irclog2html for #asterisk on 20050712

00:00.25*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
00:04.03SpaceBasslol
00:04.03SpaceBasshttp://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5788092558&rd=1
00:05.39_DAWIm bidding now!!!
00:05.39SarahEmmthat link has been pasted in here like 8 times :P
00:06.06SpaceBassyeah, sounds great right? good luck getting new firmware!
00:06.12_DAW:)
00:06.20opus__is anyone having problems dialing out with bubbahtel?? i can't make a call
00:06.35opus__:)
00:06.38SpaceBasslol
00:06.48SarahEmmopus__: what error message are you receiving?
00:06.56_DAWtoo much tarnish
00:09.33*** join/#asterisk doughecka (~Miranda@doughecka.user)
00:09.40JinxyJul 11 20:09:02 NOTICE[22827]: chan_sip.c:7733 handle_request: Registration from 'John <sip:200@192.168.0.254>' failed for '192.168.0.2'
00:09.52Jinxybut, I've got all the information right in sip.conf & the softphone
00:14.02NuggetI'll bet you don't.
00:16.18Jinxywell, I just uncommented the xlite config in sip.conf, and put the info from there in xlite softphone
00:16.31Jinxyexecuted reload on * console
00:16.44JinxyI keep getting the registration error, but it doesn't state why it is failing
00:16.47Jinxyusername/etc
00:17.39gravemindergh, there are two mpg123 instances running
00:17.44gravemindone spawned the other apparently
00:19.54Jinxyah ha, forgot a critical thing
00:20.04Jinxy[] has to be same as extension name
00:20.10Malthus:)
00:20.29Malthusno
00:20.38Jinxyno? why did it work then?
00:20.43Malthusmagic
00:20.57Jinxyplease be serious, I am trying to learn
00:21.01MalthusI have xlite/asterisk set here
00:21.02Jinxythe last thing I want is a detour
00:21.20Malthusthe username is not the same extension
00:21.47MalthusI have a numeric extension but alphanumeric username
00:22.03Malthus[name] is the username by the way
00:22.56Nuggetyou two are both saying the same thing.
00:22.58JinxyMalthus, ah ha
00:23.05JinxyI was using extension as the username
00:23.25Jinxyso by naming xlite1 to extension, which I had defined as the username in xlite, I fixed my problem
00:23.25Nuggetthere is no direct relationship between a sip entry and an extension.
00:23.52Jinxyok, so now I am logged in, what can I do to test if things are working?
00:24.05Jinxyany preconfigured automated extension that has some sexy lady on the other end?
00:24.11Malthus800
00:24.12JinxyI've not specified any external things yet
00:24.12Malthusoops
00:24.14Malthus600
00:24.26Jinxynothing happens
00:24.42Malthusoh
00:24.52Malthusthats part of the demo
00:24.59Malthuslook at your extensions.conf
00:25.07Malthussearch for demo
00:25.18Malthusgood starting point
00:25.22Jinxyya, I am going through the file right now, was hoping some of those things were already running
00:25.45Malthusis the demo in there?
00:25.52Malthussearch for [demo]
00:26.02Jinxydone
00:26.08Malthusnone?
00:26.48Jinxynada
00:26.55Jinxyworking
00:27.19gravemindwould a system using sip phones and asterisk that works somewhat like a traditional PBX where each phone has an "extension" line, voicemail line, conference line, etc? if so, how do phones indicate to asterisk which line is being used...
00:27.22Jinxythat was just an extension I tried
00:27.37gravemindmaybe the extension number passed along when dialing?
00:28.07Malthusgravemind: each SIP phone registers with asterisk
00:28.12Jinxyholy jeebus
00:28.22gravemindyes, but do sip phones come with multiple line buttons?
00:28.28Jinxyyou have to press the dial button on xlite 2x
00:28.36Malthushaha
00:28.56gravemindlike if you want to get your voicemail, you can press the voicemail button and pick up the receiver, and it'll dial that out
00:29.10Drukenthat's just a speed dial
00:29.14Malthusyeah
00:29.33gravemindwhat about if your phone is rang both by external lines and an internal extension, distinguishing between those etc
00:29.35Malthusyou configure the phone to call a specific number for voicemail etc.
00:29.53gravemindyou know, you've seen phones that have multiple lines calls can come in on, and pick which one to pick up right on the phone
00:29.57Drukengravemind: that's something you have to do in your dialplan
00:30.15Malthusgravemind: you're typing pretty fast btw
00:30.20gravemindi do that :P
00:30.24*** join/#asterisk HaHaOok (~norman@60-240-240-66.tpgi.com.au)
00:30.40gravemindwould that kind of thing be implemented as a speed-dial for the *8<exten> thing?
00:31.16Malthusbasically asterisk can do all that and more
00:31.19gravemindsorry, im just trying to grapple with how asterisk would replace a traditional PBX system
00:31.25Drukenyour thinking of voip phones like network phones, and they don't work the same way
00:31.27gravemindparticularly the "lines" coming into each phone
00:31.36gravemindhrm.
00:31.59*** join/#asterisk mansing (~mansing@250-116.customer.cloud9.net)
00:32.01harryvvgravemind you can interface asterisk to databases and show grapically what it can do. But whats your intentions?
00:32.29gravemindi could see how it _could_ be done, but i doubt it is
00:32.33*** join/#asterisk Cybertoy (~Cybertoy@ool-457852fa.dyn.optonline.net)
00:32.55*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
00:33.06Malthusgravemind: it is
00:33.16Malthuseg with xlite
00:33.19bdunnEVERYTHING IS WORKING!  Wow.  Except one thing.  Music on Hold is completely silent.  Any ideas?
00:33.27gravemindsomething like when a call comes in on line X, you ring the phone with /X as the extension, and it flashes the light for line X
00:33.36Malthusmpg123 problems
00:33.38gravemindbdunn: have you enabled a source in musiconhold.conf?
00:33.52gravemindMalthus: is xlite a windows softphone?
00:33.56Malthusyes
00:34.02gravemindoh, too bad then
00:34.05bdunngravemind - Ahhh... yes, but wrong path I think.  I'll try that.
00:34.06gravemindim stuck with kiax for now :)
00:34.16Malthusand the sip hardphones are very similar
00:34.22Malthuskiax is nice
00:34.25gravemindbdunn: also note that it takes a full restart for musiconhold changes
00:34.33Malthusif kde is your thing
00:34.36MalthusI use iaxcomm
00:34.39gravemindits not, but it works.
00:34.43Malthusyes
00:34.46gravemindand looks good
00:34.53Malthusand it supports multiple lines
00:34.56gravemindonly annoying thing is the dialpad being on a seperate tab
00:35.24Malthusso you can accept and make multiple calls at the same time
00:35.34Malthuscallerid on each incoming call
00:35.38gravemindright
00:35.41Malthusoh
00:35.50Malthusthere is an xlite for linux
00:36.00gravemind... i don't think i want it though
00:36.02gravemindsounds like it'd suck.
00:36.14iCEBrkrDoes EAGI even work?
00:36.16Malthusits as good as the windows version
00:36.27Malthusbut I stick with iaxcomm
00:36.46bdunnOKay... music on hold... can I pick any MP3 for that?
00:37.01Malthusonly reason I don't use kiax is because it would be out of place with my gone desktop
00:37.07gravemindbdunn: yes, but unless you have the real mpg123(not mpg321 as many distros have) it wont downsample corretly
00:37.11gravemindso you'll get a bunch of static
00:37.22Drukenbdunn: you don't use a mp3, but a directory containing mp3s
00:37.26gravemindMalthus: purist :P
00:37.29gravemindi use fluxbox, nothing matches
00:37.33bdunngravemind- I built this from cvs.  Should it have the right stuff?
00:37.41gravemindbdunn: mpg123 isnt part of asterisk
00:37.58Drukengravemind: actually it is on the cvs
00:37.59bdunngravemind - Ahh.. okay.. Could also be the problem then.  :-)
00:38.13Malthuslook at the wiki
00:38.17gravemindDruken: really? its nto even proper open source...
00:38.31Malthushttp://voip-info.org
00:38.34Drukengravemind: pretty sure it is....
00:38.45gravemindDruken: its open source, but its not /proper/, licensing issues to be had
00:38.47Malthusthere is a lot of info on making mp3s work with asterisk
00:39.13gravemindi chose to reencode MOH at 8khz ahead of time to save cpu
00:39.27Malthus:)
00:39.51MalthusI chose to hum along in my head
00:40.03Malthussaved me configuration and even more CPU
00:41.08gravemindMalthus: long story short, im just wondering if hardware phones come with "line buttons" that light up when a given extension is ringing you
00:41.08gravemindlike one lights up for an internal call, another for external line N, so on
00:41.08gravemindand you press that button to answer that specific line
00:41.22gravemindbut im guessing sip phones dont normally have that functionality so asterisk cant implement it
00:41.39mansinggravemind: hardphones -- look at the Sipura
00:42.00Malthussome do
00:42.07mansingalso Grandstream has a multiple SIP appearance phone
00:42.18Malthusthe ones at work do
00:42.29gravemindi wasnt thinking an entire seperate registration is needed
00:42.35Malthusbut most of the better ones include a proper LCD
00:42.46gravemindjust a way that asterisk could send a DTMF code with the ring to indicate which line
00:42.52Malthusso you can get more info that just internal/external
00:42.56gravemindand in turn, dial out a DTMF to pick a line
00:43.22Jinxyhow can I tell if I am connected to an IAX2 trunk?
00:43.31Malthusyou can configure asterisk to do what you want with the callerid it sends
00:43.47gravemindMalthus: say you have a receptionist who has access to 10 incoming lines. if two lines are ringing at once, and you want to pick up a specific one, how would you go about doing that?
00:44.21mansingwith a multiple appearance phone
00:44.30Malthusher phone would show both incoming
00:44.36mansingeach button is a different extension
00:44.42mansingasterisk manages the buttons
00:44.43Malthusbutton beside each to answer
00:44.50mansingyes
00:44.50bdunnCan someone please tell me where mphmp3 is supposed to go in the file system
00:44.55gravemindfair enough
00:45.21mansingI did that with a Sipura
00:45.22gravemindout of curiosity, could that be done with a single extension, assuming its a pure sip phone?
00:45.42mansingon incoming calls, asterisk checks each of the four exts on the sipura.
00:45.47mansingeach ext == one button
00:45.50Drukengravemind: you should be able to use the SIP stuff for that....
00:46.17gravemindhm
00:46.18gravemindokay, thanks.
00:47.31DrukenSetVar(_ALERT_INFO=
00:47.41Drukentake a look at that... might help ya out
00:48.11*** join/#asterisk cabronsito (~awefawe@201.240.45.194)
00:48.12Kattymew
00:50.03*** join/#asterisk justinks (~kennedy10@machine76.Level3.com)
00:52.57*** join/#asterisk int19h (~Miranda@PC00019127.eng.monash.edu.au)
01:00.33Jinxydo I need to do anything special if my * box is behind a linux firewall so my AIX2 trunk works?
01:00.48Jinxy*IAX2 trunk rather
01:02.28*** join/#asterisk Mistil (Twister@24-179-94-064.dhcp.chtn.wv.charter.com)
01:03.01*** join/#asterisk w0w0 (~apardo@159.Red-83-41-3.pooles.rima-tde.net)
01:03.03bdunnOKAY... Music on Hold is working... BUT VERY VERY VERY VERY quietly.  Any suggestions?
01:03.44harryvvthere is a loud setting
01:06.54bdunnWhen I set it to loud I don't hear anything at all.  :-(
01:09.02Malthusmaybe it caused you to go deaf
01:09.08bdunnAhh... music!
01:09.12iCEBrkrBlah.
01:09.19Malthus:)
01:09.29bdunnI had to go to default.... now it's working at the right volume.  No idea what I did.
01:09.33iCEBrkrEAGI() doesn't appear to be working correctly.
01:09.40bdunnThat's everything... it's ALL WORKING!  Holy crap!
01:09.46iCEBrkrbdunn: hehe
01:10.13darwin35E911 can go to hell
01:10.20iCEBrkrdarwin35: lol
01:11.04darwin35I want to shoot the people who voted for this bill and then move up the line from there
01:11.08darwin35gfrrrrrr
01:11.33Nuggetgot the ill communication
01:11.36darwin35no easy answer
01:11.54darwin35and down to less then 90days to implament it
01:12.36harryvvand the phone companies wont give up the info ?
01:12.39justinkdarwin35: whats up with the E911 grief? Most ppl want their phones, even if VoIP, to function like a phone. As another phone geek put it "if it walks like a dog and wags like a dog, then ppl expect it to be a dog."
01:12.40sivanadarwin35: we're working on an affordable solution
01:13.13darwin35open ss7 still has alot of work
01:13.41darwin35and it is going to put alot of new startups down and existing voip providers down
01:13.43Mistilany1 else have nufone and have an 800 number that isnt working?
01:13.47harryvvhow is vonage tackeing the e911 problem?
01:13.58darwin35they wont say
01:14.06Jinxyso guys, do I need to forward any ports to get an IAX2 trunk to work if my * box is behind a linux firewall?
01:14.24NuggetI think you're confusing "firewall" and "nat"
01:14.32harryvve911 cannot be enforced..if thay do thay need to hire alot more field agents.
01:14.50JinxyNugget, the * box actually has internet ip, but it is behind a linux firewall
01:15.00sivanaharryvv: here in Canada, they are enforcing it through the ILECs and CLECs
01:15.06Jinxywhich filters most of the unrelated traffic coming to it, so I need to know if I should allow in a certain port
01:15.07Nugget"port forward" is a nat concept, though.
01:15.12iCEBrkrDid 911 always have ANI?
01:15.23sivanaharryvv: the CLEC providing my PRI has threaten to shut it down if I don't comply
01:15.32JinxyNugget, I understand, but I need to know if I need to allow in any unrelated connection?
01:15.47harryvvsilvana, wow really.
01:15.52Nuggethttp://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
01:16.04darwin35they can inforce by making it the t-1 e-1 dsl/cable providers right to cut you off if you have anything running on a voip port
01:16.09sivanaharryvv: ya, they're forcing the CLEC/ILECs to enforce their networks
01:16.21Nuggetthird hit on google for "asterisk firewall iax", for what it's worth
01:16.47harryvvI think thats rediculios. I use sixtel to call into the states and I dont need no fricken 911 from them.
01:17.03sivanathat's different though
01:17.11harryvvWell I hope so!
01:17.31darwin35800/866/877 providers dont have to worry
01:17.41harryvvI guess the only way around it is be a country to county voip carrier. Or sell voip to bussiness that have pstn.
01:17.46darwin35just did/pstn providers
01:17.56sivanaonly providers that serve local residential customers
01:17.57sivanaya
01:18.32harryvvseems to me residential is the most profitable because I suspect thay wont use as much bandwith vs say a call center
01:18.33darwin35they would have to have 1 line that you could map to thier system for 911
01:19.01*** join/#asterisk NatRH (~Nat@dargo.trilug.org)
01:19.03harryvvyea, and splice off that line to a red phone in a red box on the wall in the event the pbx goes down.
01:19.12SarahEmmredbox? ;)
01:19.14harryvvits extra security.
01:19.19darwin35have them have a sipura or something mapping the call to thier pbx service
01:19.41*** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
01:19.50darwin35that would work
01:20.20harryvvhave the door triger a ringer microswitch when its opened.. and ringer switch stays on when the door is closed untill its reset. :)
01:20.44darwin35I want a red phone
01:20.53darwin35with a flashing blue light on it
01:20.57harryvvThat way, if thay do shut it by accident and 911 calls back, it will still ring.
01:21.07darwin35yes
01:21.20harryvvits just good insuance.
01:21.25darwin35I think it should be a pbx box at every company
01:21.26harryvvpeace of mind
01:21.27harryvv:)
01:22.02darwin35well we where looking into being a virtual pbx company
01:22.09harryvvOne federal company I have worked in and i suspect its req by the fire department in all buildings is a red phone box on every floor.
01:22.19*** join/#asterisk santiago (~santiago@63.245.86.188)
01:22.20darwin35for small/medium companies
01:22.49darwin35that would be the answer
01:23.01*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
01:23.06darwin35but if the pbx goes down then they cant take calls anyways
01:23.07bdunnWhat is the command to add a voice mail box?
01:23.23darwin35ee voicemail.conf
01:23.25harryvvlook at the bottom of voicemai..conf
01:23.26SarahEmmvi voicemail.conf
01:23.28darwin35addd the box
01:23.38darwin35then reload asterisk
01:23.43harryvvexamples at the bottom of voicemail.conf
01:23.53harryvvif you have samples installed
01:23.56darwin35the first time someone calls the exten it will creat the box and prompts
01:25.01darwin35but with the red phones would they dial 911 only or have a full keypad
01:25.10bdunnSeems like there was a command line way to create a voice mailbox.
01:25.17*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
01:26.02ManxPowerAsterisk will now automagically create the voicemailbox when the first message is lefst
01:26.14darwin35you could  write a bash script to add a new vm boox to the end of the file
01:26.27bdunnWhen I try to access the voicemail, I get silence, so I can't leave a message.
01:26.38*** join/#asterisk hans (fugalh@falcon.fugal.net)
01:27.26hanssingle-port fxs ATA - recommendations and antirecommendations?
01:28.04ManxPowerhans: SIPura, always SIPura
01:29.08hanswhy?
01:30.17*** join/#asterisk emp (~emp@70.57.239.37)
01:30.21ManxPowerThey make good stuff for a decent price.
01:30.46ManxPowerTheir hardphone is not all that great, but good for the price if you accept it's limitations
01:31.50Qwell<rant>
01:32.12QwellYou know what I love?  When you're told on Monday night at 6:30, that the entire network is being upgraded on Friday night.
01:32.30Qwelland, that you'll need to come in at midnight or so, and wait for them to finish, so you can fix shit and test it
01:32.44*** join/#asterisk jm_dot_com (~jmdotcom@adsl-67-119-233-167.dsl.sndg02.pacbell.net)
01:32.47Qwell...I'm leaving my rant tag open
01:33.01empis it possible to have an asterisk system use the call transfer feature on a my POT ?
01:33.08Jinxyinteresting, I can make a call through voipjet if I use iaxcomm utility on my windows box, but it won't work through asterisk
01:33.22QwellJinxy: Then your setup is wrong in asterisk
01:33.59JerJermmm POT
01:34.08hansManxPower: thanks for the recommendation. any chance on finding one locally (normal size city USA) or am I stuck with online shipping costs?
01:34.29JinxyQwell, I've looked everywhere
01:34.35QwellJinxy: Well, what happens?
01:34.44JinxyI dial, I get an error
01:34.57Qwell"an error".  yeah, THATS helpful
01:35.02JinxyI am copying it
01:35.03Jinxy<PROTECTED>
01:35.03Jinxy<PROTECTED>
01:35.25JinxyI've got the setup as suggested by voip
01:35.28Jinxy*voipjet
01:35.29SarahEmmlol jerjer
01:36.30jm_dot_comhello guys i have basic telco questions trying to find the best solution for my office setup. Ive, been activly useing asterisk for about year half and this is my first time on a irc list.
01:38.14*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
01:38.43*** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
01:39.02SarahEmmwell it's not a list, but ask away jm_dot_com
01:40.14*** join/#asterisk Newbie___ (me@211.24.146.11)
01:40.43jm_dot_comI an asterisk box colocated in our dc.  And 2 remote offices with dsl so far this has worked out great, however useing SIP providers for incoming lines however with our call volume incressing have not been happy with the way the sound qualy has been
01:41.32jm_dot_comso looking at possibly having a dids or pri coming directly into office and haveing asterisk plug into that. but wanted to konw if this would be most cost effective solution. also trying to stay within a montly budget. and
01:42.14*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
01:42.41jm_dot_comi like how now we can have unlimite calls come in and it just opens another line which is why i dont want to go with regualr pots lines as the cost adds up just to get a new number.
01:43.13SarahEmmwell, if you want to leave SIP/IAX providers, you're moving to a PRI, so.. i'm not sure what other options you have :)
01:43.58jm_dot_comWill this help to incress my voice clarity this is the realy concern, of ours as clients are begging to complain about sound quality.
01:44.22SarahEmmwell, it could be an issue of quality between the office and your * box too.
01:45.40*** join/#asterisk iq (~iq@70-59-160-156.omah.qwest.net)
01:45.40jm_dot_comwould useing hardware sip phones over the soft phones make any diffrence. our office to dc is usualy in the 30ms and from * box to SIP providers around 80 as it goes back east we are in west coast.
01:46.06iqhi
01:46.20SarahEmmit might, i'm not sure jm_dot_com :)
01:46.29SarahEmmi'm likely one of the worst people in the channel to talk to about sound quality..
01:46.36SarahEmmso someone else may be better to help you on this :)
01:46.47KattySarahEmm: i'm the worst, me thinks
01:47.00SarahEmmKatty: oh?
01:47.22KattySarahEmm: i don't know hardly anything
01:47.28SarahEmmKatty: oh.
01:47.30jm_dot_comtxs.  do you happen to know aprox how much pri did lines run. are they cheaper if went with someone like xo vs a local bell.
01:47.35SarahEmmKatty: i know stuff, i just can't hear so good :)
01:47.44SarahEmmnot sure jm_dot_com
01:47.44Twisterbout how much of an internet connection you all think id need for say..a max of 8-10 outgoing and 2-3 internal calls at once, also what would be the best codec that would give me a good balance of quality/low bandwidth use
01:47.52KattySarahEmm: but i did set up samba and let a co-worker have access to the master.cvs
01:48.04Kattyand then i figured out how to do a ssh tunnel
01:48.07Kattyso i guess i'm not all that dumb
01:48.13SarahEmm*nodnods* you're not :)
01:48.21Kattyyay :>
01:49.04Twistermeow! :)
01:49.16SarahEmmmew!
01:49.26*** part/#asterisk mansing (~mansing@250-116.customer.cloud9.net)
01:51.25Drukenmoo...
01:52.07*** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com)
01:52.08Twisterbaaaaaaa!
01:52.31Drukendamn sheep!
01:52.36JerJerwoof
01:53.05Kattymew
01:53.19Drukenwho has a phone game in the dialplan ?
01:54.12SpaceBassgame?
01:54.30SpaceBassi like to send people into a never ending IVR... does that count?
01:55.11Kattylike blackjack
01:55.15Kattypress one for another card
01:55.17Kattypress 2 to stay
01:55.26Kattyyour score is 20, my score is 18
01:55.27Kattyyou win
01:55.32Kattypress 3 to play again
01:55.34Jinxyany iax cli commands?
01:56.01JerJergotta have a really bad impresssion of Sean Conerry
01:56.05Twisterhelp iax2
01:56.11Drukennah.... a game...
01:56.36*** part/#asterisk hans (fugalh@falcon.fugal.net)
01:56.38int19hhas anyone heard of any games where callers can compete for queue position?
01:56.53*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
01:57.14bdunnOutgoing calls show caller ID with a new of Wireless Caller.  Any fix to that?
01:57.27SarahEmmint19h: ROFL!!
01:57.30SarahEmmint19h: that would rock!
01:57.47Twisterhmmm
01:57.50Twister*gears turn*
01:57.51int19hSarahEmm: yea, I thought so :)
01:58.47Jinxydoes voipjet use aix2 or aix?
01:59.01Jinxyaccording to their config, it seems like it is aix2
01:59.06SarahEmmJinxy: it's IAX.
01:59.07Jinxybut damn, I can't get it to work ugh
01:59.09SarahEmmAIX is an operating system.
01:59.12SarahEmma nice one.
01:59.15SarahEmmbut still. IAX.
01:59.25JinxySarahEmm, not iax2?
01:59.34SarahEmmerr
01:59.39SarahEmmi meant the acronym is IAX, not AIX
01:59.43SarahEmmi have no idea what the answer to your q is
01:59.47Jinxyya, I keep confusing it
01:59.54Twisteraix...mmmm..drooooool
02:00.17SarahEmmthen again, i'm Weird ;)
02:00.53int19hsheesh.... you people....
02:00.57int19hAIX is a designer label
02:01.13Jinxynothing wrong with my voipjet account, works just fine if I use iaxcomm windoze program
02:01.13SpaceBassthats A|X (pipe not I)
02:01.17int19hdamn
02:01.19int19hgood point
02:01.22SpaceBass:)
02:01.31int19h:)
02:02.13darwin35Mr President please pick up the blue phone. Mr President please pick up the blue phone
02:02.40int19hI was major into asterisk about 6 months ago, but my interest has declined substantially in recent months... tell me... is there an app for modifying the audio stream within calls yet?
02:02.42NuggetAlright, give me Hamm on five, hold the Mayo.
02:02.52darwin35SpaceBass pls pick up the white curtisy phone
02:02.55int19hi.e. filter the audio? to, lets say, cause a reverb effect?
02:03.15SpaceBassdarwin35:  give me the mayo on white and hold the red phone
02:03.46SpaceBassdamn Nugget  beat me to it... couldnt quite remember it
02:04.13darwin35all those attending cluecon pls pickup the yello com phone
02:04.18*** join/#asterisk Blake0PS (~blake@blakeops.com)
02:04.23*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
02:04.40NuggetSpaceBass, don't start with that red zone/white shit again.  We all know what this is really about.
02:04.56SpaceBassi was telling someone about that scene just last night
02:04.59darwin35no thats black and white
02:05.08darwin35red and white are phones
02:05.15SpaceBasslooks like I picked the wrong day to quit sniffing glue
02:05.24*** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
02:05.25*** join/#asterisk florinm (~florin@host-84-9-255-27.bulldogdsl.com)
02:05.29florinmhi guys
02:05.40SpaceBassis there a way to make an ATA call something as soonn as it goes off hook? if so I may have to get an old school red phone
02:05.41SarahEmmdarwin35: i'd love to, but.. mew.
02:05.41florinmanyone knows what is with goldenpbx ?
02:06.07darwin35when my tdm 40b gets here I plan to have a red/blue/yello/green line setup
02:06.25florinm??
02:06.49darwin35yes its called autodial and its in the web config
02:07.04litage`sip show peers` shows a registered device (softphone), but the softphone was shutdown 2 minutes ago. how can i fix this?
02:07.05SpaceBassdarwin35:  what web config?
02:07.21darwin35ata's have a web interface
02:07.28*** join/#asterisk Qorky (~Pooa@202.173.160.26)
02:07.33Drukenis there a good web based config for asterisk yet? possibly a free one?
02:07.37darwin35if you login to it you can configure it
02:07.45darwin35amp
02:07.47SpaceBassDruken: amp
02:07.50Qorkycan anyone help me please. when I call my meetme conference. Asterisk crashes out.
02:07.54darwin35is about the only one I have tested
02:07.54Qorkyanyone else seenthis ?
02:08.03Drukenurl ?
02:08.09darwin35you dont have a timing device
02:08.24_DAWQorky - you have zaptel card or ztdummy?
02:08.28Qorkyhmm but i do
02:08.31darwin35you7 need to compine zaptel and ztdummy
02:08.40Qorkyi have a z100p in the box
02:08.47darwin35did you run ztcfg on it
02:08.55QorkyChannel map:
02:08.55QorkyChannel 01: FXS Kewlstart (Default) (Slaves: 01)
02:08.55Qorky1 channels configured.
02:09.06darwin35ok then stop asterisk
02:09.09Qorkyi ment x100p
02:09.11darwin35type ztcfg
02:09.20darwin35and then start asterisk
02:09.22Jinxyworky worky
02:09.28darwin35and the timing should work
02:09.35Jinxystupid firewall had ip for voipjet off
02:09.36Jinxy:(
02:09.44darwin35and if not read /var/log/asterisk
02:11.29Qorkythe messages file
02:11.29Qorky?
02:11.35Qorkyit doesnt have much in there.
02:11.41Blake0PShi, if i start in.tftpd from inetd and specify -v (verbose logging) where does it log to?
02:11.52QwellBlake0PS: probably /var/log/messages
02:12.59litagewhat can i do about asterisk thinking a softphone is connected/registered when the softphone isn't running?
02:13.14*** join/#asterisk Twister (Twister@24-179-94-064.dhcp.chtn.wv.charter.com)
02:13.40darwin35reset the time out
02:13.46darwin35in the conf file
02:14.07darwin35and make sure the client logs out not just turns the client off
02:14.29litageshouldn't turning the client off (ie: shutting down the software) log off automatically?
02:14.40litagedarwin35: there isn't any way to manually log a client off from asterisk?
02:15.18darwin35not that I know of other then the time out will clear the db
02:15.31darwin35if it fails to reg in the time required
02:15.37litagethanks darwin35
02:16.29*** part/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
02:16.42darwin35np
02:16.44JinxyI need some suggestions from the pros here
02:16.51Kattytake a nap
02:16.54empis a X100P a good way to get started with an * setup?  Where are good places to buy?
02:16.56Kattyyou'll feel better in the morning
02:17.07Jinxyemp, fleabay
02:17.11darwin35x100p are no longer made
02:17.23darwin35get a tdm 400
02:17.33Jinxyemp, you can get a clone, and a small mod to the driver can get you started
02:17.34darwin35it has 1 fxo 1 fxs
02:17.37Jinxybut, they have echo issues
02:17.55Jinxydarwin35, you have to buy modules for it
02:18.02Kattywhat should i do now?
02:18.09Katty2 hours till bed time and nothing to do
02:18.13SpaceBassyou can get an x100p (clone) from ebay for like $10
02:18.15Qorkyhow can i check the timing ?
02:18.23Qorkyis working from the x100p ?
02:18.29darwin35it comes with 2 modules if you buy the dev kit for 175
02:18.33Jinxyanyway, I've got 4 POTS lines and about 8~12 analog phones
02:18.43JinxyI don't want to replace all the phones with ip ones, gonna cost me a lot
02:18.50Jinxyand I also want to give each phone its own extension
02:18.53darwin35then thet the 12 line card thats out there
02:18.54Jinxywhat is my best option?
02:19.13darwin353 tdm40b
02:19.22darwin35for about 900
02:19.26empdarwin35, where is a good place to buy online?
02:19.49Jinxydarwin35, with 4 FXS modules each?
02:19.54darwin35http://www.telephonyware.com/telephonyware/scan/fi=products/st=db/co=yes/sf=category/se=Analog CTI Cards/op=eq/tf=category,description.html?id=kTSeGr8d
02:20.03darwin35there is all the cards
02:20.09darwin35and pricing
02:20.47empnice, thanks
02:21.32Blake0PSHow do I get the in.tftpd server to log? in.tftpd -vvvvvvvvv /tftpboot/
02:21.36Blake0PSthat doesn't work
02:21.43darwin35the voicetronix card with 12 fxo/fxs ports is nice you jumper it for fxo or fxs on each port
02:21.48pfnuse tcpdump or tethereal
02:22.12Blake0PSwhich one is easier to use and view the output of
02:23.21justinkhas any company made a 12port ATA using SIP? An external solution for fxs ports?
02:23.45iCEBrkrBlake0PS: I think it logs to /var/log/messages
02:24.08QwellDidn't I say that earlier?
02:24.41iCEBrkr(10:11)< Qwell> Blake0PS: probably /var/log/messages
02:24.42iCEBrkrYup.
02:24.44iCEBrkr:D
02:25.17iCEBrkrEAGI just isn't ment to be.
02:25.20*** join/#asterisk ToR\L (toril@cpe-24-58-23-240.twcny.res.rr.com)
02:25.26ToR\Lhi folks
02:25.32JerJermoo
02:25.43darwin35not that I have seen
02:26.06ToR\LI have an asterisk server (1.0.9) for voip at home, using a sip ata... how do I stop it from sending mwi info?
02:26.26QwellToR\L: remove the mailbox= from sip.conf?
02:26.29ToR\LI have these cordless phones that say "msg waiting off" every few seconds
02:26.31Qwellor whatever that option is
02:26.32ToR\LI commented it out
02:26.40Qwelland reloaded?
02:26.41ToR\Lmr qwell
02:26.42ToR\Lyup
02:26.49QwellThen your phone sucks. :D
02:26.52iCEBrkrlol
02:26.55ToR\Llol
02:26.57ToR\Ldoh
02:27.00iCEBrkrToR\L: listen to your messages. That turns it off. :)
02:27.01ToR\Lmy 5 phones suck!
02:27.05ToR\Lno messages
02:27.10iCEBrkroh. that problem
02:27.11iCEBrkrhaha
02:27.13litagevoicetronix says this about their openswitch12:   "Standard-sized PCI form factor that easily fits into any regular PC chassis"  how? the card looks like it's 2 feet long
02:27.13iCEBrkrhrrm.
02:27.16darwin35the netweb401 rock
02:27.18iCEBrkrI forget how I fixed that
02:27.27Qwelllitage: most machines are 2 feet long
02:27.27ToR\LI set checkmwi=3600
02:27.34ToR\Lcommented out the mailbox line
02:27.38ToR\Lstill it insists
02:28.00ToR\Lvonage forums insist its in the ata
02:28.08Malthusmaybe that's why it says 'messae waiting off'
02:28.08iCEBrkrhaha qwell is the master of the obvious
02:28.09Qwelllitage: put it in one below all of the HDs...
02:28.10litageerrr, giraffe would be better
02:28.11ToR\LI have a linksys rt31p2
02:28.17Qwelllitage: You must've never seen old ISA cards man...
02:28.22Qwellthose things were fucking SCARY
02:28.31QwelliCEBrkr: I try. :)
02:28.55QwelliCEBrkr: remember those old video boards, with the ide controllers builtin?
02:29.03iCEBrkrHrrm, you could build a ceiling fan with those old ISA cards
02:29.14iCEBrkrVLB baby!!
02:29.22Qwellvlb?
02:29.31Qwellvideo logic board?
02:29.32Malthusvesa local bus
02:29.32iCEBrkrYea, the full length Vesa Local Bus cards
02:29.34Qwelloh
02:29.34ToR\La search for mwi disable on voip-info.org and google came up with nothing really good
02:29.35Qwellheh
02:29.39Qwellclose enough...
02:29.57MalthusVLB gotten beaten by its competitor 'PCI'
02:30.03iCEBrkrThat the 'fast' 'enhanced' IDE controller on it
02:30.11*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
02:30.14QwelliCEBrkr: ata33?  heh
02:30.16iCEBrkrhaha
02:30.32litagewhat's the difference between `sip show peers` and `sip show users`and `sip show registry`?
02:30.36iCEBrkrlets not forget the 16550 UART I/O chip set!
02:30.41Malthuslol
02:30.46Qwelllitage: users != peers != registered users
02:30.48iCEBrkrBLAZE'N
02:31.38iCEBrkreep, I showing my geekness. and age.. 16550 should be a unknown number.
02:32.23ManxPowerNo, the 16550AF was good, it was the older models were the bane of my tech support life for a year.
02:32.27litageQwell: what does asterisk consider to be a user, peer, and registered user?
02:32.44iCEBrkreeww! vic
02:32.47ToR\Lhehe
02:32.59ToR\LI missed acoustic couplers by | | that much
02:33.25ToR\Lbut a 300 baud vicmodem still needed a rotary dial phone for manual dialing
02:33.39Malthuslater people
02:33.40iCEBrkrEh, my 300 baud modem had tone dial.
02:36.13iCEBrkr<PROTECTED>
02:36.42fileeh?
02:36.45filewhy do you want bkw
02:36.55drumkilla[work]everybody should want file
02:36.56MikeJ[Laptop]cuz he;s sexy
02:36.59iCEBrkrI dont' 'want' him.. he might like that.
02:37.04fileyeah - everyone should WANT me!
02:37.07iCEBrkrlol
02:37.11filein a sexy way
02:37.14Qwellfile: should...but don't
02:37.17MikeJ[Laptop]me?
02:37.19file;(
02:37.19iCEBrkrWTF is up with this EAGI crap?!
02:37.31iCEBrkrMy Asterisk box is looking at me like Helen Keller.
02:37.33MikeJ[Laptop]ummmm
02:37.38MikeJ[Laptop]wow...
02:37.38iCEBrkr..as in, it can't F'n hear me.
02:37.47MikeJ[Laptop]that's wrong
02:37.50ToR\Lhmm the mailbox line I commented out was in my section of sip.conf
02:37.53ToR\Lsound about right?
02:37.59Qwellyesh
02:38.01Qwellyeah
02:38.06darwin35how did helen kellers parents punish here
02:38.10iCEBrkroh no
02:38.13iCEBrkrlook what I started
02:38.17darwin35put door knobs on all the walls
02:38.22MikeJ[Laptop]introduced her to you?
02:38.22ToR\Llol
02:38.23*** join/#asterisk cybyc (~cyb@Ottawa-HSE-ppp269044.sympatico.ca)
02:38.38MikeJ[Laptop]wow... sorry.. that was wrong too...
02:38.53darwin35how did helen keller burn her face ... she tried to answer the iron
02:39.01iCEBrkrAccording to the so called documentation, the audio file descriptor in EAGI is stderr+1
02:39.30darwin35I wear a hearing aid I can tell the jokes
02:39.30iCEBrkrdarwin35: oh, so you're allowed to poke fun?
02:39.44Qorkyso is there a way to test me timing? meetme still bombs out
02:39.53*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
02:40.05darwin35yes look at the tools included
02:40.12darwin35in the zaptel dir
02:40.19iCEBrkraye, buy me pint o'ale and ye timing shall be fixed, mate
02:41.20Qorkyyou mean this?
02:41.20QorkyOpened pseudo zap interface, measuring accuracy...
02:41.21Qorky99.987793% 99.987793% 99.987793% 99.987793%
02:41.23darwin35did you edit the meetme.conf
02:41.41darwin35and map meetme right
02:42.04Qorkyerm i have edited it.
02:42.13Qorkyunder rooms it has this
02:42.13Qorkyconf => 50
02:42.19Qorkyi have no password for it.
02:42.28DaminIt's fucking ICEBRKR
02:42.34iCEBrkrOh god.
02:42.43iCEBrkrIt's that beer drinker.. Damin
02:42.48DaminWhats' up dude?
02:42.51iCEBrkrNot a damn thing.
02:42.54iCEBrkrPlaying in hurricanes already
02:43.24iCEBrkrBut, right now, I'm bored off my ass, so I'm hacking around with EAGI stuff
02:43.48ToR\LiCEBrkr: you said you've turned off mwi?
02:44.08iCEBrkrToR\L: I think I just reset my ATA and the MWI reset
02:44.11_DAWboooo Emily :(
02:44.47ToR\Lhmm
02:45.13iCEBrkrI had a remote phone do that.. All the messages were erased but the MWI was still blinking
02:46.02fileMWI! that's what I was going to look at
02:46.12*** part/#asterisk cybyc (~cyb@Ottawa-HSE-ppp269044.sympatico.ca)
02:46.18ToR\Lreset the ata
02:46.51DaminiCEBrkr: Are you up for an N2Net Alumni OUting?
02:47.06DaminiCEBrkr: With Mac The Hack as the guest of honor?
02:47.06iCEBrkrDamin: If you fly me up there LOL
02:47.08ToR\Ldoh no that didn't do it
02:47.11iCEBrkrMac!!!!
02:47.13ToR\Lit was maybe 90 seconds
02:47.33iCEBrkrDamin: When were you thinking?
02:47.51DaminiCEBrkr: Late august..
02:48.05iCEBrkrDamin: I was thinking of maybe visiting Labor Day
02:48.15Qorkydarwin35: that look ok ?
02:48.37darwin35dmesg | less and look for the config of the card
02:48.38ToR\Lactually 30 seconds
02:48.41ToR\Lhmm
02:48.49darwin35cut and paste in a pvt window
02:49.37litagewhen using x-lite, how do you dial a phone number that contains letters?
02:49.53Twisterjust like you do on any other phone
02:50.01Twistertranslate the letters into numbers
02:51.24DaminiCEBrkr:4
02:51.59*** join/#asterisk vuvie (~vuvie@bb219-74-45-31.singnet.com.sg)
02:52.01DaminiCEBrkr: That would be cool. A Labor Day Weekend NACS Outing.. Maybe Steve and Martin can re-enact the Summer Sausage incident..
02:54.07*** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net)
02:55.57litagetwisted: what if userA and userB have "williamA" and "williamB" as their extensions, though? if you translate letters into numbers, there's ambiguity..
02:56.14iCEBrkrhahahahahhahahaaha
02:56.18QwellWhen would their extensions be letters?
02:56.37Qwells/when/why/
02:56.47litagebecause they're easier for users to remember
02:57.06litageBob: "what's joe's extension?"    Alice: "joesmith"
02:57.31Qwellusers who can't remember 4 digit extensions should be shot
02:57.35iCEBrkrhaha
02:57.48litageQwell: and if you have 30 extensions?
02:57.50iCEBrkrEasier to remember, but harder to spell on the phone.
02:58.11Qwell"Bob: Whats Pams extension?"  "Alice: piyawan-patcharaprasertsook"
02:58.11litageiCEBrkr: if you use a softphone, it's not harder
02:58.24*** join/#asterisk nifter (~nifter@190-141.SPEEDe.golden.net)
02:59.02QwellI'd smack our phone lady if she made extensions names...
02:59.12litageQwell: why?
02:59.27Qwellthat was the real name of one of our employees.  She would have either had to break the standard (first name + last name), or actually make a 30 digit extension
02:59.47ToR\Lin high school
02:59.53ToR\Lwe had a networked system
03:00.11ToR\Land your username was first three letters of your last name, then first 3 of your first name
03:00.19ToR\Lso john smith was smijoh
03:00.22ToR\Lwe had a don hart
03:00.31litageQwell: then you limit it to something like "first letter of first name + lastname, max X chars"
03:00.43Qwelllitage: Then you hit ambiguity
03:00.48QwellJohn Smith vs Jack Smith
03:01.22iCEBrkrIt's called an Excel Spreadsheet...
03:01.27iCEBrkrPrint and distribute.
03:01.31iCEBrkrPin to your cube wall.
03:01.35iCEBrkrk/thx/bye
03:01.39litagemany ISPs/network admins/etc used the old "first letter of first name + lastname, max 8 chars" rule, and it worked fine
03:01.49Qwelllitage: bzzt
03:01.52Jinxyanybody know the vista 350 time/date procedure to update adsi information?
03:02.16litageQwell: if jsmith already existed and jack smith came along, he'd get something like jsmith2
03:02.19Qwelllitage: Then you have to add numbers.  Let's use something like msmith
03:02.28Qwellmsmith2, msmith3, msmith10, msmith95
03:02.31QwellThats 8 chars
03:02.45QwellYou have enough employees/customers, and you're toast
03:02.56iCEBrkryea, so whats Jack Smiths number?? 3 or 95?
03:03.06QwelliCEBrkr: in which order did he join?
03:03.09iCEBrkrjsmith3? jsmith95?
03:03.11Qwelloh, and that
03:03.12iCEBrkrI dunno man.
03:03.19iCEBrkrwe just assign them that way :D
03:03.29Qwell"Whats Jacks number?" "jsmith...uhh...46?"
03:03.37iCEBrkrQwell: exactly
03:03.41Qwellyeah, I got you
03:03.55litageiCEBrkr: asking "what's jack smith's number?? 3 or 95?" is the same as asking "what's jack smith's number? 123456 or 234567?"
03:04.09litageiCEBrkr: it's just that "jsmith3" and "jsmith95" are more descriptive
03:04.18iCEBrkrI dunno man. It's pretty simple to just have a printed piece of paper for people to reference.  Matter of fact, it's quite common.
03:04.23Qwell"Whats Jacks last name?" "How do you spell that?"
03:04.27Qwellor, 4 digit numbers
03:04.44ToR\LiCEBrkr: can you show me an extension line from extensions.conf (minus any phone #s)?
03:04.50iCEBrkrlitage: Ever see those inserts on the phones?
03:04.55iCEBrkrhehe
03:04.57litageif you have a doc with the extensions, then spelling doesn't matter
03:05.12iCEBrkrI don't know what you're asking me fore?
03:05.13iCEBrkrerr
03:05.13iCEBrkrfor
03:05.31litageiCEBrkr: i'm just responding to your statements, that's all
03:05.37iCEBrkr; 2101 / Sipura:Port1 - Florida
03:05.37iCEBrkrexten => 2101,1,Macro(extension,2101)
03:05.42iCEBrkr; 2102 / Sipura:Port1 - Tim
03:05.42iCEBrkrexten => 2102,1,Macro(extension,2102)
03:05.46iCEBrkr; 2103 / Sipura:Port2 - Andrea
03:05.46iCEBrkrexten => 2103,1,Macro(extension,2103)
03:05.49ToR\Lhmm
03:05.52iCEBrkrPRetty simple
03:06.34litageQwell: whether your extension sheet says "joe smith => 1234" or "joe smith => jsmith95" doesn't matter, except that 1234 is harder for an avg user to remember
03:06.51Qwelland jsmith95 is harder to dial
03:07.04ToR\Lexten => xxxxxxxxxx,1,Dial(SIP/xxxxxxxxxx,30,tr)
03:07.06ToR\Lwhere the x
03:07.10iCEBrkrlitage: I dunno man, I've been working with 3-4 digit extensions since I was 7.. Not that big of a deal
03:07.11ToR\Lx's are the #
03:07.18iCEBrkrlitage: you remember the people you call the most.
03:07.19litageQwell: so what's more important? ease of dialing, or ease of remembering?
03:07.34iCEBrkrs/7/17
03:08.09iCEBrkrAnd at my last job, we had a sweet Call Manager interface.
03:08.13litageiCEBrkr: i agree, 3-4 digit extensions are easy; basic. i'm just wondering why letters are translated into #s rather than being treated as valid, separate chars
03:08.18iCEBrkrWhich I wanna code up for Asterisk.
03:09.21iCEBrkrOk, yeah, I'm getting nohwere with this EAGI crap
03:09.23*** join/#asterisk dysan (~ack@202.37.224.27)
03:10.25dysanim getting make[1]: *** [chan_zap.o] Error 1 errors and chan_zap.c error: dereferencing pointer to incomplete type errors, how do i get around these?
03:10.40litageanyways, so i got x-lite working on linux, speakers and mic and all. why might x-lite work properly, but kphone not even bother to touch /dev/dsp?
03:10.57iCEBrkrkphone always blew up for me as soon as the call was completed
03:11.34litageiCEBrkr: kphone kicked the bucket if it lost focus for me, so you're faring better with it than i  =P
03:11.37iCEBrkrerr, connected.
03:11.38*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
03:11.45iCEBrkrhaha
03:11.57iCEBrkrit'd ring, I go to answer it and it'd blow up
03:12.04litagestrange
03:12.17iCEBrkrI haven't had much luck with softphones
03:12.27litagemine connects fine, but doesn't play or record audio
03:12.35iCEBrkroh wait. I got that fwd/pulver.communincation thing working
03:13.05litagex-lite has worked perfectly on 2 win and 1 mdk machine so far for me
03:13.41iCEBrkrx-lite's interface is F'n GAY
03:13.45litageaye
03:13.53iCEBrkrCould they make it any more bulky?
03:14.28litageyeah they could. i've seen a couple other softphones that took up like 1/3 of your desktop and had like 5mm of padding around each button
03:14.37litageit was like they were expecting you to use a broom on a touchscreen
03:14.42iCEBrkrhaha
03:14.53litagethe name slips my mind hto
03:14.54litagetho
03:15.03iCEBrkrit was some java basd one, I think..
03:15.14iCEBrkror maybe it was one of the 'better' looking ones for Linux.. just a binary
03:15.20iCEBrkrI think I know which one you're talking about
03:15.44litage=P  you just have to shake your head when you see things like that. i mean, WHAT are the developers thinking??
03:16.19litagesome people undervalue desktop realestate  :(
03:16.33iCEBrkrThey're thinking they have 1600x1200 dual displays
03:17.20iCEBrkrWinAmp is the only interface that has it right.. I'm talking the original.. Not the 'modern skin'
03:18.23litageyeah i agree. i'm not a fan of those new-fangled winamp5 skins that are funky shapes and whatnot
03:18.35empmy ISP uses Interleave on their DSL lines... so my first hop can range between 40-100ms lag.  Would this pose a problem setting up a * box?  What is acceptable lag?
03:18.45ToR\LiCEBrkr: how about the mailbox line in sip.conf?
03:18.51Qwellemp: less then 100 for sure
03:18.52ToR\Lthen I won't bug ya anymore ;)
03:19.06Twisterim creating a menu that brings a call in on one trunk then exits it on another
03:19.14Twisterhere is that the cli outputs
03:19.17TwisterExecuting Dial("IAX2/NuFone@outbound-nufone/2", "SIP/broadvoice/13045905796") in new stack
03:19.17Twister<PROTECTED>
03:19.33iCEBrkrWow. My latency is lower than I thought
03:19.41iCEBrkrOK (30 ms)
03:19.45iCEBrkrOK (23 ms)
03:19.49Qwell30 to your provider?
03:20.14iCEBrkrWell, that's 'sip show peers'
03:20.22iCEBrkrMy asterisk box is remote.. my sipura's are here
03:20.41empis DSL suitable? should I be looking at T1 or colo?
03:20.48iCEBrkremp: It's doable.
03:20.58Drukenemp for how many calls?
03:20.58iCEBrkremp: You can get skiping and dropout tho
03:21.23empOr maybe I just need to find the tech that can turn off Interleave on my line :) heh
03:21.24iCEBrkrI ran my asterisk setup off DSL for awhile.
03:21.41iCEBrkremp: If you're just building a glorified answering machine for home it should be ok
03:21.43Twisterhttp://pastebin.com/311643
03:22.00Twistertheres the pastebin of my menu
03:22.27empI would like to be able to add remote extensions down the road... but that might be a while
03:22.39iCEBrkrToR\L: I have the mailbox=2101 in my sip.conf
03:23.07*** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au)
03:23.13iCEBrkremp: oh, I had more issues with using voicepulse for LD
03:24.36iCEBrkrI did a phone interview over voicepulse and my DSL was saturated.  Could have been that I didn't tinker with the codecs
03:25.14spoot_nicki'm looking for a working FOSS CDR rating/billing system. does anobody has any advice on the subject?
03:25.28*** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz)
03:25.50ToR\Lhmm
03:25.53spoot_nicktried trabas (everybody says it sucks). got it to work, didn't understand how i can pull records from asterisk's cdr table. CDRTool is free for personal use, so it doesn't suit
03:26.05litagehow fast does telephonyware.com ship orders?
03:26.40iCEBrkrspoot_nick: I don't have a complete system, but I log all my calls to MySQL
03:26.43file[laptop]you can't get everything for free.
03:26.53iCEBrkrfile[laptop]: who asked you?
03:27.01iCEBrkr:D
03:27.07file[laptop]I asked myself.
03:27.09iCEBrkroh
03:27.10file[laptop]I count as two people
03:27.13file[laptop]see?
03:27.17iCEBrkrfile[laptop]: Proceed.
03:27.27JunK-Yfile: u and u double personality? :P
03:27.32file[laptop]exactly
03:27.34spoot_nickfine, but the question applies to FOSS  =)
03:27.34iCEBrkrlol
03:27.35JunK-Yhehhe
03:27.52iCEBrkrfile[laptop]: get back to work!!
03:27.59file[laptop]meh work ended awhile ago
03:28.04file[laptop]I just wanted to clean up my config files
03:28.11file[laptop]and debug a problem
03:28.18iCEBrkrOoooooooh, personal stuff. I see.
03:30.29file[laptop]uh my cellular provider like... billed me for something they shouldn't have
03:30.57MikeJ[Laptop]what was it.... a puppy?
03:31.02JunK-Yerotic lines are bad for phone bills u know it file!
03:31.18file[laptop]nah I called a friend... local
03:31.20file[laptop]but they charged me
03:31.27file[laptop]my evenings/weekends thing didn't kick in
03:31.53JunK-Yya, shits like that happens so often, that why im checking really carefully my bills.
03:32.49*** join/#asterisk jaike (~a@203.131.137.76)
03:33.51MikeJ[Laptop]tzafrir, you around/
03:33.55file[laptop]they charged me $25
03:34.03file[laptop]billed my local calls as long distance for a night
03:34.22jaikeanyone know the cause of choppy asterisk calls? were currently using asterisk and vonage and we dont seem to have problems with voice quality with vonage like we have with asterisk
03:35.14jaikewere using sipura phones and cisco 186 atas with asterisk
03:37.54iCEBrkrMy calls are only choppy when they're important ones.
03:38.05iCEBrkrI can talk to my parents back home all night and day without a problem.
03:39.43jaikeam just wondering why vonage quality is ok while asterisk isnt ... what protocol is vonage using?
03:39.49Twisterim assuming you are using a zaptel
03:39.55Twisterinterface for vonage?
03:40.09jaikewere using sip
03:40.17Twistergot a softphone account?
03:40.27*** join/#asterisk fugitivo (~ajf@201.255.104.90)
03:40.28fugitivohello
03:40.45Jinxyhello fugitivo
03:41.18Jinxyguys, would I be limited in my ability to use * features on a analog phone compared to an ip phone?
03:42.00empoh, great... I'm in one of 3 states I can't switch my # to vonage :)
03:43.01iCEBrkrJinxy: I prefer ATA+Analog phones
03:43.03Qwellemp: Can't really use vonage with asterisk anyways
03:43.17*** join/#asterisk brenda (~nnnnn@c-67-182-205-227.hsd1.ut.comcast.net)
03:43.18fugitivoJinxy: it depends, a cisco 79xx ip phone? :)
03:43.26Qwelljaike: probably g729
03:43.28empQwell, what are some providers that are compatible?
03:43.41Jinxyfugitivo, say grandstream 101
03:43.46iCEBrkremp: there's a ton listed up on www.voip-info.org
03:43.58Jinxyfugitivo, are you saying some features are phone specific?
03:44.22iCEBrkrJinxy: The feature you're losing with a IP-Phone is taking your friends out for beer after you've set it all up. Cuz they cost too damn much!! :D
03:44.34fugitivoJinxy: well, cisco has those ip phones with big screens where you can run xml applications :)
03:44.50Jinxyfugitivo, I am talking features provided by *
03:45.00darwin35anyone have the line for musiconhold for madplay
03:45.04Jinxycall transfer, music on hold and such
03:45.10fugitivodarwin35: yes
03:45.22darwin35fugo can you pvt me with it
03:45.28fugitivoholdon
03:45.51fugitivodarwin35: default => custom:/var/lib/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000 --output=raw:- --attenuate=-4
03:46.25fugitivoJinxy: no, you're not limited
03:46.38Jinxythank you very much
03:46.51darwin35thnks
03:46.53mishehuhmm...  was there a post-1.0.9 clid bug?  I noticed I'm not getting clidname with my pri anymore.
03:46.54Jinxyfugitivo, I've got 12 analog phones, and replacing them won't be cheap
03:47.03darwin35I was missing the attenuate
03:47.08Jinxyso, I was thinking of just getting a channel bank
03:47.25Jinxyand on top of that, I've got 4 POTS lines
03:47.26fugitivoJinxy: hmm, check the wiki about the channel banks
03:47.35Jinxyfugitivo, I know about that, no biggy there
03:47.56fugitivoJinxy: ok, just be sure to get a channel bank reported to work ok with asterisk
03:52.16Jinxyfugitivo, say features like call waiting, what if my analog phone doesn't support it?
03:55.31darwin35that fixed it
03:55.51iCEBrkrJinxy: Why wouldn't an analog phone support call waiting?
03:55.52Nuggetcall waiting does not require cooperation from the phone on an analog channel.
03:55.55iCEBrkrit's a flash-hook
03:56.04Nuggetit's impossible for an analog phone to "not support" call waiting
03:56.15Jinxyoh, I was thinking visual call waiting
03:56.21iCEBrkrNugget: well, it might not have the 'flash' button which might confuse the user :D
03:56.22Jinxythat the phone needs to support
03:56.39darwin35i have a flash hook setup
04:00.15|Vulture|Any Sangoma users around?
04:00.24file[laptop]blitzrage!!!
04:00.27blitzragefile[laptop]: !!!
04:00.38blitzrageSwK_: hey
04:00.53SwK_hey
04:00.54jaike1.0.9 has a queue bug...some agents logged in to a queue do not receive calls from the queue..had to go back to 1.0.7
04:00.56SwK_what
04:01.02SwK_heh
04:01.18blitzrageSwK_: you msg'd me
04:01.27SwK_yeah
04:01.53SwK_i was gonna ask about a bug you found on RTP path
04:02.10SwK_but I figured it out... wasnt a SIP bug but a libpri bug
04:03.45|Vulture|can I bring a PRI into a TSU750, then output it to a PRI and a signle FXS?
04:04.17*** join/#asterisk SwK (buteod@12-219-156-206.client.mchsi.com)
04:04.37SwKdamned client
04:05.16drumkillajaike: care to gether details on that so that we can get it fixed?
04:06.09jaikedrumkilla: ok..will try to replicate problem again
04:06.21drumkillasweet  :)
04:12.00*** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
04:12.14*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
04:12.16darwin35ok in cvshead when I do help database it shows it using put and get yet head examples say to use set ${db
04:12.40darwin35did they go back to put and get
04:12.55drumkillano, that example just didn't get updated
04:13.09darwin35ok
04:13.45darwin35drum have a min I have issues them things not getting put into the db
04:13.59SwKdamn it
04:14.11SwKwhere is that setting for app_voicemail on which sound file to attach?
04:14.26darwin35voicemail.conf
04:14.26iCEBrkrvoicemail.conf?
04:14.31SwKyeah
04:14.37SwKin voicemail.conf
04:14.44darwin35read it
04:14.48SwKtI am
04:14.54SwKi dont see it tho...
04:15.08SwKi guess its one of the RTFSC config options
04:15.41darwin35exten => password,username,email addy,file
04:15.42*** join/#asterisk znoG (~gs@200.115.216.109)
04:15.46darwin35its all there
04:16.27*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
04:21.12*** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net)
04:27.15*** part/#asterisk jaike (~a@203.131.137.76)
04:30.21darwin35http://pastebin.ca/17514
04:30.34darwin35its not putting infor into the db
04:30.43darwin35when I set the functions
04:31.31*** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net)
04:34.17darwin35if more is needed I can paste the full exten.conf
04:37.05darwin35what did everyone jump ship
04:37.15file[laptop]yes.
04:38.28Hmmhesaysit's late
04:39.14file[laptop]it's 1:38AM, yes
04:39.20file[laptop]I should have said no, someone's waiting for me!
04:39.53Hmmhesays1:38 huh? are you off in canadia?
04:40.02file[laptop]Atlantic Canada
04:40.13file[laptop]just wanna tell you don't worry... I will be late, don't stay up and wait for me
04:40.18file[laptop]say again you're dropping out my battery is low
04:40.19file[laptop]so you know
04:40.22file[laptop]we're going to a place near by
04:40.23file[laptop]gotta go
04:40.56*** part/#asterisk OzJames (~opera@203.208.64.29)
04:42.24*** join/#asterisk viking667 (viking@203.184.23.240)
04:42.33viking667whew. Here at last.
04:42.50file[laptop]so you think.
04:43.14*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:43.24viking667anyone here using the irssi script "speak.pl" that feeds the festival text-to-speech engine? I'm not managing to get it to go.
04:44.08viking667I was told to come here - or at least, I think I was...
04:44.42*** join/#asterisk dano_ (~bismantux@216.147.160.17)
04:45.55*** join/#asterisk Luke-Jr (~luke-jr@207.192.219.246)
04:46.54johnh51:)
04:48.25darwin35man fixiing this is a pain
04:50.12*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
04:54.34*** join/#asterisk jgenender (~blah@209.181.65.238)
04:54.57jgenenderHello...I am have a bit of an issue.  I am doing sip-to-sip...
04:55.08jgenenderBasically Teliax to my asterisk box
04:55.19jgenenderand I am getting duplicate digits with DTMF
04:55.24jgenenderi.e. if I dial 1010
04:55.36jgenenderIt sometimes understands 110110
04:55.39jgenenderor 10010
04:55.40jgenenderetc
04:55.49jgenenderany ideas on what is wrong or how to fix this
04:55.55jgenenderThere is no Zaptel in this config BTW
04:58.32darwin35http://pastebin.ca/17517
04:58.42darwin35does this look correct
04:59.16darwin35i know part of a macro i s missing it has not ben writen yet
05:03.12jgenenderHmmm...time to build from the HEAD...looks like my issue is a bug: http://bugs.digium.com/view.php?id=4659
05:05.06darwin35check that you set your dtmfmode= and dtmf= rfc2833
05:05.20darwin35in your sip.conf and other fonf files
05:05.31drumkillafonf files!!!
05:05.38jgenenderI believe I did...
05:05.52darwin35dbl check
05:06.08Qwelldrumkilla: There were like 4 other typos you could have chosen ;p
05:06.39jgenenderOk...the dtmfmode=rfc2833 but not dtmf...both need to be set?
05:07.16jgenenderOk...both set...same problem
05:07.21jgenenderLooks liek its that bug
05:07.28jgenenderlike^^
05:07.34darwin35what phone
05:07.53darwin35I set both
05:07.59jgenenderI am calling an inbound number for Teliax...no matter what phone I use I get this problem
05:08.15jgenenderI tried from Cell, from standard PSTN, and from Cisco 7940
05:08.25darwin35hmm
05:08.40darwin35what day of head you on ?
05:08.50jgenenderI am on standard 1.0.7
05:09.21jgenenderEverything works fine from SIPPhone and FWD (which are dtmfband=inband)
05:09.27jgenenderTeliax is rfc2833
05:10.39*** join/#asterisk MustDie (~voip@ool-18b91f29.dyn.optonline.net)
05:10.42*** join/#asterisk cfrank_ (~cfrank@wsip-24-234-137-140.lv.lv.cox.net)
05:13.48WilliamKevenin ya'll
05:16.15MikeJ[Laptop]jgenender, did you try head, with and without that patch?
05:16.37MikeJ[Laptop]testing is appretiated
05:16.39jgenenderNo...that is next...I just wanted to see if others are having similar issues...this really is an annoying issue
05:16.48MikeJ[Laptop]what are you seeing
05:17.19jgenenderI am getting duplicate or double DTMF (somewhat random) for dtmfmode=rfc2833
05:17.22MikeJ[Laptop]that one should go in pretty quickly as soon as we get a disclaimer fromthe guy, and he has been actively responding soo..
05:17.26jgenenderWhich what Teliax uses
05:17.40MikeJ[Laptop]inbound, to asterisk?
05:17.50jgenenderYes
05:18.03MikeJ[Laptop]I thought that bug was about how we send dtmf
05:18.24jgenenderHmmm...let me re-read it
05:18.41*** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl)
05:19.12jgenenderMikeJ[Laptop]: No...i tlooks like its reading DTMF
05:19.32MikeJ[Laptop]ummmm
05:19.56jgenender"If rtp rfc2833 packets arrive out of sequence for a single DTMF event,
05:19.57jgenenderAsterisk may report double or triple digits received
05:19.58jgenender"
05:20.20jgenenderThats the same issue I am having (or so it seems)
05:20.36MikeJ[Laptop]ok, it is both sending and reciving maybe.. I am lookin at thte patch
05:21.14MikeJ[Laptop]ok.. looks like both sending and reciving...
05:21.37MikeJ[Laptop]well.. give it a try, and post results to that bug to say how it affected you...
05:21.45jgenenderI will..thanks.
05:23.13*** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au)
05:24.12jgenenderMikeJ[Laptop]: Do you know if the rtp.c becomes part of the asterisk binary or one/many of the libs?
05:24.30jgenenderI just want to determine what I need to update from a binary perspective
05:24.44MikeJ[Laptop]yes, part of the bin...
05:24.52jgenenderok thanks
05:25.00jgenenderPatching now...
05:25.00MikeJ[Laptop]but if you are going from stable to head, you need to do it all regardless
05:25.07jgenenderok
05:25.37MikeJ[Laptop]make clean, remove all the .so modules, then make, make install
05:25.45jgenenderdoing it now
05:25.47MikeJ[Laptop]ok.. .sleep time for me.
05:25.50MikeJ[Laptop]night all
05:25.58jgenendernn...and thanks
05:27.31DA-MANhow often should one rebuild head
05:28.30QwellDA-MAN: hourly
05:29.30DA-MANok!
05:29.30DA-MANhehe
05:29.36DA-MANi do mine weekly
05:33.36MustDieseimhourly
05:36.23*** join/#asterisk cj (~cjcollier@CJ.monthly-bronze.supporter.pdpc)
05:38.10*** join/#asterisk jaike (~a@203.131.137.76)
05:38.27jaikequestion guys...will setting the jitterbuffer too high have any negative effects?
05:40.48jgenenderbig delays
05:41.18mostyi'm having trouble getting a linksys pap2 to work from behind NAT, it won't connect to an asterisk server outside the NAT (but will to one inside). i've tryed playing with the NAT settings, but nothing seems to work. what could be wrong?
05:41.45cfrank_Anyone built meetme2 w/o mysql? (postgres only)
05:41.48DA-MANmosty, pap2-na
05:41.50DA-MAN?
05:42.01mostyda-man, yes
05:42.27jgenenderMikeJ[Laptop]: Don't knwo if you are asleep but the patch worked ;-)
05:42.32DA-MANdo you have nat=yes, qualify=yes on sip.conf
05:43.24mostyda-man: i had nat=yes, qualify=xxx in the general section. i've changed qualify to yes now
05:44.19DA-MANno you have to have it in the [ext] stanza, not in general
05:45.36opus__holy JESUS SON OF GOD
05:45.49opus__i'm getting cvs spam
05:46.02opus__(chris farley voice)
05:47.02Math`cvs spam?
05:47.11Qwellcvs commit spam?
05:47.26mostyda-man: ok, i created an [ext] secion, and put the nat and qualify settings in there, restarted asterisk. still doesn't register (how can i force the unit to attempt to register again?)
05:47.31jaikejgenender: big delays? choppy calls?
05:47.56DA-MANensure that register is et to yes under the line
05:48.28jgenenderjaike: This bigger the buffer the longer the delays in when the voice gets to you
05:48.39jgenenderYou may be waiting for voice...its annoying
05:49.39mostyda-man: register is already set to yes. i got it to reboot via ivr, but still no connection
05:50.21DA-MANwhats sip debug say
05:50.56infinity1hola folks
05:51.11opus__hey MATH
05:51.20opus__yo QWELL
05:51.47opus__infinity1 - sveasoft forums has a good iptables rate limitor script for that
05:51.48mostyda-man: there are a bunch of messages flying past, some mention the pap2's private lan address, others mention my nat machines public ip address
05:52.03opus__infinity1: do both
05:52.19infinity1opus__: already got QoS setup.
05:52.24opus__really
05:52.28infinity1opus__: and graphed ;)
05:52.42opus__what.
05:52.54infinity1http://prelude.brendon.com/~brendon/cacti/graph.php?rra_id=all&local_graph_id=54
05:52.59opus__no way
05:53.31infinity1i'm first a network engineer. then a voip experimenter. heh
05:53.34opus__haha that awesome
05:53.47infinity1obviously * is in purple
05:53.56*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
05:54.30jaikebye guys..thanks jgnender..will try lower settings..
05:54.32opus__Thats pretty unique
05:55.22infinity1thanks.
05:55.22Math`fuck just got an * crash on a call transfer
05:56.16Math`(cvshead)
05:56.56|Vulture|Math`: how many times did you transfer? just 1nce?
05:57.03Math`|Vulture|: for that call, yeah
05:57.07opus__where do you download movies
05:57.16infinity1bit torrent sites.
05:57.18|Vulture|hmm there is an issue with transfering multiple times
05:57.21Math`opus__: www.torrentspy.com / wwww.mininova.org
05:57.22opus__ever since btenet died with rss feeds i'm been dailyshowless
05:57.28|Vulture|Martohtar: I haven't looked at CVS current
05:57.29Math`|Vulture|: there's an issue for PARKING multiple times
05:57.46infinity1opus__: tell me about it. i haven't been downloading tv shows cuz of that
05:57.52|Vulture|I think transfers as well with a PRI
05:57.58Math`ah was SIP to SIP
05:58.04infinity1opus__: notice my traffic drop over the past few months :)
05:58.28opus__\haha
05:59.02Math`|Vulture|: note that codec transcoding might have been required on the new line (the one that received the x-fer), x-fer done via astman
05:59.09Math`I'll try to replicate it
05:59.14infinity1i really need to switch windows and start making head packages for debian.
05:59.24infinity1but its funner to just chil here. lol
05:59.34*** join/#asterisk clive- (~pirch@rndf-146-9-138.telkomadsl.co.za)
05:59.51opus__math dude thanks!!! the.daily.show.07.11.05.marci.hamilton.dsr.xvid-crimson.[VTV].torrent
05:59.59Math`np
06:00.28opus__i freaking pay for cable but have no tv
06:00.37Math`thats bad
06:00.43Math`I freaking pay for cable tv but rarely watch it
06:00.46infinity1all the cable is for inet?
06:01.03MustDieopus: no biggie, it's nothing on tv anyway
06:01.14justinkany suggestions for unlimted SIP VoIP provider that offers GSM or equavilent economy codec?
06:01.23MustDiethere are about 6000 channels and only 300 has something to watch on ;)
06:01.36opus__justink how much do you want to spend
06:01.51Qwellopus__: OT - piratebay too
06:01.52justinkas little as possible-its residential service.
06:02.19Qwelljustink: How many minutes per month would you say you use?
06:02.22DarthCluejustink: you want PSTN terminatin with that, or just VOIP?
06:02.42opus__justink - well you can get very cheap service but there is no guarntee that the service will be up the next day. your better off paying $24.99 or buying your own PRI and reselling it yourself
06:03.06Math`voipjet seems reliable
06:03.13justinkDef want PSTN termination, availity to have virtual numbers, full feature set...and prefer not to use G711 as Im a bandwidth hog.
06:03.45opus__well i work for a few providers so if i recommended them it'd be a conflict of interest:)
06:03.58Math`lol
06:04.00opus__what i want is g711 wideband
06:04.03justinkread that broadvoice uses 711, teliax offers gsm.
06:04.13DarthClueopus__: it is always a conflict of interest, that doesn't stop most of us.
06:04.16Qwellopus__: no, it'd be a shameless plug
06:04.20justinkyou and I are after diff things then opus
06:04.22Qwellnothing wrong with that...
06:04.54Qwellopus__: If somebody asks you what a good provider is...  If you believe in your company, hell, promote it
06:05.03opus__ahaah!
06:05.04opus__haha
06:05.17QwellI said if. :P
06:05.18justinkNo need for a CD quality phone call...at least for me.
06:05.22darylpis anyone having problems with free world dialup?
06:05.33*** part/#asterisk jaike (~a@203.131.137.76)
06:05.34justinkI'll check out voiphet
06:05.43Math`darylp: Im talking on it right now
06:05.43justinkvoipjet
06:05.44Qwellisn't voipjet outgoing only?
06:05.47Math`it is
06:05.48DarthCluejustink: virtual numbers could be an issue with some providers.  what countries do you intend to call?  asterlink offers 2c per minute voip termination to the US48 with a toll free number.  They support gsm and sip as well.
06:06.13opus__and if you have a credit card right now you can dcc chat me we can start an account right now
06:06.14justinkcontinental US only.
06:06.22darylphmm, then I must have a problem
06:06.34Math`opus__: who are u working for?
06:06.42Qwelljustink: If you need incoming calls, you might want to reconsider voipjet
06:07.00opus__math a few resellers, one call center.. just the dumb shit. i got my own dream so fuck em
06:07.00justinkif its outgoing only youre right.
06:07.10Math`ok
06:07.18DarthCluejustink: check out asterlink, 2c per minute with a toll free, incoming and outgoing, they can set you up for sip and gsm.
06:07.40darylpwith iax debug on should I see something when I try to call fwd from outside my system?
06:07.46justinkok-will do some digging and googling. thanks.
06:08.13darylpof course the fwd number is registered
06:08.13*** join/#asterisk marshall (~test@24.77.245.158)
06:08.25DarthCluedarylp: if it is using iax, yes.
06:08.52marshallCan someone please tell me how to move to the next line of the dialplan when a call exits non-zero
06:09.13darylpI'm seeing nothing when I call a local access number to fwd extension using pots
06:09.25infinity1awesome. i think i found a solution to my debian packages! checkinstall!
06:09.27DarthCluemarshall: pastebin the extensions.conf file
06:09.28Qwellmarshall: show application dial
06:09.44enderinfinity1: doesn't work nearly as well as you might wish.
06:09.47DarthClueQwell: but is he using dial?
06:09.55infinity1ender: don't tell me that!!!!!!!
06:09.56QwellDarthClue: oh, dunno
06:10.00marshallthanks Qwell, I'll look through that
06:10.15enderinfinity1: it's the equal to maybe making a tarball of compiled binaries into a package, w/out any logic to where they go and whatnot that makes a package desireable.
06:10.17DarthClueinfinity1: nothing ever works the way you want it to.
06:10.40marshallDarth I am using Dial
06:10.44infinity1ender: thats good engouh. i just want something to keep track of where * puts shit
06:10.51infinity1ender: cuz the * makefix sux arse.
06:10.53marshallwant a failover onto a zap channel if my IAX trunk is down for whatever reason
06:10.54infinity1er makefile
06:11.08enderinfinity1: ah, well that works then I suppose
06:11.10Math`marshall: use a dial macro
06:11.25Math`it just to exten+101 in case of failure
06:11.27infinity1ender: sweet. i'm going to try it.
06:11.45marshallMath n+101 is only on a zero exit is it not, this is a non-zero exit
06:11.53opus__darylp - try this, edit /etc/asterisk/logger.conf and make console => or what ever have everything. start asterisk with asterisk -vvvvvvvvvvvvvvvvvvgcddddddddddddddd
06:11.54marshallif the trunk is dead
06:12.08Math`marshall: if the trunk is dead the call will fail and go to n+101
06:12.32opus__darylp - is that fails i am sure there is a command 'iax2 debug' lemme check real quick
06:12.55litagewhich ATAs also have a regular RJ-11/POTS connection that the ATA fails over to in case an internet connection isn't available?
06:13.05Math`marshall: hmm it just jumps to pri+1 sorry
06:13.15Qwelllitage: I think one of the SPA's did
06:13.24marshalldoesnt seem to work like I need it to
06:13.29litagethanks Qwell
06:13.30*** join/#asterisk kd5uzz (~nntpsmurf@139.78.167.69)
06:13.47darylpyeah, I have iax2 debug turned on
06:13.57opus__modify logger.conf
06:14.49opus__chances are, you have a context/dialplan problem  or don't use fwd
06:15.16darylpit was working until this afternoon
06:15.28opus__then, they are down.
06:15.35darylpwhen I went to demo the access numbers to someone and then it stopped
06:15.50marshallMath, any other ideas? Im stumped
06:15.55opus__i think thats a moore law
06:16.11darylpwell, it's murpy's law at any rate
06:16.15Math`marshall: I got more than 1 PSTN termination providers and if 1 fails, it jumps to the second one
06:16.25Math`marshall: the 1st one uses prio1, the 2nd uses prio2
06:16.46litageQwell: any idea what sipura might call that feature?
06:16.52opus__darylp haha right on.
06:17.00marshallIm trying to prevent dead air on an inbound DID call to my PRI, if the trunk is down to the remote site
06:17.08Qwelllitage: dunno
06:17.27marshallI was hoping to bounce it to a backup cell phone or basic voicemail box
06:17.28opus__darylp - i'd help but i opened a FWD account in 2002 and forgot my password :(
06:17.33marshallon the main server
06:17.41Math`marshall: just put a dial as prio+1
06:17.42opus__hehe they wern't even around back then
06:17.55darylpwhen I use the call me app I see nothing
06:18.16marshallMath: where would I find the syntax for that?
06:18.32Math`show application dial
06:18.34Math`in the CLI
06:22.47opus__the cli is pretty funny. what is it suppose to be sql
06:24.16*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
06:24.17opus__darylp - do you have any adavnce IAX2 security features turned on?
06:24.58infinity1wow. checkinstall worked nicely for libpri
06:25.33Math`checkinstall always works nice :P
06:25.33Math`<PROTECTED>
06:25.34DarthCluedarylp: what is your fwd number?  i can give ya a call and see what it does.
06:25.38Math`uhu now I got that with fwd
06:26.17darylp671901
06:26.20darylpthanks
06:26.24wunderkinmath thats fine
06:27.07Math`wunderkin: yeah.... 10000 times
06:27.25DA-MANhehe never tried checkinstall
06:27.25DarthCluedarylp: what do you get when you type 'iax2 show registry' on the CLI?
06:27.26Math`ok I hung up before the 10 000'th time
06:27.59infinity1forget freeworld, just peer with everyone :)
06:28.05darylpall of my iax accounts, including that one, are registered
06:28.12Math`infinity1: lol
06:28.40darylpdo you get that my number is busy?
06:28.53DarthCluedarylp: yes.
06:29.01darylpyep, that's what I get too
06:29.16Math`I love it when I google something, and the first link google gives me is a mailing thread of [asterisk-users] being told to google for it
06:29.28infinity1lol
06:29.28DarthCluedarylp:what codecs do you allow?
06:29.37kd5uzzlol
06:29.38Math`what's rtp codec id 72?
06:29.51darylphmm, I don't know off the top of my head
06:30.02empwhat's the best way to allow >1 incoming calls? more physical lines? use a hosted pbx solution? something else?
06:30.06darylpI get this:
06:30.10darylp<PROTECTED>
06:30.10darylp<PROTECTED>
06:30.10darylp<PROTECTED>
06:30.10darylp<PROTECTED>
06:30.10darylp<PROTECTED>
06:30.19opus__emp - pri or provider. provider is cheaper
06:30.21Math`emp: a PRI line or a voip DID provider
06:30.25Newbie___hi guys, i am using h323, is there any way to know i am connected to a gatekeeper?
06:30.28opus__math i beat you
06:30.35Math`u did
06:30.40emp:) thx
06:30.41DarthCluedarylp: pastebin your iax.conf with user/pass XXed out
06:30.53Math`rtp.c:508 ast_rtp_read: Unknown RTP codec 72 received
06:31.08darylpthat's a lot of editing, hold on
06:31.37darylpjust ulaw
06:31.38kd5uzzare there any useful codes, etc you can get from the phone company using normal phone lines using Asterisk? CID info that sort of thing?
06:31.43darylpI have
06:31.54darylpdisallow=all
06:31.56darylpallow=ulaw
06:32.32opus__kd5uzz electromagnetism
06:32.49DarthCluedarylp: pastebin the complete cli output
06:33.07*** join/#asterisk Assid (~assid@203.115.64.62)
06:33.15darylpok, I apologize, but I don't know what "pastebin" is
06:33.20DarthClue~pastebin
06:33.21jbotpastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca
06:33.39kd5uzzopus__ cool, just what I was hoping for :-)
06:33.42darylpthanks
06:33.53darylplearn somthing new every day
06:34.06darylpdarth, you mean when I try to call myself?
06:34.10darylpthat output?
06:34.23Math`opus__: already got that unknown rtp codec 72 thingy?
06:34.27opus__kd5uzz did you know "In classical electromagnetism, the electromagnetic field obeys a set of equations known as Maxwell's equations, and the electromagnetic force is given by the Lorentz force law"
06:34.28DarthCluedarylp: yes.
06:34.43opus__Math - no, did you try cscope
06:35.01infinity1:)
06:35.27opus__math - cd /usr/src/asterisk && cscope -R
06:35.48infinity1yay. leno is on
06:36.26opus__kd5uzz - use the force
06:37.11Math`opus__: great,  The payload type range 72-76 is marked "reserved" so that RTCP and
06:37.11Math`<PROTECTED>
06:37.27darylphttp://pastebin.ca/17523
06:37.30opus__well, now you know.
06:38.33Math`wiki entry for RTCP: See RTP
06:38.41opus__math - my wild guess, with no experience what so ever, is that you're getting some kind of control packet ? rtcp? that is messing shit up, perhaps a reinvite
06:38.48opus__i'm guess rtcp stands for real time control packet
06:39.00DarthCluedarylp: what operating system?
06:39.08opus__no, wait, it wouldn't  make sense to to revinites in RTP.. hmmm
06:39.10Math`RTP COntrol Protocol
06:39.28darylpfedora core 1
06:39.45darylpI'm running asterisk at home, which is why pasting confs can be a bit of a mess
06:40.02Math`how do u exit cscope lol
06:40.15DarthCluedarylp: i would recommend using either 1.0.9 or CVS-HEAD.  you may be running into a bug in 1.0.7
06:40.48opus__math ok my second guess is that it has something to do with comfort noise
06:40.50Math`ah ok it wants ^d
06:40.53opus__math yeah d
06:40.55DarthCluedarylp: do you have to run *@h or can you run just pure *?
06:41.27darylpwell, I really like AMP, the rest of at home I don't care much about
06:41.32litagewhat's the difference between the SPA-1000 and SPA-2000 lines?
06:42.10DarthCluedarylp: i think you can install * and then add AMP on top of it without the whole *@h setup.  most of us don't like *@h at all.
06:42.14DA-MANSPA-2000 has two physical ports
06:42.20DA-MANphysical phone portz
06:42.29darylpyeah, I've noticed ;)
06:42.34Math`opus__: any cng support in x-lite
06:42.38opus__math -- hey, its for feedback error repair. http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-feedback-11.txt
06:42.59opus__'of course'
06:43.03litageDA-MAN: the SPA-1000 also has 2 POTS ports
06:43.11newmemberDarthClue: what are some of the issues with *@h?
06:43.22opus__math whats cng?
06:43.23darylpthere's a lot I don't like about it, but, it does a lot of configuration for me
06:43.27Math`opus__: comfort noise gen
06:44.08opus__oh we don't need that here i leave on the beach
06:44.12opus__live
06:44.39Math`opus__: yeah but it was your "best bet" of the problem source so I wanted to check if it wasnt enabled somewhere
06:44.43DarthCluenewmember: if you use *@h then you won't ever really know how * works, and if you don't know how it works, you won't ever know where to start when you need to fix it.  * is not difficult enough to need the gui on a fresh install.  if you can follow written directions it takes about 5 minutes to get a base install running excluding compile time.
06:45.11opus__darthclue - exactly.
06:45.42darylpyeah, I've installed pure asterisk, in fact, it's working on one of my macs, but the configuration is overwhelming at first
06:46.08darylpI'm learning quite a bit from at home, and for the most part, it's been working
06:46.17darylpI consider it a transition tool
06:46.29DarthCluejust run make samples and modify what you need from there.  it is quite simple as long as you keep a paper bag handy to keep from hyperventilating.
06:46.37darylphaha
06:46.38Math`whats so different with *@h
06:46.43opus__big mistake a@h delieverd 1.0.8 :(
06:46.46tzafrirDarthClue, excluding compile time, and the time to get the rlevant libraries, and figuring out what the heck they are, etc.
06:46.54darylpI didn't try make samples
06:47.18DarthCluetzafrir: no issues with libraries on this end, i guess that's just a debian problem.
06:48.25darylptnx darth, I have done that, in fact, I have head compiled on my mac
06:48.35opus__nice
06:48.51opus__darthclue why does sse2 break ilbc
06:48.52opus__:)
06:48.53tzafrirlet's compare that to the build-deps on debian...
06:49.22DarthCluetzafrir: that's why i don't recommend debian.  Debian is for people that want to deal with that kind of crap, not for people who need to get it done now.
06:50.22Math`DarthClue: debian is good when u know what ur doing
06:50.37*** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com)
06:50.42Math`Gentoo is good when u want to waste time recompiling
06:50.55opus__yeah
06:50.59tzafrirBuild-Depends: debhelper (>= 4.0.4), sed (>= 3.95), zlib1g-dev, libgtk1.2-dev | libgtk-dev, libreadline4-dev | libreadline-dev, libgsm1-dev, libssl-dev, libzap-dev, libtonezone-dev (>= 1:1.0.6-1), doxygen, bison, libasound2-dev, postgresql-dev, unixodbc-dev, libpri-dev (>= 1.0.6-1),, dpatch (>= 2.0.10), zaptel-source (>= 1:1.0.6-1), autotools-dev, libnewt-dev, libsqlite-dev, libspeex-dev
06:51.03opus__gentoo is for kids
06:51.19darylpI tried gentoo once...once is enough
06:51.23infinity1opus__: agreed
06:51.36tzafrirLet's remove dependecies that were added because of debian scripts, and inter-asterisk-packages
06:51.38infinity1tzafrir: hey. whats with your debian packages?
06:51.41DarthClueMath`: that's my point.  most of the people who want help with * don't know what they are doing.  most people don't want to waste time.  most people want instant gratification, therefore, i recommend a rh variant, either FC or CentOS which are known to work and work quickly with few if any complications.
06:52.09tzafririnfinity1, what do you mean?
06:52.13darylpfwiw I didn't have any real trouble compiling head on os x
06:52.17infinity1tzafrir: don't you make deb packages?
06:52.26tzafririnfinity1, I do
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06:52.33infinity1tzafrir: head?
06:52.36darylpa couple of libraries were necessary, same with ubuntu (debian)
06:52.39tzafrirnot yet
06:52.42darylpbut synaptic makes that childs play
06:52.51infinity1tzafrir: why not??
06:52.53infinity1:)
06:53.04infinity1tzafrir: i'm one of those instant gradification bitches :)
06:53.05tzafrirhead is still a moving target
06:53.16opus__darylp - i will beat you with my my wget && tar xvfz && make , skillz
06:53.16tzafrirthe relevant build dependencies:
06:53.18DarthCluedarylp: so i've heard.  i'm being encouraged by the storm troopers to switch to mac but i can't afford it at the moment.
06:53.20infinity1tzafrir: any plans to? like pakcage it once a week or something
06:53.50infinity1tzafrir: i don't follow it closely. it might move to fast for packaging once a week for all i know.
06:54.02tzafririnfinity1, no, I don't have the time for that.
06:54.13darylpmacs are expensive, I don't have a fast enough one which is why I'm not running * on it
06:54.13infinity1tzafrir: k. just checking.
06:54.16tzafrirI mean: I can automate the build process, but not the testing process
06:54.37darylpwell, the better half is hungry, and my problem can wait a little while
06:54.57hellopSo, whats the word on Outlook shared Calander replacement these days?
06:54.58darylpthanks for the help guys, I'm off to dennys and I'll attack it on a full stomach in a little bit
06:54.59DarthCluedarylp: just don't feed them after midnight...or get them wet.
06:55.04tzafririnfinity1, there debs of head, but I haven't checked them for a long time
06:55.04infinity1tzafrir: i think it would be nice to have like nightly packages if it was automated. just a thought. maybe i'm on crack.
06:55.24infinity1tzafrir: the only debs of head out there are from like dec or jan 04/05
06:55.28darylpgood advice darth, you wanna keep em on your good side
06:55.30tzafriranyway, soon we will have head packages in experimental
06:55.41infinity1debian experimental?
06:55.43tzafrirUnstable will still have 1.0 for a while
06:55.56infinity1tzafrir: that would be sweet.
06:58.19tzafririnfinity1, so do it yourself
06:58.27tzafrirthe scripts are mostly in place
06:58.41infinity1i made a few crude packages.
06:58.51infinity1very ugly.
06:59.23tzafririnfinity1, the pkg-voip svn repository has a script for building all of their packages from svn
06:59.33tzafrirHaven't tried setting it up yet
06:59.35*** part/#asterisk ryansc (user@c-24-9-254-252.hsd1.co.comcast.net)
07:00.17tzafrirBut it would be interesting. Contact me if you have any problems with it. http://svn.debian.org/wsvn/pkg-voip/
07:01.15infinity1my main goal right now is getting asterisk working in production.
07:01.32Math`infinity1: whats stopping you?
07:02.06infinity1Math`: well, i just got head install. i was runing 109
07:03.38infinity1where is the new syntax for head documented?
07:03.52infinity1thats the main reason i wanted to upgrade. i still haven't seen docs for it
07:04.07Math`what are u looking for
07:04.11Math`docs of what
07:04.43infinity1you know the replacement for using s,1 - s,2, etc. and use s,1 - s,n - s,n - etc.
07:05.14Math`uhu
07:05.18infinity1i've seen it used, haven't ran across docs
07:05.26Math`never saw that heh
07:05.47DarthClueinfinity1: did you check the wiki?
07:05.52*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net)
07:06.01infinity1DarthClue: yea. i searched around. didn't see it.
07:06.19empanyone have anything good/bad to say about teliax?
07:07.29infinity1DarthClue: you use it? did you learn it by reading other exmaples?
07:08.49infinity1oh shit. head has a color console.
07:09.00infinity1p.i.m.p. :)
07:09.02DarthClueinfinity1: no, i don't personally, but it should be pretty straightforward.
07:09.18DarthClueinfinity1: all the cool toys are in HEAD.  That's why most of us recommend it.
07:09.28*** join/#asterisk Myshenka (~gunde@217.9.101.85)
07:09.34infinity1what else do i get?
07:09.36infinity1heh
07:09.50DarthClueinfinity1: i can tell you what you don't get...the kitchen sink.
07:10.01infinity1uhh.
07:10.06infinity1the disposal?
07:11.25infinity1typing "stop now" is too much work. ctrl-d ??
07:12.03hellop"up arrow"
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07:12.16hellopinfinity1..
07:12.18DarthClueuse stop gracefully, it's much kinder.
07:12.20infinity1maybe
07:12.31infinity1DarthClue: even for devleopment
07:12.31infinity1?
07:12.51DarthClueyes, if there is nothing happening, then gracefully will stop just as quickly as stop now.
07:13.09hellopinfinity1, I type the up arrow button to go back to last command..
07:13.17infinity1lets see if the cli can interpret shortcuts
07:13.27infinity1ah boo
07:13.56DarthCluestop now is for when the drunken pulls on to the train tracks and you are traveling at 50 mph with a mile of cargo behind you.
07:14.02infinity1where did they get this cli? 3com netbuilder series? heh
07:14.24infinity1well, i like the color.
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07:34.19Newbie___hi , i am using h323, is there any way to know i am connected to a gatekeeper?
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07:38.29RoyKNewbie___: don't use h.323 with asterisk :P
07:39.04*** join/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com)
07:39.05Newbie___RoyK: i am more familiar with SIP, but the other end insist on h323
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07:40.12hellopAnyone got a Symbol MC50?  they look cool
07:40.15Newbie___looking at http://www.gnugk.org/
07:41.46tzafrirBTW: should ctrl-c on the cli (asterisk -r, not -c) save the history? any reason for it not to? (apart from "nobody has yet bothred")
07:41.47tzafrir?
07:42.12*** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122)
07:42.24TheEmperorhello
07:42.34TheEmperorcan someone tell me how to fix this? : Jul 12 15:33:05 NOTICE[1289744832]: rtp.c:275 process_rfc3389: RFC3389 support incomplete.  Turn off on client if possible
07:42.34TheEmperorJul 12 15:33:05 NOTICE[1289744832]: rtp.c:304 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256
07:45.54Math`change the dtmfmode
07:46.04Math`because your codec doesnt support rfc3389
07:46.07ZeeekTheEmperor IIRC the first has to do with transmitting silence which youneed to disable in X-Lite. Or VAD
07:46.29Zeeekwhat is the client?
07:46.36Math`ah 2833 is dtmf, nevermind my comment
07:46.50Zeeekinteresting while it lasted
07:47.25RoyKMath`: rfc3389 has nothing to do with dtmf
07:47.39RoyKTheEmperor: turn off silence suppression on the client
07:48.12brendaif a one minute conversation takes up 60k on my hard drive... how many minutes will fit on a theorectically empty 120gig hard drrive?
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07:49.35DarthCluebrenda...rough number here...but about 1.5 million minutes.
07:50.03brendawow!
07:50.16TheEmperori'm using x lite as well as a hard phone
07:50.39TheEmperordisable vad?
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07:51.24TheEmperorroyk: how to do i turn off silence supression on the client?
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07:51.39Zeeekwhat client?
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07:51.53TheEmperorZeeek: client means mine or the provider?
07:52.10brendaso about 25,000 hours
07:52.10Zeeekclient is the phone
07:52.10Zeeekwhat is it?
07:52.10TheEmperorit's an atcom 320
07:52.28Zeeekuncheck vad
07:52.59TheEmperorshould i leave agc and aeg checked?
07:53.21Zeeekaec is checked on mine so I authorize to leave it checked
07:53.28TheEmperorok
07:54.35TheEmperori put dtmf as rfc 2883, is that ok?
07:55.04Zeeekwhy change 50 things at once? It'll only make it harder to find the problem
07:55.16TheEmperorok
07:55.39TheEmperori was just wondering what the optimal settings were
07:56.01Zeeekdepends on what you're connecting to
07:56.06Zeeekthat's why there are choices
07:57.32ZeeekI wonder if I should have a coffee or install HEAD?
07:58.15DarthClueZeeek: install HEAD, you can get coffee while asterisk is building.
07:58.33Zeeekalready built - just a make install needed
07:58.42Zeeekit's an old head
07:58.53DarthClueZeeek: get new HEAD, then get coffee.
07:58.54ZeeekI'm scared to run head in production
07:59.02Zeeekis new HEAD good?
07:59.19Zeeekthis is June18th
07:59.25DarthClueZeeek: I'm pretty sure i saw bkw get HEAD from earlier today put it in production.
07:59.26Zeeekshit that's older than I thought
07:59.41DarthClueZeeek: you might be surprised how well head runs in production.
07:59.58ZeeekI know but I don't wanna be surprised by horrible problems
08:00.10Zeeek"better safe than sorry"
08:00.18DarthClueZeeek: then don't use asterisk.  asterisk is a horrible problem.
08:00.23TheEmperorstill get the same errors
08:00.29TheEmperorJul 12 15:51:27 NOTICE[1289744832]: rtp.c:275 process_rfc3389: RFC3389 support incomplete.  Turn off on client if possible
08:00.29TheEmperorJul 12 15:51:27 NOTICE[1289744832]: rtp.c:304 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256
08:00.50Zeeekno on the contrary, when you have it working and don't need any new features it is a mystery why I'm even considering updating! :)
08:01.11DarthClueZeeek: because you need to, everyone should run head.
08:01.15Zeeekhaha
08:01.22Zeeekeveryone should GET HEAD
08:01.24TheEmperorZeeek: any ideas?
08:01.29Zeeekbeyond that, I dunno
08:01.30DarthClueZeeek: that too.
08:01.48ZeeekTheEmperor what are yopu calling when you get that error?
08:02.01Zeeekjust you and your phone or a provider?
08:02.18TheEmperorZeeek: i am putting the call through to a provider
08:02.20Zeeekalso look that up on the mailing list it's a common message
08:02.21TheEmperorusing h323
08:02.25TheEmperorok
08:02.38ZeeekH323!!!! fuck. If I'da known that...
08:03.05TheEmperor?
08:03.15ZeeekI don't know anything about 323 at all
08:03.24TheEmperoro
08:03.32Zeeekmy advice is even more useless than usual
08:03.44TheEmperorhaha
08:03.46Zeeek"try reinstalling windows 95"
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08:03.49TheEmperorok thanks anyway :)
08:03.51Delvarhehe
08:03.58Zeeek"over XPPro"
08:03.59DarthClueZeeek: but he's running windows 1.0?
08:04.11RestLessGeminiHi All
08:04.15ZeeekI'm old enuf to have used WIndows 3
08:04.29ZeeekGod did windows SUCK BAD in those days
08:05.07DarthClueZeeek: I remember windows 1.0...And windows didn't suck, it created and vortex that destroyed everything that got near it.
08:05.09Zeeekmy TRS-80 was more stable (no gui tho)
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08:05.51Zeeekadding the 32k RAM was a rush :)
08:06.05Zeeekquite right
08:07.01Zeeekwhy do I have a module called "isdn" in lsmod ?
08:07.20djinHope someone can point me in the right direction for the following. We use Cisco 7940/7960's to connecto to *. Sometimes while calling another person the caller the call gets 'muted'. It seems to pickup again if when the called talks. In the mentime there is a very clear silence.
08:07.26RoyKZeeek: modinfo isdn?
08:07.47RoyKZeeek: using your favourite 'add everything there is' kernel distro?
08:07.56Zeeekbut I don't have anything related to ISDN on this box
08:08.09tuxinator_linuxMZeeek: It was crazy when I upgraded my ram from 512k to 2 Megs on my 286
08:08.33Zeeekthey wanted $12,000 to add 2 Megs to our miniVAX
08:08.57tuxinator_linuxMCost me several hundred
08:09.08tuxinator_linuxMDon't really miss those days
08:09.16ZeeekI think I remember 1 meg at $20 at one point
08:09.47tuxinator_linuxMthese were chips you had to stick on a daughter board
08:09.53Zeeekbut then my first printer cost $600 (no cable incl)
08:10.17tuxinator_linuxMI had a nice 24 pin
08:10.24justinki assume that wasnt a dot matrix for $600
08:10.41Zeeekso is there any point in removing drivers that are "used by 0" ?
08:10.50tuxinator_linuxMjustink: Evening
08:10.56ZeeekEpson MX80 dot matrix, slow as molasses
08:11.19justinktuxinator: morning to ya.
08:11.26tuxinator_linuxMZeeek: But you had some much fun with Print Shop
08:11.28Zeeekpardon my flood
08:11.29Zeeeki810_rng                2656   0  (unused)
08:11.29Zeeekhisax                 449444   0  (unused)
08:11.29Zeeekisdn                  117184   0  [hisax]
08:11.29Zeeekslhc                    5040   0  [isdn]
08:11.29Zeeekisa-pnp                30724   0  [hisax]
08:11.41Zeeeki810_rng                2656   0  (unused)any reason to care?
08:11.54Zeeekoops I went over the line there
08:12.02Zeeekhahah POrint SHop
08:12.30ZeeekI don't think there are any ISA slots
08:12.49Zeeekstill haven't chosen between HEAD and coffee
08:13.01tuxinator_linuxMZeeek: Sorry dude, a little late for that stuff
08:13.24tuxinator_linuxMZeeek: Eat a grapefruit
08:13.28Zeeekmoot point. I forgot to turn on the espresso machine :(
08:13.53ZeeekVoluto, anyone?
08:15.06tuxinator_linuxMVolu...who?
08:17.11RoyKZeeek: what is i810_rng?
08:17.12ZeeekIt's my favorite in the morning
08:17.30RoyKanyway - hisax is just the base driver. no card driver is loaded
08:17.36Zeeekrandom number generator
08:17.45RoyKah
08:17.57Zeeekstill hisaw is half a meg
08:18.12Zeeekno audio card, no usb, no scsii
08:18.15RoyKĂºsing 2.4 or 2.6?
08:18.23Zeeek2.4
08:18.33RoyKconfigure modutils to stop loading it, then
08:18.52RoyKisa-pnp might as well be unneeded :)
08:18.58ZeeekI wonder why ideèscsi says no? there is an ide
08:19.36Zeeekwell, if it works...
08:23.38*** join/#asterisk Vehemence (Gimp@bgp948716bgs.canton01.mi.comcast.net)
08:23.57*** join/#asterisk helpmeguys (~helpmeguy@222.253.87.143)
08:33.34*** join/#asterisk michael1234 (~mick@staff.lk.tsn.cc)
08:33.41*** join/#asterisk afv-13 (~NetCB@tbnb-165-208-126.telkomadsl.co.za)
08:33.50michael1234anyone in here know how to send a sip call to a billion
08:34.36djinmichael1234, not many here would be willing to support that.
08:34.38afv-13i seem to have a delay when the caller hits a key during a background, and when it moves to the next extension
08:34.52djinHope someone can point me in the right direction for the following. We use Cisco 7940/7960's to connecto to *. Sometimes while calling another person the caller the call gets 'muted'. It seems to pickup again if when the called talks. In the mentime there is a very clear silence.
08:36.03michael1234djin: I have my own asterisk box just trying to work out how to send a call to my registered billion box
08:36.51djin"billion box' ?
08:36.53*** join/#asterisk Newbie___ (me@211.24.146.11)
08:36.58fenlanderdjin: sounds like a problem with VAD. Asterisk needs a continuous rtp stream to clock the outgoing data
08:37.26fenlanderdjin: turn off any silence supression
08:37.50michael1234djin: http://www.billion.com.au/product/voip/bipac7402vl.php
08:37.57djinfenlander, I was looking into that, but does is require VAD=0 or 1?
08:38.17djinmichael1234, ah /me was thinging of voice spam ;)
08:38.28fenlanderdjin: I have vad: 0 in my SIPDefault.cnf
08:39.19fenlanderdjin, sorry, enable_vad: 0   :)
08:39.58djinfenlander, vad was disabled by default, so wasn't sure this would help.
08:41.00fenlanderdjin: might be worth checking that it is working by running ethereal on the asterisk box
08:41.13djinthere is no 'comfort noice' setting, is there?
08:41.27djinok, will try that.
08:42.53*** join/#asterisk florinm (~florin@81-178-45-7.dsl.pipex.com)
08:43.01florinmmorning boys
08:43.12florinmanyone heard about goldpbx?
08:44.03florinmno , the software :P
08:44.35florinmi found the goldbpx somewere and is fully asterisk, but all name changed
08:44.44florinmand changed the copyright info
08:44.51RoyKstrange
08:44.58RoyKgoogle didn't give any hints
08:45.00djinRoyK was a fan of Beavis & Butthead: "Fire Fire Fire Fire Fire !!!" :)
08:45.02florinmand i was wandering if this doesn't break the licence of the asterisk
08:45.14fenlanderMaybe they have a commercial license
08:45.40RoyKfenlander: i somehow doubt digium would allow people to re0wrap it
08:45.49RoyKflorinm: have you got the software somewhere?
08:45.53RoyKcan i have a look?
08:45.57florinmi didn't saw anywere saying about digium and asterisk
08:46.09Newbie___RoyK: what do u recommend if i were to to h323
08:46.16Delvarshot to the head
08:46.25RoyKNewbie___: dunno. yate might help
08:46.27florinmi need somewre to upload it
08:46.31RoyK~lart Delvar
08:46.32florinmas is 20 meg
08:46.37Delvar:)
08:46.40florinm(is allreadu v 4.0.1
08:47.25fenlanderflorinm: what is the license it came with?
08:48.04Newbie___RoyK: yate does SIP to h323 signaling
08:48.04florinmi saw no licence info on it
08:48.23florinmlet me check it :P
08:48.34RoyKNewbie___: I don't know if it does SIP any good
08:48.38RoyKNewbie___: perhaps chan_woomera
08:48.44RoyKNewbie___: that might be best
08:48.51RoyKthat's with asterisk
08:49.30Newbie___i am shit out of luck with oh323 and h323
08:50.06RoyKNewbie___: try chan_woomera
08:51.45Newbie___RoyK: installation seem simple enough
08:52.11*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com)
08:52.52Zeeeklo PoWeRKiLL
08:54.21*** join/#asterisk bogdanro (~gfdsxghcd@cnh.comtrust.ro)
08:54.28bogdanrohello
08:54.57RoyKhi
08:56.51Zeeekok now I'm in real trouble
08:57.01*** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com)
08:57.09bogdanrocan anybody here help me to setup a oh323 trunk on my asterisk ?
08:57.16limbiquehi
08:57.43limbiquei have found some wicked things on the asterisk
08:58.05limbiquewhen i call from external to a voip phone
08:58.32limbiquewhen i hangup the voip phone, the asterisk reserves an extra zap channel
08:58.38limbiqueanyone knows?
09:00.26*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
09:03.16limbiquekinda dead here?
09:04.12*** join/#asterisk elric (~m@ppp114-10.static.internode.on.net)
09:05.23djinfenlander, are you getting these "RFC3389 support incomplete." messages as well?
09:09.29fenlanderdjin: don't remember them
09:09.45fenlanderwhat is 3389?
09:11.13djinfenlander:
09:11.14djinWhat does this RFC3389 error message in the CLI/my log try to tell me?
09:11.14djinThis RFC refers to silence suppression. For example X-Lite has a configuration setting called "Transmit silence" where the behaviour can be controlled by the user. Anyway, Asterisk is complaining about silence supression. It's a harmless message, but if you can turn off silence suppression in your SIP client the message will go away. Related keywords are "VAD" and "comfort noise".
09:11.34*** part/#asterisk pif (ldm@zenon.apartia.fr)
09:11.35ZeeekI have a major problem: I can't seem to get back to STABLE.
09:11.58Zeeekhow is /etc/moidules.conf created? is it touched during install ?
09:13.24fenlanderZeek: I thought it just came out of configs/*.sample like the others
09:13.42fenlanderZeek: what is the error?
09:14.01Zeeektalking about the /etc/modules.conf file
09:14.02fenlanderdjin: must have some form of VAD still turned on somewhere?
09:14.11fenlanderZeek - oh
09:14.20ZeeekThe change from wcfxs to wctdm has fscked up something on my machine
09:14.39limbiquei have an some like, that asterisk reserves and immidiate release an extra channel when i hangup a external call from a voipphone
09:14.50limbiqueextra zap channel
09:15.09Zeeekastrisk is now dying at a random point during its startup
09:15.24Zeeekperhaps I need to delete all modules and start again
09:15.53*** join/#asterisk tengulre (~tengulre@61.185.238.166)
09:16.05djinfenlander, "Enable VAD NO"
09:16.55*** join/#asterisk indra (~indra_wat@microinfo.rain.fr)
09:17.15indraHello all
09:20.08*** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122)
09:20.11TheEmperorhi guys
09:20.17TheEmperoranyone know how to fix this?
09:20.20TheEmperorUnknown RTP codec 19 received
09:22.15fenlanderTheEmperor: sounds like confort noise - turn it off on your phone (what is it with comfort noise today?)
09:22.16*** join/#asterisk Jas_Williams (~Jason@host86-130-10-146.range86-130.btcentralplus.com)
09:22.52TheEmperorok
09:23.38fenlanderdjin: all I can say is that it works for me :-( You could try rebooting the phone I guess
09:23.50DelvarZeeek: yeah delete /modules dir and do a make install.
09:23.57*** join/#asterisk kajtzu (~kajtzu@212.226.212.95)
09:24.02kajtzuhmph
09:24.11ZeeekI just deleted the "wrong" stuff but screwed up I think on something else
09:24.23ZeeekI removed a reference to wctdm in /etc/modules.conf
09:24.45Zeeekasterisk is fine now but I bet if I reboot the drivers don't load right
09:25.04Delvaroh wait.. your talking about linux kernel modules?.. thought you were talking about asterisk
09:25.07Zeeeksomeone tell me about these post-install lines?
09:25.22ZeeekNO the modules I eliminated WERE asterisk
09:25.37Zeeekbut now for zaptel, I think I screwed up the /etc/modules.conf
09:25.38Delvarim confused!
09:25.57djinfenlander, thanks. I'll continue my search :)
09:25.59Zeeekthat's what they get for have two modules.conf files
09:26.03Delvareh its still morning and my brain isnt working
09:26.13Zeeekwhat do the post-install lines do?
09:26.28Zeeekand where did they come from?
09:26.31*** join/#asterisk jansaell (~jan@c80-216-185-161.cm-upc.chello.se)
09:26.56*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
09:26.57RoyKgoldenpbx from http://www.xoasisnetworks.com/ seems to be extremely similar to asteriskk
09:27.02puzzledmorning
09:27.48RoyKpuzzled: morning
09:28.35*** join/#asterisk helpmeguys (~helpmeguy@222.253.87.143)
09:29.27*** join/#asterisk lobbin (~lobbin@c83-253-149-45.bredband.comhem.se)
09:29.59Zeeekwhat is this:
09:30.00Zeeekzt_rbs: Tried to set RBS hook state 0 on channel WCTDM/0/0 while span WCTDM/0 lacks rbsbits or hooksig function
09:30.23Delvarhttp://www.xoasisnetworks.com/products/prodigy-voip/screenshots.php?img=ss-wop.png thats flash operator pannel :)
09:30.30Delvarmut be asterisk
09:30.45fenlanderRoyK: I don't see any goldenpbx - where is it on the site?
09:30.56Zeeekwhat is zt_rbs?
09:30.58djin2005-07-11 - Xoasis Networks offers 20% discount code on all IP PBX products to VoIP Info readers, coupon code: voip.
09:31.02djinvoip-info.org
09:32.59PoWeRKiLLSalut Zeeek
09:33.02RoyKfenlander: florinm got through some hole there and found the code
09:33.24Zeeekjour PoWeRKiLL
09:33.35Zeeekcat modules.conf
09:33.48Zeeekthat's where I am, typing in the wrong windo
09:34.19Zeeeksomeone help me with linux post-install lines ?
09:35.06TheEmperorRoyK: any idea on this? rtp.c:434 ast_rtp_read: Unknown RTP codec 19 received
09:37.58*** join/#asterisk mithro (~tim@195.177.247.8)
09:38.56RoyKfenlander: http://pastebin.ca/17529
09:39.36Zeeeklet me try another angle: I already have modprobes in rc.local for startup. WHat are the post-install lines in /etc/modules.conf ? They seem to have been added by my failed install of HEAD?
09:40.27puzzledRoyK: if that isn't * I don't know what is
09:41.52RoyKpuzzled: right
09:43.34fenlanderRTP codec 19 was for Comfort Noise in a draft spec. It is now reserved, but you will find some devices that still use it.
09:45.20RoyKpuzzled: seems this company creates 'embedded pbx systems'
09:45.40ZeeekRoyK what distro are you running?
09:45.58Zeeekno wait I need someone who has both X100 and TDM400 ?
09:47.44*** join/#asterisk outsidefactor (chrismarti@203-206-254-183.dyn.iinet.net.au)
09:48.01Newbie___i have a X100 and TE410
09:48.05ZeeekI see that zaptel Makefile writes those lines to /etc/modules.conf
09:48.21ZeeekHEAD apparently writes a lot more lines
09:48.39Zeeekpciradio etc
09:49.36puzzledZeeek: yup, check the Makefile. it's all in there
09:50.00Zeeekyeah but in the meantime I'd like to know what has to be in the file so I can reboot
09:50.02puzzledZeeek: alternatively you can just load them "manually" in /etc/rc.d/rc.local
09:50.11ZeeekI do that
09:50.15puzzledme too
09:50.24Zeeekso I don't need any of the modules.conf post-install et al?
09:50.46Zeeekit also looks like it added options
09:50.55Zeeekand what is torisa?
09:50.57puzzledI don't use it. just load them from rc.local. iirc cause I couldn't figure out the modprobe.conf stuff :)
09:51.04Zeeekand tor2 ?
09:51.23puzzledwhich card do you use?
09:51.23TheEmperorfenlander:how do i get rid of that?
09:51.37ZeeekX100P and TDM400P
09:51.42TheEmperorfenlander: the rtp 19 codec i mean, is that coming from my side or the otherside?
09:52.28puzzledZeeek: then I think you just need the wcfxo module for the X100P. dunno the module's name for the tdm400p
09:52.38Zeeekwcfxs
09:52.59ZeeekI have this in my rc.local: modprobe of zaptel, wcfxo, wcfxs ztcfg
09:53.24puzzledthink that should work
09:53.36*** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com)
09:53.45Zeeeksince they changed the tdm driver to wctdm (no more wcfxo)
09:58.50Zeeekwhat is torisa ? anyone ?
09:59.04puzzledi think those are the really old cards
10:00.00Zeeekthis backstage shit has me confused. No wonder I get so scared when I try to update
10:00.30puzzledjust make some good notes how you made it work like you want it to and keep those handy when updating
10:02.03Zeeekmuhahahaha notes! I'd have to find the notebook!
10:02.13Zeeekbefore and after
10:03.03*** join/#asterisk TimmiG (~TimmiG@83-65-129-46.paris-lodron.xdsl-line.inode.at)
10:03.13*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
10:03.42lobbinI'm having some problems with my sip phone, I can dial-in but I can't dial-up
10:05.14Zeeekplus delete all modules"what are the symptoms?
10:05.28Zeeekwhat are the symptoms?
10:05.40Zeeekasterisk -r
10:05.44TimmiGHi, I've a question, can asterisk use Advice of Charge from the ISDN side? And how I have to configure it. I use a zaphfc-card
10:06.11puzzledTimmiG: search the list archives
10:06.21TimmiGOK, I try
10:06.52Zeeeklobbin you have to tell us more about what happens and what you are trying to do
10:07.17RoyKhow are US laws with regard to GPL protection?
10:07.27Zeeekthey lock you up
10:07.52lobbinZeeek: it tries to make the call to my sip provider, but asterisk seems to send wrong information
10:08.41Zeeekhow can anyone know what the problem is if you don't describe it? Are you running @home or AMP by any chance?
10:09.44Zeeeklobbin need to see the dial line and the errors asterisk shows
10:09.48lobbinhttp://pastebin.com/311772
10:09.50Zeeek~pastebin
10:09.50jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
10:09.56Zeeekok you're ahead of me :)
10:09.59lobbin:)
10:10.17Zeeekwhat part of WRONG PASSWORD do you not get?
10:11.20lobbinIf you look at the end of that line, in the <sip:>, it seems to send asterisk as my username?
10:11.42Zeeekpaste you peer config
10:11.50Zeeeksans passwords of course
10:12.25Zeeekdigisip is the provider?
10:12.39lobbinhttp://pastebin.com/311774
10:12.41lobbinYeah
10:13.11Zeeekyou have any other SIP providers working?
10:13.34lobbinno, but it did work before I tried to put it trough asterisk
10:14.05Zeeekyou mean it works with just a phone?
10:14.35lobbinNo, I got a dlink 1120 that I'm using, I need to use it through asterisk as this pos doesn't work behind a nat
10:19.18Zeeekok I was interrupted for a moment
10:19.32Zeeeksomething is wrong with your username or password
10:20.19lobbinYeah, but what, it seems that asterisk is sending asterisk as my username?
10:20.48*** join/#asterisk ady (~root@202.5.145.13)
10:21.03lobbinYou can see the header here: http://pastebin.com/311776
10:21.09Zeeekno that's normal
10:21.31Zeeek<PROTECTED>
10:22.01lobbinok
10:22.38lobbinHmm, it does seem to send Proxy-Authorization further down
10:22.40PoWeRKiLLIs there a possibility to generate a random number without using agi in extension.conf ? (If i remember there vas a variable rand ?)
10:22.55ZeeekI think there may be in HEAD
10:23.44ZeeekPoWeRKiLL quel est ton prénom, je les confond avec les pseudos
10:24.00tzafrirPoWeRKiLL, I was looking for something similar recently
10:24.15*** join/#asterisk eldu (~damajor@tuxmania.org)
10:24.19Zeeekor it would be easy to write the app in C
10:24.24elduhello
10:24.38tzafrirI want to make a nice ivr-like context with all the silly sounds in asterisk-sounds
10:24.58Zeeekremind me not to call you for mission-critical info
10:25.38*** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122)
10:26.04tzafrirZeeek, asterisk is not exactly on-par with some "proffesional" PBX-s. But which PBX has a prompt "will you marry me?"
10:26.48lobbinZeeek: looking at the headers it seems like asterisk is sending correct headers, so why is the password "wrong"? Atleast the username is sent correctly
10:27.16Zeeeklobbin that "no record" message may mean you are trying to register with a wrong proxy
10:27.31elduis there a frontend to easily manage dialplans ?
10:27.58tzafrireldu, not exactly
10:28.00Zeeekthere are a few, but you won't hear a lot of people recommending them here
10:28.13elduhmmm
10:28.27tzafrirthere are some front ends for that, which will limit you with the choice of actions
10:28.43eldui saw that there r many web interfaces to manage sip/iax client but not for dialplans
10:28.57tzafrirOthers are basically glorified editors, and to use them you have to understand how the dialplan works
10:29.02eldutzafrir: that's what i saw
10:29.23elduok ok
10:29.31Zeeekthe manager allows adding extensons dynamically
10:29.36tzafrirtake your pick. If you want the full power, you'll have to understand how it works
10:30.15Zeeekthat's my problem with linux. I don,'t know anything about it
10:30.16elduZeeek: its a good idea but a global view of the dialplan is also a feature i need
10:30.22*** join/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net)
10:30.33MuppetMasterHello everyone.
10:30.40Zeeekyou could write something thatwould be like a FOP for dialplan - that's be cute
10:30.50tuxinator_linuxMNight guys
10:30.51eldutzafrir: no prob understanding what i did :)
10:31.09MuppetMasterI am having a real nerve wracking issue with a Sipura 3K and the latest CVS HEAD of Asterisk (as of yesterday).
10:31.12Zeeekbye tuxinator_linuxM
10:31.13elduZeeek: i'll plan to do something if i really need it ;)
10:31.38MuppetMasterI have logged the issue in the Voxilla forums for the Sipura users group.  http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&p=21329#21329
10:31.59*** join/#asterisk pelo (~tcs@adsl203-149-051.mclink.it)
10:32.26*** part/#asterisk florinm (~florin@81-178-45-7.dsl.pipex.com)
10:32.55tzafrirZeeek, things like AMP aim to do that. But the price is that they have to make some assumptions about the dialplan
10:33.06MuppetMasterIn essence, the Line 1 on my SPA3K registers, but not completely (somehow).  When I try to dial I get:  NOTICE[3209]: chan_sip.c:6208 check_user_full: From address missing 'sip:', using it anyway
10:33.14MuppetMasterAnd WARNING[3209]: chan_sip.c:5801 get_destination: Huh? Not a SIP header (3006)?
10:33.14ZeeekI've never seen AMP or @home
10:33.28MuppetMasterThing is that I have a Sipura 2K with the same configuration working without a problem.
10:33.38MuppetMasterAnd the PSTN line of the device registers without a problem.
10:33.42lobbinZeeek: I guess that something is not correct in my settings then?
10:34.02Zeeeklobbin sometimes providers have a proxy system with two different servers
10:34.13ZeeekI have one like that, it's a pain to set up
10:34.47Zeeekare you registering tothis provider?
10:35.05lobbinZeeek: Yes, register to the provider works including incoming calls
10:35.25Zeeekyou see, outgoing may require a different ip
10:35.38Zeeekwere there no instructions for configuring?
10:35.46Zeeekheh maybe not with asterisk
10:36.10lobbinI have instructions for asterisk and I followed them and ended up with this :)
10:36.33Zeeekis there always the same ip or domain name inthe instructions or more than one?
10:36.42lobbinit's always the same
10:36.58*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
10:37.18MuppetMasterI am also getting this:  Jul 12 12:32:43 NOTICE[23730]: chan_sip.c:5675 register_verify: Invalid to address: '3006' from my_ip (missing sip:) trying to use anyway...
10:38.18Zeeeklobbin it appears right - you sure there isn't a typo in user or pass? Perhaps you want to limit codecs? disallow=all allow=ulaw or sth
10:40.04Zeeekno it's clear that it doesn't like the pasword
10:40.32Zeeekyou have a username AND a number ? Maybe you're supposed to log in with the number?
10:42.46lobbinyea, I have a username and a number
10:43.02lobbinI can try, but the the userguide says it should be the username
10:47.17PoWeRKiLLZeeek pv
10:47.31PoWeRKiLLtzafrir and you find something ?
10:48.37ZeeekPoWeRKiLL I'm going to lunch - back in 30 or so min
10:48.55PoWeRKiLLok
10:48.59PoWeRKiLLBon appetit
10:49.11*** join/#asterisk chezgi_ (~user@217.219.15.242)
10:49.20lobbinZeeek: no luck changing the criterias
10:49.26Zeeekc olivier? je suis perdu entre les pseudos
10:49.47Zeeeklobbin, if it makes you feel any better, I'm having major linux stupidity (mine) here
10:50.01lobbinit does :)
10:50.28Zeeeklobbin try logging in with a phone?
10:52.27Zeeekwhat is zt_rbs ?
10:53.27Zeeekaha
10:53.43ZeeekI can't get rid of the damn calls to wctdm - well lunch for now
10:54.50MuppetMasterAlso, I can see the address is missing in the Asterisk internal Database as seen from the CLI:
10:55.04MuppetMasterWhat would cause the SIP URI to be missing in the Asterisk database as per 3006?
10:55.13MuppetMasterWhile 3000 from the same Sipura 3000 contains the SIP URI.
10:55.25lobbinI'm out for lunch now as well
10:56.28*** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl)
10:56.43onkeltimmhelay
10:57.18onkeltimmif i do an attended transfer, and the person i want to transfer to does not pick up, how do i get the call back?
10:57.41RoyKhm... now if digium was to license asterisk to someone
10:58.08RoyKwould they also allow that company to replace all references to digium/asterisk/developers by their own people?
10:58.32RoyKand rename the suite
10:58.37RoyKand set a new version on it
11:00.35*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
11:02.04*** join/#asterisk goldenear (~goldenear@m209.net81-64-245.noos.fr)
11:02.46MuppetMasterRoyK You will most likely need to sign an NDA with Digium and ask them what their licensing terms are.
11:02.58RoyKhttp://www.xoasisnetworks.com/
11:03.02RoyKsee the demo
11:03.08MuppetMasterRoyK If they are willing to do a true OEM with the license than yes.  But maybe they are not.  No one may answer that but Digium themselves.
11:04.01*** join/#asterisk alphaque (~alphaque@60.48.197.184)
11:04.40kd5uzzis there any software out that would allow a pocketPC to connect to an asterisk server as a voip phone?
11:04.57goldenearI wonder, is adding "exten => h,1,Hangup" in a dial plan useful ?
11:04.58tzafrirRoyK, did you try contacting the company ans asking them before posting on a public mailing list?
11:05.05goldenearkd5uzz, try ziax
11:05.13kd5uzzthanks
11:05.30goldenearif you run linux on your PDA
11:05.39RoyKtzafrir: nope... didn't think about it. i was just given the info i posted. should have done so, sorry
11:05.53tzafrirBTW: where can I get this libpri 4.0.1? these guys seem to live well ahead of us
11:05.58goldenearelse you can try something comercial from xten
11:06.22kd5uzzdoh..k
11:06.31goldenearkd5uzz, make a search at voip-info.org
11:06.31*** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com)
11:06.54Zeeeksomebody gotta help me wit this  zt_rbs problem
11:06.58*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
11:09.11MuppetMasterkd5uzz:  SJPhone runs on PocketPC:  http://www.sjlabs.com/products.html
11:09.18MuppetMasterXten might too, just not sure.
11:11.04goldenearhttp://www.xten.com/index.php?menu=products&smenu=xproppc
11:11.19*** join/#asterisk Morex (~blah@host81-157-165-204.range81-157.btcentralplus.com)
11:11.23MorexHi there
11:11.34goldenearhi Morex
11:11.35MorexAnybody else having trouble logging on to cvs.digium.org?
11:11.43Morex'Sup Goldenear
11:12.12fenlandercvs is good for me
11:12.16MorexHuh...
11:12.18MorexOK thanks
11:12.32*** part/#asterisk vuvie (~vuvie@bb219-74-45-31.singnet.com.sg)
11:13.50goldenearCould you help me please I have 2 (existential ?) questions about dial plan / extensions.conf :
11:14.19goldenear1) is exten=>h,1,Hangup necessary or is it implicit ?
11:14.53ManxPowergoldenear: You NEVER need exten => h,1,Hangup.
11:15.01goldenear2) when exactly it's necessary or not to use the "r" option of Dial ?
11:15.07ManxPowerExtension h is only called when the call has ALREADY hungup.
11:15.14ZeeekManxPower when you make zaptel, where does it install modules?
11:15.16ManxPowergoldenear: you almost never need "r" option to dial.
11:16.16goldeneardoes "r" really  kills call progress information
11:16.20goldenear?
11:16.44ManxPowergoldenear: Yes.
11:16.49goldenearok
11:16.59ManxPowerwhich is why you almost never need it.
11:17.29goldenearI was confused because I found a lot of extensions.conf examples using "r" all the time ;)
11:17.42ManxPowergoldenear: they are written by newbies
11:18.18goldenearI guess...
11:19.52goldenearalso, I guess Hangup is needed at the last priority for each extensions, isn't it ?
11:20.10goldenearexten => 10,1,Dial(IAX2/user)
11:20.21goldenearexten => 10,2,Hangup
11:21.45ManxPowergoldenear: Most of the time Asterisk will do the right thing, but it's still a good idea to have Hangup in that case.
11:22.05goldenearI notice that, when Hangup is not used and the called part is not registred, Calling part needs more time to hangup
11:22.54goldenearand with iaxclient based client, it will just ring ring ring infinitly :)
11:23.01*** join/#asterisk albators (~albaneagr@81-178-45-7.dsl.pipex.com)
11:23.26goldenearThat's a good idea indeed :)
11:23.49goldenearand what about a Hangup at priority n+101 ?
11:23.53Zeeekthis must be a job for bkw_
11:24.14*** join/#asterisk zeitgeist_y2k (~ceicke@mail.inmedias.it)
11:24.27goldenearexten => 10,102,Hangup
11:25.07goldenearor exten => 10,102,Busy (but I guess this is not necessary as it is implicit, right ?)
11:25.40Zeeekok, one last attempt at linux question: the files in /lib/modules/2.4.22/misc
11:25.52Zeeekare they autoloaded (like asterisk modules)?
11:26.10*** part/#asterisk MuppetMaster (~MuppetMas@177.Red-213-98-135.pooles.rima-tde.net)
11:27.01goldenearManxPower, so what do you think about Hangup or Busy at n+101 ?
11:27.48ManxPowergoldenear: If you use Busy, you don't need hangup.
11:29.05goldenearso is exten => 10,2,Hangup a good idea ? or is it just not necessary ?
11:29.43MorexDoes anybody hear use FastAGI?
11:29.47Morexhere even?
11:30.07*** join/#asterisk Twister (~jason@pool-151-205-68-214.char.east.verizon.net)
11:30.18goldenearManxPower, I guess it's killing Busy indication, so it's should be avoided...
11:35.35goldenearThank you ManxPower for all this information. Now my dial plan should be cleaner :)
11:35.58ZeeekManxPower you run HEAD or STABLE? (i seemto remember STABLE)
11:35.58TwisterYAY!!! *jumps for joy*
11:37.47zeitgeist_y2kcan anyone help me with compilation problems of res_mysql.so in combination with the bristuff package?
11:38.05ManxPowergoldenear: no, Hangup as the priority after Busy or Congestion will never be called, since Busy does not exit until the caller hangs up.
11:38.13zeitgeist_y2ksorry... it's res_config_mysql.so
11:38.20ManxPowerZeeek: STABLE only.
11:38.45Zeeeksee my speech in #asterisk-stable
11:39.57*** join/#asterisk pa (~Paolo@pa.user)
11:48.38*** part/#asterisk pelo (~tcs@adsl203-149-051.mclink.it)
11:53.47alphaquegoldenear: yes, because the Dial cmd has no timeouts
11:54.30*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
11:56.30limbiquehi
11:56.42limbiquedoes anyone knows what i can do with AGI?
11:57.35limbiqueAnd what is "zap show cadences"?
11:58.42limbique<PROTECTED>
11:58.42limbiqueJul 12 13:58:34 WARNING[229390]: chan_zap.c:2022 pri_find_dchan: No D-channels available!  Using Primary on channel anyway 3!
11:58.42limbique<PROTECTED>
11:58.43*** part/#asterisk RestLessGemini (~root@202.142.182.154)
11:58.45limbique?
11:58.48*** join/#asterisk krazykrab (~Krab@203.81.238.65)
11:59.01krazykrabhi
11:59.17krazykrabany one can help me out with Asterisk AAH
11:59.51zeitgeist_y2kkrazykrab, shoot... maybe i can help
11:59.56krazykrabthanks
12:00.14krazykrabi just installed the AAH . and have the broadvoice connection to test
12:00.16ManxPowerlimbique: Did you have a specific question?
12:00.29limbiquehi
12:00.31krazykrabbut when ever i call .. in my panel it shows incoming
12:00.54limbiqueum, yes i have many questions.. :)
12:01.05krazykrabi have followed the instruction but didnt help it
12:01.05limbiquei'm a developer
12:01.18zeitgeist_y2kkrazykrab, i have no idea about asterisk@home
12:01.25krazykraboh ok
12:01.29ManxPowerI can't really help with Flash Operator Panel, limbique
12:02.30krazykrabdoes any one knows any site addy for asterisk@home manual
12:02.56ManxPowerlimbique: Do NOT message me unless you have confidential information that cannot be put on the channel.
12:03.09limbiqueok, sorry
12:03.56krazykrabManxPower can you help me on Asterish@home
12:04.39limbiquei have something like this. when ik call from external to a voipphone, when i disconnect the call, asterisk reserves an new zap channel and releases it immidiately..
12:04.44limbiquedo you know why?
12:05.16Zeeekfucking wengo changed their server domain again without warning! </rant>
12:06.20limbiqueIs there a way to get information from a manager connection, to get all active links? (not the connections but who's connected to who)
12:06.38ManxPowerkrazykrab: no.
12:06.51ManxPowerlimbique: I cannot help you with developement questions
12:07.13ManxPowerlimbique: look at how Flash Operator Panel does things.
12:07.16limbiqueManxPower: tnx.. what do you do? installing etc?
12:07.35ManxPowerlimbique: I do network and PBX management, installation, and service for my clients
12:07.47limbiquei'll give a watch on Flash Operator Panel
12:07.53limbiqueis that a windows application?
12:07.59ManxPowerNo.
12:09.52Zeeekpart of it is :)
12:10.02limbiquea, perl
12:10.18ManxPowerZeeek: No, the GUI part is Flash.
12:10.32Zeeekwhich does run on WIndoze
12:10.38ManxPowerThe GUI works under Linux
12:10.39ZeeekFOP is a great tool
12:10.57Zeeekbecause it does run on anything Flash runs on
12:12.42goldenearOk, thank you ManxPower :)
12:13.03goldenearalphaque, what do you mean ?
12:16.22tzangerwell
12:16.24tzangerthere goes the cat
12:16.43tzangercame home at about 6:45am, went out at 7:45am, cat was on the road between that time
12:17.14PrionAnyone having any issues with Voicepulse connect?
12:17.34lobbinback..
12:18.40limbiqueManxPower: ok, tnx.. i'm a noob with linux etc...
12:19.28Zeeeklobbin any progress?
12:21.51alphaquegoldenear: sorry. meant that without a timeout, no call progress is being handled by Dial, so it continues forever
12:22.39*** join/#asterisk CdtDelta_PM (~CdtDelta_@dsl081-225-161.chi1.dsl.speakeasy.net)
12:22.52limbiquebut is generates flash... thats cool
12:24.43*** join/#asterisk FreezeS (~gido_b@83.103.170.130)
12:24.50FreezeShi guys
12:25.02FreezeSdo you know how can I route a call based on the destination number ?
12:25.46FreezeSI know I can route based on the calling number ( exten => number/s,1,Dial...)
12:25.59FreezeSbut how can I route based on the number that is called ?
12:26.01*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
12:27.45FreezeSanyone ?
12:30.10albatorsyes
12:30.29albatorsextren => numberdialed,1, .....
12:33.54lobbinZeeek: yea, right now, I added "fromuser=" and then my username, after that it send that instead of asterisk and now it works
12:34.02Zeeekgreat!
12:34.07lobbinYeah :)
12:34.24lobbinCaller ID seems fubar, but I guess that isn't Asterisk's fault?
12:38.42*** join/#asterisk lodeon (~not4u@as1-6-4.ld.bonet.se)
12:39.28goldenearalphaque, of course I know it ... I just forgot the time out in my example ( exten => 10,1,Dial(IAX2/user,20) )
12:40.35goldenearWhat I did not know is if 10,102,Busy was implicit or not ...
12:41.16FreezeSso, anyone can help me with my problem ?
12:42.41ManxPowerFreezeS: exten => _31XX,1,Whatever will route all calles to any 4 digit extension beginning with "31" to the application Whatever
12:42.42*** join/#asterisk razu (~razu@80-235-89-85-dsl.prn.estpak.ee)
12:44.23goldenearFreezeS, indeed it's just as simple as ManxPower says ...
12:44.32tzangerManxPower: I heard that Whatever() app was kind of dangerous
12:44.46tzangerit turned the temperature on my freezer way up and made my ice cream go all melty
12:45.25*** join/#asterisk toot (chris@212.20.250.186)
12:45.31darwin35anyone having issues dialing interal sip extensions
12:45.55goldenearwhat kind of issues ?
12:45.59darwin35my mapped exten work fine but when I dial sip/1001 or 1002 I get a 404 on my phone
12:46.02ManxPowertzafrir: It sometimes does that.  The developers have not personally experienced this problem so they always close bug reports about it.
12:46.15tzangerManxPower: :-)
12:46.42tzangerhee hee hee
12:46.46tzangerthat word always makes me giggle
12:46.50tzangertoot
12:46.52tzangerhee hee hee
12:46.55*** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net)
12:47.19lobbinZeeek: never mind, I solved the caller id bit as well
12:47.52Zeeek$*%%*$$%%£££!!!
12:48.27Zeeek^^^^^ read with Donald Duck voice
12:48.33goldeneardarwin35, are you sure 1001 and 1002 are well defined in sip.conf ?
12:48.41darwin35yes
12:49.03goldenearcan you see them with sip show peers ?
12:49.08ManxPowerA 404 indicates the phone is getting the request.
12:49.27ManxPowerdarwin35: Try Dial(SIP/1001@1001)
12:49.40ManxPowerYou should not need that, but it can't hurt to try.
12:49.59*** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
12:50.03darwin351001/1001                  10.0.0.2         D          255.255.255.255  5060     Unmonitored
12:50.10FreezeSmy problem is this: I have an ISDN line with multiple numbers. I want to allocate each number to a SIP phone. This way, there can only be 2 simultaneous calls, but each user can be called with a different nummber, from the exterior
12:50.22darwin351002 is offline at the min
12:50.40*** part/#asterisk alphaque (~alphaque@60.48.197.184)
12:51.03*** join/#asterisk alphaque (~alphaque@60.48.197.184)
12:51.28FreezeSI've discovered in the documentation ${DIALEDPEERNUMBER} .. how can I route based on this variable ?
12:52.01ManxPowerFreezeS: Incoming calls that haved the dialed number associated with them will match an extension line.
12:52.28tzangerFreezeS: Goto(somecontext,${DIALEDPEERNUMBER},1) ??
12:52.33ManxPowerFor example, my provider sends me the entire number dialed when a call comes in.  So I have exten => 5045551212,1,Goto(1212,1)
12:52.50FreezeSManxPower: super, thanx
12:52.54FreezeSthat's what I needed
12:53.05*** part/#asterisk alphaque (~alphaque@60.48.197.184)
12:53.08*** join/#asterisk alphaque (~alphaque@60.48.197.184)
12:53.13*** part/#asterisk alphaque (~alphaque@60.48.197.184)
12:53.38*** join/#asterisk alphaque (~alphaque@60.48.197.184)
12:53.43goldeneardarwin35, I did the test, I only have 404 when the extension dialed doesn't exist ...
12:53.44ManxPowerFreezeS: The telco may send 1 digit, 4 digits, or some other number of digits, depending how your phone company has the line set up.
12:54.39darwin35hmm
12:54.53darwin35all the mapped exten work fine
12:55.20goldenearso you sould not have a 404 :/
12:56.01goldenearalso are you in the right context ?
12:57.44Kattymew
12:57.53darwin35re reading hold a min
12:58.17Kattyis it morning?
12:58.45kajtzusup
12:58.50lobbinZeeek: thanks for the help
12:59.12Zeeeknp
13:00.50*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
13:01.09KattyZeeek: my new card came in today :>
13:01.35Kattyoh.
13:01.37Kattyno
13:01.38Kattyi take it back
13:01.45Kattyk, all better
13:02.32*** join/#asterisk albators (~albaneagr@81-178-45-7.dsl.pipex.com)
13:02.41Zeeeknew credit card?
13:02.53Kattyno
13:02.58Kattythe 4 port card for another 4 phone lines (=
13:05.35ManxPowerIt was for your old boss.
13:06.30DarthClueYesterday was the last day at my other job.  Problem being, the head man didn't show up so I couldn't tell him why and what my plans are.  He is freaking out because they are about replace the entire phone system / dialer and I was heading up the project.  I guess he doesn't realize that if ya don't pay me I won't stick around.
13:06.43iCEBrkrOk, so I spent about 5hrs hacking up the EAGI example and res_agi last night and I still can't figure out how to 'listen' to the audio.
13:06.44*** join/#asterisk toot (toot@212.20.250.187)
13:07.09tootsob - i can't reach the power for my new TDM400P
13:07.14iCEBrkrDarthClue: Yeah! Bend 'the man' over.. :D
13:07.25KattyZeeek: hmm..when i put that card in
13:07.31KattyZeeek: and define my channels
13:07.39KattyZeeek: do i have two sets of 1-4 or does it just become 1-8?
13:07.45DarthClueKatty: 1-8
13:07.59KattyDarthClue: and the second card? how does it know which card is number 2?
13:08.31Kattyports 7 and 8 i want in a different group
13:08.47DarthClueKatty: it depends on how they are detected on boot.  Worst case scenario, you have to move the lines that are plugged into the cards or move the cards.
13:08.52Kattyhopefully it won't think 7 and 8 are somehow card one, and not card two
13:09.22KattyDarthClue: k
13:09.32KattyDarthClue: anything special about putting that card in?
13:09.43DarthClueKatty: configure them and see what it thinks, if it thinks card 1 is 5-8 then just move the lines.
13:09.45Kattyshouldn't have to load drivers twice, surely
13:10.07DarthClueKatty: should use the same drivers as the other card.
13:10.11Kattyk
13:10.14Kattyjust...put the card in?
13:10.51DarthClueKatty: turn the box off first...and ya know, maybe a few other things to ensure that you don't fry it.
13:10.57Katty;>
13:10.59Kattybutbutbut
13:11.07Kattyi wanna work with the computer /on/!
13:11.30DarthClueKatty: That procedure isn't supported and will void your warranty and terminate your support contract.
13:11.48*** join/#asterisk zoa (~zoa@pirus.securax.be)
13:11.49Hmmhesaysheh
13:12.16ManxPowerIf you plug a phone line into a Digium FXS port, the FXS module will be destroyed when the first call comes in
13:12.44DarthClueManxPower: you serious?
13:12.53kajtzuManxPower: never happened on a cisco fxo/fxs interface
13:13.00KattyDarthClue: i /am/ the support contract ;)
13:13.04JerJerDarthClue:  as a heart attack
13:13.07kajtzu(and people plug them wrong all the time)
13:13.18JerJerkajtzu:  then it never rang
13:13.28ManxPowerDarthClue: Yes, I am serious
13:13.33zoaany bri experts here ?
13:13.48Hmmhesaysmorning Katty :)
13:13.50ManxPowerThe Cisco may be designed to not have that problem.
13:13.55KattyDarthClue: i'm going to put the card in over lunch (=
13:13.59DarthClueManxPower: k.  Seems like a design flaw.
13:14.08JerJerno i've seen cisco FXS ports smoke on ring of an FXO line
13:14.11darwin35ok I did a asterisk -vvvvvvvvvgc and get no errors or warnings but still cant dial local sip exten
13:14.14KattyDarthClue: someone mentioned something about going into bios and making sure they were on different interrupts or something
13:14.22darwin35this pisses me off
13:14.37KattyDarthClue: should i disregard that?
13:14.50*** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net)
13:15.09Kattyhi ariel_ (=
13:15.12darwin35I would first move the cards around before messing with the bios
13:15.16yxadarwin35 sometimes you need to reboot :)
13:15.25darwin35I have 100 times
13:15.26ariel_Katty, morning hope your dong well today.
13:15.28DarthClueKatty: yes, irq settings may cause a conflict.  as long as the devices are on different irqs it shouldn't be an issue.  what is the motherboard on the system?  maybe the mobo docs will say which slots share irqs.
13:15.31Kattyariel_: yes thanks (=
13:15.35goldeneardarwin35, what does asterisk CLI tell you when you call a local sip phone ?
13:16.10goldenearwhat the log says ?
13:16.15KattyDarthClue: k, i'll go look a little later
13:16.42*** join/#asterisk Mike9 (~sturdee@ireland.pathwaynet.com)
13:17.28iCEBrkrdarwin35: Dude, haven't you gone to bed  yet?
13:18.01DarthClueiCEBrkr: it's 815am, why would anyone want to be in bed now?
13:18.52iCEBrkrDarthClue: I just recall him being fairly active in channel last night.
13:19.00iCEBrkrJust seemed like he hasn't stopped. :D
13:19.29DarthClueiCEBrkr: yeah well, some of just don't sleep normal.  and that includes me.
13:19.40*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:19.47iCEBrkrSame here, but I had to call it a night early last night. 12:30 and I was passed out.
13:19.48darwin35http://pastebin.ca/17536 there is a sip debug
13:20.21Kattyyawn.
13:20.22RoyKanyone here that knows how i can change the audio file for a particular call to Authenticate?
13:20.40*** join/#asterisk crash3m (crash3m@crash3m.user)
13:20.55DarthClueRoyK: audio file Authentication?
13:21.15RoyKDarthClue: huh?
13:21.15iCEBrkrDarthClue: Sun and Mon were 3am nights for me.  I couldn't walk away from the keybaord.
13:21.41iCEBrkrRoyK: Yea, what do you mean by audio file authentication?!
13:21.53DarthClueiCEBrkr: Last week was hell.  A couple of days in row where I was up for 24 hours straight, crashed for 6, rinse and repeat.
13:22.07iCEBrkrDarthClue: ouch
13:22.08RoyKi mean i call Authenticate in asterisk and that plays off some file
13:22.16RoyKi want it to play my own voice in my own language
13:22.27goldeneardarwin35, and could you show the general asterisk log, not the sip debug ?
13:22.28RoyKone per purpose
13:22.42darwin35the log is empty
13:22.46KattyiCEBrkr: thx, all better
13:22.53iCEBrkrKatty: :D
13:22.54darwin35no errors or warnings
13:23.17Kattywill ssh tunneling work for softphones?
13:23.40goldeneardarwin35, but does it execute the correct commands ?
13:23.46Kattyconnect to an ssh through a firewall while specificying localport:destination:anotherport
13:23.48iCEBrkrDarthClue: I've been hacking away at this EAGI audio crap for the past 3 days.  I still don't understand what it's (not) doing.  I'd like to eventually get sphinx integrated.  I have sphinx working without a hitch.
13:23.56DarthClueRoyK, how about using Read / Playback instead?
13:24.01Kattyand then connect to localhost on port like 1234
13:24.09Kattyand have if forward to destination:whateverport
13:24.19iCEBrkrKatty: I've heard it's been done, but I'm not so sure it's 'efficient'.
13:24.20darwin35wich command ?
13:24.34KattyiCEBrkr: why?
13:24.40darwin35its registering all the extensions and every thing else works
13:24.47zeitgeist_y2kKatty, you need more than one TCP channel for SIP connections
13:24.51darwin35inbound from outdide works
13:24.54zeitgeist_y2ksome UDP too
13:24.56Kattyzeitgeist_y2k: yes, but not iax (=
13:25.00iCEBrkrKatty: I didn't the follow the conversation that closely.  :)
13:25.04darwin35all the mapped exten work
13:25.07Kattyzeitgeist_y2k: iax only uses one port :)
13:25.11JerJerasterisk doesn't deal with TCP based SIP
13:25.14zeitgeist_y2kKatty, I should read before writing :-)
13:25.25Kattyyeah, it's udp
13:25.31JerJeryou need a GRE tunnel
13:25.42Kattyis that something like a green tunnel?
13:25.45Kattyor a subway tunnel?
13:25.57iCEBrkrKatty: There was someone in here all concerned about someone 'listening in' on their VoIP calls. I kinda laughed.
13:25.57JerJeror other udp encapsulated tunneling method
13:26.02DarthClueKatty: VPN
13:26.06KattyDarthClue: oh
13:26.12JerJerVPN is a generic term
13:26.20KattyJerJer: yes, but i understand vpn
13:26.34DarthCluejerjer: it may be generic, but it is understood by many.
13:26.36JerJerbut you cannot use just any VPN and expect VoIP to work
13:26.37darwin35this makes no sense
13:26.44KattyJerJer: hahahahhahahaha
13:26.49KattyJerJer: HAHAHAHA, kthx it works
13:26.54KattyJerJer: on DIALUP even
13:27.05goldeneardarwin35, what is the context for inbound calls ?
13:27.13zoadoes somebody know if a bri card can be used to connect asterisk to a pbx ?
13:27.20DarthClueok...PPTPD or IPSEC VPN which uses GRE, etc, etc, etc
13:27.26zoacan asterisk play the net side on a bri ?
13:27.34kajtzuyou dont need gre with ipsec
13:27.41Kattythe Dark side of a bri
13:27.57kajtzuin fact, gre is completely unnecessary unless you intend to run a routing protocol on top of ipsec :)
13:28.06RoyKis the asterisk db persistent between restarts?
13:28.22DarthCluekajtzu: yeah, i know, but it's too early to sift thru the mess in my head.
13:28.36goldeneardarwin35, the asterisk log (with -vvvvvv) should show you why you have a 404
13:28.42KattyDarthClue: it needs a shower first
13:28.46JerJeryou don't want IPSEC - adds too much latency
13:28.52*** part/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com)
13:28.55darwin35http://pastebin.ca/17538 there is the sip 101 entry
13:28.55iCEBrkrOh? Sweet. NuFone offers 800 DID's?
13:28.57zoaipsec could do udp
13:29.01zoaso that doesnt have to be a problem
13:29.23DarthClueKatty: true, but I don't have to leave the house till about noon so I'm in no hurry.
13:29.41kajtzuJerJer: uh?
13:29.51KattyDarthClue: k
13:30.00Kattymy hair gets cut today :<
13:30.02Kattyi'm skeered bad
13:30.07iCEBrkrDarthClue: must be nice.
13:30.20*** join/#asterisk wiseguy_ (chivilis@85.206.10.242)
13:30.21tootcan i do anything with the TDM400P without the power supply hooked in?
13:30.24DarthClueiCEBrkr: it is.  one of the perks of working for the dark side.
13:30.24wiseguy_hellow people
13:30.31tootits  a public holiday over here and i won't be able to get one for a few days
13:30.33Kattynoooooooooo
13:30.37iCEBrkrDarthClue: hehe
13:30.38wiseguy_anyone have done a calling center with asterisk?
13:30.40JerJereverytime i've gone into a network that used an IPSEC tunnel and tried to use VoIP the latency was very noticeable
13:30.57JerJertoot:  no
13:31.02kajtzuipsec itself doesn't add much to latency
13:31.04JerJeryou must connect it to power
13:31.11DarthCluewiseguy_: yes, why?
13:31.13tootsob
13:31.14darwin35Jul 12 08:10:07 WARNING[54145] res_indications.c: Invalid ringcadance given '' at line 62.
13:31.14darwin35Jul 12 08:10:08 NOTICE[54145] res_indications.c: Duplicate entry 'stutter', skipped.
13:31.15iCEBrkrI need to get a haircut myself. I'm starting to get that 70's bed-head shag..
13:31.17JerJerkajtzu:  30-40 ms
13:31.17kajtzuwe're talking sub millisecond range
13:31.20kajtzuJerJer: nooo
13:31.20darwin35thasts new
13:31.25wiseguy_DarthClue: maybe pm
13:31.44darwin35thats the only error I have
13:31.57JerJerkajtzu:  sub ms - only if the network has hardware assisted ipsec
13:32.03goldeneardarwin35, and what's the log says when you call the sip phone ?
13:32.14goldenear(errors or not)
13:32.29kajtzuJerJer: of course it would have. what's the point of designing a ipsec network and then utilizing underpowered cpus? :)
13:32.53JerJersmall businesses are not always going to invest in the latest and greatest technology
13:33.00JerJeror home users
13:33.03clive-zoa yes klaus's cards can do NT mode
13:33.08kajtzuHiFn chips came on the market in 1998
13:33.17zoaso NT mode is what i need ?
13:33.22zoai dont know a thing about NT mode
13:33.25kajtzuand the cost of the add on module or chip isnt much
13:33.26zoaor BRI in general
13:33.26clive-zoa yup
13:33.33zoak do you have any docs on this ?
13:33.37zoaor links ?
13:33.40darwin35the log says nothing
13:33.44JerJeri use a very trivial GRE tunnel and VoIP works flawlessly
13:33.52*** join/#asterisk zpn (~xpn@207.111.174.1)
13:33.55darwin35and I pastebin the sipdebug
13:33.56tootanyone know where i can buy a Molex Power extension cable in the uk? i'm struggling to find one
13:34.10tooti imagine other people have had the same issues
13:34.12JerJerRadio Shack UK
13:34.13JerJer:)
13:34.16iCEBrkrJerJer: hehe
13:34.19kajtzuJerJer: it doesn't protect the contents either. wasnt this thread about protecting
13:34.22iCEBrkrRad-Shack
13:34.27tootah - cheers
13:34.33JerJerkajtzu:  who cares?
13:34.39JerJerencrypt at the application layer then
13:35.39kajtzuyeah, encrypted signalling and rtp anyone? ;-)
13:35.42DarthClueJust make sure all your VOIP calls are in a language that noone else can understand, like Navaho.
13:36.35goldeneardarwin35, that's the problem so ... the log should show you what happens during the call ...
13:36.40*** join/#asterisk coppice (~chatzilla@244.203.17.210.dyn.pacific.net.hk)
13:36.48JerJerencrypt=yes
13:36.50JerJerproblem solved
13:37.05ManxPowerBTW, Asterisk Business Edition is based on CVS-HEAD and ONLY gets bug fixes.  It might be the solution for people that need some of the features of CVS-HEAD, but don't want behavoral changes.
13:37.28DarthCluebureaucracy=US, problem solved.
13:37.40goldeneardarwin35, your phone may not use asterisk as the default proxy ...
13:37.49ManxPowerOf course, Asterisk Business Edition does not have Realtime documented.  Apparentlys TPTB didn't think it was stable enough.
13:38.05*** join/#asterisk santiago (~santiago@63.245.86.188)
13:38.10MikeJ[Laptop]Calling all asterisk head users... the bug tracker needs you... go adopt a bug.
13:38.13JerJerone should only ever use realtime if one has a very good reason to use it
13:38.25darwin35ok fixed
13:38.26JerJera 'business' does not need realtime
13:38.35MikeJ[Laptop]my house does ;)
13:38.37MikeJ[Laptop]hehe
13:38.43ManxPowerJerJer: *nod*  That's why ABE would work well for us.
13:38.44darwin35even thou the phone is not behind nat I had to set nat=yes
13:38.50MikeJ[Laptop]there's a smart way to watch tv
13:38.55darwin35but that did not work earlier
13:39.01fenlanderCan we expect "Asterisk Service Provider Edition" soon?
13:39.02darwin35now it just did
13:39.05DarthClueMikeJ[Laptop]: I don't want to adopt any bugs, that requires time that I don't have.
13:39.06ManxPowerAnd apparently it actually comes with a real manual.
13:39.12ManxPowerfenlander: that would be called "CVS-HEAD"
13:39.28goldeneardarwin35, ah sip mysteries ...
13:39.46darwin35yeah
13:39.48ManxPowerdarwin35: All the COOL service providers are using it!
13:40.06darwin35what nat=yes
13:40.19darwin35or the manual
13:41.18darwin35well its all working again
13:41.24[TK]D-FenderNeat info I just got on * : A company called ScopServ (on the Wiki) is selling a turn-key * solution (all hardware & software setup up and supported).  Their GUI is actually DAMN solid looking and in gearing up for MASS production and distribution.  They are aligning themselves with some major OEM people and they seem to be taking serious aim at the market,
13:41.27ManxPowerdarwin35: and what does "sip show peers" show.
13:41.37darwin35its all working now
13:41.53darwin35but earlier when I had nat=yes it was not going threw
13:42.06darwin35then I set nat=no and still did not work
13:42.20darwin35then just now set nat=yes and it all works
13:42.33bkw_what
13:42.44darwin35nat is a thorn is the butt of asterisk
13:42.50[TK]D-FenderWebsite : http://www.scopserv.com/ and an on-line demo of the GUI - http://demo.scopserv.com/
13:43.20Nivexdarwin35: maybe only in the butt of SIP
13:43.28NivexIAX and NAT do quite well
13:43.28darwin351001/1001                  10.0.0.2         D   N      255.255.255.255  5060     Unmonitored
13:43.30*** join/#asterisk rjreb (~rjreb@greatwall.amer.net)
13:43.42ManxPowerDarthClue: I have almost no SIP problems with my NAT SIP clients.
13:43.54ManxPowerNivex: IAX and NAT has some issues too.
13:44.33bkw_iax has no nat issues.. unless you're using PAT which isn't really NAT
13:44.37ManxPowerTwo IAX2 phones behind different NAT routers, but talking to the same Asterisk server can't use IAX2 transfers.
13:44.39bkw_IAX has sound quality issues
13:44.39JamesDotCom[TK]D-Fender: WOW, now you're gonna tell us you have no affilliation with the company except a satisfied customer, right?!?!
13:45.03Nivexwhy would IAX have sound quality issues?  that's dependant on the codec
13:45.10bkw_no its not
13:45.14bkw_IAX is pure ass
13:45.16coppiceNivex: wrong
13:45.16ManxPower[TK]D-Fender: I'll wait for user reports of using it.
13:45.26bkw_its got issues
13:45.38ManxPowerApparently bkw_ mind control programming has worn off and he's seeing the light.
13:45.38NivexI'm being goaded here... I can feel it.
13:45.41darwin35bkw how much to get your help with the call back busy
13:45.46DarthClueNivex: IAX has issues.  I can confirm it.
13:45.48bkw_ManxPower, no I have always known this
13:45.52bkw_asterisk in general Sucks
13:45.54bkw_majorly
13:46.01bkw_it has a few fatal flaws
13:46.06ManxPowerbkw_: The FORMER Asterisk cheerleader.
13:46.07bkw_but nobody will listen to us
13:46.08coppiceNivex: no. you're just wrong
13:46.16ManxPowerbkw_: so fork the code and fix it.
13:46.18Nivexbkw_: I hear there's a nice Windows offering you can use :)
13:46.38bkw_forking asterisk to make a better PBX is like starting with a gun with the intention of making a small childs toy.
13:46.39DarthClueWelcome to the dark side.  Once you go dark, you never see the light again.
13:46.43ManxPowerbkw_: and you KNOW the only way you'll get your issues fixed is by forking Asterisk.
13:46.44Nivexcoppice: I was referring to bkw_'s comment.  If you would like to inform me in what manner I am wrong I'd be glad to listen.
13:46.59bkw_ManxPower, not really.. just write something else
13:47.00ManxPowerbkw_: So you are moving over to YATE then?
13:47.03bkw_no
13:47.13ManxPowerbkw_: best of luck with that.
13:47.20[TK]D-FenderJamesDotCom : No affiliation.  I am trying to get * into my company and the bosses here don't like the "programmer only" maintenance needs * demands.  ScopServ is just a way of validating getting * in the door and kicking out Nortel's BCM solution we are being proposed.
13:47.39coppiceNivex: codec is a biggy in sound quality. several things affect the smoothness of the stream and they have just as big an effect on sound quality
13:47.51bkw_what coppice said
13:48.05DarthClue[TK]D-Fender: tell your bosses that you can get someone to support it no questions asked for less than the cost of your nortel system.
13:48.24puzzledbkw_: mind elaborating a bit on what you think is wrong with *?
13:48.29[TK]D-FenderJamesDotCom : Personally I'd like the just white-box it myself, but they don't like the word "home-grown" even if I DID find 1/2 dozen consultants to set it up.  These people are friggen GUI centric where everything has to be configurable bya MORON.
13:48.55bkw_puzzled, every call uses the same port.. same queue.. as the box gets loaded with traffic.. the more and more calls you have the worse it gets.
13:49.00DarthClue[TK]D-Fender: i can make that happen for less than your nortel system.
13:49.01[TK]D-FenderDarthClue : Didn't work, they want turn-key...... ScopServ made the best offering to get it "in the door"
13:49.21ManxPower[TK]D-Fender: Let us know how well it works.
13:49.22[TK]D-FenderDarthClue : So can I!  The point is they want something that makes them feel "warm & fuzzy" :(
13:49.40bkw_puzzled, can you see what i'm talking about?
13:49.45DarthClue[TK]D-Fender: pm me the specs that you need and what I need to compete with and I'll see if i can do better.
13:49.47coppicethey want to resemble pubic hair? :-\
13:49.59ManxPowerWe're in a SAVE MONEY mode.  Apparently the company has not made any money so far this year.
13:50.01puzzledbkw_: yup (as far as my understaning of C goes)
13:50.17[TK]D-FenderFranly my test server is VERY nice right now and it'd take about 10 hours to finalize it & SpanDSP I'm sure, but they want "idiot-proof" and "firiendly" and "Supported".  They are too used to "stock" systems
13:50.26bkw_now if IAX could punch holes in NAT and have a monitor thread per call.. now that might change things
13:50.37bkw_oh and a double legged IAX connection is pure evil
13:50.52*** join/#asterisk Ahrimanes (~aron@hobbes.bsd-dk.dk)
13:50.56[TK]D-FenderDarthClue : Anyone can offer me * at a better price including my doing it ALL from scratch (which I have actually just done (mostly).  But Cost isn't their point :/
13:50.59sivanaso is IAX better than SIP overall?
13:51.10bkw_Zap => IAX => BOX => IAX => ZAP  <--EVIL
13:51.11ManxPowerbkw_: I already talked to someone about better IAX2 NAT traversal.
13:51.17puzzledbkw_: a little patch here, a little patch there, rinse and repeat :)
13:51.20bkw_sivana, no
13:51.21DarthClue[TK]D-Fender: i can give them turnkey, i just need to know what I have to work with on your end.
13:51.24coppiceoverall the PSTN is best "-)
13:51.28[TK]D-FenderIts "user maintainable" and "friendly" that counts.
13:51.35ManxPowerbkw_: You've been spending too much time with timecop
13:51.44bkw_SIP will whip IAX in quality
13:51.47DarthClue[TK]D-Fender: i can do that.
13:51.54[TK]D-FenderDarthClue : You'd also HAVE to have a LOCAL physical presence.  AKA Montreal, QC.
13:52.00bkw_ManxPower, nope just telling it like it is
13:52.04sivanahrm.. and here I thought IAX was better
13:52.13[TK]D-FenderThats the curse * has with corporate approval.
13:52.22bkw_sivana, you must be drinking out of the IAX punch bowl
13:52.26sivanaya :)
13:52.27DarthClue[TK]D-Fender: is ScopServ local?
13:52.44puzzledbkw_: is there any agreement amongst the developers that these are indeed issues that need to be addressed?
13:53.04bkw_puzzled, yes but asterisk development is a one way street as of late
13:53.12zoamark seems to agree with changes needed for iax2
13:53.14Nivexand here I've been espousing the virtues of IAX to my friends of late.
13:53.18[TK]D-FenderDarthClue : Yup, Offices in Montreal and 2 surrounding cities
13:53.22zoai had a talk with him on astricon about it
13:53.26bkw_zoa knows of the flaws in IAX also
13:53.39zoai think we need an iax3
13:53.42bkw_yes
13:53.47iCEBrkrDive! Dive! Dive!
13:53.49Ahrimanesah i was there too :)
13:53.49puzzledbkw_: ok, well, realizing it is a start. babysteps I guess
13:53.50ManxPowerMaybe if mark did more coding and less traveling....
13:53.54zoaoh brian, i have a patch for sip nat for sale
13:53.57zoanot my patch though
13:53.58DarthClue[TK]D-Fender: too bad, i bet i could easily compete and give you and your bosses a system that would make all of you happy.
13:54.02NuggetA lot of IAX advocacy is just a groupthink feedback loop.
13:54.24iCEBrkrManxPower: Naa, bkw just needs to code faster.
13:54.25NuggetAnd a lot of the anti-SIP sentiment is just misplaced anger at NAT.
13:54.25zoawell i like iax2, but you need to know when to use it and how to use it
13:54.25darwin35bkw you have not answerd me
13:54.30Twisterwhen i dial *77 and make a recording it will play through playback when when i do echo "text" >> text2wave hello.wav it wont play, what options am i missing for text2wave?
13:54.52puzzledNugget: my anger is totally reserved for NAT and stupid alcatel adsl modems :)
13:54.54ManxPowerYay!  My callerid-fixup.agi is working
13:55.06darwin35cool
13:55.08iCEBrkrF NAT.  I wish NAT was never invented. :D
13:55.42coppiceNay wasn't invented
13:55.43[TK]D-FenderDarthClue : I quoted a system with bottom-dollar hardware costs on all components (best of breed purchasing).  I then added on $1500 for 10hrs@150$/hr of support (IF NEEDED) just to set up the support relationship.  I was proposing to set it all up myself.  Hard to beat that.... but that wasn't what they wanted.  They want someone else to set it up make it idiot-proof, and support it physically.
13:55.49coppiceNAT wasn't invented
13:55.59coppiceIt sort of festered into life
13:55.59Ahrimanesit was summoned from below..
13:56.08bkw_haha
13:56.16iCEBrkrcoppice: Yea, I know.. :/
13:56.28bkw_nat spells 628 so its on the same block as 666
13:56.35ManxPowerI just don't have many problems with NAT
13:56.36iCEBrkrhaha
13:56.56DarthClue[TK]D-Fender: luckily my old boss is a cheap bastard.  we are replacing our entire pbx and dialer for $20,000 and that gives us ton's of growing room.
13:57.13iCEBrkrManxPower: With a properly setup router sure.  But it's just a headache.  My setup has been playing nicely with NAT so far.  Kinda scary.
13:57.13[TK]D-FenderYou couldn't beat MY price, but you might beat ScopServ's.  Theirs is +/- $12,000 for a 4 port T1 server.  (Rack mount SCSI RAID, fully congured and GUI'd up)  Frankly for the actual increase from white-boxing I consider it an acceptable trade-off since I will have root access to it.
13:57.17bkw_DarthClue, you didn't quite that job yet?
13:57.36[TK]D-FenderDarthClue : Lucky for you your boss takes your word for it.  I know I'm right, but they don't care that much.
13:57.44darwin35I just wish asterisk would drop mpg123 and just move to madplayer it works great
13:57.52darwin35no issues
13:57.57ManxPoweriCEBrkr: it took me 1 hour to configure my SIPura, my Asterisk, and my Cisco router to handle NAT.  The phone can move between the local lan and the general internet transparently.
13:57.57bkw_darwin35, please tell us why?
13:58.02iCEBrkrdarwin35: hack the source.
13:58.11bkw_ya really
13:58.20DarthCluebkw_: i found out this morning he is in panic mode.  I am going in this afternoon to tell him i am part time only starting yesterday.
13:58.55iCEBrkrManxPower: I'm not a network guy, so that kinda stuff is a bit confusing at first.  But like I said, all my stuff just seems to work and I was rather surprised. :D
13:58.59DarthCluebkw_: he never showed up yesterday before i left.
13:59.15bkw_sounds like you need to call your big boss to get the boss fired
13:59.17ManxPowerOf course if you have a cheap-ass old Linksys router that randomly reboots when it has to NAT SIP packets, then that's not a NAT problem.
13:59.20tooti used to have a x100p clone - it stopped working - i wondered might it be anything to do with the fact my phone rings once after i hang up a call? i have now bought a tdm400p and don't want it to go the same way - any thoughts?
13:59.28iCEBrkrManxPower: hehe
13:59.38iCEBrkrRandomly reboots haha
14:00.28ManxPoweriCEBrkr: each of the 3 older (hardware rev a) Linksys routers start rebooting when I put a SIP phone behind them.  Running the latest firmware available for that hardware.
14:00.42DarthCluebkw_: neither my manager, nor the big boss showed up yesterday.
14:00.44ManxPowerI've almost finished switching the 2 locations to m0n0wall
14:00.44Kattyhi
14:00.47Kattyi have query
14:00.48Kattywhat does this mean:
14:00.52KattyJul 12 08:54:08 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Critical Request)
14:00.52ManxPower..er.. 3 locations
14:00.55KattyJul 12 08:54:12 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Non-critical Request)
14:00.57iCEBrkrManxPower: that's fizzucked.
14:01.06ManxPowerKatty: it means asterisk can nolonger see the phone.
14:01.20ManxPowerEither the phone is off, the NAT translation timed out, or the phone changed IP addresses.
14:01.23ManxPoweriCEBrkr: Yup!
14:01.30Kattymust be NAT timed out
14:01.36bkw_ManxPower, those messages happen even on the local lan
14:01.39Kattycause the phone is still on and they're set to manual ip
14:01.42ManxPowerKatty: qualify-yes will fix that.
14:01.52ManxPowerqualify=yes that is
14:01.53KattyManxPower: in sip.conf?
14:02.00ManxPowerKatty: yes, in the entry for the phone
14:02.18[TK]D-FenderDarthClue : My 6187$ home made boring server woul be replaced by a SCSI redundant supported server costing $12000.  My original solution TOTAL cost was $19,000, and Nortel's bid was $45,000.  If I go up $6000 and still get to KEEP * Halla-fucking-lugiah!  *THEN* I will twist it to my designs :D
14:03.11[TK]D-FenderFor me ScopServ is a means to an end.  I'd prefer to bank the $6,000 mind you, but if it gets me what I want, I'll do it.
14:04.00DarthClue[TK]D-Fender: is that canadian?
14:04.24[TK]D-FenderBut I do have to admit I'm DAMN impressed by their GUI especially because it does Queue statistic reports as well which is only really avaliable by the XC-AST add-on for almost $1000 CDN.  Add that to my $6187, and $12,000 looks better again
14:04.35[TK]D-FenderDarthClue : Yes, All CDN
14:04.44*** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au)
14:05.01zoai have queue statistics reports too
14:05.17[TK]D-Fenderzoa : Using what?
14:05.28DarthClue[TK]D-Fender: take it, i doubt i could beat that.
14:05.37[TK]D-Fenderzoa : And I'm something web-based and pretty.
14:06.03*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
14:06.03bkw_you do realize that chan_agent and app_queue are CRAP
14:06.11bkw_aka chan_deadlock
14:06.16bkw_and app_segfault
14:06.19[TK]D-FenderDarthClue : ScopServ's advantage is they are heading to mass-distribution to be sold as commoditiy like the Nortel BCM is without the added licensing costs.
14:06.38maikwho killed the cvs server?
14:07.00[TK]D-Fenderbkw_ : So how long before ICD becomes better documented and merged in?
14:07.14DarthClue[TK]D-Fender: I am working with someone else to head in that direction, but ours aren't quite to that level yet where we can compete with another VOIP system.
14:07.29ManxPower[TK]D-Fender: I highly doubt that ICD will EVER be merged into Asterisk.
14:07.30[TK]D-Fenderbkw_ : It didn't leave a good taste in my mouth when I first read up on it.  Too cryptic for me at the time
14:08.09[TK]D-FenderDarthClue : They are already targeting the BCM and have channel parters I know well already lined up.  This could be BIG for * in the long run.
14:08.40bkw_ICD has gotten out of hand
14:08.54[TK]D-Fenderbkw_ : So what does that leave me?
14:09.09*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
14:09.41[TK]D-Fender..besides screwed...
14:09.52bkw_nowhere
14:09.54*** join/#asterisk mcreedjr (~anon@oh-65-41-206-34.sta.sprint-hsd.net)
14:10.50[TK]D-Fenderbkw_ : And there you have it.  So I guess I'll just have to sit back and pray that *'s call center woes get resolved by SOMEBODY in some way...
14:10.59mcreedjrCan anyone point me to a good explaination of the difference between FXO/FXS? I am wondering if I need one FXO port and one FXS port to be able to send and receive calls from the PSTN.
14:11.02*** join/#asterisk wiseguy_ (chivilis@85.206.10.242)
14:11.15wiseguy_i can't login as queue agent with grandstream phone
14:11.30wiseguy_how do i need to send dtmf, as sip info maybe?
14:11.30ManxPower~fxofxs
14:11.30jbotfrom memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
14:11.53wiseguy_any one using queues?
14:11.56DarthCluewiseguy_: yes, info.
14:12.33mcreedjrJbot: So I need to use FXO ports to access my outside PSTN lines, and then I need FXS ports as extensions to hook my phones to then?
14:12.48[TK]D-FenderDarthClue : ScopServ is coming on-site with a server & 2 phones for a demo and proposal.
14:13.00wiseguy_okey, i will try
14:13.16mostymcreedjr, by jove you've got it :)
14:13.18DarthClue[TK]D-Fender: does that cost include phones or just the box?
14:13.32mcreedjrMosty: Thanks for the 'atta-boy :)
14:14.21*** join/#asterisk brookshire (~pfffft@207.111.174.1)
14:14.22*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
14:14.27DarthClueMikeJ[Laptop]!
14:14.28mcreedjrSo an FXO will answer inbound lines and also allow me to call out on the PSTN then?
14:14.37[TK]D-FenderDarthClue : just the box & support.  I'm looking to get all Polycom IP 600 phones.  SCopserv already auto-provisions Cisco & Aastra phones and they're looking to add Polycom to that list
14:14.42MikeJ[Laptop]hello
14:14.45mostymcreedjr, you only need fxs ports if you want to use regular (non voip) phones
14:15.10MikeJ[Laptop]Calling all asterisk head users, today is adopt a mantis bug day.  Adopt a bug and fix it!
14:15.23[TK]D-FenderDarthClue : For the rest of my solution I can bottom-dollar source it (like my Rhino failover / analog solution)
14:15.27MikeJ[Laptop]it's your time to give back a little bit.
14:15.47DarthClue[TK]D-Fender: how soon do you need an answer? and is having someone local a requirement?  I might be able to beat that price if it is the box and support (how many hours).
14:15.48brookshiremikej: you're awesome ;)
14:15.57mcreedjrMosty: Okay, great, makes things easier. Thank you for your help!
14:16.04mcreedjrJbot: Thanks for the help.
14:16.04jbotgern geschehen, mcreedjr
14:16.52DarthCluejbot: are you dead?
14:17.27brookshirejbot: purr
14:17.27jbotACTION *purrs purrs purrs*
14:17.30brookshire:D
14:17.36DarthCluejbot: explode
14:17.36jbotBOOOM
14:17.40DarthCluejbot: die
14:17.40jbotACTION takes two shots to the head and crumples to the ground, lifeless.
14:17.42ManxPowerjbot doesn't seem to do google searches anymore
14:17.52brookshire:(
14:17.52DarthCluejbot: google digium
14:18.00brookshire:(((((((((((((((((
14:18.13DarthCluejbot: fsck brookshire
14:18.14jbote2fsck /dev/brookshire : warning! filesystem contains idiots!
14:18.24DarthCluejbot: fsck jbot
14:18.24jbote2fsck /dev/jbot : warning! filesystem contains dickheads!
14:18.43cpatry~google cluecon
14:18.50DarthClue~cluecon
14:18.50jboti heard cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
14:19.00MikeJ[Laptop]brookshire, no.. I am naggy...
14:19.04[TK]D-FenderDarthClue : Local = MUST.  Instant failure for you.  And you don't have to corporate image and backing to convince people here.  I've already called in professional consultants here for this.
14:19.05MikeJ[Laptop]but I enjoy it...
14:19.40mcreedjrCan anyone recommend a good, cheap IP phone. I am pilot testing an Asterisk solution and I just want to try VoIP out.
14:19.46coppicecluecon puts the con in conference :-)
14:20.01DarthCluemcreedjr: Grandstream bt100
14:20.16DarthCluecoppice: cluecon will have the best of the VOIP world and then some.
14:20.20ManxPowermcreedjr: There ARE no cheap good phones.
14:20.36ManxPowermcreedjr: Try the SIPura SPA-841.  It's the best phone for under $100.
14:20.46ManxPowerWe use Polyco IP 300 and IP500
14:20.47coppiceno it won't. I'm not going :-)
14:20.58ManxPowerme neither
14:21.06ManxPowerThere are two "best" that won't be there.
14:21.11mcreedjrManxPower: I was just waiting for the no good, cheap phone thing :) Thank you for your input
14:21.34ManxPowermcreedjr: Grandstream would qualify as "cheap and no good"
14:21.38DarthCluecoppice and ManxPower: i'm not commenting.
14:21.40*** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca)
14:21.56DarthClueGrandstream qualifies as a good test phone.  Period.
14:21.58MikeJ[Laptop]coppice, but you should..
14:22.00ManxPowerAnd honestly, I don't give a flying fuck about non-asterisk related stuff.
14:22.13MikeJ[Laptop]cuz I really want to finally meet you.
14:22.19coppicehold it in asia and i'll consider going
14:22.22lyroyDoes someone could tell me how can I insert VALUES in a MYSQL database in my dialplan?
14:22.28MikeJ[Laptop]where are you?
14:22.31*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
14:22.37coppiceduh! asia
14:22.42MikeJ[Laptop]I got that much.
14:22.46RoyKhk
14:22.50*** join/#asterisk darwin35 (~darwin35@ip70-186-117-198.ma.dl.cox.net)
14:22.54RoyKif he hasn't moved lately
14:22.55mostylyroy, what do you mean by values?
14:22.55*** join/#asterisk lehel (~lehel@82.79.20.17)
14:23.01lehelhello
14:23.10darwin35ok major issues with head as of 8:45 am
14:23.12ManxPowercoppice IRCs from a remote village in China and uses GRPS and SAT links for IRC.
14:23.17MikeJ[Laptop]hehe
14:23.19RoyKlyroy: use application MYSQL from asterisk addons
14:23.23darwin35audio for voicemail craps out
14:23.43lyroymosty exten => 5,3,MYSQL(Query resultid ${connid} INSERT\ INTO\ foward\ VALUES\(\'XXXXXXXXXX\'\,\'XXXXXXXXXX\'\))
14:24.22lyroysomething like that
14:24.58darwin35it crashes asterisk  when it cant read the vm password also
14:25.09Kattyaroo!
14:25.14mostylyroy, you probably want to lookup the "asterisk gateway interface" (agi): http://www.voip-info.org/wiki-Asterisk+AGI
14:26.13*** join/#asterisk cybyc (~CYB@bi01p1.nc.us.ibm.com)
14:27.16darwin35I also get constant speaker beeps
14:27.18Kattywoah
14:27.22Kattythere are still drive in theators
14:27.28KattyJul 12 08:54:08 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Critical Request)
14:27.31Kattyerm, wrong paste
14:27.34KattyJul 12 08:54:12 WARNING[865]: chan_sip.c:907 retrans_pkt: Maximum retries exceeded on call 5dda11452a01acad3451271917c09320@192.168.0.26 for seqno 102 (Non-critical Request)
14:27.43Kattyhttp://www.driveintheater.com/index.htm
14:27.49*** part/#asterisk cybyc (~CYB@bi01p1.nc.us.ibm.com)
14:27.52Kattythat's neat :>
14:27.55DarthClueKatty: yes...We have one here that shows first run movies for less than the actual theatre.
14:28.15goldenearManxPower Two IAX2 phones behind different NAT routers, but talking to the same Asterisk server can't use IAX2 transfers. <-- it works for me :)
14:28.39KattyDarthClue: hot
14:28.44Kattyi wanna go to a drive in movie
14:28.46darwin35Jul 12 09:15:00 WARNING[70034] chan_sip.c: sip_xmit of 0x8158e1c (len 451) to 10.0.0.2 returned -1: Bad file descriptor
14:28.58darwin35its filling my log with loads of these
14:30.00goldenearManxPower, if the * server has a public ip and if the NAT is nos symmetric, IAX2 transfers will work. I did the test.
14:30.55ManxPowergoldenear: between IAX2 phones behind different NATs.  I don't mean "transfer the call".  I mean IAX2 native transfers so the phones talk directly to each other once the call is set up, bypassing Asterisk.
14:31.18ManxPowerIt really should be called "IAX2 reinvites" or something other than "transfer"
14:32.10*** join/#asterisk santiago (~santiago@63.245.86.188)
14:32.41darwin35well thats screwed
14:33.12goldenearManxPower, that what I'm talking about
14:33.27goldenearand it works :)
14:33.51*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
14:34.04ManxPowergoldenear: it will work with port forwarding, of course.
14:34.05*** join/#asterisk JakubS_ (~qbast@pe72.czempien.sdi.tpnet.pl)
14:34.42JakubS_hello
14:34.59JakubS_do all G.7xx codecs are lossless?
14:35.19coppiceJakubS_ none are
14:36.10ManxPowerJakubS_: not G711
14:36.24JakubS_are they enough to transmit fax ?
14:36.41coppiceeven G.711 is not lossless
14:37.11goldenearManxPower, it worked for me without any manual seting on the NAT/router
14:37.47ManxPowergoldenear: How do you know that the iax2 native transfer worked?
14:37.58goldenearI did the test
14:38.06ManxPowergoldenear: WHAT TEST?
14:38.29ManxPowerDid you use tcpdump?  Did you look at the console?  Did you get telepathetic reports from alians from Alpha Centauri?
14:38.57*** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155)
14:39.03ManxPowerpacket sniffing is really the only way to know if the IAX2 native transfer worked or not.
14:39.27coppiceis packet sniffing addictive?
14:39.35goldenearuser1 behind NAT 1, user2 behind NAT 2, * server with a public IP (different than each NAT/router IP)
14:39.38Nuggetheh
14:39.42mariogamboai have a question? i want to use the g729 codec but i see i need pay royalities is true this or exist a open source of the codec
14:39.52Nuggetmariogamboa: it is true.
14:40.01Ahrimanescoppice: quite so
14:40.07MikeJ[Laptop]coppice. yes... and it gets you high
14:40.09ManxPowermariogamboa: you need to pay a license fee.  Digium sells the license in partnership with the patent holder.
14:40.13mostymariogamboa, you need to buy licenses for the g729 patent. it's $10 usd per channel
14:40.16goldenearuser1 call user2 --> asterisk log show native transfer
14:40.27ManxPowergoldenear: that means nothing.
14:40.34ManxPowergoldenear: try using a packetsniff.
14:40.47Ahrimanesethereal is your pal
14:41.15mariogamboain this channel is when a ip phone call to another ip phone i need to buy 1 license or need 2 licenses
14:41.35ManxPowermariogamboa: if both phones support G729 then you don't need any license.
14:41.49Kattyhow much do you tip a hair dresser?
14:41.54goldenear(also monitoring with successive iax2 show channels show me that the connection between usr2 and usr2 is released as usr1 and usr2 are effectly still speaking )
14:41.57Kattyno cow tipping comments please (=
14:42.01bkw_Cresl1n, boi whats up
14:42.05Hmmhesaysha!
14:42.09Kattybkw_: how much do you tip your hair dresser?
14:42.09mostymanxpower: you do if you want to do voicemail, or anything else that "speaks" to something that only supports g729
14:42.10mariogamboai have 2 grandstream
14:42.11Cresl1nhowdy all ;-)
14:42.19ManxPowerbut if you want the phone to connect to the PSTN, Asterisk voicemail. "t/T" option to Dial, use Zapscan, Zap Barge, or any other application or part of asterisk that has to listen to or play audio -- then you need a licenmse.
14:42.19goldenearManxPower, why do it means nothing ?
14:42.20Kattybkw_: TELL ME
14:42.22Kattybkw_: I MUST KNOW
14:42.23bkw_Katty, I don't.. I give her gossip
14:42.28Kattybkw_: oh. k
14:42.34bkw_we gossip like crazy
14:42.41bkw_because her ex-husband married my neighbor
14:42.45Ahrimaneslol
14:42.48Kattybkw_: do you know what you're supposed to tip?
14:42.49ManxPowergoldenear: Asterisk is pretty bad this sort of stuff when it comes to the console messages.
14:42.50mariogamboabut if i don't add the g729 codec to asterisk the grandstream phones only conect in g711a
14:42.59Hmmhesaysnothing if they suck at hair dressing
14:43.00mariogamboacan't connect in g729
14:43.05KattyHmmhesays: and if they don't?
14:43.12ManxPowermariogamboa: that is correct.
14:43.13bkw_Katty, I have simple needs
14:43.15bkw_I shouldn't have to tip
14:43.20Kattyargh
14:43.23bkw_I think 13 bucks for a haircut is plenty
14:43.25bkw_for 10 min of work
14:43.25Cresl1nmariogamboa: yep
14:43.26Kattyyou're going to make me ask google :<
14:43.32mariogamboain this case manx i need to buy the license is correct?
14:43.34Cresl1nmariogamboa: that's the way it is
14:43.39ManxPowerbut if you want the phone to connect to the PSTN, Asterisk voicemail. "t/T" option to Dial, use Zapscan, Zap Barge, or any other application or part of asterisk that has to listen to or play audio -- then you need a licenmse.
14:43.43Cresl1nmariogamboa: right again
14:43.45Hmmhesaysyeah I normally will give them a buck
14:43.45goldenearManxPower, I also verified it with netstat :)
14:43.46mostymariogamboa, do the grandstream phones support gsm?
14:43.47Kattybkw_: well my hair is going to take a wee bit more than 10 minutes
14:43.55goldenearit works I tell you !
14:43.56Kattybkw_: try about 50 minutes :<
14:43.57Hmmhesays15%
14:44.04ManxPowergoldenear: What universe do you live in?  I want to move there.
14:44.05NuggetThe difference between a $12 haircut and a $120 haircut is about two days.
14:44.06KattyHmmhesays: google said 20%
14:44.13Cresl1nKatty: that's a long time
14:44.15ManxPowerMy haircuts are free.
14:44.18Kattythat's like a dollar for every 5
14:44.20Hmmhesaysgoogle wants you to put more money into the economy
14:44.25Cresl1nKatty: how much does it cost?
14:44.30KattyCresl1n: don't know yet
14:44.31mariogamboaok
14:44.35Cresl1nhrm...
14:44.39KattyCresl1n: just want to know how much to tip (=
14:44.43Cresl1nKatty: Maybe you could check froogle
14:44.46Cresl1n:-)
14:44.46KattyCresl1n: trust me, it will take awhile...my hair is down to my hips.
14:44.58KattyHmmhesays: you missed webcam last night!
14:45.02goldenearManxPower, I live on planet IAX, constellation of asterisk :)
14:45.03mariogamboaanyhere have the pinout of the wildcard te110p
14:45.08KattyHmmhesays: last picture is still up if you wanna go see (=
14:45.09ManxPowerKatty: a clippers will fix that very fast.
14:45.14KattyManxPower: noooo
14:45.21HmmhesaysI did huh, I was sleeping all night long.  sure
14:45.41KattyHmmhesays: it's in usual spot (=
14:45.45DarthClueKatty: warn us next time web cam is gonna be on.
14:45.59KattyDarthClue: it'll be like 2 motnhs
14:46.02Kattyi mean months
14:46.12Hmmhesaysyou'll have to remind me, It's been like 9 months since i've been there
14:46.17KattyHmmhesays: oh, and see new.jpg, new2.jpg, and new3.jpg for the new heels i bought (and am wearing today!)
14:46.28DarthClueKatty: that's ok, we'll see ya in less than a month at cluecon.
14:46.31goldenearManxPower, and on my planet SIP means local net or ipv6 only :)
14:46.41KattyDarthClue: ya (=
14:47.04*** join/#asterisk jhava (~icechat5@200.58.26.21)
14:47.16Kattyi'll be the Voice From Behind
14:47.23Kattyto quote a famous book
14:47.29bkw_ok that sounds just wrong
14:47.33Kattyk
14:48.02*** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net)
14:48.03sneakhi
14:48.07Kattybkw_: s'ok, i don't want people staring at me anyway (=
14:49.06goldenearManxPower, why IAX2 native transfers shouldn't work with [non symmetric] NAT ?
14:49.10*** join/#asterisk mkrufky (~mk@68.160.103.77)
14:49.15KattyDarthClue: until i steal it
14:49.29Kattyactually, i think i'll nap
14:49.49*** join/#asterisk Madkiss (madkiss@madkiss.staff.freenode)
14:49.52MadkissHi folks
14:50.27bkw_i'm about to murder greg
14:50.33Kattybkw_: :<<<
14:50.36bkw_he thinks he's about to die.. he's not feeling well
14:50.41MadkissWhat does "insecure=very
14:50.42Kattyrut roh
14:50.43Madkiss" cause in sip.conf?
14:50.51Kattybkw_: that calls for hugging, sometimes :<
14:50.52bkw_he does this every single time he feels bad
14:51.04bkw_then he goes to the dr. which costs out the ass
14:51.07DarthCluebkw_: maybe he knows that you are about to kill him?
14:51.23Kattydoctor visit costs me 30
14:51.29Kattywhich includes whatever drugs they decide to put me on
14:51.46ManxPowergoldenear: because the two phone have never communicated directly before, they have all been talking to Asterisk, so there is no dynamic NAT translations for the phones going
14:51.51Kattybut i've got insurance, too
14:51.54DarthCluei need an expiration date.  what would be a good expiration date for a jug o milk.
14:52.19*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:52.19*** mode/#asterisk [+o anthm] by ChanServ
14:52.25Kattyhow about a jug o DarthClue
14:52.39ManxPoweri.e. Phone at 172.16.7.12 is communicating with IP address 209.45.3.16.  If another device then tries to communicate with the phone, well the NAT router has never seen an outbound packet to the 2nd phone's IP address and so doesn't have a NAT translation set up for it.
14:52.52Katty:<
14:53.03Kattyyou can't leave me alone with bkw_!
14:53.20DarthClueKatty: bkw_ is harmless, mostly.
14:53.23*** join/#asterisk wasim (~wasim@wasim.active.supporter.pdpc)
14:53.26Katty:<<<
14:53.32goldenearit does, as the * server know on wich port the Phone is connecting from
14:53.43ManxPowergoldenear: Correct, but the NAT router does NOT.
14:54.20goldenearbut if the NAT is not symmetric, the used port is kept for any connection
14:54.24goldenearso it will work
14:55.08ManxPowergoldenear: NAT is done on PORT+IPADDRESS
14:55.18*** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net)
14:55.25goldenearand so ?
14:55.32ManxPowerEven if the packet is coming from the correct source port, the address will be different from that of the Asterisk server.
14:56.07ManxPowerit would be like doing a DNS query thru a NAT router, and the response comes from a different port and IP adddress.  It won't get thru the router.
14:56.12goldenearbut it will be accepted, as the phone also try to contact the other one
14:56.40ManxPowergoldenear: and the NAT router then picks a random source port for NAT and the packet fails.
14:57.04goldenearyou think symmetric NAT !
14:57.22goldenearI said, if the NAT is symmetric is won't work
14:57.31MadkissOh, does somebody know whether the voicemail-mesages are available in german language?
14:57.34goldenearbut if the nat is non symmetric, it will work
14:57.52ManxPowergoldenear: so you admit there ARE nat issues with IAX2 native transfers.
14:58.10mostymadkiss: i see a package in debian (sarge) called asterisk-prompt-de
14:58.50*** join/#asterisk mrtwister (~user@cable-1-32.cgates.lt)
14:59.00goldenearbut less than with SIP
14:59.22*** part/#asterisk mrtwister (~user@cable-1-32.cgates.lt)
15:00.00*** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net)
15:00.06*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
15:00.25goldenearand if native transfer is impossible because of symmetric NAT, the call will just smoothly keep to go through the * server
15:00.29ManxPowergoldenear: SIP reinvites simply don't work with NAT unless, maybe, you have an external "thing", SER, STUN, etc.
15:00.31Madkissmosty: Cool, thank
15:00.44*** join/#asterisk montag___ (~montag@host187-252.pool8175.interbusiness.it)
15:01.26montag___what's the best way to interconnect a PABX with asterisk for a ibrid system (asterisk with some VoIP clients <---> pabx with phones)
15:01.28montag___?
15:01.42goldenearsymmetric NAT are so evil !
15:01.53alphaquemontag___: depends how many lines. if its large, consider E1 otherwise a few FXOs
15:01.54goldenearthat's the issue, not IAX
15:02.16*** join/#asterisk lehel (~lehel@82.79.20.17)
15:02.44goldenearyep, SIP needs help to work with NAT
15:03.01lehelwhy i can't run asterisk as "asterisk" while i'm loading chan_capi ??
15:03.02montag___8 lines...i think a couple of tdm400 connected to analog extensions on pabx, but i think it's better E1. But TE410P E1 card work well with all pabx ????
15:03.21ManxPowerI recommend you stay away from the TDM400Ps
15:03.30alphaquemontag___: u'd need an E1 conn from the pbx too
15:03.44coppicei recommend staying away from anything electronic
15:03.57ManxPowerlehel: use a channel bank if your PBX has no E-1/T-1 ports
15:04.27montag___with FXO lines two problem: call delay (waiting for the first ring...), and when i receive a call from the analog pabx and i trasnfer the call to another analog pabx extension i use 2 analog lines :-(
15:04.45montag___i've tried only tdm400, there's better on the market ???
15:04.52alphaquemontag___: cant u use flash() to do that ?
15:05.00ManxPowermontag___: Channel bank
15:05.06lehelwhat do you mean channel bank ManxPower ?
15:05.19montag___alphaque: flash ???
15:05.42alphaqueyes, flash hook transfer. should free up the line connected to pbx and thus let call pass within pbx
15:06.03montag___alphaque: have u some samples for this ???
15:07.30razucan anyone help me loading ztdummy module under slackware 10.1 ... i'm getting errors :S
15:08.13*** part/#asterisk santiago (~santiago@63.245.86.188)
15:08.23alphaquemontag___: www.voip-info.org
15:08.30goldenearwho said that iax2 has some audio issues ?
15:08.58goldenearbkw_, ?
15:09.09ManxPowerbkw_ says a lot of stuff.
15:09.39goldeneardoes he take drug or something ? ;)
15:09.53ManxPowergoldenear: I hope so!
15:10.14goldenearme too!
15:10.37goldenearhe's scary ;)
15:10.54ManxPowergoldenear: he's also right much of the time.
15:10.56*** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au)
15:11.01ManxPowerand he's smart as well.
15:11.02lehelManxPower: i read about Asterisk+Channel+Bank on voip. but still 'don't get..
15:11.17goldenearso he is even more scary !!!
15:11.46ManxPowerlehel: your analog lines into FXO ports on the channel bank, the FXS ports to your PBX, then a T-1 card to Asterisk from the channel bank.  A channel bank converts t-1/e-1 channels into analog ports
15:13.37*** part/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au)
15:13.45*** join/#asterisk Malthus (~herman@port0129-abn-s-adsl.cwjamaica.com)
15:13.56*** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au)
15:14.27lehelManxPower: i've no doubt that this is a useful information, but not for me yet:-/
15:14.46leheland i don;t know what is the connection between my Fritz card
15:16.05lehelAVM Fritz! CAPI ..
15:16.26Kattylehel: you've insaned.
15:17.20lehel?
15:18.36lehelspecify Katty
15:19.50Kattylehel: you. have. insaned.
15:25.02lehelexplain
15:25.09Katty...
15:25.31lehelthen shut up
15:25.40Kattygosh
15:25.47*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
15:25.54Kattylehel: don't be hateful )=
15:26.19lehelk i'll try no to.. :P
15:26.24MikeJ[Laptop]whos a hater?
15:26.34MikeJ[Laptop]yous a hater!
15:26.45Kattyhater? i hardly know 'er!
15:27.34*** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) [NETSPLIT VICTIM]
15:27.34*** join/#asterisk johnh51 (~john@adsl-66-218-62-115.dslextreme.com) [NETSPLIT VICTIM]
15:27.34*** join/#asterisk Brumle (~brumle@brumle.com) [NETSPLIT VICTIM]
15:27.34*** join/#asterisk KryoStoffer (~kri@helium.kri.dk) [NETSPLIT VICTIM]
15:28.20lehelzbye
15:28.23*** part/#asterisk lehel (~lehel@82.79.20.17)
15:28.39*** join/#asterisk kkq (~christ@234-200-29-134.hcc.mnscu.edu)
15:28.50kkqhello
15:33.12*** join/#asterisk tomtom_ (~root@83-217-70-161.reverse.realroot.be)
15:33.14tomtom_hi
15:35.21kkqhi
15:35.41RoyKis robert webb here?
15:35.41*** join/#asterisk pdugas (~pdugas@h79.95.40.69.ip.alltel.net)
15:36.13kkqjack webb is on my television
15:36.41*** join/#asterisk LOT (~Methos@S0106000f6694b86f.ed.shawcable.net)
15:37.07*** join/#asterisk rjreb (~rjreb@greatwall.amer.net)
15:37.51ManxPower~docs
15:37.51jbotDocumentation 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
15:37.52ManxPower~mailinglist
15:37.52jbot[mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:38.49albators~seen royk
15:38.49jbotroyk is currently on #asterisk (8h 20m 42s).  Has said a total of 48 messages.  Is idling for 3m 8s
15:38.58albators~history
15:38.58jbothistory is, like, at http://ibot.rikers.org/oe/
15:39.11albators~asterisk
15:39.11jbotmethinks asterisk is the symbol that looks like a star (shift-8 on north american English keyboards)
15:39.32albators~;ls
15:39.46albators:P
15:40.26albators~did
15:40.26jbotwell, did is Direct Inward Dialing
15:40.43albators~CLIP
15:40.50albators~CLI
15:40.50jbotmethinks cli is a Command Line Interface, the best form of interface around, of course  Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction  Common Language Infrastructure (See mono or .net)
15:41.06albators~crc
15:41.06jbotcrc is, like, (Cyclic Redundancy Check) A mathematical calculation that produces a number that can be used to check the integrity of a file by regenerating the number and comparing the results.
15:41.47albators~ping me
15:41.48jbotpong albators
15:41.56albators~time
15:41.56jbotYou are educated stupid and therefore too dumb to understand nature's perfect time cube!
15:42.12albators:D
15:42.39albatorscan i crash it ;) ?
15:43.01RoyKalbators: ?
15:43.12RoyK~lart albators
15:43.33ManxPoweralbators: better people than you have tried and failed
15:43.45albatorsnever say never :P
15:44.11albatorseveryone has a bug :P
15:44.18albatorsis just matter of time to be found :P
15:45.16*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
15:45.28jeremywhitinghi all
15:46.07jeremywhitingjust checking to see if I've got this right, if I want to dial from one asterisk  to another asterisk to a zap channel that may or may not be in use
15:46.28jeremywhitingshould I just dial the second box and have alternate plans if dial doesn't work e.g. n+101, etc
15:46.59jeremywhitingor do you have to do chanisavail in there somewhere?
15:47.44MikeJ[Laptop]chanisavail is not necessary, just let it roll baby...
15:48.13jeremywhitingand if there's no channel available on the second asterisk box for dialing out, would the first dial come back to n+1 or n+101?
15:48.27*** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc)
15:48.35jeremywhitingor do I tell it which when the second asterisk box dial doesn't work with a busy or something?
15:48.53*** join/#asterisk logicalonline (~Ken@border.logicalonline.com)
15:49.00kaldemarit comes to n+101.
15:49.12jeremywhitingok, thanks
15:49.27kaldemarare you familiar with Noop?
15:49.41kaldemarquite useful for printing debug information.
15:49.48alphaquejeremywhiting: it goes to n+101 on the /second/ asterisk box
15:50.03jeremywhitingso Box1 Dial(IAX2/box2) -> Box2 Dial(Zap/g1) fails then how do I kick it back so box1 knows it failed
15:50.10jeremywhitingor was busy or whatever
15:50.19alphaquejeremywhiting: the first dial succeeded to the second box
15:50.31kaldemaroh yes, but if you have Busy in the second box, it comes to n+101 on the first one.
15:50.58jeremywhitingkaldemar: really, I sure hope so, that would be nice
15:50.58alphaquejeremywhiting: like kaldemar says, use Busy() or Congested() in the second box which would cause the first to go to n+101
15:51.07jeremywhitingok, got it
15:51.08jeremywhitingthanks
15:51.21*** part/#asterisk logicalonline (~Ken@border.logicalonline.com)
15:51.22jeremywhitingyou guys really are the best
15:51.31alphaquejeremywhiting: aww shucks. :)
15:51.32kaldemari tried that chain with several boxes some time ago.
15:52.07jeremywhitingyeah, the boss wants to use the cheaper lines for long distance from the home office while at the remote office if he can
15:52.11*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
15:52.15jeremywhitingtrying to save some $$ to pay me with I guess
15:52.25*** join/#asterisk jansen (~Jan@p5487894B.dip0.t-ipconnect.de)
15:55.29DarthClue~cluecon
15:55.29jbotrumour has it, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
15:55.44file^^^ Everyone should go that ^^^
15:56.42ManxPowerEveryone should stop spamming the channel about Cluecon.
15:56.51mutilatorNOOOOOOOOO
15:56.58mutilatorNOT CLUECON!
15:57.18ManxPowerIn fact everyone should boycott Cluecon if they continue to spam the mailing lists and the channel.
15:57.29mutilatorheh
15:57.32*** part/#asterisk Myshenka (~gunde@217.9.101.85)
15:57.33DarthClueManxPower: it would appear that your definition of spam is anything that mentions anything you are interested in, am i right?
15:57.41cpatryManxPower: i think u really have something against cluecon.
15:57.45ManxPowerDarthClue: only partially.
15:57.47anthmManxPower, perhaps everyon involved in cluecon should boycott asterisk
15:57.53Malthuslol
15:57.57*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
15:58.15ManxPowerDarthClue: Three messages to asterisk-users from bkw_ about the SAME speaker at cluecon exceeded my annoyance threashold.
15:58.26mutilatorperhaps i should go eat some hotdogs
15:58.34ManxPowerThat and the fact that he seems to send 2 or so messages (seperate messages) about EACH AND EVERY speaker.
15:58.36DarthClueit would also appear that some people drink too much of the asterisk voodoo and fail to realize just how much the cluecon sponsors actually do for asterisk.
15:58.57ManxPowerDarthClue: Sangoms?
15:59.04ManxPower..er..Sangoma does a lot for Asterisk?
15:59.11DarthClueManxPower: anthm, bkw_, file, etc.
15:59.15anthmperhaps that's because we get new speakers every day and cannot predict the future
15:59.33anthmhow can we send an email today telling you who decided to speak tommorow
15:59.40ManxPowerDarthClue: This must be some previously unknown definition of "sponsor".  Perhaps you meant "organizers"?
16:00.11ManxPoweranthm: perhaps a simple web page listing all the speakers and/or one e-mail with all the speakers.
16:00.23ManxPowerI certinally don't get 2 e-mails about every speaker at Astricon
16:00.44bkw_ManxPower, you do realize thats a bug in the mailing list software
16:00.49zoai get 9 for every speaker :)
16:00.53anthmwell I've been told the mailing list has a bug in it and sends the same email more than once
16:00.55bkw_I sent it once to all three lists
16:01.01DarthClueManxPower: 3?  why is it that I only see one?
16:01.03bkw_when you do that it triggers the bug
16:01.14MalthusI see two
16:01.16ManxPowerbkw_: no, I did not know that.
16:01.17bkw_it causes it to dupe all messages 3 times
16:01.28bkw_three per list or something stupid like that
16:01.30bkw_look at the title
16:01.31zoalook at the subjects, they are slightly different
16:01.35bkw_yep
16:01.37*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
16:01.37ManxPowerIt's still not needed to send a seperate message about each and every speaker.
16:01.55bkw_ManxPower, I would have announce them together but at the time I didn't know that Nix was going to show up
16:01.57bkw_or I would have
16:02.03ManxPowerIf Astricon did that there would be riots in the street.
16:02.18bkw_Lets count the 500+ emails I got from astricon about the hot tub
16:02.23bkw_that went on for a week
16:02.27bkw_now that was anoying
16:02.35mutilatorsend me a hottub!
16:02.39Juggienothing wrong with a hot tub
16:02.49mutilatorheh
16:02.49*** join/#asterisk zimba02 (~zimba02@200.39.213.62)
16:02.54bkw_not at all but when I got 500 emails about it over a week it was pretty anoying :P
16:02.56Juggiei got drunk in a hottub sat night, good times.
16:03.13anthmwe won't spam the list anymore
16:03.20anthmafter aug5
16:03.21ManxPowerI still have 24 messages in my trash folder about cluecon since July 7
16:03.25zoa:)
16:03.35anthmthen we will resume in sept for cluecon2006
16:03.40ManxPowerFortunatly the mailing list issue is nolonger an issue since updated my .procmailrc
16:03.42bkw_ManxPower, it wasn't intentional to annoy anyone.. so i'm sorry if it did... I'll try not to annoy!
16:03.52*** join/#asterisk Tili (~Tili@202-133-65-107-dialup.sat.net.pk)
16:04.04ManxPowerbkw_: my level of annoyance is less since you informed me about the mailinglist bug.
16:04.13anthmmaybe you should give up on hating it since it's a collection of all the ppl who give you the code you use to run your business !
16:04.13zimba02hi all, i have a question to make asterisk send tcp information, is this the right place to ask?
16:04.16ManxPowerBut still, isn't your cluecon advertizing a little excessive?
16:04.17bkw_i'll see if we can get them to update that software to fix that
16:04.48bkw_ManxPower, not any more than Astricon really
16:04.54anthmnope it's a small price to pay for all the stuff we do for free all year
16:04.58anthmfree
16:05.13DarthClueManxPower: not really.  If you really are tired of it, pay anthm and he'll make it stop.
16:05.21bkw_hahahaha
16:05.23ManxPowerDarthClue: How much?????????????
16:05.41Juggieanthm, did you ever hear of an indian giver?
16:05.51Juggiethats someone who gives away something for free, and expects something in return
16:06.03ManxPowerif I can't afford it, maybe I can create a non-profit to find the "no clie-con" messages.
16:06.07tomtom_anyone has experience with channel grouping in zapata.conf?  when i register to two groups (eg: group = 1,0) it doesn't seem to work ...
16:06.07Juggieyou work in open source, be happy if you get anything.
16:06.09bkw_Juggie, you're just mad because we won't give away confcall :P
16:06.32Juggiebkw, my meetme+agi solution is working just fine
16:06.37bkw_hehe
16:06.38Juggiei'll use that for years before i pay for open source
16:06.42fileconfcall is sexy
16:06.44*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
16:06.44anthmyah so what does that make you an "indian taker"
16:06.50bkw_hahahahha
16:07.04bkw_well to be honest people that take and take and take are worse than we are
16:07.08bkw_we give and give and give..
16:07.16bkw_atleast people could say thanks
16:07.17bkw_but they don't
16:07.24eKo1thanks
16:07.27Juggiemy thanks is helping people in here on irc
16:07.30Juggieand answering questions
16:07.31ManxPowerA Cluecon announcement is in the /topic.  If someone needs jbot to vomit up the announcement to the channel for people....well you don't want those people to come to the con anyway.
16:07.32Juggiethats my thanks
16:07.45Corydon-wAnything further on the copyright infringement alleged by xoasis on the dev list?
16:08.01bkw_Corydon I was wondering about that also
16:08.08Juggieand thats more then 99% of the people do.
16:08.19bkw_no
16:08.22ManxPowerCorydon-w: I suspect it's the same crap.  No proof that they have not purchased a commercial license.  No comment from Digium, not even a mention of trying to report the issue to Digium before posting to the mailing list.
16:08.23bkw_1% of the people do that
16:08.26bkw_99% take and make
16:08.34bkw_while the 1% gives back
16:08.36Juggiebkw, we just said the same thing.
16:08.41anthmManx, c'mon
16:08.47DarthCluetomtom_: HEAD or STABLE?
16:09.04Juggiei said, i come in here, i help people, i give back where i can, thats more then 99% of people do.
16:09.08drumkilla[work]I have forwarded that email from the -dev list to the Digium licensing director so that it can be addressed through our lawyers
16:09.10bkw_ManxPower, I took the sysmaster thing to digium months before I let the beans spill
16:09.11Corydon-wManxPower: yes, but there's no license that Digium can make to remove copyright notice
16:09.14Juggieand you just said the same thing.
16:09.31ManxPowerbkw_: I wasn't referring to you in this case *tease*
16:09.33Juggiebkw, so we agree there then :)
16:09.35bkw_hehe
16:09.51*** join/#asterisk Zeeek (~Zeeek@Zeeek.active.supporter.pdpc)
16:09.54bkw_Juggie, agree that the world is full of selfish pricks that could care less?
16:09.54bkw_sure
16:09.55ManxPowerCorydon-w: have you seen their commercial license?
16:10.09ManxPowerCorydon-w: I haven't.
16:10.11Zeeekhave I come at a bad time? I have a question for bkw_
16:10.11Corydon-wdrumkilla[work]: sure would be nice to know that the lawyers are doing SOMETHING about the infringement, even if we have no details
16:10.24bkw_if digium doesn't do anything thats just bad
16:10.27Silik0nis it ever a bad time to come
16:10.28bkw_I suspect they will
16:10.31anthmhmm so Juggie we donate like $100k in services and code per year and we want people to come and learn more about voip and that makes me an indian giver?
16:10.35Corydon-wManxPower: you cannot remove a copyright notice, period, except if you have permission from the copyright owner
16:10.48ManxPowerbkw_: Then the people that DO have copyright claims to pieces of Asterisk can start sueing 8-)
16:10.51Zeeekthis is about premiumgoldplatinumpbx?
16:11.13anthmI'd love to hear some more about that
16:11.15Zeeekfile you might know
16:11.17Juggieanthm, the part i was referencing was the bombardement of advertising.
16:11.34fileI know lots of things.
16:11.40bkw_nobody says anything when we bombard cvs with patches and such
16:11.48bkw_double standard
16:11.50bkw_:P
16:11.53ZeeekI have a friend who wanted to try HEAD but wanted to be able to revert to STABLE 1.0.6 if it didn't work out
16:12.02ManxPowerbkw_: we would if they were posted to -users or -dev and posted to the channel.
16:12.02filebkw_: let's get the anthm contribution list!
16:12.05bkw_Zeeek, it will wokr
16:12.17anthmoh like the advertising that comes with free help and megabytes of free code?
16:12.27Zeeekso he tried HEAD had some problems and went back, removed the modules redid make install on the old STABLE. But...
16:12.41anthmas i said it's takers we are talking about not givers
16:12.47bkw_cd /usr/include
16:12.49bkw_rm -rf asterisk
16:12.52bkw_before you go back to stable
16:12.59ManxPowerjbot!*@* added to ignore list.  <-- that should help
16:13.09anthmmanx you are silly
16:13.16ManxPowerbetween /ignore and .procmailrc perhaps I'll see less cluecon stuff.
16:13.18twisted[work]hussha
16:13.18KattyManxPower: silly rabbit
16:13.20ZeeekMy "friend" had a shitload of wacky linux issues suddenly because apparently HEAD Makefile writes to /etc/modules.conf alnd all that and it kept wanting to load wctdm and all that
16:13.22DarthClue~cluecon
16:13.22jbotit has been said that cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses.
16:13.32NuggetLinux is poo.  :)
16:13.37ManxPowerZeeek: 1.0.x also writes to that
16:13.50*** join/#asterisk jhava (~icechat5@200.58.26.21)
16:13.50Zeeekbkw_ ah ok if that was it it's good to know. I'm sure my "friend" will be happy
16:14.01zimba02need some help making a small tcp application work with asterisk, any help appreciated
16:14.09Zeeekbut in the meantime there were no phones inthe office for two hours
16:14.51ManxPowerzimba02: The only TCP stuff in Asterisk is for manager interface and remote console.  All VoIP is UDP.
16:15.05DarthClueCluecon is the only conference where you won't have to hear ManxPower give a presentation.
16:15.07Kattyhmm, i've noticed my polycoms acting funny
16:15.10ZeeekManxPower "my friend" was confused because he doesn't use /etc/modules but has all the modprobe stuff in rc.d/rc.local
16:15.18eKo1DarthClue: hahaha
16:15.20DarthClueKatty: you shouldn't have let bkw_ touch them.
16:15.24Kattyfunny strange, not funny haha..................after awhile, you can pick up, but not hear anything.....but the other person can hear you
16:15.30*** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl)
16:15.31Kattyrebooting the phone seems to fix it
16:15.32anthmunless he want's to he's welcome of course
16:15.40bkw_yep
16:15.44bkw_I think I offered
16:16.10anthmbut he's gonna go ignore us and not give us the same respect we give him I think
16:16.21Zeeekthen one of my firned's providers suddenly switched domains with no notice excapt all calls were answered by "your are calling the wrong server... please change to ..."
16:16.22anthmthat's too bad
16:16.31DarthClueNo respect I tell ya, no respect!
16:16.36KattyRESPECT
16:16.41Kattyshow you what it means to me!
16:16.42Silik0ngod thats a blantant rip off of FOP
16:16.43Kattydo do do do do do
16:16.44mutilatorsing it!
16:16.50ZeeekR E S P E C T Fine Out what it mean to me
16:16.54twisted[work]Katty, keep your do do to yourself ;P
16:16.54zoaid go, but it cant make it
16:16.55zimba02i'm looking to acomplish the follow, I have written a small c program to send information to an external application via tcp, i need asterisk to send cdr informatin over tcp
16:17.05Zeeeksocket.2me socket.2me socket.2me socket.2me socket.2me
16:17.12*** join/#asterisk brimstone (~brimstone@brimstone.digium.sponsor.pdpc)
16:17.15Kattytwisted[work]: pffft
16:17.20Zeeekoh shiw I won't be able to sleep now
16:17.27filezimba02: you can write custom CDR handlers ya know
16:17.33twisted[work]Katty, heheh ;)
16:17.41cpatrycdr_custom.c
16:17.48ManxPowerNo, sweetie, I'm ignoreing the conference.
16:17.51ManxPowerNot YOU.
16:17.58Kattyi'll pun you in a minute.
16:18.03twisted[work]promise? :P
16:18.06Kattyno
16:18.06anthmthe conference is all me!
16:18.08JerJercpatry:  blah
16:18.09twisted[work]lol
16:18.14zoa:)
16:18.24anthmanyone doesnt like it it's all my fault
16:18.31JerJercpatry:  anyone can register their own cdr handler by simply calling one asterisk API function
16:18.31ManxPowerI don't give a flying fuck about the conference, but anthm and the rest have made significant contributions to Asterisk.
16:18.35Kattyit's always your fault, anthm
16:18.40KattyYOUR FAULT
16:18.49bkw_yeppers
16:18.51DarthClueCome to Cluecon.  The only conference with a high female / geek ratio!
16:18.53bkw_JerJer, is right
16:19.00*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
16:19.02Zeeek3 to 2000 ?
16:19.04ManxPowerThe only time I ever put bkw_ on /ignore is when he's being an asshole.
16:19.11ManxPower8-)
16:19.13DarthClueZeeek: even better.
16:19.17bkw_I am never an asshole.. i'm always a bitch
16:19.20Zeeek4 : 2000 ?
16:19.20*** join/#asterisk Shaneful (~sharper@d154-20-37-11.bchsia.telus.net)
16:19.22bkw_:P
16:19.25Silik0ndamnit DarthClue enuff spamming about cluecon already
16:19.27filebkw_: very much so
16:19.31cpatryanthm: we have to go in city, ive a large room for myself alone! :P
16:19.32Silik0nwe get it
16:19.34NuggetI can stay home and enjoy the perfect equilibrium of a 1/2 female/geek ratio.
16:19.41ManxPowerSilik0n: Just put jbot on /ignore
16:19.58DarthClueZeeek: at least 3:40
16:20.05KattyDarthClue: how many now?
16:20.13Kattyoh
16:20.15Kattyk
16:20.19JerJerwe need some conf whores to show up
16:20.21zimba02i know, i need to solve a billing solution for a hotel, they have their own billing system reading cdr from serial port or old pbx
16:20.22DarthClueat least, but i don't have the list.
16:20.24Zeeekjeeze the estrogen ration will be really heavy! not.
16:20.42DarthClueJerJer: if enough people come, we can always get some high quality girls.
16:20.46filezimba02: so write an asterisk module to do it
16:20.57Silik0nits not that everyother word out of that boy is cluecon... I definatly understand the need to promote the conference and I sincerly hope it does well, but damn there becomes a point where its bocming annoying and counter productive
16:21.04JerJerplan a rave for the last night    :)
16:21.11ManxPowerHey!  Maybe we can have conference advert free-for-all!  In this corner is Astericon, in this corner is Cliecon, in this corner is VON!  They can duke it out on the channel, damn the annoyance!
16:21.15*** part/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
16:21.16zimba02that's where I need some help or good links to get started
16:21.16zoa<zimba02> i know, i need to solve a billing solution for a hotel, they have their own billing system reading cdr from serial port or old pbx -> we made such a tool some months ago
16:21.32ShanefulIs there a calling card application that allows a caller to make as many calls as they need . something like when done with your current call hit pound for 3 seconds to dial the next?
16:21.39Silik0nnow if he was staying STRIPPERS AT CLUECON
16:21.45Silik0nthat would be a different storry
16:21.51Silik0nbut I havent heard shit about the strippers yet!@
16:21.52twisted[work]Katty, after the help i've given you you be mean?
16:21.53Cresl1nCome to RussellCon2005!
16:22.02zimba02i have written a c program and work fined, i tried to "incorporated" to asterisk and got lost.. lol
16:22.04cpatrymouhjahahha
16:22.05DarthClueSilik0n: if enough people come to cluecon...there will be STRIPPERS AT CLUECON!
16:22.07Cresl1nit's the only conference that's all about russell
16:22.09Cresl1n:-)
16:22.12Kattytwisted[work]: i'm always mean
16:22.12twisted[work]DarthClue, hmm
16:22.14Kattytwisted[work]: VERY mean
16:22.20bkw_but keep the divorce lawyers handy too
16:22.24twisted[work]AHH
16:22.24Silik0nCresl1n i thought that was geekcon
16:22.26brimstoneCresl1n: ?
16:22.28eKo1STRIPPERS!? where?!
16:22.33brimstonesorry, wrong window again
16:22.36zoahey creslin
16:22.38DarthClueand for an additional fee, we cover the tips and lawyers fees.
16:22.38Silik0nbkw_: what happens in chicago stays in chicago
16:22.42Cresl1nzoa!!! :-)
16:22.45zoathey promised to do it on the 22nd
16:22.47bkw_Silik0n, ok
16:23.16Silik0nbkw_ be sure to let me know if you are really going to do strippers so I can break out the stripclub MP3s and bring the right stuff
16:23.23zimba02do you have any links or info where to go and searcho for the tool made months ago?
16:23.34filenights are open to go to a stripclub :P
16:23.34ManxPower"Candy" and her twin sister "Boffy" will be the "hosts" for the conference party!
16:24.06mogormanrusselcon would be cool
16:24.10mogormanall stable all the time
16:24.16ManxPowerYeah!
16:24.22anthmoh sure, fuck telephony, if it's about strippers we can go on all day talking about it!
16:24.24Cresl1nrussellcon will be a big room that we rent out for 3 days
16:24.31drumkilla[work]that's hot
16:24.49mogormanand all stable bugs get kicked around
16:24.52Cresl1nwith a steady supply of red bull and food sent to it
16:24.58mogormanuntil 1.2 rumbles out of the ashes
16:25.03Kattyi wonder if redbull is vegan
16:25.11Cresl1nI'm sure it can be
16:25.11ManxPowerdrumkilla: my Dutch Red Bull can was squished in my luggage.
16:25.12Cresl1n:-0
16:25.18drumkilla[work]Katty: I don't think it is
16:25.19Cresl1nsucks
16:25.21twisted[work]dude
16:25.24Kattydrumkilla[work]: :<
16:25.28twisted[work]Taurine isn't vegetarian
16:25.37mogormanyeah
16:25.39mogormanexactly
16:25.39Cresl1nit's just a chemical, no?
16:25.47zimba02chat zoa
16:25.47drumkilla[work]Cresl1n: that's not what I heard ...
16:25.47anthmwe can guarentee strippers....
16:25.50twisted[work]Cresl1n, yes, but it's created by bull testicles
16:25.54anthmthe develpers cam
16:25.55Cresl1nnot an animal product
16:25.55Cresl1nooh...
16:25.56Cresl1nwow
16:26.01anthm"strip the code"
16:26.04twisted[work]go look up taurine
16:26.04anthmstrip asterisk
16:26.07Nugget]:8) No Bull
16:26.12Kattyhi Nugget!
16:26.13anthmyay more efficient
16:26.15twisted[work]anthm, talk to drumkilla about releasing astwipe
16:26.16mogormanmy virgin eyes
16:26.24drumkilla[work]you'd think they were using a synthetic version now though
16:26.29Cresl1nI heard that asterisk is made of moose testicles
16:26.29bkw_virgin eyes my ass!
16:26.44mogormanlol
16:26.45Kattygosh, i'm not looking
16:26.47bkw_Cresl1n, I have a good picture of you
16:26.48mogormanbkw_ you know nothing
16:26.48ManxPowerCresl1n: Only CVS-HEAD.
16:26.58twisted[work]drumkilla, they may be... but i dono
16:27.04bkw_http://www.bkw.org/photos/spring_von2005/100_0328.html
16:27.06Kattyin fact, i'm going to lunch
16:27.13twisted[work]enjoy, Katty
16:27.17zimba02irc question, zoa replied in red to my question, how do I reply the same way?
16:27.34bkw_here is Cresl1n under the table http://www.bkw.org/photos/spring_von2005/100_0229.html
16:28.03wasimzimba02: you need to be in debt
16:28.05*** part/#asterisk zpn (~xpn@207.111.174.1)
16:28.10ManxPowerI dunno what it is about Cresl1n, but he always seems to look stoned.
16:28.18*** join/#asterisk zpn (~xpn@207.111.174.1)
16:28.18fileManxPower: pfft
16:28.19cpatryi prefer twisted on fire: http://www.bkw.org/photos/spring_von2005/100_0294.html :P
16:28.24bkw_here is zoa and Cresl1n http://www.bkw.org/photos/astricon_2004/100_0921.html
16:28.38zimba02sorry, don't know what you mean by debt (that I am with my bank.. lol)
16:28.43Cresl1noh yeah, my glasses broke
16:28.51Cresl1nthose ones I bought in san francisco
16:28.58fileCresl1n: that's sad
16:29.02bkw_you had no need for those gay ass glasses
16:29.03bkw_really
16:29.06bkw_its better they broke
16:29.19bkw_here is kram http://www.bkw.org/photos/astricon_2004/100_0920.html
16:29.23bkw_look at that look
16:29.27bkw_haha
16:29.28*** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
16:29.50cpatrymouhaha
16:29.54bkw_here is blitzrage http://www.bkw.org/photos/astricon_2004/100_0807.html
16:29.57*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com)
16:30.12ManxPowerI like the zoa/Cresl1n tongue pic better.
16:30.19bkw_<PROTECTED>
16:30.21bkw_it was funny
16:30.23ManxPowerAsterisk Porn!
16:30.32bkw_the security guard was getting on me for taking pictures of the Marta
16:30.49bkw_"its a security issue"
16:30.55bkw_bitch please
16:31.21mogormansomeone hasnt google asterisk on google images
16:31.35zoai made a new tongue version with creslin on the last astricon
16:31.36zoa:)
16:31.52*** join/#asterisk cfrank (~cfrank@bi01p1.co.us.ibm.com)
16:31.55bkw_haha
16:35.31ManxPowerMaybe I should do a talk on Asterisk Urban Myths at Astricon/Calif.  Things like "reinvite=", "the TDM card is stable", and "SIP and NAT are hard".  Total lies like that need to be dispelled.
16:36.10*** join/#asterisk [Jedi] (~fdsafasdf@213.162.200.226)
16:36.11cfrankbkw_: Is dundi dying?
16:36.16[Jedi]Hello
16:36.18ZeeekManxPower you used to say SIP and double NAT were hard
16:36.24*** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl)
16:36.25Malthuswhat's the reinvite lie?
16:36.26[Jedi]*CLI> pri show span 2
16:36.27[Jedi]Primary D-channel: 47
16:36.27[Jedi]Status: Provisioned, Down, Active
16:36.35[Jedi]how can i put it 'Up'?
16:36.54DarthClue[Jedi]: call your provider?
16:36.56[Jedi]the remote end is supposed to be enabled
16:37.38ManxPowerZeeek: until I tried them 8-)
16:37.42cpatryu sure the cable is right connected? (just in case) :)
16:37.47[Jedi]yes
16:37.49mkrufky[Jedi]: what is on the OTHER end of the cable?
16:37.56[Jedi]mkrufky: a #5ess
16:38.01ManxPowerIt took me an hour and that was mostly reading up on DNS SRV records. 8-)
16:38.02*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
16:38.02*** mode/#asterisk [+o anthm] by ChanServ
16:38.21ManxPowerMalthus: The sip.conf option reinvite= does NOT exist.  Check the source code.
16:38.47Zeeekwhat about makeitworkplease=yes ?
16:39.01ZeeekI've used that to advantage
16:39.09*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
16:39.22Zeeekhavereadwiki=yes
16:39.23DarthClueyoutookmystapler=yes?  burndownthebuilding=yes?
16:39.35mkrufky[Jedi]: i had a problem like that when i was connecting to another pbx.... Make sure that your asterisk box is set to CPE if the PBX is set to NET ... (or vice versa -- MY pbx is set to CPE, so I had to set the asterisk box as NET)
16:39.40DarthClueiamyourfather=nooooooooooooooooooooooooooooooooooo?
16:39.46Zeeekheh
16:40.00Zeeekok I submit to the jury another situation
16:40.11Kattymy phone won't work
16:40.12KattyIT WON"T WORK
16:40.14outtoluncwhatshouldthechanneldo=deadlock <G>
16:40.16Kattyfix it
16:40.30[Jedi]mkrufky: my "PBX" is a big 128 E1 switch... and it's suposed to be NET
16:40.42mkrufkyZeeek: I like your fix better.... now let's implement it :-D
16:40.44Malthusoh
16:40.45ZeeekI call the provider and the audio goes both ways between my SIP phone, asterisk and their SIP server, hence codecs and the rest are fine. Two-way audio and DTMF are fine
16:40.47mkrufkyj/k
16:40.55*** join/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com)
16:41.03zimba02zoa: do you have any info or link where to find the tool to send tcp information from asterisk?
16:41.15ZeeekBut a call to a phone number from this same provider give a call that rings wand when answered, no audi either way
16:41.21mkrufky[Jedi]: so your asterisk box should be set as CPE ... just make sure that it already is set that way -- just a suggestion
16:41.29ZeeekThis HAS to be their problem, right? (many users complaining as well)
16:41.51Zeeekthey keep saying, yeah, well, asterisk blah blah, port forwarding
16:41.56[Jedi]mkrufky: that's in zapata.conf, right?
16:42.00twisted[work]cpatry, i don't even remember that picture
16:42.09[Jedi]signalling=pri_cpe
16:42.13cpatrytwisted: so that's really bad then :)
16:42.25twisted[work]cpatry, lol
16:42.42twisted[work]cpatry, i don't remember what was going on there... but apparently it was on the exhibit floor while it was open
16:42.47*** join/#asterisk santiago (~santiago@63.245.86.188)
16:42.50twisted[work]so i'm kinda scared to think what I might have been doing
16:42.53*** part/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com)
16:43.08mkrufky[Jedi]: yes... zapata.conf should have -- signalling=pri_cpe
16:43.56cpatrymaybe you're already on playgirl then :P
16:44.17cpatryapparently, i need to taste to shots too.
16:44.24*** join/#asterisk grolloj (~grolloj@slim-eth0.horizonlive.net)
16:44.54*** join/#asterisk MustDie (voip@205.247.13.252)
16:45.23*** join/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com)
16:46.11mkrufky[Jedi]: you also need switchtype=5ess ... but i bet you already know that ;-)
16:47.05puzzled_anyone happen to remember that nice webpage about tweaking pci interrupt latencies to help cards work better with asterisk?
16:48.01Kattyshould i update my 2.4 kernel?
16:48.05zoamy asteriskguru page ?
16:48.17Silik0nKatty: update to 2.0.36
16:48.18*** part/#asterisk aminorex (~aminorex@66-191-69-132.dhcp.dlth.mn.charter.com)
16:48.21puzzled_zoa: could be. lemme check
16:48.25zoahttp://www.asteriskguru.com/tutorials/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
16:48.27zoais this the one ?
16:48.28Katty:<
16:48.29Hmmhesays2.4 to what?
16:48.38Kattyisn't 2.6 out?
16:48.40Silik0n;)
16:48.44puzzled_zoa: yup, thanks
16:48.48Hmmhesaysare you having issue's caused by the 2.4 kernel?
16:48.54KattyHmmhesays: no (=
16:48.58Silik0nkajtzu yeah
16:49.00*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
16:49.04Silik0nerr katty yeah 2.6 is out
16:49.10Silik0n2.6.11 even
16:49.15HmmhesaysI personally would not if you have no reason other than just to upgrade
16:49.18Kattyk
16:49.31Hmmhesaysespecially on a production box.... if it's a testing box, go for it
16:49.32mkrufkySilik0n: 2.6.12 is the current mainline kernel
16:49.56mkrufky...and 2.6.13-rc2 is current stable prepatch
16:50.01*** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172)
16:50.26mkrufky...and 2.6.13-rc2-mm2 is current dev tree (released today)
16:50.28Hmmhesayshrm, the more I play with perl, the more I like it
16:50.37KattyHmmhesays: how do i know when to update asterisk?
16:50.45Silik0nhah
16:50.50mkrufkyif it's not broken, dont fix it
16:50.55Hmmhesaysproduction or testing box?
16:50.56Silik0nKatty when you find something b0rked :)
16:50.56Kattymkrufky: that's a male attitude
16:50.59mkrufkylol
16:51.12*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
16:51.15KattyHmmhesays: it's not a test machine anymore
16:51.19mkrufkyi agree with you... i only feel that way on my PRODUCTION asterisk box
16:51.20KattyHmmhesays: it omgworks
16:51.21DarthCluekatty: just bring it to cluecon, we'll do it for you.
16:51.27Silik0nit might be a male attitude, but for a production box, thats always a good course of action
16:51.27KattyDarthClue: shan't
16:51.31[Jedi]mkrufky: no, my switchtype is a EuroISDN
16:51.36Hmmhesaysmy rule of thumb for production boxes is, don't update unless you find something that needs fixing that an update will fix
16:51.37Silik0nunless a newer version happens to have a feature you need
16:51.39KattyDarthClue: keep your dirty paws off my silky......err, asterisky ...uhh, box
16:51.39[Jedi]mkrufky: althought it is a lucent 5ess, it's using euroisdn signaling
16:51.56KattyHmmhesays: k
16:52.05KattyHmmhesays: so...something equivilent of doing windows updates.......
16:52.12KattyHmmhesays: is......non existant?
16:52.18mkrufky[Jedi]: OOoooh ... ok I'm not familiar with that stuff... hopefully someone else in here knows
16:52.22KattyHmmhesays: whether it's the kernel, or asterisk?
16:52.26Kattyor.......debian
16:52.42KattyHmmhesays: is there /anything/ i should be updating on a fairly regular basis? :P
16:52.50Hmmhesayssecurity patches for debian
16:53.19SpaceBassdoes debian have anything like RH's up2date or what ever?
16:53.26mkrufkyapg-get
16:53.28Hmmhesaysapt
16:53.29mkrufkyoops
16:53.31mkrufkyapt-get
16:53.33Hmmhesaysit has super cow powers
16:53.44KattyHmmhesays: how.....do i do that?
16:53.55mkrufkyKatty: type man apt-get
16:53.56SpaceBassbut will apt-get run as a damean or something that notifis you when security updates are available?
16:53.58KattyHmmhesays: apt-get update alapeanutbuttersammiches?
16:54.23mkrufkySpaceBass: apt-get CAN be set to run regularly by cron
16:54.30SpaceBassgotcha
16:54.34Hmmhesaysbrb
16:54.51SpaceBasssimilar to yum?
16:55.22mkrufkyVERY similar... but i like apt better
16:55.37mkrufkyi like apt SO much better that i even use it on my fedora boxes
16:55.46SpaceBassmy only problem with yum- and is apt-get the same? - is that i never know exactly what the package is called
16:55.49JerJertar zxf filename.tgz ; cd filename ; make install  <--- the only way
16:55.56anthm"if it ain't broke, don't fix it" is a good policy in the computer industry and is not a "male" thing
16:56.02SpaceBassfor instance... I needed gnome-kerberos and thats what its call everywhere i've ever seen it
16:56.06MalthusSpaceBass: apt-cache search
16:56.08SpaceBassbut yum couldt find it
16:56.10SpaceBassahhh
16:56.19anthmbecause computers tend to break the most whilst being "fixed"
16:56.20mkrufkySpaceBass: apt-cache search gnome kerberos
16:56.32zimba02looking to replace a panasonic pbx with asterisk in a hotel that reads cdrs via serial cable with the following solution
16:56.33SpaceBassgood to know
16:56.34SpaceBassthanks
16:56.34mkrufkyMalthus: :-)
16:56.44Kattyanthm: boy, did i get into the wrong field ;)
16:56.55alphaqueanthm: aint that the truth !
16:57.07Malthusmkrufky: :)
16:57.14JerJerzimba02:  you could make asterisk crap out cdr's on a serial interface
16:57.36anthmKatty, nah it's just a fact you come to terms with after doing it long enuf
16:57.37jeremywhitinghi all, anyone know how to tell if a line is connected to tdm04b card from asterisk cli?
16:57.46ManxPowerjeremywhiting: no.
16:57.49JerJerjeremywhiting:  zttool
16:57.52Kattyanthm: yeah i know, that's what i do to the windows server :/
16:57.53ManxPowerYou should be able to tell via zttool.
16:57.55*** join/#asterisk InfraRed (~bigboss@master.subhi.com)
16:58.03ManxPowerYou can also post a bounty to PAY someone to add that basic feature.
16:58.04jeremywhitingis zttool the only way?
16:58.08Kattyanthm: which is why it unnerved me when that guy from out here starting poking about on it
16:58.10ManxPowerjeremywhiting: yes.
16:58.11cpatryjeremywhiting: zap show status?
16:58.20wasimcat /proc/zap/*
16:58.21InfraRedhi all
16:58.25zimba02jerjer: thanks, the billing system is related to room checkout, etc.. and the software developer has opened a tcp port to recieve text
16:58.26jeremywhitingcpatry:? no such thing
16:58.40cpatryjeremywhiting: its only in cvs-head
16:58.43[Jedi]i I do intense debug in my PRI, I get "Sending Set Asynchronous Balanced Mode Extended"
16:58.50ManxPowerjeremywhiting: what is your specific issue?
16:58.54zimba02all i need to send is the sql statment to the tcp port, "inster into bla blah"
16:59.06ManxPowerzimba02: you can't do that.
16:59.06[Jedi]what may be happening in my E1?
16:59.08jeremywhitingjust trying to tell if phone line is connected right at a remote location
16:59.17jeremywhitingbecause asterisk isn't picking up when I call the line
16:59.19*** join/#asterisk jfonsecausa (~jfonseca@c-66-176-57-28.hsd1.fl.comcast.net)
16:59.24anthm;delete from blah ;grant me a shell ?
16:59.27InfraRedquestion: I am trying to change my voip termination company, i have asterisk server and 6 extensions, do i need one login per phone on the remote server? or do i just need one account for the asterisk server?
16:59.28ManxPowerjeremywhiting: Asterisk does not support line or dialtone detection on analog cards.
16:59.47jeremywhitingok
16:59.57ManxPowerjeremywhiting: I have that fairly often with one or two of my TDM400P.  Reboot the server (and hope it doesn't crash when unloading zaptel) and prey.
17:00.01ManxPowerand pray too
17:00.05zimba02is not an sql connection, it's just a tcp server reading "instert into...."
17:00.10MalthusInfraRed: just one
17:00.12jeremywhitingis there a way to see debug messages if I get a call in on the zaptel interface
17:00.12wasimInfraRed: one login per server would work, as well as one per line, its flexible
17:00.21ManxPowerzimba02: How, exactly, does this relate to Asterisk?
17:00.25jeremywhitingso after the line is connected I have to restart to get asterisk to use the lines?
17:00.30[Jedi]jeremywhiting: pri debug
17:00.30bkw_ManxPower, yes it does.. if the line is unpluged it goes into red alarm on analog lines
17:00.40ManxPowerjeremywhiting: only if the TDM card blows up.
17:00.44jeremywhiting[Jedi] not using pri I don't think
17:00.49ManxPowerbkw_: That's Zaptel, not asterisk.
17:00.52zimba02manxpower: i will be replacing their existing panasonic pbx with an asterisk box
17:00.58[Jedi]"Status: Provisioned, In Alarm, Down, Active"
17:00.59InfraRedok
17:01.00InfraRedcool :)
17:01.05bkw_you'll get an alarm notice if you have debug on
17:01.06[Jedi]how can I see the kind of alarm on the PRI?
17:01.11jfonsecausaHi, Someone knows haw to dial to a provider of DID's who does not requires registry neither passwoed or userid?
17:01.15ManxPower[Jedi]: red alarm, yellow alarm, or blue alarm?
17:01.15*** join/#asterisk iswm (iswm@iswm.user)
17:01.32ManxPowerbkw_: Sorry, you are correct about that.
17:01.36*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
17:01.38[Jedi]ManxPower: I don't know how to check the alarm type
17:01.43[Jedi]:))))
17:01.48cpatryjedi: "zap show status" give u all that info. (only in cvs-head)
17:02.00ManxPowerActually, I've only see the red alarm on X100P, never on the TDM400P.
17:02.08[Jedi]No such command 'zap show status' (type 'help' for help)
17:02.10ManxPower[Jedi]: zttool
17:02.13*** join/#asterisk zpn (~xpn@207.111.174.1)
17:02.20cpatryu need zttool then.
17:02.22[Jedi]I don't have zttool
17:02.27[Jedi]where's that?
17:02.29ManxPower[Jedi]: What part of "only in CVS-HEAD" are you not understanding.
17:02.30cpatrythen run cvs-head
17:02.31cpatry:)
17:02.40[Jedi]I can't run HEAD
17:02.41ManxPower[Jedi]: zttool will only build if you have newt-devel installed.
17:02.47[Jedi]ok
17:02.49cpatrywhy?
17:02.52[Jedi]then I'll rebuild
17:02.56[Jedi]cpatry: because I need BRIstuff
17:03.04cpatryk
17:03.05*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
17:03.09ManxPowercpatry: If you have to ask "why can't you run CVS-HEAD" then ANYTHING he tells you will be moot.
17:03.20[Jedi]why? =)
17:03.34cpatrynmouhaha
17:03.42jeremywhitingmaybe since I had to patch my wcfxs.c to detect my card it's not working quite right?
17:03.48ManxPowerThe 1.0.x people think that people that run CVS-HEAD are crazy lunatics, and the CVS-HEAD people think the same about people running 1.0.x.
17:03.48jeremywhitingasterisk still doesn't answer the calls
17:03.56*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
17:04.00*** join/#asterisk _ioscanner (~ioscanner@c-67-162-251-133.hsd1.tx.comcast.net)
17:04.05ManxPowerjeremywhiting: perhaps you could try using zttool?
17:04.12gtigeneDoes call parking work with SIP phones? I pressed # and nothing happened.
17:04.15jeremywhitingwhere's that, just in cvs
17:04.16_ioscannerHello all
17:04.37ManxPowerjeremywhiting: If you plug a phone line into a Digium FXS module, the module will blow up at the first call.  The ports may not be numbered as you expect.
17:04.44DarthCluethose of us who choose to run CVS-HEAD know who runs the asylum, we just prefer to have more padding on our walls.
17:04.44ManxPowerjeremywhiting: it's in zaptel.
17:04.45[Jedi]ok, installing newt-devel
17:04.50alphaquegtigene: did u dial() with the t or T option ?
17:04.54*** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com)
17:05.06ManxPowerjeremywhiting: make sure newt-devel is installed, rebuild zaptel, you will have zttool
17:05.06[TK]D-FenderAnyone have experience with Channel banks on lines with DSL on them?  I'm lloking to use a channel bank for and analog failover to my PRI and want to know if have dsl (behind a filter) should pose a problem with it.
17:05.31ManxPowergtigene: Is your SIP phone too brain dead to support a transfer button?
17:05.36_ioscannerI have 1-0-07/05/05 cvs stable code on a box.  Asterisk just restarts at at times.  Seem mostly when I get SIP calls
17:05.53alphaqueManxPower: some cheap sip hard phones dont
17:06.02ManxPoweralphaque: those are brain dead phones.
17:06.12alphaqueManxPower: note i said /cheap/
17:06.22gtigenealphaque: no t or T in my Dial commands
17:06.27wasim[TK]D-Fender: theoretically it shouldn't
17:06.31alphaquegtigene: then do that
17:06.33gtigeneManxPower: my phone transfers calls OK
17:06.51[Jedi]ok
17:06.54[Jedi]I have now zttool
17:06.56[Jedi]my alarms are RED
17:06.58ManxPowergtigene: If your phone does not support supervised transfers then there is no way to do supervised transfers in 1.0.x
17:07.11ManxPower[Jedi]: Red means "I don't see a line, it's a wireing problem"
17:07.27ManxPowerAny time we get red alarms we call the telco and say "we have red alarms" and they fix it.
17:07.48ManxPowergtigene: and you need supervised transfers to work for call parking
17:07.48_DAWHi All..
17:08.02gtigeneManxPower: what is a supervised transfer? Its a Polycom phone, I just press the transf soft key.
17:08.33ManxPowergtigene: You have a polycom phone.  You do NOT need "t" or "T" on your Dial line.
17:08.43ManxPowergtigene: What is the specific problem?
17:08.55ManxPowerI manage 60 polycom phones and we use call parking all the time.
17:09.11_DAW[Jedi] - your prob is with T1/PRI?  Try a loopback plug.
17:09.12jeremywhitingzttool doesn't show anything
17:09.17gtigeneManxPower: I looked up call parking in the Wiki and it said you park a call by pressing #. I press # and nothing happens.
17:09.31ManxPowergtigene: # is only for phones too stupid to have a transfer button
17:09.53ManxPowergtigene: transfer the call to the parking extension, listen to for the parking lot number, press transfer the second time to complete the transfer.
17:10.07gtigeneManxPower: thanks
17:10.16*** join/#asterisk heison (~heison@ns.somanetworks.com)
17:10.19[Jedi]ManxPower: I am the telco
17:10.34ManxPower[Jedi]: then either you have a wireing problem or the port isn't provisioned on the PBX.
17:10.43[Jedi]ManxPower: the people who manages the phone switch to which my E1 is connected says my E1 is right
17:10.58ManxPower[Jedi]: then they are either lieing or you have a wireing problem.
17:11.12ManxPowerperhaps you forgot to make a CROSSOVER cable when connecting to the PBX?
17:11.47[Jedi]the cable is right
17:12.09[Jedi]it's suposed to be right
17:12.17[Jedi]and my asterisk sends SABME messages to the poneswitch
17:12.23ManxPower[Jedi]: plug a loopback connecter into the E-1 line.  If your techs can't see a loop then you can WITH NO DOUBT tell them the line is broken.
17:12.26[Jedi]so it seems the lowest-level is working fine
17:12.42ManxPower[Jedi]: LISTEN TO ME.  Red Alarm means NOTHING AT ALL is working.
17:12.57[Jedi]ManxPower: but the remote switch says it is! that's the problem
17:13.13ManxPower[Jedi]: then you have a bad cable.
17:13.23ManxPowersounds like the TX pair from asterisk is right, but the RX pair isn't.
17:13.25jeremywhitingwhat is zttool supposed to show?
17:13.26_DAW[Jedi] loopback the cable to your * box
17:13.28[Jedi]uhm
17:13.32*** join/#asterisk M_at (~matthewt@81-1-109-38.homechoice.co.uk)
17:14.12gtigeneManxPower: Thanks again that works fine
17:14.16ManxPowerjeremywhiting: Sorry, for a TDM400P zttool does NOT show an error if the line is not plugged in.
17:14.24ManxPowerI just tested it.
17:14.27jeremywhitingoh, it's just for pri cards, etc
17:14.32jeremywhitinggot it, thanks anyway
17:14.33ManxPowersorry, bkw_ you are wrong.
17:14.39bkw_about?
17:14.44ManxPowerjeremywhiting: it's also for X100P.
17:14.49gtigeneHow do you listen in on SIP calls.
17:14.52bkw_the x100p does
17:14.53jeremywhitingnope, I'm using a tdm04b
17:14.55bkw_mine always has
17:14.59JerJergtigene:  chan_spy
17:15.08ManxPowerbkw_: zttool provides no indication for TDM400Ps about if a line is plugged in.  YES, it works for the X100P
17:15.08bkw_I have never had a working tdm board
17:15.09bkw_so I wouldn't no
17:15.11gtigenejerjer: thanks.
17:15.16bkw_s/no/know/
17:15.22alphaqueJerJer: i heard (though i could be wrong) that chan_spy doesnt work in the present -STABLE
17:15.33bkw_alphaque, talk to anthm
17:15.33ManxPowerbkw_: Is it time to tell you to shut up and say you don't know what you are talking about?
17:15.35JerJernobody should be running -STABLE
17:15.41DarthCluealphaque, use HEAD or talk to anthm.
17:15.45bkw_ManxPower, what ever
17:15.50alphaqueanthm: does chan_spy work with 1.0.6 ?
17:15.55bkw_ManxPower, if it doesn't then I would consider that a bug
17:15.57gtigenealphaque, jerjer: I am running cvs head from about there weeks ago
17:16.09ManxPowerbkw_: me too, actually.
17:16.20ManxPowerbkw_: but zttool does not show any port specific information about that card.
17:16.31alphaquebkw_: does it work with 1.0.6 ?
17:16.35DarthCluealphaque: probably not, use HEAD or pay anthm to make it work with STABLE 1.0.9, 1.0.6 is too old.
17:16.41bkw_ManxPower, it might print a warning if debug is up high enough
17:16.42*** join/#asterisk carrar (tim@osburn.com)
17:17.03alphaqueDarthClue: 1.0.6 is what freebsd's ports collection has
17:17.36alphaqueDarthClue: like was discussed earlier, it aint broken for me, so no reason to upgrade to 1.0.9. will consider a bump to 1.2 when it comes though
17:17.42bkw_ManxPower, you also have to realize I don't work with analog very much!
17:17.55anthmwhat's the dealio with the pettyness over who knows what they are talking about I conclude that you are both right depending on what card you are talking about
17:18.11anthmand we are not sure till we test both cards
17:18.18bkw_exactly
17:18.22*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
17:18.23anthmI can attest that my x100p does have a red alarm
17:18.31anthmthe other one I do not have
17:18.32ManxPoweranthm: I only mention it, since when bkw_ was channeling Elvira he always told me to shut up and that I didn't know what I was talking about.
17:18.34[Jedi]uhm
17:18.37[Jedi]now I got RED/LB
17:18.44[Jedi]what's the difference between RED/LB and RED?
17:18.51Hmmhesayscause someone must always have the finer feathers
17:18.51ManxPower[Jedi]: no idea.
17:19.30opus__rtfm?
17:19.46anthmI know read pertains to the physical
17:19.46_DAW[Jedi] RedLB = red alarm in loopback?
17:20.10ManxPower[Jedi]: Um, if you did what I suggested, you would not see that message.
17:20.12anthmno physical layer and yellow/blue is the logical
17:20.17[Jedi]_DAW but I don't have a loopback =?????????
17:20.30ManxPower[Jedi]: perhaps your "telco" is looping back.
17:20.35[Jedi]maybe
17:20.39bkw_the telco might have it in LB
17:20.39ManxPower[Jedi]: call them up.
17:20.53bkw_ManxPower, do tdm boards actually work? :P
17:20.56ManxPower[Jedi]: this is a good thing, if they are.
17:21.13ManxPowerbkw_: They seem to, other than the odd issue with the some of the ports stop working until a reboot.
17:21.18[Jedi]no, they aren't in loopback
17:21.24bkw_wired wrong?
17:21.25[Jedi](they say they aren't in loopback)
17:21.31ManxPower[Jedi]: replace the connectors on the cable.
17:21.32anthmwhat is it hooked up with
17:21.41[Jedi]ok
17:21.45anthmtry both a straight and a cross cable and see which one is better
17:21.56ManxPower[Jedi]: tell your telco to loop the line.  They will fail.
17:22.28anthmyou never know how it's comming in till you mess w it I've see side by side ports where 1 needs cross and the other straight
17:22.36ManxPowerbkw_: I have to reboot our corporate PBX Asterisk Gateway every few days to avoid problems with the TDM400P.  I'll be SO happy when we can yank the card in a week or two
17:23.07*** join/#asterisk BMG (BMG@162.80-202-207.nextgentel.com)
17:23.23ManxPower[Jedi]: Let me know when you finally realize it's a wireing problem.
17:23.32alphaqueManxPower: amen to that. i've not liked the tdm400p's either
17:23.44BMGhey, got a problem with musiconhold, if we call the MOH extention number, that works, but we put a user on hold there is silence
17:23.48alphaqueby comparison, the te410s we have are a gem
17:24.16jeremywhitingBMG, what kind of phones are you using?
17:24.42anthmI have 4 x100's and 1 olden days tdm400 in 1 box back to the aint broke dont fix it thing ;)
17:24.45[Jedi]incredible
17:24.50[Jedi]i've unloaded wct4xxp
17:24.52[Jedi]and reloaded it
17:24.59BMGjeremywhiting: cisco 7905G, SJ-Softphone and Cisco 7920
17:24.59[Jedi]and the E1 goes up
17:25.10Ariel_ok so how do I get this jbot for another channel?
17:25.30[Jedi]I have my E1 working
17:25.35[Jedi]and I've done nothing but reloading the driver!
17:25.48alphaque[Jedi]: without running ztcfg ??
17:25.52terrapenEVERYBODY WANG CHUNG TONIGHT
17:25.56M_atAnyone used the new Grandstreams? GX200s
17:26.01[Jedi]I run ztcfg when I modprobe the driver, in /etc/modules.conf
17:26.06jeremywhitingBMG: you need to find out what the sip phone sends to the server when you press hold on the phone
17:26.10jeremywhitingI think
17:26.22terrapentoday's music theme: 1980s party songs
17:26.30terrapenmusical, even
17:26.41opus__does anyone here use intercom?
17:26.46ManxPower<PROTECTED>
17:26.55[Jedi]ManxPower: why? they don't work well?
17:26.59jeremywhitinghey is Zap/3-1 port 3 on the tdm card? or is that indefinite
17:27.00jeremywhiting?
17:27.05BMGjeremywhiting: ok, any idea how to do that ?
17:27.07newmemberManxPower: what are you moving too?
17:27.10ManxPower[Jedi]: you apparently missed my tirade about 2 mins ago about them.
17:27.18ManxPowernewmember: T-1 cards and channel banks.
17:27.26jeremywhitingsip debug maybe, but I'm not sure exactly if that's the problem
17:27.49jeremywhitingI'm sort of new here too
17:27.57jeremywhitingonly been at asterisk for about a couple months
17:28.23BMGok :) we hear that the call is beeing placed at hold when we press hold, but just lack the music
17:28.48BMGanyone else got any clues ?
17:29.18*** join/#asterisk pa (~Paolo@pa.user)
17:29.38Ariel_BMG, is moh working on your system? did you install mpg123
17:29.52BMGyes, if we call the extention then it works
17:30.02anthmuse native moh perhaps?
17:30.23BMGthats what we are doing
17:30.41anthmthe guy who asked about chanspy, it can be ported to stable but it will reqire consulting fee cos it's a pita
17:31.17bkw_alphaque, was who asked
17:31.44alphaqueanthm: that was me
17:32.05alphaqueanthm: was just asking. i dont use chan_spy, though i am using the native moh patches which are part of chan_spy
17:33.06*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
17:33.11yaaarword
17:34.33yaaaranybody know why my cisco 7940 has no audio? I thought I got the channel setup about right, following the wiki. The phone can dial, and rings the phone it's calling, but neither party can hear the other. On the cisco handset I can only hear a soft 'click' about once every 2.5 seconds. ????
17:35.08yaaarif i try to leave a voicemail on an asterisk mailbox it hangs up after a half-second saying 'no audio on <channel>
17:35.11JerJeryaaar:  firewall and/or NAT is blocking RTP
17:35.15_ioscannerasterisk seem to die when I hang up a call.  I see WARNING CDR on channel 'SIP/100-8c3c' not posted and same with lacks end
17:35.16yaaarnope
17:35.19JerJeryep
17:35.28_ioscannerIs this an Asterisk or mysql problem?
17:35.39anthmusing chanspy or native moh as patches is not advised
17:35.40SpaceBassyaaar on the same lan as the * box? same subnet?
17:35.47anthmthey are not complete
17:35.48JerJer_ioscanner: you are telling asterisk to not record a cdr
17:35.49_ioscannerseem that when it tries to write to mysql it dies
17:35.54yaaarJerJer: this machine is on a public subnet attached directly to the asterisk server via a switch with absolutely no firewall or nat in between. and it's going out on a zap channel
17:35.59JerJeriptables -L
17:36.35_ioscannerI should be loggin the cdr in mysql
17:37.40yaaarJerJer: it just shows the 3 chains (INPUT/OUTPUT/FORWARD) with "(policy ACCEPT)" after each one and no rules following.
17:38.12[Jedi]If I want to make an application which records voice, I should use EAGI?
17:38.14[Jedi]or what?
17:38.16JerJerthen you have a codec problem
17:38.25JerJerare you looking at the whole picture in asterisk?
17:38.38JerJer[Jedi]:  show application record
17:38.55yaaarJerJer: sorry, not sure what you mean?
17:39.07yaaari've got asterisk running with about 7 v's and a c
17:39.23yaaarbut it just looks like a normal call that's working fine
17:39.27JerJerand logger.conf ?
17:39.37JerJerallow=ulaw in general section of sip.conf?
17:39.42JerJerand the phone set to use ulaw
17:40.21*** join/#asterisk |dennis| (~dennis@200.32.197.2)
17:40.30yaaarlogger.conf has both console and messages set to 'notice,warning,error'
17:41.22yaaarallow=ulaw is in sip.conf
17:41.27*** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
17:42.01yaaarhmmm. how can i check whether the phone is using ulaw?
17:42.21jeremywhitingyaaar: make a call and do sip show channels
17:42.25_ioscannerlooks like mysql is having a problem.  never mind
17:42.26yaaarcool
17:42.32*** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com)
17:42.53yaaaryep format ulaw
17:43.09Nuggetmysql is good at causing problems.  :)
17:44.25*** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net)
17:44.41*** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net)
17:46.00*** join/#asterisk bdunn (~bdunn@69.15.23.58)
17:46.20ManxPowerthe -users part of my .procmail rc: http://pastebin.ca/17555
17:46.27jeremywhitinghi all, if I only have 2 lines coming in to a 4 port tdm04b where do I specify to only try those two ports for outgoing calls?
17:46.46ManxPowerjeremywhiting: in your Dial line.
17:46.55ManxPoweralong with the group= in zapata.conf
17:47.09terrapenoh good god...
17:47.10jeremywhitingchannel => 1-2 in zapata.conf?
17:47.12terrapenthis song is hilarious
17:47.35terrapen"Conservative Christian, Right-Wing Republican, Straight White American Male"
17:47.36ManxPoweralong with a group=1 before the channel line.
17:47.52ManxPowerThen you can Dial(Zap/g1/whatever) and it will pick the first port that's not in use by Asterisk
17:48.45ManxPowerApparently nobody cares about my procmailrc 8-)
17:49.05anthmtry this
17:49.06anthmgroup=1,10,100
17:49.06anthmchannel => 1
17:49.06anthmgroup=2,10,100
17:49.06anthmchannel => 2
17:49.06anthmgroup=3,20,100
17:49.07anthmchannel => 3
17:49.09ManxPowerjeremywhiting: stop what you are doing.  Go to voip-info.org.  Spend the next week reading about Asterisk
17:49.09anthmgroup=4,20,100
17:49.11anthmchannel => 4
17:49.27anthmg1-4 is indivdual
17:49.32anthmg10 us 1 and 2
17:49.36anthmg 100 is all 4
17:49.43fileanthm: encore, ENCORE!
17:49.47anthmg20 is tge top 2
17:49.55yaaardoes anybody have a sample of a cisco 79xx sip.conf i can check out? sort of grasping straws here...
17:50.15jeremywhitingManxPower: yeah, I have, just never read about digium cards and such cause we weren't planning to use them
17:50.37jeremywhitingbut Junction Networks doesn't sound very good for some reason when we call, so we're trying this instead
17:53.39jeremywhitingis there a way to have asterisk cli initiate a call?
17:54.13jeremywhitingfor debugging purposes only of course
17:54.33fileapp_changrab allows you to originate a call from the CLI if you wish, it's available at http://www.pbxfreeware.org/
17:54.33anthmapp_changrab available at a pbxfreeware.org near you
17:54.35MikeJ[Laptop]jeremywhiting, yes, app_changrab from pbxfreeware.com
17:54.36MikeJ[Laptop]hehe
17:54.42*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
17:54.43MikeJ[Laptop]echo.
17:54.44filedejavu
17:54.53ManxPowerjeremywhiting: only if you have a soundcard set up on the server, and the developement headers for OSS or ALSA and you built Asterisk with those headers avainable.
17:55.06ManxPowerjeremywhiting: or you could have looked it up on the Wiki: console phone
17:55.23*** join/#asterisk salvini_fs (~felipesal@200165194043.user.veloxzone.com.br)
17:55.38ManxPowerfile: How do you run app_changrab?  Just run "changrab" from the console?
17:56.03MikeJ[Laptop]for originate, you type originate, a channel, and a channel
17:56.12file^^^
17:56.30los415u can always drop a call file in the outgoing folder
17:56.39MikeJ[Laptop]ie originate sip/blah local/1000@default
17:56.41anthmUsage: changrab [-b] <channel> <exten>@<context> [pri]
17:56.42MikeJ[Laptop]err
17:56.44ManxPowerI wasn't aware apps could register console commands.  I thought only res_, etc could.
17:56.47MikeJ[Laptop]y
17:56.52Juggiehrmf, cvs wont compile for me
17:57.05Juggiecould be a cvs update issue, cleaning all source and redownloading
17:57.11MikeJ[Laptop]ManxPower, there is no diff between an app and a res.... or a chan for that matter,
17:57.18filejust different naming convention
17:57.24outtolunci just compiled about 20 minutes ago.. just a few issues but compiled
17:57.24ManxPowerJuggie: I'll bet 1.0.9 will compile for you.
17:57.25anthmUsage: originate <channel> <exten>@<context> [pri] [callerid] [account] [timeout]
17:57.26MikeJ[Laptop]they can do whatever they want, just where you put em and what you call em
17:57.43JuggieManxPower, probally, whats your point?
17:57.46fileyou can have an app_doitall that does a channel, exports functions everywhere, app, whateva
17:58.25anthmso originate Zap/g1/18005551212 1000@mycontext
17:58.29MikeJ[Laptop]you could have a MikeJ_ismyfavorite.c that does the same thing
17:58.36anthmto call someone and make them go to exten 1000
17:58.57cpatryany chance to see it in head?
17:59.00Juggieyah its fine now, looks like cvs update just mucked something up.
17:59.46anthmwell if you look at chan_sip and chan_zap it has cli commands in it too
18:00.07anthmcli commands are a way to expose the scope of your module to the outside
18:00.26ManxPowercpatry: if it's written by anthm it will never be in the standard Asterisk as far as I can tell.  Is that correct, anthm?
18:00.30MikeJ[Laptop]and if it you don't like it.. rename it :D
18:00.40*** join/#asterisk killerwhale (~killerwha@gocastlerock.com)
18:00.50cpatryManxPower: wtf are ya talking about?
18:00.57MikeJ[Laptop]ManxPower, you like rubbing people the wrong way don'
18:00.57anthmhmm so things written by anthm never become std parts of asterisk ?
18:00.59MikeJ[Laptop]t you?
18:01.00cpatrythere's many major contributions already in head by him.
18:01.14MikeJ[Laptop]isn't there a website of those?
18:01.16bdunnI have a two SIP phones - one at the office and one at the home office.  When someone dials my extension, is it possible to have them both ring at the same time?
18:01.17anthm*shrug*
18:01.45ManxPowercpatry: In order for code to be included in Asterisk you MUST either give an unrestricted license to Digium OR you must assign them copyright to your code.  Many people refuse to do that, so their work will never be part of Asterisk.
18:02.05ManxPowerAs I understand it, anthm does not disclaim any of these cool apps.
18:02.18cpatryManxPower: take a look at http://www.cluecon.com/anthm.html
18:02.37cpatrywhat about chanspy?
18:02.51cpatryu really should take a look at that url manx.
18:02.56mogormanbut everything lately has not been disclaimed
18:02.59ManxPowercpatry: that stuff happened a long time ago.
18:03.03mogormanand some taken out
18:03.04ManxPowerI'm still waiting for anthm's response.
18:03.17mogormanbut he has the choice to do that
18:03.29ManxPoweranthm: will you disclaim app_changrab?
18:03.41bdunnCan anyone tell me how to make two extensions ring at the same time - simulring or whatever?
18:03.48filebdunn: &
18:03.55filebdunn: Dial(SIP/jcolp&IAX2/jcolp)
18:03.59Juggiehe doesnt have to disclaim whats not in the tree.
18:03.59filewill ring both at the same time
18:04.08MikeJ[Laptop]ManxPower, chanspy and app_dictate were both quite recent
18:04.11*** join/#asterisk pabelanger (~a@67.71.252.98)
18:04.14bdunnfile, Wow... that easy... thanks.
18:04.16ManxPowerfile: Oh, I repsect his refusal to disclaim some of his work.
18:04.41ManxPowerI'm just saying that if he doesn't disclaim something it will never be in the Asterisk tree.
18:04.51ManxPowerBTW, I thought bkw_ wrote chan_spy.
18:05.00anthmhard to say alot of the stuff I don't disclaim has already been refused
18:05.02Juggiei dont think he cares, thats why theres pbxfreeware.org
18:05.24anthmor i dont have the patience to modify it 1000 times
18:05.29anthmto make them happy
18:05.38ManxPoweranthm: The second issue it prolly the more significant one. 8-)
18:05.42killerwhalehey all - Looks like I'm getting rid of TDMs and moving to Wildcard TE110P.  Where can I information on how to do the following....
18:05.43killerwhale12 to data while passing all traffic through to the Asterisk Open Source PBX, which reliably routes the channels to their designated locations. This eliminates the need for an external router.
18:05.57mogormanasterisk has to have solid code
18:06.01mogormanand structure
18:06.01killerwhaleI want to read-up when I get it.
18:06.11mogormanits a pbx
18:06.14ManxPowermogorman: Have you looked at the Asterisk souce code?
18:06.20mogormanyeah
18:06.26Juggiemogorman, look at app_voicemail
18:06.27mogormana good bit
18:06.35mogormanlol that is the red headed step child
18:06.40mogormanthat many have tried to fix
18:06.48mogormanand yet it grows bigger everyday
18:07.31anthmit's an interesting debate, should asterisk be designed to have loadable addons then insist on owning them all ?
18:07.53anthmdoes apache care if they own them all?
18:08.25anthmi'll gladly work on making the core stay the same long wnough to not break any modules
18:08.44anthmit's kind of a catch-22 no?
18:08.49ManxPowerAsterisk has some significant issues, but it does seem to work amazingly well.
18:08.58*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
18:09.24ManxPowerAnd with mark never being around to steer things in the right direction.....
18:09.53*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
18:10.07ManxPoweras far as I can tell kpflemming is now steering Asterisk developement, but I'm sure that's not really the case.
18:10.24anthmprecicely why we host a weekly call and are having a conference
18:10.38ManxPoweranthm: is mark usually on those calls?
18:10.42anthmyep
18:10.46ManxPowernifty.
18:10.49anthmwhen he's in town
18:10.52JerJerstole the weekly conference call
18:10.57mogormanbut work ever gets done on them?
18:10.57ManxPowerDude, he's never in town.
18:11.12JerJerhe could call in, if properly motivated
18:11.27ManxPowerJerJer: Exactly.
18:11.29MikeJ[Laptop]ManxPower, what's up with your attitude..
18:11.33anthmanyone who wants to present an issue should attend cos him and kevin are there alot
18:11.57MikeJ[Laptop]if you are so anoyed about it, do somthing to change it
18:12.02Kattyanthm: /he/ and kevin /is/
18:12.13MikeJ[Laptop]I am soo sick of the #asterisk bitch and no action game
18:12.18ManxPowerMikeJ[Laptop]: Perhaps low blood sugar, perhaps I'm just in a bad mood.
18:12.31MikeJ[Laptop]it's not just you.
18:12.35anthmKatty, irc is a grammar free zone
18:12.39ManxPowerPerhaps I'm annoyed that 1.nextversion is taking so long.
18:12.40Kattyanthm: ...
18:12.45Kattyanthm: k
18:12.49Hmmhesaysgrammer/spelling
18:12.50ManxPoweror more specifically the feature freeze, at least.
18:12.59JuggieManxPower, i have a bet with drumkilla, 1.2 by july 31st
18:13.01KattyHmmhesays: i'll grammer your spelling in a minute
18:13.02MikeJ[Laptop]ManxPower, well perhaps if you worked towards it actually happening it would sooner
18:13.05Juggiebut its looking tougher and tougher to make it
18:13.19*** join/#asterisk Mw3 (mw3@daisy.chains.ch)
18:13.24ManxPowerOR perhaps it just seemed like a lot of this sillyness didn't happen when it seemed that mark was participating more.
18:13.25Juggietheres a case if redbull on the line :)
18:13.41anthmthat's cute misspell grammar to tease us =p
18:13.43MikeJ[Laptop]ok, so theres your answer.. bitch, don't do
18:14.05ManxPowerMikeJ[Laptop]: What do you suggest that does not require me converting my production systems to CVS-HEAD?
18:14.09MikeJ[Laptop]guidelines for 1.2 have been set, and are in process of happening
18:14.14mogormanbe...
18:14.18ManxPowerMikeJ[Laptop]: URL?
18:14.22mogorman^_^
18:14.26MikeJ[Laptop]URL what?
18:14.34ManxPowerto the set guidelines?
18:14.34MikeJ[Laptop]I suggest you fix bugs.
18:14.44MikeJ[Laptop]there is no URL...
18:14.58ManxPoweremail message?  Anything?
18:15.01ManxPowerI'm curious.
18:15.19ManxPowerMikeJ[Laptop]: You know I'm not a programmer.
18:15.28MikeJ[Laptop]the one from kevin in respose to you
18:15.48MikeJ[Laptop]the fact that there has been extended discussion both onthe dev list some time ago and on the dev calls
18:16.05ManxPowerAh, the goal of a feature freeze on x date and my suggestion that it be announced 30 days before the freeze happens?
18:16.37ManxPowerI'm not sure I would call that "guidelines", but OK.
18:16.52MikeJ[Laptop]no one has ever said there would be a feature freeze
18:17.03MikeJ[Laptop]and there likely won't
18:17.05*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
18:17.09SarahEmmr
18:17.15SarahEmmerr
18:17.17SarahEmmhihi
18:17.28anthmeveryone who wants stuff should attend the meeting where they can speak in person to address the entire developer base
18:17.42anthmits every thu
18:17.54MikeJ[Laptop]there will be a holdoff on invasive features as we fix as many non feature bugs as possible, then a RC branch, then 1.2
18:18.01MikeJ[Laptop]I have told you this over and over again...
18:18.12anthmthe most likely would welcome the imput
18:18.12MikeJ[Laptop]you seem  to need it in writing
18:18.33MikeJ[Laptop]but for god sakes stop bitching about it..
18:18.42MikeJ[Laptop]or do somthing...
18:18.43anthmcos the list and this irc is so overwhelming they probably dont even realize some of the stuff ppl bitch about
18:19.00anthmso the best policy is to get in line and bitch out loud
18:20.38ManxPowerI just see Asterisk going the way of phpGroupware and that would REALLY suck.
18:20.57MikeJ[Laptop]then help
18:21.09ManxPowerAt least Asterisk doesn't have a twice yearly loss of source code 8-)
18:21.16MikeJ[Laptop]bug marshalls are always welcome
18:21.25ManxPowerMikeJ[Laptop]: I simply believe that the problem is NOT a technical one.
18:21.36MikeJ[Laptop]like I said.. do somthing
18:21.54ManxPowerMikeJ[Laptop]: I guess I could write up release guidelines.
18:22.11ManxPowerBut isn't that something the primary sponsor of asterisk should be doing?
18:22.13MikeJ[Laptop]you could do many things.. but youare too busy complaing to do anything at the momnet
18:22.18cypromisthat is not enou8gh, you would have to enforce that spomeone reads them as well
18:22.34ManxPowercypromis: and follow them.  Exactly.  Only Digium can do that.
18:22.38MikeJ[Laptop]propose it, whatever
18:22.40znoGMikeJ is right, I don't whinge simply because I don't think I have much to offer back to the Asterisk project, and hardly the time either.
18:22.56ManxPowerI'll stop complaining then
18:22.58*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
18:23.01MikeJ[Laptop]thank you
18:25.22*** join/#asterisk virterm (~virterm@shiva.kanatek.com)
18:25.50ronnhi guys ... i have sipuras which fail to re-register after 40-50 minutes .. any one had similar problems
18:26.07bkw_sounds like a deadlock
18:26.14bkw_do calls totally stop working?
18:26.33ronndeadlock? the sipuras wouldn't register at all .. they just fail to register
18:26.41ronnget registration failed message
18:26.43bkw_does asterisk continue to function?
18:26.51ronnyes.. asterisk works just fine
18:26.53bkw_if you restart asterisk does it stop?
18:26.57bkw_or re-register?
18:27.24ronni wouldn't restart asterisk as it carriers other traffic at the time
18:27.35ronni haven't tried restarting asterisk
18:27.46bkw_how often are you registering your sipura?
18:27.52bkw_and are you behind nat?
18:27.58ronn3600 .. the deafult
18:28.26ronnyes behind nat
18:29.01*** join/#asterisk mrgoby (~mrgoby@aa.linuxbox.com)
18:29.18*** join/#asterisk guugmember (~guugmembe@200.6.235.232)
18:29.32guugmemberhello guys, any comments about this product http://www.broad-tel.com/index_en.php
18:30.52*** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de)
18:33.36enderanybody know how to boost the mic level on Sipura phones?
18:34.44MikeJ[Laptop]ender,phone or ata?
18:35.11MikeJ[Laptop]on the ata it is on the admin\advanced pagees, somthing weird like international or sonthing.. and it says gain
18:35.29MikeJ[Laptop]I would guess the phone has similar settings?
18:35.33pdugasender: if it's an SPA-841, return the unit if you can  They completely blow.
18:35.41enderMikeJ[Laptop]: phone
18:35.49guugmemberany one has tested PA - 100 series from Broad-tel?
18:35.54MikeJ[Laptop]I only have that guess
18:35.57enderpdugas: it is an 841 and it will be returned eventually
18:36.24enderpdugas: however a user is stuck using it right now and I need to boost their mic level.
18:37.01pdugasender: I've got one customer who bouht a batch of 6 then upgraded to latest firmware.  They're close to useless.
18:37.38pdugasender: you may look for older firmware and downgrade.  I have one on the shop that works okay but has never been flashed.
18:37.59enderpdugas: so is there no way to adjust it?
18:38.42pdugasender: recent version of fw had some gain adjustments.  didn't have much luck myself.  Not in front of one so I can't point to the exact setting.
18:39.27enderpdugas: k
18:39.42pdugasender: luck!
18:40.37*** part/#asterisk brookshire (~pfffft@207.111.174.1)
18:40.41*** join/#asterisk brookshire (~pfffft@207.111.174.1)
18:41.28enderpdugas: thanks, I actually found something.
18:41.44*** part/#asterisk brookshire (~pfffft@207.111.174.1)
18:41.49*** join/#asterisk brookshire (~pfffft@207.111.174.1)
18:42.02brookshirei love it when xchat messes up
18:42.21twisted[work]brookshire, haha
18:42.51enderah sweeet irssi
18:42.54brookshirehah.. EPIC 4 life!
18:42.59brookshirei don't know why i used xchat
18:43.01brookshiretoday
18:43.16twisted[work]i use xchat every day
18:43.35brookshirewierdo
18:43.36brookshire:D
18:43.38*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
18:43.40twisted[work]hah
18:43.45twisted[work]works great
18:43.56brookshireworks great at crashing
18:44.01shido6sunuva...
18:44.03twisted[work]shido6, uhm, good luck with that
18:44.07twisted[work]seeing as i'm not on a laptop
18:44.30*** join/#asterisk meppl (~mephisto@A7d23.a.pppool.de)
18:44.31twisted[work]brookshire, i don't have any of those crashing issues
18:44.46twisted[work]you must be a nub
18:45.05brookshirenah.. i blame xchat
18:45.21twisted[work]i blame your inability to use xchat :P
18:45.31enderis there a way to bump up the audio at asterisk level?
18:45.53shido6bump up the audio, eh?
18:45.58shido6zap card?
18:46.04twisted[work]pump up the volume
18:46.07shido6heh
18:46.08brookshirepump it up
18:46.18*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
18:46.19twisted[work]and dance
18:46.21SpaceBasshey folks
18:46.21twisted[work]dance
18:46.35brookshirehaha.. well there is the bass
18:46.41anthmshido6, did JerJer tell you to talk at cluecon instead of him?
18:46.42twisted[work]lol
18:46.42brookshirewe just need some treble
18:46.53SpaceBasswhen I do dial(SIP/mybroadvoiceaccount, <a number>) I get circuits busy from BV
18:47.03brookshiretit-tit-tit-ta-tit-tit-tit-ta
18:47.04twisted[work]well gee
18:47.08shido6ahhr yoooh ghonna mess with meE on dis? cauze awf maye accent?
18:47.19twisted[work]SpaceBass, your syntax is wrong
18:47.44anthmahem, shido6, did JerJer tell you to talk at cluecon instead of him?
18:48.06SpaceBasstwisted[work] figured that :) what am I missing... feel like its glaring but I dont see it... tried dial(sip/bv|||<number>) too...
18:48.25SpaceBassi dont want to set a timeout or options
18:48.42shido6anthm, yes - aye wiw tahlk in maye ahnold voice
18:48.53MiccSpaceBass, did you set the callerid properly?
18:48.58shido6abowt voip frawd
18:49.02anthmthen why is he pissed off that he is not talking?
18:49.17SpaceBassMicc i figured that was a moot point b/c BV doesnt accept outbound CID
18:49.27anthmhe aparently feels he was the one who wanted to give the speech
18:49.49shido6he can give the speech - its not something thats going to cause any havoc here
18:49.54shido6I had to reserve the spot
18:49.56MiccSpaceBass, right if you don't set the fromuser to your broadvoice number it will give you a 500 error.
18:50.05twisted[work]SpaceBass, then try using the proper syntax
18:50.08twisted[work]it's on the wiki
18:50.09anthmwell now he's pissed you may want to go work it out with him
18:50.09twisted[work]very simple
18:50.16twisted[work]and it's also in extensions.conf.sample
18:50.26SpaceBassthanks...
18:50.30twisted[work]dial(sip/provider/number)
18:50.43*** join/#asterisk criptos (~criptos@201.138.231.189)
18:50.46twisted[work]note the /    --  not a comma, a space, or a pipe
18:50.51endershido6: no, sip phone w/ a shitty mic
18:50.53MiccSpaceBass, broadvoice works sometimes and doesn't other times.
18:50.58shido6erf?
18:51.00criptosHow to disable the callwaitting at the iaxy? I dont want call waitting...
18:51.00shido6oh
18:51.12brookshirecriptos: in the provisioning
18:51.18SpaceBasstwisted[work] thanks, thats what I was looking for... no stranger to dial strings... looking for my typo... thanks for point it out
18:51.33MiccSpaceBass, so even if you get it all setup right, it depends on who is holding the bubble gum and duct tape on broadvoice systems.
18:51.35filebrookshire: MATTTTTTTTTTTTTT!
18:51.40brookshirefile!
18:51.42filehi.
18:51.47twisted[work]oh my god beck
18:51.50twisted[work]look at those inodes
18:51.51SpaceBassMicc I think there is some pixi dust involved too
18:51.54fileomg Becky!
18:51.56twisted[work]*becky
18:52.01twisted[work]they are so big
18:52.06filethey're like coconuts
18:52.19twisted[work]inodes?  like coconuts?
18:52.23fileyessss
18:52.28brookshiredang.. i really need to get gtkiaxyprov out
18:52.32brookshireTODAY I SAY!
18:52.33brookshire:D
18:52.44fileI need to get my accounting stuff done, but I don't think I will
18:52.49filenot today atleast
18:52.56twisted[work]heh
18:53.05MiccI'm still having dtmf issues with cvs head. I think I had better get the stable build so I can rule that out.
18:53.28fileI have a feeling the box will go kaboom-like
18:54.33guugmemberany one has tested PA - 100 series from Broad-tel?
18:55.27SpaceBassyaaar what version did you upgrade from?
18:55.39SpaceBassyaaar I haven't been able to go from 6.4 to 7.4 for some reason
18:55.52twisted[work]brookshire, i'll believe it when I see it
18:56.01brookshiretwisted.. lol
18:56.13brookshirewell... i just need to put it in cvs
18:56.20brookshireand then write the documentation
18:56.26twisted[work]do you have cvs commit?
18:56.52twisted[work]if so, do it!
18:58.00InfraRedwhere should my asterisk -> provider login information be stored?
18:58.05*** join/#asterisk drbrown (~chatzilla@65.121.240.182)
18:58.09*** part/#asterisk brookshire (~pfffft@207.111.174.1)
18:58.16*** join/#asterisk brookshire (~pfffft@207.111.174.1)
18:58.30drbrownis CVS HEAD broken???
18:58.33drumkilla[work]brookshire: it's already in CVS
18:58.41Kattyborken???!!!!11oneoneone
18:58.43brookshiredrumkilla!!!
18:58.45brookshireis it?
18:58.46brookshirewhere?
18:58.52*** join/#asterisk krazykrab (~Krab@203.81.238.65)
18:59.05drbrownwon't complile, it seems to get stuck on version.h
18:59.18krazykrabhelp needed on Asterisk@home .. All circuit BUsy now
18:59.21twisted[work]drbrown, is this a fresh checkout?
18:59.22drumkilla[work]drbrown: make clean
18:59.38drbrownyes, 2 miniutes ago on 2 different systems
18:59.47drbrownstuck in the same spot
19:00.35krazykrabanyone can help ?
19:01.04drbrownhttp://pastebin.ca/17562
19:01.27drbrownThat is the spot where it gets stuck on the compile, just keeps repeating itself
19:02.38Kattydo do do ...can't ping this
19:02.49twisted[work]lol
19:03.04twisted[work]damn, you're right!
19:03.33krazykrabAll Circuit Busy!!
19:03.47twisted[work]krazykrab, chances are that's a telco message
19:04.01*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
19:04.03drbrownshould I file a bug report, or am I the only one having this problem??
19:04.19brookshireno.. just talk with drumkilla
19:04.23twisted[work]lol
19:04.24brookshirecomplain to him, lol
19:04.36drbrownk
19:04.40brookshirei was just joking
19:04.41brookshirei don't know
19:04.47brookshireJOSH KNOWS!
19:04.54twisted[work]i do not
19:04.57brookshiredo too!
19:05.04fileI don't either
19:05.05brookshirehe is the * of *
19:05.10twisted[work]no
19:05.21filethe Josh club does not know.
19:05.30M_atIs the first * because you can't legally put the word there?
19:05.34brookshirewell he knows, he just won't help you
19:05.37Kattydoes the peanut gallery know?
19:05.42bkw_drumkilla you did get what I ment about adding that new cvs module to the cvsup server at digium
19:05.47twisted[work]let me check
19:05.51bkw_it will get picked up by the mirror
19:05.58krazykrabtwisted- this circuit busy thing coming again and again i this i have missed somthing
19:06.03twisted[work]nope... the peanut gallery (aka, my co workers) does not know
19:06.09Kattytwisted[work]: k
19:06.15drumkilla[work]bkw_: no, we couldn't figure it out, haha
19:06.23brookshireyay! gtkiaxyprov
19:06.25bkw_its in /etc/cvsup/sup
19:06.26bkw_I think
19:06.46drbrownwonder why I can't compile
19:06.51bkw_its a directory with per module with some files in it
19:07.10bkw_go to the all one
19:07.31twisted[work]bkw_, it's /usr/local/etc/cvsup
19:07.32bkw_add it to list.cvs
19:07.39bkw_not the binary
19:07.42bkw_the sup files
19:07.55bkw_it might be in /usr/local/etc/cvsup/sup
19:07.58bkw_not sure how he did that
19:08.02twisted[work]yeah
19:08.05twisted[work]that's wher eit is
19:08.07bkw_but the all directory has a list.cvs
19:08.11bkw_add it to the list
19:08.15bkw_and the mirror will pick it up
19:08.23bkw_note i say "the" mirror
19:08.24bkw_the one and only
19:08.44drumkilla[work]bkw_: want a cookie?  :p
19:08.50bkw_look like krazykrab didn't get his help in time
19:09.02Kattyaww
19:09.02bkw_I can't do cookies
19:09.03Kattysniffle
19:09.05bkw_cookies bad
19:09.09Kattypoor wittle krab
19:09.39filemy parents are on barbecuing spree
19:09.43fileer on a
19:09.48Kattygosh
19:09.59M_atWhat are they bbqing?
19:10.03fileburgers
19:10.07filethe day before that it was hot dogs
19:10.11Kattyveggie burgers? :>
19:10.12denonstir fry kitty cat
19:10.24denonhaha, sorry kajtzu
19:10.26denoner Katty
19:10.50drumkilla[work]bkw_: i forwarded the info to malcolm, so we'll see
19:10.50Kattyk, all better
19:11.17*** join/#asterisk MicC_ (~sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com)
19:11.37MicC_is there a newb channel?
19:11.54anthm#snipe
19:11.54Kattyprobably lots
19:12.01twisted[work]d'oh
19:12.24MicC_just had a couple of SIP related questions
19:12.48filethen ask 'em
19:13.04MicC_thats what was hoping for
19:13.16MicC_what is a good (free as in beer) IP softphone.
19:13.21filefor what OS?
19:13.27M_atX-ten's X-Lite works for me
19:13.57MicC_Winblows
19:14.15fileX-Lite
19:14.19MicC_thanks
19:14.23bjohnsonI get "Maximum retries exceeded on call " but nothing shows up in show channels or anything.  I've even restarted asterisk.  What the heck could it be?
19:14.25twisted[work]free as in beer?
19:14.28twisted[work]beer is not free!
19:14.36filetwisted[work]: poisoned beer is
19:14.41Kattyboo, beer
19:14.45Kattyoh
19:14.45MicC_next question: I would like a couple of phone #s what is a good SIP provider that is relatively inexpensive?
19:14.46twisted[work]file, hushua
19:14.47Kattyi was going to go do something
19:14.55Kattystop distracting me, silly irc
19:15.03citatsmmmm beer
19:15.14M_atMicC_ : Are you using Asterisk?
19:15.22MicC_yeap yeap
19:15.23MicC_:P
19:15.24*** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net)
19:15.30MicC_got it and AMP installed today
19:15.32criptosHow Can I know what firmware version a iaxy is using?
19:15.43MicC_AMP anygood btw?
19:15.50*** join/#asterisk Umaro (~umaro@209.140.74.64)
19:15.50MicC_I am a commandline man myself
19:16.01M_atYou behind NAT?
19:16.12MicC_yes...I can do 1 to 1 if I have to
19:16.24*** join/#asterisk meppl (~mephisto@A7d69.a.pppool.de)
19:16.28M_atYou may prefer to look for IAX termination then - works better than SIP where there is NAT
19:16.36Hmmhesaysugh, why do people use telephone dialers anymore
19:16.51MicC_Hmmhesays: proof of concept
19:16.52MicC_:P
19:16.55*** join/#asterisk RedHatUser (~asaa@baydeinter-27-83.baydenet.com.br)
19:16.57MiccMicc_, are you mocking me? ;)
19:17.07Malthuslol
19:17.09MicC_ug
19:17.16Malthuswhat's a telephone dialer
19:17.16MicC_that a coincidence?
19:17.38HmmhesaysI'm thinking just to piss me off
19:17.53MiccI normally go by Mic but that is usually taken so I go by Micc.
19:17.54MicC_M_at: ok...you know what my next question is...what is a good ....
19:18.13MicC_I usually go by MicC but it was taken
19:18.23Miccyeah, hahaha, by me.
19:18.38M_atwww.iaxprovider.net is a good starting point
19:18.41UmaroHey guys, anyone worked with * on a SGI box, or signate's enterprise PBX thing?
19:18.42MiccNow I'm gonna have to make sure I take this name on all the big sites. :D
19:18.46MicC_this is the first time this has happend
19:18.59Umarosupposedly it's open source, but they don't have a source release anywhere, of course
19:19.03MicC_Micc: too late
19:19.04MicC_:P
19:19.08MiccMicc_, and we ended up in the same channel. strange.
19:19.14MicC_weird
19:19.18MicC_you Irish?
19:19.22brookshireumaro??
19:19.30MiccNope, german and polish I think.
19:19.37MicC_lol
19:19.39Miccmmm.. sounds like yummy saussage.
19:19.45MicC_Mic is slang for Irish
19:19.51MicC_Mac is scottish
19:20.00MiccMine is short for Michael.
19:20.01MicC_pikey=pikt...etc.
19:20.06MicC_ah
19:20.30MicC_M_at: Thanks for the help...one last question. What is a good IAX provider?
19:20.32brookshirewe had it working on an altex here
19:20.33MiccIts a nickname. Started on my 12th birthday when I received a 2400 baud modem and signed up on my first bbs and couldn't yet type.
19:20.43MicC_how old are you?
19:20.57*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
19:20.58*** join/#asterisk Nix (~Nix@81.213.125.220)
19:21.03MiccIts been with me for almost 16 years. As I am almost 28 now.
19:21.20Umarobrookshire: oh yeah? cool.
19:21.21drumkilla[work]there was a 'drumzilla' in here once
19:21.26twisted[work]yeah
19:21.28twisted[work]you scared him off
19:21.32drumkilla[work]:)
19:21.34Malthuslol
19:21.55Umarobrookshire: how was the performance? trying to figure out how much of signate's "enterprise pbx" is them actually making performance modifications to asterisk, and how much of it is that they just use SGI boxes
19:22.33drumkilla[work]twisted[work]: do you know if malcolm has to kill cvsup for the changes to take effect?
19:22.43brookshireumaro: i doubt they make zero changes to asterisk
19:22.56drumkilla[work]no, you doubt they make *any* changes
19:23.05brookshireyes
19:23.11brookshirei doubt they make *any* changes
19:23.18brookshirein fact i'm pretty sure of it
19:23.25drumkilla[work]nubs ...
19:23.32brookshire:D
19:24.36MiccMicc_, NuFone is a great IAX provider.
19:24.42bdunnI can't seem to work this one out... I have this in Extensions.conf, and I need it to ALSO ring ext 3004 when 3001 rings.  Please help if you can.  Thanks!
19:24.43bdunnexten => 3001,1,Macro(stdexten,3001,${BDUNN}0)
19:24.43bdunnexten => bdunn,1,Goto(3001|1)
19:24.51MicC_MicC: HA! 30 years old used it on an Atari 800 BBS at 1200 baud
19:24.57MicC_ph33r my l33t3ss
19:25.23drumkilla[work]fight!!!@!12!
19:26.10M_atPut it away
19:26.20file[laptop]noooooo it's so cheap
19:26.59MiccMicc_, you've got me beat there. Where are you from?
19:27.30MicC_Saint-John, NB CANADA
19:27.31MicC_:P
19:27.38brookshireoh no!
19:27.41MicC_not exactly a technopolis
19:27.46brookshiretwo micc's
19:27.54file[laptop]Saint John?
19:28.00file[laptop]MicC_: I'm in Riverview
19:28.07MicC_oh jesus
19:28.13M_atHe wont save you
19:28.19sivanafriggin beauty
19:28.23file[laptop]Dieppe is french
19:28.35MicC_My dad lives in Moncton
19:28.46MicC_do you know Darcy or Logan Cunning?
19:28.48MicC_haha
19:28.50M_atOh course Dieppe is french - it's in france
19:28.53file[laptop]can't say I do
19:29.01MicC_lol...just checking.
19:29.14SarahEmmi love when people find out i'm from toronto canada. 'oh, do you know X? he lives in montreal!'
19:29.30sivanaSarahEmm: ya, but do you know Jim Blumsen?
19:29.45denonSarahEmm: hey, I know someone from quebec, do you know her?!
19:29.47denon:)
19:29.51sivanahe visited toronto once
19:30.18SarahEmmlol
19:31.38MicC_ah
19:31.49MicC_nufone only offers DID in michigan
19:31.52MicC_:(((((
19:32.34bdunnI can't seem to work this one out... I have this in Extensions.conf, and I need it to ALSO ring ext 3004 when 3001 rings.  Please help if you can.  Thanks!
19:32.36bdunnexten => 3001,1,Macro(stdexten,3001,${BDUNN}0)
19:32.37bdunnexten => bdunn,1,Goto(3001|1)
19:32.59bdunnSeems like this should be so easy but I just can't figure it out.
19:33.35*** join/#asterisk SteveL (~Steve@smtp.burlesonisd.net)
19:33.54*** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr)
19:33.56jhiverhi all
19:34.55*** join/#asterisk [Gator] (~qe2315us@66-118-23-3.texlinkcom.com)
19:35.13jhiverwell, the chan looks pretty calm for now
19:35.18*** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net)
19:35.23anthmoh yeah?
19:35.30anthmemacs rocks!
19:35.37denonanthm: linux sucks!
19:35.43jhiverlol
19:35.57anthmsolaris bay-bee
19:36.07denonanthm: ok, fine .. how about "vi kicks emacs' butt!"
19:36.10jhiverasterisk blows! (your mind, of course)
19:36.20denonsolaris? oh come now .. even sun has given up on solaris
19:36.28brookshirehehe..
19:36.29anthmdem fighting words
19:36.32brookshireasterisk blows (you)
19:36.34brookshire:D
19:36.44jhiver:)
19:36.45drumkilla[work]if only ...
19:36.45anthmemacs can pretend to be vi!
19:36.51drumkilla[work]we'd all be so set
19:36.55jhiveranthm: except slower
19:36.57brookshirehaha.. russell would definately be a lucky man
19:37.07Malthusvi sucks
19:37.08file[laptop]Russell Wussell
19:37.11Malthusvim rocks
19:37.13anthmESC-x ...  vi-mode
19:37.18jhiveryeah, vim is good
19:37.26file[laptop]nooo ESC-x ... perl-mode
19:37.31*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
19:37.45jhiverI used to use emacs a hellofalot actually
19:37.55jhiverit was a bad habit from uni :)
19:37.59jhivernow I use only vim
19:38.07jhiverfor a while I was using both :)
19:38.14denonI just use echo >> and never make mistakes
19:38.21anthmbah
19:38.35eKo1sed for editing, echo for appending
19:38.42denonediting? wussat?
19:38.55jhiveryeah, emacs is good for heavy coding
19:39.01anthmif you use any of the features I added to asterisk you are using emacsified code mwa ha ha ha
19:39.02jhiverespecially with the speedbar
19:39.16bjohnsonvacuums suck
19:39.18SpaceBasseKo1 why not just ed
19:39.20SpaceBass:)
19:39.26jhiverbut for config files and remote file editing, vim is just so cool
19:39.41bjohnsonkate is great
19:39.46Malthusremote file editing?
19:39.47brookshirejove!
19:39.50jhiverit looks nice yeah
19:39.51sivanawhich Manager Proxy is the best?
19:39.53brookshireall your editors suck!
19:39.55brookshireMUAHAHAHAHAH
19:39.57bjohnsonsftp + kate
19:39.57anthmsee if ed can edit 4 files at once and let you edit the same file in 2 places at the same time =]
19:39.58eKo1[master@dubraska ~]$ emacs
19:39.58eKo1bash: emacs: command not found
19:40.02jhiverMalthus: like editing a file over ssh
19:40.07bjohnsonall your editors are belong to us!
19:40.09eKo1mwuahhahaha
19:40.21file[laptop]what a fight!
19:40.27anthmmaybe cluecon should have code races!
19:40.28brookshiresomeone sip us up the bomb
19:40.33MikeJ[Laptop]heh
19:40.40anthmmake and app that can ... blah ... GO!
19:40.46jhiverbrookshire: LOL
19:40.53Malthusjhiver: you mean running vim on the local machine and editing the remote file, or running vim on te remote machine?
19:41.05jhiverMalthus: running vim on remote
19:41.12Malthusoh
19:41.13eKo1load app_emacs.so
19:41.20eKo1weeeee
19:41.30MalthusI thought there was a vim feature I didn't know about!
19:41.57anthmor how bout setup asterisk contest
19:42.16*** part/#asterisk RedHatUser (~asaa@baydeinter-27-83.baydenet.com.br)
19:42.19anthmempty boxes with on os all the way to make a call to some sip addr to see who wins
19:42.21eKo1the vi extensions in bash are much more powerful than emacs in my opinion.
19:42.22cpatrycontest? who have the sexiest voice around ?:P
19:42.37twisted[work]me!
19:42.43*** join/#asterisk rjreb (~rjreb@greatwall.amer.net)
19:42.46anthmcan it play video games ?
19:42.48anthmand talk
19:42.52anthmand render images ?
19:43.01anthmsend email
19:43.08anthmconnect to aim and irc?
19:43.10jhivereKo1: why would you want vi extensions on bash when there's perl?
19:43.21anthmyay for perl!
19:43.30twisted[work]anthm, an editor does NOT need to do all that shit
19:43.37twisted[work]but yes, VIM can indeed do most of that.
19:43.42anthmyes it does!
19:44.04anthmit's on page 1460 of the big book of what every editor should do
19:44.08twisted[work]lol
19:44.10Silik0ncpatry I do
19:44.46cpatryno, its when brian laugh! hehehe
19:44.57anthmwhich was drafted with emacs btw!
19:45.12*** join/#asterisk reddawn (~reddawn@69.238.222.176)
19:45.18jhiverSilik0n: maybe you should do some asterisk sound files to prove it :)
19:46.06jhiverI'd love a suave: "Comedian Mail, would you like to listen to your messages, master" kind of voice ;-)
19:46.08eKo1jhiver: uhm, 'set mode perl ' doesn't exist.
19:46.12twisted[work]you don't want that.
19:46.13twisted[work]no
19:46.13twisted[work]no
19:46.18twisted[work]bad touch
19:46.51jhiverwhat don't you want?
19:47.32jhivereKo1: what's 'set mode perl' supposed to do?
19:48.04anthmi'd guess it would make it render perl code all pretty
19:48.04eKo1Nothing, since it doesn't exists.
19:48.28jhiverlol
19:48.34Corydon-wBut you can do :set syn=perl
19:48.40jhiverwhat would you want it to do?
19:48.46jhiver(if it existed)
19:49.11Corydon-wwhich changes the interpretation of the file, in terms of syntax color highlighting, to that of Perl
19:49.14anthmallow you to be able to enter any key on the keyboard mutiple ways?
19:49.39jhiverme I'd like 'set mode perl' to slap vim user if there is no 'use strict' and 'use warnings' statements at the top of the file :=)
19:50.16*** join/#asterisk mrtwister (~user@cable-1-32.cgates.lt)
19:50.23jhiverlike: "set mode perl" <enter> *SMACK* USE STRICT! moron!
19:50.27*** join/#asterisk FaxTerix (~administr@lineaAK59.velocom.com.ar)
19:50.28anthmuse YourOwnJudgement;
19:50.33FaxTerixGood days
19:50.54jhiveryo
19:51.17jhiveranthm: nonono
19:51.25jhiveruse strict, use strict, use strict
19:51.37UmaroI'm going to have to agree with jhiver on this one
19:52.03jhiveranything else is pure and utter rubbish, and use strict; should be perl default behavior for fucks' sake
19:52.35jhiverI can't believe the amount of time i've lost because of no 'use strict' code
19:53.07anthmbah
19:53.16UmaroWhat's the most sane way for me to do silence detection in *?
19:53.35UmaroI've been trying to use BackgroundDetect, but it just takes an insane amount of resources
19:54.45anthmuse Crutch; ?
19:55.22jhiverI didn't know this backgrounddetect app
19:55.28jhiverinteresting!
19:55.38FaxTerixDoes anybody have used TxFax? Because I'm having a problem and I can't figure out how to resolve it.
19:56.04jhiverout of curiosity, what are you doing that requires such detection?
19:57.17Umarojhiver: answering machine detection. And before you get red in the face, this is only for calling existing customers to confirm appointments
19:58.17jhiver?
19:58.30jhiverI'm not that judgmental :)
19:58.41Umarojhiver: sorry, a lot of people are, I've found.. lol
19:59.25jhiver:)
20:00.13eKo1funny, i've never encountered voip spam.
20:00.17Silik0ndamned ricers
20:00.20eKo1but then again, i don't live in north america.
20:00.27jhiverit's because it's not mainstream enough
20:00.37mutilatori don't get voip spam per say
20:00.38jhiverI guess this ought to happen with skype at some point
20:00.47MicC_thanks for your help guys.
20:00.48MicC_I am out
20:00.51mutilatorbut i do have 2 phone numbers i use when signing up for junk
20:00.52jhivercya
20:01.07mutilatorjust never use them unless i need to activate something
20:01.08gambolputtyhow would any voip spam problem be solved?
20:01.09tomtom_anyone has experience with channel grouping in zapata.conf?  when i register to two groups (eg: group = 1,0) it doesn't seem to work ...
20:01.20*** join/#asterisk So3kris (~jan-wille@soekris.xs4all.nl)
20:01.28So3krishello
20:01.43*** join/#asterisk Bicster (~Bicster@Bicster.active.supporter.pdpc)
20:02.10So3krishaha nice
20:02.14So3krisik have a 4521
20:03.33wasim(hic)
20:03.45Bicsterhic
20:03.53wasimhail -drunks!
20:03.58Bicsterthose guys rule
20:04.26Bicsterl33 with a capital T
20:04.40denonhey wasim, your country has Internet again!
20:04.47jhiverUmaro: http://www.voip-info.org/tiki-print.php?page=Asterisk+Wishlist search for app_machinedetect.c
20:04.47DaminHmmm..
20:05.10wasimdenon: yes, we're back on the glass centric thing, and off that lousy e3 on satellite
20:05.26denonwasim: that outtage sounded pretty ugly
20:05.28DaminCan you set "port=" directives for each IAX peer? I.E. have one on port 4569 and another on say 5000?
20:05.54DaminI know you can define it globally...
20:06.01DaminBut can you do it in a context?
20:06.03wasimdenon: it was, and we're due for another one, for a week 18th-23rd
20:06.20denonwasim: that's when they're going to down it for the repairs?
20:06.29denonI heard they had to get it approved by the other countries first
20:06.39Bicsterdenon: that's when the department of homeland security will be taking it down to upgrade the carinvore servers
20:06.45denonhehe
20:06.54Umarojhiver: is that better, od you think?
20:06.55So3krisi'm trying asterisk out for a couple of days but some questions. ik have the getting started for me and im wondering if you get only a email with the voice message of that you can listen your voice mail via the server. iḿ wondryy in some exampels the voice mailmailbox nummer is the same as the callerid and other exampels not
20:06.59wasimdenon: yeah, they've patched it up temporarily, and then giving enough lead time to get alternates up
20:07.28denonwasim: get a boat and lay out a couple'a oc-192s :)
20:07.33denonyou could make a fortune
20:07.35empis *@home a good way to get started?
20:07.43wasimthe worst of it was, it was just bloody 20 km off the coast, and it took them two weeks to get a ship in to fix it
20:08.02wasimdenon: we're getting smw-4 sometime in oct to this single point of failure should go
20:08.08denonhuh, 20km .. could almost send a diver
20:08.46yaaarBicster: which one?
20:08.47iCEBrkrOK, there has to be SOMEONE that has tinkered with EAGI's and the audio stream...
20:08.53jhiverUmaro: no idea
20:09.01jhiverUmaro: just pointing you to it
20:09.06Bicsterhehe yaaar ... mrs. wasim doesn't seem to like the idea of having other mrs. wasims around
20:09.14Umarojhiver
20:09.20Umarojhiver: er, oic
20:09.25jhiverit's me :)
20:09.34FaxTerixDid anybody have problems sending tiff images using TxFax?
20:09.43*** join/#asterisk znoG (~gs@200.115.216.109)
20:09.50*** join/#asterisk DannyF (~dannyf@h47n1fls32o865.telia.com)
20:09.51yaaarBicster: you must not be too familiar with Islam
20:10.02Umarojhiver: yeah, that's about the same as the backgrounddetect app
20:10.10yaaar(that was the joke)
20:11.40tomtom_it seems that i can't use any other group number but 1
20:11.54yaaarBicster: ferocious....but dumb as a post
20:12.23Umarojhiver: but thanks for finding it :)
20:12.32Bicsterdid the bugblatter beast make it into the movie?
20:12.40wasimthere's a movie?
20:12.52M_atI believe it was mentionned
20:13.17Bicsterwasim: it's been discussed at great length in -drinkers
20:13.30Bicstercoppice hated it, nobody else saw it :)
20:13.55M_atIt's great if you understand the true THHGTTG
20:14.15M_atBicster: Reeally it's a 30 second AAC file
20:14.17So3krismailbox=1000 is this a the dailing number ?
20:14.31*** join/#asterisk santiago (~santiago@63.245.86.254)
20:15.30Bicstereven worse
20:16.07BicsterI see people all the time who let their phone ring longer than necessary, just so they can annoy others with their 'cool' ringtone
20:16.28M_atOh I don't - cant stand that
20:16.30twisted[work]my ring tone simply says "AFLAC!"
20:16.36Bicstertwisted, awesome :)
20:17.09BicsterI was really pissed off that my last mobile phone didn't come with any non-musical ringtones.  I had to hunt one down and install it, which took a few hours.
20:17.11M_atGet a decent symbian handset
20:17.16M_atIt can say whatever you want
20:17.39*** join/#asterisk albators (~albaneagr@host-84-9-255-27.bulldogdsl.com)
20:17.52twisted[work]i just made mine
20:18.24twisted[work]heh
20:18.34tomtom_no one any clue about the group thing?
20:18.48So3kris!SGI++
20:19.08BicsterSGI is still around?  heh, stock price $0.59
20:19.09So3krisSGI has a new 1u server
20:19.21Bicsterwhy would anyone buy an SGI product when the company is clearly not long for this world?
20:19.21twisted[work]tomtom_, considering the vagueness of your problem, no
20:19.38M_atbecause they look cool
20:19.46tomtom_twisted[work]: i shouldn't start pasting logfiles in here right away , now should i?
20:19.52Bicsterwell yeah .. the SGI 1600SW on my desk _does_ look cool :D
20:19.56So3kristhe are cool and the are fast en good
20:20.22So3krisAnd the run unix
20:20.23twisted[work]tomtom_, no, but you could describe your problem in greater detail.
20:20.25tomtom_the issue seems to be that if i use any other channel group in zapata.conf than g0 it doesn't work
20:20.30So3krisThe real sgiÅ› than
20:20.36twisted[work]okay, what does it say?
20:20.48tomtom_well, i'm getting an all circuits are busy when i dial out
20:20.56tomtom_unaible to create zap channel
20:21.09twisted[work]you have to have group=x in zapata.conf
20:21.26twisted[work]where x is the group number, and it goes from above the first channel definition until it runs into another group line
20:21.26shidoirix
20:22.07tomtom_yeah, i have that .. when i put group = 0 and dial on g0 it works
20:22.16tomtom_when i put group = 1 and dial on g1 it doesn't work
20:22.19eKo1wasn't sgi doing something with *
20:22.29tomtom_when i put group = 0,1 it works only on channel 1
20:22.49twisted[work]pastebin your zapata.conf
20:22.50tomtom_both on asterisk 1.0.7 and 1.0.9
20:22.57twisted[work]because I know for a fact this works
20:23.01tomtom_although it does work on similar systems
20:23.10tomtom_yeah, it does .. only not on this system
20:23.36tomtom_everyone is busy/congested at this time is what i get
20:23.47*** join/#asterisk Inv_arp (~junya@adsl-3-251-24.mia.bellsouth.net)
20:23.51twisted[work]pastebin your zapata.conf
20:24.08SpaceBassanyone know a fix for a cisco phone that crashed during a firmware upload?
20:24.45tomtom_twisted[work]: or shall i post it private?
20:24.48DaminSpaceBass: Get it replaced under SmartNet...
20:24.55twisted[work]pastebin your zapata.conf
20:24.57DaminSpaceBass: You did get SmartNet on it, correct?
20:24.58SpaceBassDamin was afaride of that
20:24.58twisted[work]pastebin
20:24.59twisted[work]pastebin
20:25.03tomtom_yeah ok
20:25.05tomtom_i get it
20:25.13SpaceBassDamin purchased smartnet... like 4 weeks ago.. vender is taking their time
20:25.27SpaceBassDamin its also still under warrenty... called cisco to open a case
20:25.38DaminSpaceBass: You can try rebooting it and see if it spews out any packets w/ Ethereal. It probably has some dying gasp recovery mechanism..
20:26.23SpaceBassDamin will try that... thanks
20:26.44SpaceBassDamin not entirely sure what happened... was trying to update the logo and it locked up on getting an IP... then when I rebooted it just hangs
20:26.51BicsterSpaceBass: as in fish, or as in audio below 60Hz?
20:26.51tomtom_twisted[work]: http://pastebin.com/pastebin.php?dl=312113
20:27.14SpaceBassBicster close... as in the musical instrurment(producing audio below 60hz)
20:27.25Bicsterok
20:27.32twisted[work]tomtom_, is this bristuffed?
20:27.35Bicsterbut they don't produce much noise in space ;)
20:27.36tomtom_yeah
20:27.41twisted[work]heh
20:27.44SarahEmmooh
20:27.46twisted[work]no guarantees there
20:27.48SarahEmmi thought it was space fish :)
20:28.00tomtom_well i have systems with the exact same software image that do work :/
20:28.13twisted[work]bristuffs changes zaptel iirc.
20:28.28twisted[work]in which case, i'm now lost
20:28.39Bicsterbristuff fixes zaptel :P
20:28.42*** join/#asterisk Caro_away (~caro@201.133.229.15)
20:28.45twisted[work]HAH
20:28.48Caro_awayhi all
20:28.49Bicsterhehe
20:29.00Bicsteractually IIRC bristuff doesn't mess with zaptel
20:29.11twisted[work]chan_zap
20:29.22Bicsterok, it messes with that
20:29.25SpaceBassnote: never ever order anything from pcuniverse.com
20:29.37twisted[work]yeah, and that's where his problems are going to come from
20:30.40Caro_awayi have a little problem with tdm400p  and te110p i have in my first pci the te110p and the second pci my tdm400 2 fxs 2 fxo in my zaptel.conf have fxo_ks=1-2 and fxs_
20:31.05Caro_awayin 2-3 but how i can configure the te110p like e1
20:31.17*** part/#asterisk Bicster (~Bicster@Bicster.active.supporter.pdpc)
20:31.42Caro_awaybecause my tdm400 works fine but the te110p i can´t found info how configure in mode e1
20:32.33FaxTerixGood bye
20:33.55eKo1Caro_away: isn't it a jumper setting
20:34.12albatorsca  be jumper or software
20:35.18twisted[work]te110p has a jumper for t1/e1
20:35.20anthmwho want's res_monitorthatmuxes ?
20:35.23twisted[work]jumper ON means e1
20:35.26twisted[work]jumper OFF means t1
20:35.34shidomodes
20:35.35shidodood
20:35.36shidomodes
20:35.37Caro_awayah
20:35.41shidoscrew the jumper
20:35.43Caro_awayi don´t check that
20:35.52Caro_awaymaybe is my mistake in this case
20:36.09twisted[work]shido, on the 110, you need to set the jumper.
20:36.16*** join/#asterisk KarolM (asterisk@toronto-HSE-ppp4040107.sympatico.ca)
20:36.18brookshireoh.. haha
20:36.25brookshirethank god we have that documentation up
20:36.25shidoyou can modprobe it and change it from the command line
20:36.38shidoor have I been hit in the head too many times
20:36.44twisted[work]shido, on the 410/405 yes, but on the 110, IIRC no
20:36.50eKo1you can change jumper settings from the command line?!
20:36.55shidoyeah eKo1
20:36.58brookshirefor the quad spans
20:37.00Caro_awaybut in zaptel.conf how i can put my conf is correct span=1,0,0,cas,hdb3  bchan=1-15 dchan=16 bchan=17-31 is correct
20:37.01shidoI have a box in the philippines
20:37.03brookshirenot the single one
20:37.10shidoand I didnt want them to shut it down everytime -
20:37.15Caro_awayyep is single e1
20:37.16mogorman<PROTECTED>
20:37.17shidoand found u can change the jumper settings
20:37.19mogormandont be a nubb
20:37.22mogormanyou can change both
20:37.24mogormanfrom source
20:37.27brookshirethen wtf
20:37.30mogormani mean the mod probe
20:37.35brookshirewhy are we making this stupid documentation then
20:37.37mogormant1e1override
20:37.41twisted[work]mogorman, from source?
20:37.43mogormanis the flag
20:37.51mogormanwell you could go edit the source
20:37.57mogormanand change default twisted ^_^
20:38.01brookshireyou guys need to tell jan what's up
20:38.03SarahEmmerr
20:38.03mogormanbut yeah i have typos a plenty
20:38.07twisted[work]that's a bit dumb, considering it's easy to swap the jumper
20:38.10SarahEmmwhy bother with a jumper if oyu can override it in software?
20:38.13shidono
20:38.16shidowhen u have a remote box
20:38.21shidoand if you have someone "change" the jumper
20:38.24shidoand have your box stolen
20:38.26shidothat sux
20:38.34shidoI would much rather do it from here remotely
20:38.38shidogiveing NO one access
20:38.41mogormanand you can change it no matter where the jumper is
20:38.47shidoyes
20:38.51mogormanpeople just like the ability to do it
20:38.53mogormanby hand
20:39.02SarahEmmerr...
20:39.05yaaarhmmm. anybody know why calling my voip (asterlink) 800 number is just ringing without connecting to my * box? i'm looking at the console (with abou 7 v's) and it doesn't say anything at all. 'iax2 show registry' shows my registered to them....
20:39.12SarahEmmhow often do boxes change from T1 to E1 once put on site? :P
20:39.33twisted[work]apparently quite often
20:39.34shidoif you are a carrier and your E1 gets cut off you have to quickly change to analog
20:39.36mogormanexactly
20:39.42anthmask asterlink support, PAGING DARTH!
20:39.45shidoand t1 channel abnks are easier to find
20:40.34twisted[work]do you not then still have to have someone on site?
20:40.40DarthClueyaaar: call me, sending you a pm now...
20:40.48yaaark
20:40.49twisted[work]to change the equipment?
20:40.51shidoyeah but not open the box
20:41.01twisted[work]but they could just as easily steal it
20:41.04*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
20:41.09shidothe box is in an apparatus that allows you to use a keycode
20:41.16shido3 to get to the box
20:41.24shidou just give them 1 to open the cage
20:41.27shidonot the one to get the box out
20:41.30shido:)
20:41.41shido"plug this in and go home"
20:41.52shidopaypal him a few bucks, done
20:44.28lesouvagetzafrir: The work for TBook is almost done. It took more time then expected.
20:45.25*** join/#asterisk brookshire (~matt@207.111.174.1)
20:48.13yaaarneat. thanks DarthClue
20:48.46*** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk)
20:48.57*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
20:49.13sivanais the sip.conf username field a global variable in *?
20:49.29SarahEmmyou're asterlink, DarthClue?
20:50.07DarthClueSarahEmm: i work for the dark side, we are many, many things.
20:50.12anthmsee, we get bitched at for too much advert and every day ppl are suprised by stuff
20:50.28SarahEmmheh
20:50.34MRH2is there any way to set ALERT_INFO on a Channel (when using a call file to auto-dialout)
20:50.38SarahEmmyou need to offer 416/647 DIDs ;)
20:51.00DarthClueSarahEmm: we are working on it, but we have to negotiate with certain rebel forces to make that happen.
20:51.28anthmfor the record cluecon IS asterlink and asterlink is run by anthm and DarthClue, bkw_ and file work with me.
20:52.21MikeJ[Laptop]SBC has more :)
20:52.26SarahEmmheh okay DarthClue
20:52.34*** join/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it)
20:52.35SarahEmmwho are you, sivana?
20:52.37mutilatoryep, i'm not sbc
20:52.47sivanaI am... sivana :P
20:52.50twisted[work]sivana is sivana
20:52.55sivanahehe
20:52.57SarahEmmerr
20:52.59SarahEmmwhat provider i meant ;)
20:53.01sivanavoctel.com
20:53.36sivanabut you need wholesale voip right?
20:53.56SarahEmmme?
20:54.01mutilatormine are only in michigan tho..
20:54.10sivanaya
20:54.14SarahEmma 647/416 DID and an 888 DID... i'm still looking around.
20:54.25SarahEmm888/877/866/whatever
20:54.28SarahEmmvanity, tho
20:54.31sivanaya, we got them too
20:54.39DarthClueand Canada...must be usable in Canada.
20:54.55twisted[work]hah
20:55.06sivanaya
20:55.46SarahEmmyeah.
20:55.54SarahEmm'k. i'll keep that in mind
20:55.57SarahEmmstill looking around a bit.
20:56.57Silik0nwho cares about canada?
20:57.09MikeJ[Laptop]canadians
20:57.36pabelangerTerrance and Philip do...
20:57.39DarthClueMustDie: you coming to cluecon?
20:57.50MustDiei never been to chicago
20:57.52sivanaI am Canadian!
20:57.54MustDieso i will have a reason to go
20:57.56MustDie:P
20:57.58Silik0nMikeJ[Laptop] and canadians matter since when?
20:58.14MustDieSilik0n: since south park i guess
20:58.15MustDie:P
20:58.16Silik0ni mean they are famous for what Theo DeRaddt and a hockey strike?
20:58.17sivanain a bottle?!.... brilliant!
20:58.19albatorsanyone knows why i can't get caller id on my tdm card to my analog phone ?
20:58.50pabelangerSilik0n: don't forget beer.
20:58.54albatorsif i have call on hold, then i can see the caller id
20:59.02Silik0noh yeah beer++
20:59.17pabelangeralbators: you line provisioned for the service?
20:59.25albatorsyes
20:59.36twisted[work]molson ice != beer
20:59.42albatorsi see the caller id in the debug
20:59.48albatorsbut on phone nothing
21:00.02Silik0nhey hose hound get outta here eh
21:00.07albatorsand if i get the second caller , then on phone it will show the caller id
21:00.14twisted[work]Silik0n, WTF?
21:00.27Silik0ntwisted[work] never saw Strange Brew?
21:00.30twisted[work]no
21:00.35MustDietwistah 1
21:00.36MustDie!!!
21:00.54Silik0n2 guys who's whole life is sitting around getting drunk and cracking on canadians
21:00.59twisted[work]oh
21:01.00albators:D
21:01.01Silik0nthats the plot of the entire movie
21:01.05*** part/#asterisk ilTizio (~tcs@adsl203-149-051.mclink.it)
21:01.12twisted[work]sounds like terrance and phillip
21:01.14twisted[work]only older
21:01.16Silik0nhah
21:01.33albatorsand other question, how to answer to the second call ;) since i never managed ;)
21:01.49drbrownhas anyone had any problems with today's cvs HEAD compiling?
21:01.55*** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu)
21:02.04albatorsi don't use HEAD since 3 days ago
21:02.07DarthClueyaaar: you coming to cluecon?
21:02.10albatorssince it stoped working!
21:02.34drbrownalbators: does yours not compile either?
21:02.56albatorsit compiled but asterisk eat all my resurses :P
21:03.06*** join/#asterisk opus_____ (~opus@dahphish.org)
21:03.07terrapenoh    my   god
21:03.09opus_____hi
21:03.13terrapenlook what i found on fark
21:03.14terrapenhttp://www.tufts.edu/vet/cfa/hoarding/FORMgallery.htm
21:03.18albatorscpu 80% and asterusk not working :P
21:03.42albatorseat cpu and ram, but not working :P
21:04.10opus_____me hungry for cpu and ram soup
21:04.15albators;)
21:04.22albatorsso about the caller id
21:04.25albatorsany idea?
21:06.12*** join/#asterisk AST07070NYC (~spam@pool-68-239-181-169.nwrk.east.verizon.net)
21:08.10AST07070NYChello, does anyone know of an example on the web which uses call files which pass data.... ie call NXXNXXXXXX and play file /var/foo.gsm ???
21:10.18*** join/#asterisk emp (~emp@70.57.239.37)
21:10.26Drukenanyone here with a working AMP install that is willing to help me out?
21:11.18empI just installed *@home and configured X-Lite, when it logs in, it gives "Login failed! Contact Network Admin.  Your number is: 200"  any hints?  I'm just following the walkthrough at this point
21:11.45ChkDigitWhat is a complete list of the special extension names (s for the starting one, etc...)?
21:12.13eKo1check the wiki
21:12.25ChkDigitI had hoped this would be quicker.
21:13.06Hmmhesaysquintum cli is such a pain
21:13.41eKo1ChkDigit: no, the wiki is quicker and safer
21:13.48*** join/#asterisk data2 (~data@213.221.191.134)
21:13.50eKo1quintum?! NO!!
21:14.23eKo1I'm never using anything from them again.
21:14.30Hmmhesaysoh yeah? why is that?
21:14.59twisted[work]quintums are fucking ridiculous to set up
21:15.17eKo1yeah.
21:15.24Hmmhesaysi'm not saying the setup is hard... it's really quite simple, but their cli is a bitch to write anything for
21:15.35eKo1that, and the sip stack on the unit I bought sucks major donkey balls.
21:15.41*** join/#asterisk santiago (~santiago@63.245.86.254)
21:15.41DarthClueChkDigit: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf under predefined extension names.
21:15.58Hmmhesaysit really depends on 1st or 2nd generation quintum and what firmware you have
21:16.09eKo1I have the 1st gen. stuff
21:16.40Hmmhesaysyeah that's why sip was an after thought for 1st gen,  the second gen is a whole different animal
21:16.52eKo1i bet the 2nd gen. stuff is way better. but after that experience, i'm not doing anything with quintum anymore.
21:17.03twisted[work]i'm running a 2nd gen device
21:17.07twisted[work]the cli fucking blows
21:17.12twisted[work]and the setup is still a bitch
21:17.15*** part/#asterisk mkrufky (~mk@68.160.103.77)
21:17.21Hmmhesaysnaw, pretty simple once you get used to it
21:17.25twisted[work]i would not reccomend it to my worst enemy
21:17.35Hmmhesaysit's no worse than cisco
21:17.50twisted[work]HAH
21:17.55MiccSo is dtmf detection a problem for asterisk or do I just have something screwed up?
21:17.56twisted[work]cisco's cli at least gives you command help
21:18.10twisted[work]quintums says "refer to the user guide"
21:18.19MiccI've had the same intermittent problems with SIP and ZAP.
21:18.20twisted[work]and the users guide is so far from correct it's not funny
21:18.32Hmmhesaysthat is an inaccurate statement
21:18.37twisted[work]no it's not
21:18.46MikeJ[Laptop]one of my back burner projects was to re-work command completion and partial commands and context sensitive help in cli to be more cisco-like
21:18.56Hmmhesayssorry, the cli reference is accurate
21:19.02twisted[work]no it's not
21:19.12Hmmhesaysif you are reading the one for your firmware revision it is
21:19.15twisted[work]most of the commands in the user guide that came with this thing do not exist
21:19.42twisted[work]as well as the ones on the website
21:19.57Damintwisted: Whatcha working on?
21:19.58Hmmhesaysheh, name one that doesn't match on the cli reference
21:20.16twisted[work]Hmmhesays, kinda difficult seeing as it's been a month since i've touched the damned thing
21:21.04HmmhesaysI hear the same stuff about quintum all the time, they've got learning curve... so what
21:21.13twisted[work]that's not a learning curve
21:21.19Hmmhesaysget the right cli reference and you'd be fine
21:21.25twisted[work]that's like saying the special olympians can compete in the olympics
21:21.45Hmmhesayssounds like you are complaining just to complain
21:22.05twisted[work]no, i'm stating my opinion on something that is a pain in the ass and has no accurate reference
21:22.16Hmmhesaysyour statement is not accurate though
21:22.21twisted[work]my statement is highly accurate.
21:22.26twisted[work]if you were here I could prove it
21:22.38Hmmhesaysand you can't prove it now?
21:22.56twisted[work]uh, no, because the box is not set up any longer
21:23.01twisted[work]it's sitting here on my desk
21:23.41Hmmhesaysso put an ip in it, update the firmware, grab the right cli reference and tell me what in the reference is not accurate. because I am quite curious
21:23.55twisted[work]Hmmhesays, i'm not going to the effort just to appease you
21:24.05empwhat log file can i look in for * authentication messages?
21:24.19twisted[work]if you don't want to listen to my opinion then ignore it.
21:24.46Hmmhesaysheh, honestly it seems more like you just don't like the fact that your statement could be inaccurate
21:24.53enderdoes the Polycom 301 actually have a mic built in for speaker phone?
21:25.02data2if anyone can do a reasonable rate for a Sri Lanka Mobile? could they please msg me, ta.
21:25.22twisted[work]Hmmhesays, no, i don't like the fact that you can't just accept what i have dealt with as the truth
21:25.49twisted[work]end of conversation
21:25.59twisted[work]bring it up again and you will be quelched
21:26.04Hmmhesaysheh
21:26.38Hmmhesaysso kick me
21:26.46eKo1fight fight fight
21:27.11Hmmhesayscome on
21:27.17*** join/#asterisk tris (tristan@camel.ethereal.net)
21:27.26eKo1all this antagonism over quintum...
21:27.35Hmmhesaysheh, this isn't about quintum at all
21:27.56Hmmhesaysthat much is obvious
21:32.56twisted[work]oh, but it is.
21:33.05Hmmhesaysheh, obviously you have deeper issues
21:33.17*** mode/#asterisk [+q *!*@66.173.103.108] by twisted[work]
21:33.30SwK[work]CRAZED BUTT CLOWNS!
21:33.37Assidwhats +q again?
21:33.50twisted[work]if you had spent the amount of time I have downloading manuals, trying different shit, and wasting time trying to figure out that piece of shit, you would understand where i'm coming from
21:33.57twisted[work]+q is quelch
21:34.04twisted[work]or quiet
21:34.07Assidaah
21:34.08Assidk
21:34.08twisted[work]however you want to look at it
21:34.15Assidi thought it was for the channel
21:34.22Assiddidnt know you could do it per user
21:35.30twisted[work]telling someone they have 'deeper issues' when they are simply explaining the difficulty they've had with a product and their dislike for it is bad enough.  taunting an op is just bad.
21:36.03data2hmm, just checked my gateway box... asterisk 0.9.1 was just out last time i was on this channel :(
21:36.05twisted[work]Hmmhesays, i have nothing against you.  we have had many another conversation/discussion that has been fine.  when you try to tell me that i'm lying i get pissed off.
21:36.07SwK[work]but taunting ops and ircopers is fun sometimes
21:36.31empdo I need an outbound proxy IP for x-lite to do a echo test?
21:36.35twisted[work]SwK, heh, yeah, but basically calling them liars is one thing.
21:36.43Assidxlite is weird
21:36.44SwK[work]twisted[work]: LIAR!
21:36.45twisted[work]*one another
21:36.51SwK[work]heh
21:36.55Assidsomehow everyone i call says they hear me funn
21:36.57Assidfunny
21:37.06twisted[work]if you weren't sitting right behind me I might not have known you were joking
21:37.08Assidactually i dont know if its xlite.. or carrier
21:37.27SwK[work]Assid: heh xLite is ok but its not the best phone
21:37.29Assidyour lying!! he aint behind you.. hes on the left of the chat screen!
21:37.31*** mode/#asterisk [-q *!*@66.173.103.108] by twisted[work]
21:37.36Assiderr. right..
21:37.38twisted[work]Assid, lol
21:37.40SwK[work]what do you expect for a free version of a commercial product
21:37.43twisted[work]he's physically right behind me
21:38.00Hmmhesaysheh, no I didn't say you were lying, I said your statement was inaccurate as of right now with the current firmware and cli reference
21:38.02Hmmhesaysperiod
21:38.12bkw_you tell em!
21:38.16Assidhahahaa
21:38.17SwK[work]Assid: actually twisted is correct we sit with about 15 feet between our backs
21:38.18twisted[work]hah
21:38.31Assidhrmm
21:38.38*** join/#asterisk gst (~gst@85.124.173.170)
21:38.45DaminWTF piece of equipment are you guys talking about anyway..
21:38.48AssidSwK[work]: what do you suggest i use?
21:38.55SwK[work]Assid: a hard phone
21:39.00Assidbah
21:39.02Assidnot availableh here
21:39.03twisted[work]heh... at least
21:39.05Assidnot for cheap anyways
21:39.17Assideverything costs a min. of over $100
21:39.17SwK[work]no seriously tho... check the settings of your soundcard and mic etc...
21:39.22twisted[work]actually, xlite is fine, as long as your audio device is good and the machine can handle it
21:39.26SwK[work]yeah
21:39.33twisted[work]I use it occasionally on my mac
21:39.46Assid3Ghz.. enough to voice chat man!!!
21:39.46SwK[work]crappy sound card, mic gain turned up too much etc can cause xlite headaches
21:39.57Assidhrmm
21:40.00twisted[work]Assid, haha.. yeah, that should be okay to handle the application
21:40.24gstis it somehow possible to reply in an AGI (or py-asterisk program) with a "user not found" (after an extension matches and the script is called)?
21:40.28SwK[work]call 800-555-1212 it'll talk to you
21:40.31MikeJ[Laptop]-q *!*@
21:40.38Assid800?
21:40.41Assidcool..
21:40.48MikeJ[Laptop]-q *!*@* ..is it quiet in here?
21:40.49SwK[work]yeah 800-555-1212 its powered by tell me
21:40.50Assidisnt that bad use of office equipment?
21:41.04Assidno clue whats tell me
21:41.09twisted[work]MikeJ[Laptop], no
21:41.15Assidim guessing some knd of directory service?
21:41.25SwK[work]Assid: voice activated directory assistance
21:41.28twisted[work]tellme
21:41.37twisted[work]i'd use tell me
21:41.40twisted[work]18005558355
21:41.46SwK[work]Assid: ask for crazy stuff like "buttplugs" or "stained blue dress"
21:41.46twisted[work]you can actually have "conversation" with it
21:41.49fileHello, and thanks for calling Tell me!
21:41.54fileMain menu.
21:42.00Assidyeah.. but i wanna know if someone else can hear me fine
21:42.04bkw_here are some of your choices
21:42.10bkw_stock quotes
21:42.12filegod, we're such telephony geeks
21:42.12bkw_weather
21:42.14bkw_news
21:42.25Assidneed human behind the network
21:42.27*** join/#asterisk dysan (~ack@202.37.224.27)
21:42.49SwK[work]well seeing its voice activated and those systems sorta suck, if you get get a good response you'll know you are transmitting audio ok
21:42.57SwK[work]OR... call yourself on your cellphone
21:43.03Assideeks
21:43.06Assidim not calling india!!!
21:43.13*** join/#asterisk fugitivo (~ajf@158-190-126-200.fibertel.com.ar)
21:43.15Assidwaaaay too expensive
21:43.22SwK[work]so you gotta US DID?
21:43.22fugitivohello
21:43.26Assidyep..
21:43.33SwK[work]or find dell's 800 number and call them
21:43.36Assidbut .. will be funny to hear myself
21:43.40SwK[work]they have people there all the time
21:43.58SwK[work]or if you just want to test xlite there is a echo app for the dialplan
21:44.03twisted[work]Assid, simply call echo() then
21:44.09twisted[work]or whatever the echo app is
21:44.10SwK[work]loops whatever you say backto yourself with a pretty good delay
21:44.11twisted[work]i can't remember
21:44.19SwK[work]yeah what twisted said
21:44.27MustDieswk: i think Dell is a indian company, go and call them then ;)
21:44.37SwK[work]MustDie: hah
21:44.38SwK[work]I have
21:44.41fugitivoanyone is using the mp108 fxs?
21:44.43twisted[work]dell is not an indian company, although they outsourced my job a long time ago
21:44.47twisted[work]to... india.
21:44.49SwK[work]fugitivo: yes
21:44.53SwK[work]fugitivo: it works
21:45.00SwK[work]lemme guess DTMF problems
21:45.11fugitivoSwK[work]: it works fine? or some problems?
21:45.24SwK[work]it works fine
21:46.09SwK[work]comes with a not so standard setting for DTMF on RFC2833 (it comes default at 96 and should be 101) otherwise its ok
21:46.20SwK[work]comes in a 4 port and a FXS version too
21:46.29SwK[work]err FXO version
21:46.38fugitivogreat, thanks
21:46.42empis the * handbook valid with the current release?
21:47.03Assidhrmm
21:47.04fugitivoSwK[work]: how much did you pay for it?
21:47.06Assidecho does nothing
21:47.08Assidthats weird
21:47.16Assidexten => 399,1,Echo
21:47.29SwK[work]fugitivo: dont remember got it from voipsupply I think
21:47.43SwK[work]i dont order the stuff I just install it
21:47.49fugitivook, thanks
21:48.05dysanwhich config files do i need to edit to make outgoing calls using x-lite? i have 3 zap compatible cards installed and working
21:48.46DarthCluedysan: sip.conf and extensions.conf, check the wiki, if you need links, ask.
21:49.08twisted[work]http://www.voip-info.org/wiki-Asterisk
21:49.47DarthCluehttp://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf and http://www.voip-info.org/tiki-index.php?page=Asterisk+config+sip.conf
21:51.08MiccSorry, maybe that wasn't that funny.
21:51.27brimstonei laughed
21:51.45*** join/#asterisk Twister (Twister@216.30.232.108)
21:52.03MiccI mean, it wasn't that funny for twisted.
21:52.26*** join/#asterisk talkwebhosts (~freenoder@c-24-127-182-46.hsd1.ca.comcast.net)
21:55.22*** join/#asterisk pdugas (~pdugas@h79.95.40.69.ip.alltel.net)
21:55.25Assidokay 4.2/4.4 and 3.3/3.4  K/sec for GSM / ILBC
21:55.32Assidquality similar
21:58.22elduis there a free video+voice sip client ?
21:59.51So3krishttp://www.voip-info.org/tiki-index.php?page=VOIP+Phones#id216825
22:00.23eldumy bad
22:00.31eldui miss this link lol
22:01.20So3kris?
22:01.56MRH2ne1 know if I can set alert_info variable on a channel, or as an overall default?
22:02.15elduno matter i know this wiki but i havent figured out this link
22:03.35MRH2I can set in the dialplan fine
22:04.54yaaarok guys. catch you tomorrow. thanks for the help DarthClue and happy birthday.
22:05.46So3krisIḿ bizzy white the voicemail but something is nog fully clear. is  mailbox nummer the nummber that you call toe listen the voicemail or a identique nummber that haves no connection white the nummbers that your dailing
22:06.02*** join/#asterisk niZon (~ilt@S0106deadbeefbeef.wp.shawcable.net)
22:08.57*** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
22:09.57*** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155)
22:11.00MRH2what I was hoping to do was dial out from channel to destination and setting the alert_info asterisk dials out to the 'originate channel' so that the phone placing the call will autopickup
22:13.08*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
22:13.08*** mode/#asterisk [+o bkw_] by ChanServ
22:14.06terrapeni need a decent audio-in device for my powerbook
22:14.11terrapenthe iMic just sucks ass
22:15.06opus_____hey
22:15.23opus_____in realtime, for sipusers => , how come I MUST have ipaddr= to the address that the phone is on?
22:15.48opus_____I have 'host=dynamic' in my sql table, but, it wants ipaddr=192.168.0.247 and won't work without it
22:17.59mariogamboai have a problem with my tdm22b i configure in zaptel.conf fxo_ks=1-2 and fxs_ks=3-4 but i don't recive tone for my analog phone in the port 2 what is missing?
22:19.04angler_mariogamboa, have u configured zapata.conf and also loaded the wctdm driver?
22:19.13mariogamboammm no
22:19.50mariogamboahow i can do thant?
22:20.32mariogamboain my zapata.conf i have context=from-pstn  signalling=fxs_ks  rxwink=300
22:21.03mariogamboaand another options like usercallerid etc. i need to add somenthing more?
22:21.47*** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
22:21.48angler_zapata.conf    signalling=fxo_ks channel => 1-2  signalling=fxs_ks  channel => 3-4
22:22.06TripleFFF2sdfdoes one know how to chage the which app which is refering to ?
22:22.21*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
22:22.36*** join/#asterisk file[laptop] (~file[lapt@mctn1-142166195139.nb.aliant.net)
22:22.42TripleFFF2sdfEG: box that moh works.. # which mpg123
22:22.42TripleFFF2sdfEG2 box that doesnt
22:22.42TripleFFF2sdf<PROTECTED>
22:22.56angler_mariogamboa, after that load wctdm...... modprobe wctdm
22:23.32twisted[work]it's good to see more digium folk in here providing quick tech help
22:23.37angler_:)
22:23.42TripleFFF2sdfheh
22:25.02*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
22:25.21TripleFFF2sdfsame paths
22:25.22drbrownhas anyone had any luck running asterisk as a non root user?
22:25.27TripleFFF2sdfjust one is a symlink
22:25.33TripleFFF2sdfdarn mpg
22:25.47TripleFFF2sdfno diff in asterisk conf.. no dif in mpg123 size not version
22:25.49TripleFFF2sdfnor perms
22:25.53mariogamboawhat version of zaptel have you angler
22:26.05TripleFFF2sdfbut mpg just exits on box
22:26.18mariogamboabecause the wctdm module is loaded
22:26.38angler_latest head cvs
22:26.43mariogamboaand no recive tone in the analog phone withh the modification in the zapata.conf
22:26.44angler_run "ztcfg -vv" now
22:27.05mariogamboayep
22:27.33mariogamboai have channel 1-2 fxo  3-4 fxs  5-31 cas/user
22:28.23angler_asterisk is started correct?
22:28.28mariogamboayep
22:28.34*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
22:28.46mariogamboai can recive the asterisk*CLI>
22:28.51shmaltzI get this message exactly every minute:
22:28.52shmaltzJul 12 18:36:36 NOTICE[28197]: pbx.c:1689 pbx_extension_helper: Cannot find extension context 'INVALID'
22:28.54shmaltzwhy?
22:29.05*** join/#asterisk bprice20 (~bprice20@cpe-24-194-236-170.nycap.res.rr.com)
22:29.10RoyKshmaltz: rtfm?
22:29.24shmaltzRoyK, ok, now what? I read it all
22:29.44TripleFFF2sdfshmaltz Add exten i,1,ahngup()
22:29.56TripleFFF2sdfi for invalid
22:30.01shmaltzTripleFFF2sdf, I know the solution, but why am I getting it?
22:30.02TripleFFF2sdft for timeout
22:30.12TripleFFF2sdfcoz something sending to inavalid exten
22:30.18TripleFFF2sdfa goto or somethign else
22:30.22shmaltzwho?
22:30.28shmaltzI got no active channels
22:30.44RoyKshmaltz: pastebin your dialplan
22:30.45TripleFFF2sdfcall you friend pastebin.. ask him to receive your dialplan ,, and tell him to come tell us the url to his party
22:30.55TripleFFF2sdfRoyK beat me to it
22:31.35TripleFFF2sdfso any reasin mpg123 now spawning on one box but ok on another ?\
22:31.43TripleFFF2sdfsame confs same exec
22:32.13MikeJ[Laptop]TripleFFF2sdf, if it is yesterday\today cvs head, update.. there were issues
22:32.29shmaltzRoyK, It's too long
22:33.36TripleFFF2sdfnah its a old one
22:34.38TripleFFF2sdfwell one month
22:34.40TripleFFF2sdfmax
22:34.43TripleFFF2sdfused to work
22:34.46Juggieyah eugh
22:34.49Juggie* crashes for me now
22:34.50TripleFFF2sdfjust doesnt spanw mpg
22:35.10mariogamboalogin root
22:35.21mariogamboapassword
22:35.45Juggiefor me * crashes when i put a call on hold
22:36.11shmaltzOk, thaks guys
22:36.15shmaltzI know why
22:36.38shmaltzsome stupid provider is trying to register or use our server, sip debug told me that
22:37.25sivanaheh
22:39.19MiccWhat version of asterisk is nufone using? Aren't they using realtime?
22:40.20TripleFFF2sdfhehe
22:40.27TripleFFF2sdfanyone got a mpg idea ?
22:40.32TripleFFF2sdfunless i change the player
22:40.35TripleFFF2sdfbut not good
22:43.41mariogamboacool i really go to florida to take the asterisk tranning and dcap
22:45.23twisted[work]while you're there, don't forget to hide from the hurricanes
22:45.25*** join/#asterisk heka (~heka@82.114.68.124)
22:47.33*** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
23:00.47mariogamboaok
23:01.33litageif you/your company needed to hire an intermediate+ linux/asterisk employee, what sorts of questions would you ask?
23:02.28DougRoyerTo do what?
23:02.48DougRoyerProgram? Configure? Compile? Add-On's?
23:03.22litageprogram, config, add-ons
23:06.10*** join/#asterisk cinix (~ax@24-52-166-190.lndnnh.adelphia.net)
23:06.11twisted[work]litage, don't be a competetor ;)
23:06.28DougRoyerI think that asterisk development experience is rare. So you would want an experienced C/C++ developer on Linux with knowledge of SIP/ VOIP.
23:06.37cinixI am attempting to dial a local SIP client when I receieve an incoming call from my voip provider, and this is what I get:
23:06.40cinixNow forwarding IAX2/voicepulse-in-01@66.234.228.170:4569/3 to 'Local/matt@outgoing' (thanks to SIP/matt-9713)
23:06.53cinixGot SIP response 482 "Loop Detected" back from 24.52.166.190
23:06.54twisted[work]oh wait
23:06.55twisted[work]you're in .au
23:06.57twisted[work]n/m
23:07.12cinixand then it makes some new local channel and tries to call that and everything falls apart :-/
23:07.37twisted[work]cinix, turn off forwarding on your phone
23:08.25twisted[work]that means that whatever device is on SIP/matt sent a 302 Moved Temporarily or another forwarding code along with a URI
23:08.46twisted[work]asterisk will accept that URI and attempt to send it to that location
23:08.58cinixoh hmm.. thought I had disabled forwarding. lemme check
23:09.04twisted[work]which means SIP/matt is forwarded to uri: matt@<your * ip>
23:09.44cinixI had notransfer=1 set in sip.conf, I suppose that's not valid.
23:09.48cinixgot it from iax.conf
23:12.06*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
23:14.09cinixokay you actually meant on the client, found the forwarding optinos but they aren't set
23:14.44*** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net)
23:17.06jeremywhitinghi all
23:17.29jeremywhitingI'm having serious problems getting callerid on receiving calls to work with tdm04b
23:19.15jeremywhitingfsk_serie made mylen < 0 (-13)
23:19.34jeremywhitingCallerID feed failed: Success
23:20.08jeremywhitingI've seen a few posts like this in pastebin after google search, and on asterisk forums, but no solution
23:21.57*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com)
23:22.15jeremywhitinganyone ever experienced this?
23:23.49MikeJ[Laptop]jeremywhiting, there is a bug on a crash similar to this in mantis..
23:25.48jeremywhitingmantis?
23:26.15DarthCluejeremywhiting: bug tracker.
23:26.20jeremywhitingit doesn't crash though
23:26.25*** join/#asterisk dack (~dack@S0106000f664f0871.vc.shawcable.net)
23:26.39DarthCluemay still be related.  you running HEAD or STABLE?
23:26.43jeremywhitingit just doesn't get callerid on any zap channel is all
23:26.45jeremywhitingstable
23:26.54jeremywhitingwell, latest gentoo I should say
23:27.02DarthClue1.0.9 or something else?
23:27.08jeremywhiting1.08
23:27.11jeremywhiting.8
23:27.13dackHow come I can't seem to get the T extension to trigger on AbsoluteTimeout?  It always just hangs up right away.
23:27.29Qwelldack: T or t?
23:27.34dackQwell: T
23:27.38dackabsolutetimeout.
23:27.47DarthCluejeremywhiting: you should upgrade to .9, .8 has a known caller id issue.
23:27.55jeremywhitingok, thanks
23:27.59Qwellthere was a bug in HEAD a few days/weeks ago, that had to do with exten t
23:28.10Qwellit affected me, and upgrading fixed it
23:28.11dacki'm on 1.0.8
23:28.18Qwellthen you should be ok, heh
23:28.54DarthClue1.0.9 is the recommended stable
23:28.55dackI do "exten => T,1,Playback(beep)"
23:29.09dackbut i get no beep on absolutetimeout
23:29.25Qwellperhaps its quiet?  Have you tried a NoOp
23:29.27Qwell?
23:29.32DarthCluedack: what is your absolutetimeout set to?
23:29.39jeremywhitinggot it
23:29.51jeremywhitinggentoo dev in charge of asterisk is just a bit behind then I guess
23:29.58Qwelljeremywhiting: They always are.
23:30.03Qwell(no offense to the gentoo devs)
23:30.21jeremywhitingdo I need libpri and asterisk-sounds, etc in line with it. e.g. 1.0.9 for it to work too?
23:30.33Qwelljeremywhiting: its recommended...
23:30.39DarthCluejeremywhiting: yes.
23:30.41dackDarthClue: 10 seconds
23:30.44Qwelloverpaid?
23:30.45jeremywhitingok, thanks
23:30.51DarthClueer..underpaid.
23:30.55dackgotta go, work on this later
23:31.33*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
23:32.12bdunnIf I want to record my own voice prompt - what should I search for on voip-info?
23:32.29Qwellrecord
23:33.21bdunnTHanks.... I think that's the one this I didn't search on.  :-(  Duh
23:34.36*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
23:34.37DarthCluehttp://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Record
23:35.46*** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
23:35.49*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:35.51*** join/#asterisk brookshire (~matt@207.111.174.1)
23:36.06*** part/#asterisk Morex (~blah@host81-157-165-204.range81-157.btcentralplus.com)
23:36.10Ariel_evening everyone
23:36.27SarahEmmhihi
23:36.39brookshirehttp://www.digium.com/downloads/gtkiaxyprov/
23:36.39SpaceBasswell after my cisco 7912 died during an update they are sending me another, no questions asked... thought that was nice
23:36.43brookshireyay! gtkiaxyprov
23:37.03SpaceBasshey Ariel_
23:37.51Ariel_SpaceBass, nice. I had a Polycom IP-500 speaker go out and they switch the phone with only me tell them the s/n. This was swapped out by Polycom directly.
23:38.31SpaceBassyeah, I had to tell cisco the s/n and no probs... pcuniverse.com on the other hand as still not registred my smartnets after 4 weks
23:38.35SpaceBassI'm about to call Visa
23:40.49*** part/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
23:46.11*** join/#asterisk rv_weasel (~no@adsl-66-142-40-127.dsl.kscymo.swbell.net)
23:47.09rv_weaselcan someone recommend a good IAX or sip phone for linux that is not harder to set than setting up all of asterisk
23:47.39Ariel_rv_weasel, there is a linux version of xlite out there.
23:48.20rv_weaselsweet,  ill look that up.  thanks
23:49.45SpaceBassrv_weasel:  for iax there is iaxcomm and gnonephone or something
23:50.24*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
23:51.24rv_weaseliaxcomm is RPMified for my distro!!!
23:51.52NuggetLinux is poo!!!
23:52.28SpaceBassyes, long live windows for workgroups
23:52.52Hmmhesaysclassic
23:53.18Ariel_Nugget, ????? pooo?? oh you muct then be either an OSX or FBSD person.
23:53.49Ariel_Commodore 128  (still have one)
23:53.49rv_weaselwindows 3.1, without TCP/IP support!!!  that is my fav
23:54.02SpaceBassI have my c64
23:54.21NuggetI'm just in favor of using the right tool for the job.  And there just aren't that many jobs where Linux fits that equation.
23:54.31Nuggetzaptel is the only one I can think of off-hand
23:54.42jeffgusNugget, iscsi server?
23:54.49rv_weaseli have calcs that are more powerfull than a C128
23:54.50jeffguswith lvm backend?
23:54.56jeffgusand GFS?
23:54.58Ariel_SpaceBass, I have an extra 1541 I need to sell Wonder how much I can get for it on Ebay.
23:55.07SpaceBassand my collection of original OS discs includes the first netware, os/2 warp...
23:55.17rv_weaselNugget: how bout running the internat
23:55.27Nuggetwhat the hell is an internat?
23:55.27Ariel_Frist Netware I have one it's for the apple
23:55.33SpaceBassAriel_:  I've never looked up my IIe or IIgs or c64... they are just worth to much to me :)
23:55.44SpaceBassand by worht too much, I mean im lazy
23:56.07rv_weaselNugget: only newbs freak about typos in chat.  guess what that makes you
23:56.28rv_weaselwhich would explain you adversion to linux
23:56.28NuggetYou're the guy who thinks that the internet runs on linux, but I'm the newb.  I see.
23:56.46SpaceBassAriel_:  when I got my CNE 3.x there was still a netware client for apple... have that disc next to my netscape (mozilla) 0.8a disc (3.5 HD)
23:56.54Nuggetlinux could disappear today and "the internet" would be largely unaffected.
23:57.03rv_weaselNugget: perhaps you should look at some data that is not published be MS
23:57.18jeffguswell the internet itself would be fine
23:57.22Nuggetperhaps you should visit your ISP and see what they're running.
23:57.24jeffgusbut a lot of web servers would disappear
23:57.35rv_weaselexcept for the fact that 60% or more of the websites would disappear
23:57.56jeffgus60% is apache, not linux
23:58.00SpaceBassi will admit that i just tried linux on the desktop again, thinking I could salvage my intel workstation, and in a windows domain enviroment its just not ready... but its come a long way
23:58.06jeffgusbut linux does run a good portion of apache servers
23:58.09rv_weaselISP's cisco and the like.
23:58.17SpaceBasson the otherhand, I'd NEVER use a windows server for routing, nat, firewall...
23:58.32kd5uzzbx_dundi.c:1399: warning: implicit declaration of function `compress'
23:58.32kd5uzzpbx_dundi.c:1400: `Z_OK' undeclared (first use in this function)
23:58.34Nuggetneither would I.
23:58.36SpaceBassso, I guess Im going to buy a powermac to replace my desktop
23:58.49NuggetI use routers for routing, as does most of the internet.
23:58.53kd5uzzany ideas?
23:58.54rv_weaseli am 100% linux now.  except for my 166mhz laptop.  that is win98
23:59.03SpaceBassNugget:  what kind of routers?
23:59.27Nuggetnot linux boxes. :)
23:59.39*** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-209-168-226.dsl.sfldmi.ameritech.net)
23:59.56jeffgusFC "routers" run linux

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