section
16:01.28*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)
16:02.18kreilmeierrushrowr: have you investigated this problem or are you guessing?
16:02.59rushowrNot sure what you mean, I'm doing testing on my system now, and there's no progress through the dialplan after the dial
16:03.11cytrakhmm I keep setting my zapata.con for echocancelling but whenever I do a zap show channel 1 or any other channel  128 taps unless TDM bridged, currently OFF
16:04.04jsharpAre you looking at active channels?
16:04.22SkalTurahey
16:04.44SkalTurai need to get different language speeches on my asterisk box (the standard once, voicemail etc.), where to look for them?
16:04.46*** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net)
16:05.04SkalTurai'm looking for a finnish set
16:05.42*** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc)
16:05.52fourcheezeSkalTura: do it yourself
16:05.55fourcheezeor google
16:06.03fourcheezeor find a sexy female to do it
16:06.12SkalTurafourcheeze: well i don't have a clue what keywords i should type into
16:06.41rushowrSkalTura - try googling finnish asterisk pbx voice files
16:06.44[TK]D-FenderHey, can anyone here confirm is ther is a major problem at Broadvoice right now?  A site I'm debugging is timing out all over the place...
16:06.46jbalcombhow do I resolve the PCI Parity Error on my Dell 2850 that appears to be a conflict between an on-board NIC and my Digium TE411P card?
16:07.22bweschke[TK]D-Fender: which proxy? I'm on boston right now and I appear to be Ok
16:07.36SkalTuranothing with those keywords neither :(
16:07.40fourcheezewww.voip-info.org/wiki-Asterisk+sound+files+international
16:08.08*** join/#asterisk Jon335 (n=jon335@ottawa-hs-64-26-167-111.d-ip.magma.ca)
16:08.29SkalTuravoip-info.org is timing out :(
16:08.54rushowrit's slow sometimes
16:09.04jsharpMust be on a windows machine.
16:09.12SkalTurasly? try like nothing moves
16:09.17fourcheezeSkalTura: http://72.14.207.104/search?q=cache:ywnwKngRF9sJ:www.voip-info.org/wiki-Asterisk%2Bsound%2Bfiles%2Binternational+asterisk+voicemail+finnish&hl=en&gl=uk&ct=clnk&cd=1&client=firefox
16:09.34rushowrkreilmeier (or anyone else) http://pastebin.com/618155
16:09.53fourcheezethe short answer is there doesn't seem to be finnish
16:10.20SkalTurano finnish :( shame
16:10.28SkalTurai guess i gotta get someone to talk then
16:11.55*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
16:11.59*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
16:12.08*** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net)
16:12.24*** join/#asterisk HamYaI (n=HamYai@125.24.7.238)
16:12.59HamYaIhow can we pass some parameters to agi scripts?
16:13.04*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
16:13.29*** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
16:13.39Jon335Has anyone created a weather script for canada?
16:15.27asterboyjon335, sounds cool...what weather script works with *?
16:15.43austinnichols101Jon335: just need a weather source similar to what's available with noaa
16:15.45*** join/#asterisk frenzy (n=frenzy@196.46.104.33)
16:15.54asterboy~noaa
16:15.59austinnichols101would be easy to modify weather.agi
16:16.08frenzystupid question... How do I forward a particular extension to multiple SIP
16:16.08asterboy~agi
16:16.10jbotagi is, like, the Asterisk Gateway Interface...  similar to CGI for web applications AGI lets you script call control and access databases using your favorite language.  AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages
16:16.17austinnichols101actually I think it uses national weather service
16:16.45fourcheezefrenzy: forward from where/
16:17.05frenzylocal SIP forward
16:17.16jsharpHave one extension dial multiple phones?
16:17.20SkalTurai assume there is hardware to use a GSM SIM card with asterisk for receiving calls and making calls
16:17.22austinnichols101then get stephen hawking to do the text2speech
16:17.37jsharpexten => 1234,1,Dial(SIP/foo1&SIP/foo2&SIP/foo3)
16:17.41frenzye.g. calls on 123 to send the call to 333 and 444
16:17.45asterboyhttp://www.voip-info.org/wiki/index.php?page=PBX+Hunt+Groups
16:18.12asterboyvoip-info.org down again?
16:18.33*** join/#asterisk futura (n=user@12-210-203-61.client.insightBB.com)
16:18.40*** join/#asterisk salviadud (n=ralfalfa@201.138.132.150)
16:18.48asterboyI can't get access to voip-info.org...anyone else reporting same?
16:18.57Dovidasterboy: its up and down
16:19.00frenzyyup itd donw
16:19.04asterboylike a yo-yo
16:19.07[ProB]CrazyManquestion: TxFAX(filename[|caller][|debug]), when I now want to get debug information do I have to make -> TxFax(filename.tiff| |debug) ?
16:19.10Dovidlol
16:19.11*** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no)
16:19.12salviadudi report the same
16:19.27salviadudnot working for me either
16:19.59asterboyI apprecieate voip-info, but it really needs not only a face lift, but some reliable connectivity and equpment.
16:20.18salviadud$$$
16:20.20asterboyby face lift, I mean better content.
16:20.24asterboyyes $$$
16:20.28asterboylike most everything.
16:20.40asterboyTime to make a donation.
16:20.41salviadudif puff daddy would only contribute
16:20.45asterboylol
16:20.47RoyK£££
16:21.01salviadudwe should get some rappers on this thang
16:21.06asterboypuffy would if you made a black panther party out of it.
16:21.20jsharpVoip to your mother
16:21.22jsharpDawg
16:21.26asterboylol
16:21.28salviadudhaha
16:21.31salviadudi like that one
16:21.32*** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-66-36.d-ip.magma.ca)
16:21.35*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
16:21.56jsharpBustin caps in the telco's asses.
16:22.43salviadudthe least we could do...
16:22.49salviadudthe church of asterisk
16:22.58asterboyIf I land some telco contracts, I'll pump some money in.
16:23.14salviadudyou gonna bring up the G's holmes?
16:23.24asterboyThe way some of the guys brag about their monthly * income on here, there should be tons of cash already.
16:23.32*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
16:23.33salviadudthat be awesome...
16:23.36*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
16:23.39nettieguys anyone have a suggestion for my network addres translation problem please?
16:23.39fuzzbawlhow do you transfer calls? Is there a * command or something?
16:23.44salviadudwell, to be truthful, the money is great
16:23.58salviadudi get paid to say "yep, i know asterisk"
16:24.12salviadudright now, there are no real projects with asterisk
16:24.20rushowrI wish my money was great, I've been trying to make a go of freelance work since I left an ITSP, and I'm dyin'
16:24.20salviadudyet, they keep me here "just in case"
16:24.33Jon335So, about the NOAA weather source, the best I could find is this: http://tinyurl.com/ot7em
16:24.43salviadudrushowr, you gotta go small office first
16:24.51salviadudthen, climb to the top
16:27.32*** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net)
16:27.38kippihey
16:27.52SkalTuraabout money
16:27.56kippiI am looking for a cms system to link into my asterisk box, any ideas?
16:27.56SkalTurasalviadud: you done corporation PBXs?
16:28.12salviaduderrr, no
16:28.13SkalTurai'd like a private conversation with someone who has done corporate PBXs
16:28.23jsharpCorporate Asterisk PBXs?
16:28.26SkalTurai need to know howto price it right
16:28.35salviadudohhh, the mula
16:28.39salviadudjust overprice
16:28.44SkalTurajsharp: for big companies, asterisk pbxs yes
16:28.47salviadudyou can never go wrong with that
16:29.25fourcheezeSkalTura: work out how much you're worth, add a bit then double it
16:29.53SkalTurafourcheeze: ok... so i'm thinking now something like... perhaps 9k would be nice
16:29.53SkalTuraeuros...
16:29.53fourcheezeso
16:29.54fourcheezesay 20k
16:30.01jsharpHardware + 50%, 10-12 hours of time for a "basic" system.
16:30.04SkalTurathat's a bit top over of my comfort zone (9k)
16:30.09jsharpThen hourly for customization.
16:30.20fourcheezeSkalTura: what's your hourly rate?
16:30.57SkalTurajsharp: connecting 3 different stores (couple hundred kms apart), internal phones, then customer service system, and so that when someone calls to say at location a to that company, the call comes out to pstn at location b to save costs
16:31.10fourcheezeso not basic
16:31.17SkalTurafourcheeze: 48e per hour, but note that i have never before done PBXs, just coding and stuff like that
16:31.31fourcheezeok,we'll your price is too low for the corporate market
16:31.50fourcheezeat least 100Euros
16:32.02fourcheezeplenty of people charging 300-400
16:32.04SkalTurafourcheeze: you've done stuff for corporate markets?
16:32.15fourcheezeactually, not asterisk, but other things, yes
16:32.16SkalTuraholy shit
16:32.30SkalTuradidn't think prices were that high!
16:32.31SkalTurashit
16:32.39fourcheezesomeone else will contradict me if I'm wrong
16:32.46fourcheezeEuro~=$
16:32.47SkalTurai have to do PBXs for the largest finnish hosting company too :O
16:32.56jsharpCorporate mentality: More money == better
16:33.02SkalTuranope, 1euro = ~1.17USD
16:33.13fourcheeze1.17 ~= 1
16:33.25fourcheezein that case
16:33.31fourcheezeprice high
16:33.34HamYaIanyone knows whether we can pass parameters to cmd AGI's scripts?
16:33.36fourcheezeand aim to take on another consultant
16:33.46SkalTurai have next to 0 experience with PBXs
16:33.53fourcheezedefinitely get help then
16:34.03SkalTuraunfortunately, this stuff is being subcontracted to me
16:34.33SkalTuraand that subcontractor will piss his pants when from the usual 48e/hour, instead i go charging 250e/hour
16:35.07*** part/#asterisk futura (n=user@12-210-203-61.client.insightBB.com)
16:35.32SkalTurafourcheeze: you think that's really necessary? So far all stuff seems really easy
16:36.14SkalTura*everything
16:36.15fourcheezethere are lots of wrinkles
16:36.29SkalTuraaah, 'undocumented features'? ;)
16:36.31fourcheezeit all depends on how much you trust your ability to think on your feet
16:36.49SkalTurathink on my feet oO; what does that mean
16:37.01fourcheezedunno how that translates to finnish
16:37.16jsharpthink on your feed = adapt to problems you might have
16:37.17fourcheezeto be able to respond to rapidly changing situations
16:37.27SkalTuraaah ok
16:37.32SkalTurai think i'm quite good with that ;)
16:38.06fourcheezetry it then
16:38.23fourcheezethe worse that can happen is that it all goes wrong and you get sued
16:38.29fourcheeze:-)
16:38.33SkalTuralol x)
16:39.03jsharpFind an inexpensive consultant that you can keep on a leash if you get into problems, though.
16:39.16SkalTuraif things are handled well (ie. live system isn't brought down until the new one is ready & tested) --> worst case scenario --> it sucks & i don't get paid
16:39.33SkalTurajsharp: found already, it's called #asterisk @ freenode...
16:39.36SkalTura>;D
16:39.39jsharpHeh
16:39.48jsharpThats about as cheap as it comes, then.
16:40.03SkalTuradude, my budget is 0e
16:40.08SkalTuraj/k
16:40.32fourcheezethe thing you have to remember is that nothing is more important to a business than phones
16:40.39fourcheezemost businesses anyway
16:40.45fourcheezeeven in the age of email
16:40.51SkalTuraindeed
16:41.01fourcheezeso do price high
16:41.03SkalTurait's hell of a responsibility, thus hell of a payout ;)
16:41.05fourcheezeand get help
16:41.25fourcheeze9k Euro doesn't buy much of a traditional phone system
16:41.36SkalTurafourcheeze: afaik that buys next to nothing?
16:41.40fourcheezethat's really entry level
16:41.52fourcheezecertainly once you start doing multi-site
16:42.01SkalTurathis is going to be built to replace the current legacy system there, and to save costs... ladder is more important, saving costs is all they care
16:42.10fourcheezeit costs 2K just for a card on one exchange to allow it to talk to another
16:42.28SkalTuraouch!
16:42.44Jon335OK, so if I got the weather in XML, how hard would it be to turn that into voice?
16:43.10SkalTuraJon335: text2speech
16:43.29SkalTuraJon335: voip-info.org, i think i saw there already what you want to do
16:43.47Jon335SkalTura:http://tinyurl.com/rg4g5 this is the xml
16:44.19fourcheezeJon335: well you'll have to transform that to text first
16:44.20SkalTurafourcheeze: so 200-300e/hour that is... 3 sites and all.. i guess 20k is infact well priced, qutie on spot... they've been talking about savings of 50kE/yr already tho
16:44.26*** join/#asterisk saftsack (n=saftsack@p54A7D706.dip.t-dialin.net)
16:44.26Jon335But I would need to convert the tags to real words somehow
16:44.50SkalTuraJon335: small piece of scripting fun that is :)
16:44.53fourcheezeSkalTura: if you're a subcontractor you need to have a meaningful conversation with the main person
16:45.07fourcheezegive him your numbers and what you expect him to sell it on for
16:45.13fourcheezeor her
16:45.17*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
16:45.18SkalTuraJon335: i can write you that convertor, if you make my time worth :) Should not take that many hours
16:46.10SkalTurafourcheeze: indeed, he's been busy selling it already, and i don't have a clue about the numbers he's been talkin' bout yet... Everything is still kind of pre-examination mode still
16:46.41hfbOkay, I found an old unused machine that supported pci 2.2.  The zaptel TDM400P seems to work.  I ran zttest and pretty much saw 99.987793 with an occasional 100.0000000.  I'm now compiling all the packages needed.
16:46.45*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
16:46.45*** mode/#asterisk [+o anthm] by ChanServ
16:47.08SkalTurafourcheeze: that main guy, who subcontracts from me, i just built an e-mail system lately
16:47.30Jon335SkalTura:No one has written an agi xml parser already?
16:47.34SkalTurahey, can i use an old modem as PSTN interface? :O
16:47.44SkalTurai got struck with a wild idea that is x)
16:47.45*** part/#asterisk cfh (n=luca@82.193.23.6)
16:48.01SkalTuraJon335: most probably not the format you need
16:48.07SkalTura*not to the
16:48.09DovidSkalTura: nope
16:48.17SkalTuradamn
16:48.25DovidSkalTura: hehe. thats everyones dream
16:48.29Dovidgoto get a voice card
16:48.47Dovidbesides ur supporting digium, the ones who gave a great PBX for free
16:49.12mogormanwoohoo
16:49.19mogormansupport me ^_^
16:49.20SkalTurablegh, that sucks... if i do a little bit of soldering and use the modem to manage the call, and sound card as digital<>analog converter? ;)
16:49.33kippiI am looking for a cms system to link into my asterisk box, any ideas?
16:49.57fourcheezekippi: I've considered using plone for that
16:50.02SkalTurakippi: depends what you are looking out of that cms, and interworking with *
16:50.03jsharpSkalTura:  That's an uglyugly hack.
16:50.10*** join/#asterisk rushowr (n=rushowr@cpe-65-189-187-159.columbus.res.rr.com)
16:50.16SkalTurajsharp: indeed, but it would work, yes? ;)
16:50.24*** join/#asterisk lorinc (n=ang@caracas-3939.adsl.interware.hu)
16:50.25rushowrI'm back, sorry about the drop
16:50.34jsharpNot without some hacking of Asterisk to handle the call flow.
16:50.35rushowranyone ever have any thoughts on the 'g' issue for Dial()?
16:50.55SkalTurajsharp: ok, so it would work, but definately not worth the effort x)
16:51.16jsharpOnly if your time is free, and we've already established it at 48e an hour.
16:51.25fourcheezeg    - Proceed with dialplan execution at the current extension if the    destination channel hangs up.
16:51.25kippiI want somthing that will make a note of calls make etc to the cust
16:51.34Jon335SkalTura: To make it harder the SDK says you have to cache the weather data for 2 hours
16:51.49fourcheezekippi: oh you mean a CRM?
16:52.01rushowrfourcheeze: problem is it isn't progressing
16:52.04fourcheezerushowr: never tried that - doesn't it work?
16:52.18kippiYep
16:52.26Hmm-workdigium gave a great pbx for free? a lot of work came from people who have never worked for digium
16:52.40rushowrfourcheeze: didn't work for me
16:52.44mogormanhey lets not do this again this morning
16:52.46Dovidif not for them u wouldnt have it
16:52.47fourcheezerushowr: what did you do?
16:52.47mogormanplease
16:52.50Dovidtget started it
16:52.59redondosCan anyone please tell me what kind of connector this card has? (I'm assuming BNC, but I need to be sure.) http://xrl.us/ki6u
16:53.01kippithe only one that i can find is SugarCRM
16:53.08rushowrfourcheeze: http://pastebin.com/618155
16:53.45mogormanthey are rj45 jacks
16:53.54SkalTuraJon335: not that hard, you know
16:54.00mogormanman tormenta cards are old.....
16:54.24redondosmogorman: Was all that for me? You sure it's RJ45?
16:54.30mogormanpostive
16:54.32redondosmogorman: What should I use instead of that card?
16:54.32mogormani have one
16:54.33jsharpYes, those are RJ45s.
16:54.40Jon335SkalTura: I would pay you to make the script, but I don't have the money
16:54.41fourcheezerushowr: I'm not sure why that doesn't work, but have you tried G ?
16:54.58mogormanat least get digium or sangoma
16:55.04mogormanthat thing is gonna give you heart ache
16:55.12[TK]D-Fenderbweschke : one of Broadvoice's CST proxy's seems to be assy today.  I switched them to a Miami one and they are up and running fine
16:55.16rushowrfourcheeze: one sec, reviewing
16:55.24redondosmogorman: Ouch, why do you say that?
16:55.38rushowrfourcheeze: G transfers the call after it's picked up, I don't want that
16:55.43fourcheezerushowr: ahh yeah
16:55.45jsharpI love this auction.   All the functionality you get with a single board.  Wow.
16:55.46fourcheezehmm
16:55.52rushowrfourcheeze: I want processing after the hangup
16:56.02rushowrwe're doign postcall billing
16:56.06mogormanit is prone to problems
16:56.13SkalTuraJon335: btw, asterisk@home has somekind of weather forecast thing built in
16:56.23mogormanyeah agi script
16:56.26mogormanits on the web
16:56.27mogormansomewhere
16:56.28redondosmogorman: ok, thank you.
16:56.41SkalTurawhat's festival btw?
16:56.43redondosJust one more: can you use X100P cards to connect a normal phone to them?
16:56.47mogormanit has a lot of older issues, mostly to do with the parts on the boards
16:56.48redondosSkalTura: text-to-speech software
16:56.58jsharpX100P cards are for connection to lines, not phones.
16:57.03*** join/#asterisk blaylock (n=seth@snap.helixsystems.com)
16:57.03mogormanboth of the newer sangoma and digium hardware are of much higher grade
16:57.07redondosjsharp: thanks.
16:57.41blaylockanyone know which file I need to edit to configure the line monitoring feature with polycom 601 phones?
16:58.13redondosmogorman: So if that card has an rj45 jack, is it incompatible with E1 lines that end in a BNC connector?
16:58.45asterboyblaylock, are you talking about a Polycom feature, or just *.
16:58.52blaylockwell both
16:58.54jsharpYou can get a BNC to RJ45 balun.
16:59.03asterboyextensions.conf
16:59.06blaylockim not sure if its just the phone, or if its both * and the phone
16:59.09redondosmogorman: Just using a BNC<->RJ45 thingie?
16:59.11redondosk, cool :)
16:59.13*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
16:59.16SkalTurabtw, anyone know some hardware to connect a GSM to *? like just insert SIM card, or i can spare a phone too, that's no prob
16:59.19redondosThank you very much.
16:59.27mogormanbut not sure
16:59.30Hmm-workugh gsm
16:59.32asterboyHere is my line: exten => s,3,Monitor(wav,/var/spool/asterisk/monitor/Distance_${CALLERID})
16:59.43*** join/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50)
16:59.43mogormani havent seen a bnc cable since i was 12
16:59.44redondosWhat do you prefer over GSM, Hmm-work ?
16:59.51redondosmogorman: I live in South America.
16:59.51blaylockthanks asterboy
16:59.57*** join/#asterisk af_ (n=af@ip-164-94.sn2.eutelia.it)
16:59.58blaylockill give that a whirl
16:59.58Hmm-workanything hardwired
16:59.58redondosmogorman: That should explain it.
17:00.00mogormanheh
17:00.06redondos:/
17:00.11asterboyAlso, I use these lines to playback:
17:00.14mogormanthere is old tech everywhere, i just dont have to see it much
17:00.19asterboy<PROTECTED>
17:00.22redondosmogorman: You're very lucky :)
17:00.23asterboy<PROTECTED>
17:00.25SkalTurai would like that when people calls to my gsm #, it gets to routed Wifi phone and then to my phone, and the call gets recorded
17:00.33*** part/#asterisk frenzy (n=frenzy@196.46.104.33)
17:00.36Hmm-workgod my voice is shot
17:00.42nettieanyone kniow if nat=route is supposed to help when I want to call local phones behind NAT using a remote asterisk server please?
17:00.46mogormani know it redondos
17:00.48blaylockhmm, are we talking about the same thing asterboy?
17:00.50asterboyYou can also use zapbarge to monitor a line.
17:00.56asterboy<PROTECTED>
17:00.57asterboy<PROTECTED>
17:01.06blaylockim talking about that extra device that goes on the side of the phone
17:01.21asterboyoh, ya I need one of those also.
17:01.22blaylocktells you who is using their phone
17:01.34jsharpSkalTura:  I dunno about native Asterisk hardware, but there's a couple of companies that make SIP to GSM gateways.
17:01.36asterboyoh, so not just listening in.
17:01.43blaylockright
17:01.47blaylockyeah i dont need that
17:01.53SkalTurajsharp: neat, know the names?
17:01.54blaylockjust know who's line is active
17:02.14jsharpVoiceBlue is the first one I can think of.
17:02.28asterboycouldn;t you get that from CLI and debug info.
17:02.30asterboy?
17:02.53blaylockheh, not sure, im kinda new with asterisk
17:02.56asterboyyou want it at the phone, so you know what other extension is using the line.
17:03.14blaylocki figured there would be some config file that i could add something too...that type of think
17:03.17redondosIs there anything at all that would let me connect normal phones to an asterisk box?
17:03.20blaylock*thing
17:03.36jsharpredondos:  TDM400P with FXS modules.
17:03.37asterboyredondos, ATA adators
17:03.43jsharpOr those.
17:03.53SkalTuranow where can i get one of those...
17:03.59asterboylike an unlocked Vonage or Sipura or IAX adaptor.
17:04.05redondosAlright, thanks.
17:04.07jsharpATA-186
17:04.09asterboyGrandstream makes em also
17:04.15redondosI never saw those. Can you show me one so I know what to look for?
17:04.22jsharpGrandstream HT-286/386/486
17:04.28asterboyebay for Grandstream or "FXS"
17:04.31SkalTurawow, found a finnish company selling some of the 2n stuff
17:04.33redondosk, thanks
17:04.59*** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
17:05.38brettnemewww grandstream
17:05.47asterboyblaylock, I don't have it working yet, but I like the idea of being able to see who has an extension open at the Polycom phone.
17:05.53SkalTurabut no pricing :(
17:06.18asterboyI know you can have shared lines but that won't tell you 'who' exactly is using the line
17:06.30Jon335SkalTura: I have A@H, but the script only works with NOAA, which is only in the US
17:06.53brettnemasterboy: you can watch extensions on the polycom phone.. but only up to 7
17:06.55blaylockasterboy, yeah i think its in the directory file actually
17:07.13blaylockim looking at a polycom SIP manual now
17:07.14asterboyoh yes, that's right.
17:07.31SkalTuraJon335: oh
17:07.40SkalTurawell, i'm outta for awhile, l8er
17:07.44blaylockthat will be fixed in the new upcomming release i hear
17:07.46asterboyHence the add on module to monitor more lines.
17:07.59asterboyrelease of firmware?
17:08.00brettnemthe add on module won't monitor more than 7 lines!!
17:08.07brettnemyes, the firmware is broken.
17:08.10asterboyya I heard that.
17:08.13blaylockasterboy, no asterisk
17:08.22asterboydidn't know the firmware was broken though.
17:08.35*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
17:08.48brettnemyep.. polycom phone, regardless of configuration or expansion modules will only monitor a maximum of 7 lines period
17:09.10brettnemin SIP mode..
17:09.27asterboyIs there a way to send lines of text to the display of a Polycom phone?
17:09.33yuta-vcnethi nettie, regarding the phones behind NAT, do they support STUN?
17:09.37asterboyKinda like the clock text?
17:09.42brettnemnot the < IP600
17:09.44brettnemthe 600 maybe
17:09.49brettnemwith the browser
17:09.57Dovidanyone know if i can send text to an xlite phone ?
17:09.58*** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br)
17:10.05Jon335So haw would I go about making a Weather XML parser in agi? Feed at http://tinyurl.com/rg4g5
17:10.12*** part/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
17:10.12*** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com)
17:10.14brettnemDovid: from what?
17:10.15yuta-vcnetI have been playing with a similar setup, I think nat=no would work
17:10.20*** join/#asterisk GuruDom (n=domiplus@66-202-165-66.rev.knet.ca)
17:10.29brettnemJon335: just write the code.. should be easy
17:10.30Dovidfrom the CLI or an AGI
17:10.33nettieyuta-vcnet I dont think so :(
17:10.45nettieyuta-vcnet those are polycom
17:10.52nettieyuta-vcnet ip501
17:10.52redondosjsharp: will this balun work for an E1 line? http://xrl.us/ki63
17:10.56redondosmaybe someone else knows?
17:11.07Hmm-worki've used baluns on e1's before
17:11.18brettnemDovid: asterisk's text support is really weak.. I'd try it with sipsak first.
17:11.29nettiethey have a field for an external ip and port
17:11.37redondosk, great.
17:11.43Dovidkk
17:11.45nettiebut unfortunately our is dynamic:(
17:12.01rushowrfourcheeze: thanks for the help mate. I'm gonna see what I can do
17:12.03brettnemnettie: what the heck are you doing?
17:12.24nettiebrettnem: I have a remote asterisk box in a colo
17:12.44nettieI have a couple of polycom ip501
17:12.50brettnemannnd?
17:12.53nettieon my lan behind nat
17:12.59nettieI can defiintely call outside
17:13.01brettnemok, so what is the problem?
17:13.02Jon335Any good resources for learning AGI?
17:13.14Hmm-work~google
17:13.16jbotrumour has it, google is a search engine found at http://www.google.com/
17:13.16brettnemtry the wiki
17:13.17brettnem~wiki
17:13.31Dovidnettie: what problem are u having ?
17:13.38nettiebut I can't call/transfer calls inside my lan
17:13.50brettnemhave you read the wiki guides on nat?
17:13.53nettiesure
17:13.58Dovidprob. a config issue
17:14.02brettnemcause it'd work if you did. ;)
17:14.02Dovidcan u get calls in and out ?
17:14.03Hmm-workyou have two phones behind nat?
17:14.07nettiesure
17:14.10*** join/#asterisk point (i=1000@213.27.44.55)
17:14.10nettieI can call out
17:14.16nettieboth phones are behind nat
17:14.17brettnemnettie: qualify=yes
17:14.19fourcheezeis it possible to get a polycom phone working without all the tftp stuff?
17:14.20brettnemnettie: nat=yes
17:14.24nettieasterisk isnt behind nat
17:14.24brettnemnettie: canreinvite=no
17:14.34brettnemfourcheeze: yes
17:14.53brettnemfourcheeze: you can ftp or manually config with web interface or on the keypad
17:14.55blaylockahh, I found it asterboy
17:15.00nettiebrettnem that's how it is configured
17:15.03fourcheezehow do I stop it looking for tftp ?
17:15.05nettieI'll triplecheck
17:15.05asterboydo tell
17:15.13fourcheezebrettnem: ^^
17:15.15jbalcombbrettnem what's the scoop on 'canreinvite=no'?
17:15.17brettnemI bet you have a linksys router.
17:15.20*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
17:15.20*** mode/#asterisk [+o russellb] by ChanServ
17:15.26blaylockfile called <mac addr of phone>-directory.xml
17:15.27brettnemjbalcomb: what do you mean, 'the scoop'?
17:15.40nettiebrettnem I get only one way audio
17:15.41jbalcombbrettnem what does it do and why should it be turned off for a NAT setup?
17:15.48blaylockthe mac addr of phone is the phone which has the buddy thing attached to it
17:15.48nettieand
17:16.14blaylockthen you turn on the <bw> tag
17:16.14fourcheezebrettnem: there doesn't seem to be anywhere to put sip settings into it
17:16.19brettnemnettie: check the SDP and make sure the hosts can communicate with the stated ips.
17:16.27Dovidnettie: one way audio usually has to do with NAT
17:16.36nettieok
17:16.43nettiegimme as ec to rety
17:16.45nettieretry
17:16.50nettieand dump some packets
17:16.56brettnemjbalcomb: canreinvite makes the phones set the calls up direct from phone to phone without passing audio to asterisk.. good for lans, but if you are going lan to lan you will have problems..
17:17.01*** part/#asterisk rushowr (n=rushowr@cpe-65-189-187-159.columbus.res.rr.com)
17:17.13brettnemfourcheeze: have you tried the web interface? it's all in there, I promise.
17:17.15Hmm-workreinvites are a pain in the ass
17:17.23jsharpAnd they don't always work
17:17.26brettnemreally reinvite should be avoided unless you know what you are doing.
17:17.27kardecallanIs there anybody that know to configution MFC/R2 in the Asterisk?
17:17.32asterboyblaylock, so it's not just a directory, its a buddy list thing.
17:17.37asterboyup to 7 anyway
17:17.40fourcheezebrettnem: phone has IP number but browser doesn't get there
17:17.40justinu|Zzzcheck the SDP? who do you think these people are? :P
17:17.48brettnemasterboy: that's right
17:18.01brettnemjustinu: oh come on, they have to learn
17:18.04b00meranybody around who can help me construct a section of my sip.conf for an avaya 4620SW?
17:18.08justinuheh, i'm just teasing
17:18.18justinui tell people look at the SDP, and they look at me like i'm nuts
17:18.24b00merDoes this look right?
17:18.25b00mer[avaya]
17:18.25b00mertype=friend
17:18.25b00merhost=dynamic
17:18.25b00merdtmfmode=inband
17:18.28b00merusername=200
17:18.28brettnemjustinu: they look at you?
17:18.30b00mersecret=200
17:18.38brettnemfourcheeze: sounds hosed to me
17:18.40justinubrettnem: amazingly enough, i deal with people in person too
17:18.41jbalcombbrettnem: that's set in the sip.conf right? any issue if the phone and server disagree? if reinvites are enabled then I couldn't monitor/record calls?
17:18.47brettnemjustinu: <GASP>
17:18.50nettieok
17:18.53nettielet'see
17:19.01brettnemin RL? OMG!
17:19.11jbalcombjustinu wow, you get paid extra for that?
17:19.24justinui suppose i do
17:19.36*** join/#asterisk cyburdine (n=jburdine@208.2.145.2)
17:19.42nettiebrettnem: doublechecked the config ... I can confirm it's as you said and I only get one way audio
17:19.43blaylockasterboy,  yeah i think so
17:19.45jbalcombjustinu is that on your resume under 'soft' skills? ;)
17:19.45Dovidb00mer: please dont post configs in here. use pastebin.com
17:19.49asterboyok
17:19.53brettnemjbalcomb: not sure what you mean if the phone and server disagree.. the phone doesn't care about reinvites.. it just sends audio where it is told
17:19.57nettiebrettnem now I check the SDP
17:19.58justinujbalcomb: it reads "mad people skillz, yo"
17:19.59cyburdinehey gang
17:20.03jbalcombjustinu nice
17:20.29jbalcombbrettnem it can be set on the phone and/or in sip.conf for that extensions entry?
17:20.56brettnemjbalcomb: what is the setting to which you are referring to is on the phone?
17:21.11jbalcombbrettnem for reinvites
17:21.12*** part/#asterisk gr0mit (n=w10277@206.41.25.138)
17:21.24brettnemreinvite is not a setting on the phone.. like i said, it doesn't care.
17:21.35*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
17:21.38cyburdinecan someone point me to a list, or recommend a good turnkey asterisk ip pbx vendor
17:21.51brettnemI build them for a fee. :D
17:22.08Hmm-worksame
17:22.16jsharpPretty much any of the consultants will build you a PBX.
17:22.21Hmm-workgive me money, i turn your key
17:22.34jsharpThat's not my key you're turning.
17:22.38Hmm-workyeeYAYyah
17:22.57nDuffcyburdine: the Bristol Group are reasonably decent.
17:22.57brettnem"PBX"
17:22.59Hmm-workmy mistake
17:23.01jbalcombbrettnem ok, maybe i'm misunderstanding this setting
17:23.02jbalcombForce INVITE:     No       Yes (Always refresh with INVITE instead of UPDATE)
17:23.09brettnemI say we make the "P" in PBX be public
17:23.21brettnemjbalcomb: where are you seeing this?
17:23.23nDuffcyburdine: the system they built us was built on Red Hat 9, which was old then (and is now ancient), so I'd hope they're using something newer by now...
17:23.34jbalcombbrettnem account config page on my GXP-2000
17:23.46nDuffcyburdine: ...but they delivered what they said they would when they said they would, had competant tech support, and interfaced with the phone vendors for us.
17:23.47brettnemeek.. Grandstream.
17:23.51Hmm-workwho the fark cares what underlying OS you're using on small installs
17:23.56jsharpYou need bleeding edge OS on a PBX?
17:24.00jsharpYeah.
17:24.09b00merok... anybody help with getting a phone to register... here is my sip.conf section for the phone: http://pastebin.com/618266
17:24.11nDuffHmm-work: it's bloody near impossible to get some modules to build on RH9.
17:24.11cyburdinethanks nDuff, I'll look them up
17:24.18Hmm-worklike what?
17:24.21nDuffHmm-work: rxfax/txfax
17:24.22cyburdineanyone work with fonality?
17:24.24brettnemjbalcomb: I wouldn't have the phone do anything.. turn it off.
17:24.27russellbHmm-work: try windows
17:24.39Hmm-workrussellb: I accept your dual
17:24.43russellbheh
17:24.54jbalcombHmm-work jsharp could be relevent to non-backwards compatible updates to the PBX software. The older your OS the closer you are to being deprecated.
17:25.03russellbor run macosx and get a T1 card
17:25.03jbalcombbrettnem ok, cool. thanks.
17:25.09brettnemI think it's time for an OS war.. or at least a healthy bashing of mysql for no reason
17:25.12russellbpoint is that it matteres :)
17:25.21russellbbrettnem: lol
17:25.21jsharpBleh mysql.  MSQL
17:25.21Hmm-workmy OS is bigger than yours!
17:25.23cyburdinethey seem to have a much more robust control panel than AMP and that's what we're looking for
17:25.26jbalcombbrettnem MySQL 5 should still be BETA
17:25.30Dovidb00mer: what comes up in the CLI
17:25.34nDuffbrettnem: why for *no* reason? There are plenty of reasons... some of them just happen to be based on historical behaviour rather than present product status.
17:25.39nettiebrettnem: when I call the pap2 from the polycom the pap2 can heard the polycom but the polycom cant hear the pap2. when I call the polycom from the pap2 no one hear eachother.
17:25.40b00merDovid: you think you might have some insight?
17:25.44brettnemah ha.. there we go
17:25.49Doviddont know it too well
17:25.51Dovidlook at the wiki
17:26.04jbalcombDebain is great but Red Hat ES 4 is better
17:26.14jsharpCan I run Asterisk on a Windows server and have it talk to my MS-SQL server?
17:26.15Hmm-workI use debian for most installs
17:26.16russellbI've never heard of Debain
17:26.18brettnemnettie: sounds pretty messed up..
17:26.21b00merDovid: Mar 23 07:25:30 NOTICE[6804]: Registration from '200 <sip:200@10.1.1.60>' failed for '10.1.1.230'
17:26.34Dovidhmm
17:26.42nDuffbrettnem: there wouldn't be nearly so many folks with resentment against MySQL if their dev team hadn't been asshats back in the 90s and tried to get the world to believe that relational and transactional integrity (like views and other things they also didn't support) were useless and wasteful and should generally be avoided.
17:26.43jbalcombjsharp yeah, umm.. just uhh.. needs umm. the ODBC driver and maybe cygwin..
17:26.44nettiebrettnem can I use some asterisk debugging to see what could be wrong?
17:26.44Dovidwhat kind of phone ?
17:26.52b00meravaya 4600SW
17:27.00b00mer4620SW
17:27.04jbalcombrussellb ah, well, it's kind of new. bleeding edge if you will.
17:27.11*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
17:27.11russellbnice
17:27.15brettnemnettie: you really should check the SDP.
17:27.20nDuffbrettnem: it's the having been forced to maintain and fix software and databases built by folks who believed that that leads to there being so many people who hate MySQL with a passion, not any serious problems in the modern-day product.
17:27.25Hmm-worklast dance with mary jane, one more time to kill the pain
17:27.26brettnemnettie: is the asterisk server on a public IP
17:27.38nettiebrettnem sure
17:27.52brettnemnDuff: it's as f-d up as PERL is.. it's built for lazy programmers
17:28.15brettnemand if you don't agree with me, you're just in denial.
17:28.15b00merwhere do I issue sip debug command?  how do I get to an asterisk console?
17:28.23brettnemasteisk -r
17:28.24*** join/#asterisk charles___ (n=charles@fw.invosat.com)
17:28.24brettnemsip debug
17:28.26b00merthx
17:28.38*** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F2AA7.dip0.t-ipconnect.de)
17:29.09charles___Hey guys, anybody using MCC ?
17:30.48Kattyhihi
17:31.12*** join/#asterisk MarcoZZZ (n=admin@62.48.121.129)
17:31.16nDuffbrettnem: They have relational integrity. They have transactional integrity. They have views. I still use Oracle and PostgreSQL exclusively, but I don't get into feature wars with MySQL folks anymore because the really hard-hitting arguments don't apply (unless one takes the point that "well yes you have it, but if it hasn't been in released builds for at least a few years it isn't really mature yet").
17:31.33Hmm-worksome things will never change, they just stand there looking backwards half unconcious from the pain
17:31.40b00mercan someone provide me an example sip.conf section for a hardphone?  trying to setup an avaya 4620sw with no luck
17:31.46MarcoZZZhello
17:31.57brettnemnDuff: dosen't sound like you care much. :)
17:31.58Hmm-workhey Katty
17:32.11Kattyheylo.
17:32.20Kattyyou going to cluecon again this year?
17:32.26Hmm-worknot sure, maybe
17:32.32Kattythat'd be swell, if you did.
17:32.44Hmm-worknot if it is in that lame @$$ suburb
17:33.07Kattyhaha
17:33.09Kattyyeah, don't blame you.
17:33.16Kattyhopefully it'll be downtown
17:33.39fourcheezebrettnem: I just did a nmap on the polycom and all its ports are closed
17:33.48b00merIs voip-info.org down for anybody else?
17:33.59MarcoZZZHi, I have 3 VoIP accounts, is it possible to configure asterisk to answer to these 3 accounts with a predefined vocal message? Or do I have to launch multiple instances of asterisk with different sip.conf files?
17:34.00justinuthe polycom takes about 2-3 minutes to load it's webserver up after boot
17:34.35brettnemfourcheeze: all?!
17:34.48KattyHmm-work: but i do hope you come....i could use an escort.
17:34.57nettiebrettnem you want to know if there's the NAT address in the SDP?
17:34.58brettnemMarcoZZZ: no it will listen to as many accounts as you configure.. start reading the wiki
17:35.09brettnemthe private address.. in the c= line
17:35.22nDuffbrettnem: It's been almost 5 years since I last had to fix a POS completely corrupted database created by use of a buggy application built against a database schema with no relational integrity. At that time I cared a lot. Since then... yah, I have different things to care about. Some of them have to do with people who have no clue how to use databases (ie. building queries by concatenating strings), but most don't.
17:35.27nettiebrettnem it's there..
17:35.31Hmm-workWe'll see, bkw_ asked me about a week ago too
17:35.36MarcoZZZbrettnem: can you give me the address of the wiki please?
17:35.47nettieI'll check the pap, maybe it's the pap which has the problem damn.
17:35.56Kattyanthm, dear, do you have any paperwork for cluecon yet?
17:36.05brettnemMarcoZZZ: http://www.voip-info.org
17:36.14brettnemnettie: what's there?
17:36.23bkw_Katty, very soon dear ;)
17:36.25MarcoZZZbrettnem: that server is down
17:36.34MarcoZZZor it's terribly slow
17:36.34bkw_wrapping up some work on hotel and we'll be ready to roll
17:36.39brettnemnettie: look, it's more complated than that
17:36.43Hmm-workfingers tips have memory they can't forget the curves of your body and when i feel a bit naughty i run it up the flag pole and see
17:36.47Kattybkw_: alright.
17:37.03brettnemnettie: it's ok to get private ips from the phones, but when the request comes from asteisk TO the phon it needs to be public there.
17:37.04KattyHmm-work: hmm..you haven't emailed me any new musics in awhile.
17:37.33nettiebrettnem it's in the Via field
17:37.38KattyHmm-work: i have a new bjork album (homogenic) if you want anything from it
17:37.41nettiein the Via filed I have a private ip
17:37.52nettiebrettnem the private ip of the pap2
17:37.56brettnemthat *might* be ok..
17:37.59MarcoZZZwww.voip-info.org doesn't work
17:38.33Hmm-workKatty: there's a name i haven't heard in awhile
17:38.38brettnemMarcoZZZ: try it now
17:38.46nettiebrettnem I read the debug
17:38.50nettieit's always public there
17:39.01KattyHmm-work: i can tar it up for ya.
17:39.04nettieonly the VIA is private
17:39.12brettnemthat I think is ok
17:39.21fourcheezebrettnem: all 65536 ports are closed
17:39.27nettiebrettnem if I dont answer the call on the pap
17:39.35nettiethe vmbox gets in and I can hear it
17:39.42fourcheezebrettnem: I've tried a factory reset
17:39.44nettieso it must definitely be a nat issue
17:39.53Hmm-workKatty: sure
17:40.03brettnemfourcheeze: seems weird that no ports are active.. is it really getting an IP on the network?
17:40.16KattyHmm-work: have an zeppelin albums?
17:40.25*** join/#asterisk lzhang (n=lewiszha@67.95.13.46)
17:40.25KattyHmm-work: s/an/any
17:40.44jbalcombI think all 192 countries and Taiwan should switch thier date format to YYYYMMDD damnit.
17:40.56justinutelling time in swahili makes a lot more sense
17:40.59brettnemAND Taiwan?
17:41.00justinudawn is hour 0
17:41.01fourcheezebrettnem: I've given it a static IP
17:41.07Hmm-workKatty: yeah a few
17:41.08brettnemand you can ping it?
17:41.12fourcheezeyes
17:41.18brettnemweird
17:41.22KattyHmm-work: which ones?
17:41.25jbalcombbrettnem CIA factbook lists countries 1 - 192 and then the last line is Taiwan with no number
17:41.28brettnemI really don't know.
17:41.36brettnemjbalcomb: haha!
17:41.38fourcheezebrettnem: would it help to get it to download a configuration?
17:41.44KattyHmm-work: i brough a static-x album into work today too.
17:41.48brettnemfourcheeze: I don't think so..
17:41.58brettnemhave you tried configuring from the keypad?
17:42.00brettnemwhat software rev?
17:42.11Hmm-workKatty: i have to look at home, my cable ip changed so I can't remote in
17:42.24KattyHmm-work: kay.
17:42.30justinuever heard of dyndns.org? :P
17:42.34*** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131)
17:42.39KattyHmm-work: i'll tar up the bjork album. you want the static-x one? it's machine.
17:42.50jbalcombAlso, it seems rediculous that America and two little countries are the only ones left using the supposed /English system of measurements. England doesn't even use it!
17:42.53Hmm-worki can grab that one of  usenet
17:43.04Fedoracore6hai all
17:43.05nettiebrettnem I can see the private ip also in Call-ID: abfda9b8-7a50d491@172.31.253.5
17:43.09fourcheezebrettnem: rev2.6.6.0032
17:43.12lzhanghey guys, I'm running an asterisk with both Zap lines and VoIP lines... my problem is that on inbound calls from the Zap line, when the caller presses digits in the auto-attendant, asterisk may not pick up all of them, especially if they are pressed slowly enough... but on SIP calls, DTMF is picked up perfectly
17:43.15brettnemnettie: that's ok.
17:43.20brettnemfourcheeze: this is a polycom phone?
17:43.30fourcheezebrettnem: there doesn't seem to be anywhere to configure it
17:43.30KattyHmm-work: just want bjork then?
17:43.35Hmm-worksure
17:43.37justinusounds like a bootloader rev
17:43.39fourcheezebrettnem: it's a 601
17:43.41Hmm-workyou can't gmail it though, 10meg limit
17:43.54brettnemyeah, I think that's a bootloader..
17:43.54russellblzhang: you can try the relaxdtmf option in zapata.conf
17:44.01KattyHmm-work: i'll put it on the server for you to download.
17:44.05fourcheezebrettnem: so it needs a firmware?
17:44.10lzhangrussellb: thanks
17:44.28Hmm-workcool
17:44.37fourcheezejustinu: brettnem: so how do I get it to load ?
17:44.42KattyHmm-work: just get me those zeppelin albums ;)
17:44.51Fedoracore6i try run my update script but have one error like this
17:44.55Fedoracore6pbx.c:2356 __ast_pbx_run: Invalid extension '2', but no rule 'i' in context 'kodsubjek1'
17:44.59justinufourcheeze: does your phone ever say "running application sip.ld"?
17:45.04brettnemfourcheeze: try setting up a tftp server with a fresh copy of the bootloader and sip firmware... I'd try that..
17:45.08justinuwhile it's booting?
17:45.08brettnemjustinu: good thought there..
17:45.10KattyHmm-work: actually, why don't you get on the gmail thing, and i'll send it across that.
17:45.25fourcheezejustinu: not sure, I'm doing this remotely with someone else's hands
17:45.25asterboyWhy would a Polycom phone *not* save changes made via http?
17:45.34justinubut i agree with brett... start with a new bootloader.
17:45.38fourcheezebrettnem: ok, I@ll give that a go in the morning
17:45.38Hmm-worki gotta download it on this pc
17:45.42fourcheezeI've lost the will to live now
17:45.47justinui'd recommend the 3.1.1 bootloader... that'll reformat the phones flash completely.
17:45.52KattyHmm-work: does the gmail.com login thingy not support file transfers?
17:45.54justinuthen go with the latest sip image... 1.6.5, iirc.
17:45.58Kattyspeaking of file
17:45.58fourcheezejustinu: brettnem: ok , thanks for the help
17:46.01Kattyfile[laptop]: file dear!
17:46.03Hmm-workparanoia paranoia everybodies coming to get me
17:46.13asterboyfourcheeze, thats ok, we'll keep you alive as long as possible so we can torture you further.
17:46.23NuggetI always feel like somebody's watching me.
17:46.30asterboywe are
17:46.32KattyNugget: that's because i am
17:46.38*** join/#asterisk ramtha (n=ramtha@p50889673.dip0.t-ipconnect.de)
17:46.40ramthahi
17:47.02asterboyand that tin foil hat your whereing actually helps to amplify the signals we need to disect your mind.
17:47.05Nuggetand I have no privacy (oh oh oh)
17:47.10brettnemI think you are just down the street from me anyway
17:47.22brettnemhaha
17:47.22asterboyso much for my engrish...whereing.
17:47.43brettnemwho needs civil rights? I wasn't using them anyway
17:47.57asterboyAnyone else deal with Polycom *not* saving settings via http?
17:47.59justinufourcheeze: also... a tethereal/tcpdump capture of the phones traffic is very useful in debugging these problems: try a command like: tethereal ether host xx:xx:xx:xx:xx:xx
17:48.04asterboyWhat the heck could that be?
17:48.06ramthamy telco routes calls from pstn to my voip clients over asterisk box. some thing i dont understand: in pstn, there is something like "ii" or "ni" bevor every callpartynumber
17:48.09justinuwhere x's are the MAC obviously.
17:48.11nettieasterboy I do
17:48.15nettiejust got them
17:48.15ramthaii is internation number format
17:48.20ramthaand ni national number format
17:48.47KattyNugget: aww. you ruined my view!
17:48.49justinui've never seen that ramtha
17:48.52asterboynettie, you have Polycom's ignoring setting changes made via http?
17:48.52ramthabut i dont know, how to tell asterisk, to put 00 bevor international format and 0 bevor national format
17:48.55lzhangrussellb: When I set relaxdtmf=yes in zapata.conf, does that mean that DTMF is more likely to be detected or less likely
17:49.05nettieasterboy no, they work perfectly
17:49.07ramthabecause i dont see the "ii" and "ni" in asterisk..
17:49.07Hmm-workwow the quickest way to piss off a tech,  ask a question that is answered in the second sentence of the manpage you have been given
17:49.16justinuheh
17:49.18nettieasterboy no, I just added the line and works
17:49.21ramthaok
17:49.22KattyHmm-work: i probably did that to you a couple times ;)
17:49.39nettieasterboy apart the nat problems I'm dealing with but I dont think are related to the phones
17:49.57asterboyso your settings are being saved no problem via http?
17:50.01nettiesure
17:50.27ramthathe incoming call is from international number like this: 16342XXXXX (the number in original ist 016342XXXXX) from polen the number is displayed 4822XXXXX (original ist 0049XXXXXX)
17:50.36ramtha0048 not 0049
17:50.51justinuthe phone might be trying to save it's configs to a TFTP/FTP/HTTP server, asterboy
17:50.58justinuthey try and do that
17:51.13ramthahow can i figure out if a call is national, pt a 0 before the nummber, if its international put 00 befor the number
17:51.26justinuasterboy: i'm not sure if it'll solve the problem, but can your phone upload it's logs to the provisioning server?
17:51.35asterboyyes
17:51.44justinuhave you looked thru those logs for anything?
17:52.16KattyJunK-Y: you around?
17:52.24russellblzhang: more likely
17:52.34russellbi think ... let me check again
17:53.04russellbyes, that's correct
17:53.40*** join/#asterisk octothorpe_ (n=octothor@unaffiliated/octothorpe)
17:53.51*** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com)
17:53.55Hmm-workis it bad that I will go out of my way to just copy and paste answers out of the manual when I could just type them out?
17:53.57asterboynothing in the logs
17:54.07lzhangrussellb: it seems like I'm getting the opposite effect, now it seems more difficult to get the DTMF digits to read
17:54.17lzhangit could just be my imagination
17:54.18Jon335I need a unlimited US/Canada long distance provider, preferably without a DID
17:54.29asterboywhat is that key sequence to upload a log file?
17:54.38lzhangI'm going to try with it set to 'np' again
17:57.29blaylockasterboy, page 32
17:57.50lzhangI'm reading up on relaxdtmf, and I'm seeing that it should be more likely for the numbers to be detected... but it  still seems to be the opposite behavior of what I'm expecting
17:58.00lzhangare there any other options I can investigate?
17:59.11asterboyah, found this: As an additional diagnostic tool, both log files can be uploaded on demand to the boot server by pressing and holding the four arrow keys until a confirmation tone is heard or for about three seconds.
17:59.37fuzzbawlrecommendations? I need a few asterisk-friendly SIP hardware phone that can do: speakerphone, call hold/transfer, callerid, and cheaply. ADSI is a plus
17:59.55fuzzbawland by cheaply I'm wanting around $100
18:00.15asterboygrandstream
18:00.50justinusip phones don't do ADSI
18:00.59justinu~adsi
18:01.01jbotadsi is, like, Active Directory Service Interface
18:01.05justinulike wrong
18:01.09*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
18:01.21fuzzbawlSIP phones would do something like xml?
18:02.05justinuyeah
18:02.06*** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com)
18:02.22justinuthe aastra 480i, polycom 601, and cisco 7960 all have xml microbrowsers
18:02.23fuzzbawli feel stupid :P
18:02.38Kattyhi justinu (=
18:02.45justinumorning katty :)
18:02.54NetgeeksHi Katty, Justin
18:02.54RoyK~adsi is also
18:02.56jbot...but adsi is already something else...
18:02.56RoyKAnalog Display Service Interface
18:03.02RoyK~adsi is also Analog Display Service Interface
18:03.04jbotRoyK: okay
18:03.04Kattyallo, Netgeeks
18:03.12RoyK~adsi
18:03.13jbotextra, extra, read all about it, adsi is Active Directory Service Interface.  Analog Display Service Interface
18:03.27justinuthanks roy
18:03.44KattyNetgeeks: i can lift weights again :>>>
18:03.49RoyK~no, adsi is Analog Display Service Interface or perhaps Active Directory Service Interface for the Mickysoft ones.....
18:03.50justinuuh-oh
18:04.00*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
18:04.00fuzzbawlare there any decent voip hardware outfits that won't screw me over like voipsupply did?
18:04.07justnulling2can't connect to cli get "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" and the file is there, what can it be?
18:04.07justinuroyk: say "jbot, no adsi is...."
18:04.19RoyK~no adsi is Analog Display Service Interface or perhaps Active Directory Service Interface for the Mickysoft ones.....
18:04.21jbotRoyK: okay
18:04.27NetgeeksKatty, I didn't know you couldn't lift weights!  what did I miss out on in my coding zeal?
18:04.27RoyKworks with ~ as wlel
18:04.28russellbjustnulling2: asterisk probably isn't running ....
18:04.32RoyK~adsi
18:04.34jboti heard adsi is Analog Display Service Interface or perhaps Active Directory Service Interface for the Mickysoft ones.....
18:04.44KattyNetgeeks: maybe...my lymph nodes have been swollen the better part of 3 weeks
18:05.00NetgeeksThat sounds uncomfortable
18:05.05Kattyit wasn't too bad, really.
18:05.17Kattybut i was too afraid to do much of anything for fear of them getting worse.
18:05.20*** join/#asterisk SwK (n=Silik0nJ@c-24-92-158-154.hsd1.ga.comcast.net)
18:06.02asterboy<PROTECTED>
18:06.05asterboyhttp://pastebin.ca/46716
18:06.10KattySwK: mew.
18:06.22*** join/#asterisk stoffell (n=stoffell@d51A58202.access.telenet.be)
18:06.23justnulling2@russellb but is it
18:06.29NetgeeksWell, it's good to hear you are back in top condition again!
18:06.45asterboynot suer what [SoNcasC] Failed is.
18:07.23asterboy#
18:07.24asterboy0322224431|copy |3|00|UtilCopyC: curl_easy_perform failed: curlRes: 23, respCode 150
18:07.26KattyNetgeeks: not exactly in top condition yet..the right one is still a smidgen swollen.
18:07.27asterboy#
18:07.29asterboy0322224431|copy |3|00|UtilCopyC: curl error: failed writing received data to disk/application.
18:07.42asterboyits like there is a write protect on the disk
18:08.56NetgeeksGoing down on thier own, or did the Doc give you something?
18:09.23KattyNetgeeks: they gave me duricef, an anti-biotic that's a cousin of penecillin and a bit stronger, but it didn't phase it.
18:09.31KattyNetgeeks: i'm guessing it's viral.
18:09.50fuzzbawlbiotic's suck
18:10.40Kattycipro sucks, that's for sure. they tried to me on that first and it made my heart race so fast i had to stop taking it after 2 pills.
18:11.13fuzzbawlcipro doesn't work on too many people anymore. After the big anthrax scare anyway
18:11.49Netgeeksleast you went to the doc!  I was never a fan of docs, but I got graves disease and finally had to go see one... When I finally went I had lost 80 lbs. of which a good portion was muscle mass.  I could barely walk....  some quick meds, and I was all better in 3 months
18:12.04Netgeeksbut now I have to get blood drawn once a month.. ewww
18:12.22justinuviruses are the real enemy
18:13.13justnulling2how do i change location of asterisk.ctl?
18:13.49*** part/#asterisk cyburdine (n=jburdine@208.2.145.2)
18:14.02KattyNetgeeks: graves?
18:14.04nettiebrettnem anything elese I could check?
18:15.40NetgeeksKatty: it's when the immune system attacks the thyroid gland, forcing it to swell and go crazy making thyroid enzymes... when I finally saw the doc, I was about 100 times the normal levels in my blood....
18:15.40justinuasterboy: this line repeated a few times is troubling to me
18:15.40KattyNetgeeks: yikes :< were there no symptoms?
18:15.48justinuasterboy: 0322154457|cfg  |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f200e15b-phone_cfg.zzz
18:15.56NetgeeksThere were lots of symptoms, but they came on over a year and so I kind of ignored them
18:16.11Kattyah.
18:16.23KattyNetgeeks: any little thing wrong with me, and i'm paranoid and freak out..
18:16.46justinuhypochondriac?
18:17.01Kattyjustinu: yeah, mildly
18:17.09Kattyjustinu: had a really bad UTI awhile back that started it.
18:17.12justinuprobably a survival technique
18:17.17NetgeeksI had a real nasty skiing accident in college, so I tend to fault the lingering effects of it for everything... ;P
18:17.23mishehuUTI?
18:17.28mishehuurinary tract?
18:17.28justinu~uti
18:17.30justinu:)
18:17.32Kattymishehu: aye.
18:17.34mishehupissing flames?
18:17.37Kattymishehu: not really
18:17.41NuggetI have to take my cat in for radiation therapy next week to (hopefully) deal with a tumor in his thyroid.
18:17.41justinupissing razor blade
18:17.41Kattymishehu: it was a lot worse...
18:17.47Kattymishehu: almost got into the kidneys
18:17.51mishehuKatty: pissing liquid hot magma?
18:17.52justinuouch
18:17.53asterboyjustinu, the .zzz part is strange.
18:17.54mishehueww
18:17.55Kattymishehu: really really painful.
18:17.58mishehuand ouch.
18:17.59Kattymishehu: almost passed out
18:18.13justinuasterboy: i don't think that's the strange part... but the parse error is strange to me... can you verify your phone1.cfg file is ok?
18:18.16Kattyand it hit at 4am..and of course no one was around at /all/
18:18.20Kattyreally freaked me out.
18:18.29mishehuKatty: did you go to the er for that?
18:18.38Kattymishehu: no, i did
18:18.41Kattydidn't
18:18.45*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
18:19.05Jon335reccomendations? I need a unlimited US/Canada long distance provider, preferably without a DID
18:19.28Kattymishehu: my mamma drove down here and took me to the doctor.
18:19.34docelm0plainvoip.com
18:19.38Kattymishehu: after about 6 hours or so the pain went down a good bit.
18:20.30docelm0www.plainvoip.com US Domestic Termination $.0009USD/Minute
18:20.44Netgeeksthat cheap?
18:20.53asterboyhella cheap
18:21.01NetgeeksI'm signing up now
18:21.02attais it possiple to add a stunserver into a register => .... function for a extern SIP provider??
18:21.05justinulol
18:21.14justinulet us knnow how the quality is, netgeeks
18:21.53*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
18:21.57NetgeeksThats 5.4 cents per hour in the US.... nice  ;)
18:21.57KattyAriel_: hey hun (=
18:22.09asterboyjustinu, I don't have a phone1.cfg.
18:22.10KattyNetgeeks: you have any Jamiroquai albums?
18:22.13Ariel_Katty, hello hope your doing well today
18:22.18justinuasterboy: then what do you have?
18:22.19asterboyI can put it in...but I thought the phone used the mac
18:22.21Ariel_hello everyone
18:22.24KattyAriel_: yup, pretty good...got back to my weight lifting last night.
18:22.30asterboymac-phone.cfg
18:22.37justinuok, can you verify that's ok?
18:22.38Ariel_Katty, great to hear it.
18:22.43justinuall xml tags are closed, etc.
18:22.48NetgeeksKatty, nope, doesn't look like I do
18:22.51docelm0What can I say guy's..  I try..  :)
18:22.54KattyNetgeeks: okies. thanks for looking.
18:22.57asterboya girl wight lifting...must have some steroids raging through ya.
18:23.07Kattyasterboy: uh, not quite.
18:23.12asterboywatch you don't go through to many razors.
18:23.15docelm0Katty, what can you bench?
18:23.22Kattyabout 60, now
18:23.33Kattybut i can get up to 80, if i keep it up
18:23.38asterboya guy with a 200lb beer gut.
18:24.01docelm0wow..  I curl that much..
18:24.05docelm0:)
18:24.05Katty:<
18:24.10Kattyi curl 10 right now
18:24.18Kattybut i'm not that big either
18:24.29docelm0What 5'6" ish?
18:24.35Katty5'2"
18:24.44docelm0wholey shitzu..
18:24.48docelm0itty bitty
18:24.55docelm0I would be a giant next to you
18:25.00Kattyi'm not short! i'm a ......good height.
18:25.08docelm0Your vertically challenged
18:25.09Kattyi barely reach Hmm-work's shoulder.
18:25.11Netgeeksmy older sister stands a whopping 4'11'
18:25.19*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
18:25.20charles___Esgabon, guen, gagua !
18:25.21*** join/#asterisk subdolus (n=subby@dsl-202-72-158-93.wa.westnet.com.au)
18:25.22Kattylike /just/ barely.
18:25.22asterboyjustinu, the phone1.cfg is the same as mac-phone.cfg
18:25.31docelm0Im 6'6"
18:25.44KattyHmm-work: and that album is in your gmail too, just so you know.
18:25.47CunningPikeIs Windows Sound Recorder retarded?
18:25.47docelm0And there are a few in here that can verify this..  :)
18:25.51Netgeeksyou are a giant next to almost anyone, Docelm0
18:25.53docelm0CunningPike, yes
18:25.56CunningPikelol
18:25.59Kattydocelm0: you'd probably freak me out.
18:26.05docelm0Netgeeks, ever been to astricon?
18:26.07Kattyanthm does :<
18:26.15docelm0Katty, EVERYTHING freaks you out
18:26.21Netgeeksdocelm0: nope, never been to astricon
18:26.22Kattyoh shush.
18:26.24CunningPikeRegardless of the settings I choose, I get format_wav.c:169 check_header: Unexpected freqency 22050
18:26.29docelm0Gotta come this year..
18:26.38docelm0Im gonna try and talk steve into letting me speak
18:26.39docelm0:)
18:26.50Kattydocelm0: you going to go to cluecon?
18:26.58docelm0Nope..  Not really a Dev
18:27.02Kattykay
18:27.11docelm0At least I try not to unless I REALLY half to
18:27.28Kattymy company like sending me to the closer ones.
18:27.43justinumy cat sleeps in the funniest positions on my amplifier
18:27.52justinusleeps with it's eyes open too sometimes
18:27.55Kattyand i hate going anywhere alone....especially big cities.
18:28.07justinubig cities rule
18:28.11Kattyhopefully someone will meet me at the amtrak station and help me get to cluecon so i don't have to go there alone.
18:28.12*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
18:28.19justinui'm sure someone will help you out
18:28.19Kattytwisted could probably be bribed.
18:28.33docelm0Ya.. beer
18:28.34Kattyprobably.
18:28.42Kattydocelm0: twisted won't drink around me
18:28.43docelm0or ass
18:28.54digimeanyone know where to download a trial of Pocket PC SIP Softphone v2.2
18:28.56Netgeekswhere/when is the next astricon?
18:28.57*** join/#asterisk Nodren (n=nodren@64.193.95.10)
18:28.59docelm0really?   We drank it up last year at Lucky Strike
18:29.04docelm0Dallas, TX October
18:29.05Kattydocelm0: really really.
18:29.08Kattydocelm0: he won't do it
18:29.12Jon335Can I put a FQDN as externip in sip.conf?
18:29.18justinunot really
18:29.18Kattydocelm0: drunk males freak me out too
18:29.22*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
18:29.23docelm0Jon335, no
18:29.27Kattyhey Qwell[] (=
18:29.32Netgeekslol
18:29.34Qwell[]hi
18:29.38docelm0Katty, told you EVERYTHING freaks you out
18:29.43Kattydocelm0: exactly.
18:29.45justinuheh
18:29.55Kattyi'm just a skiddish person :P
18:29.59Jon335So what should I do about having a dynamic IP?
18:30.05docelm0Katty, Still one of the Asterisk Hotties tho..  :P
18:30.05*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)
18:30.05asterboyeven one-eyed monsters?
18:30.06justinuboo!
18:30.10Kattybah
18:30.11docelm0Jon335, PRAY!
18:30.16*** part/#asterisk point (i=1000@213.27.44.55)
18:30.30Jon335docelm0: lol
18:30.34Kattywell, maybe amongst geeks, docelm0
18:30.40Kattybut i don't nearly compare to most girls.
18:30.51*** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
18:30.52Ariel_Jon335, yes you can
18:30.54b00meranyone have any luck with a Avaya 4620IP?
18:31.02docelm0Considering our selection at Sausage Fest 04 and 05 yes..  Your a hottie..
18:31.27docelm0There were only a handfull of cuties there..
18:31.42Kattycause, what, there were 3 girls there?
18:31.45docelm0But techie girls freak me out..   Its weired talking to someone know knows what I Do
18:31.49asterboyjustinu, I get this when I make a change via http to enable the headset memory and then upload my log files to view the results:
18:31.50Jon335Ariel_: Two say no, one says yes who's right?
18:31.52docelm0EXACTLY!
18:31.52asterboy0323112618|cfg  |5|00|Edit|Simple|error 0x0 when locking (setting up.headsetMode="1")
18:31.58Kattyyou know, that makes me really sad.
18:32.04Kattyat cluecon last year, hardly /no one/ talked to me
18:32.06docelm0What?
18:32.07Ariel_Jon335, I use in my setup a FQDN
18:32.19asterboyThe phone does not reset.
18:32.23Katty3 or 4 people said hi......and just wandered off.
18:32.32Jon335Ariel_: I'll try it
18:32.52justinuasterboy: have you verified that your XML is valid?
18:32.57asterboyput in some breast enhancments, and lots of guys will talk to you. :P
18:33.00Kattygirls don't bite, donchaknow.
18:33.02docelm0Well Katty come to Dallas..  We go out and kick round a few suds..  But then again twisted told me my personality would probably make you uncomfortable
18:33.10nettiebrettnem
18:33.11docelm0Katty, Nope they SUCK!
18:33.12nettiebrettnem I fixed it
18:33.15fuzzbawlKatty does Dallas?
18:33.19fuzzbawl:P
18:33.22nettiebrettnem I had ALG enabled on the router
18:33.22asterboyis there a utility to verify it?
18:33.23Kattyfuzzbawl: i've never been to dallas.
18:33.28Kattydocelm0: and twisted is probably right.
18:33.29justinuasterboy: yeah, find an XML editor
18:33.35asterboyvim them
18:33.42nettiebrettnem I disabled it .. works perfectly now. :)
18:33.44justinusomething that verifies the tags for you
18:34.02docelm0Ask damin or twisted about what I did last year..   haha
18:34.02Kattyi don't trust males on liquor.
18:34.08justinukatty: probably a good instinct
18:34.14docelm0no Damin is better ..   He passed out under a piano in the hotel
18:34.22docelm0I at least made it to my room
18:34.25b00merwhat do you use for a username if the phone only asks for an extension and password?
18:34.29asterboyI trust women on liquor.
18:34.39docelm0asterboy, YES!   HELL YA!
18:34.42Kattyheh, i'd probably start crying if i was drunk.
18:34.54docelm0Katty, your too reserved..  Need to get out more..
18:35.01Qwell[]docelm0: yeah, Damin wins
18:35.06Qwell[]docelm0: hands down :P
18:35.09Kattydocelm0: now i don't doubt that in the least.
18:35.12asterboyor worse fall asleep
18:35.14docelm0Qwell, were you there?   Ya you were!..   hahaha
18:35.33b00merI would start crying if I got my phone to work with asterisk... but I must be on the wrong channel
18:35.33asterboylet yourself be taken advantage of.
18:35.36Kattydocelm0: my social circle here consists of 5 people.
18:35.47Kattydocelm0: and their 3 cats.
18:35.47docelm0Katty, Need I say more?
18:35.48Netgeeksb00mer: lol
18:36.00asterboyand a bunch of geeks on an asterisk channel.
18:36.02Kattydocelm0: well sorry if i'm just not miss social butterfly :P
18:36.07docelm0back to programming T1's on a cisco..
18:36.12asterboyget drunk more.
18:36.17docelm0Katty, :{P
18:36.18*** join/#asterisk xterminus (n=cmauch@00104bc8bd59.click-network.com)
18:36.25Kattyasterboy: i don't drink.
18:36.31asterboythen get high
18:36.35Kattyuhh, no
18:36.38asterboyacid?
18:36.46asterboymushrooms?
18:36.48[TK]D-FenderKatty: mew.
18:36.54Kattymister fender!
18:37.06Kattyasterboy: no, and no (=
18:37.07justinu~asterboy
18:37.09jboti guess asterboy is a weed smoker
18:37.17asterboychocolate is a drug.
18:37.22justinua good one too
18:37.29[TK]D-FenderKatty: You don't know how much I need it.... I'm 2 days away from moving, hardly packed and ^%$#@$ beyond belief.
18:37.32justinucheap, readily available, satisfying
18:37.37asterboycaffine is the smartest molecule.
18:37.41Katty[TK]D-Fender: what's up?
18:37.55asterboya good hump is satisfying.
18:38.04Kattyasterboy: yeah i don't do that either.
18:38.06*** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
18:38.21asterboyshit, you might as well dig a grave.
18:38.29Netgeeksoh, this is impressive... I just got an email from APC that is titled  "What to do when your Smart-UPS says 'Replace Battery'".  Pardon me for my ignorance, but I thought that perhaps that means, I should replace my UPS battery..  but maybe I need an email to tell me that
18:38.32asterboyyour already dead.
18:38.33Kattyi'll get around to it eventually, asterboy
18:38.46Kattysure as hell not wasting it on anybody
18:38.53[TK]D-FenderKatty : Torn between the split and thoughts of whether I'll be going back.  I still love her, but don't know if I can sacrifice what I'm leaving in hopes of to go back.  I need a therapist...
18:39.10Katty[TK]D-Fender: ugah, sounds complicated and messy.
18:39.43asterboydr. phil, thearpy for the masses.
18:39.47[TK]D-FenderKatty : Yup, and every outcome makes me feel like shit even though she practically evangelizes me...
18:40.05Katty[TK]D-Fender: hang in there.
18:40.06asterboyyou need to expand your social network.
18:40.17Katty[TK]D-Fender: if you wanna go see a therapist, do it....no shame in it.
18:40.19asterboyhave more than 1 girlfriend.
18:40.23asterboysay 2 or 3 at least.
18:40.32Kattyi have several girlfriends.
18:40.35asterboywhere one failes the others will pickup.
18:40.35[TK]D-FenderDr. Phil has a point.... unfortunately most of the time he makes you want to beat him senseless with a big stick before you get to that...
18:40.37tmccraryI have a PRI card and a TDM 4-port FXO card in the same box. Can I just set channels in my config or do I need to determine which channels got assigned before hand?
18:40.50tmccraryThey're both Digium equipment
18:40.50robl^Netgeeks: at least the email wasn't something like "What do you do when your spouse asks if it could be more satisfying?"
18:40.53asterboylol , ya dr. phil is hilarious
18:40.55fourcheeze[TK]D-Fender: get some children instead
18:41.02Kattyugah, children.
18:41.04Kattystick with cats.
18:41.05justinu[TK]D-Fender: women problems? :(
18:41.11Netgeekscats >> children
18:41.15Kattymister fender needs lots of hugging.
18:41.18fourcheezechildren are great
18:41.27asterboychildren rock!
18:41.28fourcheezebut I won't go on about how great
18:41.36Kattychildren make me want to pull my hair out
18:41.36fourcheezebecause I know some people do that with cats
18:41.36Netgeekswhy would I need children when I still am one?
18:41.38fourcheezeand it sucks
18:41.48asterboyuntil my little guy dump a bunch of screws inm y * server
18:42.03GoRKtmccray: the channels will be assigned in the order in which the modules are loaded in your case
18:42.07Netgeekschildren = responsibility = yuck
18:42.09fuzzbawlare sipura phones any decent?
18:42.09asterboyservers don't run well with a bunch of loose metal on the motherboard.
18:42.15Kattytoo many assholes out there that would just up and leave if they found out you were expecting too
18:42.33[TK]D-Fenderfourcheeze : Thats a good part of why I'm leaving her.
18:42.38robl^hehehe.  children are great from birth til about 18 months..  then they scare me
18:42.41fourcheezeshe wants kids?
18:42.42asterboyyep that is nature.
18:42.42[TK]D-Fenderjustinu : The only kind....
18:42.50GoRKtmccray: if you load the tdm module first it will be chans 1-4 if you load the t1 card first it will be channels 1-24; the next loaded module will take the next available channels
18:42.54KattyNetgeeks: i'll second that.
18:43.08KattyNetgeeks: at least cats can entertain themselves all day long usually
18:43.08[TK]D-Fenderfourcheeze : no, *I* do.  She has one and is impaired against more.
18:43.15fourcheezeahh ok
18:43.16tmccraryah, thanks Gork
18:43.17GoRKtmccray: if you had multiple cards served by the same driver it would be trickier, but in your case it will be the order of the modules
18:43.24fourcheeze[TK]D-Fender: well that's difficult
18:43.30fourcheezeyou have my sympathy
18:43.34asterboyand make a mess of the litter box.
18:43.37tmccraryMy distro is autoloading the modules, so I guess I'll just go by dmesg
18:43.39justinui found a woman who helps me, doesn't cause me grief
18:43.41asterboysmells like a grow op.
18:43.43fourcheezewhich you probably don'tneed
18:43.45asterboyammonia
18:43.47asterboyewwww
18:43.49NetgeeksKatty: yep, I've got two cats, and if it wasn't for cleaning the litter box, they could go weeks without human interaction
18:43.52GoRKtmccray: if you just want to, set the modules up to auto load and then just see which card they are assigned to with ztcfg
18:43.54[TK]D-Fenderfourcheeze : Wish there was a conversion rate to something I could use... it doesn't go very far here...
18:44.02KattyNetgeeks: small price to pay for having cats, in my opinion
18:44.10[TK]D-FenderKatty : Oh... and I'm leaving her our cat.
18:44.15KattyNetgeeks: they're such lovable creatures.
18:44.23fourcheeze[TK]D-Fender: one of the few things I've learned so far is to work out your worst case scenario
18:44.28Katty[TK]D-Fender: :<
18:44.29fourcheezefind out what that really is
18:44.36b00merDoes anybody know how I login to asterisk from a phone which has extension and password... no mention of username?
18:44.56fourcheezeb00mer: what kind of phone?
18:45.00b00merAvaya
18:45.09Kattyi've never heard of an Avaya before.
18:45.09fourcheezeis it a sip phone?
18:45.15b00meryes
18:45.19fourcheezeAvayas are normally not sip
18:45.20b00meravaya 4620IP
18:45.21justinuavaya used to be lucent, or at&t
18:45.24Kattybut i don't work with them much
18:45.29b00merits running sip
18:45.30tmccraryIf I use ztcfg to determine the channels, won't it just tell me what my config says to do (versus what the kernel wants to do)?
18:45.50Katty[TK]D-Fender: oh! good news...my company is moving to a bigger building.
18:46.00Katty[TK]D-Fender: and i get an office! with a door!
18:46.02robl^Avaya used to be "Lucent"  which used to be "AT&T"..  i.e  Partner or Merlin phones
18:46.10Katty[TK]D-Fender: rather than my shiny purple cubicle.
18:46.17fuzzbawlbrb
18:46.23b00merMaybe all you can help with this question:  If I was going to setup asterisk for 100 person office.... what would be my phone of choice... cavate... all the features need to work
18:46.35justinupolycom
18:46.37jsharpWhat features do you need?
18:46.39Katty[TK]D-Fender: plus we'll have a /real/ server room :>>>
18:46.42Netgeeksb00mer: polycom or cicso
18:46.42justinuip301/501/601 depending on price/needs
18:46.46b00mernice... well made phones
18:46.49Kattyspeaking of servers.
18:46.49Nodrenhey justinu
18:46.55Kattywhere's a good place to get a nice 4 post rack?
18:47.00b00merspeaker phones... and 2-4 lines per phone
18:47.10fuzzbawlKatty, closed rack or open?
18:47.10fourcheezeb00mer: not avaya
18:47.13fourcheeze:-)
18:47.14b00merKatty: Wrightline
18:47.14Kattyfuzzbawl: open
18:47.18justinuspeakerphone... then you want the polycom 501
18:47.21b00merfourcheeze: got that
18:47.30fourcheezeif you want to find out what it's doing
18:47.36fourcheezedebug the sip connection
18:47.36robl^polycom ip501, Snom 320/360 all work well
18:47.37fuzzbawlKatty: MCM electronics usually can ship decent racks for about $150
18:47.45Kattyfuzzbawl: do they have a website?
18:47.50justinusnom 360... blah i wouldn't consider that over a 501
18:47.53fuzzbawlKatty: mcmelectronics.com
18:47.55Kattyk
18:48.05fourcheezesnoms are good if you want lots of buttons
18:48.10fuzzbawlbrb
18:48.17fourcheezepolycoms seem to be the most robust
18:48.31fourcheezeand if you're deploying 100 then polycoms are also good for that I'm told
18:48.34jsharpI like the polycoms and Cisco 7940s we've got.
18:48.40asterboyMy polycom seems to have a locking problem.
18:48.42asterboyEdit|Simple|error 0x0 when locking
18:48.45b00merthe issue I have is that the users are not techie... will not appreciate the fact that is voip
18:48.56fourcheezethat's ok, they don't need to know
18:48.58jsharpThey don't need to know.
18:49.00b00merthey only care about the look / feel / quality of the phone / features
18:49.06fourcheezethat's not an issue
18:49.14jsharpJust make it so they get ringy-dingy dialtone and they'll be happy.
18:49.14fourcheezethat's just called a (l)user
18:49.28b00merif the message light doesn't work... I will be strung up
18:49.43justinuMWI works fine on polycom
18:49.47b00merok
18:49.57b00mercool... off to buy some samples... this Avaya is a POS
18:50.02jbalcombwhat is the default password for admin on the polycom IP 501?
18:50.05[TK]D-FenderKatty : In the move I pray to god they ditch that stupid channel bank / analog "solution" you're running on.
18:50.22[TK]D-Fenderjbalcomb : 456
18:50.28Katty[TK]D-Fender: doubtful :/
18:50.32tmccraryin my dmesg output, my TDM card shows up first and PRI second, however when I configure with ztcfg, if my TDM card is set to channels 1-4, ztcfg ssegfaults at the end of the command
18:50.37Kattyoh
18:50.38tmccraryis that normal operation?
18:50.43[TK]D-Fenderjbalcomb : Sorry I didn't get back to you.  I'm royally screwed this week.
18:50.50fourcheezeb00mer: what's your budget?
18:51.00GoRKwell my telco just crashed their entire infrastructure it seems
18:51.06asterboyvoip-info is back up.
18:51.14Katty[TK]D-Fender: i'm kinda scared to use anything /but/ analog cards.
18:51.14asterboyno info on Polycom locking setting though.
18:51.16tmccraryit's been up for me all day
18:51.18asterboyAnyone?
18:51.18*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
18:51.21Katty[TK]D-Fender: mostly cause i don't know what i'm doing.
18:51.41GoRKpolycom locking settings? cant you change both passwords and disable web config?
18:51.55[TK]D-FenderKatty : Remember "move" is supposed to rhyme with "improve".  Cut costs, improve efficiency / reliability, add functionality. 2/3 guaranteed, and on resale of the CB should be "no loss"
18:52.06jbalcomb[TK]D-Fender thanks
18:52.29Katty[TK]D-Fender: the channel bank is provided by our isp
18:52.37Katty[TK]D-Fender: the isp which i /hate/ by the way
18:52.52malverian[work]There's a LUG in Gainesville tonight that is about Asterisk.. wonder if I should go :-P
18:53.00[TK]D-FenderKatty : Like [av]bani always says "Buy Sangoma and Polycom gear and get free 24/7 support from [TK]D-Fender!" ;)
18:53.09Katty:P
18:53.10develi haven't just tried it, but in extensions.conf, can you 'include => ${EXTEN}'?
18:53.31Katty[TK]D-Fender: i try not to bug people unless i really don't even know where to start
18:53.32Netgeeksdevel: ??
18:53.39KattyiDunno: bouncer.
18:53.39*** part/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net)
18:53.42GoRKdevel: you could theoretcially do that only with global variables; however you really should be using a macro
18:53.51malverian[work]"Mike Crown from VOIP Connection, Bill Merriam, and Clinton Collins" are the keynote speakers. Never heard of them.
18:53.52GoRKdevel: it wont work with EXTEN, no
18:54.02*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
18:54.05GoRKdevel: and with a global it is kind of confusing at that, too
18:54.05iDunnoKatty: well, sometimes it has to be done, no? :)
18:54.12KattyiDunno: obviously.
18:54.21develcool.  ok, thanks, GoRK
18:54.31KattyiDunno: i just like hurling accusitions and such.
18:54.48justinukatty: analog is a big headache
18:54.56jbalcomb[TK]D-Fender ok, just get the info as soon as you can so I can submit the PO. It will take a week or two after that.
18:55.06iDunnoKatty: ahh - well, fair enough.
18:55.18GoRKdevel: or maybe even exten => _X.,1,Goto(${EXTEN},s,1)
18:55.19NetgeeksI fear any jobs I get that involve analog cards
18:55.28*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
18:55.37develthat's the stuff, GoRK
18:55.44develi hadn't thought of that.
18:56.15GoRKdevel: that's *very* close to what a macro does though, you really ought to investigate macros
18:56.27jsharpWhat's wrong with analog?
18:56.34asterboyGoRK, I want the web config up, but changes I make are not being saved.
18:56.51asterboyGetting this in my logs: Edit|Simple|error 0x0 when locking
18:56.52develyeah, i've got a stack of macros, i just want to be able to catch special cases (i.e. if the context didn't exist, it would just keep going)
18:57.16Netgeeksanalog is finicky....
18:57.40justinuanalog signalling sucks
18:57.52justinudisco supervision is a pita
18:57.57justinuno answer sup
18:58.06GoRKasterboy: eh? Hmm have not ever seen that one; sorry. i do not use the web config at all though. are you using a boot server for provisioning?
18:58.13*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
18:59.03Axel69hi, i have a analog gateway with 8 lines i want to configure it to the asterisk to provide outside calls but the gateway is with no public ip in another lan
18:59.03[TK]D-FenderKatty : For you : free answers here :)
18:59.10GoRKasterboy: if you are just configuring the phones one by one and not using a provisioning server, you might try just resetting to defaults then reconfiguring them; or reload the firmware to reformat the flash
18:59.10Axel69can anyone help me with that
19:00.08Ariel_Axel69, waht type of gateway?
19:00.14malverian[work]mog_work, Alright.. I'm finally going to continue working on app_sphinx ;)
19:00.26justinuasterboy: I recommend formatting your phone's filesystem and starting over
19:00.34justinui've had the polycoms start to act all funky, and that fixed em
19:00.40Axel69is a quintum
19:01.00jbalcomb[TK]D-Fender Do I have to do something to be able to access the web interface on the polycom IP 501 phone?
19:01.12mog_workwoohoo!!!!!!!!!!!!!!!!!!
19:01.12Axel69i try it as a extension but only can send 1 call
19:01.27justinujbalcomb: wait a few minutes after the phone boots
19:01.32Katty[TK]D-Fender: yay :>
19:01.39jsharpAxel69:  What happens if you send more than 1 call?
19:01.45*** join/#asterisk yuta-vcnet (n=asdf@82.71.50.245)
19:01.46Ariel_Axel69, you might have to call quintum. But is it an sip or h323 gateway?
19:01.58*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
19:02.38Axel69is a sip
19:02.49[TK]D-Fenderjbalcomb : I highly recommend getting a web browser....
19:03.02Axel69well realy is both, you can program whatever you need
19:03.14jbalcomb[TK]D-Fender what does that mean? I have a web browser.
19:03.19Ariel_Axel69, sip will work better
19:03.32[TK]D-Fenderjbalcomb : Thats what it takes to access the web interface :)
19:03.41NetgeeksFender was being Sarcastic...
19:03.43malverian[work]jlewis, I'll be damned.. Jon Lewis from Atlantic.Net
19:03.48[av]bani[TK]D-Fender: can i poke you about a polycom bug?
19:03.49[TK]D-Fender*I* never
19:03.49Ariel_but your going to have to setup it up to forward calls to your asterisk boxes ip address and then acept inbound calls to any port from your IP address
19:04.06[TK]D-Fender[av]bani : Hasn't stopped you yet, has it? ;)
19:04.09[av]bani:)
19:04.18jbalcomb[TK]D-Fender oh, you are the funny one. ;)
19:04.30Axel69yes
19:04.33[av]bani[TK]D-Fender: Dial(SIP/4000) works fine, but if SIP/4000 is dialed from queue, the polycom keeps ringing after caller has hung up
19:04.35[TK]D-Fenderjbalcomb : Should be enabled by default
19:04.48[av]bani[TK]D-Fender: other phones (grandstream, cisco) dont have the problem -- only the ip601 does
19:04.57justinuis it getting a cancel?
19:04.58Axel69the problem is that i have to configure it not as a trunk...because it doesn't have public ip
19:05.04[TK]D-Fender[av]bani : news to me... otherwise my CSR's would be after me with pitchforks & torches again...
19:05.07[av]banijustinu: yes, but apparently it's not acking it
19:05.10jbalcomb[TK]D-Fender whats the username and password for the web interface?
19:05.21justinui wonder what it doesn't like about said cancel
19:05.22[av]banijustinu: but the sip debug peer 4000 doesnt show the cancel being any different than Dial()
19:05.25[TK]D-Fender[av]bani : 601 only?  Thank God I was quick to get my 600's :)
19:05.31justinuit completely ignores the cancel?
19:05.35[av]banijustinu: apparently
19:05.36[TK]D-Fenderjbalcomb : Polycom/456
19:05.38justinuwow
19:05.45[av]banijustinu: and i can't see anything obviously different
19:05.48[av]banii have two dumps
19:05.48justinuthat makes little sense
19:05.49jsharpAxel69:  Why does it need a public IP?  Is it behind NAT as seen from the Asterisk box?
19:05.52[av]baniif you care to look
19:06.01justinui'll be happy to look when i've finished hacking up manager.c
19:06.05[av]baniif you're in a position to tell :)
19:06.06[TK]D-Fenderjbalcomb : But as with everyone I highly advise against wasting time in there...
19:06.12Axel69yes
19:06.29jbalcomb[TK]D-Fender why is that? How else shall I configure my phone?
19:06.30Axel69but how can i program it as a trunk?
19:06.37SkalTuraw00t, asterisk is easy
19:06.44Netgeeks[av]bani does the cancel have the same CSeq: number as the invite?
19:07.13SkalTuratried my first time yesterday --> no hassles i got 2 soft phones put up, and convo started between then, now linked 2 asterisk boxes and routed calls between them without probs :)
19:07.15[av]banigoing to look in a sec when i find the logs again
19:07.20*** join/#asterisk X-Gen (n=x-gen@dsl-145-197-193.telkomadsl.co.za)
19:07.21*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
19:07.24jsharpProgram what as the trunk?  Your quintum or Asterisk?
19:07.47[TK]D-Fenderjbalcomb : Through the provisioning XML files as I'm sure I've mentioned more times than most would have cared for.
19:08.13[TK]D-Fenderjbalcomb : The XML configs give you the real power ove the phones.
19:08.15jbalcomb[TK]D-Fender yeah, you can write me up one of those too while you're at it. ;)
19:08.18[TK]D-Fenderover*
19:08.19Axel69the quintum
19:08.34[TK]D-Fenderjbalcomb : I'll add it to my billable hours on your PO ;)
19:08.37justinujbalcomb: just use the sample
19:08.41Fedoracore6hai all i doing ar delete function but, what happen in my data base, the fill still did no delete ....its my code wrong
19:08.42Fedoracore6http://pastebin.com/618460
19:09.00Axel69the asterisk has al the extensions.....i will go outside calls from the asterisk to the quintum
19:09.01Fedoracore6or some thind i must add in my code
19:09.35[av]baniNetgeeks: yep the cancel has the same cseq
19:09.41*** join/#asterisk terrapen (n=cjs@166.70.135.60)
19:09.45asterboy<PROTECTED>
19:09.54jsharpWill you be going outside calls into the quintum back into Asterisk?
19:09.55Ariel_Axel69, your going to have to setup an account with dyndns and use an url to access your gateway
19:10.00Netgeeks[av]bani: okay, that easy answer is out the door
19:10.01asterboy<PROTECTED>
19:10.52Axel69uppps...how do i i do it? you have a tutorial for that?
19:12.18justinubani: is the contact in the cancel ok?
19:12.28justinuphone isn't trying to send an ack or anything to someone else?
19:12.31SkalTuraWhat am i missing, as asterisk does not impose the challenge i was told it would?
19:12.34*** join/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com)
19:12.37jbalcomb[TK]D-Fender yeah, no PO till I get your info so umm.. your chickens ain't near enough to hatchin'
19:12.46SkalTuraeverything has been piece of cake to do so far
19:12.57SkalTuratrunking, outbound routing, voicemail
19:13.28opc0dehey can anyone point me to a list of recommended motherboards for sangoma/asterisk?  I'm putting together a system to use with asterisk so I want some motherboard recommendations
19:13.51GuruDomtyan motherboads
19:13.55jbalcomb[TK]D-Fender do I have to put the SIP server info under the SIP config /AND/ the Line config to get this thing to make a call?
19:13.59[av]banihmm the only diff is i'm changing the callerid before i enter queue
19:14.08[av]banivs direct dial
19:14.18opc0deGuruDom: tyan? is that what most people recommend?
19:14.26jbalcombopc0de Digium has some recommendations and not-recommendeds
19:14.35SedoroxAsterisk & skinny.... Asterisk can act as a skinny server, but not a skinny client, correct?
19:14.39GuruDomWell Tyan is the most stable of all the systems out there
19:14.43asterboynice irssi link: http://www.garion.org/irssi/features.php
19:14.43GuruDombut there is a cost
19:14.47jbalcombopc0de also on the wiki there are a few notes on hardware regarding what people have worked with
19:14.53opc0deGuruDom: what's the cost?
19:15.07opc0dejbalcomb: yeah, I've seen a few messages on the wiki, I was hoping for a more comprehensive list
19:15.38GuruDomfor instace: I have a "Vonage like" voip company and I use Tyan sysyems because i do not want my servers to go down due to motherboard failure
19:15.58jbalcombopc0de just buy a dell 2850
19:16.00asterboyok, thx for the suggestions on the phone lock thing,
19:16.17jbalcombopc0de and run RH ES 4
19:16.25GuruDomSo it really depends what you are useing the system for
19:16.26*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
19:16.31[av]baniFrom: ""Sales"" <sip:Hold Queue@192.168.42.1>;tag=as205eef2d
19:16.37[av]baniis that an acceptable sip url?
19:16.41opc0dejbalcomb: yeah, is that a system known to work well?
19:16.44asterboyya, that sucks.
19:17.00asterboymakes you wonder why even have an expansion module.
19:17.05jbalcombopc0de yes, i spoke with several digium reps and they recommended it to me
19:17.05[av]banii'm wondering if the polycom doesnt like a space in the sip number field
19:17.10asterboyguess to make it easy on dealing with extensions.
19:17.20asterboybut you kinda need to know if it's in use.
19:17.34asterboyguess you'll find out when you go to transfer.
19:17.39MstlyHrmls[av]bani: I think it might need to be hex encoded
19:17.40jbalcombopc0de its also that RH ES 4 is required for digium support contracts
19:17.47[av]baniMstlyHrmls: eg %20 ?
19:17.48justinubani: the "" is unacceptable
19:17.55[av]banijustinu: i changed that, no diff
19:17.57justinuok
19:18.01jbalcombopc0de and RH ES 4 is available from Dell on the 2850
19:18.02opc0dejbalcomb: what's the exact model name of the 2850?
19:18.03[av]banibut other phones have no problem with it
19:18.12[av]banigrandstream and cisco dont complain
19:18.14justinuanother guy was having trouble with 400 Bad Requests due to ""
19:18.21[av]banipolycom doesnt complain, it just doesnt stop ringing
19:18.46[av]baniso i'm wondering if  sip:Hold Queue@  is confusing the polycom
19:18.49[av]banithe space
19:18.54jbalcombopc0de i just the 2850 and customized it. i think you definitely need the rizer card though and Dell recommended leaving the on-board SCSI RAID in place
19:19.01MstlyHrmls[av]bani: yeah %20
19:19.07*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
19:19.23*** join/#asterisk mroth_imm (n=chatzill@63.65.26.220)
19:19.32SedoroxCan Asterisk act as a skinny client?
19:19.34*** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
19:19.37Fedoracore6some udy can help me , cos my code for delete did not function
19:19.47Fedoracore6http://pastebin.com/618460
19:19.58Fedoracore6i try , bu still fail
19:20.14mroth_immis anybody aware of any downsides to allowing the kernel to balance interrupts from the ethernet device across CPUs on an SMP system?
19:20.24asterboylol, playing with "/lastlog -hilight" in irssi and thought brettnem was telling me something, hence my last few lines.
19:20.25opc0dethanks for the help guys
19:20.27*** part/#asterisk opc0de (i=adam@CPE006008148866-CM000f9fa8c50a.cpe.net.cable.rogers.com)
19:20.43asterboyThe more I start to leanr irssi, the more I fall in love with it.
19:20.53asterboys/leanr/learn/g
19:20.55mroth_immi'm concerned about retransmits if the packet data is split across different CPUs and re-assembled in a different order than expected
19:21.13ManxPowerFedoracore6, why are you not using the pgp AGI libs?
19:21.32[TK]D-Fenderjbalcomb : Either should work.
19:21.37mroth_immthe advantage to balancing the interrupts is it keeps the CPU utilization even
19:21.43ManxPowermroth_imm, I don't think any of us are worried about it.
19:21.47mroth_immjust trying to determine any possible gotchas
19:21.56mroth_immManxPower: excellent news
19:22.16[TK]D-Fenderjbalcomb : I believe if you put it in the msater vs line that you won't have to specify it for multiple accounts on the same server.  Not that I suspect you should need more than one anyways.
19:22.19Fedoracore6ManxPower : hemm pgp AGI libs ... i did no bro
19:22.27mroth_immanother question then: on my FC3 boxes, i see a user-space daemon called irqbalance
19:23.03mroth_immbut the 2.6 kernel also seems to balance irqs itself (irq affinity + acpi irq routing)
19:23.08asterboyIf I can't use the http interface for Polycom phones, no big deal...everything is best done at a command prompt anyway.
19:23.43ManxPowerasterboy, the HTTP interface takes a while to come up.
19:24.12justinupolycom's web interface sucks.... give up
19:24.23mroth_immis there any legitimate reason to have the irqbalance daemon running (it continually updates the smp_affinity files under /proc/irq/<irq> ) on such a kernel
19:24.27mroth_immor is it just something left over from the 2.4 days?
19:24.32ManxPowerwonder of all wonders!  searching for 'PHP AGI'....
19:24.40Fedoracore6ManxPower : where i can use the  pgp AGI libs
19:25.01[TK]D-FenderFedoracore6 : My best guess : in a PHP script for AGI :)
19:25.19Fedoracore6hehhehehe
19:25.23Fedoracore6:D
19:25.42ManxPowerFedoracore6, use google
19:25.53Fedoracore6huhuhuh ok :D
19:26.18asterboyManxPower, do you suggest disable then and the phone will boot faster?
19:26.33asterboyor after making a change you need to wait a while.
19:26.59[TK]D-Fenderasterboy : the only thing that makes polycom's faster is being dropped off a roof :)
19:27.07justinuheh
19:27.13[TK]D-Fenderasterboy : another great reason to provision them :)
19:27.16asterboylol
19:27.33asterboyI do provision them, (ftp server).
19:28.06asterboytftp is great cause you don't need to rename files for upgrades.
19:28.20justinuHTTP provisioning is teh shizzle
19:28.21asterboybut couldn't get Polycom to talk to my tftp on linux.
19:28.25asterboyIt did on windows.
19:28.40asterboystupid-ftpd works well.
19:28.56asterboyvim does a good job of editing XML.
19:29.19Fedoracore6bro i already find pgp AGI libs, but ..... how come i must start using this
19:29.25*** join/#asterisk KranZ (n=user@imail.bestline.net)
19:29.41asterboyIt can be slow when your ftp is down though and the phone needs to reboot without it.
19:29.52Fedoracore6is like some plugin
19:30.19asterboyI'll try an upgrade later and see if the phone will save settings via http.
19:30.28*** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net)
19:30.34asterboyActually, irrc, the phone would not save settings even from the user menu.
19:31.08*** join/#asterisk KranZ (n=user@imail.bestline.net)
19:31.28Fedoracore6pgp AGI libs or php AGI libs .... huhuhuh
19:31.35Fedoracore6i confuse
19:31.37asterboyI have the IP 300, 500 and 600...all great phones.
19:31.40*** join/#asterisk KranZ (n=user@imail.bestline.net)
19:31.55asterboyThe zaptel X101P cards are not great quality though.
19:31.59*** join/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net)
19:33.07asterboyWhile waiting for some ATAs to arrive, I was installing an IP300 temporary at a potential clients residence for VOIP service.
19:33.14asterboyShe was a troll.
19:33.25asterboyI had CSI visions of her spitting into my IP300.
19:33.43asterboyI terminated the service, took the phone out and decided to drop them as a client.
19:33.48asterboyobsessive or what.
19:34.26asterboymoral of the story, don't let trolls spit on your IP300.
19:34.38asterboystock plenty of ATAs
19:34.39[av]baninope! it's not the space
19:34.42jsharpI hate it when that happens.
19:34.46asterboylol
19:34.48[av]banii used %20 and the polycom still shits itself, keeps ringing
19:34.52fu3hi lads
19:34.56*** join/#asterisk davidcsi_ (n=davidcsi@146.Red-83-32-41.dynamicIP.rima-tde.net)
19:35.03MstlyHrmls[av]bani: huh
19:35.25salviadudreally?
19:35.27MstlyHrmls[av]bani: can you toss an ethereal capture up somewhere?
19:35.32fu3So.. get this... after weeks of work, the telco has now told me that my non-pri T1 cannot operate two-way and that i Need to get a DSS trunk, and that it takes five days to swing lines from the POTs system to being a DID on the T1.
19:35.36fu3what a bunch of bullshit
19:35.39salviaduda girl spit on your ip300?
19:35.48[av]baniMstlyHrmls: will a sip debug peer xxx do it?
19:36.12ManxPowerfu3, that is why you get PRI
19:36.21jsharpYour non PRI line can't run both ways?
19:36.25MstlyHrmls[av]bani: not sure, I tend to work with captures when I can; I'm willing to have a look though
19:36.27fu3yeah.. well needless to say, im ditching these guys and buying the PRI direct from qwest.
19:36.36fu3jsharp.. thats what they tell me.
19:36.39jsharpYou have a T1 from Joe Bob's Bait Shop and Telephony service?
19:36.44salviaduddid you hang up the phone with fashion?
19:36.47fu3from "intertech"
19:36.50*** part/#asterisk terrapen (n=cjs@166.70.135.60)
19:36.51salviadudlike "good-bye jerks"
19:36.58salviadudor "sayonara suckers"
19:37.04fu3haha actually I called them on their lies, and then said "comments?"
19:37.08fu3and got no response
19:37.25salviadudevil bastards, they had it comin'
19:37.30fu3yes, they did.
19:37.30Fedoracore6huhuhuhu
19:37.33jsharpI'd have gone over and peed on their desks.  But that's just me.
19:37.35fu3theyve been lying to me for years.
19:37.43fu3all these years they said I HAD to go through them
19:37.43salviadudyou see, you can't bullshit the open source community
19:37.45fu3but no.. I dont.
19:37.49*** join/#asterisk ToTo (n=ToTo@host114-166.pool870.interbusiness.it)
19:37.52salviadudwe know when something sucks
19:37.56fu3hell yes.
19:38.06fu3I just got my polycomm 301, 501 and 601 demo's today.
19:38.11fu3at least that worked out :)
19:38.15jsharpYou will love and cuddle your PRI after dealing with E&M Wink start.
19:38.23*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
19:38.23Fedoracore6eheheehee
19:38.30fu3E&M wink was a snap actually.. once i realized I had to use that type.
19:38.58fu3in any case, im going to ditch InterTech's bullshit t1, and buy the T1 direct from qwest.
19:39.13fu3who guaranteed me that they can turn over my old lines to DID in less than 3 hours, if scheduled in advance. (No big deal)
19:39.25fu3and they said it IS two-way, and it CAN pass caller-ID across the line.
19:39.27ManxPowerquest is a bunch of liers.
19:39.33fu3and i'll get direct access to my 911 records.
19:39.40ManxPowerfu3, callerid name only or name and number?
19:39.43fu3I havent found them to be liars yet..
19:39.45fu3name and number
19:39.54ManxPowernot on an E&M.
19:40.05fu3of course, im not dealing with the lowest level people either.
19:40.11fu3no, im not getting callerID now.
19:40.12fu3but I need it,
19:40.15fu3and thus.. PRI.
19:40.41ManxPower*nod*
19:40.55fu3So..  I hope to get this bullshit straightened out over the next few days.
19:41.02fu3I just hate going so far, and then having to go back to square one.
19:41.03ManxPowernow all you have to do is managed to get out of your contract with the current carrier
19:41.30fu3i dont have a contract :)
19:41.34fu3so.. NO problem!
19:41.51ManxPowergood luck with that.
19:42.01fu3thanks.
19:42.08ManxPowerthe current carrier will try to bill you for SOMETHING, early termination or install or something.
19:42.21fu3no, it doesnt work like that.. this is a state of minnesota thing.
19:42.24fu3not for me anyway
19:42.27*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:42.43ManxPowerah.
19:43.00fu3so, im lucky in that respect.
19:43.21ManxPowerI have a policy of never doing work for govt, charity, or religious orgs
19:43.30*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
19:43.35fu3yeah, it can be a real pain in the ass
19:48.31Kattywhat's a good album?
19:48.36jbalcomb[TK]D-Fender it seems you are incorrect. PO cancelled.
19:48.44jbalcomb[TK]D-Fender bwhahahaha
19:48.50[av]baniwhat?
19:49.14*** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net)
19:49.27jbalcombManxPower you have a policy against being able to charge rediculous amounts of money for simple services? ;)
19:49.47ManxPowerjbalcomb, not all.  It's the cornerstone of my business.
19:50.04b00merany recommendations on where to buy a Cisco 7940g at a good price?
19:50.06b00merebay?
19:50.22jbalcombb00mer atacomm.com if not eBay
19:50.40jbalcombb00mer maybe even Craig's List
19:50.51Kattyb00mer: voip-supply.com might have them
19:51.01b00merhmm... hadn't thought of craigslist... good calll
19:51.03brettnemmy house is on craigslist. :)
19:51.21b00merthe 7940g is better than a 7940 correct?
19:51.27*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
19:51.37*** join/#asterisk AlexCTI (n=alex@68-66-149-78.miamfl.adelphia.net)
19:51.38b00mergig + improvements?
19:51.50Fedoracore6ok thank you all
19:51.52Fedoracore6:d
19:51.54Fedoracore6i gtg
19:51.58Fedoracore6tired
19:52.01Fedoracore6byeb
19:52.02*** join/#asterisk garsan (n=garsan@dsl-201-133-99-34.prod-infinitum.com.mx)
19:54.02astra^^i need some support on * . it is not going well for me.. low ASR and ASD
19:54.05*** join/#asterisk lesouvage (n=lesouvag@82.74.19.41)
19:54.48astra^^some proffesional support ..?
19:55.01SkalTuraASR and ASD?
19:55.50lesouvageWhat can I do woth the asterisk settings to avoid an awfull echo wile using sjphone on a qtek9000. Technically it is working but calling a cellphone from the qtek9000 through an asterisk box the cellphone has an echo with a delay of almost 3 seconds.
19:56.53astra^^SkalTura:ie very low calls. many call getting dropped.cant figure out the problem
19:57.19SkalTuraastra^^: aah, how's the line quality when the calls don't get dropped?
19:57.29astra^^exelent
19:58.04SkalTuraok, loud & clear i assume...
19:58.13astra^^yes
19:58.20*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
19:58.31SkalTuraastra^^: is this with what kind of calls?
19:58.52astra^^?
19:58.56[av]baniany * coders around want to take a look at a sip debug? polycom<->app_queue bug
19:58.57SkalTuraVoIP<>VoIP, or perhaps VoIP<>PSTN?
19:58.58astra^^i dint get u ?
19:59.19SkalTuraso am i even on correct tracks
19:59.27astra^^voip voip
19:59.50astra^^* is fwding to another server
19:59.53SkalTurai was quessing PSTN<>VoIP Phone, so was thinking along the lines of poor phone lines etc.
19:59.58[av]banijustinu: wanted to look at the app_queue bug?
20:00.10justinusure
20:00.29SkalTuraastra^^: checked packet loss? Note, that i'm not applying any * knowledge here, just some basics as somewhat knowleadgable about electronics & networking...
20:00.30*** join/#asterisk jskcrtech (n=J@53-pool1.ras14.floca.alerondial.net)
20:00.34ms345anyone have the sip firmware for a cisco 7960 handy?  cisco site requires login for download....
20:00.43justinuillegal to give it out
20:00.53justinuw'ere law abiding folks here
20:00.58ms345even if I own the phone?
20:01.00*** join/#asterisk ke4zvu (n=savirc@srv.fgp.com)
20:01.03justinuyeah, unfortunatly
20:01.08justinucisco has some lame policies
20:02.50ms345the phone is advertised as supporitng SIP but requires $$$ login to use that functionality.... nice...
20:03.01justinuthat's right
20:03.17justinuone reason why i don't recommend cisco
20:03.33docelm0ms345, what version?
20:03.50ms345any
20:04.06ms345i read about the upgrade then no downgrade issue with > 5 but really don't care
20:04.12docelm0Well pick one..  I have an enterprise login for cisco
20:05.39ms345I'll research...
20:06.41jsharpI'd recommend 7.5.  It works well for me.
20:07.36justinu7.5 seems to be working for most
20:07.58docelm0I have 8.2 just came out 10MAR06
20:07.59*** join/#asterisk ahattar (n=ahattar@static-68-236-175-229.ny325.east.verizon.net)
20:08.04docelm0or 7.5
20:08.07docelm0or whatever
20:08.10docelm0you choose..
20:08.10*** join/#asterisk stoffell (n=stoffell@d5153FC83.access.telenet.be)
20:08.46jsharpI haven't seen any problems with the limited deployment of 8.2 I have.
20:09.00docelm0huh?
20:09.05docelm0What the hell did you just say?
20:09.32jsharpI'm only running 8.2 on a few phones, and I haven't seen any problems with it yet.
20:09.32docelm0ms345, hay make it quick.. :)
20:09.37mockerMy question to the list about the stability of the TE406P and TE410P make it sound like those are pretty unreliable cards.
20:09.38docelm0ohh ok
20:09.44mockerAnyone here have good luck w/ them?
20:09.57docelm0I have a TE410 and dont have any issues
20:10.28justinua lot of people like them
20:10.30justinusome complain
20:10.32mockerWeird.
20:10.39mockerdocelm0: What motherboard are you using?
20:10.46justinuseems like interrupt conflicts are the most common troubles
20:11.01justinuand some other weird mobo incompatibilities certain people have/had
20:11.20mockerjustinu: I wish there was a list of motherboards certified by digium.
20:11.34justinuyeah, you just gotta use what people are succesful with, i guess
20:11.34docelm0Tyan
20:11.57justinuthe main problem i've seen is HDLC errors caused by hardware issues
20:12.15*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
20:12.16jsharpWith the turnover rate of motherboards, its hard to keep anything like that current.
20:12.33*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
20:12.51docelm0I am using a Dual Opteron 248 Tyan 1U Rack Server
20:12.53mockerjsharp: Doesn't really have to be current, or exhaustive..
20:13.07qseekwhat is the difference between an itanium and a pentium
20:13.11qseekprocessor
20:13.19docelm032bit
20:13.23jsharpOne sucks.  The other one sucks.
20:13.24jsharpOhwait.
20:13.31KranZi have dual 244s on Tyan
20:13.36KranZhad nasty hdlc errors
20:13.47docelm0I dont have any problems
20:13.49KranZtill i bought a sata card and disabled the onboard
20:14.15KranZcouldnt move the pri card b/c its 32bit and there is only 1 32bit slot
20:14.28angom_wKranz: is that the only way to get read of hdlc errors ?
20:14.36justinuqseek: itanium is a totally different architecture
20:14.48justinuno binary compatibility
20:14.48KranZangom_w: for me, yes
20:14.50docelm0qseek, yes..  32bit
20:14.51docelm0:)
20:14.57*** join/#asterisk GoRK (n=GoRK@209.40.175.194)
20:14.58qseekyeah i was trying to see how to compile the G.729 codec source code
20:15.02justinuit's like comparing a pentium to an Alpha AXP or something
20:15.02KranZangom_w: the easier way would have been just to move the t1 card to a different slot
20:15.09Abydos313maybe you could add pci slots with a riser card?
20:15.10docelm0qseek, YOU HORRIBLE PERSON!
20:15.15qseeki am
20:15.19justinuheh
20:15.19qseekwhy ami horrible
20:15.26docelm0your stealing from Digium
20:15.27justinustealing g729.. shame :P
20:15.29qseekno i am not
20:15.34docelm0How are you not?
20:15.39Beirdohow is he?
20:15.41docelm0qseek, did you buy the licensing?
20:15.43qseeki was just trying to see this code someone had put out on the net
20:15.49justinuperhaps he lives in a country where the patents aren't valid
20:15.54KranZdocelm0: digium doesnt own the rights to g729
20:15.58Beirdostealing from the patent owners, perhaps
20:16.03Beirdobut not from digium
20:16.09docelm0KranZ, I know but thats what Mark would say..  :)
20:16.17docelm0KranZ, he's chewed my ass a couple times
20:16.19KranZcan't argue with that
20:16.26Beirdowell then he's wrong
20:16.39qseekwell i did not write the code
20:16.43mockerdocelm0: What model Tyan board do you have?
20:16.47qseeki found it on one of the links some guy wrote it
20:16.49docelm0Beirdo, Digium sells licensing for the g729 module
20:16.50*** join/#asterisk Katty (n=angela@64.82.232.54)
20:16.57docelm0readytechnology?
20:16.58qseekand i was just trying to compile it ...
20:16.59docelm0ya..
20:17.02qseekdidnt say i was going to use it
20:17.07ke4zvuhi, i've just begun using Asterisk with cisco SIP phones.  if a call is ringing at an extension, is there any way in the dial plan or what not for the called user to press a button without "answering" the call that would instead send the call to the called user's VM?
20:17.17KranZqseek: are you messing with the intel code?
20:17.18qseekwow i got the short end of that stick
20:17.23angom_wKranZ: I have a system that is throwing some hdlc errors, will try changing the card to another slot but... /proc/interrupts show the card in its own irq number...
20:17.25qseekall i did was ask a question
20:17.32qseekand everyone started chewing me out :(
20:17.38KranZangom_w: yeah, that doesnt make a difference
20:17.39justinunot everyone
20:17.58docelm0qseek, Consider this..  Asterisk == Free..  How hard is it to shell out $10 a license?
20:17.59KranZangom_w: chances are its on the same physical bus as your harddrive or onboard nic
20:18.09qseeki dont mind shelling out the 10 buck
20:18.12qseeki was just trying to learn
20:18.15KranZangom_w: when devices are on the same bus, they fight for bandwidth
20:18.17docelm0It helps digium and us
20:18.19Beirdodocelm0: yeah, but choosing to get it from elsewhere isn't stealing from digium, it's stealing MAYBE from the patent holder (unless you license it from elsewhere)
20:18.21qseeki know i know
20:18.40angom_wKranZ: so, changing the slot might do it to another bus ?
20:18.41KranZangom_w: in my case, my sata drives would cause the aborts whenever there was significant disk usage
20:18.44KranZyeah
20:18.47qseekhow is g.729 a licensed thing..if u follow the protocol and develop your own...
20:18.54qseekthat is not cheating is it?
20:19.02Beirdobecause they have a patent on the algorithm
20:19.07angom_wok
20:19.15qseekjust  a philosophical question
20:19.17Beirdothat's my understanding of it anyways
20:19.21KranZangom_w: a quick way to see if it's your disk is run "hdparma -tT /dev/hda"
20:19.22docelm0Beirdo, no shit there sherlock..  But just the same..  Asterisk is free..  The least you can do is help out digium and keep the project floating
20:19.30Abydos313is there no open source g729 codecs?
20:19.33qseeki dont think that holds any water...
20:19.33Beirdoheh.
20:19.39justinuheh
20:19.47GoRKtechnically only digium has the authority to hook proprietary-licensed modules into asterisk so you couldn't really license a g729 plugin to asterisk from elsewhere; you could license g729 maybe and then write/use an open source implementation i guess but that would be nearly impossible to do without spending  a fortune
20:19.57Beirdoif they'd start selling X100P again, MAYBE I'd buy stuff from them
20:20.23GoRKthe x100p didnt even work very well; why the love for it?
20:20.32BeirdoGoRK: unless you redistribute, you have the right to do whatever you want to GPL code
20:20.37jbalcombIs this going to get me the current stable release of Asterisk?
20:20.38jbalcombsvn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
20:20.43qseekbeirdo: i would have said u can buy x100p clones  but then u might chew me out on that too :)
20:20.44justinuno
20:20.53justinujbalcomb: download the tarball from asterisk.org
20:20.59BeirdoI'm not going to buy a $300 card just to get one line of FXO
20:21.17qseekBeirdo: did you try the new sangoma board
20:21.22qseekthat is not  as expensive
20:21.24jbalcombBeirdo can't you just use a Voice Modem?
20:21.24Beirdoqseek: I DID buy crappy clones :)
20:21.34Beirdomaybe.
20:21.45docelm0sigh
20:21.51qseekor u could get a handytone 488 from grandstream
20:21.55qseekthose are pretty good...
20:21.58BeirdoI think that once I move, I'll just use my SPA3000, and do that
20:22.13qseekthere u go beirdo...
20:22.15jbalcombSPA-2002 works grizzeat!
20:22.29qseekso does anyone have any idea how PRI pricing works...
20:22.29Beirdonevertheless, digium's products aren't aimed at the part of the market I'm in, so not much I can do about that
20:22.39qseekI am totally  lost in that area
20:22.45jbalcombBeirdo what market is that?
20:22.50Beirdoexcept the IAXy
20:22.51qseekwhat are local loop charges...
20:22.59jbalcombqseek you pay for access and you pay for a local loop
20:23.09Beirdoa geek with a PBX at home... or small business with 2 lines, etc
20:23.11GoRKerr it's $140 not $300; and that price is only $41 more than they sold the x100p for anyway (Despite $99 being overpriced for what is basically a winmodem), and the physical interface is arguably better on the tdm400 card too
20:23.26jbalcombqseek the local loop is actually the wires to get onto the /network/ and is priced by distance to the CO
20:23.45qseekso does that include all local calling charges..or u have to pay for those seperate
20:24.03KranZincluded
20:24.10jbalcombqseek local calls are included generally just line a standard phone line
20:24.10Abydos313on spa3k which do i configure for service to sip account? pstn or line1
20:24.37jbalcombqseek they usually offer a package deal as well for toll calls and long distance calls
20:24.41qseekwouldnt I get DIDs for those PRI channels? rather than getting a line
20:24.45GoRKsmall biz/home use BRI is totally the coolest way to go
20:24.52BeirdoBRI?!
20:24.54Beirdoare you nuts?
20:24.59angom_wKranZ: hdparma or  hdparm  ?
20:25.00jbalcombqseek your local loop providor is not always the same company you order your PRI from
20:25.05BeirdoI ain't paying for BRI service
20:25.11qseekah i see
20:25.27qseekso do u have any recommendations about providers?
20:25.39KranZhdparm
20:25.43qseekI am trying to see what would be most feasable
20:25.46jbalcombqseek the company you order the PRI will handle the local loop providor if its another company and you will only get one bill
20:25.59KranZangom_w: if you're using sata drives, do /dev/sda and not /dev/hda
20:26.12jbalcombqseek might as well find out who even offers service in your area before you starting thinking about who to pick
20:26.14angom_wOk
20:26.30*** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk)
20:27.15jbalcombhdparma is short for Hoodlum Parma which is a trailer park friendly city just south of Cleveland, OH
20:27.42qseekwell if I had my server in a colo which had cross connects to a provider
20:27.48qseekwould that not be cheaper?
20:29.10jbalcombqseek not sure I'm following you there.. you want to host your PBX at a remote data center and have your office phones make calls out through the remotely located PBX?
20:29.23Octothorpe~X-Rob
20:29.25jbotfrom memory, x-rob is not a palindrome
20:29.33qseekno i am trying to set up remote users on IP accessing my BOX..
20:29.36Octothorpe~seen X-Rob
20:29.45jbotx-rob is currently on #asterisk (14h 50m 13s), last said: 'so whoever set the pricing went 'Ooh, we can make more money on this one!' and put $500 on it'.
20:29.45qseekjust setting up phone lines for them through the PRI
20:29.58KranZheh
20:31.06jbalcombqseek seems pheasible and likely the data center is closer to a CO than you are so it would be cheaper.
20:31.35qseekso what is usually the rate for a PRI...
20:31.42qseekhavent investigated yet
20:31.53qseekI found a T1 provider for 400 including local loop charges
20:32.03KranZqseek: 400 to 1200
20:32.18KranZdepends heavily on what city/market you're in
20:32.23qseekdallas
20:32.31KranZtx is a good market for that
20:32.41jbalcombim paying about $600 for each of mine. $300ish for the service providor (cogent) and $300ish for the local loop providor (verizon)
20:32.56KranZqseek: which provider does 400?
20:33.02qseekspeakeasy
20:33.21qseekso any recommendations for providers in the dallas area?
20:33.24GoRKyeah t1's in tx are cheap; i can get a pri from a local clec for $360
20:33.33KranZGoRK: what city?
20:33.39GoRKamarillo
20:33.55qseekyeah but then u only get local terminations in amarillo
20:34.05jbalcombif our office wasn't in Brunstuckey we would get a much better price
20:34.09qseeki am trying to find out how to get as many POPs as I can
20:34.18qseekand do unlimited local in that area
20:34.55jbalcombqseek sounds like a money maker. I'm in for $75/hr. jbalcomb@hotmail.com
20:35.02KranZheh
20:35.06GoRKwell admittedly we could get virtual numbers from another provider or do IP trunks to our own local location somewhere else
20:35.14Zodiacalqwell any ideas why my ip communicator's status says: No CTL installed? im trying to use it with chan_sccp. i got my cisco hardphone working with sccp but having trouble getting the ip communicator working...
20:35.53qseekjbalcomb
20:35.56*** join/#asterisk ToTo (n=ToTo@host114-166.pool870.interbusiness.it)
20:36.00qseeknot a money maker yet :)
20:36.01*** join/#asterisk synthetiq (n=sdfs@204.124.238.248)
20:36.04qseekhopefully soon though
20:36.06jbalcombqseek never is
20:36.15jbalcombqseek always hopeful
20:36.15synthetiqports 49k-64k would they have anything to do with voip?
20:36.25KranZsynthetiq: not usually
20:36.35synthetiqwhat do u mean not usaully
20:36.58KranZunless you set up a device to use them
20:37.05synthetiqcould it be some nat traversal deal?
20:37.05KranZrtp traffic is typically <20k
20:37.07KranZyeah
20:37.16KranZif they go through a nat, then its a different story
20:37.23qseekok people thanks for your help.....
20:37.38KranZqseek: shoot me a msg if you set your targets on austin
20:38.02qseekkranZ : I will keep that in mind....
20:38.20qseekright now i only want to focus on Dallas, Houston
20:38.22*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
20:39.13*** join/#asterisk Peaceful (n=Peaceful@70.98.162.62)
20:39.22jarrodqseek
20:39.26jarrodwhats the name of your wompany
20:39.27Beirdohe's gone
20:39.37*** join/#asterisk SwK (n=Silik0nJ@c-24-92-158-154.hsd1.ga.comcast.net)
20:39.40Beirdolooks like "Nortel Networks" to me :)
20:39.42jbalcomb*poof*
20:39.55PeacefulIs it possible to alter volume level (gain?) for specific participants of a conference call in asterisk?
20:40.01*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
20:40.14qseeksorry got bumped :(
20:40.21*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
20:40.23jbalcombPeaceful sure, if you have a phone that has configurable gains
20:40.51KranZthere are in-conference menu options which allow a user to adjust their volume
20:41.10jbalcombKranZ that sounds too easy
20:41.11PeacefulWhat about for an admin to adjust others' volumes?
20:41.22KranZbut there's not an option to raise a user's gain
20:41.26jbalcombKranZ how am i gonna get my $75/hr if everything is so easy?
20:41.33PeacefulWe get some people that come in really loud, and others that come in real soft
20:41.42justinuwoohooo... AMI patch complete.
20:41.48KranZPeaceful: doubt it, but you might want to do a feature request for that
20:41.51*** join/#asterisk Assid (n=assid@59.183.27.23)
20:41.58jbalcombjustinu what does it do? I use that crap.
20:42.25PeacefulWell, but theres a way for each user to do it themselves using in-conference options?
20:42.26justinuyou use originate action?
20:42.35KranZPeaceful: if they could integrate that into the web interface, it would be a neat option
20:42.52KranZPeaceful: just to raise the volume of the conference
20:43.08KranZPeaceful: soft people get louder, loud people get louder
20:43.19PeacefulKranZ: Ick.  Not what I'm looking for.
20:43.22justinui had a problem with AMI not having anyway to link what channel is returned from an originate action
20:43.40justinuactually, there is a way, but you have to wait until the channel answers to get the Channel name
20:43.46qseekKranz: I thought conferencing worked on the principal of letting the loudest get through :)
20:43.50PeacefulKranZ: I'm using PRI cards, is there a way I could temporarily boost the gain on specific zap channels?
20:43.54justinuit's not acceptable to me to wait until the channel is answered before I can know the uniqueid/channel name
20:44.09KranZPeaceful: im not sure if you can even boose the gain on a pri channel
20:44.23justinuyou can
20:44.26KranZqseek: somehow the loudest always get through
20:44.28justinuyou can set gains/channel
20:44.35Peacefuljustinu: how?
20:44.38qseekno on the CPE justinu
20:44.47qseekit can only be done on the CO site
20:44.48*** part/#asterisk davidcsi_ (n=davidcsi@146.Red-83-32-41.dynamicIP.rima-tde.net)
20:44.59justinuyou can set the TX gains and RX gains/channel
20:45.02KranZjustinu: isnt that an analog only feature (per asterisk)
20:45.04justinuwe've done it
20:45.12justinuno, it applies to T1 also
20:45.28KranZi have a faint memory of messing with that and watching the meter
20:45.53justinuplease don't ask why I needed to do such a thing, it would take a long time to explain
20:46.03Peacefuljustinu: can you change those gains while asterisk is running?
20:46.06justinuno
20:46.11Peacefuloh.  Dang.
20:46.19justinurequires ast to be restarted, or chan_zap.so unload/load
20:46.19KranZi think there was a feature request for that tho
20:46.23PeacefulIck.
20:46.32PeacefulWhere are feature requests recorded?
20:46.35*** join/#asterisk Dovid (n=Dovid@89-138-33-253.bb.netvision.net.il)
20:46.38justinubugs.digium.com?
20:46.38Peaceful...assuming they are recorded somewhere
20:46.48KranZbugs.digium.com
20:47.04SplasPoodDoes anyone know how to globally disable RFC3389 (Comfort Noise) on an AS5300?
20:47.07KranZit was either a feature request, or a patch
20:47.17justinuSplasPood: g729?
20:47.22Nodrenwhats a good tutorial that'll explain how extensions.conf, zapata, asterisk, and sip works?
20:47.31justinu~thebook
20:47.33jbotthebook is, like, Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
20:47.33KranZNodren: www.voip-info.org
20:47.38SplasPoodjustinu: no, ulaw..  between the 5300 and asterisk
20:47.39justinucheck that link, nodren
20:47.46Nodrenohh i havnt seen asteriskdocs
20:47.47Nodrenthanks!
20:47.48Nodren:D
20:47.48justinuSplasPood: ah... not sure, sorry
20:48.00KranZSplasPood: is it interfering?
20:48.01justinunodren: you can either buy the book at the bookstore, or download the PDF ;)
20:48.14*** join/#asterisk terrapen (n=cjs@166.70.135.60)
20:48.23Nodreni saw that oreilly book on pdf
20:48.28KranZSplasPood: * should nag, but i've never had a problem
20:48.28justinuyeah, read it
20:48.35justinuyou'll be a whole different man ;)
20:48.45Nodrenhaha
20:49.03*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
20:49.21SplasPoodKranZ: asterisk does nag, and it...  Doesn't appear to cause a problem... However for caller id numbers KNOWN to the 5300 a no vad in the stanza seems to take care of it...  there's gotta be some sorta 'defaults' for calls originating with unknown CID
20:49.27justinuif you can do something in an hour that would take your coworkers at least a week to figure out, is it ok to work for an hour, and slack off the other 39?
20:49.44qseekamen justinu
20:49.50KranZSplasPood: well, CID and rfc3389  have nothing to do with each other
20:49.54Nodrenwell thats not a fair assesment
20:49.57KranZSplasPood: sometimes you just dont get CID info
20:50.22Nodreni work for a website and do php coding, the girls who do tech support cant do it nearly as fast as me.. doesnt mean i can slack off when i have to do tech support
20:50.46KranZSplasPood: err... hmm..
20:51.05*** part/#asterisk Peaceful (n=Peaceful@70.98.162.62)
20:51.14justinuok, so maybe I should only slack off for 20 hours?
20:51.20justinui'm still twice as fast
20:51.24Nodrendont slack off at all
20:51.25KranZSplasPood: could you paste your config to pastebin.ca?
20:51.33justinuNodren: that's crazy talk
20:51.34Nodrenthen they'll hire more people to replace the position they just promotd you out of
20:51.35Nodren:D
20:51.55*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
20:51.58justinuno way
20:52.03Nodrenthey did where i work
20:52.06justinuthey can't find people with skillz
20:52.11Nodreni've been promotd, and my old job was replaced by 2 people
20:52.13justinupeople are fuckin' morons
20:52.19KranZheh
20:52.20Nodreni started less then a year ago
20:52.21justinui'm serious
20:52.27Nodrenand i've gotten 3$ in raises
20:52.27Qwell[]Zodiacal: That's normal.  YOu can ignore the CTL stuff
20:52.29justinui can't believe the people we interview
20:52.59Zodiacalqwell hrmm.. i have no options except the config button, and that only lets me see the status log and a few other version things..
20:53.15Nodrenwell everyone has those kinds of interviews
20:53.17Nodrenjust dont hire them.
20:53.18Nodren:P
20:53.20SplasPoodKranZ: any specific piece?  there's a bunch to it...
20:53.20Zodiacalits reading the firmware and SEP file.. but it doesn't get registered. when i do sccp show devices
20:53.22Nodrenheh
20:53.27Zodiacalqwell any thoughts?
20:53.43Nodreni got this job over more qualified people who lied and thought they could get away with it
20:53.43Qwell[]Zodiacal: I'd search the wiki
20:53.53KranZSplasPood: how about around where you put the "no vad"
20:54.02KranZSplasPood: anything before and after that might be relevant
20:54.16justinua lot of people like to lie these days
20:54.28justinuthey look good on paper, but fail horribly in an interview
20:54.32Nodrenwhich is why you dont hire those people
20:54.43justinusome of them get pissed when I ask them to take a programming test
20:54.45*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
20:54.48Nodrenmy boss was looking for a coder when he hired me, but the position was full time tech support and office admin
20:54.54Nodrenhaha
20:54.55SplasPoodKranZ: well the no vad is in a stanza for other DIDs and works as expected... Its when calling from an unknown number that I'm trying to figure out
20:54.59Nodrentheres a really sucky test
20:55.03Nodreni had to take
20:55.10Nodrenit had all sorts of trick questions
20:55.17justinui make them take the brainbench tests
20:55.18Nodrenthat could have two answers
20:55.24jbalcombjustinu i lie on my resume but i'm sweet in the interview. Just read some trade magazine articles and case studies. ;)
20:55.27justinuthose seem very good
20:55.28mog_worki think you should ask questions that have to do with company culture
20:55.36mog_worklike i would ask questions about family guy
20:55.42justinuthere's no culture at work
20:55.43justinu:(
20:55.46justinubunch of zombies
20:55.47Nodrenif you goto elance.com or rentacoder.com
20:55.50Nodrenthey have testing standards
20:55.51mog_workor other funny things
20:55.58Nodrenthat there users take
20:56.07Nodrenand the testing standards are from 3rd party companies
20:56.10Nodrenso those are good testing grounds
20:56.17Nodrensince so many tried and true coders can do em
20:56.22nDuffNodren: I was hired as a coder; they hired me to do that and 20% system administration, and then I ended up spending the next year as 95% sysadmin 5% coder.
20:56.37*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
20:56.41jbalcombnDuff yeah, love that one
20:56.42Nodrenyeah they want to make me a sys admin now too
20:56.49Nodrenand i'm refusing as best i can
20:56.52Nodrencause i love coding
20:57.01Nodrenthey just dont know any good linux geeks
20:57.17jbalcombnDuff 3rd day on my new asterisk sysadmin job they asked me if i new any php and if wanted to do some developing
20:57.22*** join/#asterisk Creperum (n=ilya@82.207.62.67)
20:57.24Hmm-work<PROTECTED>
20:57.27Nodrenits not easy to find a trustworthy linux guy, who can do shell scripting, is resourceful and wont jack up your box after you give him root
20:57.38synthetiqhellp? im getting unable to open /dev/zap/channel
20:57.41synthetiqany ideas why
20:57.42jbalcombNodren whats the company?
20:57.46synthetiqwhat cuases that
20:57.48KranZsynthetiq: ztcfg -vv
20:57.53Hmm-workmost people in here aren't going to jack your back
20:57.56jskcrtechNodren:  not really
20:58.05Hmm-workespecially the people in this chan that are known
20:58.05jskcrtechNodren: Thats why you find consultants with references
20:58.11nDuffNodren: Really good system administration includes coding -- everything from building custom tools to writing and debugging drivers.
20:58.13jbalcombHmm-work yeah, not your /back/ just your backside.
20:58.43Hmm-workyeah nDuff: people like that don't come cheap
20:59.13jbalcombnDuff really good system administration has more to do with seeing the big picture, managing projects, and operating as a business partner to the rest of the organization
20:59.30*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
20:59.46nDuffjbalcomb: that's a different aspect, but yes.
21:02.34*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
21:02.43Dovidanyone ever get contacted for a job and they want it dirt cheap cause asterisk is free and they expect the service from u to be the same ?
21:03.05synthetiqkranz
21:03.05jsharpMany times.
21:03.11KranZsup
21:03.28synthetiqim getting unable top open /dev/zap/ctl
21:03.42nDuffjbalcomb: I'm coming from the position of a small startup with a very limited budget and management who would rather build everything in-house if it'll result in immediate savings. We have new management in recently with large-company experience who aren't so afraid to buy instead of build... so it's a different kind of environment. Less need for miracle workers, but more sustainable.
21:03.58*** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net)
21:03.59KranZdo a "lsmod" and make sure you zaptel drivers are loaded
21:04.04synthetiqthey wont load
21:04.14synthetiqbecause it doesnt know about /dev/zap/channel
21:04.15KranZdo a dmesg after  you try to load them
21:04.17KranZand read the error
21:04.23synthetiqbrb
21:04.41*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
21:04.44Dovidi usualy just dont do it
21:05.15Dovidcause they knicle and dime everything
21:05.25KranZnickle
21:05.26nDuffjbalcomb: different from where it used to be, that is. And *hopefully* more sustainable. Recent change... can't tell for sure 'till things settle out.
21:05.28jsharpAnd they want a 40 hour project done in 20 hours for $15/hr
21:05.36Dovidhaha
21:05.42Dovidtell them to hire a college kid
21:06.01*** join/#asterisk rfmonk (n=rfmonk@205.241.253.220)
21:06.04jsharpLike those rentacoder projects.  "Build me an ebay".  Budget - $250.
21:06.09justinuheh
21:06.16synthetiqkranz: http://pastebin.ca/46738
21:06.20SkalTurajsharp: tell'em i do it for them in 10hours for $250/hr, and in 20 hours for 110$/hr, or in 40 hours for 50$/hr X) j/k
21:06.42Dovidi charge 50 for configs
21:06.45jbalcombSkalTura funny but we actually do something like that
21:06.48SkalTurajsharp: that's not even best of it! Some guys ask for impossibilities for 30$!
21:06.48Dovidu dont want it , bbye
21:06.49synthetiqthats the dmesg output
21:07.05KranZsynthetiq: you have 2 410p's?
21:07.06jbalcombDovid $75/hr
21:07.20SkalTurajsharp: like this one guy wanted something which fetches bids for keywords --> ehrm, there is a security pic, and doing AI that can surpass that == Not Easy!
21:07.26nDuffneither I or the fellow who hired me (local computer shop owner who wanted to be his own ISP) had any idea of the scale.
21:07.37synthetiqMar 23 16:09:58 WARNING[6263]: chan_zap.c:902 zt_open: Unable to open '/dev/zap/channel': No such file or directory
21:07.37synthetiqMar 23 16:09:58 ERROR[6263]: chan_zap.c:6817 mkintf: Unable to open channel 1: No such file or directory
21:07.37synthetiqhere = 0, tmp->channel = 1, channel = 1
21:07.40jbalcombnDuff agreed. different situations definitely had thier needs
21:07.48synthetiqzaptel and wct4xxp is laoded
21:07.48Dovidam i too cheap ?
21:07.50jbalcombsynthetiq hows about pastebin?
21:07.52KranZsynthetiq: looks like it loaded up correctly
21:07.58KranZsynthetiq: did you run "ztcfg -vv"?
21:08.03synthetiqyes
21:08.10KranZwhat was the output
21:08.19synthetiqline 0: Unable to open master device '/dev/zap/ctl'
21:08.33KranZyou got the latest drivers?
21:08.36jsharpDo you have /dev/zap in your filesystem?
21:08.38opus_does anyone know how I can disable Native Bridging in chan_local?
21:08.40*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
21:08.41synthetiqthey are old
21:08.48KranZd/l the latest
21:08.52synthetiqfuck
21:08.55KranZheh
21:09.05*** join/#asterisk sgrgc (n=torque@unaffiliated/sgrgc)
21:09.12synthetiqso i have to upgrade asterisk and everything
21:09.24KranZnot necessarily
21:09.36KranZwhat version you running now?
21:09.36synthetiqwhat i did is upgrade the kernel
21:09.44synthetiqcvs head from octobe
21:09.46KranZyou need to rebuild the drivers then
21:09.47synthetiqoctober
21:09.50synthetiqi did
21:10.24KranZtry compiling the latest drivers and see if you can get ztcfg to work
21:11.11KranZno need to reboot, just 'rmmod wct4xxp zaptel' when you're done compiling
21:11.14KranZthen modprobe them
21:11.44synthetiqdid and same error
21:12.21KranZwhich kernel and asterisk you using
21:13.39synthetiq2.6.12
21:13.48synthetiqcvs head october
21:13.57qseekwhat is the linux distribution
21:14.02synthetiqfedora 3
21:14.40qseekthere was a thread on this somewhere...u have to modify one of the files before your recompile...a bug in the distribution if i remember correctly..
21:14.43qseeklet me look it up
21:15.12KranZi had an issue on a kernel upgrade right around the .12 release
21:15.17KranZthe driver wouldnt load
21:15.30KranZi cant remember if it was the kernel or the driver wouldnt compile on the new kernel
21:15.33synthetiqwell i went back to the .9 and it stil would load
21:15.43synthetiqmaybe i have to comile under that
21:15.46synthetiqcompile
21:16.04synthetiqill do tha tnow
21:16.07KranZyou should always recompile on a kernel change
21:16.13synthetiqyes i did
21:16.15KranZzaptel and libpri
21:16.40synthetiqyep
21:17.01qseekcd /usr/src/kernels/2.6.9-34.EL-i686/include/linux
21:17.01qseekmv spinlock.h spinlock.h.old
21:17.01qseekwget http://nerdvittles.com/aah27/spinlock.h
21:17.12qseekshutdown -r now
21:17.16qseekthis worked for me
21:19.14*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
21:21.37synthetiqthanks
21:21.58qseekthen rebuild your zaptel drivers
21:23.20*** part/#asterisk sgrgc (n=torque@unaffiliated/sgrgc)
21:27.34[av]banijustinu: ready to look at the app_queue bug?
21:27.39*** part/#asterisk Utah_Dave (n=boucha@0-1pool149-100.nas31.salt-lake-city1.ut.us.da.qwest.net)
21:27.54ManxPoweropus_, you must have missed line 10 of doc/localchannel.txt
21:28.17SkalTurahey
21:28.24SkalTurai need to somehow easily test my asterisk system
21:28.45SkalTurai would need so that i can have multiple cons to * boxes from same computer, and atleast X-Lite doesn't allow me to do so
21:28.51SkalTuraie. i could call myself on the same computer x)
21:30.01SkalTuraany ideas for this?
21:30.01qseekskaltura: you mean registered users to the * from a single computer?
21:30.06SkalTurayup
21:30.15*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
21:30.19qseeku could dial 7777 to simulate an incoming call
21:30.23SkalTuraso i could say connect both test user 1 & 2 from the same computer and call from #1 to #2
21:30.26qseekor is it 6666 :)
21:30.46qseekif u dial 7777 it would call u back on the same xlite client
21:30.55justinu[av]bani: go ahead
21:31.00SkalTuraoh
21:31.04SkalTuraqseek: and if i answer?
21:31.07qseekor u could download another IAX phone and call between the IAX and Xlite
21:31.14qseekit will have 2 way voice path...
21:31.42SkalTuraqseek: pointers to good soft phones i could use for this?
21:32.00qseekwell u could use the asteriskguru idefsk
21:32.06qseekit is a free iax client
21:32.53*** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org)
21:33.26angom_wKranZ: what should I look for in the output of the hdparm -tT device ?
21:33.36cytrakis there a way I can pass to the voicemailman the extension that I'm calling from instead of having to enter my extension number to check on voicemail ?
21:33.57SkalTurai called to 7777 and it immediately hungs up but no return call
21:33.58qseekSkalTura: so you could simulate 3 connections at once on it... line 1 dials 7777 to call line 2; then line 2 transfers the call to that extension
21:34.57*** join/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-66-36.d-ip.magma.ca)
21:35.08ManxPowercytrak, ${CALLERUDNUM} contains the information, "show application voicemailmain" will tell you what you need to know.
21:35.32ManxPowerOf course, that means you can only check your voicemail from your own extension, not anyone else's extension
21:35.44qseekSkalTura: are u just trying to test if your box is setup right?
21:36.04SkalTuraqseek: yeah things like that, general testing, i'm learning to use *
21:36.21ManxPower${CALLERIDNUM} that is
21:36.38qseekSkalTura: then we have to see your extension config
21:37.44*** join/#asterisk GuruDom (n=domiplus@66-202-165-66.rev.knet.ca)
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21:39.46synthetiq2.6.12 does not have sping lock.h
21:39.51synthetiqspinlock.h
21:39.57qseeki am sorry synthetiq
21:40.00*** join/#asterisk ToTo (n=ToTo@host114-166.pool870.interbusiness.it)
21:40.03qseekthat was a redhat solution
21:40.07qseeku have fedora
21:40.23SkalTuraqseek: thanks for the help
21:40.30SkalTuraqseek: got what i wanted tested :D
21:40.46cytrakManxPower: thanks ..
21:41.20qseekhere is a solution link http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
21:42.47GuruDomAnyone have an g729 problem with FC5?
21:43.49MavvieIs John Bigelow on this channel?
21:44.27qseekSkalTura: you are welcome
21:45.43Dovidqseek: why use fc ? CentOS is RHEL without the serial num. much better
21:45.45Dovidbeter*
21:46.39qseekDovid: i wasnt using that , synthetiq was using it
21:46.45GuruDompersonaly i like freeBSD, but when experimenting with the new FC i got a number of issues
21:47.07NuggetAsterisk on FreeBSD is viable, but I'm not sure I'd recommend it.
21:47.11*** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net)
21:47.27GuruDomwhys that?
21:47.43*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
21:47.46Nuggetit's hard to avoid needing zaptel/ztdummy, and that really blows in FreeBSD.
21:48.10*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
21:48.17Nuggetany time you have troubles you'll be met with "dunno, it works fine in Linux"
21:48.26GuruDomwel actually freebsd has zaptel drivers for astersisk
21:48.31Nuggetyes, I'm aware of that.
21:48.36Nuggetand I'm telling you that they blow.
21:48.40Dovidanyone else here use CentOS ?
21:48.46qseeki do
21:49.12SkalTuraqseek: this is crazy, all i've been told that asterisk is hard really hard
21:49.21SkalTuraqseek: so far everything has been qutie easy!
21:49.23jskcrtechDovid: Yes I use centos
21:49.26SkalTuraqseek: in fact stupidly easy
21:49.46Dovidlol
21:49.47SkalTuraqseek: and even crazier is that first PBX system i will be building is a big ass corporate system (8 sites and so forth)
21:50.20jskcrtechDovid: I use centos/debian/gentoo/fedora/rhel/suse
21:50.32*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
21:50.46qseekskaltura: if u need help with configuration dont forget to holler at me :)
21:51.38SkalTurai'm sure there's gonna be problems with hardware
21:52.09SkalTuraso i will be asking 'stupid' questions here
21:52.29qseekno questions are stupid
21:52.35Dovidthe only stupid question is the one that isnt asked
21:52.41qseeki am still trying to figure out this PRI stuff
21:52.59jsharpThere are no stupid questions.  Only stupid people.
21:53.05Dovidwe all learnt, we werent born wit it
21:53.25qseekand still havent figured out how to compile my own apps on asterisk :(
21:53.47Dovidplay, play and play some more thats how u learn
21:54.40SkalTuraqseek: heck, i didn't compile anything for this: a@h ;)
21:54.46SkalTuratho compiling ain't such a worry
21:55.33qseekskaltura: well i am trying to do a custom app
21:55.39SkalTuraoh :O
21:56.33*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
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22:02.40*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
22:02.41a1fahey
22:02.43a1faguys
22:02.46a1fathis is sooooo fucked up
22:02.52a1fahttp://pastebin.ca/46744
22:03.01a1faBT101 phone is trying to NTP the SIP PROXY
22:03.06a1fainstead of NTP server
22:03.09a1fahow odd
22:03.21*** part/#asterisk lzhang (n=lewiszha@67.95.13.46)
22:03.24*** join/#asterisk lzhang (n=lewiszha@67.95.13.46)
22:04.24lzhanghey guys, I'm having trouble with DTMF detection on Zap. If I press say 422, most of the time asterisk thinks I pressed 42 (missing one of the 2's) I tried relaxdtmf, but it seemed to make things worse, not better. Any suggestions?
22:05.23a1falzhang : let me guess
22:05.27a1faGRANDSTREAM?
22:05.32jsharpTwiddle your gain settings.
22:05.49jsharpYou may be clipping your DTMF or not giving the decoders enough.
22:05.50lzhangI am calling in by SIP from a Polycom 300
22:05.59lzhangonto a Zap POTS line
22:06.03*** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net)
22:06.15*** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
22:06.18jsharpOh.  Inband DTMF on the Polycom?
22:06.45lzhangI'm testing using my polycom, although the same issue exists from regular phones and cell phones
22:06.49jsharpIf you're calling out onto the zap line, relaxdtmf has no effect.
22:07.06jsharprelaxdtmf only takes effect if you're calling into the zap line.
22:07.08lzhangI am calling from the outside into asterisk auto-attendant
22:07.32jsharpPlay with your zaptel gain settings, then.
22:08.13lzhangok I will try it
22:08.38jsharpRemember, you have to restart (not just reload) for zapata.conf gain settings to take effect.
22:08.39jbalcomblzhang that is odd, we have the same issues with the lower model polycoms but not on our GXP-2000s
22:09.01lzhangjsharp: shit... I never restarted
22:09.07jsharpWell, there ya go.
22:09.14lzhanglet me try that first haha
22:09.19jbalcomblzhang DTMFinfo = inband, rfc2833, or SIP?
22:09.38lzhangjbalcomb: not sure, where can I find that info?
22:09.47jbalcomblzhang maybe the polycom calls rfc2833 something like 'AVT'
22:10.01lzhanghmm
22:10.03jbalcomblzhang it'll be in your phones config and also in the sip.conf I believe
22:10.31lzhangI see... I will take a look at that too. I guess I'm going to try relaxdtmf again first
22:10.39redondosI just installed Asterisk on gentoo. I was trying it out in Ubuntu. Now, the 'default' context, all it does is "include => demo". If I copy the 'demo' context and give it the name 'redondos' and then replace with  "include => redondos" I can't call my Asterisk box. I just get a "404 not found" in my softphone (twinkle). Debug and verbose are set to 50, though I don't see anything in the console. What went wrong?
22:10.47lzhangthanks a bunch guys
22:10.55*** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84)
22:11.05*** part/#asterisk jgomata (n=jgomata@201.143.139.160)
22:11.48*** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com)
22:13.33a1faya dude
22:13.37a1fafuck cheap SIP phones
22:13.41a1faget a $400 CISCO
22:13.44a1faand be worry free
22:13.49a1faor $100 Linksys
22:13.56a1fajust dont get anything below $140
22:13.58a1faplease
22:14.02[hC]Im having some shitty echo problems with some linksys spa-941's ive bought.
22:14.05a1fasave you a lot of time troubleshooting bullshit
22:14.17a1fa[hC] : lol cheapo
22:14.39[hC]They're just sipura ATAs in a cisco looking shell, I figured they'd be alright.
22:15.26*** join/#asterisk Damin_PDA (n=pocketir@75.sub-70-212-106.myvzw.com)
22:17.27jbalcombI'm getting "apt-get install zlib1g-dev"
22:17.53jbalcombeverything says install zlib-dev but I already have it. Any other packages needed?
22:19.48*** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84)
22:21.09[hC]what is your error message?
22:21.10redondosok, problem solved.
22:21.23redondosabout festival... how can I invoke it specifying a different language?
22:23.46*** join/#asterisk denon (i=denon@synapse.subneural.net)
22:23.46*** mode/#asterisk [+o denon] by ChanServ
22:25.10jbalcomb[hC] aw geez, pasted the wrong thing.. "/usr/bin/ld: cannot find -lssl" is the error message
22:25.39[hC]apt-get install libssl-dev
22:25.39[hC]:)
22:25.57*** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com)
22:26.09*** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net)
22:26.11jbalcombredondos: you really should atleast try to figure stuff out before you ask..
22:26.44jbalcombredondos: festival --help shows "--language <string>" Run in named language, default is english, spanish and welsh are available
22:26.46redondosjbalcomb: about the first question: you are right
22:26.51redondosabout the second question: http://www.voip-info.org/wiki/view/Asterisk+cmd+Festival
22:27.17redondosjbalcomb: it doesn't mention that you can specify a different lang. I know you can do it from the command line, but not from the Festival built-in * function.
22:28.47jbalcombredondos ah, my apologize then. I got your question wrong.
22:29.30redondosjbalcomb: it's ok. maybe I can invoke festival in another way? any ideas?
22:29.46*** join/#asterisk watchy (n=watchy@70.238.56.18)
22:29.55watchyanyone here got experience with poly 501s?
22:29.59*** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net)
22:30.32Damin_PDAyes..
22:30.50jbalcombredondos: I have yet to work with festival in Asterisk. I'm building a new server as we speak and I will be working it in with festival over the next couple of weeks.
22:30.54watchyi'm having issues provisioning mine
22:31.06jbalcombwatchy: Just installed two today but not doing provisioning yet.
22:31.19watchyjbalcomb: well to me its a bitch
22:31.27watchyi got them reading from tftp no issue though
22:31.46jbalcombwatchy: [tk]defender says its the best thing ever and he says it almost every day.
22:31.57watchyi believe they are nice
22:32.19redondosjbalcomb: all right. thanks. festival? it's one of the best text-of-speech implementations I've ever seen, but it still can't be used for production systems IMHO
22:32.22Damin_PDAwatchy. use ftp instead..
22:33.32watchywhy im not having issues with getting files
22:33.36watchyor confs
22:34.45*** join/#asterisk veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
22:35.34*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
22:35.47watchyhow do i reset the phone so it gets all the settings from the tftp?
22:37.08watchywhats reset local config?
22:37.23*** join/#asterisk hfern (n=hfern@h-64-105-50-227.dllatx37.dynamic.covad.net)
22:39.20*** join/#asterisk miztic (n=gerard@rarcoa.com)
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22:47.28SplasPoodwatchy: Menu, Settings, Advanced, Password Entry, Admin Settings, Reset...
22:47.38watchytheres 3 reset options
22:48.07SplasPoodwell what exactly is the problem?
22:48.10SplasPooddid you define the tftp info?
22:48.32watchyit pulls some stuff from my tftp it seems
22:49.07watchylike sip server etc
22:49.34watchybut not like my extensions username/passwords
22:49.46watchyi just did a full reset of my phone lemme look at the logs brb
22:50.14watchynow i'm getting a config file error on bootup on the phone
22:50.14watchywtf
22:55.41Jon335Does anyone know of a unlimited US/Canada long distance provider, that doesn't include a DID?
22:55.45ms345anyone know how to get past  "TFTP size error" when upgrading firmware on a 7960?  It looks like the phone replies back with tftp error #5 after the 2nd packet of the .bin is sent.
23:01.10*** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home)
23:01.20*** part/#asterisk yuta-vcnet (n=asdf@82.71.50.245)
23:02.24lzhangwatchy: does it say invalid bootrom or something of that nature?
23:04.29*** join/#asterisk maxx4life (n=max4life@71-35-210-12.slkc.qwest.net)
23:09.54*** join/#asterisk watchy (n=watchy@70.238.56.18)
23:09.56watchyhrm
23:10.01watchyi guess i broke my poly
23:10.12watchyit says config error then 0x0
23:10.19[hC]digium's new site is very awkward.
23:10.51*** join/#asterisk AKUI (n=FastNet@69.71.137.248)
23:11.58lzhangit's very corporate-ish
23:12.07watchyok im switching to fucking ftp
23:12.08[hC]well
23:12.11[hC]it looks like it could be cool
23:12.16[hC]but its like... disorganized, yet pretty
23:12.34[hC]they broke it apart in a very awkward way
23:12.56*** join/#asterisk HamYaI (n=HamYai@125.24.1.45)
23:13.05[hC]I wonder what the digium font is. looks like its just arial or something
23:13.21watchyatleast you can look at it
23:13.22lzhangthat's not arial
23:13.30watchymy interets so messed up i cant even load it
23:14.11[av]banidigium's new site almost looks like a domain squatter parking site
23:14.32lzhanghehe
23:14.53lzhangit's sort of pretty, but wtf is the blue ghost thing on the left
23:17.20[hC]where the hell did the g729 register binary go to
23:17.47[hC]its not linked via the g729 page any more
23:18.49[hC]found it on ftp. oh well.
23:22.35watchyshould i have my polycoms register?
23:26.24SplasPoodyes
23:28.22*** join/#asterisk Umaro (n=umaro@68.142.142.105)
23:28.31watchyok the phones getting my sip server from sip.conf
23:28.34UmaroHey guys.. anyone know of a digium/sangoma reseller in las vegas?
23:28.38watchylets see if it pulls the line settings
23:29.39*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
23:30.37watchywhat bootrom and sip version should i use?
23:31.03watchy1.6.2 and 2.6.2?
23:32.31watchyhmm
23:32.49SplasPoodwell
23:32.52SplasPoodthe latest you can get
23:33.01*** join/#asterisk _deg_ (n=deg@201.22.48.112.adsl.gvt.net.br)
23:33.02watchyomg
23:33.05watchyit fucking works
23:33.06SplasPood2.6.2 is old, I forget what bootrom was current /w 2.6.2
23:33.09SplasPooderm
23:33.15SplasPood<PROTECTED>
23:33.32SplasPoodI'm running 1.6.5/3.1.3
23:33.42Nodrenwhats a good site that details all the functions for extensions.conf?
23:33.52SplasPoodwww.voip-info.org
23:34.12Nodreni'm already there, i'm not seeing anything good, except a detailed example
23:34.20SplasPoodhehe
23:34.24Nodrenbut it doesnt tell me what everything does
23:34.53SplasPoodsearch harder, most apps/functions have their own page and are pretty well documented
23:34.56*** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
23:35.02_deg_anyone knows why my moh play music fast sometimes?
23:35.13_deg_some mpg123 issue?
23:35.16_deg_asterisk 1.0.9
23:35.35*** join/#asterisk Mr|White (n=mrwhite@69.0.36.217)
23:35.36watchyi need some new polycom firmwarez
23:36.38SplasPoodwatchy: the previous version is available publically on www.polycom.com
23:36.43SplasPoodwatch: 1.6.4/3.1.2
23:37.20SplasPoodI must depart now tho, back on in 45min
23:37.50watchysweet
23:38.45*** join/#asterisk rikstah (n=rick@66.78.236.255)
23:40.02Nodrenwould you guys recommend using a GUI to create my dialplan? or is it better to just write it up?
23:40.20*** join/#asterisk Op3r (n=op3r@202.71.189.90)
23:40.37Op3rany knows call recording?
23:40.45rikstahNodren, you could use realtime that allows the dialplan to be created in mysql, then use some web/app frontend
23:41.18Op3rexten => 301,1,Playback,transfer|skip     ; "Please hold while..."
23:41.18Op3rexten => 301,2,Monitor(wav,${EXTEN}-${TIMESTAMP}-in,m)
23:41.18Op3rexten => 301,3,Dial,sip/301|20|to       ; Ring, 20 secs max
23:41.30Op3rdid I do that correctly?
23:42.10*** join/#asterisk tessier (n=treed@66-162-45-90.gen.twtelecom.net)
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23:54.57*** part/#asterisk Jon335 (n=jon335@ottawa-hs-209-217-66-36.d-ip.magma.ca)
23:55.08*** join/#asterisk ms345 (n=mike_sim@dsl027-163-193.atl1.dsl.speakeasy.net)
23:58.43watchysomeone stab me
23:59.00*** join/#asterisk kink0 (n=k@62.37.205.161)
23:59.05kink0hello
23:59.40kink0I know ussing h323 is a headache, but my peer requires h323... what is your experiences with h323 in Asterisk ?

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