irclog2html for #asterisk on 20050305

00:01.42iceypMar  5 12:59:47 NOTICE[69726]: chan_iax2.c:5405 socket_read: Rejected connect attempt from
00:01.43mesiiceyp: I get the error message "No authority found" ?!?
00:01.44Shidook, mr... what do you have for the user on voip.fast.co.nz ?
00:02.00Shidoiceyp what do you have for the user 2202?
00:02.07iceypShido I have a SIP/2202 extension, I just want to allow guests
00:02.21iceyp<PROTECTED>
00:02.25*** join/#asterisk dallaschumly (dallaschum@c-67-174-74-91.client.comcast.net)
00:02.44mesiI put this in my extensions.conf and called extension 7 in this context:
00:02.45mesi<PROTECTED>
00:02.46mesi<PROTECTED>
00:03.02*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
00:03.07iceyp[guest]
00:03.07iceyptype=user
00:03.07iceypcontext=home
00:03.07iceypcallerid="Guest IAX User"
00:03.10iceypthat right?
00:03.29mesiiceyp: wouldn't I have to register to your server tehN?
00:03.53Shidoiceyp no , then we need to ring guest@voip.fast.co.nz
00:03.53iceypexten => 2202,1,Dial(SIP/2202,20,rtm)
00:03.54Shidonot 2202
00:03.59Shidono
00:04.03Shidook
00:04.04Shidofind
00:04.10Shidothen we ring 2202 at the end of the guest
00:04.11Shidofor example
00:04.16iceypahh right
00:04.21Shidoiax2/guest@voip.fast.co.nz/2202
00:04.26iceypmmmmm
00:04.27Shidoso the entire line would be.....
00:04.50Shidoexten => whatevernumberyawant,1,Dial(IAX2/guest@voip.fast.co.nz/2202)
00:04.55Shidoand errr
00:05.00Shidowhat do u have for codecs set in general
00:05.02iceypso how do i allow people to dial EXTEN@voip.fast.co.nz
00:05.10mesiiceyp: Got my message?
00:05.10Shidocuz in the guest entry you have pasted u dont have anything set
00:05.31mesiiceyp: I think I got your voicemail box :-)))
00:06.14iceypsweet
00:06.20iceyplet me check the phone
00:06.21Shido:)
00:06.28mesiiceyp: Got my message? :-)
00:06.29Shidoleft a vmail for ya iceyp
00:06.47mesiYeah, me too. There's an answering machine there on 2202 :-)
00:06.50iceypargh someone vacumed this morning and unplugged my phone
00:06.51iceyplol
00:06.58iceypthanks guys
00:07.03Shidoplugged it back in?
00:07.09mesi*LOL*
00:07.10Shidoring...
00:07.27Shidocodec ur sounding -
00:07.30Shidogeneral stanza
00:07.35Shidoand set a codec for general or guest
00:07.44mesiOk, Shido first ;-)
00:07.49Shidodisallow=all
00:07.50Shidoallow=gsm
00:07.53Shidoallow=ilbc
00:07.56Shidoor something other than ulaw
00:08.01Shidothen issue a reload
00:08.20Shidomr. roboto man ;)
00:08.30mesiOh dear, my config is a mess! I would have to clean that up some time.
00:08.50Shidono no, only iceyp needs to change it in the guest user
00:08.55Shidothen issue a reload
00:09.02iceypis this set right ... bandwidth=low
00:09.06Shidoeveryone else doesnt have to set that... because his will choose the codec
00:09.11Shidothats fine
00:09.12mesiShido: Anyway, my config still is a mess. ;-)
00:09.19Shidowhat I need you to change is the guest user in iax.conf
00:09.21Shidoadd disallow=all
00:09.23Shidoallow=gsm
00:09.25Shidoallow=ilbc
00:09.39Shidootherwise ulaw will be used... dont want that...
00:09.57owhI'm looking in the source for dial and I can see where it plays the announce message, but I'm stuffed if I understand how I can play to the caller, not just the callee.
00:09.59iceypShido  i added it to general not to guest user
00:10.01iceypis this ok?
00:10.02Shidook
00:10.04Shidothats fine
00:10.09Shidoreloaded?
00:10.11iceypand what about bandwidth=low
00:10.15Shidothats fine
00:10.16Shidoleave it
00:10.48mesiHm... I don't really understand how iceyp set up this guest user.
00:10.49Shidowhat are you running asterisk on
00:10.52Shidoiceyp ?
00:11.02*** part/#asterisk eKo1 (~bernd@63.245.57.70)
00:11.06Shidoits just a user
00:11.48iceypp4 w.8 gig
00:11.51iceyp512megsw ram
00:11.57iceypp4 intel not celeron
00:12.07iceypno
00:12.22*** join/#asterisk TSCHAK (~thomas@69-161-192-45.clspco.adelphia.net)
00:12.27mesiSo [guest] is only added to iax.conf, right?
00:12.28*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
00:12.37mesiIt's not a unix user.
00:12.38Shido?
00:12.46iceypbrb
00:12.48iceypon a call
00:12.57mesiiceyp: I want to make a voice call to your box.
00:13.00Shidook
00:13.13mesiPlease tell me when I can do this, and have you received my voicemail? :-)
00:13.44Shidoexten => 31337,1,Dial(IAX2/guest@shido.gotdns.org/3000)
00:13.56Shidoin iax.conf you just add a [guest] user
00:13.56mesiShould I call you?
00:14.00Shidowith a type=user
00:14.08mesiYes, got that.
00:14.14Shidocontext=whatevercontextyouwannalookfortheextension in
00:14.17Shidono secret
00:14.24Shidoand a codec if not set in [general]
00:14.26Shidoand issue a reload
00:14.44mesiRight, it is not so complicated. I am just not used to it yet :-)
00:14.45*** join/#asterisk jjhall (~chatzilla@24-116-128-106.cpe.cableone.net)
00:14.50Shidothen in the dialplan (extensions.conf) the extension u take on the end needs to be available in the context you provided in the guest user in iax.conf
00:14.58Shidoor u can use the s, if u wanted to
00:15.32mesiShido: I am using ordinary phones from my pstn and they are connected to my computer. It is always a hassle to dial such numbers with them.
00:15.45mesiI would have to add every number I use to my extensions.conf.
00:15.53Shidono
00:15.56Shidou can make a macro
00:16.12Shidoor match a pattern u dial
00:16.13mesiBut there are many different hosts.
00:16.17Shidoto reach whatever extension u want
00:16.22Shidostill not a problem
00:16.28mesiHm...
00:18.27mesiThere's a simple problem with such prefixed numbers: From my pstn  phones I have to call the asterisk computer, it is number 27. So this is the msn I call an  not any wildcard number from extensions.conf. :-\
00:19.04iceypsorri shido
00:19.06iceypmum called
00:19.39iceypon voicemail
00:19.42iceypmesi i got it
00:19.49mesiiceyp: great! ;-)
00:20.00mesiiceyp: Could you understand it? Or was it too fragged?
00:20.18iceypclear as
00:20.26mesiiceyp: Can i call you live now?
00:20.41tuxinator_linux2If my corporate location has a subnet of 10.0.1.0/24 and each remote location has a subnet of 10.0.2.0/24, 10.0.3.0/24 etc...  Would I need an * or proxy server at each location?  Thw wan would be created using MPLS found at http://www.netifice.com/products/siteToSite/MPLSPrivate.html
00:22.54mesiEverybody dead?
00:22.59tuxinator_linux2not me
00:23.06*** join/#asterisk tzafrir_home (~tzafrir@bzq-179-40-134.cust.bezeqint.net)
00:23.06*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
00:23.24tuxinator_linux2~dead
00:23.25jbotyes :(
00:23.31Shidodying?
00:23.32Shido:)
00:23.37Primertuxinator_linux2: as long as valid routes exit, there's no reason for either
00:23.42Primerexist, even
00:23.43iceypthanks so much greg
00:23.44tuxinator_linux2~dying
00:23.45jbotit has been said that dying is The process by which you turn plain white material into colored material.
00:23.52ShidoNO
00:23.59mesiiceyp: I call you on 2202, ok?
00:24.05Shidoyou do not need an asteriskbox at each location...
00:24.09tuxinator_linux2jbot: not that kind of dying
00:24.11walnuck~asterisk
00:24.12jboti heard asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org
00:24.17iceypmesi sure
00:24.18Primer~vagina
00:24.19jbotsomebody said vagina was something that i dont have but i like to suck, SUCK PUSSY YEAH!!!
00:24.24walnuck~asterix
00:24.25jbotasterix is, like, a fearless fighter of the Roman tyranny
00:24.38walnuck~obelix
00:24.43Shido~digium
00:24.45jboti heard digium is http://www.digium.com
00:24.45tuxinator_linux2bad jbot, your mother would be ashamed
00:25.11iceypgrr, shido seems to still cut off on voicemail, but i think its a bug in bsd *
00:25.17iceypbecause i cant run AGI's
00:25.25iceypthey also cut off
00:25.30walnuckthat's disgusting..someone should moderate the bot
00:25.49mesiiceyp: ok, right. There's still your answering machine :-(
00:25.50Shidohrmm
00:25.53tuxinator_linux2Primer, could you explain what a "valid route" is?
00:26.11*** join/#asterisk Nethab (~Anon6069@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
00:26.20tuxinator_linux2~walnuck
00:26.26*** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com)
00:26.30Primertuxinator_linux2: you know, that thing that allows you to say, ping a box on another subnet from the subnet you're on
00:26.36tuxinator_linux2okay
00:26.40tuxinator_linux2thanks, primer
00:27.18KalD|Workanyone know which radioshack p/n w/ work for an IAXy power supply?
00:27.49tuxinator_linux2Primer, I may have thousands of locations, if I did a HA * cluster of some sort I think it could handle it.
00:27.55*** part/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-13-25.w81-248.abo.wanadoo.fr)
00:28.05Shidotip positive
00:28.30Shidomeasuring 5.0mm outer-diameter and 2.5mm inner-diameter
00:28.46Shido<PROTECTED>
00:28.59Shido9V DC
00:29.17mesiShido: I know a solution for my problem. When  calling from pstn phone on msn 27 I can define an extension 27 which is this callthrough example from the webpages. But it won't use Dial but Goto a different context calling an extension with the entered number :-D
00:29.26Shido<PROTECTED>
00:29.45sivanathe monkeys fly at noon
00:30.15Shido273-1771 and 273-1715
00:30.37tuxinator_linux2~monkeys
00:30.37jbotThis problem, like many others in the computer industry, can be solved by the application of monkeys.
00:31.24ionixHey guys, I would like to start a IAX/SIP termination server. I am currently working for a big CLEC will killer rates for long distances. What is important for you?? DIDs ? North American rates ? International rates ? Customer service ?
00:31.39ionixI think I can be 50% less than nufone ;)
00:31.54ionixwe have 4X OC48 across the world...
00:31.55Shidoionix, go for it
00:32.05sivana1. customer service 2. North America technically inclused Mexico
00:32.08jjhallI'm having some issues with MOH.  When it plays back, it has a very electronic distortion sound, and chops as well.  I've googled the mail list, and ensured my version of mpg123 is correct (0.59r.)  Any ideas?
00:32.17sivana3. Good rates
00:32.37ionixsivana: I meant Canada/USA for good rates (Blame the mexico CLEC :) )
00:32.40buddahi completely agree with 1. customer service
00:32.57Shidothat will be your biggest issue.
00:33.01ionixwhat kind of customer service ? Email or phone ?
00:33.04Shido24/7 customer service
00:33.09sivanaany provider that is promising North America but not Mexico is in the wrong :)
00:33.17buddahnothing is more agrivating then purchasing service from a company, and not getting good support. like when something isnt configured correctly and nothing is done to resolve it
00:33.18iceypme
00:33.20ionixShido: We already have 200 employees working in our call centre
00:33.22sivanaat 24 hr email response
00:33.26sivanaat least
00:33.35Shido24/7? ionix
00:33.39ionixShido: yeh
00:33.42Shidohow many of them are trained in asterisk?
00:33.51Shidosip configurations
00:33.54Shidorouter configs
00:34.04Shidonat environments
00:34.06ionixShido: 0 :) This is customer care...
00:34.13ionixThe NOC is trained though...
00:34.27ionixbut if we offer the service, I could take 10 people and train them
00:34.28sivana1. customer service 2. rates both US/CA, international 3. DIDs
00:34.38Mocsivana so 1800 DID ? ;)
00:34.52*** join/#asterisk billb-uk (billb@82-32-200-114.cable.ubr06.newt.blueyonder.co.uk)
00:34.57billb-ukhey all
00:35.05sivanahey Moc, our service is confirmed
00:35.10sivanaand operational
00:35.15Mocwoohoo ;)
00:35.35sivanajust working on an automated process now
00:36.03Moceasy, give me ssh access, you wont have to do anything ;)
00:36.04*** join/#asterisk Sedorox (~Sed@Neptune.client.wlgrv.pa.sed6.net)
00:36.44sivanaI mean with provision on my carrier's side
00:36.46sivana:)
00:37.57billb-ukim have problems with dtmf tone recognition with one of the carriers my company uses - i hve ser sitting in front of asterisk, and finding that if i set the dtmfmode inband it works for people from PSTN => SIP, but one the rtf2833 method works for SIP => sip... anyone else had these issues?
00:39.10Shidobleh
00:39.13Shidoset it for rfc2833
00:39.16Shidonot rtf2833
00:39.23ionixhehe
00:39.29Shidothen unload chan_sip.so and load chan_sip.so
00:39.39billb-ukshido - sp mistake - surely you can realise that
00:40.16ShidoIm not bitching :)
00:40.22Shidouser error is prominent
00:40.26Shido:)
00:41.03sivanaya... who knows
00:41.05billb-ukhmm.. well anyway, can asterisk do an auto detection?
00:42.22*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
00:45.20rvhii am using cvs stable. how do i find out if there is any bug fix after my last checkout?
00:46.24Hmm-workread the changelog
00:46.35*** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com)
00:46.53rvhigot to have a better way...
00:47.06JunK-Ybillb-uk: auto detection of ?
00:47.52rvhithere is no way to show a file has been changed?
00:48.29JunK-Yrvhi: huh?
00:49.31rvhiif there is a small bug fix in cvs stable, i want to be able to find out the change
00:49.47rvhii use some other patches
00:49.58jjhallAnyone here familiar with music on hold?
00:50.13rvhiif there is just a file change, i can manually do it, without going through the whole patch process again
00:50.26JunK-Yrvhi: go into the bug tracker and see whats has been ported to stable.
00:50.40JunK-Yrvhi: run head btw.
00:51.05rvhiit is for production, wouldn't head be a problem?
00:51.31rvhii kept seeing broken/crash/not-working on the mailing list
00:51.44JunK-Yrvhi: im running head on my prod sites since more then 2 months.
00:52.32billb-uk<JunK-Y>: the dtmf tones
00:53.17rvhiwhat do i get extra in head?
00:53.44rvhii am thinking about using ast_data to simulate realtime db
00:53.50JunK-Ybillb-uk: sure * handles dtmf.
00:54.01rvhiother than that, any benefit?
00:54.08JunK-Yrvhi: head is a much more powerful code.
00:54.28rvhipowerful, in term of ?
00:54.39rvhifeature-wise, difference?
00:55.05rvhibtw, cvs diff gets this error: Disk quota exceeded
00:55.32JunK-Yrvhi: and df -H gives ya what?
00:55.46JunK-Yfeatures, yes.
00:56.19rvhilots of space on my local drive
00:56.39rvhican you list a few features?
00:56.48JunK-Yhttp://bugs.digium.com
00:56.48*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
00:57.07PTG123hey i am getting an out of g729 licenses when i try and make a call
00:57.14PTG123i have 3 licenses, and no one is on the phone currently
00:57.23PTG123i just made 3 calls though that worked, and now none work
00:57.26PTG123anyone have any idea?
00:57.29PTG123~seen kram
00:57.31jbotkram is currently on #asterisk.  Has said a total of 22 messages.  Is idling for 3h 17m 18s
00:57.31Shidothat sucks
00:57.32Shido:)
00:57.55PTG123g729 is the best codec ever :)
00:58.00JunK-Ystevekstevek: why?
00:58.10PTG123one of my only pet peeves with iaxclient :
00:58.12stevekstevekI hate patents.
00:58.28PTG123yah but g729 does the job super well, its worth the $10 a channel by far :)
00:58.36*** join/#asterisk Newbie___ (some@218.111.221.237)
00:58.47stevekstevekIt isn't the $10.  It's the principle.
00:58.57stevekstevekand the annoyance, as you can see.
00:59.03Nethabof paying for compression of my own voice?
00:59.12Luhiwui'm having problem trascoding from ULAW/GSM to G729/H323, anyone knows if it should work?
00:59.39stevekstevekmy point exactly.  It seems that 50% of people's problems seem to come from G.Patentedcrap
00:59.41*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
00:59.44Luhiwui mean ULAW/IAX :)
00:59.45ariel_Luhiwu, your going from one server to another one
00:59.50JunK-Ystevekstevek: ive to agree a bit with ya.
00:59.57iceyphow much bandwidth does G729 use? <-- that is the one u have to purchase right?
01:00.00stevekstevekanyway, my spewing is done.  Goodnight :)
01:00.08JunK-Ystevekstevek: cya.
01:00.10stevekstevekuse speex ABR at 8kbps.
01:00.15*** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au)
01:00.20JunK-Yiceyp: goto wikis
01:00.22iceypand its good quality? like gsm etc
01:00.22ariel_g729 is about 8k but you have to add the ethernet overhead
01:00.23JunK-Yall these infos is there.
01:00.32PTG123its the principle that someone wants to make money for their hard work? :)
01:00.38iceyp8KB or 8kb
01:00.47PTG123iceyp: 8kbps
01:00.53PTG123lower case
01:00.55JunK-Y8 kilobytes
01:00.56PTG123yes it works great over a modem
01:00.57Luhiwui'm having problem trascoding from ULAW/IAX to G729/H323, anyone knows if it should work? it does translate if i use SIP but not if i use H323
01:00.57PTG123:)
01:01.03PTG123its kilobits
01:01.11iceypso u can make like 4 calls on a 64K line?
01:01.11JunK-Yya bits
01:01.19JunK-Yin french, its fucked up.
01:01.22PTG123steve: speex has to many issues, and compression doesn't come close to g729
01:01.31PTG123ice: in theory
01:01.33PTG123ice: not in practice
01:01.38iceypk
01:01.38PTG123ice: 3 calls would be safe
01:01.45JunK-Ydue to overhead.
01:01.49PTG123right
01:01.53PTG123your first call has the most overhead
01:01.57PTG123think its like 12-14kbps
01:01.58iceypis the quality good tho? is there a trial channel we can use?
01:01.59PTG123for the first call
01:02.01ariel_speex  take more cpu power from the test I was doing.
01:02.01PTG123coule be as high as 20
01:02.11PTG123iceyp: better quality then any other codec
01:02.17JunK-Yim using ulaw all the time :)
01:02.22PTG123i like the quality better then ulaw even, b/c you are less prone to network issues
01:02.28PTG123which i have with ulaw and not g729
01:02.30PTG123due to bw it uses
01:02.30iceypPTG123 where do i purchase it, and how do you laod it?
01:02.42PTG123icey: www.digium.com and when you buy it they send you install instructions
01:02.43ariel_digium.com
01:02.44JunK-Yiceyp: www.digium.com
01:02.54PTG123really simple to get working
01:03.01PTG123you need 1 license per simeltaneous call
01:03.02iceypand whats stopping me from sharing that codec with others? not sayoing i would
01:03.07JunK-Yariel_: btw im reading docs related to SS7, really cool.
01:03.11PTG123its licenses based
01:03.14PTG123node locked to your machine
01:03.27iceypso there is no way to hax it? :)
01:03.30PTG123everytime you start it, it contacts digium and sends them your mac address, and asks permission to use it
01:03.33PTG123iceyp: always a way :)
01:03.41PTG123iceyp: but not super easy
01:03.44wildcard0i have one that doesn't do that
01:03.46PTG123and for 10 why waste the time
01:03.57wildcard0i pay for licenses and just use them
01:04.24iceypyeah, what if their server goes down?
01:04.30wildcard0ya exactly
01:04.31iceypthen my calls stop?
01:04.36PTG123it may cache the license not sure
01:04.38PTG123no
01:04.39JunK-Ycvs server: include/asterisk/channel_pvt.h is no longer in the repository
01:04.41JunK-Ywoohoo!
01:04.43PTG123b/c i think it does it only when your server starts
01:04.55wildcard0so you can't restart your server if theirs goes down
01:05.15iceypok
01:05.20iceypis it under software?
01:05.31iceypgot it
01:06.55ariel_JunK-Y, yes it is cool but it does not play nice with asterisk at this stage.
01:07.03iceypok... does nufone and voip providers support g729 gernerally?
01:07.14*** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net)
01:07.24JunK-Yhttp://www.voip-info.org/tiki-index.php?page=Asterisk+G.729+Licensing
01:07.53iceypthey just let u download the code
01:07.54iceypc
01:08.27mesiQ: I have got a computer and an isdn card installed. How can I Dial() an asterisk "*" ?
01:08.58ariel_mesi, you don't have a phone?
01:09.22NukemizerI hate to ask but I am hoping to find out which version of RH is best to use,FC2 or FC3 for running asterisk ?
01:09.41ariel_Nukemizer, which one are you better with?
01:09.41mesiAriel: I have 8 phones ;-) But I want my computer to dial one of my ordinary phones and activate the speaker to immediately play voice :-) Therefore I have to send it *48*
01:10.27NukemizerI use Mandrake.. so neither one, but I need to run the test for Digium tech support
01:10.31ariel_mesi, that is hard unless the phone is able to auto answer.
01:11.12ariel_Nukemizer, FC2 or 3 are ok. But I tend to like the FC2 better at this stage.
01:11.42mesiariel: it is able to autoanswer as long as you can send "*48*" on the line :-)
01:12.01mesiAh! DTMF would do it!
01:12.02Nukemizerariel: is that because of stability or more stuff works with * ?
01:12.49Nukemizerariel: is FC2 2.4 Kernel versus 2.6 on FC3 ?
01:14.52PyroSteveok guys ... asterisk non-related plan
01:15.24Nethabi currently use fc2 but that's because when i built it fc3 wasn't out yet
01:15.33rvhido i have to get timing source somewhere to run *?
01:16.00rvhiif i don't load pri card, * won't play a message
01:16.01Nethabyou only need timing source for pci line cards either fxo or pri or isdn
01:16.13Nethabif your pure IP you don't need it
01:16.43rvhiif the dialplan is 1 answer, 2 playback and 3 hangup
01:16.50rvhiit didn't play a message
01:16.53PyroStevebuilding A key system <---> key system ata <----> (fxo)spa3000(eth) <----> vpn/ip/internet <----> (eth)spa3000(fxo) <---> key system ata <----> key system for building B
01:17.22rvhiit starts working if i load the driver for pri card
01:17.34PyroStevehopefully that will allow one extension to dial another extension at building b
01:18.00PyroSteveanyone tried that before ? Maybe even with asterisk ?
01:18.27Nukemizerariel_ , Nethab: Thanks for input
01:20.30*** join/#asterisk kks (~kks@203.115.210.253)
01:20.58ariel_Nukemizer, any time.
01:21.11Nethabnp
01:21.29Newbie___put * behind a router, and it gives u hell
01:21.44ChujiPyroSteve : That should work fine
01:21.55tzangerNewbie___: uh, * is always behind any number of routers...
01:22.32Newbie___tzafrir: i got 2 *, 1 behind a router and the other is not
01:22.40Newbie___never have any problem with the 2nd one
01:23.02rvhidoes the router do NAT?
01:23.19Newbie___i try using nat=yes and no
01:23.51Newbie___funny thing is, i am able to use softphone with 2nd * even if the port is not 5060
01:23.54Chuji~sip+nat
01:24.12Chuji~nat+sip
01:24.13jbotsomebody said nat+sip was just fine if you have the SIP client behind NAT and Asterisk on an official IP.....
01:24.40Newbie___i am trying internet-> * ->router -> PCs
01:25.01Shidoturn on nat processing
01:25.04Shidoon the phones
01:25.04ChujiThat should be no problem then
01:25.07Shidoand the router
01:25.29Newbie___on the router it self, there is no nat
01:26.31tzangerbehind a router != behind NAT
01:26.39tzangerI have *no* issues with double-nat
01:26.54tzangermind you I don't use SIP and I port-froward udp/4569 to the right box on both sides...
01:27.54Newbie___and damn, whitebox linux upgrade site is down, wtf
01:32.33*** join/#asterisk kks (~kks@203.115.210.253)
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01:45.51PyroSteveChuji: Well I just relized that the configuration would only allow one call between systems at a time
01:46.25PyroSteveChuji: I think I can do asterisk boxes with T1 cards in each asterisk box, as well as, each key system
01:46.55PyroSteveChuji: that way 23 calls can be made at a time
01:50.46*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net)
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01:57.11nix000anyone running * on the wrt here ? i am trying to do the same.
02:02.53Primerwrt?
02:03.01Primeras in, wrt54g?
02:03.27Primeras in, really really underpowered mips based router made by linksys (aka cisco)?
02:03.58*** join/#asterisk coolhp (~coolhp@mtl149-99-187-66.dedicated.sprintdsl.ca)
02:04.02coolhpGood evening all !
02:04.18coolhpAnyone have any idea of what this is : Don't know what to do if second ROSE component is of type 0x6 ?
02:04.37cbachmannix000 I've played with compiling asterisk for it today some.
02:05.04ta[i]ntedPrimer i swear they try to squeeze everything on the wrt lol
02:05.07*** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net)
02:05.17Primerta[i]nted: that just sounds wrong
02:05.36ta[i]ntedwart
02:05.38ta[i]ntedlol
02:05.53ta[i]ntedit's an interesting idea
02:06.00ta[i]ntedwifi hotspot pbx
02:06.47Primerthat can handle what, 1 call, if that?
02:06.51cbachmanI'm more interested in running it as a cheap router/nat/sip->IAX conversion box.  For the price I paid, wireless is a bonus (turned off at the moment anyway)
02:06.55Nukemizerwifi telco can only degrade further the quality of voice communication
02:07.22ta[i]ntedcbachman u can't afford a 40$ p2 box?
02:07.24cbachmanJerJer indicates he's had at least three calls running on his, if I recall.
02:07.30Nukemizerisn't cell service bad enough ?
02:08.01cbachmanta[i]nted, I have P2's and P3's I can nab from work, they are just more heat and noise in my small place.
02:08.25ta[i]ntedwhat distro can u put on the wrt
02:08.26cbachmanbesides, I've always had a fondness for running small embedded devices, and stretching their limits.
02:08.32Primeropenwrt
02:08.40Primerwww.openwrt.org
02:08.50ta[i]ntedlike primer said.. it will probably support one call
02:08.50Primercbachman: you must be a masochist!
02:09.12ta[i]ntedand if something else is chomping cpu cycles on it, ur one call will sound like ass
02:09.19ChujiPyroSteve : Yes, that is a much better solution
02:10.39cbachmanI look at it as a way to learn more about asterisk
02:10.44ta[i]ntedside question: how do u people find time to work on voip projects... my g/f is always complaining
02:11.09Chuji~astriholics
02:11.10jbotastriholics are people that spend every waking hour working with Asterisk. They need a life!
02:11.19ta[i]ntedlol
02:11.26ta[i]ntedastribots
02:13.34Chujita[i]nted : A lot of people here do Asterisk for a living
02:13.49BrianR___I got an extra fxo card at work.. I should set up an asterisk box at home..
02:13.50ChujiNot a very good living, but a living none the less
02:14.40BrianR___I'm doing an asterisk deployment at work...
02:15.01BrianR___Could make a pretty decent small business as an asterisk phone system installer...
02:15.05TrepaliumI'm just trying to learn about * before I try deploying it work.
02:15.07BrianR___Excellent markup on the software :
02:15.08BrianR___)
02:15.36BrianR___Doing an asterisk and norstar key system integraiton.
02:15.42BrianR___err.. integration
02:15.59ariel_BrianR___, there are a few of us just doing that. Trying to make a living out of asterisk installations.
02:16.36Newbie___and is not easy living
02:16.46ariel_but since asterisk is not paying the bills I am also doing windows networks.
02:16.58ChujiIt looks much more appealing on the outside
02:17.22*** join/#asterisk modulus_ (modulus@rm-f.net)
02:17.23ChujiOnce you commit, it's a lot of work to be profitable selling *
02:17.28ChujiDo it cuz you like it
02:17.36Chujiand hope it keeps food on the table
02:18.03Newbie___i do it to pay my loans
02:18.46ta[i]nteddo u guys know where i can find a list of international countrycodes, prefixes, and prefix types?
02:19.02*** join/#asterisk Sedorox (~Sed@2001:4830:2018:a:20f:eaff:fe91:3778)
02:19.04ta[i]ntedi'm trying to do least cost routing but i can't find such a list
02:20.04Chujita[i]nted
02:20.04*** join/#asterisk kks (~kks@203.115.210.253)
02:20.07Chujihttp://www.numberingplans.com/
02:20.14ariel_ta[i]nted, there was a list posted on the user list a few days ago.
02:20.25Chuji~e164
02:20.27jbotrumour has it, e164 is the numbering scheme for many telephone numbers, e164.org maps them into dns
02:20.32ta[i]ntedon digium?
02:20.52ta[i]ntedwow it's not free? shessh
02:23.10ariel_enum is not a bad Idea. So is dundi if they get all the kinks fixed.  Using them will save you some money. But your calls might not be as clear in some cases.
02:27.39TrepaliumWhat kind of information do you need to hook a TE110P to a T1 from the local phone company, and is it difficult to do?
02:28.12*** join/#asterisk Sedorox (~Sed@2001:4830:2018:a:20f:eaff:fe91:3778)
02:28.17BrianR___helps if you know the phone number and/or DID ranges that go with the PRI
02:30.13ariel_Trepalium, Well you need to know if it's a normal T1 with e&m wink or an PRI they both will work you just have to ask them.
02:30.30BrianR___aah yes. that would be helpful too.
02:30.36BrianR___Get PRI signalling if you can..
02:32.27TrepaliumHmm..  Okay, good to know.
02:32.31filekram: VON!!!!!!!!
02:37.03parintegrated T1 or PRI, which is better to go with?
02:37.13ariel_PRI
02:37.27BrianR___PRI signalling is less aggrivatin
02:37.27BrianR___g
02:37.36par'k thx.
02:38.24ariel_~seen pfn
02:38.25jbotpfn <500@adsl-69-107-210-254.dsl.pltn13.pacbell.net> was last seen on IRC in channel #asterisk, 11d 44m 48s ago, saying: 'only with iax'.
02:41.57*** join/#asterisk dfunnell (~dfunnell@port-222-152-55-43.fastadsl.net.nz)
02:42.53dfunnellLooking for some help configuring a TDM400P, anyone online who can help?
02:43.32*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
02:44.21ariel_dfunnell, yes but I am trying to feed my child right now. In about 15 I should be back.
02:45.52dfunnellSounds a bit more important than my TDM!
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02:47.09Nuggetheh
02:52.10dfunnellThanks ariel, look forward to hearing back from you.  Description of problem is as follows:
02:52.16*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
02:54.02dfunnellHave built and installed Asterisk with latest release on RHEL 3.  Have TDM400P with 4 x FXS ports (forget proper part number) and quad port Eicon Diva Server Card.  Everything is running fine, including Eicon card and CAPI, but cannot seem to use TDM.
02:55.31scrubbwhy not RHEL4?
02:55.37dfunnellRunning 'modprobe zaptel' and 'modprobe wcfxs' both give no response, which I believe means that the driver can talk to the card?
02:55.53scrubbwhat does zttool show?
02:56.02dfunnellNot RHEL4 as bought RHEL3 just before RHEL4 came out ;-)
02:56.24scrubbits a subscription, if you are registered with RHN you can download and upgrade for free.
02:57.18dfunnellReally?  Might keep that in mind, but have to get this box built in a hurry, so may be able to hit customer up for on site upgrade later!
02:57.23scrubb:-)
02:57.30scrubbdid you try zttool?
02:57.36dfunnellHave not seen zttool before.  Is recognising the card, shows it as unconfigured.
02:57.44ariel_dfunnell, you did download and compile the zaptel drivers?
02:57.45dfunnellStarting to think I may have missed a step?
02:58.08dfunnellHi ariel, yep downloaded and complied zaptel successfully.
02:58.11scrubbdid you configure the /etc/aptel.cfg
02:58.15scrubbzaptel.cfg
02:58.38ariel_zaptel.conf in the /etc/ and zapata.conf in the /etc/asterisk directory.
02:58.42Sedoroxdfunnell: stupid question.. did you compile the zaptel.. before asterisk.. or after?
02:59.50dfunnellsedorox - not stupid q at all, need to check my notes, compiled in order suggested on ww.voip-info.org, will come back on that one.
03:00.19Sedoroxok
03:00.44SedoroxI know I had problems when I put in my zaptel.. because I did it after I installed asterisk.. so... just giving my $0.02
03:00.47*** join/#asterisk Zilas (~root@c-24-30-75-206.mw.client2.attbi.com)
03:01.36dfunnellariel - did configure /etc/zaptel.cfg, added line saying fxoks=1-4... sound right?
03:01.48ariel_Sedorox, even if you compile them after zttool will still work. just you can't use them in asterisk.
03:01.55Zilashello
03:02.08dfunnellariel: ALso tried fxsks=1-4 just in case I had it backwards!
03:02.20ariel_dfunnell, no it's zaptel.conf and there are more then just fxoks=1-4
03:02.27Zilascan anybody help me with one stupid Q?
03:02.48GoshenZilas: ok, one
03:02.49TrepaliumThere are no stupid questions, only stupid people.
03:02.57Zilas:)
03:04.01Goshenoh wait..you already asked it... :)
03:04.27dfunnellariel: after zapata.conf:found on console, get message saying:
03:04.28dfunnellMar  5 16:05:27 WARNING[3402]: chan_zap.c:771 zt_open: Unable to specify channel 1: No such device or address
03:04.40Inv_arpZilas: just ask ques,  no reason to ask to ask
03:04.41dfunnellA couple more messages then bombs out.
03:04.54Sedoroxariel_: true.. he said it found ut.. just unconfigured...
03:05.13dfunnellOk, I need to configure card first?
03:05.19ariel_dfunnell, ok now it's time to edit the file and put the correct settings
03:05.25dfunnell;-)
03:06.32ariel_I have a TDM11b which is a 1 port FXS and 1 port FXO. I am going to pastebin.ca my settings so you can use them to configure your card.
03:07.43dfunnellSounds perfect.  What is this pastebin.ca magic you speak of?
03:08.38dfunnellAlso, do I need to bounce box to get config to load, or just try re-starting asterisk?
03:11.22ariel_dfunnell, http://pastebin.ca/6880 it's just samples
03:12.14ariel_you are going to have to stop asterisk then service zaptel restart if it's loaded correctly.
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03:17.30*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca)
03:17.31dfunnellAriel: Excellent, thanks, modifying conf now.  service zaptel restart reports 'zaptel: unrecognized service', but lsmod lists it as being loaded.  Possibly misconfigured?
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03:26.29dfunnellariel/sedorox: Can't find notes regarding which was installed first - Asterisk or Zaptel.  Does Asterisk need to be re-built if they were built out of order?
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03:31.29ariel_dfunnell, yes
03:32.25ta[i]ntedis there a way to play MOH while executing dial application?
03:32.44dfunnellariel: Ok.  Any need to back up configuration before make install, or are settings, etc. retained?
03:32.49ta[i]ntedso instead of ringing user hears soft music
03:33.12ariel_there retained as long as you don't do make samples
03:33.40drumkillata[i]nted: the 'm' option to Dial
03:33.41Inv_arpdfunnell: zaptel is not a redhat service
03:33.47ariel_But have you modprob zaptel , modprob wtdfxs , then ztcfg -vvvv
03:34.15kramwow
03:34.30drumkillakram!!!!!!!!!!!!!!!
03:34.33kramrussell!
03:34.36drumkilla:D
03:34.41kramhey did you know aiferion has you in their credits?
03:34.50kramor aeferion however it's spelled
03:34.58drumkillanope - sure didn't
03:35.14kramyah
03:35.21kramyou're a core contributer to their fork
03:35.25kramjust FYI
03:35.25kram:)
03:35.37drumkillaI have never even heard of that, heh
03:35.40dfunnellAreil: Oh, I see.  Modprobe wtdfxs, then ztcfg -vvvv, before installing Asterisk?
03:36.05kramwhat's wctdfxs?
03:36.17drumkillakram: it's a little jewel I added the other day
03:36.26*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
03:36.29drumkillait's kind of complicated, you might not understand  :p
03:36.33ariel_kram, bad spelling
03:36.36kramoh :)
03:37.42Inv_arpsimple 3 way conferencing = meetme?
03:37.45Sedoroxdfunnell: really youjust have to have it installed on the system so it finds the libs
03:38.33tzangeraiferion?
03:39.49ariel_dfunnell, sorry it's modprobe wcfxs
03:40.08dfunnellariel - yep, got that, I was just copying and pasting.
03:40.13dfunnellJust to summarise:
03:40.20dfunnellmodprobe wcfxs
03:40.28dfunnellztcfg -vvvv
03:40.34dfunnellmake install asterisk
03:40.44BrianR___tzanger: Get your norstar mwi to go?
03:40.58ariel_dfunnell, what did you get after the ztcfg -vv
03:41.11tzangerBrianR___: haven't had any time to work on it
03:41.15tzangerI'm fighting off a damn cold
03:41.21tzangerI'm hopped up on AC&Cs right now
03:42.06dfunnellariel - just ran it, realised I made a mistake, fixing now, will re-run soon.
03:42.28ariel_brb
03:42.55*** join/#asterisk tessier_ (~treed@222.253.74.118)
03:49.23dfunnellariel - making progress, now have '4 channels configured.'
03:53.16dfunnellNow that they are running it seems that my Diva server won't start.  Great!
03:53.27dfunnellWonder if I have a resource conflict or something.
03:54.53dfunnellAriel - logging off now to restart box and see if I can't fix this new problem... thanks a million for your help. D.
03:56.07tessier_Greetings from Saigon!
03:56.55*** join/#asterisk footnote (~jhicks@67.141.135.121)
03:58.10brc_file,
04:00.40*** join/#asterisk datareactor (datareacto@203.81.192.33)
04:00.57ManxPower~docs
04:00.58jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
04:02.32mesiHow can I send "*4814" out to an ISDN-card?
04:03.21mikegrbwith pixie dust
04:03.45*** join/#asterisk jsolares (~jsolares@200.12.33.64)
04:03.46footnotemmm pixie dust
04:07.41*** join/#asterisk adjacent (~scott@64.203.220.105)
04:08.46datareactorcan anyone guide for hardware recommedation for *
04:09.20Ron-NaWhen do you get CONGESTION?
04:09.50datareactorif i want to use 4 port 4 digium card
04:09.51QwellRon-Na: When idiots on the freeway decide they want to look at an accident.
04:10.10Ron-Naif I type:  sip show peers,  I get a status OK (200 ms)
04:11.53*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
04:16.06*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)
04:20.55PTG123Hey i found a new ip500 for $169 think thats a good price? :)
04:22.57*** join/#asterisk tessier_ (~treed@222.253.65.14)
04:27.17MocPTG123, yea
04:27.37WilliamKPTG123, new?
04:28.41PTG123yes
04:28.47PTG123i just bought it from ebay
04:29.00PTG123whats the main diff between the cisco 7960 and 7940
04:29.07PTG123any reason to get the 7960?
04:29.40Juggiedoes the 7940 support sip?
04:29.41MocPTG123, only have 6 line instead of 2...
04:29.47MocJuggie, yes
04:29.58MocPTG123, why you get cisco ??
04:30.56PTG123because it has a cool big screen
04:31.01PTG123what would you recommend
04:31.17SedoroxI like cisco's 'cause of the style...
04:31.43footnoteSS7 my arse
04:31.47PTG123i wanna buy a couple of phones just to have a sampling of everything out there that people would want to use
04:31.51PTG123moc what do you like?
04:31.57Mocoh well.. hehe, Im sold to Polycom all the way.  I ONLY wish it had backlight ..
04:32.05QwellPTG123: When you're done testing the cisco, you know where to ship it.  heh
04:32.13PTG123why poly?
04:32.16PTG123qwell :)
04:32.36SedoroxPTG123: I use a BT100.. works fine.. nothing fancy... I really like the design of the Avaya phones..
04:32.49MocPTG123: check the begining of : http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones
04:33.05Mocit nearly all stuff cisco dont support
04:33.10footnoteI had a 7960 on my desk at Cisco :)
04:33.43PTG123Sedorox: got sa url
04:34.06MocPTG123, I dont feel limited with my polycom...
04:34.30MocWhen I got my cisco, I found it so limited, I was very disapointed about IP Phone
04:34.35SedoroxI was just looking at them on ebay.. I haven't used them... BTW.. a nice place to buy VoIP stuff from: http://www.voipsupply.com
04:34.39PTG123Moc: really?
04:35.04Sedoroxhttp://voipsupply.com/index.php?cPath=95 <--- listing of IP Phones
04:35.07MocI just bought 13 ip 500 from voipsupply, excellent service (I got my Sipura ATA from him also)
04:35.09PTG123Sedorox: they don't sell any avaya
04:35.27PTG123Moc: well i just ordered a ip500, but was looking at a cisco just bc. of the style
04:35.27Sedoroxyea.. I know.. like I said.. saw it on EBay...
04:35.51Sedoroxand it seems the census of the channel is Polycom, from the past few weeks I'm been here.. :-p
04:36.23Mocit was cisco before I start talking about the polycom.. Im like nearly the first one to try it, and show it vertue
04:36.32SedoroxPTG123: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=58329&item=5757046508&rd=1&ssPageName=WDVW I think its nice looking... little on the 'spensive side tho
04:37.21PTG123moc: i just wish it had a nicer lcd :)
04:37.27MocPTG123 ???
04:37.42mesiHow can asterisk make a call itself at a given time? Is that possible?
04:37.44MocI find polycom LCD / font alot better
04:37.58Moceasier to read, and a full name can fit correctly
04:38.10Mocand the IP 600 Display is just amazing ..
04:39.24MocPolycom phone are inteligent also, if my name is Marc Olivier Chouinard, it will trunk it to Marc O Chouinard
04:39.37Moccisco will do : Marc Olivie
04:39.39Mocthat it ..
04:40.00PTG123hehe
04:40.55*** join/#asterisk techie (gus@asterisk.horizonte.us)
04:41.11MocUntil someone show me a better phone, Polycom is the best VoIP phone outthere..
04:41.16PTG123so you think  polycomm ip600 or ip500 is the way to go?
04:41.21Mocand at the price, it very nice
04:41.59Mocwell it depend, ip500 is good, but if you need real PoE, having special cable is anoying, so IP 600 is a better bet
04:42.18Mocip 500 can't do XML Browsing, and have 3 line
04:43.06MocIP 600, have XML browsing, have 6 line (with independant LED), with standards PoE + the speaker,mute and handfree it at lower right
04:43.15PTG123my cisco 3500XL will support poe won't it
04:43.28PTG123i like the xml browsing
04:43.37PTG123can i have a direct list on an xmls erver with extensions
04:43.42PTG123and they can browsw it to see who is online?
04:43.58MocPTG123, polycom support provisionned Directory
04:44.17MocPTG123, polycom support that, but * dont ;)
04:44.31PTG123hah
04:44.35PTG123well i have my own directory system
04:44.40PTG123that i use more for softphones then anything else
04:44.44MocI mean, Directory is a xml file with the phone MAC (config is uploaded/downloaded if modified)
04:44.47PTG123an asterisk module
04:45.05PTG123it shows if people are at their desk, etc
04:45.11PTG123but it woul dbe cool to have th ephone support that
04:45.13Mocjust read the page I show about what you can do special.
04:45.33MocI love the directory on the polycom, beable to set ignore, or forward and set ringtone depending of who is calling
04:45.48Mocand having directory that get updated on the server if you modify it on the phone
04:46.03PTG123well can the directory on the ip500 be updated live?
04:46.07Mocthe phone support Instant messaging, and line apperance
04:46.23MocPTG123, I dont know, you can update it on the phone, and the phone will upload it on the FTP server
04:46.37PTG123yah but the xml one would be more live?
04:46.37Mocif you ask to refresh the config, the phone will reboot completely
04:47.19Mocprobably
04:47.35PTG123so your sure i shouldn't go cisco
04:48.31Mocwell Im sure I wouldnt go with them in a deployment
04:48.49NuggetI wish I could set custom ringtones on my cisco.
04:48.58Nuggetbut I do really enjoy building apps with the xml stuff
04:50.10NuggetI wrote a cgi that exports my os x address book to the cisco directory.
04:50.15PTG123williamk: how come you are selling them?
04:50.30WilliamKPTG123, decommed them from a customer's site
04:50.47WilliamKthey decided they didn't like Cisco and spent 60k on proving it
04:50.50mikegrbWilliamK: I'll give you $50 ;)
04:51.18QwellI'll also give you $50
04:51.19NuggetI'll pay $51  :)
04:51.32QwellNugget: he has 50 of them, don't overbid us :p
04:51.35mikegrbyay, we all win
04:51.44mikegrband nugget screw himself
04:51.49WilliamKWhat is this a live auction? =)
04:51.53Qwellyes
04:51.57Qwelllive dutch auction
04:52.05Nuggetik ben een vliegende bidder!
04:52.12QwellNugget: wrong kind
04:52.15PTG123williamk: how much you selling them for
04:52.38WilliamKPTG123, was thinking aprox 200-250 for the 7960s
04:52.41Juggiethe cisco phones are decent
04:52.44Juggiebut not well thought out
04:52.50WilliamKhaven't looked at pricing the 7940s
04:52.51Juggiemitel sip phones are much better
04:53.11Juggiei'm waiting for my shipment of 5220 dualmodes to see if they have some kind of directory support
04:53.16Juggiethe 5055 did not
04:53.30Nuggetthe next phone I buy will be one of the hitachi wireless deals, I think
04:53.30Newbie___any one has any idea what == Unregistered channel type 'Tor'
04:54.43PTG123williamk: good condition?
04:54.50PTG123WilliamK: i'll buy one
04:55.41mikegrbNugget: I'll buy your zyxel one for $10 then ;)
04:56.28WilliamKPTG123, very good condition
04:56.41Nuggetyou coming to SXSW/NTN, mike?
04:57.40*** join/#asterisk obelisque (~samifruit@Toronto-HSE-ppp3887338.sympatico.ca)
04:57.56obelisquehello guys
04:58.03obelisqueSomeone can help me?
04:58.28PTG123williamk: well sell me one for $200 :) i'll paypal you right now
04:58.48MocNugget, you can set custom ringtones on a 7960,7940
04:59.03Nuggetjust one.  not line or caller dependent.
04:59.12obelisqueI have problems with my X100P help me
04:59.23loudmagic word ?
04:59.31Qwellloud: now?
05:00.06MocNugget, ha nope, you need Polycom for that (or use the ALERT stuff when dialling to the phone)
05:00.31Nuggetalert doesn't work if you're using a polyphonic ring
05:00.43Mocit doesnt ? that suck
05:00.57obelisquemy rx ztmonitor level is always on
05:01.42Moccisco need to add more feature, but my guess it the phone is too limited
05:01.55Mocit made to run skinny, in dummy mode
05:02.05Mocthey got SIP in, but got limited
05:02.25mikegrbNugget: -EWRONGCITY
05:03.17Nuggetbah
05:04.11mikegrbyes
05:04.59obelisquehelp
05:05.25mikegrb~help
05:11.19*** join/#asterisk ShadowMaster1 (~askme@host89-133.rancor.birch.net)
05:12.55ShadowMaster1Is anyone actually awake?  I'm a newbie at Linux and at Asterisk..  I need to find out what version a chan_sip.c
05:12.59ShadowMaster1file is..
05:13.29TrepaliumVersion of a particular file...?  Why?
05:13.50*** join/#asterisk kks (~kks@203.115.210.253)
05:13.59ShadowMaster1My Voip provider has sent me a patch file, and as I just installed the PBX last week, I'm curious if that patch is already included.
05:15.07ShadowMaster1What tehy sent me is:
05:15.07ShadowMaster1diff -u -r1.564 chan_sip.c
05:15.07ShadowMaster1--- channels/chan_sip.c14 Nov 2004 15:13:13 -00001.564
05:15.08ShadowMaster1+++ channels/chan_sip.c14 Nov 2004 20:57:20 -0000
05:15.40ShadowMaster1As the patch file header.  Does that make sense as to what version they are trying to patch and if I am running a newer one (how do I tell)?
05:16.33obelisquehelp me with echo cancelling!
05:16.41TrepaliumWell, it's hard to say if that version is their internal CVS server or the public one.
05:16.51Qwellobelisque: ask a damn question
05:16.58TrepaliumThe easiest way is to just apply it, and see if you get rejects.
05:17.58ShadowMaster1They had said in their posting that it was to be applied against the public source, and that they were working with Asterisk on the patch development..  That without it the SIP protocol took too much bandwidth..
05:18.14Qwellbroadvoice?
05:18.17obelisqueztmonitor is displaying 30% of the RX even if there is no calls processed
05:18.22JuggieShadowMaster1, ask them for the bug number on mantis.
05:18.30ShadowMaster1I had trouble getting it to apply.  I think it wants the source and my install does not seem to have the source.
05:18.32obelisquewhyé?
05:18.51*** join/#asterisk mitcheloc (~mitchel@69-169-28-46.anhmca.adelphia.net)
05:18.54ShadowMaster1Let me see if it has a bug #..  BRB...
05:18.59Mocpcq ;)
05:19.20obelisquefrench!
05:19.22Juggieyou need to have the source on ytour box, apply and recompile and install to apply the patch.
05:19.39QwellShadowMaster1: Is it broadvoice?
05:20.32obelisque( # = Audio Level  * = Max Audio Hit )
05:20.32obelisque<----------------(RX)----------------> <----------------(TX)---------------->
05:20.32obelisque<PROTECTED>
05:20.41obelisqueand its always steady
05:20.44ShadowMaster1I don't have the source on my box.  And I know nothing of how to compile it.
05:20.56loudIS IT BROADVOICE ?!
05:20.58ShadowMaster1Yes, it's Broadvoice...
05:21.05QwellThank you.
05:21.16loudIn that case, the patch is not needed anymore, since asterisk stable version brings it.
05:21.24QwellThere you have it.
05:21.42ShadowMaster1That was what I was hoping to hear..  That it was an old bulletin that I could ignore..
05:22.05mitchelocheh it would be nice if BV announced that to everyone...what they really need is iax2 support =)
05:22.33obelisquehey whats better...sip or iax2?
05:22.41ShadowMaster1Thank you folks..  This is my first attempt at working with Linux..  Been using Windows for years, but never had a reason to need to learn Linux..  This is my stepping stone..  Thanks for the help..
05:22.41*** join/#asterisk DHuang (~DHuang@203.41.13.154)
05:22.47TrepaliumDepends on what your hardware supports.
05:22.53DHuangG'day!
05:23.11mitchelocShadowMaster1: heh, asterisk was my gateway drug too
05:23.27DHuangA quick one:  how to turn off native bridge for SIP -> SIP during the calls?
05:24.07ShadowMaster1Drug is about right..  Have missed quite a few hours of sleep this week..  But I'm finally able to atleast call from my PC to my voicemail <grin>..  Can't call between PC's yet, but that will come in time..
05:24.25ShadowMaster1using the XTEN soft phone.
05:25.27DHuanghow to make asterisk not to bridge call for certain context?
05:25.40*** join/#asterisk owh (~onno@OptusSatelliteServices.22bjc76f09.optus.net.au)
05:26.10footnoteDHuang: if they're on the local lan together they don't bridge if they don't have to, right?
05:26.42obelisquedo you guys know something better than sjphone (free)?
05:26.59Mocxten
05:27.02ShadowMaster1XTEN
05:27.09footnotexten!
05:27.15obelisqueits free?
05:27.23footnoteno, you gotta pay me five bucks
05:27.24ShadowMaster1Well, I'd give XTEN 3 votes...  hehe
05:27.25mitchelocyes, but feel free to donate to their bloated budget ;)
05:27.32TrepaliumXten X-Lite is very good.
05:27.37obelisqueok whats your paypal address?
05:27.38footnotemitcheloc: shh! :)
05:27.41footnotehehe
05:27.55owhHey all, been googling until I'm blue in the face, fiddled with source code, still none the wiser. I would like to play an announcement to both the caller and the callee at the same time at the beginning of the message.
05:27.56footnoteobelisque: it was a JOKE son :)
05:27.58mitchelocmine eh? lol, nah if you want to donate donate to mark
05:28.14footnotemitcheloc: he almost gave me money
05:28.29owhs/beginning of the message/beginning of the call/
05:28.42footnotefree beer!
05:28.48mitchelocbah, be nice to him
05:28.51obelisquehey guys...I am preparing a project for school... Around 600 phones...Does asterisk can handle this?
05:28.58footnotei let him off the hook, no paypal :)
05:29.00mitcheloclol 600 phones? is that enough?
05:29.12mitchelocwhat kind of "project" is this?
05:29.14DHuangfootnote:  it's from local(NAT) -> public -> public  and when it bridge I can only hear one side
05:29.25TrepaliumYou better have big servers to handle that...
05:29.47footnoteDHuang: hrm
05:29.52obelisquemmm
05:30.36footnote"right topology"
05:31.52DHuangfootnote: yea.. I know
05:31.55owhSome background might help. The message says:"This is an experimental VoIP call using a wireless and a satellite link, be patient." The purpose of the message is two-fold, one to tell the users, the other to have the call settle down while the satellite link sorts out it's bandwidth.
05:32.11footnoteok, time to go to waffle house.
05:32.36footnotebaon egg cheese plate, wheat, scattered smothered, hold the mayo.
05:32.39owhfootnote: Will you be swapping manly stories?
05:32.44footnoteowh: naw
05:32.52loud"this is a satellite link, please be patient, your call will be completed within an hour".
05:33.00loud:>
05:33.02footnoteowh: waitress, big (.Y.)
05:33.15owhYum
05:33.21owh(The waffles :-)
05:33.26footnoteyeah yeah
05:33.27footnoteme too
05:34.05owhSo, anyone got any suggestions?
05:50.13ShadowMaster1Ok, question time again..  With Broadvoice's service..  Does this line look right??
05:50.13ShadowMaster1register => <accountid>@sip.broadvoice.com:<password>:<account id>@sip.broadvoice.com/<extension>
05:50.59ShadowMaster1it keeps telling me password is an invalid port.
05:51.31loudno.
05:51.44louduse the common reg function, let me show you .. hold
05:51.51ShadowMaster1thanks
05:52.18loudregister => bvnumber@sip.broadvoice.com/ext
05:52.36ShadowMaster1but then where do I put the password?
05:52.52loudbelow, in your [sip.broadvoice.com] context.
05:53.34loudgot time, let me pastebin .. hold.
05:53.41ShadowMaster1Ok, now I really am confused..  Thanks
05:53.48ShadowMaster1i'll hold
05:54.40loudhere you go, http://pastebin.ca/6886
05:55.40Qwellfor some reason, I can never connect to that one
05:57.30loudyou had to mention it man
05:57.33loudMar  4 22:49:48 NOTICE[15088]: chan_sip.c:8726 sip_poke_noanswer: Peer 'sip.broadvoice.com' is now UNREACHABLE!
05:58.18ShadowMaster1Ok, config updated..  Lets see if it likes it or spews errors..
05:58.23loudcool
05:59.29Qwellloud: I meant pastebin.ca :p
05:59.54loudAh:>
06:00.36ShadowMaster1-- Got SIP response 404 "Not found" back from 147.135.0.128
06:00.52TrepaliumOh, look.  More updates for FC3.  I'm getting tired of these kernel updates, though.
06:00.53ShadowMaster1Mar  5 00:06:44 NOTICE[1698]: chan_sip.c:4044 sip_reg_timeout:    -- Registration for 'ShadowMaster@sip.broadvoice.com' timed out, trying again
06:00.53ShadowMaster1<PROTECTED>
06:02.37*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:02.39loudi doubt ShadowMaster is your username.
06:02.48loudit must be an us number.
06:02.49ShadowMaster1Yes..  That's my user ID..
06:03.06loudmaybe when you signed up, but not the sip user.
06:03.18loudMine's JackDaniels for example.
06:03.22ShadowMaster1Hummm...  I can try that..  so your suggesting I put the tel # they assigned me as the account ID?
06:03.26loudBut i use a 619XXXXXXX
06:03.32loudYes sir.
06:03.51ShadowMaster1Ok, let me try that..  That does kind of make sense...
06:07.25Ron-Naloud & ShadowMaster1: I am at the same setup now ;-)   how is the register statment in sip.conf?
06:07.50QwellRon-Na: scroll up about 15 minutes
06:08.10ShadowMaster1Not sure because I'm not getting it..
06:08.37Juggiedid you by chance check the wiki
06:08.47ShadowMaster1i'm trying register => 214453xxxx:password@sip.broadvoice.com/8400
06:08.53ShadowMaster1and it's giving me a 404 error
06:09.00Juggiedid u check the wiki?
06:09.47ShadowMaster1wiki??
06:09.56Trepaliumheh.  RTFW is such a popular response.
06:10.18QwellIn most cases, its an appropriate response
06:10.27ShadowMaster1guess that means no...  I've seen RTFM..  Never the W for whitepaper...  cute...
06:10.35QwellW for wiki
06:10.40Qwell~rtfw
06:10.41jboti heard rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php
06:10.56Juggiehttp://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broadvoice
06:11.03Juggiei'll even direct link you
06:11.10TrepaliumLots of good info there, but it can be hard to find what you're looking for at times.
06:11.22ShadowMaster1I'm loading it now..  I've read thru so much stuff that much of it has turned to mush in my brain..
06:11.45TrepaliumAt least it hasn't turned your brain into mush yet.
06:13.27ShadowMaster1Well, the error message has changed..  hehehe..  Thanks for pointing out that page..
06:13.28Juggieyou can also try a 1 in front of that number
06:14.22ShadowMaster1my brain is the next to become mush..  It'll be server with that scattered, covered and diced hashbrowns taht someone was talking about earlier...
06:14.53Juggiedid u get it working?
06:17.25ShadowMaster1No, now it no longer gives me 404 errors.  Instead Mar  5 00:23:13 NOTICE[1698]: chan_sip.c:6818 handle_response: Failed to authenticate on REGISTER to '<sip:214xxxxxxx@sip.broadvoice.com>;tag=as43172473'
06:17.47*** part/#asterisk DHuang (~DHuang@203.41.13.154)
06:19.08Juggiedid u try a 1 in front of the number?
06:19.37ShadowMaster1yes, with the 1 I get a 404 again..  So I suspect I'm closer...
06:19.44RestLessGeminiShadowMaster1 : can you see it registered when you do SIP SHOW REGISTRY
06:19.48RestLessGemini?
06:20.19JuggieShadowMaster1, then it must be the password or the extension at the end
06:21.06ShadowMaster1funny enough, yes, it does show up when queried..
06:21.19Juggieit likely says unregistered
06:21.28Juggiewhat happens if you call your did
06:21.33Juggiedoes the incomming call work
06:22.14ShadowMaster1No, state is "Auth Sent"....  and if I call it, I don't get thru...  Just tried that on my PSTN line.
06:22.30*** join/#asterisk ManxPower (~eric@152.sub-166-145-186.myvzw.com)
06:22.43ShadowMaster1The web page didn't show the /extension on it.  I left that in.  Was that wrong of me?
06:23.31RestLessGeminihmm okay
06:23.33RestLessGeminigive me a min
06:23.42ShadowMaster1no prob..  Thanks for the help...
06:23.50ShadowMaster1brb.. Gonna grab a drink..
06:25.10*** join/#asterisk marc324 (~marc32344@69-28-224-214.dsl.teksavvy.com)
06:26.31ShadowMaster1ok, I'm back
06:27.18RestLessGeminiShadowMaster1: read PM
06:28.51RestLessGeminitry those which i  gave you in PM
06:31.14*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
06:33.30PTG123Mar  4 23:30:24 NOTICE[65152]: chan_iax2.c:4337 register_verify: Peer 'ptg' is not dynamic (from 66.235.234.131)
06:33.31PTG123Mar  4 23:30:24 NOTICE[65152]: chan_iax2.c:6481 socket_read: Registration of 'ptg' rejected: Registration Refused
06:33.34PTG123what would cause that?
06:33.37ShadowMaster1Think it's time for me to get a password reset...
06:34.00RestLessGeminiPeer 'ptg' is not dynamic (from 66.235.234.131)
06:34.08loudput host=dynamic on iax.conf
06:34.08PTG123we have host=dynamic
06:34.08RestLessGeminithis is the cause PTG123
06:34.12PTG123yah we have that
06:34.13RestLessGeminiput host = dynamic
06:34.15PTG123still nothing
06:34.38RestLessGeminithen try removing default ip if you have that in your settings
06:34.43ShadowMaster1With all these different confgs, I'm suspect it's my screw up now..
06:35.00hardwirerequested/capability 0x2/0xf802 incompatible  with our capability 0x4.
06:35.01hardwirewtf
06:35.03PTG123paste error :)
06:35.04hardwireiax2 -> iax2
06:35.29loudthats ulaw
06:35.34loud0x4 i mean.
06:35.49hardwireok
06:35.53hardwireI have both set to gsm
06:35.54hardwireodd
06:38.14hardwireI think it must not be authing
06:43.06ShadowMaster1Password reset..  (and they answer the support lines quick for a change)...  And now it shows registered on the sip show registry....
06:43.25ShadowMaster1Thanks all for the help..
06:44.26ShadowMaster1Tomarrow, I tackle connecting it to the extensions...  Night everyone and thanks again for the help...
06:44.58RestLessGeminiyw.. nite ShadowMaster1
06:47.54hardwireugh
07:11.56*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
07:15.58*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
07:17.44*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)
07:18.10*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
07:18.17*** join/#asterisk iceyp (~icepick@max.unix.co.nz)
07:20.21iceypis it free to test g729?
07:20.33iceypcan i download it and use it without any problems?
07:23.44iceypalso where is the cheapest place to get a 7960
07:27.43*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
07:28.45*** join/#asterisk asteriskforuk (~vircuser@i-195-137-59-254.freedom2surf.net)
07:29.16asteriskforukhello all
07:29.56iceypheya
07:30.06asteriskforukne1 have a sample dialplan.xml to use on a 7940/60 to get rid of the "Dial" button for each call?
07:31.37*** join/#asterisk ST-3 (ser@dipsy.tch.org)
07:32.40iceypmmm nah, where u buy ur 7960 from?
07:32.42iceypI want to buy one
07:32.58iceyptrying to find the cheapest location
07:33.35asteriskforukwe got our 7940's off ebuyer.com
07:33.44iceypthey cheap?
07:33.52asteriskforuku in uk?
07:34.06iceypnah
07:34.08iceypNZ
07:34.31asteriskforuk190 GBP
07:35.00asteriskforuku'll have a lot of fun with the SIP upgrade
07:35.09iceypsip upgrade?
07:35.18iceypis there problems/?
07:35.31asteriskforukthey come with the SCCP images for Cisco call manager usually
07:35.41iceypahh right
07:36.03asteriskforuku'll need a cisco login to download some sip images
07:37.39iceypcant download it off the net?
07:39.06asteriskforuknah
07:39.07iceypebuyer.com dont have any voip phones
07:39.59asteriskforukwhat keyword u typed?
07:42.03*** join/#asterisk invi_ (~undisclos@dsl-crow-209-5-162-157-cgy.nucleus.com)
07:42.18*** part/#asterisk asteriskforuk (~vircuser@i-195-137-59-254.freedom2surf.net)
07:43.02*** part/#asterisk invi_ (~undisclos@dsl-crow-209-5-162-157-cgy.nucleus.com)
07:45.02*** join/#asterisk Romulan (~Vortexia@cpc2-rdng7-3-0-cust163.winn.cable.ntl.com)
07:45.41*** join/#asterisk DHuang (~DHuang@203.41.13.154)
07:45.47DHuang:-) hi again
07:46.47RomulanHi, can anyone tell me what the updates are in version 1.0.6 ? Thks
07:47.01DHuanghow to lookup a username in the sip.conf context bebefore the Dial??
07:47.06DHuangRom: check the CVS
07:49.51*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
07:49.56Ron-NaWhere to start? I want to get the CDRs on a web page, .... do I need to put it first in a database?
07:50.38DHuangRon-Na: prob easier to store in DB
07:52.16RomulanSorry, DHuang, were would I find that ?
07:53.04DHuang1st one on google... http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005346.html
07:53.20RomulanCheers, Thks
07:53.26DHuangRomulan: np.
07:54.04Ron-NaDHuang how to get it into a database? Realtime?
07:54.32DHuangRon-Na: what do you want to do??? store or retreive?
07:54.56Ron-NaI want to see the records of dialing
07:55.09Ron-Na... to make a billing out of it (later)
07:55.11DHuangwhat DB is it stored in?
07:55.32Ron-Nano database at all now, .. just clean installation of Asterisk
07:56.02DHuangRon-Na: setup CDR to DB so easier to do SQL query later...  check out http://www.voip-info.org
07:56.24DHuangThen you can use PHP, ASP...to put on to the www site
07:56.26Ron-NaI checked but I did not find any useful hint, ... what
07:56.31Ron-Naphp
07:56.46DHuangor perl.. whichever language you like
07:56.54*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
07:57.05DHuangor just download the calling card application 1.1  got examples there
07:58.00Inv_arpsimple 3 way conferencing = meetme?
07:58.52Inv_arpRon-Na: i use php/sqlite  werks nice
07:58.52DHuanginv_ yes
07:59.10Inv_arpDHuang: so to do conference i have to xfer my caller to a room
07:59.14DHuangRon: or you can use AGI too
07:59.57DHuanginv: that's right.... can auto transfer when they call the exten
07:59.58Ron-NaDHuang I have installed ASTCC and it finally works, but now I need to proof that it works correct ;-(
08:00.37DHuangRon:  Hmm.. didn't you say no DB??
08:00.53Ron-NaDHuang: I read something, to put the Master.cvs into an extra directory and pull it into a database from there, ..
08:01.09iceypanyo ne here using 729 codec?
08:01.14*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
08:01.24Inv_arpiceyp: nah too broke to get it
08:01.24DHuangiceyp: yes... g729 is kewl
08:01.34Inv_arpgsm  :)
08:01.35DHuang$10 per channel
08:01.41*** part/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
08:01.52Ron-NaDHuang: yes, you are right, ASTCC makes it also into a database, ... maybe I can check out this one first
08:02.04iceypDHuang is it downloadable to test?
08:02.35DHuangRon: yeap.... try it out with DB.. b'cos it has samples try this one  http://areski.net/areskicc-doc/
08:02.35Ron-NaDHuang: where are you (sounds chinese)
08:02.52DHuangRon: ya.. I think my Surname is a dead giveway
08:03.12DHuangice: test?? Hmm...  what do you want to test?
08:03.23*** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net)
08:03.38DHuangRon: I'm in Oz... Australia
08:04.18Ron-NaDHuang: nice place, ... I only was once in Australia, ... but I am from Austria, stranded in Taiwan, ....
08:04.51*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
08:05.26DHuangRon: Yeah... Austria... nice place too... was there 2 years ago for work.. and might go there again next month..  Are you made in Taiwan too?
08:05.26Inv_arpheh
08:07.08Ron-NaDHuang: ??? I came 15 years ago to Taiwan, ... built up a trading company with German speaking countries in Europe, used two BBSes for low communication to Europe, swapped to Internet as soon it was available, built up an ISP, ... brought up the company to 6 million US$, than my wife kicked me out from my own company
08:07.29invi_anybody: where do i specify codec to b used for dundi connections?
08:08.04Inv_arpRon-Na: woah
08:08.24DHuangRon: Geee....
08:08.28Ron-Nainvi_:  I am still very unclear about dundi. Why would somebody use it, when he can use ENUM?
08:09.09Ron-NaDHuang: 5 years passed by, now I can recover .... and hope that I can get all done with VoIP / Asterisk, ...
08:09.10DHuangRon: so what you doing now???
08:09.28Inv_arpDHuang: so for conferencing i just set up an ext? i have a small office i dont want other people joining in mine i want them in their own
08:09.50Ron-NaI am Internet Consultant, but recently back to VoIP (I did it already in my ISP in 1997 - but line speed was not good enough)
08:09.55Inv_arpi dont want conferecing to exceed 3 parties
08:10.24invi_Ron-Na: im wondering bout it too...
08:10.25DHuangRon: I see... I did VOIP in 1999 in my ISP too.. hehehe..
08:10.43Ron-NaInf_arp: just use a password for the conference, ...
08:10.56Ron-NaDHuang: what are you doing now?
08:11.09DHuanginv: yeap, just setup the rom.. and when they dial certain ext it redirect to that meetme room
08:11.45Ron-NaI got a project to port a video phone from H323 to SIP, ... but need somebody who can help me to do that (share the project)
08:11.51Inv_arpDHuang: but what about a seperate conf  i have to setup another room?
08:11.52DHuangRon: I'm working for a holistic medicine company in Australi  (http://www.medec.com.au)  in R&D
08:12.09DHuangRon: our factory is in German...
08:12.14Ron-NaDHuang: I studied Electronic Medicine, .. hehehehe
08:12.44DHuangRon: hehehe..... Pulsating Energy is our product...
08:12.58DHuangInv: yes, setup as many room as you want..
08:13.09Ron-NaDHuang: my favourite was EEG and EKG, ... in my first company in Austria I developed a handheld EKG, which I wanted to produce in Taiwan, ... got a girl friend instead, ...
08:13.44iceypthere are free g729 codecs?
08:13.47DHuangRon: harhahrha..... :-)
08:14.03DHuangiceyp: no free g729, asterisk can use by-pass
08:14.08Ron-Nabe back soon
08:14.12Inv_arpDHuang: ok i know what to do...      php/agi to setup rooms dynamically
08:14.26iceyphttp://www.voip-info.org/wiki-ITU+G.729 <-- non comercial use
08:14.37DHuanginv: ya..
08:15.03Inv_arpiceyp:  experimental
08:15.25DHuangiceyp:  i have g729.dll for windows if you want to test out your asterisk...
08:16.02iceypi want to use it for an open source project
08:16.17iceypwell comunity project
08:16.36DHuangiceyp: Hmm... then you might have to buy license for it...
08:16.47iceypohh ok
08:16.48invi_anybody here with dundi know-how?
08:17.02DHuanginv: what's dundi?
08:17.38invi_www.dundi.org
08:18.01DHuangHmmm dundi is disabled
08:18.09Inv_arp<PROTECTED>
08:18.29*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
08:18.36invi_www.dundi.info
08:18.53DHuangOk... inv: what u want to know?
08:19.05invi_where do i specify codec to b used for dundi connections?
08:19.12iceypthe interl g729 apears free for non comcerial
08:21.46DHuanginv: I think in the SIP.conf
08:22.27DHuangiceyp: try http://www.voiceage.com/freeimplement.html
08:23.36DHuanginv: why using dundi?
08:24.10*** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl)
08:24.30invi_DHuang: "DUNDi is fully-distributed with no centralized authority whatsoever"
08:24.59DHuanginv: I know... then everyone have to going dundi?? to make full use?
08:25.00MuppetMasterHello everyone.
08:25.14MuppetMasterDoes anybody spot a logic flaw in this RealTime statement?  | 139 | from_libretel        | s            |        3 | GotoIf          | GotoIf($["${CALLERIDNUM:0:9}" = "931010300"]?from_libretel|1|1:5)                                                        |
08:25.35MuppetMasterBy posting I just did!  LOL
08:26.10invi_DHuang: search 4 dundi & ull c how big it is getting
08:26.26DHuanginv: yeah?? Ok... shall try on google now
08:26.59invi_DHuang: http://dundi-map.netmonks.ca
08:28.03*** join/#asterisk Blake0PS (~blake@c-24-245-24-84.mn.client2.attbi.com)
08:29.11iceypDHuang that doesnt load
08:29.32Blake0PSI have asterisk CVS. How do I update it to the newest version?
08:29.36DHuanginv: impressive...
08:29.37*** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com)
08:29.49DHuangBOPS: to the update and make clean; make install
08:30.30DHuangiceyp: hmm.. they must moved it.. just a sec
08:30.37iceypta
08:32.03Blake0PSis there a howto somewhere on how to update?
08:32.11DHuangiceyp: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
08:32.23*** join/#asterisk mack_jpn (~mack_jpn@210-194-200-154.rev.home.ne.jp)
08:32.36DHuangBOPS: just do as how you get the CVS at the 1st place.. then recompile it
08:32.56Blake0PSall my settings will remain unchanged?
08:33.07DHuangif you are worried.. backup 1st
08:33.37Blake0PSgenerally are settings unchanged?
08:33.53DHuangBOPS: ya.. that' s right.. it doesn't update the config files
08:34.11Blake0PSthanks :)
08:34.20DHuangnp
08:34.26MuppetMasterBlake0PS:  As long as you don't install the config changes, yes the make/make install will leave the conf files unchanged.
08:34.48DHuanginv: did you patch your asterisk for the graphs??
08:36.05iceypgraphs?
08:36.27DHuangto put your node in the duni-map
08:37.03MuppetMasterI have a DUNDi peering setup and have tried to dail through it.  The problem is, I don't know of any valid DUNDi addressable numbers.  Anyone know one?
08:37.06MuppetMasterThat could be used for testing?
08:37.40MuppetMasterDHuang:  Which patch is needed for the graph?
08:38.09DHuangMM: sorry, I'm very new to dundi...  ptach the pbx_dundi.c
08:38.31MuppetMasterDHuang:  No worries, as am I.  But where is the patch to patch pbx_dundi.c?
08:38.46DHuang[16:26] invi_: DHuang: http://dundi-map.netmonks.ca
08:40.25DHuangdo you guys know how to use the "database get" command?? what's family and what key?
08:42.26iceypwhats dundi for?
08:44.05hardwireplk
08:44.06hardwireerr
08:44.06hardwireok
08:44.11hardwire3 concurrent calls outbound over starband
08:44.14hardwirenot too terrible
08:44.48MuppetMastericeyp:  Peer 2 Peer number discovery.  A replacement for ENUM.
08:45.03iceypohh ok, it new?
08:45.16MuppetMastericeyp:  Allows you to have a decentralized number discovery system.  It is has been in the Asterisk head since about Oct 2004.
08:45.34MuppetMastericep:  http://www.dundi..info
08:45.40hardwirehow can you limit the amount of iax channels inbetween peers?
08:50.26mamcintyjust wondering, what codec are you using for those three channels?
08:50.42hardwireg729
08:50.46hardwireiax trunking
08:51.15hardwire500ms jitterbuffer
08:51.28hardwireand that wasn't really enough
08:52.51*** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net)
08:54.35hardwireI need to push around 8.
08:54.45hardwirebut I don't have the dialplan tobounce it around like that yet
08:56.04hardwireanybody doing anything similar?
08:56.21hardwireits pretty pokey over our vsat connectino as well .. with dedicated 512kbps up :)
08:56.22hardwirewell
08:56.23hardwirechoppy
08:56.30Blake0PSAfter grabbing the CVS and make/make install. I get "Illegal Instruction" when I run *
08:56.59DHuangtry -vvvvvvvvd
08:57.05DHuangasterisk -vvvvvvvvd
08:57.14*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
08:57.18hardwireheh
08:58.00Ron-NaQuestion:   ASTCC allows you to add 4 trunks into one route statement. Can anybody explain me that?
08:58.56DHuangRon: beats me..
08:59.16Ron-NaDHuang: why?
08:59.42DHuangRon: sorry.... just mean not sure why you want to have 4 trunks for 1 route...
09:00.39Ron-NaDH yes I want that!!! I want to make sure that I am not using one trunk more than it's capacity, ... e.g., if I have several gateways in xxx, but only two lines at each gateway, ...
09:01.04*** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com)
09:01.08Ron-NaDH (I don't want to tell you another reason in public)
09:02.03Blake0PSDHuang : it gets past res_adsi.so and then "Illegal instruction"
09:02.17*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
09:03.04*** join/#asterisk Tony[] (~chatzilla@i-195-137-6-171.freedom2surf.net)
09:04.01Tony[]hi .. does asterisk have the ability to know about multiple routing providers and choose between them to find the cheapest on a per call basis?
09:04.22DHuangBOPS: instersing....
09:04.33DHuangTony: check out the ASCCT
09:04.44Tony[]??
09:04.57Ron-NaASTCC
09:04.57Tony[]I don't run asterisk, it was just an idea I had
09:05.01*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
09:05.26Ron-NaTony, how do you have the data for the providers?
09:05.38Tony[]sure .. it's a number prefix
09:06.05Tony[]prefix the number with a code which selects the routing provider
09:06.09DHuangTony: yes, it does, you can the cost for the route
09:06.18*** part/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
09:06.42DHuangTony: check out  http://areski.net/areskicc-doc/ ass see if this is what you want
09:06.45DHuang<PROTECTED>
09:07.22Tony[]ok ... I figured the next step would be to get routing companies to provide their call costs in XML format to make life easy for the PBX
09:07.51Ron-NaTony if you got that, than send me a copy ;-)
09:08.07iceypanyone know where i can get a cheap 7960?
09:08.22DHuanghahaha..... should just put on www.voip-info.org
09:08.29Tony[]there are numerous UK providers appearing call18866.com call1899.co.uk
09:08.42Tony[]with simple prefixes, all with real cheap rates
09:09.05Tony[]but you have to search quite closely to get the best prices
09:09.18Blake0PSDHuang : any other ideas for "Illegal instruction"?
09:09.33DHuangTony.. store eveything on DB and run query I guess
09:09.45DHuangBOPS: sorry... looking now
09:10.03iceypTony[] build a site :>
09:10.11*** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk)
09:10.12Tony[]yeah .. but everytime the rates changed you'd need to update the DB
09:10.13iceypTony[]  u got quite a good list already?
09:10.23iceypof providers etc?
09:11.03Tony[]I only have 5 or 6 providers for the UK so far, but between them I've cut my call costs down to about £3 a months
09:11.13DHuangBOPS: would be the config files.... not sure which one you need to change... you need to check 1 by 1
09:11.16iceyphehe sweet
09:11.31Blake0PSDHuang : Okay, thanks.
09:11.58Tony[]the free national calls within the UK is the biggest saver
09:12.18Tony[]other people are pleaces like telediscount.co.uk and telestunt.co.uk
09:12.50Tony[]anyway .. gotta run .. just wanted to air the idea see what people thought
09:12.50DHuangBOPS: sorry... easier way is rename them and try to load asterisk
09:13.15DHuangtony: ya.. and free call from UK is good... keep up UK
09:13.54Tony[]I figured the next logical step would be to get the providers to provide their rates ina standardised XML format
09:14.21DHuangor setup a service for the providers and charge them back.. ehehhe
09:14.26Tony[]which included Country of Origin (UK for example) and destination of call and cost,  using ISO standard country codes
09:15.08DHuangTony: shouldn't take long.. 1 week coding + docs
09:15.12Tony[]maybe it could be provided as an RSS/RDF feed?
09:15.19Tony[]from the providers
09:15.36Tony[]anyways ... gotta run ...
09:15.39Tony[]cya people
09:17.22DHuanganyone konw how to lookup username in sip.conf using AGI?
09:17.38DHuangor know where I can find the script for it?
09:27.03Nebukadnezamorning
09:27.15DHuangg'day!
09:27.23NebukadnezaDHuang: what exactly do you want to do (maybe i can write you a small python agi)
09:28.19DHuang:-)  when dial extension 1700XXX it lookup sip.conf for that context username and set the callid to that username.
09:28.58Nebukadnezai dont know how to set the callid ... but just the lookup should be no problem :)
09:29.24DHuangI know how to SetCallID  just need agi to return the username
09:30.08DHuangie. agi take 1 input and return username
09:30.15Nebukadnezawhy dont you set the callid in the agi too?
09:30.38DHuangcan do that too... :-p  sorry... I'm very new with AGI
09:30.44Nebukadnezadito :)
09:34.06Nebukadnezaso ... which argument should the script take?
09:34.38DHuangthe script will take the CONTEXT name
09:35.09DHuangie. [fwd-outgoing]  <-- fwd-outgoing
09:35.35Nebukadnezakay (this should be passed as standard ENV to a agi?)
09:36.02DHuangyeap... is will be ${CONTEXT}
09:36.05Nebukadnezathe context in which the agi is running
09:36.13DHuangthat's right
09:36.44*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
09:37.00Nebukadneza:)
09:37.09*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
09:37.18DHuanginvi: any luck?
09:38.13Nebukadnezaokay - and to what should the callid be set?
09:38.55DHuangcallid is set to that context username, ie. specified in that context   username=17001234567
09:39.14DHuangor callierid=17001234567 if it's set
09:45.05Blake0PShow do i copy an entire directory somewhere else with linux
09:45.13NebukadnezaBlake0PS: mv
09:45.14Blake0PScommand line
09:45.34Blake0PSisn't that move?
09:45.36NebukadnezaDHuang: sec ... @ agiilib.py readin'
09:45.44NebukadnezaBlake0PS: ah ups sorry
09:45.45Nebukadnezacp -R
09:45.57Blake0PSthanks
09:46.20DHuangBOPS: mkdir backup; cp * backup
09:47.33HjemmeRoyKcd .
09:52.00PTG123Mar  5 02:49:06 WARNING[65817]: channel.c:1934 ast_request: No translator path exists for channel type IAX2 (native 63519) to 256
09:52.04PTG123what causes that?
09:52.21DHuangwhat's 256?
09:52.33PTG123no idea :)
09:52.38PTG123ulaw?
09:52.49DHuangtry iax debug and see what happen
09:53.39PTG123iax2 debug didn't print naything extra
09:53.40PTG123should it?
09:54.08PTG123this was working 30 minutes ago
09:54.10PTG123now its broke
09:54.24*** join/#asterisk Nebukadneza (~daddel9@i3ED6E7F3.versanet.de)
09:54.36DHuangwhat device are you calling from/to???
09:54.37Nebukadnezarehi
09:54.45DHuangNebukadneza: welcome back
09:54.54Nebukadnezawhat was the last i said before the laggout? :)
09:55.01PTG123using a sip phone
09:55.02PTG123xpro
09:55.06PTG123trying to cal a normal number
09:55.09DHuang[17:45] Nebukadneza: cp -R
09:55.19NebukadnezaDHuang: *shrug*
09:55.43DHuangPTG123: normal number??? in which network?
09:55.54PTG123going out through an iax link to another box
09:56.12PTG123i have 2 ld providers, tried them both
09:56.53NebukadnezaDHuang: sorry  ... currently agilib readin
09:57.05*** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com)
09:57.16DHuangNebukadneza: :-(
09:57.31*** join/#asterisk dwC- (~dwc@69.42.74.4)
09:57.44NebukadnezaDHuang: will be finished soon
09:57.47DHuangPTG123: so SIP phone on IAX2 calling ??
09:57.53PTG123yah
09:57.55DHuangNebukadneza: thanks... :-p
09:57.59PTG123the first codec number is what and thats the second
09:58.01PTG123native BLAH
09:58.03PTG123whats it refering to?
09:58.19DHuangHmm... blah =??
09:58.36DHuangwhat device you calling?? other side?  xpro?
09:59.21PTG123cell phone
09:59.22PTG123xpro on one end
09:59.24PTG123but you know
09:59.27PTG123i am having more problems
09:59.31PTG123i think i screwed up something
10:00.12DHuangPTG123: :-( so calling like a PSTN/Mobile through a provider?
10:00.12PTG123nah i am wrong
10:00.15PTG123other stuf fis working
10:00.18PTG123yah dhuang
10:00.36DHuangPTG: Ok, check what codec your provider use.. and set that in IAX.CONF
10:00.42PTG123what is it tring to translate from/to?
10:01.00DHuang??
10:02.22PTG123i don't understand why this suddenly stopped working
10:04.22*** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
10:05.09PTG123this is so frustrating
10:05.31DHuangwas working and you change nothing and now stop working?
10:05.43PTG123well what i changed i thought i changed back
10:06.08DHuanghehehe... did you stop and start asterisk?
10:06.21PTG123the problem is i am kind of manipulating these peer records directly :) But i am just not sure what that line means
10:06.23PTG123yah i did
10:06.52DHuangOh... hehee...  part of learning :-p
10:07.03PTG123those two codecs
10:07.07PTG123one is 65k one is 256
10:07.11PTG123which are they refering to
10:07.18PTG123the peer, my asterisk server, theirs?
10:07.19PTG123or what?
10:07.39NebukadnezaDHuang: one second pls ...
10:07.43DHuangwhere u set those?? in iax.conf?
10:07.50DHuangNebukadneza: :-p Thanks..
10:07.55PTG123where is it getting them from
10:08.47DHuangcan you put the debug or info on http://www.pastebin.com/
10:09.55PTG123oh my fucking god
10:10.04PTG123did a gmake clean
10:10.05PTG123in all my dirs
10:10.07PTG123now it works :)
10:10.19PTG123the only problem i have now
10:10.23PTG123is caller id isn't being set
10:11.29NebukadnezaDHuang: may you help me for one sec?
10:11.33Nebukadneza(just one sip call pls)
10:12.06DHuangOk.... np
10:12.18Nebukadneza01@nebuk.homelinux.org via sip ... :)
10:12.20DHuanglet me load my fwd..
10:12.48NebukadnezaDHuang: did you hear someething?
10:12.54DHuangNope.. :-(
10:12.58Nebukadnezaargh
10:13.07Nebukadneza-- Attempting native bridge of SIP/fwd.pulver.com-08135438 and SIP/01-335f
10:13.13Nebukadnezathis is the problem (imo)
10:13.26DHuangOh... todo with nat
10:13.35Nebukadnezajep :*
10:13.46Nebukadnezai set canreinvi=no, but ...
10:14.24DHuangnet=yes  and canreinvite=no
10:15.00Nebukadnezanat=yes ... sec
10:15.31Nebukadnezacan you try again?
10:15.35DHuangk
10:16.09Nebukadnezahm ... connection a bit laggy :/
10:16.12DHuangr u german?
10:16.21Nebukadnezayep
10:16.33Nebukadnezais my english that bad? :)
10:16.35DHuanghehee... tell from the voice....
10:16.42Nebukadnezalol
10:16.45Nebukadnezamkay
10:16.54Nebukadnezaabout your agi ...
10:17.03DHuangNo... working in German company, so can pick up the accent...
10:17.09Nebukadneza*g okay
10:17.42Nebukadnezaso what should the callerid be?
10:17.53*** join/#asterisk mady (~root@61.11.24.250)
10:17.57DHuang... doh versanet.de is in Germany...
10:18.14Nebukadnezaits a german isp :)
10:18.39DHuangNeb: ok.. callerid is in the file...  1st grab callerid=  if not exit grab username=
10:18.46*** join/#asterisk mady (~root@61.11.24.250)
10:18.50DHuangDanka...
10:20.01NebukadnezaDHuang: callerid= from the sip.conf?
10:20.21DHuangNeb: yeap, from sip.conf   or from the database key
10:20.54Nebukadnezahm kay ... thats a bit more work :)
10:21.24DHuangYeah.. I know, that's why I thought someone must have written the script..
10:21.51DHuangNeb: be back in 30 mins... dinner time
10:22.33Nebukadnezak
10:22.36Nebukadnezaill try till then
10:23.21madyhello all i am looking for the solution for the following error can any body help me out
10:25.36madyNotice: Configuration file is /etc/zaptel.conf
10:25.36madyline 0: Unable to open master device '/dev/zap/ctl'
10:26.46HjemmeRoyK~seen wasim
10:26.47jbotwasim <~wasim@203.81.213.118> was last seen on IRC in channel #asterisk, 4d 20h 51m 30s ago, saying: 'yay! fresh feta cheese!'.
10:29.46NebukadnezaDHuang: parsing the asterisk config is quite difficult :/
10:30.23HjemmeRoyK~lart Nebukadneza
10:31.05NebukadnezaHjemmeRoyK: Oo?
10:31.26DHuangback..
10:31.36Nebukadnezawb
10:31.42Nebukadnezathat was a fast dinner :)
10:31.46DHuangNeb: Oh... then don't worry, I'll use php with mysql...
10:31.53NebukadnezaOo?
10:31.54DHuangNeb: ya... finished my dinner....
10:32.03Nebukadnezai didnt say that its impossible :)
10:32.09Nebukadnezas/that//
10:33.11DHuangNeu: I know it's possible.. but if it's too much work.. don't worry.... I just need to learn AGI...
10:33.48HjemmeRoyKNebukadneza: how can parsing be difficult?
10:34.26NebukadnezaHjemmeRoyK: it isnt if your a really good programmer ...
10:34.46DHuangHRoyK: maybe you can help..... All I want it to have retreive username/callerid  from sip.conf given a Context
10:34.47Nebukadnezabut i am not one of those :/
10:35.30Nebukadneza:/ okay - then ill continue writing my own agi
10:36.20DHuanghehehe..... good practice...
10:38.22PTG123man
10:38.28PTG123if only i knew why callerid isn't working
10:38.50DHuangjust set it b4 dialing
10:40.06*** join/#asterisk wvital (~wvital@wvital.net1.nerim.net)
10:40.24PTG123um
10:40.29PTG123well i set caller id in the sip peer :)
10:40.32PTG123yet it doesn't set it
10:40.50DHuangno set it before Dial() command
10:42.13*** part/#asterisk wvital (~wvital@wvital.net1.nerim.net)
10:42.21*** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net)
10:43.53*** join/#asterisk mesi (~player@dsl-082-083-054-206.arcor-ip.net)
10:44.48PTG123i use the same dial for everyone
10:44.53PTG123so i need to set it on account leevel
10:45.08*** join/#asterisk pif (~pif@zenon.apartia.fr)
10:45.14DHuangYa... so set it b4 you dial :-)
10:45.15HjemmeRoyKNebukadneza: really good my arse
10:45.22Nebukadnezaarse? Oo
10:45.50PTG123it usually works
10:45.53PTG123don't know why it doesn't now
10:46.04DHuangneed to set callerid=yes
10:46.05PTG123i wish i knew what command was called
10:46.09PTG123when you do a dial from a sip phone
10:46.19PTG123where do you set callerid=yes?
10:47.03DHuangsorry.. that's for zapata
10:47.06DHuangnot SIP
10:47.09PTG123heh
10:47.12PTG123damn you
10:47.15PTG123got me excite d;)
10:47.21DHuang:-o
10:49.00HjemmeRoyKNebukadneza: what is so hard about the parsing? do you need to parse it outside asterisk?
10:49.14NebukadnezaHjemmeRoyK: jep - with python
10:49.23Nebukadnezaand astconfig.py (from pyst) seems kinda ... shitty?
10:49.48HjemmeRoyKwhy not perl?
10:49.55HjemmeRoyKperl is fscking desiged for parsing
10:50.30*** join/#asterisk datareactor (datareacto@203.81.192.33)
10:50.41Nebukadnezai dont like spaghetti
10:50.46datareactorhow can i implement call waiting in *
10:51.02NebukadnezaWait(seconds)
10:51.14Nebukadnezaor what do you mean?
10:52.30datareactorAlerts the user of an incomming call while participating in another call, and allows the user to switch back and forth
10:54.03PTG123shit
10:54.07PTG123i wonder if its my problem
10:54.09PTG123or my provider
10:54.48DHuangI thinik your phone have to support it
10:54.50PTG123is there something i can put in the iax entry
10:54.52PTG123to set it by hand
10:58.48*** join/#asterisk SHiSo (Lord@did75-11-82-231-40-252.fbx.proxad.net)
10:58.49PTG123man guess i will workj on this in morning
10:58.51PTG123lame :)
10:58.54DHuangPTG: I have the same problem.... ehhehe
10:59.25DHuangthat's why I want to AGI to do the lookup so can set callerid
10:59.49PTG123well it use to work
10:59.53PTG123but i wrote all my own stuff
10:59.58PTG123and i am afraid thats causingt he issue
11:00.03DHuanghehehe
11:00.06PTG123i was happy to have incoming calls working today :)
11:00.44tuxinator_linuxAny of you build * on 2.6?
11:01.02DHuangyeap 2.6.9-041221
11:01.16DHuangalso on 2.4.x
11:01.37tuxinator_linuxDo I need to "  ln -s /lib/modules/`uname -r`/build" like it says at http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x123.html
11:02.03DHuangthe make exe should detect it
11:02.09tuxinator_linuxcool
11:02.21tuxinator_linuxI just finishd installing CentOS
11:02.39Mavvieaha, /etc/shadow
11:02.41tuxinator_linuxso I am ready to install my Dev Kit and *
11:04.01*** join/#asterisk LarsAC (~chatzilla@pD95004EB.dip0.t-ipconnect.de)
11:04.48DHuanghehehe..... asteriskathome is good too
11:05.26tuxinator_linuxI hear that
11:05.50tuxinator_linuxI am running yum update now
11:05.52LarsAChow can I change the message of my voicebox ?
11:06.33DHuangupload your own msg
11:06.43DHuangyum is good
11:06.58tuxinator_linuxyum sure makes it easy
11:07.22HjemmeRoyKNebukadneza: but... if you're having problem parsing the asterisk config.......
11:07.32LarsACyum ?
11:07.46HjemmeRoyKyummy?
11:08.11HjemmeRoyK~yum?
11:08.17HjemmeRoyK~lart LarsAC
11:08.24*** join/#asterisk shaZwaz (~adnans@203.81.196.167)
11:08.48shaZwazhi room
11:09.02tuxinator_linuxhttp://linux.duke.edu/projects/yum/
11:09.18DHuangyum yum...
11:09.53tuxinator_linuxSould I play with latest CVS for both Digium driver and *?
11:10.19DHuangfor non-production why not
11:10.56tuxinator_linuxDo I put the dev kit in first or do software first?
11:11.01tuxinator_linuxIt says hardware
11:12.29tuxinator_linuxokay, I will go put in hardware now
11:12.49Nebukadnezaargh! the festival TTS is cruel!
11:13.04DHuangya.. don't like festival.. too much work
11:13.34Mavviefestival > my voice :-)
11:16.50shaZwaz~kill HjemmeRoyK
11:16.52jbotACTION shoots a excited anti-neutron gun at HjemmeRoyK
11:20.09tuxinator_linuxHjemmeRoyK: did it hurt?
11:20.36DHuang~kill me
11:20.38jbotACTION shoots a hyper-charged fluxelectron gun at dhuang
11:23.50tuxinator_linux~kill windows
11:23.51jbotACTION shoots a magneto-ionized pseudophoton gun at windows
11:24.29tuxinator_linux~weather KSJC
11:24.46DHuang~weather YPPH
11:24.47tuxinator_linuxa little chilly for VON
11:25.20tuxinator_linuxwarm there
11:25.32DHuangcold for end of summer....
11:25.40tuxinator_linuxthe seasons are switched, aren't they?
11:25.58DHuangthe weather is stuffed..
11:26.21DHuangsome parts of Taiwan is snowing.... 1st snow in 100 years
11:26.29tuxinator_linuxwierd
11:26.51tuxinator_linuxHey, I got an error while compiling zaptel
11:27.06tuxinator_linux[install] error 127
11:27.42DHuangzaptel doesn't work on 2.6 I tthink
11:28.00tuxinator_linuxhmm
11:31.47*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:32.48*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
11:33.11tuxinator_linuxhttp://www.voip-info.org/tiki-index.php?page=Asterisk+Zaptel+Installation, found some info
11:33.51tuxinator_linuxtrying it now
11:34.16DHuangif it work let me know.. then I'll upgrade my kernel too
11:35.46*** join/#asterisk ClubBarf (~me@195.157.221.139)
11:36.24tuxinator_linuxI get the same error, '/bin/sh: restorecon: command not found' 'make: *** [install] Error 127'
11:36.36ClubBarfAnyone here use X-Pro or Eyebeam with Asterisk?  I'm seeing weirdness when I try to transfer a call...
11:36.41DHuanghehehe.. need to install dev stuff
11:37.49tuxinator_linuxany clue?
11:38.00DHuangyum install make
11:38.30tuxinator_linux"nothing to do"
11:39.13DHuangyum install "missing command"
11:39.37HjemmeRoyK~lart himself
11:41.21ClubBarfAnyone here know why I can transfer an incoming call from X-Pro, but not an outgoing call.
11:41.23ClubBarf?
11:41.37*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
11:41.49DHuangya... b'cos it's Asterisk
11:41.59ClubBarfHelpfull.
11:42.04ClubBarfReally.
11:42.09DHuangI know.. ppl been very helpful here
11:42.20DHuangI've been here for the past 3 hours.. and no one can help me
11:42.33ClubBarfWhat's up with your box?
11:42.58DHuangnot my box... looking for AGI.... to lookup username/callerid in sip peer
11:43.58Zeeekwhy?
11:44.19DHuangZeeek: so when you dial out by IAX it will set to the right callerid...
11:44.22Zeeek(sometimes it helps to know)
11:44.39DHuangby that context
11:44.47Zeeeklet's look at a black box: the input is?
11:45.23Zeeekbtw how many users?
11:45.33Zeeek1à or 10,000?
11:45.41DHuangokay... input ${CONTEXT}   --> output SetCallidNum(of that context)
11:45.41Zeeek10 or 10,000?
11:46.01DHuangfor anyone that is dialing out of that context
11:46.06Zeeekhow many users?
11:46.35datareactorhi Zeeek i have question ?
11:46.36Zeeekhow many contexts?
11:47.29Zeeekwait that doesn't make sense... if the CID is set because of a context, it could just be set in the contex
11:47.36Zeeekso I seem to be missing some vital info
11:47.41datareactorhow can i implement call waiting in *
11:47.43Zeeekhi data
11:48.04Zeeekdata in what technology? It's ont he conf files
11:48.15Zeeekzapata.conf, sip.conf, iax.conf
11:48.30Zeeek<PROTECTED>
11:48.36DHuangin sip.conf or iax.conf
11:48.51DHuanguser = as many
11:48.52Zeeekcallwating=yes
11:48.56*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
11:48.57DHuangcontexts = 1000
11:49.07Zeeekthere are 1000 named contexts?
11:49.23DHuangZeek yes, each user has their own context
11:49.43Zeeekare you using some database or realtime stuff?
11:50.33DHuangusing realtime
11:50.45ZeeekI'm not qualified to help then, sorry
11:51.07DHuangthat doesn't matter... how would you use if it's using sip.conf???
11:51.30ZeeekI'd put it in the context in extensions
11:51.39Zeeekor look it up in astdb
11:51.55Zeeekfor that matter if it's callerid that IS in sip.conf
11:52.03Zeeekcallerid=
11:52.42ZeeekHej!
11:52.46HjemmeRoyK:)
11:52.52HjemmeRoyKhei
11:52.59ZeeekHay
11:53.34ZeeekDHuang if you have a config for each user insip.conf there is a callerid=Wilson Stark <2003>
11:53.52Zeeekso now I'm really lost as to why you need AGI
11:56.16ClubBarfAnyone here using eyebeam or x-pro with *?
11:56.21Zeeekoh well, I guess I'll never know
11:56.29ZeeekX-Lite?
11:56.43ClubBarfNo, x-lite doesn't have the transfer feature.
11:57.04ClubBarfI'm seeing oddness in x-lite and eyebeam when I transfer a call.
11:57.06Zeeekso wxhat's your problem?
11:57.08Zeeekooops
11:57.16Zeeekwhat about it
11:57.33DHuangZeek: but not working in IAX.CONF
11:57.42ClubBarfWhen I make a call, then try to transfer it, the call gets dropped.  If the call is incoming, it can transfer fine.
11:57.47DHuang:-p
11:57.51datareactorsorry i was away i am using sip sho
11:58.02DHuangCB: try to debug and see what's wrong
11:58.03datareactorsdsds
11:58.20*** join/#asterisk mrempire (~user1@h71032.upc-h.chello.nl)
11:58.20ClubBarfI have, but * isn't telling me anything usefull.
11:58.33DHuangCB: is it SIP or IAX?
11:59.00ClubBarfSIP.
11:59.07ClubBarf<PROTECTED>
11:59.38ClubBarf's' fails, then it falls back to default, which fails.
12:00.01ClubBarfBut if the call is incoming to the softphone, it transfers fine.
12:00.12DHuangOh...  ok.. try    sip debug     and look at the message coming back from the phone
12:00.32DHuangnormally tells you the error msg in the SIP header
12:02.26DHuangZeek: any idea??? set CallerID in iax call?
12:04.31datareactorZeeek u there
12:04.38Zeeekset callerid in iax.conf
12:04.55DHuangtried.. and not working :-(
12:05.03Zeeeksomething s wrong then
12:05.10Zeeekwhat do you set it to for example?
12:05.25DHuangcallerid=13760
12:05.39datareactorshould i only need callwating=yes in sip.conf how can i switch b/w calls
12:05.43Zeeekusually is't callerid=Your Name <2035>
12:06.26*** join/#asterisk MikeJ[Jayden] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
12:06.26ZeeekI don't know if you can switch between calls
12:06.26DHuangzeek: shall try that.. thanks
12:06.26Zeeekdata^^^^^
12:06.47Zeeekthis is the way it says to set callerid in the many documents available on the web
12:06.50Zeeeksuch as
12:06.52ZeeekStarter tutorial:
12:06.52Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
12:06.52Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
12:06.52Zeeekhttp://www.automated.it/guidetoasterisk.htm
12:06.52ZeeekTHE reference of the moment:
12:06.53Zeeekhttp://www.asteriskdocs.org
12:07.08datareactorZeeek Ok
12:07.43Zeeekdata I don't know if you can switch
12:07.58ZeeekI'm sure you can in ZAP
12:08.00HjemmeRoyK~docs
12:08.00jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
12:08.09HjemmeRoyKdatareactor: what protocol?
12:08.17ZeeekSIP he said i think
12:08.19HjemmeRoyKzap? sip? mgcp? h323? iax? htcpcp?
12:08.41datareactorHyemmeRoyK sip
12:08.46ClubBarfNope, nothing new from the debug messages.
12:08.59HjemmeRoyKdatareactor: I beleive you can do it with using groups....
12:09.07ClubBarfGetting the same output as I got from the generic console.
12:09.13HjemmeRoyKand then trap calls if >1 is active
12:09.26HjemmeRoyKin EAGI or something, playing a MEEEEEP to the user
12:10.02datareactorHjemeeRoyK can you explain little further
12:13.04HjemmeRoyKdatareactor: I don't know how to do it, really, but I beleive it can be done with using http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup and similar to limit the number of calls to each SIP user, and then, if limit reached, signal caller that number is busy, and signal callie with a MEEEEEEP somehow
12:13.10HjemmeRoyKdatareactor: but do some research :)
12:13.34datareactorHyemmeRoyk Thanks buddy
12:14.06HjemmeRoyKnp :)
12:14.21HjemmeRoyKdatareactor: but please tell me if you find a good solution. I need this as well :)
12:14.29HjemmeRoyKjust haven't had time to get around writing it
12:15.11datareactorHyemmRoyK i will send u
12:18.51DHuangZeek: the callerid is the phone you are calling from.... I want to set to is set to the callerid of the outgoing Context provider.
12:19.10shaZwazhi Zeeek
12:19.25Zeeekyeah so set it in a macro?
12:19.35Zeeekinclude the dial command
12:19.38DHuangeg. your phone callerid = 1234  and your outgoing fwd-outgoing context is callerid=192939
12:19.48ZeeekI do that all the time
12:19.51DHuangZeek: sorry... any hint?
12:19.53HjemmeRoyKanyone that knows how I can send a MEEEP to an active SIP channel?
12:19.57Zeeekmake a macro to call each provider
12:20.08DHuangyou have a script handy??
12:20.12Zeeekoput a setcallerid() in the macro before the dial command
12:20.15*** join/#asterisk TheEmperor (TheEmperor@218.111.48.52)
12:20.28Zeeekno but there are ample docs you might want to take time to read:
12:20.35Zeeekhttp://asteriskdocs.org
12:20.40ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
12:20.54DHuangZeek: kewl.. thanks..
12:23.21ClubBarfAnyone know what the heck would cause this:
12:23.24ClubBarfStarting SIP/2001-b2bf at from-sip,2000,0 failed so falling back to exten 's'
12:23.50*** join/#asterisk hemant (hemant@220.226.53.133)
12:23.53hemanthi all
12:25.35ClubBarfHey, hemant.
12:26.13DHuangZeek: so how you lookup the callerid in certain "CONTEXT" ???
12:26.57ClubBarfAnyone know of a sip softphone I can test with asterisk (not x-lite, x-pro or eyebeam)
12:26.58ClubBarf?
12:27.12DHuangfirefly
12:27.22ClubBarfDoes that support call transfer?
12:27.34DHuangdon't know.. never tried
12:27.46ClubBarf:p
12:28.01ClubBarfI think the xten softphones may be bugged.
12:28.19DHuangfirefly support G729 without paying
12:28.22ClubBarfMy hardware phone doesn't have the same call transfer problem.
12:28.41ClubBarfHmmm...
12:28.44ClubBarfInteresting...
12:29.22DHuangOh well...  I'll resolve this callerid issue with string matching then.... Thanks...
12:30.14*** join/#asterisk anarcat (~anarcat@anarcat.mtl.istop.com)
12:30.24HjemmeRoyKhm
12:30.28anarcathello
12:30.37anarcatare iaxclient questions offtopic here?
12:30.44ClubBarfDoes no-one use eyebeam at all?
12:30.53HjemmeRoyKanyone here that uses the "context=" argument in queues.conf?
12:34.27*** part/#asterisk anarcat (~anarcat@anarcat.mtl.istop.com)
12:35.58HjemmeRoyKdatareactor: getting somewhere?
12:37.08MikeJ[Jayden]:).. playing with the jbot
12:37.08tuxinator_linuxDo I need to 'modprobe' on a "Dev kit" install?
12:37.32MikeJ[Jayden]modprobe loads drivers, so if you want to use the cards, yes
12:37.45tuxinator_linuxdon't I sound silly
12:37.54HjemmeRoyKhm
12:37.55HjemmeRoyKbriiz-custom has 0 calls (max unlimited) in 'ringall' strategy (11s holdtime), C:2, A:2, SL:0.0% within 0s
12:38.26HjemmeRoyKops
12:38.27HjemmeRoyKsorry
12:38.27HjemmeRoyKbut..... what does the C:2 A:2 SL:x 0s mean?
12:38.40DHuangOkay.... I'm off.... l8r guys
12:39.47mesiexten => 301,2,Dial(Modem/ttyI1/1234:14,7) gives me an error message: Requested device
12:39.49mesi<PROTECTED>
12:40.31mesiBut 1234 shouldn't be the device, but the outgoing msn ?!?
12:40.36ClubBarfFirefly doesn't seem to have a transfer call option.  Any other sip softphones (not firefly, X-Lite/X-pro/eyebeam)
12:41.30MikeJ[Jayden]cluebarf, the wiki will set you free
12:41.31mesiClubBarf: Many sip phones lack such features. Minisip doesn't even have a dtmf keypad :-( And kphone somehow cannot really send keypresses.
12:41.31MikeJ[Jayden]~docs
12:41.44jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
12:41.47HjemmeRoyK~lart mesi
12:43.50tuxinator_linuxI'm not sure which driver to load for the 'dev kit', any ideas?
12:44.18MikeJ[Jayden]tdm...
12:45.22tuxinator_linuxit's a tdm400p
12:45.46MikeJ[Jayden]y
12:45.57MikeJ[Jayden]wctdm
12:46.16tuxinator_linuxfound it, sorry
12:46.19MikeJ[Jayden]and zaptel
12:46.31MikeJ[Jayden]~cookie
12:46.32tuxinator_linuxthanks MikeJ[Jayden]
12:48.01*** join/#asterisk afe ([tyNO6myLG@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
12:48.06tuxinator_linuxerror, http://pastebin.com/249644
12:48.32tuxinator_linuxdo I need to configure zaptel.conf first?
12:49.49MikeJ[Jayden]heh
12:49.58MikeJ[Jayden]upgrade your kernel lately?
12:50.58tuxinator_linuxlooks like I need to do zaptel.conf first
12:51.25MikeJ[Jayden]yes, but that error looks more like zaptel compiled against wrong source
12:51.27tuxinator_linuxThe card is lit now
12:51.46tuxinator_linuxI am running 2.6
12:51.52MikeJ[Jayden]uname -r
12:52.12MikeJ[Jayden]and what kernel sourcecode do you have in /usr/src/
12:52.17tuxinator_linux2.6.9-5.0.3.EL
12:52.20Ron-NaI tried to switch to realtime, ... just with one sip phone out of sip.conf and into the database, added res_mysql.conf with the database info, ... but now this phone cannot register, ... anybody out who can help?
12:52.26tuxinator_linuxIt is working better
12:52.30MikeJ[Jayden]ok
12:52.36tuxinator_linuxI can push buttons on the phone
12:52.49tuxinator_linuxno dialtone yet, but I will read the docs
12:52.55MikeJ[Jayden]don't know res_msysql
12:53.12*** join/#asterisk afe ([tAVpE8ADX@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se)
12:53.43Ron-NaI just copied it over from res_odbc.conf
12:53.44MikeJ[Jayden]did you do the stuff in extconfig.conf
12:54.33Ron-NaMike yes, sipfriends, ... (three lines)
12:55.01MikeJ[Jayden]?
12:55.16*** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk)
12:55.33DrPeteWhats the reset key combo for a 7960, i forgot
12:56.01MikeJ[Jayden]you need 4 hands to do it
12:56.12MikeJ[Jayden]application # somthing
12:56.15DrPetehehe yeah, i remeber, whats the keys
12:58.11MikeJ[Jayden]don't remember
13:01.47tuxinator_linuxhmmm, take a look, http://pastebin.com/249647
13:03.28*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
13:05.18tuxinator_linuxcould someone translate 'line 0: Unable to open master device '/dev/zap/ctl' '
13:06.41MikeJ[Jayden]I still think you do not have zaptel compiled against the right source
13:06.48*** join/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk)
13:06.50tuxinator_linuxhhm
13:07.15MikeJ[Jayden]rewind 15 minutes....
13:07.34__Sparks_Is it possible to setup speed dials with asterisk, that will follow the rules in extensions.conf?
13:08.10__Sparks_For example, dialling *001 will call a number, but follow the rules set out in extensions.conf?
13:13.06__Sparks_I know I can get *001 to call a specific number, routing a specifit way, I just want it to follow the normal rules if that number was dialled normally
13:24.19*** join/#asterisk ckruetze_ (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
13:29.58*** join/#asterisk hemant (hemant@220.226.22.106)
13:33.10X-Genho hum
13:41.35*** join/#asterisk cjk (~cjk@80.92.75.91)
13:41.47cjkhi
13:42.29cjkis there a way to start the mp3players for moh on a remote server and just get the music tream via a tcp/ip socket. or something similar?
13:44.08*** join/#asterisk Damin_Mobile (~pocketirc@172.sub-166-155-136.myvzw.com)
13:44.38MikeJ[Jayden]cjk.. .there is a patch in mantis for this, it is against head..
13:45.06MikeJ[Jayden]a few of us have been discussing alternate ways to do this, so there will likely be somthing soon
13:45.21Damin_MobileHoly Crap! IRC from my PDA!
13:45.55Damin_MobileTHIS
13:46.17Damin_MobileROCKS!
13:46.21fileDamin_Mobile: where are you? lol
13:46.38Damin_MobileThe Restroom
13:47.18filewhen do you get in tomorrow?
13:49.04MikeJ[Jayden]cortesy flush please
13:49.20*** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com)
13:50.47cjkMikeJ[Jayden], thanks
13:51.11cjkis it possible to share the astdb file amongst different asterisk servers?
13:51.16Damin_Mobile9:30 pm
13:51.35fileDamin_Mobile: better then my flight for sure
13:52.14Damin_Mobilecjk: Try Realtime w/ mysql in cvs head
13:52.17__Sparks_Any idea why this doesn't work - exten => _0104.,1,Dial(IAX2/call1899/01234567${EXTEN:3}) - It should send 401234567 - but doesnt?
13:52.33*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)
13:52.52Zeeekwhat does it send?
13:52.52file__Sparks_: it should send 012345674 and whatever else you entered after the 4
13:53.28__Sparks_file - but doesnt the :3 mean it will remove three digits, nor four?
13:53.38afeI know this is off topic, but is here anyone with the time to help me with a small postfix problem?
13:53.42file__Sparks_: yes
13:54.21cjkDamin_Mobile, yeah thats what i will try. i just wanted to know if it could be possible with ths astdb file shard on a common nas
13:54.23__Sparks_oh i see, sorry!!
13:55.01sezuanHi! I try use iaxComm with asterisk 1.0.6(both under linux), but the voice quality is very poor. Looks like many dropped packets. Is there an issue with Kernel 2.6 or ALSA?
13:55.45__Sparks_File, sorry, this is what i want - 01234567 4 then the rest - but it doesn't seen to work
13:55.54bkw_file
13:55.57bkw_whats up
13:56.11file__Sparks_: it should, cause that's normal stuff everyone does... say what it's doing instead
13:56.20filebkw_: just awoke, thinking about tomorrow
13:56.25bkw_hehe
13:56.47bkw_i'm about to early checkin for my flight tommorow
13:57.01fileyou early bird you
13:57.13__Sparks_file, I get "WARNING[2877]: chan_iax2.c:5546 socket_read: Call rejected by 213.61.187.150: No such context/extension"
13:57.37*** join/#asterisk Luhiwu (~marsosa@200.63.89.243)
13:57.41bkw_duh
13:57.44file__Sparks_: read it.
13:57.47bkw_no such context/extension
13:58.15MikeJ[Jayden]:)
13:58.20bkw_also
13:58.23filebkw_: what are we doing Monday?
13:58.25bkw_who told you to dial IAX/peer/exten?
13:58.26MikeJ[Jayden]I see everyone is quite chipper this morning
13:58.30bkw_you should dial IAX2/user@peer/exten
13:58.55bkw_file we are gonna take over the town
13:59.12fileand break the hotel phone system?
13:59.17bkw_na replace it
13:59.23bkw_with aterisk while they aren't lookin
13:59.26filehold it hostage!
13:59.32__Sparks_File, but if i have the following "exten => _*04.,1,Dial(IAX2/call1899/${EXTEN:3})" then dial *04 012345674.. it is fine
13:59.35bkw_STAY BACK OR TH EPHONE SYTEM GETS IT
13:59.44bkw___Sparks_, dude
13:59.47bkw_don't dial like that
13:59.50bkw_user@peer/exten
14:00.12*** mode/#asterisk [+o file] by bkw_
14:00.16__Sparks_this is for outgoing PSTN calls through my IAX account
14:00.19bkw_so
14:00.21file__Sparks_: your dialplan logic is screwed up, rethink it
14:00.45bkw_ok you're trying too hard
14:00.53bkw_thats usually what catches people
14:01.07bkw_be one with the PBX
14:01.11bkw_:P
14:01.15MikeJ[Jayden]use the force.
14:01.20__Sparks_file, I am unsure why "exten => _*04.,1,Dial(IAX2/call1899/${EXTEN:3})" is wrong - it works fine!!
14:01.34fileI meant your original thing...
14:01.45filethat you seem you want to use, because it's what you asked for help with
14:01.50bkw___Sparks_, it may work.... but its not wise to do that.. because the remote end isn't getting told a username in some cases...
14:02.09bkw_those cases are far and few .. but its better to be exact when possible
14:02.12filebkw_: that was so professional
14:02.25fileI wonder how many pairs of shorts I have
14:02.33bkw_file you don't need shorts
14:02.41bkw_just run around in a toga
14:02.45fileha
14:02.45bkw_:P
14:03.06afeTOGA TOGA TOGA!!!
14:03.19afeurmm... oops... got a bit carried away there
14:03.24__Sparks_what I am tyring to do with that is, a company I am working for has a large telephone list, with all their extentions staring with a 4. I want to ba able to call people there by dialling 010 (I would prefer *010) then their extention number
14:04.01__Sparks_Is there a better way?
14:04.06MikeJ[Jayden]why not just be able to dial the ext
14:04.11bkw___Sparks_, you lack some basic understanding of how things fit together.... give a few hours of tinkering with it.. and at some point it will all snap and you'll go "OH SHIT.. duh!"
14:04.35bkw_:P
14:04.42__Sparks_bkw_ you are quite correct - that's whay I am here asking for help ;-)
14:04.43bkw_and i'm not trying to be funny.. its how it usually happens.
14:04.54fileah The Killers
14:05.11bkw___Sparks_, well go to voip-info.org and look around at examples
14:05.34bkw_because what you're asking can't be told over IRC with ease without 1. your configs on a pastebin... 2. us in the box to show you.
14:05.35__Sparks_I tried searching there for "Speed Dial" but could find anything relevent :(
14:05.51bkw_wtf are you trying to do ?
14:05.55MikeJ[Jayden]file, which song?
14:06.07fileMikeJ[Jayden]: destiny is calling me...
14:06.10fileMr. Brightside
14:06.12MikeJ[Jayden]good tune...
14:06.14bkw_I see you lookin at me like i'm some kinda freak..... get up outta your seat...
14:06.25bkw_why don't ya do something
14:06.39MikeJ[Jayden]are they playing them on the real radio... w/ xm, I don't listen tothe real radio anymore...
14:06.45bkw_man.. my eye dried out really bad lastnight and my contact got stuck to my eye.. OUCH
14:06.46filenah - TV
14:06.47sezuanIs there a public asterisk where I can try iaxComm to check if it's a configuration problem with mine?
14:06.51MikeJ[Jayden]IT'S A DANCE PARTY!!! wooo hoo
14:07.26filebkw_: so apparently one of the people I've known for a good 4 or 5 years is gay, never had a clue!
14:07.53bkw_DING DING DING
14:08.02bkw_once your around me for a day.. you'll have gayday
14:08.05bkw_er gaydar
14:08.05bkw_haha
14:08.08filegayday, uh huh
14:08.12filegaydays more like it
14:08.18bkw_hahahahahah
14:08.18filesince I'm around you for 6 days :p
14:08.27bkw_well I meant gaydar
14:08.28file6 long agonizing days :p
14:08.35bkw_got the long part right :P
14:08.45MikeJ[Jayden]fish storries
14:08.46fileyeah, everyone but you :p
14:08.50__Sparks_I am trying to setup a speed dial list - dial *001 and you get connected to a normal PSTN destination, with the call rputing through the IAX provider "18866" - this works fine - then I want to be able to dial *010 then the extention number of a remote office, Asterisk then sends the call ober the IAX provider "18866" inserting the first part of the telephone number before the extention number - this is all routing to the PSTN
14:08.51bkw_haha
14:09.12MikeJ[Jayden]I feel so left out :(
14:09.25X-GenMikeJ[Jayden]: u gay ?
14:09.26bkw___Sparks_, you lack basic understanding of the dialplan.. post your extensions.conf to pastbin.ca
14:09.29MikeJ[Jayden]sparks, do the 4xxx ext overlap with anything local
14:09.32ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
14:09.35MikeJ[Jayden]gay is a funny word
14:09.40MikeJ[Jayden]I'm married
14:09.45bkw_queer is a weird word!
14:09.50bkw_:P
14:09.54fileyou say yes, I say no, you say start, I say go go go
14:10.00bkw_hahahahhah
14:10.12bkw_file is that a description of day 1?
14:10.18__Sparks_MikeJ[Jayden] - yes, that's why I want to use the *010 first
14:10.29filebkw_: corruption level has approached maximum
14:10.34bkw___Sparks_, put your extensions.conf on pastebin.ca
14:10.37bkw_we'll show you whats wrong
14:10.45bkw_file not just yet dear
14:10.51MikeJ[Jayden]why *010, why not somthing else, like 5 or somthing.. keep it simple,
14:10.58bkw_ya really
14:11.35bkw___Sparks_, lets see da conf file.. we fix you up right fast
14:11.36Zeeekwhy not *#*#
14:11.40bkw_um
14:11.45bkw_that don't work well with sip phones ya know
14:11.48bkw_:P
14:11.52__Sparks_MikeJ[Jayden] because I will have a lot of speed dials, and currently have *001, *002 etc pointing to other destinations
14:11.55Zeeekwhy not?
14:11.59MikeJ[Jayden]why not 23458786435765854758456
14:12.01__Sparks_bkw_ okay, let be go get it!!
14:12.09filethis term 'speed dials' is making me angry
14:12.13bkw_ya really
14:12.20bkw_SPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPPEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEED DIAL
14:12.26fileoh no, bkw is on speed!
14:12.28MikeJ[Jayden]jbot, kill the term speed dials forever
14:12.30jbotACTION shoots a charged pseudoelectron gun at the term speed dials forever
14:12.33Zeeekno 77333 which is "speed"
14:12.39MikeJ[Jayden]jbot, kill the term speed dials
14:12.41jbotACTION shoots a inverse anti-quark gun at the term speed dials
14:12.48bkw_~lart MikeJ[Jayden]
14:12.48MikeJ[Jayden]that's better
14:12.57Zeeekdial abbreviations
14:13.02bkw_~lart MikeJ[Jayden]
14:13.05Damin_MobileSo file. Do your parental units realize that you will be spending 6 days hanging  around  with a bunch of older, gay men?
14:13.08MikeJ[Jayden]~troutslap bkw_
14:13.10jbotACTION slaps bkw_ around with a large trout
14:13.10bkw_jbot you're dumb
14:13.20bkw_~moose
14:13.21jbotit has been said that moose is moose-penis
14:13.21fileDamin_Mobile: what, do you think I'm crazy?
14:13.30Damin_MobileIn a hotel ?
14:13.31MikeJ[Jayden]yes it has
14:13.41Damin_MobileHaha..
14:13.42bkw_haha
14:13.48fileanyone can keep bkw at bay... all you need is some Jager
14:13.53filethen hit him over the head with the bottle
14:13.58MikeJ[Jayden]nice
14:13.58bkw_WHAT?
14:13.59bkw_abuse
14:14.04bkw_man I thought you loved me
14:14.12bkw_:P
14:14.13filebkw_: hold me hold me hug me hug me
14:14.19MikeJ[Jayden]hey... bkw.. you are streaming from von arn't you//
14:14.19bkw_hehe
14:14.24bkw_MikeJ[Jayden], yes
14:14.35Damin_Mobilebkw: irc from the palm ;>
14:14.36MikeJ[Jayden]ok.. so I can feel involved..
14:14.38bkw_ibook + isight
14:14.47MikeJ[Jayden]:)
14:14.59MikeJ[Jayden]you around today, I need to go eat somthin
14:15.45bkw_just breathe on me...
14:15.49bkw_oh baby
14:15.51bkw_just breathe on me...
14:15.53MikeJ[Jayden]I did a good deed this morning, helped somone get zaptel running  :)  one good deed a day and thats it tho
14:16.16MikeJ[Jayden]felt good for a minute, now I just feel dirty :)
14:16.16bkw_MikeJ[Jayden], if thats the case.. i'm all petered out till like 2010
14:16.40*** join/#asterisk ClubBarf (~me@195.157.221.139)
14:16.43MikeJ[Jayden]heeh
14:16.45bkw_what really irks me is when you help someone and they send you 2 bucks or something for your time
14:16.49bkw_and you spent like 2 hours on it
14:16.50fileI need to take my accounts out of my local mailserver so it won't download mail from my ISP...
14:16.55MikeJ[Jayden]and you feel dirty
14:17.20MikeJ[Jayden]hey.. I take any money when none was expected
14:17.26bkw_people freak out when I tell them my rate is 120/hr
14:17.34MikeJ[Jayden]$.02 would probably piss me off
14:17.39bkw_I have had that too
14:17.42MikeJ[Jayden]120/hr is nothing for contract
14:17.56MikeJ[Jayden]I can do that just for basic router\firewall work
14:18.06bkw_ya but I require a two hour min.
14:18.08bkw_:P
14:18.09ClubBarfbkw_: is that dollars?
14:18.13bkw_yes
14:18.28bkw_I gotta eat somehow
14:18.31ClubBarfWhat's freaky about $120p/h?
14:18.32bkw_and afford more mac hardware
14:18.52filebkw_ is obsessed with mac hardware
14:18.58bkw_ClubBarf, people seem to think that since asterisk is open src.. that they should get help for free too
14:19.07bkw_I do give free help
14:19.17ClubBarfAh, you meant in here...
14:19.23bkw_yes
14:19.24ClubBarfI thought you meant your client base.
14:19.32Zeeekthere is an incredible amount of free help available - better than 99% of pais software
14:19.41Zeeek<PROTECTED>
14:19.42bkw_what really ticks me off too is when i say I and instantly I have 10 people private msg me asking for help
14:19.42ClubBarfThat's what you get for coming in half way through a conversation...
14:19.43*** join/#asterisk sezuan (sezuan@port-212-202-57-119.dynamic.qsc.de)
14:19.47MikeJ[Jayden]pais software :)
14:19.57bkw_paid
14:19.58bkw_you ninny
14:20.00bkw_:P
14:20.04bkw_file is a whore
14:20.04ZeeekPersonal and Invisible Software
14:20.11bkw_haha
14:20.11fileOH NO!
14:20.16fileBoRiS is a whore.
14:20.20bkw_ONCE A WHORE!!!
14:20.20MikeJ[Jayden]now that's not nice
14:20.27MikeJ[Jayden]~file
14:20.28file*ALWAYS* A WHORE!
14:20.44Zeeekbkw_ turn off queries
14:20.44bkw_i'm not putting my new contacts in till I get to SJC
14:20.45X-Genbkw_: u lucky there aint a windows version of * ,imagine the support issues then
14:20.46Damin_Mobiledon't tell his parents
14:20.52bkw_X-Gen, their is
14:20.56bkw_thanks to Damin
14:20.58bkw_;P
14:21.00bkw_astwind
14:21.01X-GenNOOOO !
14:21.03fileI'm gone, brb
14:21.09bkw_it runs under co-linux on windows
14:21.22MikeJ[Jayden]asterisk runs ok on windows w/o co-linux
14:21.27__Sparks_bkw_ okay, it's here! - http://pastebin.ca/6901
14:21.50__Sparks_Line 177
14:21.53ClubBarfhey, stop splitting hairs, you know what the guy meant.
14:21.56bkw_ACK
14:21.59MikeJ[Jayden]:)
14:22.04bkw_someone needs to learn what Macro is
14:22.05ClubBarf:p
14:22.22tuxinator_linuxTime for bed, night all
14:22.25Damin_MobileAnd assoon as collnvx 0.6.2 comes out .there's another release
14:22.25bkw_exten => _*01.,1,Dial(SIP/${EXTEN:3}@sipgate1)
14:22.27bkw_that
14:22.29ClubBarfAnyone here use softphones with their *?
14:22.33bkw_exten => _*0104.,1,Dial(IAX2/call1899/01234567${EXTEN:4})
14:22.35bkw_see
14:22.39bkw_the . is bad
14:22.43bkw_if you dont knwo what it means
14:22.46bkw_now is *1 PSTN?
14:22.48bkw_for the US?
14:22.55bkw_er *01?
14:23.36MuppetMasterHello.
14:23.38bkw___Sparks_, please answer me
14:23.43bkw_is *01 the US pstn?
14:23.47MuppetMasterHas anyone had luck getting H.263 and specifically Eyebeam working with Asterisk?
14:23.52__Sparks_bkw_ yes
14:23.55bkw_ok
14:23.57MikeJ[Jayden]don't answer him :)  you'll regret it
14:23.57bkw_here is a flaw
14:24.01bkw_you don't use .
14:24.05MuppetMasterI have disallow=all and then allow=h263, etc.
14:24.08bkw_its bad unless you know what the hell you're doing
14:24.17bkw__*011NXXNXXXXXX,1,
14:24.20bkw_thats the proper way
14:24.29bkw_because the first *01 is winning
14:24.30ClubBarfOooh, I have a quick tip for any UK users (and maybe US users if you don't mind a touch of lag while your packets go to the UK and back) - call1899.com is doing completely free voip to pstn calls to landlines in the UK, US, Canada and Germany.
14:24.32bkw_vs the *0104
14:24.47__Sparks_okay, I see - I thought the extra 04 would make it route to the second one
14:24.51bkw_nope
14:24.57bkw_the first wins becuase it matches before the second one
14:25.04bkw_and the order in your dialplan can make a diff
14:25.12__Sparks_i see.
14:25.13bkw_if you move *0104 above the other
14:25.16bkw_it should
14:25.25MuppetMasterand I have videosupport=yes set
14:25.36bkw_MuppetMaster, please check the bug tracker
14:25.38bkw_before you ask
14:25.43bkw_their is a patch just for this on there
14:25.44bkw_GO TEST IT
14:25.46bkw_NEXT!!!
14:25.55ClubBarfNice.
14:26.23MuppetMasterBKW_:  Ah, didn't realize this would be a bug, as nothing on the wiki.  http://www.voip-info.org/tiki-index.php?page=Asterisk%20video
14:26.24ClubBarfReminds me of when I was tech support, yeeeears ago.  "You're fixed...  NEXT!"
14:26.40bkw_bugs.digium.com
14:26.40Zeeekwhat is the "catch" with 1899 that permits free calls?
14:26.46ClubBarfZeeek - I can't see one.
14:26.51__Sparks_bkw_, Thanks, I understand now :) - in your example "_*011NXXNXXXXXX,1," what are the N's for?
14:27.00GoshenMupperMaster: add it to the wiki :)
14:27.02bkw_read the extensions.conf.sample
14:27.05bkw_it has the info
14:27.07ClubBarfI've been using 18866 for a while (same company, different tarrifs)
14:27.07MuppetMasterbkw_:  what do I search for on the bug tracker?
14:27.08bkw_N=2-9
14:27.09__Sparks_Zeek - I am using them, and cant see any!
14:27.12bkw_Z=1-9
14:27.15MuppetMasterGoshen:  Will once I find it!
14:27.16bkw_X = 0 - 9
14:27.26bkw_NXXNXXXXXX is whats valid in the US
14:27.36__Sparks_thanks :)
14:27.38HjemmeRoyKÅ = Å
14:27.44ClubBarfOh, and Zeeek - the setup is ubereasy.  They seem to use * at their end.
14:27.57bkw_:P
14:28.06MuppetMasterbkw_:  Nothing on the bug tracker that I can find.  What is the keyword to search for?
14:28.06ClubBarfSo you can use SIP or IAX, I *think* (I've only tested SIP)
14:28.13bkw_MuppetMaster, you're blind as a bat
14:28.14bkw_hold please
14:28.16Zeeeknothing is free ever. It's a law of entropy
14:28.21bkw_http://bugs.digium.com/bug_view_page.php?bug_id=0003709
14:28.21Zeeekso....
14:28.23MuppetMasterbkw_:  i must be
14:28.35MuppetMasterbkw_:  Thanks!
14:28.37bkw_NEXT!!!
14:28.37drumkillabkw_: TOMORROW
14:28.42bkw_drumkilla, yes?
14:28.44drumkillabkw_: ring the bell!
14:28.50bkw_I'll bring it
14:28.51ClubBarfZeeek: if you use them via a normal phone line, they charge 3p per call.  I'm guessing the VoIP stuff will eventually match that.
14:29.03ClubBarfBut for now, it's fwee!
14:29.11drumkillabkw_: I don't need anything ... I'm just excited  :)
14:29.18ClubBarfExcept to mobiles, which are 10p/min, 2p/min weekends.
14:29.18Zeeekyou said you could call the UK and USA free? or did I misread
14:29.21bkw_me too
14:29.26bkw_but you reminded me to bring the bell
14:29.35bkw_and people can ring my bell if they like :P
14:29.37bkw_har har har
14:29.37ClubBarfUK, US, Canada and Germany, all free.
14:29.37fileyah! bring the bell!
14:29.42drumkillafile!
14:29.47fileRussell!
14:29.49bkw_drumkilla, what time do you get there?
14:29.52ClubBarfwww.call1899.com - take a look.
14:29.56drumkillabkw_: 1-something
14:30.00bkw_same here
14:30.02bkw_1:06
14:30.04drumkillaoh realllly
14:30.05Damin_Mobiledude
14:30.08drumkillalet me check
14:30.14bkw_what flight?
14:30.40bkw_say united flight 415
14:30.41bkw_haha
14:30.46filePlease hold.
14:30.47bkw_that would be freaky if it was
14:30.50bkw_hahaha
14:30.51MuppetMasterAdded the bug details to the Wiki, now will see if I can get it to work.
14:31.14drumkillabkw_: 1:44 PM
14:31.18bkw_kewl
14:31.21bkw_we can wait for you
14:31.22bkw_if you like
14:31.27bkw_twisted gets in like 30 min before me or so
14:31.30bkw_maybe an hour
14:31.36X-GenClubBarf: u need a UK number 2 use this service dont u ?
14:31.44drumkillawell that would be cool, then we can take the same cab
14:31.48ClubBarfYeah, seems like it.
14:31.49bkw_yep
14:31.50Zeeekit seems to be for BT users
14:31.53ClubBarfI used my mobile number.
14:31.54bkw_drumkilla, you have my cell?
14:32.02Damin_MobileTouch my Asterisk
14:32.04drumkillabkw_: nope
14:32.07bkw_you do now
14:32.08bkw_:P
14:32.10bkw_I have yours
14:32.30X-GenClubBarf: hook your mobile number to your * box so i can route through you please :)
14:32.32ClubBarfX-Gen: they wouldn't let me use my 0845 number either.
14:32.54ClubBarf:p
14:33.21bkw_another hour and I can check in on my flight
14:33.23bkw_:p
14:33.29ClubBarfI've done that already.  I can call someone in the US, and the number that the call originates from appears to be my mobile number.
14:33.44ClubBarfWhich isn't quite what I want...
14:33.48bkw_I can't handle this confusion.. i'm unable come and take me away...
14:33.55fileto come eh?
14:34.02ClubBarflol
14:34.07bkw_ya
14:34.08bkw_really
14:34.08drumkillawrong channel!
14:34.08Damin_Mobilebkw has your cell changed since astricon?
14:34.09bkw_you dirty bitch
14:34.23bkw_take this to #mp
14:34.33ClubBarfAnyone here use a softphone with their *?
14:34.51MuppetMasterGetting this on the latest head when I try to compile:  make: *** No rule to make target `include/asterisk/channel_pvt.h', needed by `channel.o'.  Stop.
14:35.04Damin_Mobilefile; dont worry. Ill meet you ar your gate even if Bkw goes mia.
14:35.11ZeeekCB I have but not X-Pro
14:35.14MuppetMasterClubBarf:  Yes, XPro and Eyebeam
14:35.24ClubBarfMuppetMaster: oooh, cool.
14:35.35fileDamin_Mobile: good good
14:35.43ClubBarfMuppetMaster: have you been able to transfer calls between softphones?
14:35.50MuppetMasterYes.
14:35.56MuppetMasterClubbarf: As well as off to a Sipura.
14:36.01__Sparks_bkw_, It workd now :-) - I was actually a definition for *01 to route calls somwhere else - but now I know not to do this :)
14:36.02filejust look for a confused teen that proceeds to take out a blueish wifi phone and look for a signal when he gets off the plane
14:36.12Damin_Mobilebkw; you really do need the moose penis sign though.
14:36.21MuppetMasterSo is there an issue with the latest CVS HEAD?
14:36.43ClubBarfMuppetMaster: when I dial out using my softphone (kphone, x-pro or eyebeam) I can't transfer the call.  If the call is incoming, I can.
14:36.54ClubBarfMuppetMaster: You havn't seen anything like that?
14:37.05MuppetMasterClubBarf:  No, I have not.
14:37.16X-Geni got HEAD lastnight, and it didnt run, i had to delete the modules before rebuilding it
14:37.18ClubBarfwhat build are you using?
14:37.18Damin_MobileAlright.... time to sign off and do laundary.
14:37.41MuppetMasterClubbarf:  You do have cantransfer=yes in your sip.conf for the softhphone clients you are using?
14:37.48MikeJ[Jayden]bkw, what are the new dates for Chicago again?
14:38.04ClubBarfnope.
14:38.20__Sparks_ClubBarf, X-Gen, - sign up with SipGate, and get a free UK PSTN Number
14:38.52MuppetMasterVoipuser.org is also good for UK #s.
14:39.02Zeeekthose numbers are expensive to the caller
14:39.05ClubBarf__Sparks_: the number has to be a geographic or mobile number - 0845 and 0870 are no-go with call1899.
14:39.12*** join/#asterisk SuShI`` (~sksushi@ANice-106-1-40-211.w80-11.abo.wanadoo.fr)
14:39.14MuppetMasterZeek:  The 0844 is 3pc per minute.
14:39.19SuShI``hi ??
14:39.20Zeeekthat's expensive
14:39.27ClubBarfHey SuShI``
14:39.36__Sparks_ClubBarf, you can habe a UK Geographic number with Sipgate for Free!
14:39.37MuppetMasterZeek:  Cheaper than calling my mobile outside of the UK.
14:39.51MuppetMasterYes, I do have a Reading number with Sipgate.
14:39.51ClubBarf__Sparks_: I may have to sign up with that then...
14:39.52*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
14:40.02Zeeeklook for voipjet mobile pricing
14:40.17MuppetMasterZeek:  Voipjet mobile pricing is more than 3pc/min.
14:40.19ZeeekI can call mobiles in France cheaper than calling them from France
14:40.23SuShI``i need someone  who can help me connect my telip
14:40.32ClubBarfMuppetMaster: The really odd thing is that I CAN transfer calls from my softphones, as long as the call in inbound to that softphone, and not originating there.
14:40.35__Sparks_ClubBarf, I use them for incomming, and 1899 for outgoing (to the UK)
14:40.43MuppetMasterClubbarf:  That is strange.
14:40.53ClubBarfMuppetMaster: Isn't it?!?!?!
14:41.08Zeeekin UK does the called cellphone pay too? Or just the caller?
14:41.20MuppetMasterZeek:  Just the caller.
14:41.21ClubBarfIt's like the harpy of softphones jumped into my * and decided to mess with my head.
14:41.30MuppetMasterZeek:  Unless you are roaming outside of the UK, then both.
14:41.34Zeeekok. Cause in the States the called phone pays too in most plans
14:41.39Zeeekwhich SUCKS
14:41.46SuShI``are there a tutorial for asteriskN
14:41.47SuShI``??
14:41.48MuppetMasterZeek:  Correct, in the US the caller doesn't know the difference.
14:41.48Zeeekthey think it's "just minutes"
14:42.01Zeeekbut minutes aren't free
14:42.03ClubBarfZeeek: the called person only gets charged if their mobile isn't a UK mobile - i.e. they're using global roaming.
14:42.07ZeeekStarter tutorial:
14:42.07Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
14:42.07Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
14:42.07Zeeekhttp://www.automated.it/guidetoasterisk.htm
14:42.07ZeeekTHE reference of the moment:
14:42.08Zeeekhttp://www.asteriskdocs.org
14:42.15ZeeekSuShi^^^^^^^^^^^
14:42.24SuShI``thx :D
14:42.29Zeeekyeah roaming is exp everywhere
14:43.01ClubBarfIt's often cheaper to buy a SIM card in the country you're in on a pay-as-you-go contract, and throw it away when you go home.
14:43.06ZeeekAgreed
14:43.20Zeeekin fact you can buy a phone and a SIM here for $85 or so
14:43.22filethat's what I'm doing for VON
14:43.29filejust grabbing a T-Mobile SIM card
14:43.38ClubBarfAlthough that means you have to leave a voicemail on your old sim telling everyone your new temporary number...
14:43.41cjkwhere can i get a list of predefined variables in asterisk?
14:43.42MuppetMasterGood call, when I did a make clean and then a make this problem went away:  make: *** No rule to make target `include/asterisk/channel_pvt.h', needed by `channel.o'.  Stop.
14:43.47MuppetMasterOn the latest CVS HEAD
14:43.57filealways do make clean
14:44.06ClubBarfMuppetMaster: which CVS release are you using?
14:44.08Zeeekwhat does make dirty do?
14:44.12X-GenMuppetMaster: SHOW ME THE $$$ :D
14:44.15ClubBarfMy build is mid december.
14:44.17MuppetMasterClubbarf:  I just downloaded, so the one from today.\
14:44.21ClubBarfMaybe that's my problem.
14:44.24MuppetMasterX-Gen:  A big thanks!
14:44.25bjohnsonZeeek: gets you arrested
14:44.25X-GenZeeek: shows u a piccie of dkw
14:44.41ClubBarfI'll have to rebuild from a new CVS snapshot and try again.
14:44.53Zeeekheh I just saw a new subject (for you guys!) on the blog thing I use:
14:44.55Zeeek"penis"
14:44.58X-Gensipgate only gives u a german number
14:45.14Zeeekthe person said "hey since there is a group called "pussy" I though we should have this too
14:45.35Zeeeksipgate has Swiss and UK stoo
14:45.55Zeeeksipgate was a little flaky when I tried it several months ago
14:46.03*** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:2)
14:46.51Zeeek<PROTECTED>
14:47.06SuShI``a tutorial for h323 ?
14:47.09Nebukadnezahm ...
14:47.11SuShI``i am too noob :(
14:47.19Zeeekdon't use h323
14:47.21Nebukadnezamay someone help me to test my SIP setup?
14:47.49SuShI``my telephone are only on h323
14:47.57Zeeekget new phones
14:48.03MuppetMasterSo, which of the three files do I use to do the patch?  http://bugs.digium.com/bug_view_page.php?bug_id=0003709
14:48.31ClubBarfNebukadneza: sure.
14:48.34SuShI``and IaX ?
14:48.38SuShI``what this?
14:48.49NebukadnezaClubBarf: can you call 01@nebuk.homelinux.org via SIP?
14:49.13ZeeekSuShl I gave you the stuff to read
14:49.24Zeeekplease go read some of it and you'll know
14:49.35SuShI``oki no problem :D
14:49.41Zeeeknone at all
14:49.49NebukadnezaClubBarf: not working? :(
14:50.06ClubBarfJust looking up nebuk.homelinux.org
14:50.28Nebukadnezaah kay
14:50.31ClubBarfIt's not a VoIP provider, is it?
14:50.39Nebukadnezaits my homeip via dyndns :)
14:50.52Nebukadneza01@62.214.231.243  should work too
14:50.57ClubBarfI'm not sure how I would dial that, is all - I have a hardphone.
14:51.08Nebukadnezaoh okay
14:51.11Nebukadnezadont know that on hardphones too
14:51.19ZeeekClubBarf add an extension in the dialplan
14:51.20*** join/#asterisk fugitivo (~ajf@201.255.108.146)
14:51.21ClubBarfMy software phone has no microphone.
14:51.28dan2bkw_: ping
14:51.59NebukadnezaClubBarf: hm ... thats bad
14:52.00Nebukadnezabut thanks
14:52.13dan2does asterisk have sip invite auth?
14:52.43ClubBarfNebukadneza: my software phone is purely for testing.  It doens't need a microphone.
14:52.48*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
14:52.59lesouvageI have a problem with getting musiconhold up and running. I have everything in place (mpg123, musiconhold=default in zapata.conf, uncoment the default line in musiconhold.conf and some example .mp3 in /usr/share/asterisk/mohmp3, symlink mpg123 in /usr/bin) On the asterisk command line I got the message Mar  5 07:29:22 NOTICE3126: res_musiconhold.c309 monmp3thread: Request to schedule in the...
14:53.01lesouvage...past?!?!.  I run xorcom asterisk and had musiconhold working. What am I doing wrong?
14:53.04NebukadnezaClubBarf: hm ... doesnt matter if you have a mic :) if you hear me thats enough for testing
14:53.44Zeeeklesouvage : http://www.voip-info.org/wiki-Asterisk+Request+to+schedule+in+the+past
14:54.57ClubBarfNebukadneza: I'm having trouble authing with your *
14:55.10NebukadnezaClubBarf: hm ... normally you dont need to auth
14:55.12ClubBarfIt's prolly my sip.conf - hang on a mo.
14:55.19Nebukadnezahm okay
14:55.26Nebukadneza(thanks)
14:57.23ClubBarfNebukadneza: can you hear me?
14:57.27NebukadnezaClubBarf: more or less
14:58.40NebukadnezaClubBarf: better here :)
14:58.55ClubBarfI couldn't make out anything you were saying.
14:59.02Nebukadnezayeah ... i got that :)
14:59.02ClubBarfIt was stuttering badly
14:59.10Nebukadnezaouh ...
14:59.23Nebukadneza1000kbit down / ~350kbit up dsl line here
14:59.36Nebukadnezaulaw, alaw and gsm are allowed on my side
14:59.37ClubBarfEither half of the UDP packets were going missing, or you're uploading a lot
14:59.41ClubBarfOr both...
14:59.53Nebukadnezahm sec - ill check my current upload
14:59.57ClubBarfI'll see if I can't get sip.conf configured to allow gsm only.
15:00.14Zeeekyou want to check that stuff, put an echo extension up for anyone to call
15:00.14ClubBarfThat should lower the bandwith requirements a lot.
15:00.39Nebukadnezaupsie ... my upload is about 26kbyte :/
15:00.48Nebukadnezathere we have the problem
15:01.13*** join/#asterisk coppice (~chatzilla@103.202.17.210.dyn.pacific.net.hk)
15:01.42ClubBarfNebukadneza: if you kill your upload for a little while, we can try again.
15:01.49Nebukadnezaat it ...
15:02.03Nebukadnezai played around a little with my htb/qos script
15:02.14ClubBarfI've set up sip.conf to allow only gsm between your * and mine.
15:02.20Nebukadnezakay
15:02.23ClubBarfThat might help.
15:02.44ClubBarfNope - same problem.
15:02.48Nebukadnezakay
15:02.54Nebukadnezai hear you more or less clear
15:03.05Nebukadnezadamned
15:03.06ClubBarfYour download is clear, that's why.
15:03.16Nebukadnezahm ...
15:03.26Nebukadnezamy upload is only about 15kbyte ...
15:03.54ClubBarfYeah, that *shouldn't* cause a problem, but who knows?
15:04.01Nebukadnezajeah
15:04.12Nebukadnezabut i cant figure out where the upload is coming from :)
15:04.14ClubBarfPlus for some reason it's still defaulting to ulaw
15:04.15Nebukadnezawait a min
15:04.18*** join/#asterisk eKo1 (~bernd@63.245.57.70)
15:04.38SuShI``Zeek : In sip.conf / general bindaddr = 0.0.0.0 must I remplace by local adress?
15:04.49Nebukadnezahm ... sec
15:04.53ClubBarf'k.
15:05.10eKo1Question: Say I'm in priority n and I have Dial(sip/...). If the call fails with a 4XX error, it just jump to priority n+1 right?
15:05.27eKo1s/just/should
15:05.36SuShI``Question: In sip.conf / general bindaddr = 0.0.0.0 must I remplace it by local adress?
15:05.41Nebukadnezaargh - still about 20k up
15:05.59Nebukadnezaand i dont know from where :/
15:06.05Zeeekusually you can leave 0.0.0.0
15:06.07ClubBarfyour firewall not telling you which IP on your lan it's coming from?
15:06.14Zeeekit means bind to all
15:07.00NebukadnezaClubBarf: there are too many connections to guess ... youve seen iptables debugging output?
15:07.04ZeeekSuShl fill in an address if you need to choose between several adapters
15:07.23ClubBarfYeah, I have...
15:07.41ClubBarfYou'll need to pipe it through grep -v to filter out the non-essential stuff.
15:07.51MuppetMasterWell, I downloaded and compiled the latest head and H.263 with Eyebeam still does not work (per this tweak http://bugs.digium.com/bug_view_page.php?bug_id=0003709).  Any ideas?
15:08.56ClubBarfI can't say I've tried getting video passthrough working with * yet.
15:09.13ClubBarfI intend to, but eyebeam doesn't seem to send DTMF properly to my * either.
15:09.21Nebukadneza*grml* jeah please wait a min again
15:09.41ClubBarfNebukadneza: I'm not goin' anywhere for the time being...  :p
15:09.45Nebukadneza:)
15:09.47SuShI``context = bogon-calls ?
15:09.58NebukadnezaClubBarf: got it - there was a sftp running
15:09.59eKo1Hmm...I was expecting it to jump to priority n + 101.
15:10.01SuShI``should replace it by other thing?
15:10.07eKo1Since it is an error.
15:10.24ClubBarfNebukadneza: Ok, I'll try you again.
15:11.45lesouvagezeek: thanks I spend hours and it kind of drived me crazy because the first time I had it working in just  5 minutes.  It up and running now.
15:11.52MuppetMasterClubBarf:  First time for me today.
15:13.50*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
15:14.05NebukadnezaClubBarf: thanks again :)
15:14.15Nebukadneza(Oo the gnophone ebuild is broken)
15:14.20ClubBarfNebukadneza: no problem.
15:14.22Nebukadnezabut i think iax is a good idea too
15:14.34Nebukadnezaare there good iax hardphones?
15:14.35ClubBarfyeah, methinks so too.
15:14.54ClubBarfI *think* the snom phones might support IAX.
15:15.09ClubBarfThey're linux embedded devices, so it would make sense to me.
15:15.21Nebukadnezahm jeah
15:15.34SuShI``Must I replace context = bogon-calls by other thing or just leave it?
15:15.43ClubBarfMuppetMaster: sorry, I was on the phone to Nebukadneza and missed that comment...
15:15.46NebukadnezaClubBarf: btw: what about iax and nat ... ?
15:15.51Nebukadnezai was having some problems with sip
15:16.12ClubBarfNebukadneza: I couldn't tell you for sure.
15:16.18Nebukadnezaokay ...
15:16.27ClubBarfIs there a NAT gateway between your softphone and your *?
15:16.42Nebukadnezanot really ... but still i was having problems
15:16.56Nebukadnezai got the * running on my server/router (direct connection to the i-net)
15:17.11Nebukadnezaand my sofphone on some machine in the lan behind this router
15:17.12lesouvageClubBarf: I have to Snom200 on my desk.  I can look for answers.
15:17.14ClubBarfAh, so * is running on your firewall?
15:17.24Nebukadnezawhen asterisk was trying to establish a native bridge ...
15:17.25Nebukadnezayes
15:17.37ClubBarflesouvage: Yeah, would be nice to know if Snom supports IAX.
15:18.05ClubBarfI have a dodgy Origo hardware phone, and I think it needs upgrading to a Snom.
15:18.53ClubBarfNebukadneza: Well, I can't see NAT being a problem then.
15:19.47NebukadnezaClubBarf: there is NO nat :/ but asterisk treats it like one
15:19.59Nebukadneza(Oo iaxcomm: opensource without the source beeing available)
15:20.04ClubBarfYou're not technically doing NAT to the SIP command stream - it's going to the *, which is where it terminates.
15:20.27ClubBarfSIP command stream?  I mean the sip port data.
15:20.46ClubBarfYou know what I mean, right?
15:21.10Nebukadnezajep
15:21.49ClubBarfMuppetMaster: I would like to talk to you about Eyebeam at some point.
15:22.23NebukadnezaClubBarf: hm ... which iax codec does the highest compression?
15:22.28ClubBarfMuppetMaster: I want to get eyebeam working properly too, so it might be an idea to talk at some point.
15:22.31MuppetMasterClubBarf:  No problem.  I only just got it today.  The g711/g729a interaction via Asterisk is fine.  But not having much luck with the h.263 video side, either through Asterisk or via FWD.
15:22.41lesouvageClubBarf: In the web interface I see Sip and H323 in the menu. Should IAX in that same menu if it is supported.
15:23.04ClubBarflesouvage: Yeah, it would be.
15:23.53*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
15:23.58ClubBarfI guess it's just using sip or H.323 then.  IAX doesn't give you any real advantage unless you have multiple calls going between 2 * servers anyway.
15:24.02lesouvageCLubBarf: it isn't so I guess it's not supported.
15:24.44ClubBarfMuppetMaster: Does DTMF work on your eyebeam?
15:25.02MuppetMasterClubbarf:  Yes, it does.  Although I have had problems with g729 and X-Pro.
15:25.34ClubBarfx-pro works flawlessly in G729 for me, with the exception of the call transfer gremlins...
15:25.58ClubBarfBut I can't get DTMF to work in RFC (or inband for that matter) in eyebeam.
15:26.22ClubBarfSounds like I should really get the latest CVS snapshot.
15:26.24lesouvageWhat should I change if I want to hear music when I put someone on hold.
15:27.16ClubBarflesouvage: http://users.pandora.be/Asterisk-PBX/MusicOnHold.htm
15:27.26ClubBarflesouvage: That worked for my *
15:28.09ClubBarfI now have some nice Bach, Handel and Mozart on my *
15:29.58ClubBarfMuppetMaster: What kind of problems with G729 in X-Pro?
15:30.19lesouvageClubBarf: I have that running now. But when I press the hold bottun the person I'm calling with hears music while I hear a beep. I also want to hear the  music myself.
15:30.24MuppetMasterClubbarf:  Does not seem to want to pass the DTMF out of band to the Asterisk.
15:30.51ClubBarfOdd - I have the same problem with eyebeam, but not with X-Pro!
15:31.07*** join/#asterisk TrunkD (~trunkd@AOrleans-252-1-44-37.w83-115.abo.wanadoo.fr)
15:31.13*** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:2)
15:31.36ClubBarfWhen I dial my voicemail, and enter my password, for example "1000" - the asterisk console tells me that I dialled either "1" or "01"
15:31.53ClubBarfIs that what your's is doing?
15:33.33MuppetMasterClubbarf:  No, just a moment and I will post the error.
15:33.40ClubBarfOk.
15:35.09MuppetMasterClubbarf:  Just got a call, so may be a bit.
15:35.48ClubBarfwell, I need a loo break, so that's just good timing.
15:37.46NebukadnezaClubBarf: iaxcomm looks really nice ...
15:38.20NebukadnezaClubBarf: you can use different audio devices for ringing, phoning and the speaker
15:39.57ClubBarfNebukadneza: I think you can do that with most softphones.
15:40.01ClubBarfI *think*
15:40.31NebukadnezaClubBarf: linphone, kphone and sjphone (the only sipphones i found for *nix) cant do it :)
15:40.35ariel_you can do something like that with diax.
15:40.52ariel_which I like since it's small and easy to setup.
15:41.21ClubBarfNebukadneza: ah, yeah - you're right.
15:42.02ClubBarfBut you can do it with X-Lite, X-Pro, Eyebeam and most windows softphones.
15:42.11ClubBarfI *think*...
15:42.59ClubBarfIt's so you can have one soundcard for a headset (or a USB phone) where the calls go, and let the ringer go through your main speakers.
15:43.10ClubBarfAt least, that's what I think it's for.
15:43.49ClubBarflesouvage: I think that's more down to your phone than *
15:45.31*** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr)
15:48.25*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
15:48.55cjkhow can i check which transcoding operations are currently going on ?
15:52.26Nebukadnezaahm
15:52.29NebukadnezaWTF is with the iax quality?
15:54.01ClubBarfNebukadneza: I've never actually used iax, but since it's only function is to negotiate a connection for your audio codec (gsm,G711, G729 etc) to talk over rtp, it shouldn't affect call quality.
15:54.12Nebukadnezahm kay
15:54.22GoshenCould someone in the UK help me test my incoming IAX connection by dialing my UK number?
15:54.26obelisqueyou guys can help me with my echo problem?
15:54.31ClubBarfActually, that's not true.
15:54.37ClubBarfIt's not it's only function.
15:54.40*** join/#asterisk zotz (~zotz@24.231.32.191)
15:55.37ClubBarfThe audio/video packets are actually inside the IAX protocol, so when I come to think about it, it *COULD* affect call quality...
15:55.57ClubBarfinside the IAX packets, even.
15:56.08Zeeeklesouvage what asterisk version ?
15:56.17mmlj4um, * does video calls?
15:56.26ClubBarfmmlj4: yeah.
15:56.31mmlj4kewl
15:56.37ClubBarfBut only in passthrough mode.
15:56.43lesouvageZeek: xorcom on vmware
15:56.46mmlj4explain?
15:56.52Zeeekwhich is what asterisk version
15:56.57Zeeek1.0.?
15:57.09Goshenmmlj4: http://www.voip-info.org/tiki-index.php?page=Asterisk+video
15:57.20ClubBarfmmlj4: * just routes the video from point A to point B - it can't understand them.
15:57.24Zeeekbecause music on hold is broken in 1.0.6
15:57.47mmlj4routing seems sufficient, no?
15:57.51lesouvagezeek: 1.0.5
15:57.56*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
15:58.25Zeeeklesouvage the trouble with xorcom is you need to ask them what the problem is since you don't really configure it
15:58.30Nebukadnezahm ... maybe its iaxcomm related
15:58.33obelisque1.0.6
15:58.44mmlj4ah, ok, it can route windows messenger crap
15:58.55Zeeekmoh is slightly broken in 1.0.6
15:59.24ClubBarfmmlj4 - not just windows messanger - any industry standard videoconferancing kit should be compatible.
15:59.27lesouvagezeeek: but there is access to all the .conf files. I can change whatever I want to change (if I was a real expert)
16:00.52ClubBarfmmlj4: H.261 and H.263 video streams, basically.
16:02.04*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
16:02.21*** join/#asterisk file2 (~jcolp@mctnnbsah25-142166093180.nb.aliant.net)
16:07.27ClubBarfOut of interest...  Can you use any linux compatible modem as an FXO?
16:07.27NuggetNo.
16:07.34ClubBarfIt's just that intel one?
16:07.47Nuggetthat one specific intel one with that one specific firmware.
16:08.14ClubBarfAh.
16:08.28Goshenso I have to use spanDSP to get faxing in asterisk?
16:12.11ClubBarfthx Nugget.
16:13.16GoshenThe Asterisk Fax Manager looks interesting...what are people using for faxing with asterisk?
16:13.49shaZwazPRI :)
16:13.49eKo1A fax machine hopefully.
16:14.03Goldeneardoes asterisk have something like a "call by email function". ie I would prefer to dial bill@krosoft.net that use his phone number.
16:14.11shaZwazhehee
16:14.35eKo1Goldenear: Make an AGI or an app for it.
16:15.09GoldeneareKo1: so nothing has been done in this way yet ?
16:15.10shaZwazc'mon u can't expect * to do EVERYTHING ...
16:15.28shaZwaz..that too for free
16:16.06*** join/#asterisk luisgrin (~luis@209.99.227.220)
16:16.21*** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:2)
16:16.58luisgrinhi im new to asterisk i want to make a quiz i got x100p and a livecd
16:17.07tzangerwow I must be in a bad mood
16:17.10luisgrinbut i did not start with it
16:17.41NuggetGoldenear: yes, asterisk does exactly that.
16:17.41tzangerI just blasted some guy on the list for buying a clone card and asking for help, and then some other twitt for telling me that I'm high on myself (separate incident)
16:17.49Nuggetgoogle for "sip uri dialing" and check out e164.org
16:17.57ClubBarfluisgrin: you want to make a quiz?
16:18.08Nebukadnezais anyone here having problems with iaxcomm (such as high noise)?
16:18.19tzangerNebukadneza: we just solved that problem
16:18.19GoldenearNugget: I would like to do that with IAX, not SIP
16:18.28Nuggetok.  so do it with iax, not sip.
16:18.29eKo1I'm contemplating whether I should make an AGI or an app for this project of mine...
16:18.42tzangerNebukadneza: this is on linux and I'm guessing you're using an es137x or intel8x0 type card?
16:20.09*** join/#asterisk freat[laptop] (~freat[lap@node-40242662.mdw.onnet.us.uu.net)
16:20.21ClubBarftzanger: when you say "we just solved that problem" - you one of the dev's?
16:20.33GoldenearNugget: but an IAX uri is something like user@asterisksrv/phonenum ... and I would like something like name@domain.com
16:20.34Nebukadnezatzanger: ensonique solo-1 :/
16:20.34*** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net)
16:20.51tzangerClubBarf: no, but I have been working with stevekstevek on the jitter buffer and I personally solved the problem Nebukadneza's talking about
16:20.56luisgrini only want to know if some body has expereince with it
16:20.58tzangerNebukadneza: what version of iaxcomm are you using
16:21.01tzangerrc2?
16:21.08ClubBarftzanger: so you ARE a dev, just not a core dev.
16:21.10luisgrinin order to make some quiz
16:21.11Nuggetyes, an iax uri is a different format.
16:21.17tzangerClubBarf: not really I just help out
16:21.25Nebukadnezatzanger: current (1.0rc2)
16:21.40tzangerI have done development (I'm the R&D manager at work, after all) but not specifically for asterisk
16:21.45tzangerNebukadneza: grab cvs head
16:21.50Nebukadnezakey
16:21.53Nebukadnezathx
16:21.55tzangeror if you have the source to 1.0rc2 I'll show you where
16:22.05ClubBarfAh, well, whatever, I just thought I would say thx for the time you put into the project anyways.
16:22.07tzangerNebukadneza: I'm not sure if stevekstevek pushed the fix to cvs or not but the fix is easy
16:22.25tzangerClubBarf: well on behalf of the real developers I will say you're welcome.  :-)
16:22.30ClubBarf:p
16:22.43tzangerClubBarf: you're absolutely right though, they've done an amazing amount of work and put in countless hours to this project
16:22.50Nebukadnezatzanger: Oo cvs [login aborted]: cannot get working directory: No such file or directory
16:22.57ClubBarfAmazing is putting it mildly.
16:23.11Nebukadnezaseems like the iaxclient.sf.net people got no cvs?
16:23.13ClubBarfThis stuff would cost hundreds of thousands from Avaya or Panasonic.
16:23.48tzangerClubBarf: perhaps not hundreds of thousands but yes at least 20k
16:23.50tzangerNebukadneza: hmm
16:23.51GoldenearNugget: so calling iax2:user@domail.com won't work :(
16:23.54tzangerI pulled CVS head just fine
16:24.05tzangerGoldenear: IAX2/user@peer works just fine
16:24.09tzangerthat's how you do it
16:24.17ClubBarfI've only been working with it in earnest since yesterday, and it's clear it's awesome stuff.
16:24.28tzangerNebukadneza: :pserver:anonymous@cvs.sourceforge.net:/cvsroot/iaxclient
16:24.31Nebukadnezatzanger: Oo kay my fault
16:24.32tzangerthat's my CVSROOT
16:24.42Nebukadnezatzanger: i tried to login in a already deleted dir :)
16:25.00ClubBarfWell, put a couple of 410's in a quad P4 or Opteron system and you have a box that WOULD cost hundreds of thousands.
16:25.08tzangerand then in iaxclient/lib/audio_portaudio.c
16:25.11Nebukadnezaman am i dumb :)
16:25.24ClubBarfMy sip only * isn't worth quite that, but free is still my favorite price.
16:25.31Nebukadnezatzanger: i just need to co the iaxclient dir right - no libs or anything else?
16:25.36tzangerline 495, the #ifndef MACOSX either say #ifdef WIN32 or #if 0 (your choice)
16:25.43tzangeriaxclient has everything in it
16:25.50Nebukadnezak
16:25.52tzangercvs co iaxclient
16:25.55Nebukadnezaah just see it :)
16:26.03Nebukadnezas/just/already/
16:26.09Nebukadneza*argh* my english engine is fcked up
16:26.14tzangerNebukadneza: your "noise" issue is static/crackling right?
16:27.02Nebukadnezatzanger: i dont know - its fine on the iaxcomm side ... but the other sides reports heavy noise
16:27.11tzangerhmm well try this anyway
16:27.28Nebukadnezaat it
16:27.28tzangerI think this was for an iaxclient side issue but it may help both depending on the card
16:27.37Nebukadnezabut it seems like i need a new wx lib
16:27.57tzangertry testcall first
16:27.58shepherdbut if you can afford a couple of 410, you are already paying thousands a month for phone service
16:27.58shepherd:)
16:27.59tzangeror rather
16:28.03tzangerNebukadneza: without rebuilding
16:28.05Nebukadnezatestcall works
16:28.08tzangertry testcall and see if the problem persists
16:28.11tzangertestcall is clear?
16:28.12Nebukadnezahm kay
16:28.29Nebukadnezatestcall compiles clear ...
16:28.33tzangernot compiles
16:28.38tzangerdoes the rc2 testcall work just fine
16:28.41tzangeror is it noisy too
16:28.51X-Genwith x-lite is it possible to log into 2 sip proxys at the same time ?
16:29.23Nebukadnezatzanger: sec - trying to figure out how to register testcall with asterisk
16:29.48tzangerok
16:29.51ClubBarfJust compiling the latest CVS snapshot - wish me luck...
16:29.54NebukadnezaUsage is XXX < wtf is wrong with those peole?
16:30.02tzangerNebukadneza: yeah it's not cool
16:30.05shepherdhehe
16:30.12tzangertestcall is specifically "raw"
16:30.20Nebukadnezai know ...
16:30.22ClubBarfX-Gen: I don't think it is.
16:30.26shepherdyesterdays was a little unstable
16:30.31Nebukadnezabut ... could you help me registering testcall with astereisk
16:30.33ClubBarfI could be wrong, but I don't think you can.
16:30.35Nebukadnezaasterisk
16:30.42Nebukadnezai just dont see a doc out there
16:30.44*** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar)
16:30.49ClubBarfX-Gen: just get Asterisk to log into them for you.
16:30.59Goldeneartzanger: user@peer won't call "user" :( I was just wondering if there was a way to call somebody with a sip like URI but with IAX2 ...
16:31.26eKo1Goldenear: yes
16:31.34tzangerGoldenear: you can't call a user
16:31.39tzangerGoldenear: users place calls to you
16:31.42ClubBarfCreative use of your dialplan lets you have as many sip proxies/PSTN providers/IAX providers as you feel the need for.
16:31.44tzangerGoldenear: peers you place calls to
16:31.55Goldenearisn't a SRV record on the DNS possible with asterik/IAX2 ?
16:31.59tzangerGoldenear: if you need to call and be called from some URI you need both a user and peer entry for them
16:33.15tzangerNebukadneza: testcall user@host
16:33.20tzangeryou can't specify a number or password
16:33.24tzanger(at least I haven't)
16:33.31Nebukadnezai cant specify a password? Oo
16:33.34tzangerit will call the 's' exten in user's user entry
16:33.40Nebukadnezathats abd
16:33.40tzangerNebukadneza: it's very dumb :-)
16:33.41Nebukadnezaargh
16:33.44tzangerNebukadneza: it's just for testing
16:33.46tzangerfor example
16:33.50Nebukadnezawith my asterisk setup im unable to test then :(
16:33.58tzangertestcall guest@misery.digium.com
16:34.37Nebukadnezathanks
16:34.39Nebukadnezabut ... noise :/
16:34.45ClubBarfX-Gen: try looking for an X-Lite or Xten forum or IRC channel, they may know how to get X-Lite to talk to more than 1 sip gateway.
16:35.17X-GenClubBarf: k thanks
16:35.23ClubBarfX-Gen: np, mate.
16:36.13Goldenearwon't it be possible that a call with the URI iax2:john@domain.net will automatically call IAX2/guest@asterisk_server_for_the_domain/john_phone_number ?
16:36.23tzangerNebukadneza: hmm
16:36.30tzangerNebukadneza: do other softphones work fine on that system?
16:36.42Nebukadnezajep
16:36.46Nebukadnezakphone (with sip)
16:36.52tzangerhmm
16:37.25*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
16:37.26tzangerGoldenear: what exactly are you trying to accomplish?
16:37.30Nebukadnezamaybe gnophone would be a solution ... but it doesnt compile too :(
16:37.54tzangerNebukadneza: you're changing too many variables when you switch from iaxclient to kphone for me to be able to determine the cause
16:38.23GoldenearI'm would like a user can call a other one via IAX2 only by knowing its email address
16:38.32Nebukadnezatzanger: hm jeah :/
16:39.08Goldenearand I would like to know if there something standard for that in asterisk (or somebody working on this)
16:39.23tzangerGoldenear: again, what exactly are you trying ot do
16:40.04ManxPowerGoldenear: I don't know, but I do know that you won't be able to dial e-mail addresses very easily from phones.
16:40.06GoshenGoldenear: ENUM will help if they are registered with ENUM
16:40.07Goldeneartzanger: What's not clear for you ?
16:40.17tzangerGoldenear: whatever it is you're trying to do
16:40.20tzangerwhat are you trying to do
16:40.38ManxPowerGoldenear: What *I* don't understand is why you want users to jump thru hoops to dial someone.
16:40.44filewoot VON
16:41.19tzangerfile: Count Asterisk von Digium
16:42.50GoldenearManxPower, tzanger : I would like to use IAX2 in a Skype like app. So I would like users to be able to call the by they email address (like msn also).
16:43.12Goldenearto be able to call the others*
16:43.24ManxPowerGoldenear: As far as I know you would have to build that feature yourself.
16:43.52ManxPowerI'm someone that thinks that Softphone Suck, so I would never have a need for that feature.
16:44.04*** join/#asterisk coppice (~chatzilla@103.202.17.210.dyn.pacific.net.hk)
16:44.28*** join/#asterisk {^DaNi^} (~dcp@eth1.bina-jojma.d1n.net)
16:44.41tzangerGoldenear: sure - build a lookup db that, given an email address, returns an IAX2 peer
16:45.02GoldenearManxPower: a softphone on a PDA is something very usefull :)
16:45.20tzangerGoldenear: then you can even put it in a macro so that you say emailDial(user@email.dom) and it looks it up and dials the right place
16:45.22*** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18)
16:45.26ManxPowerGoldenear: It still doesn't hold a candle to PSTN service.
16:45.59Goldeneartzanger: good idea
16:46.40tzangerGoldenear: anything's possible, I just needed to understand what it was you were trying to do and a simple cross-ref like that is not hard :-)
16:47.10coppicedoes anyone know what has happened to ipvolution.com?
16:47.38Goldeneartzanger: I understand :)
16:48.04tzangercoppice: not I, sorry
16:50.44Goldeneartzanger: do you think asterisk could directly manage the mail address db ? I mean when a user (with a endpoint supporting this feature) register to asterisk it will be able to give both its IAX phone number and its email address.
16:51.01tzangerGoldenear: if you code the app to do it, sure
16:51.23tzangerwhen you add the user to your iax.conf you can also add their email to the db
16:51.36GoldenearI'm not a vey skilled coder but I can try to do it :)
16:52.39MuppetMasterI applied the version three of this patch manually:  http://bugs.digium.com/bug_view_page.php?bug_id=0003709.  And yet I get this error:  Mar  5 17:51:38 WARNING[29226]: channel.c:284 ast_best_codec: Don't know any of 0x80000 formats.
16:52.48Goldeneartzanger: so, if asterisk already manage the user's email in iax.conf a part of the work is already done :)
16:53.00MuppetMasterHas anyone gotten H.263+ working with Asterisk (ie - Eyebeam Video)?
16:53.16tzangerGoldenear: it doesn't and there's no reason a PBX should care to, which is why I'm saying you'll have to code it up yourself
16:53.17*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
16:53.20*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
16:53.36ManxPowerMuppetMaster: Sounds like you should attach your report to the patch/bug entry.
16:54.00MuppetMasterWill do.
16:54.28PTG123whats the bug
16:55.26*** join/#asterisk pointer-gaim (~pointer@router.cathey.us)
16:55.32Goldeneartzanger: ok. why a PBX should care to ? because asterisk is IP/computer based and because an email is often easier to remember that a phone number :)
16:56.02Goldeneardon't you think this can be useful (especialy for a soft phone) ?
16:56.17tzangerGoldenear: for your particular application
16:56.22tzangeryour soft phone should have a directory
16:56.26tzangerwhere the IAX2 info is kept
16:56.29ManxPowerI can't dial an e-mail address from my cell phone.
16:56.29tzangerthat is the PROPER solution
16:56.37tzangerusers don't care to know the email address even for email
16:56.47PTG123since numbers are so cheap
16:56.51PTG123its smarter to use them
16:56.52tzangerthey want to click on "Mom" and have it translate to sexygranny694u@hotmail.com
16:57.06PTG123then it works from an ip or a normal phone
16:57.11tzangerit has nothing to do with numbers or email addresses, the PROPER solution is to have a proper phone directory
16:57.30PTG123i wrote my own patch to asterisk, that keeps directory info on all users
16:57.34MuppetMasterManxPower:  Added.  Always like to check here before I post something on Matis.
17:00.27GoldenearManxPower: my cell phone can write email ... and I'm sure yours can too ;) but I don't want to replace calling by numbers... I want both to have the choise :)
17:00.48*** part/#asterisk X-Gen (~x-gen@rrba-146-120-223.telkomadsl.co.za)
17:01.21MocI had this weird problem this morning, my * wouldnt handle anycall, if I did iax2 show channels, it show a few calls active (that arnt).  I was in verbose 0, as soon I set verbose 999, voila, all the 'open' call got closed, and new call got throught ...
17:02.06PTG123moc: i had some weird problems with licenses hanging.. think they are used up
17:03.23ManxPowerI accidently installed CVS-HEAD on a produciton server last night.  The system locked up when loading wcfxs/wctdm.
17:04.01PTG123heh i found a good version of the cvs 3 weeks ago
17:04.05PTG123and only use that now :)
17:04.17*** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net)
17:04.28pimpwellwas wondering the best place to get an 800# from
17:04.40pimpwellVerizon?
17:04.49PTG123pimpwell: you want it terminated over iax?
17:05.18pimpwellnot yet, this will be connecting to my asterisk box but more like a PBX if anything
17:05.27pimpwellerr voice mail system
17:05.29pimpwellsorry
17:05.47PTG123what do you mean
17:05.51pimpwellI wont be calling out with it
17:06.12PTG123pimpwell: but will it be going into your box?
17:06.17PTG123and you handle voicemail etc
17:06.22PTG123or you just want it to handle everything
17:06.22pimpwellya it will.
17:06.31jjhallMorning everybody.  I had distinctive ringing working with my SPA1000 using the setvar function.  I just had a call come in that should have used a different ringer (ALERT_INFO=Bellcore-r3) but it did not work.  I did a quick search in Mantis and on the mail list, but did not see anything relevant.  Any ideas?
17:06.33PTG123who currently provides your normal phone service?
17:06.36pimpwellVerizon
17:07.10pimpwellI'm located in New York
17:08.01*** join/#asterisk Blackvel (~blackvel@dsl-082-083-173-045.arcor-ip.net)
17:08.02PTG123verizon just gives you a normal phone line, or terminated via sip?
17:08.12pimpwellmost likley normal
17:08.33pimpwellIll chekc now, but Verizon is always expensive from what I've seen
17:08.55PTG123well what i am wondering is why don't you just use a VOIP provider that can provide both?
17:10.15pimpwellya the cheapest solution is best for me
17:10.25pimpwellor something I'll consider
17:10.31cjkhow can i check which transcoding operations are currently going on ?
17:10.36pimpwellit's a small start up business
17:10.55PTG123that would def. be cheapest
17:11.19PTG123how many minutes you think you would use, and how many channels, etc?
17:11.21PTG123for this business
17:11.21pimpwellknow any providers off hand?
17:11.25pimpwellnot much
17:11.27pimpwellat all
17:12.04PTG123I'd help you out, but i am not completely ready to start accepting customers.. but go with teliax.com
17:12.05pimpwellin minutes, Id say 60 minutes a day THE MAX
17:12.08PTG123they can do new york
17:12.09PTG123are cheap
17:12.14PTG123they have a pay as you go plan
17:12.17PTG123works well
17:12.26PTG123or a small business unlimited if you like
17:12.29PTG123they can do 800 #s, etc
17:13.03dan2can someone update the topic for me
17:13.28*** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net)
17:14.42dan2BroadVoice outbound configuration has changed with in the last two hours, http://tinyurl.com/5xwzk
17:14.56dan2the url explains what needs to be done to make outbound calls working again
17:15.38*** join/#asterisk mesi (~player@dsl-082-083-054-206.arcor-ip.net)
17:15.52*** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net)
17:15.55*** join/#asterisk Duy (~duy@port-83-236-189-65.static.qsc.de)
17:15.55shepherdj #asstricks
17:15.56shepherd;askldjfa;skdfja
17:15.59jjhalldan2: Did they send that via e-mail or is it on their page?
17:16.20dan2jjhall: I'm working on getting the page updated as fast as I can now, the email is from me, I work for them
17:17.07jjhallOK.  Since you work for them, and obviously care about Asterisk, how about IAX and/or other codecs besides 711u?  ;-)
17:17.58*** join/#asterisk Shido (~greg@d57-87-253.home.cgocable.net)
17:18.12Blackvelanyone using java + asterisk? I need some advices. Any J2EE expert and asterisk in here? Help would be much appreciated
17:18.27jjhalldan2: Do we still need fromuser=<phone number.?
17:18.48Blackvel<PROTECTED>
17:19.01Blackvelbristuff-0.2.0RC7j works great while RC7f has audio problems
17:19.55dan2jjhall: yes
17:20.10dan2jjhall: we have g726 and g729 as well
17:22.15shepherdgsm
17:22.20shepherdhehe
17:22.26PTG123g729 is the shiznit
17:22.35jjhalldan2: How about GSM or iLBC?
17:22.48shepherdasterisk supports gsm
17:23.02shepherdprobably ilbc too
17:23.03jjhallYes, Asterisk does GSM very well in fact.
17:23.11jjhallThat as well.  I use both of them
17:23.12*** join/#asterisk djin (~djin@gridfox.xs4all.nl)
17:23.24shepherdyeah
17:23.29jjhallI want Broadvoice to use it as well.
17:23.31shepherdno reason to us g729
17:23.34shepherduse
17:23.35dan2jjhall: my boss likes iLBC, we're working on it
17:23.43jjhallPerfect!
17:23.50PatrickDKilbc is good for dropped packets
17:23.56PatrickDKbut it requires ALOT of cpu power
17:24.09*** join/#asterisk meshugga (philip@loeblich.linuxteam.at)
17:24.11meshuggahi
17:24.12shepherdhmmm
17:24.25shepherdhow many lines you working with?
17:24.26jjhallNormally I don't care, but my crappy cable modem gets rate capped periodically, and I would like a lower-bandwidth solution I can use instead of the full uLAW 80+ K
17:24.44PTG123no reason besise g729 is the best codec :)
17:24.48meshuggaanyone knows why LOG_DEBUG in a module doesnt produce any debug output, even if defined debug => debug in logger.conf?
17:24.58PatrickDKI use gsm, works well over cablemodem
17:25.01PTG123quality of ulaw, with the lease bandwidth usage of them all
17:25.18PatrickDKg729 is no way near ulaw quality
17:25.21shepherdbut it's g729!
17:25.26PTG123yah it is
17:25.32shepherdtherefore it sucks
17:25.34shepherd:)
17:25.47jjhallThe way I look at it, GSM is good enough for cell phones, and most people won't know the difference.
17:26.11eKo1Cell phone calls suck anyways.
17:26.21shepherddepends on who you use
17:27.06*** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com)
17:27.54meshugga<PROTECTED>
17:28.03meshuggano clue why i wont get any debug output?
17:28.27eKo1What debug output?
17:28.45meshuggathe module i am trying to debug does "LOG_DEBUG
17:28.54eKo1What module?
17:28.59meshuggaand i configured logger.conf to output debug information
17:29.03meshuggachan_bluetooth
17:29.17eKo1pfft...good luck with that.
17:29.20meshuggabut anyway, it is not about the module
17:29.22shepherdhaha.. yeah
17:29.30meshuggait is about that there is *nothing* in /var/log/asterisk/debug
17:29.30shepherdhow many v's did you use?
17:29.48meshuggashepherd: how many am i supposed to use?
17:30.13shepherdheh.. the more v's the more verbose :)
17:30.20shepherdbut i usually don't log anything
17:30.28dan2anybody around?
17:30.28HjemmeRoyK~seen zoa
17:30.32jbotzoa <~zoa@ip-212-239-162-26.dsl.scarlet.be> was last seen on IRC in channel #asterisk, 19h 55m 2s ago, saying: 'brian!!!!'.
17:30.32dan2ops even
17:30.35dan2I need the topic updated
17:30.42meshuggai can do as many as i want, doesnt change anything
17:30.58meshuggamainly because of the fact that "verbose" output is done differently to debug
17:31.19shepherdhmmmm... ok
17:31.20meshuggathat is, ast_verbose(VERBOSE_PREFIX_1 ...
17:31.28meshugganot LOG_DEBUG
17:31.48meshuggaany source hacker here?
17:32.09HjemmeRoyKwho is zoa?
17:32.18meshuggai am really willingly to debug this stuff, but i cant due to the lack of information of how to get debug outpout :(
17:33.13*** join/#asterisk TheEmperor (TheEmperor@218.111.48.52)
17:33.24eKo1Well, I suggest you start looking at the source code.
17:33.30eKo1Use the source Luke.
17:33.32meshuggai AM looking at the sourcecode
17:34.05meshuggabut i will not trace down every other helper function
17:34.37shepherdwhere is chan_bluetooth?
17:35.51shepherdoh.. 3rd party
17:36.08shepherdask them: http://www.crazygreek.co.uk/content/chan_bluetooth
17:36.39meshuggaquit...
17:36.50meshugga-quit
17:36.56meshuggai know where i got that module
17:37.15techiequit
17:37.28shepherdwell your best bet is to ask the developer
17:38.39shaZwazanyone know of a quality IAX soft phone ?
17:38.56shaZwazto be used for Call Center setups ?
17:41.07ctooley1110044515                      enhanced                [0.0]
17:41.07ctooley1110044515                      dnid            [unknown]
17:41.07ctooley1110044515      gnumber called: 2146588351
17:41.07ctooley1110044515              GCS::DB::search: calling [SELECT * FROM person WHERE gnumber = ?]:[5127912046]
17:41.19ctooleysorry about that
17:41.21ctooleywrong window
17:42.22PTG123shaz: i was just discussing that with goldenear
17:43.04shepherdshazwaz: there are a couple in development
17:43.11shaZwazthat am interested in using it for a cc setup
17:43.13shepherdyou can look on: http://www.voip-info.org/tiki-index.php?page=Asterisk+phones
17:43.21shepherdthere might be some on there
17:43.25PTG123i wrote one, and i actually use it for that
17:43.26shepherdmost are sip though
17:43.37PTG123was just saying i am too lazy to put it on a webpage, and maintain the project :)
17:43.50shaZwazAheeva is providing one with its CC solution
17:43.51shepherdptg: you could always donate it :)
17:44.12PTG123heh if someone wants to maintain the project, i'll gladly release it under bsd license :)
17:44.22PTG123free for use by anyone
17:44.36shepherdsf!
17:44.38shepherd:D
17:44.50shaZwazI have tried Fire Fly , it works ..say not bad
17:45.03PTG123i don't have the time to do that :)
17:45.31hardwireanybody ah HTB master?
17:46.03PTG123so any volunteers :)
17:47.46ClubBarfAnyone got the latest CVS build?
17:48.20shaZwazI heard Zeesk saying that MOH broken in 1.0.6
17:48.37shaZwazanyone verified that ?
17:48.41ClubBarfI'm seeing lower quality audio with the lastest build, is that my end or something they're aware of?
17:49.13shepherdthey just raised the # of threads
17:49.16shepherdso it might
17:49.20eKo1Is there an option for disallowing a sip peer to call itself?
17:49.33MikeJ[Jayden]I am told that MOH is broken in 1.0.6, but works in CVS Stable.. don't use stable so I am not sure
17:49.41ClubBarfOh, and MOH seems to work in the latest build.
17:49.47*** join/#asterisk nextime (~nextime@danex.i-m-c.it)
17:50.37ClubBarfI'm listening to Mozart through my VoIP hardware phone right now.
17:50.47jjhallI just got MOH working last night using the moh-native function
17:50.48ClubBarf(my softphone has me on hold)
17:51.11*** join/#asterisk convey (~chatzilla@208-216-127-234.cust.gti.net)
17:51.12jjhallIt wasn't working (weird electronic sounding distortion) when using the standard method.
17:51.13ClubBarfYeah, seems to work fine in this build.
17:51.19eKo1So, when is moh going to stop using mpg123
17:51.28eKo1?
17:51.36ClubBarfWhy would it stop using mpg123?
17:51.42MikeJ[Jayden]you can use natve moh now w/o mph123
17:51.45jjhalleKo1: My understanding is moh-native does not use mgp123
17:51.45eKo1Because mpg123 sucks ass.
17:51.52jjhallmpg123 even
17:51.54QwellClubBarf: because mpg123 only works with like 1 version
17:52.01eKo1And the project is discontinued.
17:52.08ClubBarfWorks fine for me.
17:52.22MikeJ[Jayden]like I said, you don't need mpg123 now
17:52.22Qwellmpg123 is discontinued?
17:52.24ClubBarfBut then, moh isn't a terrably taxing thing...
17:52.35eKo1Qwell: Yeah, check their website.
17:52.41jjhallAnyone here know if there are any issues with Asterisk not passing setvar variables through when initiating a call to an ATA?
17:52.42Qwellhmm
17:52.45MikeJ[Jayden]mpg321 is still alive
17:52.46Qwellhow long has that been so?
17:53.10Qwelloh
17:53.19Qwellits just "not currently maintained"
17:53.39ClubBarfWell, if my * isn't calling mpg123, is it playing native?
17:53.48eKo1That's the sugar-coated way of saying 'we dropping it'.
17:53.55conveydoes anyone have a clean copy of adsi.conf? Mine is corrupt an looks nothing like my example in /usr/src/asterisk/configs...
17:53.57QwelleKo1: true, true
17:54.21shaZwazClubBarf, if u have loaded formatmp3.so
17:54.40eKo1I think format_mp3.so is the way of the future.
17:55.10MikeJ[Jayden]maybe... there is some other stuff hopefully in the works for MOH as well
17:55.15eKo1Maybe I'll bring up that issue on the next dev. conf..
17:55.34eKo1Correction, I will bring up that issue.
17:55.40ClubBarfconvey - try backing up your /etc/asterisk then running make samples
17:56.35conveywill do.
17:56.53conveyClubBarf: thanks
17:57.06ClubBarfnp.
17:57.44ClubBarfshaZwaz: thanx.
17:58.01*** join/#asterisk memic (~memic@dsl-084-056-108-191.arcor-ip.net)
17:58.52*** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net)
17:59.11ClubBarfSo, anyone know if the low audio quality is up my end, or something wrong with the current CVS?
18:02.56GoshenThe Asterisk Fax Manager looks interesting...what are people using for faxing with asterisk?
18:03.12PTG123goshen: got a url?
18:03.26Goshenhttp://tafm.sourceforge.net/index.htm
18:03.45Goshen~tafm
18:04.09PTG123whats the point of using that?
18:04.12Goshenjbot: TAFM is http://tafm.sourceforge.net/
18:04.13jbotGoshen: okay
18:04.22Goshensend and recive faxes via asterisk
18:04.37Goshenno wasted paper when people send you faxes...
18:04.37conveyClubBarf: worked like a charm.
18:05.00Goshenyou can file the fax images on your computer, no need for a crappy file cabinet
18:05.28PTG123i just want a good client for sending faxes
18:05.34MikeJ[Jayden]anyone using head and using IAX w/ trunking?
18:05.45*** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com)
18:06.48MikeJ[Jayden]anyone.. cmon, not everyone at once
18:07.03GoshenPTG: over asterisk, or using a pc faxmodem?
18:07.40PTG123over asterisk, basically want a print driver, that somehow gets the tif to asterisk and asterisk sends fax
18:07.53PTG123like the hylafax clients that exist
18:08.05Goshenthat would be nice
18:08.15PTG123yes it would :)
18:08.28PTG123i'll put up a bounty :) if i wasn't so lazy to go figure out how to do that
18:08.36shepherdfax uses tif ?
18:08.40eKo1Faxing is so antiquated...
18:08.42PTG123yes
18:08.53PTG123we do alot of faxing every day, client would save us time
18:09.18shepherdi wonder if there is a html2tif
18:09.51MikeJ[Jayden]there is all the ghostscript stuff... it can do the conversion piece if you need it
18:09.57shepherdhttp://www.convertzone.com/net/cz-art%20print%20driver-html-tif.htm
18:09.59shepherdhehe
18:10.21shepherddang it's windows
18:10.21*** join/#asterisk clive- (~pirch@rrba-146-113-179.telkomadsl.co.za)
18:10.42eKo1Dang it, this stupid spanish keyboard doesn't have that wigly char..
18:11.07Silik0nthe tilde?
18:11.12Goshenwigly chars are so antiquated ;)
18:11.24*** join/#asterisk zapa (~zapa@201.135.137.236)
18:11.34Blackvelanyone used jasterisk or jast agi/jastserver before?
18:11.37PTG123well there is no N is spanish is there
18:11.38eKo1No, the one that looks the the approximation symbol.
18:11.39PTG123so maybe n just types it
18:11.52eKo1ñ
18:12.05eKo1^the wigly symbol above that
18:12.17GoshenWhat is the difference between https://www.e164.info/ and e164.org ?
18:12.46eKo1One is secure, the other isn't.
18:13.07Goshene164.info is secure?
18:13.10shepherdyou can use the alt+keypad :)
18:13.18hermieeKo1: guess it's time to open the charmap for a tilde
18:13.21eKo1shepherd: I'm not on Winbloze.
18:13.21Goshene164.org calls your number to ensure it is you
18:13.32eKo1Goshen: Well, it's https...
18:13.46Goshenlol
18:14.02Goshendifference between e164.info service and e164.org service
18:14.26eKo1Don't know. I don't use ENUM yet.
18:14.44eKo1Since most of my traffic is voip-->pstn.
18:15.46jjhallI've thought about setting up ENUM, but haven't gotten the energy to do it yet
18:16.07eKo1Once everything is voip<-->voip, then I'll consider it.
18:16.19*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
18:16.22eKo1But that change is at least 2 decades away.
18:16.50PatrickDKthe point of e164 is to let ya do voip -> voip for people that have it
18:16.53Goshennot really...because places like vonage and other large voip providers register their numbers, then when you try to call someone on their serivce...
18:16.55PatrickDKinstead of using pstn
18:17.01Goshenyou calls gets routed to them over VOIP
18:17.01jjhallOnce everything is voip <-> voip it won't be as necessary.  The best use right now is to send the call via IP if possible, PSTN if not.
18:17.22GoshenI read they had 11 million people already
18:17.44Blackvelwho has some idea to check, if DISA get's passed extensions?
18:18.06jjhallMy question is, how many major providers use it?  Vonage? Broadvoice?  If they aren't using it, then it doesn't do much good except for techies like us that register our own numbers.
18:18.28Blackvelis only the ZAP code reponsible for incoming extensions? Is there any way to debug incoming extensions? (-vvvv doesn't list anything)
18:18.33jjhalldan2: If you are still around, mind if I /msg you?
18:22.08*** join/#asterisk ionix (~ionix@MTL-ppp-149804.qc.sympatico.ca)
18:23.08*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
18:26.06eKo1Man, I'm bored. I think I'll start figuring out how to program modules.
18:27.39*** join/#asterisk viLeR (1000@ip-33-7.telesat.com.co)
18:28.42*** join/#asterisk Tili (~Tili@202-133-65-96-dialup.sat.net.pk)
18:29.20Goldeneardoes anybody know if Linux supports USB phones ?
18:30.31Hmm-workbeing all a usb phone is, is a external speaker and mic i would imagine so
18:31.24Goldenearok, so it should work with iaxComm
18:32.29Blake0PSwhat is the process to install * from CVS? I keep messing it up.
18:32.32QwellThats a different question all together.
18:32.49QwellBlake0PS: cvs up && make && make install
18:32.54Qwellbasically the same as installing stable
18:33.16QwellOf course, if you already have it installed, you'll need to remove the modules dir
18:33.33Blake0PSQwell : I had it installed, then deleted /etc/asterisk/
18:33.38Qwelleww
18:33.39Qwellwhy?
18:33.51Blake0PSI had no idea how to upgrade my copy
18:34.03Qwell/var/lib/asterisk/modules
18:34.04Qwellor some such
18:34.11QwellThat needs to be removed.  That should be it
18:34.26Qwell/etc/asterisk/ is where all your configs reside.
18:34.33Blake0PSi backed those up
18:34.37Qwellgood job
18:34.49Blake0PSwhat is the correct way to upgrade a CVS copy
18:34.50Qwellwent from a catastrophe to a minor annoyance
18:34.54Qwellsee above
18:35.02Qwelljust remove the modules, make, make install
18:35.14QwellYou'll want to upgrade everything else though too, like zaptel
18:35.23Blake0PSI don't use zaptel or the other thing, just asterisk
18:35.29QwellThen you're fine
18:35.44Blake0PSthanks :)
18:38.56ManxPower~docs
18:38.57jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
18:41.37Blackvelanyone knows stuff about bristuff?
18:41.46Blackvelusecallingpres=yes
18:41.49Blackvelwhat is that?
18:42.01Blackvelimmediate=no
18:42.01Blackveloverlapdial=no
18:42.17Blackveldoes that still exist in bristuff 0.2.0-RC7j?
18:42.32*** join/#asterisk Duy (~duy@port-83-236-189-65.static.qsc.de)
18:46.03Blackvelchanges to zapata.conf happen in realtime?
18:46.03Goshenblackvel: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CallingPres
18:46.17GoshenBlackvel: after reload I beleive
18:46.55*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
18:47.35BlackvelGoshen: do you know DISA and asterisk internals very well?
18:47.49ManxPowerBlackvel: What is your problem with DISA>
18:47.52ManxPower?
18:48.02Blackvelcalling from my analog pbx
18:48.10Blackveli can't pass anymore extensions directly :(
18:48.20Blackvelonly when I have the DISA dailtone
18:48.27Blackvelhow can I track this down?
18:48.30ManxPowerset immediate=no
18:48.53Blackvelhad set
18:48.54ManxPowerDo your question is NOT about DISA, it's about why you don't get a dialtone when NO using DISA.
18:48.55Blackveljust upgraded
18:48.58Blackvelnothing works anymore
18:49.12Blackvelhm
18:49.15Blackvelbut I do get a dailtone
18:49.23dan2jjhall: go for it
18:49.25ManxPowerYou get a dialtone when not using DISA?
18:49.29Blackvelbut disa doesn't dail the extension anymore :(
18:49.36ManxPowerWhy are you using DISA?
18:49.48*** join/#asterisk xeaded (Saas@dhcp024-209-097-180.woh.rr.com)
18:49.59Blackvelbecause my analog pbx which does not support blockmode
18:50.04Blackvelit sends number by number
18:50.06Goshenblackvel: no, I just learned about DISA last night
18:50.08BlackvelI have to use it
18:50.15BlackvelI mean
18:50.17ManxPowerYes.  That is not a problem.  Asterisk works just fine in that mode.
18:50.24Blackveli upgraded to asterisk 1.06/0.5 and now I have problems
18:50.26ManxPowerAssuming you have immediate=no
18:50.48ManxPowerWell, we have at least 12 ports like that and don't use DISA.
18:50.59Blackveli am not sure why everyone tries to shoot me into my knees all the times :)
18:51.10Blackvelwell
18:51.16ManxPowerBlackvel: Because you are trying to do something the wrong way.
18:51.26Blackvel6 month ago i was talking to someone who told me my pbx is so silly and I have to use dSIA
18:51.30Blackveli can remember it was you even
18:51.34mesiWho will join me in the FWD coffee house? FWD: 514
18:51.43ManxPowerBlackvel: Perhaps I was being lazy.
18:51.49Blackvelbut it would have to support blockmode?
18:51.51ManxPowerHonestly, DISA is almost never needed.
18:51.52Blackvelanyways
18:51.57Blackvelsome code has changed
18:51.58mesiBlackvel: Same for me here. I use disa, too.
18:52.03ManxPowerBlackvel: "block mode" does not apply to analog.
18:52.06Blackveleither zaphfc code
18:52.10Blackvelor asterisk
18:52.25xeadeddoes anyone know why i get the error `CURLOPT_WRITEDATA' undeclared when i try to compile asterisk?
18:52.27ManxPowerBlackvel: If you are using ISDN then you are NOT using analog.
18:52.30Blackvel"something" doesn't send the extensions from my pbx analog phone into asterisk :(
18:52.38BlackvelI use both
18:52.48BlackvelISDN pbx with zaphfc and hfc card inside *
18:53.00Blackvelbut analog telephone to that 8 analog port of this pbx
18:53.04Blackveland the pbx sends number by number
18:53.07Blackvelnot the total number
18:53.11Blackvelanyways
18:53.16ManxPowerBlackvel: Well with ISDN I can see using DISA just because I have no idea how to debug an ISDN BRI connection
18:53.25BlackvelI used DISA + direct number calling before with asterisk 1.0.2
18:53.28mesiBlackvel : I use such an 8 phone pbx too ;-)
18:53.33Blackvelbut now DISA doesn't accept the whole number
18:53.37ManxPowerBut for ANALOG ports, you don't need DISA.
18:53.41BlackvelI am really p****
18:53.53Blackvelit sends number by number
18:53.56mesiOk, but my pbx sends a total number.
18:54.08Blackveldigit by digit
18:54.14*** join/#asterisk riksta (~rick@81-178-195-88.dsl.pipex.com)
18:54.16*** join/#asterisk invi_ (~invi_@dsl-crow-209-5-162-157-cgy.nucleus.com)
18:54.26ManxPowerBlackvel: Yes, for ISDN that could be a problem and DISA would make sense.
18:54.31Blackvelno
18:54.34BlackvelISDN works great
18:54.34ManxPowerFor analog, no DISA does not make sense.
18:54.41Blackvelit sends the total number as extension into asterisk WITHOUT dISA
18:54.59Blackvelbut i use disa for analog
18:55.01ManxPowerBlackvel: ALL analog ports send digit by digit.
18:55.06invi_anybody: can i specify codec 2 b used 4 dundi?
18:55.13Blackveloh my english
18:55.25Blackvelhow can I say it
18:55.32Blackvelit was working before
18:55.36ManxPowerinvi_: Not a lot of people are going to help you if you talk like a 12 year old.
18:55.36Blackvelsomewhere there are code changes
18:55.39Blackvelnow it does not work anymore
18:55.45Blackvelwhat shall I do?
18:55.51ManxPowerBlackvel: So revert to a known working version of Asterisk.
18:55.55ManxPowerThen report it as a bug.
18:56.05BlackvelI do not know if its a bug in zaphfc or asterisk
18:56.19Blackveland for the older version I have some sipura problem
18:56.26tzangerManxPower: I went a little too far in that X100P clone post on -users.  :-)
18:56.30Blackvelthey can'T call me (I dont want to go back :( )
18:56.50ManxPowertzanger: Maybe so, but we all get sick of specific questions.
18:57.08tzangerat any rate, I apologized for saying I hope nobody helps him
18:57.22ManxPowerBlackvel: Plug a damn analog phone into the Port on Asterisk that goes to the PBX and start doing standard "I'm having trouble dialing" troubleshooting.
18:57.36tzangerI am, however, a little bemused that people like John and Rich would rather blast me than give the guy his answer, since they both know how to fix it
18:58.09ManxPowertzanger: I care a lot less about X100P clones since Digium no longer sells the X100P.
18:58.25tzangertrue enough
18:58.32Blackvelhm
18:58.37Blackvelno
18:58.39Blackveli wont
18:58.48Blackvelthat zaphfc is an isdn NT card
18:59.06Blackveland I need DISA
18:59.43ManxPowerBlackvel: I'm confused then.  I thought you said that you don't use DISA with the ISDN ports.
18:59.45Blackveland I need the d*** debug code to check what stupid part of whatever is not accepting the whole number and sends it in as extension (that was working 1 year)
18:59.49Blackvelright
18:59.49ManxPowerI thought you said they are working fine.
18:59.51Blackvelonly with analog
18:59.57ManxPowerI thought you said that the problem is with the analog ports.
19:00.00Blackvelbut not analog port
19:00.09Blackveli have no analog card
19:00.10ManxPowerTHEN PLUG A PHONE INTO THE ANALOG PORT AND TROUBLESHOOT THAT.
19:00.24Blackvelin waht analog port?
19:00.25ManxPowerIf you don't have an anlog port then I guess you can't have analog cards then.
19:00.47ManxPower<Blackvel> it sends the total number as extension into asterisk WITHOUT dISA
19:00.47ManxPower<Blackvel> but i use disa for analog
19:00.53Blackvelonly the pbx has 8 analog ports + analog telephone (and with it it doesnt work)
19:00.55ManxPowerNow you just said you have analog ports.
19:01.19Blackvelanalog ports in the pbx
19:01.24Blackvelnot inside asterisk with analog card
19:01.30ManxPowerBlackvel: I don't really care what ports your PBX has, only the connection from Asterisk to the PBX.
19:01.31BlackvelI use zaphfc
19:01.35ManxPowerNow WHAT KIND OF CONNECTION is that?
19:01.36Blackvelisdn
19:01.39Blackvelzaphfc
19:01.41ManxPowerThen I can't help you.
19:01.49BlackvelI am sure you can
19:01.50Blackvel:)
19:02.02Blackvelsomewhere there is some code
19:02.06Blackvelwhich accepts the numbers
19:02.11Blackveland sends them into asterisk as extension
19:02.15Blackveland that code was working before
19:02.25Blackvelbut upgrade to latest zaphfc and asterisk 1.0.5/1.0.6
19:02.26Shidolallaal
19:02.31ManxPowerBlackvel: I have never used an ISDN BRI for any kind of voice communictations in my entire life.
19:02.31Blackvelgot the code not working anymore
19:02.33Blackveldunno
19:02.47Blackvelmy hope was
19:02.52Blackvelsince you know the internals very well
19:02.55eKo1Blackvel: Well, maybe you should downgrade to the working version.
19:03.00Blackvelthat you have some tip
19:03.06Blackvelwhere I can debug the extensions
19:03.10BlackveleKo1: no way
19:03.12ManxPowerBlackvel: I don't know the internals very well.  I know configuration for specific parts of Asterisk very well.
19:03.18Blackvelold code has sipura problem
19:03.22Blackvelnoone can call me anymore
19:03.30Blackvelokies
19:03.32Blackveli am lost then
19:03.35eKo1Sipura problems?
19:03.44Blackvelsipura with g729 couldnt call me
19:03.46ManxPowerBlackvel: downgrade Asterisk, upgrade the SIPura.
19:03.48Blackveldunno why
19:03.55ManxPowerThen report it as a bug
19:03.55eKo1Upgrade the Sipura then.
19:03.59Blackveli dont upgrade/downgrade anymore
19:04.03Blackvelwhy the sipura
19:04.08Blackvelnow it works?
19:04.11eKo1What Sipura is it?
19:04.11Blackvelits asterisk problem then
19:04.15Blackvelsipura 2000
19:04.27ManxPowerBlackvel: You just said that the reason you needed to upgrade was because the SIPura has problems.
19:04.33Blackvelno
19:04.44Shidohere we go
19:04.46Blackveli tried to say (but maybe said something different)
19:04.48ManxPowerBlackvel: Why do you need to upgrade to 1.0.5/1.9.6?
19:04.49eKo1Didn't the old firmware have problems with simultaneous g.729 calls.
19:04.54Shidodo you have a friend or a user and a peer in sip.conf Blackvel ?
19:05.02Blackvelthat a sipura couldnt call me
19:05.03Blackvel;)
19:05.12Shidowhats in your sip.conf for your sipura
19:05.15Blackvelfri3end
19:05.19Blackveli have no sipura
19:05.25Shidobreak it out to a user and peer
19:05.26ManxPowerShido: Your fetish for user/peer is distrubing at times.
19:05.27Blackveli got SIP called over intrent
19:05.31ShidoManxPower I know
19:05.36Shidoi like my peers and users
19:05.42ShidoI spank my friends
19:05.46Blackvelversion is fine now
19:05.49eKo1How dirty...
19:05.51Blackvelzaphfc RC7j
19:05.51ManxPowerShido: I like peers/users for GATEWAYS, but friend for phones.
19:05.54Blackvelno voice problems
19:05.59Blackvelbut that number calling problem
19:06.07BlackvelI am really ONLY looking for debug ways
19:06.09Blackvelnothing more
19:06.18Blackvelmaybe its asterisk or it is zaphfc
19:06.25Blackvelor libri
19:06.27tzangerManxPower: a very very good distinction
19:06.27Blackvellibpri
19:06.28Blackvelor whatever
19:06.33tzangerI just use peer/user for everything, personally
19:06.49*** part/#asterisk dan2 (~beta3@dan2.active.supporter.pdpc)
19:07.10eKo1Blackvel: good luck then...
19:07.32Blackveli know
19:07.36Blackvelmission impossible project
19:07.51BlackvelI will shoot the person who has changed that part
19:07.52Blackvel:)
19:08.03Blackvelit was running since 1 year
19:08.10Blackveleven with different versions of all drivers
19:08.11eKo1If your a l337 h4x0r, it should bew no problem.
19:08.11Blackvel;)
19:08.16ManxPowerWell, YOU are the one that upgraded.
19:08.21Blackvelhehe
19:08.24Blackvelof course
19:08.27Blackvelthat is true
19:08.30ManxPowerGranted, 1.0.x should never break anything.
19:08.39Blackvelmaybe its zaphfc
19:08.41mesiBlackvel: your digit by digit thingy sounds like the perfect match for the callthrough macro found on voip-info.com websites :-)
19:08.46Blackvelbut i need to find that out first
19:09.15Blackvelit does?
19:09.17*** join/#asterisk tih (tih@athene.hamartun.priv.no)
19:09.44BlackvelI always used #91309110000
19:09.48Blackveland then pick up the phone
19:09.54Blackvelso I could call it directly
19:09.56Blackvelwas great
19:10.00invi_anybody: can i specify codec 2 b used 4 dundi?
19:10.37Blackvelhm
19:10.43Blackvelwhere do you expect the failure?
19:10.48mesiBlackvel: What you thinking about?
19:10.53Blackvelfor sure it is the part before disa
19:11.01Blackvelwhich listens for DTMF
19:11.05Blackveland accepts it
19:11.10Blackvelwhat is before disa?
19:11.12Blackvelzaphfc?
19:11.20eKo1Anybody here use * with ss7?
19:11.32Blackvelmesi: thinking about the macro?
19:11.42Blackvelwhats the complete url? would be glad to have a look at it
19:12.11Blackvelbtw
19:12.13Blake0PSwhere does asterisk install to by default?
19:12.20Blackvelvoip-info.com has no webpage :)
19:12.26mesiBlackvel: Try it. It reads every single digit and when pressing # it dials out, but you can have several macros and exchange the Dial() command with others.
19:12.28Blackvel./usr/sbin
19:12.30*** join/#asterisk buddah (~hnic@208.179.86.5)
19:12.40Blackvelmesi: but I dont use that stuff
19:13.04BlackvelI really don't dail digits with pickup phone
19:13.07Blackvelthats silly :)
19:13.12mesiBlackvel: I mean .org
19:13.23Blackveli dont even have to press #
19:13.28mesiBlackvel: well, I do. How would you call an extension from an isdn phone then?
19:14.01mesiBlack: I call my computer on no 27 and then it is at extension 27. Whenever I want a different extension, I have to go on calling.
19:14.29Blake0PSI just make/make install, now asterisk is not in /usr/sbin
19:14.48Blackvelwell
19:14.51Blackvelisdn
19:15.18Blackvelyou have to dail on-hooked
19:15.23Blackvelyou dont need disa
19:15.26Blackvel_. works
19:15.42mesiBlackvel: What should _. do?
19:15.45mesimatch everything.
19:15.46Blackveland e.g 911000 + pick up sends in as extension directly
19:15.51*** join/#asterisk Gronker (~Gronker2@adsl-220-79-68.ags.bellsouth.net)
19:15.51Blackvelyes, matches everything
19:16.01mesiSo it matches the number of my computer box and I still can't call different extensions.
19:16.05Blackveli use goto afterwarsd
19:16.13Blackveldunno
19:16.15Blackvelworks here
19:16.26Blackveli have meetme room 500
19:16.30Blackveli enter in isdn telephone 500
19:16.34Blackvelpick up the telephone
19:16.34mesiBlackvel: Doesn't work here. When I call 27 (my computer) or 271 makes no ddifference.
19:16.40Blackveland 500 as extension gets called
19:16.57Blackvelwell
19:16.59Blackveli dont have that
19:17.05Blackveli use prefixes
19:17.15mesiBlackvel: Here I have 4 different isdn numbers. So that's just a few. I cannot use no 500 for my computer.
19:17.25mesiBlackvel: I guess my pbx is too stupid.
19:17.28buddahcan i hook a modem into an ATA?
19:17.31Blackveldiff -u bristuff-0.2.0-rc2a/zaphfc/zaphfc.c bristuff-0.2.0-RC7j/zaphfc/zaphfc.c|more
19:17.31buddahor would that not work?
19:17.36Blackvelis that the correct command=?
19:17.39mesiBlackvel: This is where disa comes in.
19:17.44Blackvelor would i have the latter first?
19:18.00Shidofaxing
19:18.03wildcard0hmm.  can i use math in asterisk variables?  like   SetVar(foo=${EXTEN}/100+3)  ?
19:18.04Shidobut not for ppp dialup
19:18.11buddahoh
19:18.18buddahhmm
19:18.21buddahok
19:18.31rikstawildcard0: see the wiki
19:18.49buddahso we'd have to get a digium card that allows us to plug analog into our * box?
19:18.50wildcard0riksta, which wiki page?  i looked around some already and didn't see anything that matched
19:18.58*** join/#asterisk oej (~oej@64-151-42-78-dhcp-kc.everestkc.net)
19:18.59rikstawildcard0: let me look
19:19.01buddahor should we jsut put them into our switch and avoid *
19:19.03wildcard0thanks
19:19.24mesiBlackvel: http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20call%20through
19:20.47rikstawildcard0: can't find the exact page, i know it's there somewhere, but not really got that much time to look right now
19:21.13Blackvelmesi: thanks
19:21.29rikstawildcard0: search for expressions
19:22.11wildcard0ah.  thanks.  i knew i'd seen that at some point
19:22.15miguellinuxhi, is there a command to check if a Zap is in use?
19:23.48Blackvelshow channels
19:24.25*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3771030.sympatico.ca)
19:24.33DaLionhey
19:24.51*** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
19:25.20jjhallOff topic, anyone know of a good free webcam software package?  I'm currently using FWink, but I want one that will save several pics of history instead of just updating a single picture.
19:25.42Blackveli could try to set a digit timeout of 1 sec
19:25.47Blackvelmaybe that works then also without disa
19:26.01Blackvelbut i guess the digits are not sent into asterisk at all
19:26.02tzangerhahaha
19:26.03tzangerI love Fark
19:26.08*** join/#asterisk mhnoyes (~mhnoyes@user-38lc148.dialup.mindspring.com)
19:26.09tzanger"Dick back to upright status after stroke"
19:26.10Blackvelany everyone refuses to tell me
19:26.13Blackvelhow I could debug this :)
19:26.35miguellinuxBlackvel, ok it shows the configured channels but which are talking an which not?
19:27.23sivanajjhall: check out webcam XP
19:27.50filehey, if you type in your pw, it will show as stars
19:27.55file******* see!
19:28.02sivanalol
19:28.04sivanaya right
19:28.19Blackvelconfigured? uh
19:28.25Blackvelshow channels should list nothing
19:28.30Blackvelif none are active
19:28.38Blackvelthats the case for my zap even
19:28.42jjhallsivana: Thanks.  Is the private version free?
19:28.54sivanathink so
19:29.09sivananot 100% sure
19:31.44*** join/#asterisk sivana (~sivana@165.154.13.35)
19:35.27*** join/#asterisk Sedorox (brandon@2001:4830:2018:a:20f:eaff:fe91:3778)
19:35.57*** join/#asterisk edO (~edO@67.Red-217-126-161.pooles.rima-tde.net)
19:36.29GoshenDoes anyone know of any other PTSN to VOIP other then ipkall.com(Washington state number), and voipuser.org?
19:36.36Goshenfree that is
19:37.12rikstayou should only be using voipuser if you are in the UK really
19:37.47jontowi don't know about free.. but i like this FXO card :)
19:39.16Goshenriksta: untrue...you should only be using voipuser if you get inbound UK calls, or you contribute via paypal, or you donate in other ways
19:40.34ManxPowerGoshen: VoicePulse, LiveVoIP, Teliax, Nufone, and about 500 others
19:40.51Goshen[12:36] <Goshen> free that is
19:41.02ManxPowerGoshen: Keep dreaming.
19:41.12Goshenipkall.com is free
19:41.35Goshenas is voipuser.org(gives UK inbound number)
19:41.39eKo1Define free.
19:42.00Goshenno cost to me on the incoming call
19:42.36*** join/#asterisk mogorman (~mogorman@146.229.176.173)
19:43.35miguellinuxBlackvel, ok you are right, I was using other command, thanks
19:44.04GoshenWhat are your favoite US toll free providers? I know Nufone and iax.cc provide them, but what are people using?
19:44.41miguellinuxGoshen, toll free... FWD
19:45.15miguellinux*18xx
19:45.32ManxPowerI don't like the fact that LiveVoip do not support G726 or Speex codec and they default to the ulaw codec.
19:45.41ManxPowerI'll also be trying out Teliax soon
19:46.28Goshenmiguellinux: yes, I use FWD for outbound toll free, works great...I am looking for a toll free provider to buy toll free lines from
19:47.05miguellinuxmmm Iconnecthere
19:47.16Goshenyou used them miguellinux?
19:47.28miguellinuxthey use to have a plan like this... some time ago
19:47.32*** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org)
19:48.04Goshenthats ok, I am just looking for feedback from people that use them
19:48.35miguellinuxI had but not on asterisk, just connected to cisco Ata 186
19:49.11*** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu)
19:49.43miguellinuxI'm in Peru... we have not too much bandwidth then I have problems with all providers... I'm not a good reference...
19:52.20*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
19:56.40*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
20:04.53*** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com)
20:04.53Shidobj
20:06.35bkw_yo yo yo
20:06.39bkw_Shido, bj what?
20:07.45Shidoheh
20:07.48Shidobjohnson
20:07.57MikeJ[Jayden]wow,that got your attention :)
20:07.57sivanalol
20:08.22ShidoI think im gonna buckle down and finish this stupid web site this weekend...
20:08.26Shidoso much crap to do
20:08.35Shidoour site sux ass
20:08.52Shidoevery other person that signs up tells us , your page needs an enema
20:08.59sivanaI think I'll have the first * mgmt system in .NET
20:09.28SexyKenHey bkw, You there?
20:10.13GoshenShido: what site?
20:10.17MikeJ[Jayden]well that answers your question :)
20:10.37*** join/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk)
20:10.55*** join/#asterisk jjhall (~chatzilla@24-116-128-106.cpe.cableone.net)
20:11.35*** part/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk)
20:12.25*** join/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk)
20:14.27sivana~seen aginamu
20:14.30jbotaginamu <~AgiNamu@216.230.151.230> was last seen on IRC in channel #asterisk, 20h 52m 18s ago, saying: '.'.
20:15.14sivana~seen jbot
20:15.15jbotjbot is currently on #ipaq (3d 17h 33m 53s) #how (3d 17h 33m 53s) #bzleague (3d 17h 33m 53s) #storm (3d 17h 33m 53s) #orkut (3d 17h 33m 53s) #asterisk-doc (3d 17h 33m 53s) #uphpu (3d 17h 33m 53s) #va (3d 17h 33m 53s) #asterisk (3d 17h 33m 53s) #nslu2-linux (3d 17h 33m 53s) #magnia ...
20:15.33sivana~putz
20:15.34jbotsomebody said putz was a person who constantly asks jbot questions in a channel instead of using /msg
20:16.25Mochi all
20:16.31sivanahello
20:17.12sivanacan anyone ping: admin.voctel.com ?
20:17.54mesiI can.
20:18.23sivanaok
20:18.32mesi20 packets transmitted, 19 received, 5% packet loss, time 19024ms
20:18.32mesirtt min/avg/max/mdev = 149.661/219.283/306.376/40.998 ms
20:18.47mesiNot very goog.
20:19.05sivanareally?
20:19.08sivanainteresting
20:20.19GoshenMesi: going to call you
20:21.42__Sparks_I am havibg trouble recroding calls - can someone give me a hand! - I tried the following "exten => _*009.,3,Record(/tmp/asterisk-recording:gsm)" but it didnt work!
20:22.24*** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net)
20:23.02*** join/#asterisk Dutts (~dutts@81.168.70.41)
20:23.15Duttshi all...
20:23.18invi_i have some ppl telling me that they can hear a bit of echo of their voice on their end. no echo on my end. im using TDM22B. any ideas?
20:24.10*** join/#asterisk lildivil (user@ool-18bc24d7.dyn.optonline.net)
20:24.31Duttscan anyone tell me when the leds on the back of the TDM400P come on? I am trying to install it, it doesn't appear even in my BIOS PCI devices list and I think I might have done something wrong
20:25.09fileis your motherboard PCI 2.2 compliant?
20:25.26tzangerDutts: only when the port is initialized
20:25.27Duttshmmmm ahh.... maybe not, it's pretty old, say 4-5 years old?
20:25.28*** join/#asterisk linagee (~linagee@netblock-66-245-227-114.dslextreme.com)
20:25.32tzangerDutts: i.e. after ztcfg
20:25.39fileif not then the TDM card won't work in it
20:25.52tzangerfile: well yeah, but ztcfg should fail then too I think
20:25.53linageeif you have two optical carrier network names, (OCNs) how do you know what path they take through the phone network?
20:26.03fileof course ztcfg will fail
20:26.07filethe drivers for the card won't even load
20:26.11Duttsit's a gigabyte 6BXD
20:26.14filethe card won't appear to exist
20:26.22eKo1linagee: What phone network?
20:26.29Duttsjust appears dead as a dodo..... not i BIOS PCI list as I said, and cat /proc/interrupts shows nothign either
20:26.55invi_Dutts: tried another PCI slot?
20:27.23linageeeKo1: say i call with my cellphone to my home phone. i know that my cell phone is "on" OCN 6672. I know that my home line is on OCN 9740. is there any way to do anything with these numbers?
20:27.25tzangerDutts: lspci -v -- does it show up?
20:27.26Duttsyeah..... even one that had my NIC in which defintely worked
20:27.31linageeeKo1: please help. my mind is hungry. :)
20:27.44Duttswhat should I be looking for? does it appear as Wildcard or something?
20:28.05lildivilhello all.
20:28.18eKo1linagee: I don't think so.
20:28.19invi_Dutts: TigerJet
20:28.20lildivili wonder if any1 can help me with this problem...
20:28.40lildivili started getting failed to authenticate on invite messages from my broadvoice provider on outbound calls... any clues?
20:28.47tzangerDutts: depends
20:28.57Duttsinvi_ & tzanger: no it's not there.... so I guess my mobo isn't PCI 2.2 compliant then =( doh!
20:29.00tzangermy particular PCI ID is b100:0003
20:29.05tzangerDigium doesn't hav etheir own PCI ID
20:29.06linageeeKo1: weird. an OCN number ties a bunch of switch "names" (the CLLI) together.....
20:29.19tzangerso mine shows up as "Individual Computers - Jens Schoenfeld Intel 537"
20:29.34lildivileverything was fine until today.
20:29.41*** join/#asterisk _6Flamez_ (lklk@00045a809589.click-network.com)
20:30.13linageeeKo1: what, there is no traceroute for phones? :)
20:30.14eKo1linagee: I have no idea how the CLLI is produced.
20:30.20linageeeKo1: magic. :)
20:30.29eKo1voodoo magic
20:30.49lildivilanyone has any ideas?
20:30.54linageeeKo1: i'm on SNMCCA11DS0
20:31.16*** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
20:32.41linageeeKo1: weird. OCNs are actually what tie the internet together... not just the phone system...
20:32.48invi_i have some ppl telling me that they can hear a bit of an echo. no echo on my end. im using TDM22B. any ideas? anybody?
20:32.52*** part/#asterisk Nebukadneza (~daddel9@i3ED6E7F3.versanet.de)
20:33.27*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
20:34.15eKo1linagee: You can always investigate by calling the appropriate LECs.
20:34.42modulus_hi eko1
20:34.51modulus_charlie bravo
20:34.53eKo1Hola.
20:34.55modulus_eko eko do you read?
20:34.57CleanerXwoho
20:35.00CleanerXcheck this out:
20:35.05linageeeKo1: "Hi, this is ____(Name) from Pacbell. I'm a phone engineer and I'd like to know the path of ___(OCN) to ___(OCN)" hehehe. :)
20:35.07*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
20:35.11CleanerXhttp://cgi.tu-harburg.de/~sxhl0490/ssf/index.php
20:35.36CleanerXif i read correctly thi girl has created a java proxy for securing voip traffic
20:35.41CleanerXwith java
20:35.46linageeCleanerX: hehehe. go girl!
20:36.30linageeCleanerX: i'd wonder if you can hide traffic to go through an HTTP proxy. :)
20:36.42lildivilso no1 has any clues about "Failed to authenticate on INVITE" messages....
20:37.17linageeCleanerX: sure the latency would suck, but say you lived in a country where there was no free speech. ;-)
20:37.18eKo1lildivil: Well, it's obvious what the problem is..
20:38.01eKo1CleanerX: I already found a security flaw in that: Java.
20:38.25CleanerXwell then port it and find another one: your os
20:38.28*** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
20:38.35*** part/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr)
20:39.03linageeeKo1: what are you talking about. java is a sandbox. just don't track in the mud. ;-)
20:39.03eKo1I know of a company that is doing that sort of thing on a larger scale.
20:39.43lildivileko: i'm not sure if i understand... what's obvious?
20:40.10Goshendoes this look right for rolling over to another provider if the first one doesn't work? http://pastebin.ca/6911
20:40.41eKo1lildivil: It means you're having authentication problems.
20:41.18lildivileko: with my provider??
20:41.26lildivileven though i register fine?
20:42.02eKo1You're using SIP right?
20:43.40lildivileko: yes, broadvoice is my provider
20:44.05eKo1Do a debug and look at the headers being sent.
20:44.06linageegiant udp l-something p-something
20:44.45ManxPower~docs
20:44.46jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
20:45.05eKo1OK. I'm outta here.
20:45.11*** part/#asterisk eKo1 (~bernd@63.245.57.70)
20:45.17ManxPowerGoshen: Not even close.
20:45.56ManxPowerYour dial is priority 3, so if the number is busy Asterisk will try to jump to priority 104
20:46.02ManxPowerBut you don't have a priority 104.
20:46.19ManxPowerAlso, for ALL busy numbers your Asterisk will dial once via each provider.
20:46.43ManxPowerYou need to look at ${DIALSTATUS} (see README.variables and "show application dial") to determine the CAUSE of the call ending.
20:47.10ManxPowerIn fact if you look at macro-stdexten in extensions.conf.sample you'll even see an example of how to do what you want.
20:51.02mesiThat's a great example.
20:51.17GoshenManxPower: thank you
20:51.35mesiCan I call skype members in any way with asterisk? Any known gateways or so?
20:51.42ManxPowerGoshen: You can thank me by finding me a job in Europe.
20:52.00GoshenManxPower: talk to mesi :)
20:52.59ManxPowerHello, mesi.
20:53.26GoshenMesi: no skype is a closed system
20:53.39GoshenMesi: there is one way...using FWD communicator on the pc
20:53.48Goshentime to go, ttyl
20:54.29mesiManx: Hello :-)
20:54.54mesiManx: What qualification do you have?
20:55.58Groobyhe's the president of Asteriskville
20:56.49tzangerhaha
20:57.10tzangerwereas I run the little run-down ghetto near the sewage treatment plant
20:57.36Groobywith lots of spanish fleas
20:57.51tzangerhaha
20:58.00tzangerno I thought that was leif
20:58.15*** join/#asterisk kietlak (~kietlak@11-mo3-6.acn.waw.pl)
20:58.27mesiManxPower: You're the president of Asteriskville? How about going for all those bounty stuff for asterisk? Like skype support and so on?
20:58.39Groobylol
20:58.41tzangerManxPower doesn't care for money
20:58.44tzangerhe wants men
20:58.49ManxPowerI am not president of anything.
20:58.51Groobyskype support..hehehehe
20:59.04Groobyand where do callers create the skype account?
20:59.33Duttshmmmm just looked on gigbyte's site and it says it is PCI 2.2 compat.... any other ideas?
20:59.36mesiWith the skype server? The API is open, I think.
20:59.44ctooleyAnyone got any advice for writing a billing system for a phone company?  This is starting to be a giant headache.
20:59.52ManxPowermesi: For the past 8 years I've managed the IP network for a Louisiana based real estate company.  Currently we have about 16 offices in Louisiana and Mississippi.  I've also managed the local linux servers, as well as the firewall, e-mail server, virus scanning, spam filtering, web server.
21:00.09ManxPowerAlso managed the Cisco routers and VPN connections, as well as data T-1 and Frame relay.
21:00.19sivanactooley: give me a month :)
21:00.33sivanactooley: I'm about 1/2 way done
21:00.36tzangerctooley: dont' listen to him, he wants you to run .net
21:00.43sivanalol
21:00.46ctooley.what?
21:00.48tzangerit's a trap!
21:00.59ctooleythat sounds like a microshaft trap
21:00.59ManxPowerI've used asterisk for about 1.5 years, currently manage 3 "small" asterisk personal asterisk servers for various people and also manage 3 smallish Asterisk systems for the company I do consulting for.
21:01.08Groobyasterisk .NET?
21:01.10Groobyhehehehe
21:01.11*** join/#asterisk footnote (~jhicks@67.141.135.121)
21:01.11ManxPowerBefore that I managed tech ops for three small ISPs.
21:01.18ManxPower(for about 3 years)
21:01.21sivanano it's not... I need someone to do a python or php version once I get this one done
21:01.22Chuji~.net
21:01.23jbotwhen .NET system goes online, human decisions are removed from the office environment. It contains Application Center 2000, SQL server 2000, Exchange 2000, Host Integration Server 2000, BizTalk 2000, Commerce Server 2000. They seperate every single bit of simple operation into many buzzword-servers. Don't you hate your money?
21:01.25sivana:P
21:01.36footnoteboy, you guys weren't kidding.   h.323 is a major PITA to get working.
21:01.39sivanalol
21:01.47spackle~microsoft
21:01.48jbotfrom memory, microsoft is Malfunction Inadequate Cruddy Reckless Operating System Originally For Trash
21:01.52ManxPowermesi: Is there anything else that you would like to know?
21:02.06footnotejbot is optimist
21:02.11mesiManx: That's great, I suppose it should not be too difficult to find a job in Europe.
21:02.48Groobyi already got dip on ManxPower
21:03.06GroobyI am sending him to China to be the president of Chinese Democratic party
21:03.06tzangerdip?
21:03.07Grooby:-D
21:03.12ManxPowermesi: The major problem is, of course, that I either need a job waiting for me before I move to Europe (since the employer has to fill out all the immigration stuff) or I need to go to Europe, find a job, then come back to the USA to apply for the work / residence VISAs
21:03.14tzangeris he a snack or something?
21:03.26ctooleysivana, what was the purpose of writing in .NET?  For the masochism of it?
21:03.31tzangerManxPower: I thought you were looking for work in toronto?
21:03.46footnoteGrooby: too much MSG
21:03.49ManxPowertzanger: That is still an option.  But Europe is back on the list.
21:03.52tzangerahh
21:04.03ManxPowertzanger: It's not like there are jobs jumping to get my services in Toronto.
21:04.12footnotemonosodium glutamate.
21:04.14Groobythat's an idea
21:04.22Groobyoutsourcing 1-900 numbers to asia
21:04.26Grooby:-D
21:04.34footnotesatanic
21:04.42sivanactooley: because, that's what we use in house (MS Shop), but I'd like to find someone to work with a php/perl port
21:04.47ManxPowertzanger: I found out more info from the immigration lawyer that I talked to and it's much more of a wait than I expected.
21:04.49Groobyok..back to my taco bell food
21:04.59tzangerNOT PHP
21:05.03tzangerthere are enough PHP things
21:05.03GroobyManx, why toronto?
21:05.07tzangerperl or python!
21:05.07sivanaor whatever
21:05.14tzangerManxPower: that blows
21:05.22tzangerI blame post 9/11 fears
21:05.23sivanatzanger: ok.. you're hired :)
21:05.24footnotetzanger: ruby :P
21:05.24ManxPowerMy ultimate goal was always Europe, Canada was just a transitional thing (if you consider 5 years to be "transitional")
21:05.38tzangerwhat's ruby got over python or perl?  I've never used it, I'm just curious
21:05.49footnotetzanger: 31337-ness
21:05.59ManxPowertzanger: I just want to get out of here.  The USA has been becoming less and less free since the early 1980s
21:06.04footnoteit's more like python than perl
21:06.16footnotebut different
21:06.17Duttsanyone got any ideas why my TDM 400P card isn't working? my mobo is PCI 2.2 compliant but I just get nothing from it?
21:06.29tzangerDutts: is the wctdm module loading?
21:06.34ManxPowerEurope, of course, has it's own issues, but not like the USA.
21:06.38tzangerDutts: if so, does ztcfg -vvv report goodness?
21:06.42tzangerDutts: come on, give us a hint
21:07.13footnoteManxPower: cheap flights to Morroco from EU!
21:07.34ManxPowerfootnote: I'm not sure I'd want to live in Morroco.
21:07.39Hmm-workanyone know what how long a lithium ion battery will hold it's charge on the shelf?
21:07.41*** join/#asterisk andrew` (~andrew@adsl-67-119-27-29.dsl.snfc21.pacbell.net)
21:07.43footnotewell, just in case.
21:07.48ManxPowerBut all offers will be seriously considered.
21:07.59footnotegreat hash in morroco...
21:07.59Duttstzanger: sorry, bit of a n00b... how do I see if the wctdm module is loading?
21:08.09lildivilis anyone on broadvoice here?
21:08.15footnotejust an observation of course
21:08.16Groobyi am on braodvoice
21:08.20tzangerDutts: run modprobe wctdm :-)
21:08.29lildivilgrooby: what version of * are you on right now?
21:08.45Groobyv1.0
21:08.48Groobystable
21:08.54Groobyfrom cvs
21:09.06DuttsI get Can't locate module wctdm... I'm running *@Home
21:09.16ManxPowerfootnote: The quality of the hash in a country is not really on my list of requirements.
21:09.17Duttswtcdm even
21:09.25Duttssorry I meant wctdm
21:09.27Dutts=)
21:09.32footnoteManxPower: scratch holland then
21:09.40lildivilgrooby: i am on 1.0.5 stable.  today i started getting 401 Unauthorized
21:09.40footnotehrm
21:09.44footnotefrankfurt too
21:09.49ManxPowerfootnote: I did not say it would EXCLUDE a country, just not a requirements.
21:09.51tzangerDutts: I have no idea baout @home
21:09.53footnoteohhh
21:09.54footnotehehehe
21:10.02Groobywierd
21:10.08ManxPowerQuality hash indicates an open minded social policy and THAT is a good thing.
21:10.11tzangerDutts: why that module's not present in @home is strange
21:10.13lildivilgrooby: did you have to add the rbroadvoice fix when you compiled *?
21:10.22Groobynot really
21:10.25footnoteManxPower: ok, explain afghanistan then
21:10.33footnotenepal
21:10.33ManxPowerfootnote: an aberation.
21:10.38Groobyi thought the patch was already submit into tree after last december
21:10.42footnotetwo abberations!
21:10.55Duttstzanger: I think there's an alternative I read on digium's onstall guide, one sec I'll have a look
21:11.00lildivilgrooby: that's what i thought too... everything was great up until today.
21:11.03Groobyi can still dial out and my gf just called here
21:11.07Groobywhere you at?
21:11.15lildivilgrooby: NY...
21:11.15footnoteManxPower: I have a friend that *really* digs Portugal.
21:11.21ManxPowerI guess I SHOULD have said "decriminilaztion(sp!) of soft drugs frequently indicates a liberal social policy"
21:11.25Duttstzanger: yeah it says 'Do a modprobe wctdm,(if you have latest cvs otherwise its modprobe wcfxs) '
21:11.25footnotehehe
21:11.40Groobyhmmm.....
21:11.46Duttsbut wcfxs reports No such device
21:11.46footnoteWhich is funny because decrim is really conservative.
21:11.53lildivilgrooby: on the phone to tech support now, probably something got screwed with my account again.
21:12.00footnotethey get all that law and order stuff backwards
21:12.07Groobyagain?  sounds like you have lots of problem w/ them
21:12.12ManxPowerI'm more interested in a good sized english speaking population, good public transportation, low crime (well low compared to the USA at least), friendly, educated, interesting place.
21:12.18footnoteit's liberal for govt to worry about what's in my pipe.
21:12.30lildivilgrooby: not lately.. but i did a few months back.
21:12.33tzanger??  they changed the fucking module name AGAIN??
21:12.42Groobylildivil.  I see....
21:12.50tzangerDutts: no such device means it odesn't see it, I'd asked yo ubefore if lspci -v showed it up
21:12.56lildivilgrooby: big time problems.  they were switchin carriers and my account was in the black hole for like 2 weeks.
21:13.05Duttstzanger: dunno.... I figured I was on the old one as it says it's wctdm if you've got the latest cvs....
21:13.07ManxPowerI've lived in or near New Orleans for 12 years at least.  Partying is not high on my list of priorities.
21:13.21Groobyouch
21:13.25Duttstzanger: no it doesn't....
21:13.29Duttsappear that is
21:13.47footnoteManxPower: heh, i know what you mean.  I'm from Jackson MS.  We stayed there every weekend for about eight years.
21:13.51tzangerDutts: well that's your problem.  :-)
21:13.51Duttstaznger: so I figured my mobo wasn't pci 2.2. compat but then I checked and it is..... so now I'm stumped
21:14.02ManxPowerfootnote: Benelux would by my prefered region in Europe, but as I said all serious offers will be considered.
21:14.03tzangerDutts: try the same card in another system
21:14.07tzangerDutts: just to see if it shows up
21:14.21footnoteManxPower: best wishes. cold winters tho
21:14.28Duttstzanger: yeah might have to..... so the leds on the back don't do anythign uless the port is up? no flash on boot up or antyhing/ just too see if it's even alive?
21:14.30footnotebrrr
21:14.31tzangerthe *great* think abou tlinux is you can literally take hte HDD and put it in another system and so long as you haven't completely customized it it'll adjust to the new ysstem almost seamlessly
21:14.32ManxPower"we want to hire you, but can't pay you much" is not a serious offer.  I get THOSE all the time.
21:14.36tzangerDutts: nope
21:14.42tzangerManxPower: AMEN
21:14.47footnotetzanger: I can do that with QNX! :)
21:14.50Duttstzanger: bah.... ok cheers mate fo ryour help I'll g=ive it a go in another system
21:14.58footnoteOr FreeBSD, or NetBSD, or ...
21:15.08tzangerfootnote: true
21:15.17tzangerfootnote: I was saying compared ot win32... but winxp is pretty good too I have to admit
21:15.21footnoteoh
21:15.31footnoteI dunno, I been boycotting them for seven years now :)
21:15.32sivanaall hail WinXp
21:15.40ManxPowertzanger: I spent 3 years working for ISPs that were on the edge of going bankrupt.  It's a terrible, stressful, depressing enviroment and I have no interest in going back to it.
21:15.41sivanaand MS :P
21:15.48footnoteI think NT 3.51 was the last thing I touched by M$
21:16.04tzangerManxPower: I hear ya :-)  I worked for an ISP too but they were (are) doing really well
21:16.06ManxPowerHard work is fine, but when they don't even pay you enough to have a phone line that's unacceptable.
21:16.10*** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl)
21:16.17QwellManxPower: preach on, brotha!
21:16.31footnoteManxPower: I recommend DSP development.
21:16.51ManxPowerfootnote: LOL!  I'm not a programmer.
21:16.56QwellManxPower: meh, soap boxes were made for a reason
21:16.59footnoteah c'mon. you can do it.
21:17.02ManxPowerI'm a network / *nix admin.
21:17.05QwellIf you aren't going to stand on it, somebody else will.
21:17.15footnoteManxPower: close enough!
21:17.33*** join/#asterisk mindCrime (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com)
21:17.34file2ManxPower lies
21:17.36file2he is actually a robot
21:17.43Qwellfile2: shh
21:17.45footnoteTI C6000 assembly code baybee
21:17.48footnotewoop
21:17.56ManxPowerfile has found me out!  My secret identity is JBOT!
21:17.57modulus_how about z?
21:18.12Qwellchortle?  mud much?
21:18.14spackle~manxpower
21:18.15jbotextra, extra, read all about it, manxpower is your God.
21:18.45footnotehrm. anybody here using Diabolic stuff?
21:18.49footnoteer, Dialogic
21:18.57Qwelldiabolic stuff, yes.
21:19.02Qwellfootnote: I use asterisk.
21:19.07*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-14-106.w81-248.abo.wanadoo.fr)
21:19.12footnoteI mean with *
21:19.26j0are there any test conferences out there i can join?
21:19.46tzangerwhat's the preferred conf app, meetme[2] or app_conference?
21:19.50tzangerwhat are teh dev conferences using
21:19.51footnotegood. i hate dialogic.
21:20.10modulus_i hate diabolic
21:20.15modulus_HAHAHHAHAAHAHAAA
21:20.16footnotehehe
21:20.23mesitzanger: well for meetme I need ztdummy, do I need it for app_conference too?
21:20.31footnotemodulus_: you mean MUAHAHAHA
21:20.34tzangermesi: not sure, I'ma  conf newb
21:20.41modulus_*eviLGrin*
21:20.44mesitzanger: I see.
21:20.49spacklemodulus_ = Mandark
21:21.05spackleHA               HA HA HA              Ha           HA HA HA
21:21.10footnotedam bias letfis
21:32.20mesitzanger: Sorry, I cannot test this. app_conference isn't installed here.
21:34.01lesouvageI want an internet radiostation for my musichold.  I followed the instructions of http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but it's not working. Any suggestions?
21:36.25ariel_lesouvage, I don't think that internet radio's are good for an asterisk server. It takes b/w away from voip.
21:36.53*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
21:37.07lesouvageariel: what do you mean with b/w?
21:37.23BlackvelIf I have some outgoing call with Dial command
21:37.35Blackvelwhich I also pass some variables to
21:38.05Blackvelcan I make it work, to call in the reply (the 2nd command) an AGI with the same variables I had just passed to Dial command before?
21:39.14lesouvageariel: what I understand is that it is not such a good idea. Thanks for the reaction/advice.
21:40.11mesiHow can I download app_conference from cvs?
21:43.35ariel_lesouvage, b/w is Band Width.
21:43.58lesouvageariel:ok
21:44.16*** join/#asterisk Damin_Mobile (~pocketirc@ip68-99-51-230.cl.ri.cox.net)
21:49.50Damin_MobileRock N Roll!
21:52.46Damin_MobileFile; Got a wi-fi card. ;)
21:53.42fileDamin_Mobile: excellent
21:53.46fileDamin_Mobile: how well is it working out?
21:53.56Damin_MobileGood...
21:54.27filewhat chipset?
21:57.28Blackvelwho got FWD running with asterisk behind NAT?
21:57.37Blackvelmaybe with SARP?
21:57.58Damin_MobileNo clue. It is from san disk
21:58.33filefun
21:58.54fileBlackvel: just set externip... map the ports, it's all good
21:58.56ariel_what is sarp?
21:59.04BlackvelI wish
21:59.09ManxPowerand localnet= of course.
21:59.16Blackvelhttp://sarp.sourceforge.net
21:59.18fileunless you want to... tempt fate!
21:59.28Blackvelexternip works with others
21:59.30Blackvelbut not with fwd
21:59.33Blackvelbesides the register is fine
21:59.38Blackveli can not upgrade to latest cvs
21:59.39fileyes it does, I've done it before
21:59.39ManxPowerfile: Why would you want to tempt Fate.  She's a fickle mistress.
21:59.50Blackvelhm
21:59.51Blackvel:(
22:00.06Blackvelshouldn't the fwd echo test work then?
22:00.10*** join/#asterisk dabba (~d@raptor.lgw.ip6net.net)
22:00.13Blackvelbut I can't hear any audio
22:00.25ManxPowerBlackvel: sounds like your portforwarding is not set up right.
22:00.30ariel_Blackvel, switch from sip to iax for fwd
22:01.20ariel_Blackvel, http://www.fwd.pulver.com/advanced/iax
22:01.23ManxPowerYou need to forward the RTP ports in addition to the SIP control port 5060/UDP.
22:01.31Qwell10000-20000, right?
22:01.49ManxPowerQwell: Well that's the default ASTERISK users.  Who knows what non-Asterisk devices use.
22:02.04ManxPowerCisco defaults to 16384 - 32758 UDP.
22:02.15ManxPowerX-Ten defaults to something else.
22:02.22*** part/#asterisk dabba (~d@raptor.lgw.ip6net.net)
22:02.26ionixwhere can I download the SER proxy ?
22:02.35ManxPowerionix: ask on #ser.
22:02.41fileor Google
22:02.41ionixk thx
22:02.45ionixdidn't know it existed
22:03.25ManxPowerionix: Not many Asterisk users need SER.
22:03.54ionixgood for them :) But I do
22:04.19ManxPowerMost Asterisk users that think they need SER really don't need it.
22:04.35ionixhmm
22:04.39Damin_MobileYou didn't know about Google?
22:04.46ionixDamin: hehe #ser
22:05.07ionixManxPower: I will be offering VoIP to Canadians. Planned growth in 12 months is 10,000 subscribers.
22:05.13ionixI guess SER would be usefull ;)
22:05.27ManxPowerPretty much the only time Asterisk users need SER is of they have a large scale deployment of SIP endpoints that are behind NAT and you want the devices to send audio directly between each other, rather than via Asterisk.
22:05.34Nuggetyou're doomed to fail if you can't even locate the tools you expect to need.
22:05.38ManxPowerionix: Then you are not "most Asterisk users"
22:05.40*** join/#asterisk dabba (~d@raptor.lgw.ip6net.net)
22:05.49fileeveryone and their mother is doing VoIP now
22:06.03ManxPowerionix: Even then, do you REALLY think there is going to be a lot of inter-customer traffic?
22:06.15ionixNugget: Actually, I googled it and found iptel but I was expecting an URL with SER in it.
22:06.20ManxPowerfile: Yeah.  But they are doing it poorly and with lack of features.
22:06.27ionixManxPower: I want to use SER to register SIP phones. Then transfer that to the Nextone via SIP
22:06.27fileManxPower: hell yes
22:06.42ManxPowerionix: I don't see Asterisk in that mix.
22:06.54fileuh oh
22:06.59ionixAsterisk would be used for conferencing, voicemail etc
22:07.16fileyup yup
22:07.33QwellManxPower: hmm, thanks for the tip
22:07.43QwellI thought 10k-20k was a SIP thing.  Period.
22:08.06QwellYou would think that there would be a set range though, wouldn't you?
22:08.17ManxPowerQwell: Not at all.  The two endpoints figure out what ports they use.  There's not specification.
22:08.35ManxPowerQwell: The people that designed RTP were obvious long time meth users.
22:08.45ManxPowerEither that or they were just plain sadistic.
22:08.59Damin_Mobile10,000
22:09.33TrepaliumGiven how much faster computers are these days, it's hard to believe that Speak&Spell is the apex of text-to-speech technology.  =/
22:09.38ManxPowerWe tell Asterisk to use 16384 - 32678 and make sure all our phone are set for that port range.
22:12.08QwellManxPower: Just because, or?
22:12.17QwellYou just like 16-32 bit ranges?
22:12.35*** part/#asterisk dabba (~d@raptor.lgw.ip6net.net)
22:12.43QwellTrepalium: I had a speak n spell...those things rocked.  I wish I could find one for my son.
22:12.48ManxPowerQwell: all our routers are Cisco and at one time we were strongly considering Cisco phones.
22:12.54QwellManxPower: ahh
22:12.56Qwellmakes sense
22:13.02ManxPowerand it's a nice range.
22:13.06ManxPowerpowers of 2 rock
22:13.08*** join/#asterisk zotz (~zotz@24.231.32.191)
22:13.10QwellManxPower: If you ever have extra, you know where to ship them. ;]
22:13.14*** join/#asterisk Rick_Hunter (~rhunter@adsl-69-209-173-100.dsl.sfldmi.ameritech.net)
22:13.15QwellManxPower: SIP^2
22:13.52ManxPowerWe may eBay some of our non-SIPura/non-Polycom phones.
22:13.57Mochi
22:14.04QwellManxPower: I'll give you $50 for a 7940!
22:14.17Goldenearcan I give both a name and number extension for an IAX2 endpoint. so I can call john either with IAX2/guest@peer.net/1234 or with IAX2/guest@peer.net/john.
22:14.24ManxPowerQwell: Did you see my diatribe on "insulting" earlier.
22:14.31QwellManxPower: Nope, do tell.
22:14.44ManxPowerQwell: don't worry about it.
22:15.06ManxPowerQwell: That's why people without children will soon take over the world.
22:15.09QwellYou're insulting me by proxy?  How rude. :P
22:15.16ManxPowerI'll put in a good word for you when the revolution comes.
22:15.20MocManxPower, so your getting rid of your cisco ? ;) hehe
22:15.22Qwellexcellent
22:15.40Qwellout of curiousity, why are cisco ip phones so expensive?
22:15.42ManxPowerMoc: The 7905G is a cool phone, but there are lots of cheaper good phones out there now.
22:15.43GoldenearIs there a place where I can find how to do that ?
22:15.45QwellBecause they can, or what?
22:16.01ManxPowerQwell: Because they are some of the best phones in the world.
22:16.04QwellI could see paying $125-150 for a 7940
22:16.10QwellI'm sure they are, yeah
22:16.11TrepaliumThe five letters on them, C, I, S, C, and O.
22:16.15footnotebah
22:16.17Blackvelhm
22:16.18Blackvelweird
22:16.21Blackveli had one test
22:16.23QwellTrepalium: each letter is $35?
22:16.24footnoteI used to work for Cisco.
22:16.29Blackvelthis FWD setup seems working
22:16.30Qwellfootnote: hook it up then
22:16.31footnoteIt ain't THAT great.
22:16.32MocManxPower, Im sold to polycom ;) i would have get a IP 300 for that range
22:16.33ManxPowerQwell: HA!  Remember to add a power supply and SIP firmware to the cost of a Cisco phone.
22:16.38QwellManxPower: indeed!
22:16.59TrepaliumThey're good phones, but you pay for the name (and percieved quality), like most name brand stuff.
22:16.59Qwell$30 for the power blovck(monitor died, typing blindly), and about $100 for the firmware
22:17.04Mocyea, and it aint a very flexible phone either ..
22:17.11ManxPowerI like Polycom as well.  That's the phones we will standardize on.
22:17.11footnotecheap plastic
22:17.25footnotei want the MANLY model phone dammit
22:17.31footnotediamond plate
22:17.42QwellI honestly can't see myself paying more then $150 for a 7940, $175 for a 7960
22:17.47Qwellused or otherwuse
22:17.50Blackvelwho owns some sipura 2000
22:17.52Qwellwise...I love jack and coke
22:18.00ManxPowerfootnote: Well splatter some deer blood on it after you've had a 6-pack of beer.  Manly enough for you?
22:18.03Blackveland would have time for a short FWD or whatever test call?
22:18.08MocI got spa2000 and 3000, Wish I had the provisioning tools ..
22:18.15footnoteManxPower: still plastic, already tried all that
22:18.18Moctime is something I dont have throught
22:18.22Blackvel:)
22:18.40Blackvelwho uses this on a regular basis: http://sarp.sourceforge.net?
22:19.12footnotethey don't have mailing list?
22:19.19QwellManxPower: Are the ciscos honestly worth $250-300?
22:19.32QwellI rarely trust somebody's opinion...I'll trust yours.
22:19.47BlackvelI have some problems (lack of understanding) how to setup sarp
22:19.49bjohnson_is Brian Capouch here?  I want to know what is difference between his asterisk-bc and asterisk-cvs ipkgs
22:19.58footnoteQwell: I had one on my desk.  It was pretty much a phone.
22:20.08Qwellfootnote: But, you didn't pay for yours, correct?
22:20.08footnoteThat said Cisco on it.
22:20.13footnotehell no
22:20.17Qwellso, heh
22:20.17MocQwell, get a polycom phone instead ...
22:20.22footnoteI wouldn't pay that much for a phone!
22:20.23QwellI mean, I know they're good, but...
22:20.27Mocyou have alot more for the buck ..
22:20.29Qwellfootnote: neither would I
22:20.36MocI did ;)
22:20.36QwellMoc: suggest a specific model?
22:20.46Qwellmany people do, but I'm a cheapskate
22:20.47MocQwell, IP 500, or IP 600
22:20.52footnoteIt's just a stupid phone for cryin out loud
22:20.53cbachmanbjohnson_ where is the asterisk-bc ipkg?
22:21.03Qwellfootnote: Thats what I'm saying.
22:21.08MocI got a 7960, IP500, IP600 and SoundStation IP3000
22:21.10QwellMoc: Those are SIP I assume?
22:21.12footnoteand then you gotta pay maintenance fee to get bug fixes
22:21.17MocQwell, yes
22:21.20Blackvelwho uses iaxtel here? :)
22:21.24Mocip500 can be found for about 180$
22:21.26QwellMoc: estimated price?
22:21.27Qwellahh
22:21.31QwellBlackvel: I do
22:21.36Qwellsorta...
22:21.47Qwell$180 is still too much IMO
22:21.53Blackvelit was working some time ago (not sure when)
22:21.59BlackvelI am trying to get it working again
22:21.59QwellI would pay $180 for a 7960, tops
22:22.06MocQwell it depend of what you are looking for..
22:22.07footnoteQwell: $200 and I can deal.
22:22.14footnoteIf it's QUALITY
22:22.16BlackvelI would be interested for a call, if you have time maybe
22:22.18Qwellfootnote: I could do $200, with a powerblock
22:22.27Moclol
22:22.28QwellI wouldn't do more then $150 for a 7940 though
22:22.31Qwellmy wife would kill me =x
22:22.33mesiAre there some kind of online phonebooks that I can register with an show people, that I want them to call me?
22:22.37Mocyou really ratter have a limited Cisco phone ?
22:22.42footnoteI refuse to pay more than $200 for a phone, period.
22:22.47Qwellfootnote++
22:23.06MocQwell, check the biggining of : http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones
22:23.10QwellSomebody answer an honest question for me.
22:23.11footnotehrm
22:23.15Mocmostly everything in there, Cisco can't do
22:23.21footnoteI oughta build a hackerphone.
22:23.29footnoteColdfire based.
22:23.30QwellWhat is the difference between a $10 analog phone, vs a $100 IP phone?
22:23.33footnoteopen source
22:23.40footnoteQwell: needs a processor
22:23.42Qwellyes, I understand its IP vs analog
22:23.53Qwellyes, I understand it needs a low power proc
22:23.56footnoteneeds ad<->da
22:24.05Qwellbut, $90 difference, honestly?
22:24.07cbachmanoh, asterisk-bc is 1.0.3-1
22:24.08footnoteyeah
22:24.20QwellThe cheapest IP phone is what, $89?
22:24.23TrepaliumThe price will probably come down as they become more and more popular.
22:24.28MocQwell, if your looking for a cheap phone, either get the sipura phone, or a cheap analog phone with a SPA-1001
22:24.53QwellTrepalium: hopefully
22:24.53footnoteTrepalium: yeah
22:24.53TrepaliumThe volume is just too low.
22:24.53footnotenot for long ;)
22:24.53QwellMoc: I've been considering a spa
22:25.03MocQwell, spa will allow you to have a very cheap wireless solution..
22:25.06Qwellprobably an spa 2000(is that the one?)
22:25.15QwellMoc: wireless bridge, or what?
22:25.18*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
22:25.23Moccompare to my IP 600 at 300$ + a Wireless headset for 300$ ;)
22:25.30Qwellouch
22:25.32MocQwell, cheap 20$ VTech wireless phone
22:25.43QwellMoc: plus the cost of an SPA
22:25.51Mocyea it about what, 64$
22:25.53Qwelland you need the base station, heh
22:26.00footnotemade in china out of old styrofoam peanuts
22:26.05footnotecheeeep plastic
22:26.06QwellI do like the vtechs though.  Tons of sales at bestbuy and the like
22:26.21MocI got my last vtech wireless phone for 25$ CND so, Im sure it cheaper in the US
22:26.39Mochave 2 line caller id/name display
22:26.48*** join/#asterisk marc324 (~marc32344@64-34-29-65.dsl.teksavvy.com)
22:27.04QwellSPA is only $65?
22:27.12Blackvelfor SARP I have to set this:
22:27.13Blackvel$rtp_start_port = '7070';
22:27.13Blackvel$rtp_end_port = '7080';
22:27.15QwellI was assuming more like $100+
22:27.26Blackvel$proxy_port = '5060';
22:27.29Qwellanyhow, need food...
22:27.30footnotehrm.  a coldfire+TMS320C549
22:27.31MocQwell: http://www.voipsupply.com/product_info.php?cPath=96_117&products_id=320
22:27.35ManxPowerBlackvel: why don't you set it to the same as Asterisk's default?
22:27.36footnoteyou could do that cheep
22:27.39QwellMoc, ManxPower: thanks for the help
22:27.40Blackvelwhat do I do set this sarp proxy to?
22:27.44QwellMoc: I don't like voipsupply ;/
22:27.47Blackvelyes I could do this
22:27.52ManxPowerQwell: no problem.  You can thank me by finding me a job in Europe.
22:27.57BlackvelI wonder if I have to use different SIP and RTP ports
22:28.00MocQwell, i love the, had good service fro, them
22:28.06Blackvelor the same as asterisk is using
22:28.08*** join/#asterisk algorithmn (~na@ool-18bce89c.dyn.optonline.net)
22:28.12QwellMoc: heh
22:28.18Mocjust ordered 13 phone the other day, got it in 2 day (and im in canada)
22:28.19Qwell"Thank you for pointing that out, the pricing has been changed."
22:28.24ManxPowerBlackvel: It does not REALLY matter, but it does make things simplier.
22:28.29Blackveldo I have to change asterisk to use that proxy somehow?
22:28.37QwellSo, the second time, the price was still wrong
22:28.38Qwell"It should be $309.95, I'll check and see what happened there. "
22:28.51QwellThey hosed the pricing TWICE in a row.
22:28.51*** join/#asterisk Bruns (bruns@pool-141-153-151-58.nwrk.east.verizon.net)
22:29.02QwellI no longer trust them.  They lost my business.
22:29.05ManxPowerQwell: I would not have pointed it out the SECOND time.
22:29.08Blackvelsarp and asterisk are on the same linux server anyways, the dsl router forwards the correct ports to the server yet
22:29.11Brunsanyone here got a moment to help me figure out something with broadvoice?
22:29.18MocI love Cnd/Quebec law, if they showed that price, it the price the HAVE TO SELL IT ..
22:29.22Blackveland did I understand it the right way
22:29.24ManxPowerBlackvel: You are going to have problems.
22:29.25QwellManxPower: Sometimes I'm an asshole, sometimes I'm too kind,
22:29.31*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:29.33ManxPowerQwell: Me too.
22:29.34Qwellthe first time, I should have been an asshole, and bought it
22:29.38Blackvelthat the sarp proxy even works when it is behind the dsl router
22:29.43Blackveloh thats bad
22:29.44QwellThen sued if they didn't honor the pricing? :P
22:29.57BlackvelI would have to run both on different machines?
22:29.59QwellNo, but really, the price was about $100 off retail
22:30.02ManxPowerBlackvel: SIP sepcifies port 5060 for SIP signaling.
22:30.07Blackvelyes
22:30.08QwellThen the second time, it was like $50 above retail
22:30.12Blackvelasterisk is configured to this
22:30.13ManxPowerYou can't have two devices opening the same port on the same IP.
22:30.16MocQwell, no need, we have a customer protection office, just send them the info, and they deal with the marchant
22:30.20QwellSo, voipsupply is unfortunately out
22:30.24Blackveli am not sure what that proxy does
22:30.32Blackveldoes it open it's own connection?
22:30.38QwellMoc: "we"?
22:30.48MocQuebec Province
22:31.04Qwellahh
22:31.16QwellI was being kind, and letting them know about the error
22:31.30bjohnson_cbachman: http://lestblood.imagodirt.net/archives/83-Asterisk-on-OpenWRT.html .. asterisk-bc is from src local http://12.176.248.4/ipkg
22:31.33QwellI would have been just fine, if they would have "fixed" the problem the first time
22:31.47Qwellbut instead, they broke it even more, raising the price far above retail
22:32.32Blackvelwhen sarp opens its own connections to the sip providers
22:32.41Qwellanyhow, I have a steak waiting for me...bbl
22:32.44Blackvelhow would I have to configure asterisk to use SARP?
22:33.13ManxPowerBlackvel: You don't.  You just tell Asterisk to send the call somewhere on the Dial line.
22:33.26ManxPowerAsterisk sees it as JUST ANOTHER SIP DEVICE.
22:33.29Blackvelis it for sipura ATAs/grandstream the variable: outgoing proxy server?
22:33.59Blackvelyou have to register sarp to *?
22:34.04Blackvelahhh, you have to change the dail command?
22:34.21Blackvelto pass it over to SARP?
22:34.47BlackvelI would be so lucky if there would be some dummy FAQ about SARP+ASTERISK on voipinfo
22:35.49Brunshmm, I've started getting errors with broadvoice suddenly today - Mar  5 17:21:44 NOTICE[29947]: chan_sip.c:7539 handle_response: Failed to authenticate on INVITE to '"Brian" <sip:3174361024@sip.broadvoice.com>;tag=as5a8f087e'
22:36.05Brunspassword is correct (checked with broadvoice
22:37.54BlackvelSARP: They will rewrite your SIP messages and have some kind of UDP/RTP proxy for the media stream
22:38.17Blackvelfor this is, is it even needed that SARP opens it's own connections?
22:38.33Blackvelit would be sufficent to rewrite the UDP packets which are sent out on port 5060
22:38.52GoldenearMay be my question has been asked a million times ... but I can find the answer anywhere... :( so please could any one answer me ?
22:39.06*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
22:39.06*** mode/#asterisk [+o bkw_] by ChanServ
22:39.08Goldenearcan I give both a name and number to an extension for an IAX2 endpoint. so I can call john either with IAX2/guest@peer.net/1234 or with IAX2/guest@peer.net/john.
22:39.37Qwell${JOHN} ?
22:39.51Nugget1234 and john are not the same extension.
22:39.58Nuggetthere's no law that says you can't have both, though.
22:40.30bkw_on the road again
22:40.34Goldenearhow I define that ?
22:40.38Goldenearin iax.conf ?
22:40.41Nuggetthe same way you define any extension.
22:40.44Nuggetno, in extensions.conf
22:41.03bkw_lets see how good this GPRS does going 70 for a few hours
22:41.04Qwellbkw_: fun
22:41.33Blackvelbkw: you are on GPRS right now? with pda?
22:41.55bkw_no
22:41.56bkw_ibook
22:41.59bkw_plus usb cable
22:42.01bkw_plus v180
22:42.16bkw_this works pretty good so far
22:42.22Blackvelibook?
22:42.23Blackvelwhats that?
22:42.25Blackvellaptop?
22:42.33bkw_you live under a rock?
22:42.34Nuggetapple.com/ibook
22:42.37Qwellheh
22:42.44bkw_go for it file
22:43.00drumkillabkw_: on the way to Tulsa?
22:43.05bkw_I  might check my email in  afew
22:43.11bkw_drumkilla, yes
22:43.14bkw_i'm gonna IRC the whole way there
22:43.24Blackvelno not really, but too busy with some stuff in my life to check over and over again for the latest news
22:43.30Qwellbkw_: setup a webcam on the dash
22:43.37bkw_I do have my isight
22:43.40bkw_I could ya know
22:43.44Qwelldo it@
22:43.46Blackvele.g I bought finally a pda with navigation software
22:43.51tuxinator_linux~whereis MikeJ[Jayden]
22:43.54bkw_I don't got da bandwidth fer that
22:43.56drumkillawith a 1 frame/minute refresh rate
22:44.13fileSOON VON! SOON SOON SOON!
22:44.15Blackveleven that has been available also years ago :(
22:44.34tuxinator_linuxfile going to VON?
22:44.41bkw_so am I
22:44.41fileyessssss
22:44.46drumkillaso am I!!!
22:44.59Qwelldrumkilla: Buy me a plane ticket, and I'll be there
22:45.00bkw_haha
22:45.03bkw_this is kinda neat
22:45.08tuxinator_linuxI will be at Meet *
22:45.11bkw_greg is driving .. i'm on the internet
22:45.13Qwellbkw_: WATCH OUT FOR THAT TRUCK!
22:45.26tuxinator_linuxwhere is the page for this cam?
22:45.32drumkillawe'll be at the booth all week, everyone stop by  :)
22:45.33bkw_omg
22:45.37filewe'll all be hanging out at the asterisk pavilion so if anyone is there, come visit us!
22:45.38bkw_this bitch about run into us
22:45.44bkw_she needs to learn to drive
22:45.47Qwellbkw_: Start watching the road. :p
22:45.55bkw_i'm not driving
22:45.57bkw_greg is
22:46.02tuxinator_linuxSomone have a spare exhibit ticket?
22:46.03Nuggethttp://bash.org/?195969  <-- bkw_
22:46.04Qwellgreh: Start watching the road. :p
22:46.32GoldenearIf I understand correctly, I have to use two lines for each endpoint in extensions.conf :
22:46.33Goldenearexten => 1234,1,Dial(IAX2/John,,rm)
22:46.35bkw_haha
22:46.35Goldenearexten => john,1,Dial(IAX2/John,,rm)
22:46.36GoldenearIs this correct ?
22:46.37bkw_i'm not driving
22:46.42NuggetGoldenear: that's one way to do it.
22:46.45bkw_no
22:46.51bkw_Goldenear, nope not at all right
22:46.53Nuggetanother would be "exten => john,1,Goto(1234)
22:46.57bkw_they both do the same thing
22:47.02bkw_but stop and think about it
22:47.13Qwellexten => john,1,Goto(${JOHN})
22:47.15Qwellwould work too
22:47.23bkw_rm -rf /
22:47.25bkw_that works too
22:47.28*** join/#asterisk mrgoby (~mrgoby@141.211.162.97)
22:47.29bkw_:P
22:47.31tuxinator_linuxAny of you using 2.6?
22:47.40drumkillaI'm using 3.2
22:47.44NuggetI'm using 5.3
22:47.46tuxinator_linuxkernel 2.6
22:47.50Nuggetkernel 5.3
22:48.10ManxPowerdrumkilla: I accidently installed CVS-HEAD on a production server last night.
22:48.15mrgobyi got 2.6 running on my laptop...  asterisk seems okay.... though....  heh....  i only use it when i have to /testing
22:48.19ManxPowerdrumkilla: I think I'm going to need therapy.
22:48.21drumkillaManxPower: ha
22:48.32Goldenearexten => john,1,Goto(1234) : That's exactly what I want, thanks :)
22:48.33tuxinator_linuxI wrote a Wiki page and need it to be corrected
22:48.37ManxPowerdrumkilla: hard lock when loading wcfxs/wctdm.
22:48.42bkw_getting a little packet loss
22:48.44tuxinator_linuxhttp://www.voip-info.org/tiki-index.php?page=Asterisk+CentOS-4.0+Zaptel
22:48.46tzangerbeer and yogurt... I think there might have to be a law against that
22:48.48drumkillaManxPower: when I meet up with Mark tomorrow, I'll get 1.0.7 out because of that SIP problem
22:48.51Nuggetso correct it, tuxinator_linux.
22:48.51bkw_whtat do I expect at 70mph
22:48.58bkw_bet the wind is catching them :P har har har
22:49.07drumkillaManxPower: and I'm going to do an RC on the bug tracker for future releases :)
22:49.09tuxinator_linuxI lack enough experience to correct it
22:49.09ManxPowerdrumkilla: have you considered a 1.0.7rc1?
22:49.25drumkillayeah, that's what I will do
22:49.34ManxPowerdrumkilla: 1.0.4 and 1.0.6 both had showstopping bugs.
22:49.36ManxPowerThat's not good.
22:49.45ManxPowerOf course if you ran 1.0.x yourself..... *tease*
22:49.50drumkillawell, even if I did
22:49.51tzangerlook at it this way, ManxPower... they're stable...  :-)
22:49.57drumkillaI probably wouldn't catch stuff
22:50.04drumkillanot everything, anyway
22:50.20drumkillaanyway, we'll have RC's from now on ...
22:50.35Qwelldrumkilla: on stable?
22:50.39drumkillayes
22:50.42Qwellhmm
22:51.14Qwell1.0.6-rc-drumkilla_is_your_daddy
22:51.14Qwell?
22:51.21drumkillaexactly
22:51.41drumkillaI might just shorten it to 1.0.6rc1
22:51.42Qwellwhatever works
22:51.50file1.0.6rc1-muffin
22:51.52Qwellyeah...
22:51.59Qwellas long as the changelog reflects it
22:52.14filemy muffin is private omg
22:52.15Qwellbkw_: ?
22:52.30Nuggetdigium needs a public cvsweb.
22:52.31Qwellfile: filesmuffin.sourceforge.net
22:52.41QwellIts GPL, according to my references
22:52.44drumkillaI will need people to verify that the RC's are working alright
22:52.46fileha
22:52.50drumkillaso I'll have a bug open for that
22:52.53Qwelloh, nm, its bsd
22:53.45ManxPowerOh!  I always thought file's muffin was under the Artistic License!
22:53.54QwellManxPower: autistic
22:53.57tzangerManxPower: no it's the ManWhore license
22:54.09drumkillaLGPL - he allows proprietary linkage to his muffin
22:54.09Qwellfile: For the record: I'd do ya
22:54.14Qwelldrumkilla: :P
22:54.18ManxPowertzanger: I wasn't aware he charged.
22:54.20*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
22:54.20*** mode/#asterisk [+o bkw_] by ChanServ
22:54.32QwellManxPower: The first one is always free.
22:54.33*** join/#asterisk ionix (~ionix@65.94.92.140)
22:54.40QwellIf you're no good, then he charges
22:54.41ariel_good night all
22:54.43bkw_guess its just laggy
22:54.45bkw_haha
22:54.45bkw_or my ip changed
22:54.49Nuggetheh
22:55.14filepoor poor bkw_
22:55.14Qwellmmhmm
22:56.39Blackvel23:52:07 ERROR (CRITICAL): Could not bind to 5060 (Address already in use)
22:56.47Blackvellooks like SARP tries to bind to 5060
22:56.53Blackvelbut there is my asterisk running yet
22:56.58Blackveltoo funny
22:57.12BlackvelI have no clue how to set it up
22:57.13Blackvel:)
22:57.27bkw_duh
22:57.34bkw_bind them to diffrent IP's
22:57.37bkw_or diffrent ports
22:57.47bkw_this is like first grade Unix/Admin stuffs
22:58.16drumkillabe nice!
22:58.19sivanahehe
22:58.21Blackvelwell
22:58.28sivanaI love it when bkw_ helps out
22:58.29Blackvelwhat should I do with that RTP stuff?
22:58.39BlackvelI would have to add the SARP ports to my firewall?
22:58.39fileI should go have a shower, pack, make sure I have everything, and panic
22:58.43Blackvelport forward them?
22:58.54Blackvelhow do I have to configure asterisk with SARP?
22:58.54sivanafile: where ya going?
22:58.59filesivana: VON.
22:59.02sivanacool
22:59.03bkw_be nice?
22:59.05Blackvellooks like SARP is registering at the providers?
22:59.09bkw_honestly
22:59.11Blackveli have no clue about this s****
22:59.13drumkillahehe
22:59.19fileVONNNNNNNNNNNNNNNNNN!
22:59.21fileVON VON VON
22:59.34filebbs
22:59.34bkw_hehe
22:59.39bkw_file don't piss yourself
22:59.43drumkillaI might.
22:59.52drumkillabkw_: we should tackle file at the airport
23:00.03bkw_why?
23:00.04drumkillaa big group tackling.
23:00.09drumkillabecause he always says that on IRC
23:00.39drumkillatime to go, bye guys
23:00.52Blackvelwould i have to use in asterisk "sip outgoing proxy" to actually make use of the sip proxy SARP?
23:00.52ckruetze_VON is too far away, VON Europe maybe or maybe VON somewhere else, but not North America :(
23:01.03bkw_yep
23:01.03bkw_we should
23:01.04bkw_and I might just end up with a rental car
23:01.04bkw_depends
23:01.04bkw_why?
23:01.06bkw_no GPRS?
23:01.12bkw_heheh
23:01.14bkw_i'm mobile baby
23:01.39DaminHmmm...
23:01.39bkw_MOBILE
23:01.44DaminHmm..
23:02.06Daminbkw_: I got a SDIO WiFi Card for my Pocket PC. ;)
23:02.17bkw_hehe
23:05.23bkw_soooo
23:05.27bkw_what the hell is up in here
23:06.33DaminNot much..
23:07.35bkw_ok that bitch is going FAST
23:07.40bkw_hope she gets pulled over
23:09.25*** join/#asterisk jsolares (~jsolares@200.12.33.64)
23:09.31ckruetze_bkw: Next time use a 3G mobile, then we could all see what is going one.
23:09.44bkw_its quiet in here
23:10.04algorithmnstill rather active compared to other rooms...
23:10.59*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
23:12.04bkw_haha
23:12.06bkw_3g eh
23:12.25bkw_that sucks here in oklahoma
23:12.35bkw_plus they wanna charge you like 100 bucks for unlimited
23:12.35bkw_with bad coverage
23:12.36bkw_because where i'm at right now doesn't even have sprint coverage
23:12.36terrapenthis weather blows
23:12.38bkw_but t-mobile has it
23:12.41*** join/#asterisk iceyp (~icepick@firewall.unix.co.nz)
23:12.44terrapenrain rain rain
23:13.19SexyKenHey BKW
23:13.29tuxinator_linuxI have t-mobile wireless on my laptop
23:13.32NuggetPOOSE MENIS.
23:13.34iceypanyone know a cheap source for 7960's?
23:13.40Nuggeticeyp: ebay
23:13.58Mociceyp, get polycom phone instead, it a damn better phone
23:14.06iceypmoc url?
23:14.08terrapenbullshit
23:14.13terrapenthe 7960 is a great phone
23:14.18bkw_I love my 7960
23:14.19bkw_great phone
23:14.22Mocterracon, it a great limited phone
23:14.24bkw_Moc just hates them
23:14.29bkw_limited my ass
23:14.29terrapena little bit of a pain to get upgrade to the latest firmware but once you get there, its awesome
23:14.30iceypi got a budgetone
23:14.31bkw_it does its job
23:14.33bkw_and does it well
23:14.33terrapenlimited whatever
23:14.33iceypwant to play with cisco tho
23:14.41Mocit receive/make calls ;)
23:14.45MocPolycom does more ;)
23:14.48SexyKenbkw- I sent Digium an email regarding that custom coding.
23:14.49terrapenthe quality of the plastic is much nicer than polycom
23:14.54terrapenmoc, so does the 7960
23:14.54bkw_thats what a phone is fore
23:15.05bkw_SexyKen, what custom coding?
23:15.20iceypis there any way to test g729 on fbsd 4.10 befofre purchase to see if it workes properly?
23:15.20SexyKenCustom coding for Asterisk we spoke about it a few weeks ago.
23:15.26terrapeniceyp, i suggest eBaying one of each and then selling the one you don't like
23:15.29bkw_it will not work on freebsd
23:15.36terrapeni have an IP500 and a 7960
23:15.36SexyKenThe custom login stuff
23:15.38bkw_SexyKen, I don't recall speaking with you about it
23:15.41terrapenand i want to sell the IP500
23:15.49iceypbkw_  even the intel version?
23:16.11bkw_nope
23:16.12MocOk Polycom aint good for the people who want a simple phone thought
23:16.12bkw_no code for YOU
23:16.13iceypi know it works with fbsd 4.2.9 or something, its on digiums website
23:16.18bkw_:P
23:16.22Mocthe config can be too complex for some people
23:16.44bkw_digium could make a freebsd one
23:16.44bkw_and an OS X one
23:16.44bkw_and a solaris one
23:16.44bkw_whats the point
23:16.50Nuggetthere's no such thing as "freebsd 4.2.9"
23:16.57iceypi know
23:16.59algorithmnsolaris t3 card would be sick
23:16.59iceypmeh
23:17.10terrapenbkw, because its as simple as a recompile most likely and if it makes the customer happy....
23:17.12iceyp5.2.1
23:17.33Nuggetthe holdback on g729 for other operating systems is the code to probe the mac address.
23:17.42ManxPowerSomeone just tried to access my "talking weather report" script (that's not been active in a long time) and he ended up getting my personal extension.  Good thing I didn't answer the phone "Hello, Sweetie.  So, what are you wearing?"
23:18.00ManxPowerThe poor guy was pretty suprized when a human answered the phone.
23:18.04terrapennugget, that's like a 5 minute patch
23:18.10Nuggetso send the patch to kram.  :)
23:18.13bkw_man there for a sec i thought I left my power cord for my laptop behind
23:18.20terrapenif i had the source :P
23:18.30Nuggetsend a reference implementation to him.
23:19.16terrapenwould a C snippet that gets the MAC addy in fbsd and printf()'s it be enough?
23:19.20tzangerinteresting
23:19.34tzangerI wonder if pinging the remote side is worth it or if full frames should just be sent every 15 seconds instead
23:19.38terrapenthe only gnarly part is getting the interfaces
23:19.45terrapenbut thats not even that bad
23:20.00terrapeni wonder how similar Darwin and FreeBSD are
23:20.21bkw_very
23:20.27bkw_cvs-head compiles on it out of the box
23:20.38bkw_I would do a CVS checkout right now and compile it
23:20.40bkw_but that would SUCK
23:20.42Nuggetheh
23:20.42bkw_haha
23:21.10bkw_this is a class 10 device here..
23:21.48tzanger"class 10 device" ?
23:21.54*** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net)
23:21.59bkw_my phone
23:22.01iceyp~like class 10 lazer
23:22.03jbotACTION smooches class 10 lazer on the neck
23:22.03iceyp;P
23:22.07bkw_I wish it was class 12
23:22.14bkw_I could talk and use the internet at the same time
23:22.20bkw_only get 48k down..
23:22.34terrapenughhhhhh
23:22.40terrapenmac address is not in ifaddrs.h
23:22.42terrapenhrmmm
23:23.00ManxPowerUm, the mac address is on the network card.
23:23.29terrapenyes, but im looking for the header file in *BSD that will get me a structure with the mac in it
23:23.58ManxPowerterrapen: Can you look at the source for the bsd equiv of "ifconfig".
23:24.06iceypataa 186's they still any good?
23:24.09iceypata*
23:24.10terrapenyep, that would be a good place to look
23:24.15ManxPowerat least on linux that prints the MAC address.
23:24.24Nuggetthe bsd equivalent of ifconfig is ifconfig.
23:24.26Nuggetjeez.
23:24.33Nuggetyou think linux invented that shit?
23:24.42ManxPowericeyp: Cisco ATA-186 is a good device but 3x more expensive than the SIPura devices and the SIPuras are just as good.
23:24.45bkw_haha
23:25.04ManxPowerNugget: no, but linux does like to call stuff different names.
23:25.07iceypManxPower i'm looking at second hand, $60 for a ata
23:25.09Nuggetfair point.
23:25.31ManxPowericeyp: It's prolly locked to Vonage and you need to have the service provider unlock it, assuming themy will.
23:25.35tzangerCeleron 1.8 just doesn't cut it
23:25.38iceypso who here works for digium; secondly how many people use bsd here?
23:25.55ManxPowericeyp: how many people are on #asterisk-bsd?
23:25.57mamcintySeveral places have the SPA-2000 for about $70
23:26.02iceypManxPower empty
23:26.10ManxPowericeyp: There is your answer.
23:26.11iceyphow many people in #asterisk-linux ?
23:26.20terrapennet/ethernet.h
23:26.22terrapenthere we go
23:26.26mamcintyIt has two FXS. I have one and I works REALLY well.
23:26.43terrapennope
23:26.47terrapeni take that back
23:27.59mamcintybeen using it everyday for a year or so
23:28.31ManxPowerHell, I think even Darwin35 (who thinks BSD is his God) is using Asterisk on Linix.
23:29.20ManxPowerUgh!  My playmate for tonight had to cancel.  Car troubles.
23:29.25*** join/#asterisk Legend (~legend@24.244.142.133)
23:31.09MocI never got used to be on BSD, I had to switch to Linux..
23:31.28bkw_not a war
23:31.30bkw_evil
23:31.49ManxPowerYay!  Backup playmate is coming over.
23:32.21Moclol
23:32.39Mocbkw_  hehe
23:32.44bkw_yo
23:32.51Mocgetting ready ?
23:32.56bkw_for?
23:33.00Mocvon
23:33.13bkw_i'm sitting a stoplight in Okmulgee Oklahoma
23:33.16bkw_headed to tulsa
23:33.20bkw_on my laptop via GPRS
23:33.35Moc;) good, battery status ?
23:34.00Mocbtw watchout for those evil people over there ..
23:34.09bkw_yes
23:34.23filebkw_ wants to touch my muffin
23:34.29bkw_no no file
23:34.33bkw_butter your muffin
23:34.33Mocfile, he on his way ;)
23:34.34bkw_get it right
23:34.36fileuh huh
23:34.36bkw_ok
23:34.55ManxPowerWell butter might facilitate touching.
23:35.10bkw_haha
23:35.24bkw_this is interesting going down the road on IRC
23:35.27bkw_how fucking geeky is that shit
23:35.47Mocbkw_, hey, you would be doing what reading book ?? that geeky !! ;)
23:36.12bkw_haha
23:36.14bkw_i'm checking email.. IRC and AIM
23:36.15Mocbkw_ you got LPC10 over GPrM working ?
23:36.25Mocgprs
23:36.34bkw_nope
23:36.38bkw_got nothing that speaks it really
23:36.47bkw_guess I could loopback to asterisk on my ibook
23:36.55Mocyour stuk with an ibook !!! poor you..
23:37.03Moc;)
23:37.15Mocyea
23:37.15bkw_ibook ROCKS
23:37.37Mocibook are for girls and grand parents ;)
23:37.48bkw_no they aren't
23:37.52Shidohehe
23:38.02bkw_an apple is a real computer
23:38.05bkw_so dont knock it
23:38.14BlackvelYES YES YES
23:38.18Blackvelstrike strike strike
23:38.23bkw_haha
23:38.24filebkw_: btw I upgraded your old laptop to 802.11g :p
23:38.27Blackvelmy bristuff works!
23:38.34bkw_YAY
23:38.36Moc;)
23:38.37bkw_way to go
23:38.48BlackvelManxPower: it was some problem in zaphfc code
23:38.57Blackvelit changed so it works now with overlapdial
23:39.00Mocthose mac/bsd user are soo easy to upset ..
23:39.03Mochehe
23:39.09bkw_no
23:39.12BlackvelI found the coder
23:39.18Blackvelhell, I am so happy now
23:39.20bkw_i'm right and you're wrong.. its just that simple.. no need to argue or get upset
23:39.30bkw_:P
23:39.33terrapenits a gnarly little bit of code to print the MAC address of an ethernet interface in FreeBSD
23:39.35terrapenhrmmmm
23:39.52fileugh
23:39.55filefalling in love with MySQL 5
23:39.58bkw_now call the interface eth0
23:40.01Mocduck
23:40.05tzangerfile: you gotta be kidding me
23:40.18bkw_file use a real server
23:40.19bkw_MSSQL
23:40.21fileno, I'm not
23:40.24MocIm moving slowly from mysql to pgsql... it hard..
23:40.27Blackvelany sipura guys there who would give me a call?
23:40.33tzangerMoc: how come?
23:40.33filebkw_: no! NOOOOOO!
23:40.39bkw_file trust me
23:40.43bkw_don't let the MS name on it fool ya
23:40.46bkw_its really nice
23:41.04Nuggetmysql is ucky.
23:41.05bkw_I should try to ssh
23:41.08MocI never liked pgsql...
23:41.10tzangermysql blows goats
23:41.12tzangerMoc: how come?
23:41.14NuggetI've been working with it all week and I'm about to kill someone.
23:41.29tzangerI'm geniunely curious why people use mysql after getting a taste of a real db
23:41.32filewe use at work tons, works fine
23:41.39fileer it at work
23:41.52Mocpc crashed once, and pgsql db was dead, restored it, then to see close the pc again, and db was still dead..
23:42.00tzangerMoc: interesting
23:42.04NuggetI spend half my time with mysql just fighting off grumpiness over how little it does.
23:42.09MocI never lost nothing with mysql..
23:42.37Moc(I know the recent release pgsql support transaction, so my DB crashing problem shouldnt occur again)
23:42.44Mocalso I like using distro binary
23:42.50tzangerNugget: the thing I *hate* about mysql is that it has no consistency... it will mangle my data and not even tell me
23:42.51Nuggetrecent?  you mean all releases since 1996 or so.
23:43.02tzangerwhen I set up constraints I EXPECT THEM TO BE FOLLOWED
23:43.08Nuggettzanger: yeah, that too.
23:43.15Mocand the pgsql that come with it have this damn problem that I need to assign right PER Table for user, I can't grant all on Database DB to user
23:43.27tzangerMoc: yes you can
23:43.33tzangerI just did it last week or the week before
23:43.42tzangernow I admit
23:43.48Moctzanger, the binary that I use dont ...
23:43.52Blackvelnoone with sipura wants to give me a call?
23:43.54tzanger6.x was slow, but it also shipped with fsync turned on
23:43.58Mocredhat enterprise 3
23:44.14tzanger7.4.x and 8.x prereleases are zippy as hell
23:44.23ManxPowerThe BF must be bored -- he's sending me obscene test messages.
23:44.30ManxPowertest == text
23:44.36bkw_haha
23:44.54Goldenearis it possible to use regexten=john,1234 in iax.conf ?
23:45.41Goldenearso that box "john" and "1234" will right john's endpoint ?
23:45.41bkw_dont think
23:45.47bkw_why in the hell are you doing that?
23:45.50bkw_why not just put one
23:45.54bkw_then usee a goto for the other
23:46.14bkw_the goto will only work if you're really reged
23:46.19bkw_but really why are you using regexten?
23:46.28mamcintyBlackvel: What did you want to know about Sipura?
23:46.31Goldenearsimpler management :)
23:46.33Blackvelnothing
23:46.39bkw_its not really
23:46.42Blackveljust had a problem sipura's calling my * server
23:46.49bkw_Goldenear, it only puts an exten with a noop in a context
23:46.50bkw_thats it
23:46.51Blackvelthat should now be fixed with my latest version upgrade
23:46.52bkw_nothing simple about it
23:47.02bkw_its main use is for dundi
23:47.14mamcintyah, okay
23:47.27mamcintysome of the old firmware versiosn due odd things
23:47.29Blackveldo you have a sipura up and running?
23:47.53bkw_I brought a couple of DVD's too
23:47.55bkw_and my ipod
23:47.56ManxPowerGoldenear: Would you like my advice (it's usually, but not always the right advice)?
23:48.00mamcintyI noticed that when I upgraded to the latest version a few days ago asterisk stopped recognizing my flash hooks
23:48.07GoldenearManxPower: sure
23:48.12ManxPowerbkw_: attending an ordy?
23:48.25mamcintyI have one SPA-2000 that I use everyday here at home
23:48.25bkw_ManxPower, no
23:48.45ManxPowerGoldenear: You want people to be able to dial IAX2/john@happydomain.com or dial IAX2/guest@happydomain.com/john.  Is that correct?
23:49.00Goldenearcorrect
23:49.24ManxPowerGoldenear: Does "happydomain.com" resolve to your asterisk server?
23:49.29mamcintyIts a very powerful device
23:49.44bkw_mamcinty, what device are we speaking of?
23:49.51Goldenear(at least IAX2/guest@happydomain.com/john)
23:50.01GoldenearManxPower: yes
23:50.01mamcintythe Sipura SPA-2000
23:50.32ManxPowerGoldenear: In case #1: Caller claims to be user "john" without a password and not passing an extensions.
23:50.43ManxPowertherefore you need in iax.conf [john]
23:50.47ManxPowertype=user
23:50.52ManxPowerdisallow=all
23:50.57ManxPowerallow= lines for the codecs you want
23:51.04GoldenearI only want case #2
23:51.11bkw_allow= takes multi  codecs now
23:51.13bkw_in the pref order
23:51.17bkw_codec,codec2,codec5
23:51.19ManxPowercontext= to point the call to exten => s in a context in extensions.conf
23:51.30ManxPowerbkw_: only in CVS-HEAD
23:51.36bkw_nope
23:51.38bkw_in stable too
23:51.43Nuggetspiffy
23:51.57bkw_drumkilla, backported the codec stuffs
23:52.02ManxPowerGoldenear: in Case #2 you have a user claiming to be "guest" connecting and asking for the extension numbered "john"
23:52.03bkw_because it really did fix a problem
23:52.15GoldenearManxPower: yes that's what I want
23:52.34bkw_regexten isn't needed btw
23:52.43bkw_don't think twice about it.. you don't need it
23:52.51ManxPowerGoldenear: So you need a [guest] in extensions.conf with type=user and disallow=all and allow= for the codecs you want and a context= line to point the call to a context in extensions.conf containing exten => john,1,whatever
23:53.21jessterIs there a diff between an INVITE 1 and INVITE 101 ?
23:53.33ManxPowerI don't know if you need an empty secret= line or just need not to have a secret=something line.
23:53.34jesster..on a CSeq
23:53.41fileto all those out there, the Hitachi WIP-5000 is an awesome wifi voip phone and works great with asterisk
23:53.47ManxPowerGoldenear: do you understand what I'm explaining?
23:53.55GoldenearManxPower: yep :)
23:54.27ManxPowerGoldenear: always remember that extensions don't have to be all numbers.  It's just kind of pointless to expect someone to dial non-numbers from phones.
23:54.32fileI in no way work for the company that made it 'nor did they give me a phone
23:56.07GoldenearManxPower: I want each user to have both a name and phone number as extension... so the exten => john,1,Goto(1234|) seems to be what I need
23:56.19bkw_ding ding ding
23:56.20bkw_yep
23:56.42*** part/#asterisk Blackvel (~blackvel@dsl-082-083-173-045.arcor-ip.net)
23:56.58filebkw_: so close to VON!
23:56.59ManxPowerGoldenear: Yup.
23:57.01GoldenearBut it's a little fastidious to configure that for each user ... I may use a macro or something for this ...
23:57.14jessterWe'll be at VON yippie!
23:57.17ManxPowerGoldenear: I'm more of a traditionalist.  People should dial numbers, not names.
23:57.26bkw_i'll be there
23:57.26bkw_w00t
23:57.33fileand so will I!
23:57.33jessterwerd bkw!
23:57.39ManxPowerWhen is VON Europe?
23:57.58GoldenearManxPower: I agree with you ... but in my case I also need names :)
23:57.59filebkw_: so are you going to come meet me at the airport?
23:58.09bkw_yes
23:58.09ManxPowerI may stop being a recluse and attend.
23:58.15bkw_so is drumkilla
23:58.16bkw_and twisted
23:58.26bkw_we are gonna really have fun with it
23:58.31filethis can't go well
23:58.33bkw_drumkilla, gets in 35 min after me
23:58.42bkw_drumkilla, says we need to attack you
23:58.49fileblasphemy! no attacking the poor file
23:59.03filewhat are we doing Monday? do we know at all?!?
23:59.11bkw_can we fsck him?
23:59.22filewe need to corrupt drumkilla
23:59.28bkw_haha
23:59.28filebring him to the dark side
23:59.30bkw_whats this we stuff?
23:59.36bkw_you got a mouse in your pocket?
23:59.38fileI'll help!

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