IRC log for #asterisk on 20070708

05:22.41*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
05:22.41*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support.
05:26.20*** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com)
05:28.07CrashSysI have a global var 1001=SIP/1001. I have ${ARG1}=1001. How can I load the Global variable from ${ARG}? Something like $[${ARG1}] ?
05:28.41CrashSysOr would it be ${[${ARG1}]} ?
05:31.14CrashSys${${ARG1}} ?
05:33.53CrashSysHmm... looks like the last one will work... :)
05:40.07BSD_Techexten = *40,n,GotoIF(${AVAILSTATUS}=""|?i,1|set) ?
05:42.38CrashSys${${ARG1}} worked for what I need... I call the dial macro with 1001 and it loads the SIP/1001 to the dial cmd...
05:43.01CrashSysor was that not for me?
05:51.18*** join/#asterisk Jameswf (n=jfinstro@ip70-162-108-73.ph.ph.cox.net)
05:51.27Jameswfallo all
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06:01.27Jameswfping
06:04.14BSD_Techgot it
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06:04.49Jameswfthats random
06:12.37BSD_Technight
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06:37.06*** join/#asterisk JuliusSpencer (n=julius@60-234-130-224.bitstream.orcon.net.nz)
06:38.00JuliusSpencerhi
06:39.33Jameswfho
06:40.16JuliusSpencerI have a problem at the moment with my tdm400p card
06:40.35JuliusSpencerjust wondering if anyone here might be able to help
06:40.53Jameswfpost the problem we shall see
06:41.18JuliusSpencerthe module wctdm seems to be loading, but when running ztcfg I get an error:
06:41.33JuliusSpencerZT_CHANCONFIG failed on channel 1: No such device or address (6)
06:41.50Jameswfdo you have your channels defined in zaptel and how
06:42.00JuliusSpencerI was here earlier today, and I tried changing pci slots and checking the power was plugged in to the card etc
06:42.15JuliusSpenceryeah; only with the channels in the config do I get the errors
06:42.17JuliusSpencer:
06:42.22Jameswftype: cat /proc/zaptel/*
06:42.38JuliusSpencerfxsks=1,2 fxoks=3,4
06:43.09JuliusSpencerthere's no files in /proc/zaptel
06:43.17JuliusSpencerthere are I mean :)
06:43.20Jameswfthen no zap device is loaded
06:43.52Jameswfpost the lspci output for the card
06:44.08JuliusSpencer00:05.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
06:44.43Jameswfrun update-pciids and see if it changes
06:45.24JuliusSpencer00:05.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interfac
06:45.26JuliusSpencerlooks the same
06:45.37JuliusSpencernever run that command before :)
06:49.41JuliusSpencerhi Jameswf , you still with me?
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06:54.04Jameswfsorry playin on google as I work with rhino not digium
06:54.12Jameswfhttp://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5#53FXOFXSComboCardsDevices
06:54.18Jameswfread 5.3.2
06:55.04JuliusSpencerthanks I'll check. I've spent probably about 2.5 hours using google to solve this,
06:55.22Jameswfrhino cards just go :);)
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07:03.19tzafrirJuliusSpencer, run lspci -v
07:04.08tzafrirVarious TDM400P cards have various sub-vendor IDs
07:04.11JuliusSpencerk
07:04.20tzafrirWhat version of zaptel do you have?
07:04.26JuliusSpencerSubsystem: Unknown device b199:0001 ...?
07:04.43JuliusSpencerzaptel-1.2.18
07:06.26tzafrirJuliusSpencer, modinfo wctdm | grep ^alias
07:07.03tzafrirthis will show you what vendor IDs / product IDs / SubVendor-ids the driver looks for (in this case)
07:07.38Jameswftzafrir smells a nock off?
07:07.56tzafriryeah
07:08.47JuliusSpencerso what should I postin here?
07:08.53JuliusSpenceror should I post somewhere else
07:09.07Jameswfgotta love china... Had a customer tell me they got 4 bad ciccos in a batch of 100 from china
07:09.24Jameswf*cisco
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07:10.48JuliusSpencerok I posted it here: http://pastebin.ca/608351
07:12.04JuliusSpencerI'm pretty sure it's the real deal :)
07:13.22Jameswfdid you run kudzu?
07:13.26JuliusSpencerme?
07:13.27JuliusSpencerno
07:13.31JuliusSpencerI don't think so
07:13.42JuliusSpencerI could try :)
07:13.46JameswfNO
07:13.50Jameswfkudzu bad
07:14.07Jameswfwhere did you buy the card?
07:17.12brian4 bad ciccos?
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07:17.31JameswfI am having a bad S day
07:17.39brianciscos?
07:18.18Jameswfyeah
07:18.37JuliusSpencerdigium
07:20.01Jameswfif you do "lsmod | grep zaptel" what do you get
07:20.09JuliusSpencerit's there.. i'l lpost it :)
07:20.40JuliusSpencerhttp://pastebin.ca/608357
07:25.39Jameswffor giggles run modprobe wct4xxp
07:25.51Jameswfthen cat /proc/zaptel/*
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07:29.25JuliusSpencerstill nothing....
07:29.32JuliusSpencerin /proc/zaptel
07:30.04Jameswfhow are the modules loaded on to the card
07:31.17JuliusSpenceryou mean plugged into the card, two green two red.
07:31.30JuliusSpencerI tried swapping the config as a test
07:31.32JuliusSpencerbi change
07:31.52Jameswfok again im not a digium guy o's and 's please
07:32.26JuliusSpenceroh two fxs s and two fxo s
07:33.11Jameswfok put the fxo's before the fxs's be sure the card is connected to power
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07:34.34JuliusSpencerI'll give it a go... still seems like a sw problem :)
07:35.01Corydon76-homeJuliusSpencer: did you plug in the Molex connector to the back of the card?
07:35.13JuliusSpencerMolex... is that like the 4 pin power cable?
07:35.19JuliusSpencerdaisy chained from the hard drives
07:35.26Corydon76-homeOkay, good
07:35.28Jameswfmolex = brand name,
07:35.57Corydon76-homeIf you're looking at the card, gold pins toward you, L bracket on the left, is it green-green-red-red?
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07:36.01Jameswftry another lead as well.  If the card is not properly powered with s' it will flip out
07:36.11JuliusSpencerheheh
07:36.36Corydon76-homeCheck your dmesg, too
07:37.05Corydon76-homedmesg will tell you if any of the daughter cards are bad
07:37.20JameswfThis is why i use Rhino no drama :)
07:37.39Corydon76-homeThere's no drama if you know what you're doing
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07:38.34JameswfOk you got me after the first 30 i do it while i sleep
07:39.24Jameswfon that note what OS are you using JuliusSpencer
07:48.45JuliusSpencerI checked the power with a volt meter and it gave 5V
07:49.09*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
07:49.10JuliusSpencerOS is Fedora
07:55.21JuliusSpencerok I swapped the modules arounf
07:55.24JuliusSpenceraround
07:56.48JuliusSpencerI'm not opening up the computer again
07:56.50JuliusSpencerheh
08:08.27JuliusSpencerhmm I still get the same problem after swapping the modules around.
08:08.58Jameswfmy guess is bad mobo or hardware issue
08:14.54JuliusSpencerhmmm
08:20.43JuliusSpenceranyone else got any ideas?
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08:53.56*** join/#asterisk SktyNick (n=nick@202.43.236.188)
08:56.18SktyNickEvening guys. Im suffering mental blanks much tonight. I have an extension that sets a handful of variables, then needs to hangup the call and execute a macro which will call the original caller back (Based on the CID num). Is there a way to do post hangup processing in 1.2? I know it can be done with g in the Dial app but this isnt what I need.
08:57.45SktyNickEffectivly currently its executing the macro which is attempting to bridge the two calls, which is not at all what I want to do - I want to drop the first call then execute the macro, obviously I cant place the macro in the extension after the hangup though.
08:57.58SktyNickTrying t avoid using Call files :-)
08:58.03Strom_Mh extension
08:59.38SktyNickPerfect - Thanks.
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09:20.52SktyNickStrom - is h executed during the hangup process or after? Ie. I have a hangup, followed by my h extension which starts with a wait(5) then runs a macro, the thing is the call remains active and back to suare one - instead of trying to create a new call, it simply trys to bridge the two.
09:21.07Strom_Mwell, duh
09:21.16Strom_Mif you want a new call, use a call file
09:21.51SktyNickIs there a way to do it without a call file? I mean I can if I really had to, however it would be nice to simply do it within the Dialplan directly?
09:22.19Strom_Myou'd have to go through some serious dialplan gymnastics
09:22.29Strom_Mwhat's the problem with call files/
09:22.31Strom_M?
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09:23.13brianSktyNick, You can use the Asterisk Manager Interface to place a call
09:23.33Bladerunner05Using grandstram gxp-2000 and asterisk if I press the speaker key the ohter part can't hear me.
09:23.37SktyNickBecause the number it has to call is going to change depending on the options the user gives us during the first call (Think of it as a callback with 80% more crap :-)
09:24.05Strom_MSktyNick: so you dynamically generate teh call file upon hangup
09:24.09Strom_Msimple enough
09:24.17brianSktyNick, You should probably create a AGI script to do that for you and then use AMI to originate a call to the original caller.
09:24.40brianSktyNick, That's how I handle callbacks and it seems to work fine
09:25.16SktyNickYeah AGI is sounding like a neater way to do this. Think I will just get our coder onto it in the morning. Thanks for the help Strom + Brian
09:25.39brianSktyNick, You need to use AMI to originate the call
09:26.35SktyNickYeah thats NP. We are already using AMI to tie Asterisk into our Nagios stuff.
09:26.53brianSktyNick, Sounds like cool stuff. What exactly are you all doing?
09:27.23brianSktyNick, You're a telemarketer aren't you? AREN'T YOU?!?!
09:28.21brianSktyNick, Was it you that called me just I was sitting down to eat dinner?
09:28.23SktyNickArgh no, I do a bit of consultation for Call Centres though :-) I actually work for a Web Host so integration of our systems is a huge part of everyday stuff for us.
09:28.38brianSktyNick, *cough* can you get me a job *cough*
09:28.44brianSktyNick, Sorry I've got a cold
09:29.14SktyNick*cough* Not unless you want to move to Aust. and work for free *cough*. Yeah cold seems to be going around :-)
09:29.24brian:(
09:29.28brianwork for free?
09:29.30brianawwww
09:30.08SktyNickCold is a subset of being a whip cracking tightass srry :)
09:30.33brianI see.
09:30.37tzafrirIt's not for free: the experince gained there is priceless. Is that a better phrasing?
09:30.50brianHow will I live in Australia with no money?
09:30.55SktyNickWow! You are either in marketing or HR heh heh.
09:31.44SktyNickPlenty of empty boxes in dark laneways Brian, if you pick a decent one there may even be a unsecured Access Point in range :-)
09:34.15kiscokidwhere do you get power for your laptop?
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09:38.54SktyNickYou can charge it while your in the office each day, failing that I hear Wireless Power is just around the corner :|
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09:40.32MJ[LappyHey :) I'm having a fiddle with SMS stuff and Asterisk - would ideally like it so I can send an SMS from my mobile phone (in the UK) to my sipgate number, and have Asterisk handle it from there. But - do I still need to use an SMSC etc for that?
09:41.04MJ[LappyIf I've got it right, the SMSC stores the SMS until the receiving end requests it?
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09:56.04tzafrirI wonder if zaptel actually need to look for the kernel source under /usr/src/linux
09:56.27tzafrirconsidering that there is totally no guarantee that this is the source of the current kernel
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09:57.32tzafrirand that if one has installed a kernel from source, the symlink /lib/modules/REVISION/build should be in place
09:59.46SktyNickIn a call file - Can you use priority labels?
10:01.10tzafrirSktyNick, hmm... I guess not. Those don't really exist in the parsed dialplan
10:02.18SktyNickHrmm cool. Will get around it
10:02.54tzafrirSktyNick, look at 'show dialplan <contextname>'
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11:12.41romano2khi everyone! anacron sends me this message frequently: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)"
11:12.54romano2ki apt-getted asterisk on my debian box, what should i do?
11:42.57matskromano2k, ~book
11:43.09matsk~book
11:43.14jboti heard book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
11:43.44JTromano2k: is the asterisk daemon actually running?
11:44.59matsknoop
11:45.11matsknot with that message
11:45.16JTmatsk: not so
11:45.21JTit may or may not be running
11:45.58matskok so you recommend a "ps aux | grep asterisk" to reveal that ;-)
11:46.13JTyes
11:46.47matskI now remember that an old asterisk version didn't create the ctl file
11:46.57matskso a touch solved it
11:51.39tzafrirromano2k, is asterisk actually running?
11:52.06tzafrir/etc/init.d/asterisk start
11:52.54tzafriralso: if you ran asterisk directly as root it may have created some files as root preventing asterisk from running as user asterisk
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12:00.10romano2kJT: sorry for the latence, nop, it is not
12:00.29romano2ktzafrir: not running
12:01.09tzafrir/etc/init.d/asterisk debug
12:01.24frawdhi, any idea of why a zaptel card would send too many interrupts? I understand a missed interrupt with zttest showing "8192 samples in 8193 sample intervals 99.987793%", but I have lines approximately once per minute showing results like: "8192 samples in 7345 sample intervals 89.660645%", along with a low volume bip while listening conversation...
12:01.48romano2ktzafrir: 3 lines, do i paste here or in a website ?
12:01.53JTromano2k: it'd help to have it running before attemting to reconnect or run a remote command on it :)
12:02.01tzafrirwell, here
12:02.04romano2kDebugging Asterisk PBX:
12:02.04romano2kUnable to set high priority
12:02.04romano2kUnable to setgid to 111!
12:02.21romano2kJT: i'm not willing to do anything with asterisk at the moment
12:02.29romano2kit's just installed and configured for further use
12:02.35tzafrirYou need to run that as root
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12:03.00tzafrirto check why asterisk has failed to start
12:03.26romano2koh i'm sorry
12:03.47romano2kit's quite long this time :)
12:06.40frawdI understand that it looks like a common zaptel IRQ problem, but I've done every suggested thing in forums (IRQ sharing, disable acpi, disable IRQ balancing, IDE things, lowering IO, check on other motherboard with same AMD64X2 processor, .....) since last month and I couldn't resolve this issue, so anyone has an idea?
12:09.27frawdfor info: card=openvox A400 with 2 FXO, kernel=debian 2.6.18 with IMQ and layer7 patch, 3 network cards with low traffic, 2 SATA drives in RAID1
12:10.53JTromano2k: logic would dictate to remove the anacron job then or at least disable it :)
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12:28.58CpuIDfrawd, nice l7 :)
12:29.04CpuIDyoull have to go userspace soon though :)
12:33.24tzafrirfrawd, is there actually is a problem?
12:33.41tzafrirgrep wctdm /proc/interrupts
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12:44.21coppicetzafrir: I'm just putting the debian build files into the spandsp package. 0.0.4pre3 has a file called libspandsp3.install. Is that right?
12:45.09tzafrircoppice, the files are generally maintained in the pkg-voip repository
12:45.12tzafrirlet's see...
12:45.34coppiceI was asked to add them to the package, so I'm doing it
12:47.03frawdtzafrir, the problem is a disturbing bip during conversations once in a while
12:47.10tzafrirThe files are under svn://svn.debian.org/pkg-voip/spandsp/trunk
12:47.31frawd# grep wctdm /proc/interrupts
12:47.31frawd<PROTECTED>
12:47.31coppicealso, there is stuff messing about with mmx.h, but that file was removed some time ago because of iff licencing
12:47.44tzafrirdebian/libspandsp3.install is indeed there
12:47.59coppiceyes, but shouldn't it be 4 and not 3?
12:48.25tzafrirthat number is bumped internally in debian
12:48.43frawdCpuID, going to l7 userspace is planned
12:48.57CpuIDcoo :)
12:49.06CpuIDyea i changed my kernel over ready for userspace last week
12:49.10CpuIDhavent got my rules active yet though
12:49.13CpuIDon the TODO :)
12:49.47tzafrirkdelibs of kde3 are kdelibs4. Go figure
12:50.36tzafrircoppice, mmx.h was re-added later after being cleared?
12:50.59tzafriror reimplemented or whatever?
12:51.14coppicei got rid of it completely, but the debian stuff seems to be playing with it still
12:51.48frawdCpuID, looks promising but i keep on kernel version until userspace is proven stable...
12:52.08frawdanyway, any idea on that wctdm bip/interrupts problem?
12:52.46CpuIDya :)
12:53.09CpuIDfrawd, cat /proc/interrupts, is wctdm sharing any irqs with otehr hw?
12:53.16frawdnop
12:53.19CpuIDive found wctdm to be nasty with irqs
12:53.32frawdindeed
12:53.34CpuIDoccasional artifacts due to hdd usage, even when not sharing
12:53.43frawd# grep wctdm /proc/interrupts
12:53.43frawd<PROTECTED>
12:53.46CpuIDim hoping to change out my digium hw sometime soon
12:53.54CpuIDpossibly try sangoma
12:54.03tzafrirthere's a patch called "nommx" to remove mmx.h, but it is not applied
12:54.27frawdi'm not sure sangoma would be much better on that issue
12:54.38frawdit still does need interrupts
12:55.13frawdcould it have relation with delays of ext3 on software RAID 1?
12:55.53CpuIDhmm you know...im runnign sw raid on that box actually
12:55.59CpuIDi have a feeling im using reiser though
12:56.24CpuIDya reiser for everything on that box
12:56.41frawdi tried with reiser as well...
12:57.17CpuIDweird
12:57.24frawdi also tried with hundreds of options in kernel (disable patches, try all preemption parameters, change HZ values, ....) without success
12:57.28CpuIDcouldnt say myself, i havent done a lotta testing as yet
12:57.46frawdsame problem in 2 different (but similar) motherboards
12:58.32frawdmaybe an APIC problem with those specific processors?
12:58.57frawdi really have no idea, even of how to debug or identify the source of the problem...
13:01.05frawdi also tried the zaptel watchdog option, but zttool reports no interrupt loss... it seems that these interrupts could be delayed a bit, generating that small tone-bip when too many delayed interrupts are received at once...
13:01.28frawdor maybe i look in the wrong direction
13:01.50frawdbut what more could i do to check/resolve that?
13:16.20bcnxHi all. I noticed that my music on hold (with the m flag in the Dial command) only works if I proceed the extension with a Answer() command. Am I right assuming that external callers are charged right away, without someone having picked up the call?
13:27.36russellbbcnx: yeah, the CDR would reflect the call was answered at the time of Answer()
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13:56.50bcnxrussellb: I have two Dial commands one after the other, the second only has hold music: I guess I should put the Answer command after the first Dial()?
13:57.17bcnxif not people calling my compay will start paying before someone even answered their call ...
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14:47.30coppicethe * mailing list seems seriously screwed up since the change. does anyone know why?
14:52.47BSD_Techyou did not talk dirty to  it
14:52.58BSD_Techyou did  not send it flowers
14:53.33coppiceI sent it nice warm loving e-mails, and they came back 4 days later
14:54.56BSD_Techhmmm maybe thats  why I have not seen any activity on  the asetrisk-user and asterisk-bsd lists
14:55.14tzafrircoppice, there is a thread about this. It seems to be a that some people have delays whereas others don't
14:56.26coppicesome would call this a downgrade
14:56.26coppicesome would call it a screwup
14:56.26coppiceamericans would call it a negative upgrade
14:56.29*** join/#asterisk guillote_GNU (n=guillote@190.7.27.17)
14:57.05Kwakwaor a complete success :)
14:57.33coppiceit meets all out goals - especially the "own" ones
14:59.38BSD_Techdigium strikes again
14:59.42BSD_Techlol
15:01.04*** join/#asterisk msetim (n=marcos@200-103-130-66.ctame706.dsl.brasiltelecom.net.br)
15:01.07BSD_TechKram goes to EU and the staff start playing with things go figure
15:03.12[TK]D-Fendercoppice, "Mission Accomplished" <-------------
15:14.59BSD_TechTK I got it fixed at 4 am
15:15.03BSD_Techit works
15:15.16BSD_Techbut I want to take it another step
15:15.39*** join/#asterisk wunderkin (n=wunderki@65.39.91.91)
15:21.00*** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net)
15:21.46marlhi, i am trying to setup a call file, at present i have channel set to ZAP/3/numbertodial and context set to internal and extension set to 205, this calls the zap channel first and then the ext, but i want it the other way around, eg. call the ext first and then wen answered make the connection to the Zap number, anyone able to sujest wat i have to do to make it work?
15:22.24marlhave tried simply swapping the channel and ext, but that fails
15:25.25shido6heh
15:27.34shido6whats the context name and extension u want to dial
15:27.36[TK]D-Fendermarl, change your channel
15:28.03[TK]D-Fendermarl, And then obviously where it will lead.
15:28.28marldo u meen just put the ext number in channel? 205?
15:29.50[TK]D-Fendermarl, Ask yourself what KIND of channel will let you go through the dialplan to meet your needs
15:31.17Jameswfdoes the channel matter so much as the context it sits in?
15:32.58[TK]D-FenderJameswf, Channels do not reside in contexts.  Your concept of *'s heirarchy is very confused
15:34.14shido6well...
15:34.24shido6you could use Channel: Local/extension@context
15:34.39Jameswfno but channels have nothing to do with the way a call is handled it is only the medium
15:35.01[TK]D-FenderShido : sure, just SPIT out the answer and have them forego any conscious effort....
15:35.06shido6LOL!
15:35.40[TK]D-FenderJameswf, And you'll want to seperate your newfound attachment between "call" and "channel"
15:35.41Jameswfif(!knowledge){rtfm};
15:35.59Jameswfwtf are you talking about
15:36.46[TK]D-FenderJameswf, This is the part where anybody with any intent on trying to find the answer will have gone HERE : http://www.voip-info.org/wiki-Asterisk+channels
15:37.16[TK]D-FenderJameswf, Where THIS should have stood out : "Local: Loopback into another context"
15:37.16JameswfI have all the answers... they are on my Blackberry
15:37.18Jameswf:)
15:37.39[TK]D-FenderSMRT
15:39.07*** join/#asterisk yonahw (n=yonahw@IGLD-84-228-84-252.inter.net.il)
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15:54.05shido6Local:/205@ext-local this is wrong
15:54.40shido6Local/205@ext-local
16:08.34*** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com)
16:10.14*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
16:22.31rbdcan I use AGI's 'get full variable' to get a SIP header variable? e.g. something like: get full variable SIP_HEADER(X-my-custom-var)
16:25.47*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)
16:26.24BSD_Techcool I just fixed the bsd asterisk port 1.4.6 + zaptel build and works
16:29.27*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
16:31.22*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
16:34.39BSD_Techwhen is libpri going to be updated to 1.4.1
16:35.11Qwellwhen it needs to be
16:39.36bcnxwhos' got a propeller hat for me
16:40.36*** join/#asterisk oej (n=olle@apollo.webway.se)
16:41.54Jameswfoh snap
16:45.04bcnxall, if a sip registration expires for a phone, will it still ring when dialed?
16:46.06*** join/#asterisk zotz (n=zotz@24.244.163.157)
16:51.27*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
17:01.25shido6the quick answer is dont bet on it. If Asterisk doesnt see the phone as registered or doesnt know the IP of the phone it wont ring
17:02.04*** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it)
17:03.56*** join/#asterisk gardo (n=gardo@121.97.176.180)
17:05.08bcnxshido6: I experience odd stuff with sip phones: they only seem to be able to work when set to registration
17:05.19bcnxI'd like to use them without registration however
17:06.03shido6you CAN do that
17:06.04bcnxwhen I switch from host=dynamic to host=X.X.X.X and I reconfigure the phones not to register, they work, but after some time they don't anymore
17:06.08shido6if you know their IP addies
17:06.13shido6when they change... :)
17:06.25shido6if after "some time" they dont work
17:06.33shido6then force the phone to register more frequently
17:07.15bcnxueah, I had them set to 120 seconds, but I think I saw them not ringing when they are re-registering
17:07.26*** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net)
17:07.30shido6where are these phones in relation to  your ast box?
17:07.33shido6nat'd ?
17:07.33bcnxI could be dreaming though, have been working 10 hours striaght now
17:07.41bcnxno, LAN
17:07.59bcnxso no real need for registration
17:08.14bcnxexcept for the fact that they don't work without
17:08.23shido6what kind of phones?
17:08.26*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
17:08.27*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
17:08.35bcnxlinksys spa942
17:09.09*** join/#asterisk brut- (n=brut@66.173.4.254)
17:09.15*** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com)
17:10.54k31thbcnx: wats the advantage to not using reg ?
17:11.20bcnxI'm not sure that the phones are available during the reg process
17:11.37bcnxif you can confirm they are, I'll switch back to a frequent registration config
17:15.18bcnxshould I set them to registration with low expiration time? 120 secs?
17:15.35*** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca)
17:16.05Slingkyhi guys! does somebody know how to pass "*67" to sip provider
17:16.59Slingkyi always get "all circuits are busy now"
17:19.10*** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net)
17:26.03escribzzSlingky check your dial plan in the device your using
17:26.20escribzzits probably set to interprate the *code
17:26.37*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
17:28.17Slingkyright now, i'm trying to solve this from x-lite softphone before trying to solve in ata
17:28.17Slingkyis there something in x-lite that doesn't allow * codes ?
17:31.10shido6does your provider accept those digits as valid?
17:32.33Slingkyyeah, it's acanac and accept *67 to disable caller id
17:33.00Slingkyi even tried an ata connected directly to acanac and by doing *81 and phone number, it was working
17:33.09bcnxshid6: I just did the test, all phones are configured for registration every 120 seconds now and just saw two phones not responding in a group vall
17:33.10Slingkybut i have to figure how to do this with *
17:33.16bcnxvall=call
17:34.31Slingkydoes 8|. supposed to allow *671231231234 ???
17:35.45[TK]D-FenderSlingky, forget about your SIP phone sending the code, make sure that ASTERISK can do it the way you think its supposed to be able to manually.
17:36.05[TK]D-FenderSlingky, by default X-lite doen't have a real dialplan IIRC.
17:37.02Slingkyi just want to be able to block outgoing caller id
17:37.20Slingkyi don't understand why it's so difficult to configure
17:37.38Slingkyi tought it was a basic thing many people would want to do
17:37.40Slingkyno ?
17:37.49[TK]D-FenderSlingky, just DO IT.
17:38.14[TK]D-FenderSlingky, Make 1 simple dial line and TEST it, and don't ask us why its not working without providing us detailed CLI output.
17:38.17bcnxSlingky: welcome to the world of asterisk and open source. Not the most easy world, but a rewarding one.
17:38.18gardoanyone knows the way to transfer a call to a meetme room w/ xlite for linux?
17:38.43Slingkysorry, i don't complain about support, sorry if it is interpreted that way. english is not my primary language also
17:38.47Slingkythanks you for helping me
17:38.53Slingkyyou're nice
17:39.37Slingkyok, could you just tell me how to paste the cli log file ? cause output pass the screen very quickly
17:39.49gardoanyone using xlite and meetme?
17:40.13[TK]D-FenderSlingky, on peut egallement suive en francais , si tu va etre plus a l'aise..
17:40.28Slingkycool, êtes-vous de la france ou du québec ?
17:40.38[TK]D-Fender<- Montrealais
17:40.48Slingkycool, je suis de st-jean, rive-sud de mtl
17:41.12[TK]D-FenderSlingky, avec quoi connecter-vois a *?
17:41.27[TK]D-Fender(SSH, direct, etc)
17:41.48Slingkyje suis direct console car je roule trixbox sur vmware
17:41.56[TK]D-FenderSlingky, avec PuTTY (windows) t'es capable de fair un "scroll-back" de plusieurs lignes.
17:42.04[TK]D-FenderTrixbox?
17:42.08*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
17:42.10[TK]D-Fender~trixbox
17:42.10jbot[trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
17:42.25[TK]D-FenderSlingky, pas grande chance d'aide la....
17:42.36[TK]D-FenderSlingky, on ne le support pas ici.
17:42.59Slingkyje roule * 1.2.18
17:43.05[TK]D-FenderSlingky, pis personne ne veut rien savoir des problems de configuration
17:43.25Slingkyje vais tenter de parler que d'*
17:43.36tzafrirEnglish, please...
17:44.00[TK]D-FenderSlingky, l'affaire c'est qui ils son't pas VOS configuration.  C'est teinte au MAX pas l'interface FreePBX
17:44.27[TK]D-Fendertzanger, we're fine here, don't worry.
17:44.30*** join/#asterisk mgamble (n=me@static-1M-b1-1.highspeed.eol.ca)
17:45.05tzangereh?
17:45.20Slingkyet si j'avais asterisknow, ça serait différent ou pas ?
17:45.31mgambledoes anyone know how to get the p-asserted-identity field from a SIP peer as a variable to use in the dial plan?
17:46.23[TK]D-FenderSlingky, pas de tout.  Tous ces interface le configure comme QU'IL veut.
17:46.47[TK]D-Fendermgamble, "show function SIP_HEADER"
17:47.10mgamblethank you!
17:47.23Slingkypeux-tu me dire comment j'active un scroll plus long dans putty
17:47.40Slingkyil retiens plusieurs lignes mais ça bloque assez vite quand même
17:47.43[TK]D-FenderSlingky, laisse-faire les GUI, tu vas perdre du temps pour des problems non-evident et perdre control de vorte system
17:47.56*** part/#asterisk mgamble (n=me@static-1M-b1-1.highspeed.eol.ca)
17:48.06Slingkywindow, ok, 2 secondes
17:48.15Slingkylines of scrollback
17:48.42[TK]D-FenderSlingky, C'est dans :las base de "Option:Windows"
17:49.18[TK]D-Fendertzanger, bad autocomplete, wasn't for you, sorry
17:49.34tzanger[TK]D-Fender: ahh :-)
17:49.37tzangerhow goes?
17:49.50Slingkyregardes, ça semble dire ceci:
17:49.51SlingkyX-Asterisk-HangupCause: Unallocated (unassigned) number
17:50.25Slingkyon dirait qu'il n'aime pas le caractère "*"
17:50.29[TK]D-FenderSlingky, paste TOUT <--------
17:50.34Slingkyou le feature codes, je ne sais pas tout
17:50.35[TK]D-Fender~pb
17:50.36jbotA Pastebin is a place to paste your stuff without flooding the channel.  Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org
17:50.37[TK]D-Fender^^^^^^^^^^
17:50.55[TK]D-FenderSlingky, un linge de mem veut RIEN DIRE.
17:50.58escribzzHas anyone come up with a firewall solution for the return path soultion for the 7970's in asterisk over wan?
17:51.02tzangeryeah un REIN DIRE!!
17:51.09Qwellreindeer?
17:51.20tzangerQwell: no, reindire
17:51.31[TK]D-Fenderqwell : put DOWN the cookbook!
17:51.47escribzzor the 7961 or 7941 all the same :)
17:52.55Slingkyje crois que tu vas capoter...
17:52.56Slingkyhttp://pastebin.ca/609106
17:53.58Slingkyligne 147, ça commence je crois
17:54.42[TK]D-FenderSIP/2.0 404 Not Found
17:54.54[TK]D-FenderTo: <sip:*675148048384@66.49.255.38>;tag=as49c94d71
17:55.22[TK]D-FenderSlingky, C'est bien inclus dans le # enveoyer, sauf c'est pas acceptable de son bord.
17:55.36[TK]D-FenderSlingky, donne-mois le liens pour son site.
17:55.44Slingkyquel site ?
17:55.48Slingkyle provider ?
18:03.09[TK]D-FenderSlingky, oui
18:04.16[TK]D-FenderSlingky, peut-etre tu devrais faire *67 tout-seul pour effectuer l'effet sur l'apple qui SUIVE aussi....
18:04.57*** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil)
18:06.08Slingkymon provider sip est acanac
18:06.48Slingkysi je fais *67, * n'aime pas ça
18:07.34Slingkyil dit "the person you're trying to call is onavailable"
18:08.05Slingkysi je fais *67 depuis un ata, j'ai un genre de beep mais plus de tonalité pour composer, peut-être veux-tu un log
18:08.30*** join/#asterisk DanielX (n=danielx@201.240.80.68)
18:10.51*** join/#asterisk ruied (n=ruied@bl7-219-142.dsl.telepac.pt)
18:12.36ruiedcan I use postgres database to create users and extensions?
18:12.51mvanbaakruied: yes
18:13.25Slingkyregardes depuis le ata, si je fais *67, ça donne ça:
18:13.26Slingkyhttp://pastebin.ca/609133
18:16.00Slingkyok, j'ai plus pour toi
18:16.05ruiedmvanbaak: qhat do I need for that? just postgresql and asterisk? do you know a good info site for that?
18:16.32Slingkysi je prend le softphone x-lite, je peux communiquer avec la boîte vocale en faisant 8*123 et ça fonctionne
18:16.55Slingkymais si je fais 8*67, ça donne all circuits are busy now
18:17.03Slingkyça t'aides-tu ?
18:22.08ruiedwhat is the odbc for? is it to create some kind of real time interface between pgsql and asterisk? do I always need obdc?
18:28.25Kwakwahttp://www.voip-info.org/wiki-Asterisk / http://www.voip-info.org/wiki/view/Asterisk+RealTime
18:30.24tsurkoCould somebody explain the difference between ${ARGn} and ${VALn} in func_odbc.conf?
18:31.04ruiedKwakwa, thanks
18:32.38*** join/#asterisk yonahw (n=yonahw@IGLD-83-130-176-175.inter.net.il)
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18:39.50[TK]D-Fender#
18:39.50[TK]D-FenderLooking for *67 in from-internal (domain 192.168.11.112;user=phone)
18:39.50[TK]D-Fender#
18:39.50[TK]D-FenderReliably Transmitting (NAT) to 192.168.11.114:5060:
18:39.50[TK]D-Fender#
18:39.51[TK]D-FenderSIP/2.0 404 Not Found
18:40.14[TK]D-FenderSlingky, ca c'est ton configuration de FreePBX qui'il est a faut
18:44.46Slingkyquoi, je ne comprend pas ce que tu veux dire ?
18:46.33[TK]D-FenderFreePBX le refuse, pas to ITSP.
18:47.19*** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net)
18:48.06Slingkyça c'est quand je fais pas 9 devant le *67
18:48.25[TK]D-FenderSlingky, et avec?
18:48.33Slingkypeut-être as-tu raison et mon provider n'accepte pas *67
18:49.24Slingkycependant, ça n'explique pas comment le linksys spa2102 ata réussi à dire à mon provider de bloquer le caller id lorsque je fais *81
18:51.16Strom_Mmauvais numero!
18:51.23Strom_Mje ne sais telebec
18:51.34*** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net)
18:52.20SlingkyStrom_M : est-ce de mon problème que tu parles ?
18:53.21[TK]D-FenderSlingky : il-y-a trop de varibles dans votre "probleme". il faut les isoler
18:53.46Slingkyok, tout part d'un seul problème
18:53.47[TK]D-FenderSlingky, utilize X-Lite en attendant.
18:54.06SlingkyJe veux bloquer le caller id
18:54.57[TK]D-FenderSlingky, No, ton configuration du SPA c'est UNE problem.  Done configuration de FreePBX C'est une autre, est on ne sais meme pas si ton ITSP va ACCEPTER un appel au "*67" tout-seul
18:56.02[TK]D-FenderSlingky, C'est le dernier qu'il faut confirmer en premier
18:56.46*** join/#asterisk Strom_M (n=strom@63.164.47.227)
18:57.52Slingkyok, mais on peut partir de ce qui fonctionne ? l'ata spa2102 est capable de dire à mon provider de bloquer le caller id
18:58.12Slingkydonc, je crois que le provider peut le faire. via *67 ou via des packets ip, je sais pas
18:58.22Slingkymais c'est quand même ça de connu...
19:01.06*** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net)
19:02.16[TK]D-FenderSlingky, NON, melange pas ces fonctionalites!
19:02.30[TK]D-FenderOUBLIE ton ATA pour l'instant.
19:02.59[TK]D-FenderSlingky, * ne reussi meme-pas D'ESSAYER de passer *67 tout-seul a ton ITSP
19:10.25Slingkyécoutes, le code *123 sert à accéder le voicemail chez mon provider acanac
19:10.37Slingkysi je fais 9*123, ça fonctionne
19:10.46[TK]D-FenderSlingky, et 9*67?
19:11.48Slingky9*67 ou encore 9*675148048384 me dit toujours "all circuits are busy now"
19:12.19*** join/#asterisk rtcg (n=rtcg@static-71-244-46-30.dllstx.fios.verizon.net)
19:12.27*** join/#asterisk Weezey (n=weezey@206.210.109.233)
19:12.39Weezeyanyone have the user/pass for voip supply's FTP?
19:12.48WeezeyI'm looking for F1000 firmware.
19:13.46[TK]D-FenderWeezey, go ask them for it.
19:14.03[TK]D-FenderSlingky, montre-moi l'appel....
19:17.56Slingkyhttp://pastebin.ca/609226
19:19.19*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net)
19:23.36[TK]D-FenderSlingky Je vois aucun essaye de composer ce # a l'exterior, et ce ne'st pas le *67 quie j'ai demande no-plus.
19:24.31Slingkydésolé, je ne te suis pas, qu'est-ce que tu veux que je te montre ?
19:26.13[TK]D-FenderSlingky, ou la-dnas as-tu essayer de composer *67 sans RIEN DE PLUS?
19:27.15Slingkyje te le fais, 2 secondes
19:27.56Slingkyavec le 9 en avant tu veux dire ?
19:29.23Slingkyparce que ma route est comme suit: 9|.
19:30.18tzangerwtf my a/c is fucked I think
19:30.22tzangerit's been on for the last 4 hours
19:30.28tzangeronly in the last 20 minutes has it started working
19:30.36tzangerdelta-t has been at most a half a degree
19:30.39tzangerand now it's at like 10oC
19:30.42tzangerand it'll hit 18
19:30.44*** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com)
19:30.46tzangerweather's been constant
19:31.32[TK]D-FenderSlingky : je veut voir que ton configuration essaie d'envoyer *67 a ton ITSP.  Comment tu le configure pour le faire m'interesse pas.
19:32.24*** join/#asterisk Dovid (n=Dovid@bzq-88-155-222-40.red.bezeqint.net)
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19:33.27Slingkyhttp://pastebin.ca/609245
19:35.16[TK]D-FenderSlingky, Vois-tu qu'il passe l'appele a ton ITSP?
19:35.41*** join/#asterisk ManxPower (n=manxpowe@015-790-961.area5.spcsdns.net)
19:36.27Slingkyil semble le passer, donc ça voudrait dire qu'acanac, mon provider, ne supporte pas le *67
19:36.30Slingkyc'est ça ?
19:38.56[TK]D-FenderSlingky, c'est ou leur address IP dans cette appel?
19:40.07[TK]D-Fendertzanger, No, wonder why the two of us ARE the only conversation in here and realize that maybe, juat MAYBE this place is a little less like a ghost-town today with us here :)
19:41.48tzafrir_laptopactually not totally a ghost-town. tzanger was writing here before ;-)
19:42.03tzangeryep
19:42.05tzangerhahaha
19:42.10tzangerI was offtopic like I usually am
19:42.26tzangerI haven't got any problem with alternative languages in here, *especially* if the channel's just idle or mostly idle
19:42.49tzangerwhen it's busy as hell it gets troublesome but man, french electrons, english electrons, I dun givea shit
19:43.23tzafrir_laptopבסדר
19:44.29tzangerright back at ya.  :-)
19:44.41Slingkyregardes ici: c'est ce qui se produit quand j'appelle mon cellulaire et ça fonctionne
19:44.42Slingkyhttp://pastebin.ca/609254
19:45.55[TK]D-FenderSlingky, ca c'est la premiere appel ou * a meme ESAYER de utiliser ton ITSP
19:45.55[TK]D-Fender-- Executing Dial("SIP/60-0855eaa8", "SIP/Acanac/5148048384|300|") in new stack
19:46.16[TK]D-FenderSlingky, Ton configuration est pourri.
19:46.45Slingkyce que je viens d'envoyer fonctionne pourtant
19:46.52[TK]D-FenderSlingky, il a jamais essayer d'evoyer *67 direct vers ton ITSP comme tu voulais
19:46.55Slingkyc'est lorsque j'essaie d'appeler un numéro standard
19:47.06Slingkyoublies le *67
19:47.10Slingkypour le moment
19:47.30Slingkyje t'ai envoyé ce qui se passe quand je fais un appel sur mon cellulaire au 5148048384
19:47.48Slingkycomme ça on va essayer de voir le ip de mon itsp làdedans pour commencer
19:48.32Slingkyle host de mon provider est: 66.49.255.42
19:48.53Slingkypourtant je ne vois pas cette adresse dans le pastebin et pourtant ça fonctionne
19:49.10[TK]D-Fender*SIGH*
19:49.53[TK]D-FenderSlingky, Malgre, je peut rien faire avec tout ca.
19:50.33Slingkypas grave merci beaucoup quand même pour ton temps
19:56.06*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
19:56.31tzanger[TK]D-Fender: not le sigh?
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20:06.28*** join/#asterisk Tee`` (i=tee@stop.rooting.us)
20:07.58Tee``Quick question, I can't seem to find a more related channel- Does anyone know if IDEFisk Softphone Biz supports conferencing over IAX?
20:09.33*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)
20:09.48*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net)
20:10.31*** join/#asterisk seele_ (n=seele@dns.datawareltda.com)
20:11.31seele_hello I need help with phpagi, when I try to execute a example the log returns AGI Script example.php completed, returning 0, any suggest?
20:19.17*** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net)
20:22.55[TK]D-Fenderseele_, we suggest DETAILS are essential to us helping you.
20:23.27seele_thanks is working now ... a typo error with php path
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20:29.08Dovidselle_: Do u have the include in the begining of ur AGI ?
20:29.13Dovidand u have php on ur box ?
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20:32.17[TK]D-FenderDovid, .. its FIXED.  Let it go......
20:34.03*** part/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-28-192.hsd1.ct.comcast.net)
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20:42.28Dovidhehe
20:42.37Dovidgoto love php when u forget a ; or a }
20:43.31mvanbaakDovid: most languages will bork on you when you forget that
20:44.49Dovidyea. soooo much fun
20:44.52*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
20:45.37NuggetIt's weird that people have started to use php as a general-purpose scripting language.  It's so dismally inappropriate for that.
20:49.56mvanbaakNugget: php cli is not too bad for general-purpose scripting
20:50.20Nuggetexcept hardly anyone has it installed, it makes code re-use a challenge, and.... it's php.
20:50.49mvanbaakmeh
20:50.58mvanbaakphp-cgi is almost the same
20:51.02NuggetI think it is abysmal for general-purpsoe scripting.
20:51.03mvanbaakif you run it with -q it's the same
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20:51.17NuggetIt's the same as long as the other guy compiled php the same way you did.
20:51.30Nuggetwhich is absolutely insane for a scripting environment
20:52.21mvanbaakI agree there are better languages for general-purpose scripting
20:52.40*** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-87.ph.ph.cox.net)
20:52.56mvanbaakbut if you are an admin and create scripts for your own enviroment only it's ok to pick php if that is what you know best
20:53.11NuggetI guess we just disagree aboutt that.
20:53.18mvanbaakyup
20:53.24mvanbaakI wont use it for that
20:53.31mvanbaakbut I can imagine ppl will
20:53.40NuggetOh, you don't have to imagine!  :)
20:53.54NuggetPeople do -- I'm just saying that the fact that people do doesn't make it a good idea.
20:54.06mvanbaakthen we agree
20:54.36NuggetThere are people who use microsoft excel to construct databases, because it's "what they know" but that doesn't mean that excel is a good choice for designing a database.
20:54.40mvanbaakit's all a matter of personal taste
20:55.06[TK]D-FenderYay, another pointless scripting-nazi war!
20:55.15mvanbaaklol
20:55.27[TK]D-FenderClearly you have fogotten that PostgreSQL > MySQL!
20:55.41mvanbaakoracle ftw !
20:55.57[TK]D-FenderOr wait.... BSD = Denial that Linux is the future!
20:56.00NuggetPainting your car orange or wearing white socks with sandals is a matter of taste.  Using PHP as a general-purpose scripting language is just plain bad.  :)
20:56.25mvanbaaklinux is soooooooo 90's
20:56.46[TK]D-Fenderand BSD is sooooooo 80's ;)
20:56.50mvanbaakuhhuh
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20:57.03Habbieah
20:57.11Habbiea FUD performance, performed by a number of well-meaning idiots
20:57.13Habbiehow quaint
20:57.14mvanbaakthe one true OS is ultrix
20:58.08[TK]D-FenderHabbie, You clearly have no sense of humour, nor a sufficient sarcasm detection center of your brain :p
20:58.15mvanbaakyesterday I installed win 3.11 just to have some fun with it
20:58.15Habbiea sarcasm detector
20:58.17Habbiewhat a useful invention!
20:58.19Habbie:}}
20:58.40[TK]D-Fendermvanbaak, I've got a friend who keeps a legacy PC around for his old games....
20:58.44mvanbaakactually it performed pretty ok on my 486 DX2
20:59.04Habbiekeeping a legacy PC around for gaming makes sense until you buy a box that can run dosbox fast enough, really
20:59.06mvanbaak[TK]D-Fender: same here
20:59.17mvanbaakHabbie: not really
20:59.23Nuggethttp://www.theonion.com/content/video/report_70_percent_of_all_praise  <-- sarcasm detector
20:59.39mvanbaaktry to play the old original volfied on a modern pc with dosbox
20:59.43HabbieNugget, hehe
20:59.53*** join/#asterisk treeshoo (i=hello@bas2-toronto12-1167861059.dsl.bell.ca)
20:59.55mvanbaakit's way too fast
21:00.02Habbiemvanbaak, dosbox is pretty tunable
21:00.52Habbiethat is a great game, by the way :)
21:01.05mvanbaakit sure is
21:01.09Habbiehttp://www.abandonia.com/games/811/Volfied has dosbox notes it seems (haven't actually read it)
21:01.52*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
21:03.45tzangervolfied
21:03.50tzangerthere's a game I have not played in forever
21:04.00tzangermind you I just got Hades Nebula and a C64 emulator
21:04.03tzangerthat was a good game
21:04.51fujinmost c64 games were awesome ;\
21:04.55tzangeryep
21:04.59tzangerthis one particularly so
21:15.20Nuggetfeh, the Atari 800 was way cooler than your crappy Commodore 64.  :)
21:16.05NuggetThe Atari had four channel sound, the C64 only had three channel sound.  Plus the Atari had twice as many sprites and a GTIA graphics coprocessor.
21:28.44*** join/#asterisk ManxPower (n=manxpowe@015-820-709.area5.spcsdns.net)
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21:42.07tzanger<PROTECTED>
21:42.11tzangerbut I liked my c64 better
21:47.58macTijnamiga ftw :)
21:53.28mvanbaakI wonder what the key to fire the guns in volfied was
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23:05.25fujinHi asterisks!
23:09.21crayz_How would I go about generating dummy calls for test purposes? I'm trying to test inbound calls coming in via IAX2, to make sure the dialplan is working right
23:09.53jameswf7777
23:09.56*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
23:14.17*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
23:14.25kiscokidcrayz If you have two Asterisk machines you could use call files
23:16.16*** join/#asterisk Jameno123 (n=james@alkaline.cvg3.bytehosting.com)
23:18.10JTfujin: asterisks, what is thatg?
23:18.13JT-g
23:18.56Jameno123know any reason chan_agent would auto-logout an agent, for no reason? (no debug information, nothing in queue log, ect)
23:19.17Jameno123After ti logs them out, it refuses to let them login again for a random amount of time.
23:19.57Jameno123it doesnt generate any Manager Events, nothing.
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23:24.40Jameno123(wondering how to trace it)
23:25.01Jameno123agent callback
23:25.15Jameno123it seems like because its on a remote server that returns CONGESTION if the agent is busy.
23:25.31fujinJT: eh
23:25.38fujinPeople who are in #asterisk
23:25.41fujinI dunno ;]
23:33.07ruied_I've created a crd table in postgresql (asterisk database) and installed odbc, what do I need to add in 'extconfig.conf' so asterisk can write to the cdr table
23:33.20ruied_not crd, cdr...
23:38.09*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
23:39.53fujinI remember reading a tutorial on it somewhere
23:42.05snuff-homehttp://www.voip-info.org/wiki/view/Asterisk+cdr+pgsql
23:42.18snuff-homevoip-info is ur friend
23:43.36ruied_thanks, going to read... :)
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23:50.15ruied_<PROTECTED>
23:50.36ruied_in voip-info, seems that I just need to configure cdr.conf and have the table and user created in postgres... don't I need odbc?
23:51.33snuff-homeprobably not since ODBC is a generic container for accessing many different db's
23:51.59snuff-homeand * has a built in pgsql module for realtime/db cdr
23:52.58*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
23:54.20ruied_snuff-home, but it seems that is doesn't have a module for sip user accounts... that's what seems strange to me...
23:54.34*** join/#asterisk ManxPower (n=manxpowe@015-821-344.area5.spcsdns.net)
23:55.35ruied_why does it have for cdr and for users it's seems to be needed an odbc?
23:55.47Jameno123hrmm.. does joshua colp come on IRC?
23:56.19Jameno123he seems to work in chan_agent alot, maybe he could help fix this odd issue that started with asterisk 1.4.4 :(

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