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) |
05:28.07 | CrashSys | I have a global var 1001=SIP/1001. I have ${ARG1}=1001. How can I load the Global variable from ${ARG}? Something like $[${ARG1}] ? |
05:28.41 | CrashSys | Or would it be ${[${ARG1}]} ? |
05:31.14 | CrashSys | ${${ARG1}} ? |
05:33.53 | CrashSys | Hmm... looks like the last one will work... :) |
05:40.07 | BSD_Tech | exten = *40,n,GotoIF(${AVAILSTATUS}=""|?i,1|set) ? |
05:42.38 | CrashSys | ${${ARG1}} worked for what I need... I call the dial macro with 1001 and it loads the SIP/1001 to the dial cmd... |
05:43.01 | CrashSys | or was that not for me? |
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05:51.27 | Jameswf | allo all |
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06:01.27 | Jameswf | ping |
06:04.14 | BSD_Tech | got it |
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06:04.49 | Jameswf | thats random |
06:12.37 | BSD_Tech | night |
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06:37.06 | *** join/#asterisk JuliusSpencer (n=julius@60-234-130-224.bitstream.orcon.net.nz) |
06:38.00 | JuliusSpencer | hi |
06:39.33 | Jameswf | ho |
06:40.16 | JuliusSpencer | I have a problem at the moment with my tdm400p card |
06:40.35 | JuliusSpencer | just wondering if anyone here might be able to help |
06:40.53 | Jameswf | post the problem we shall see |
06:41.18 | JuliusSpencer | the module wctdm seems to be loading, but when running ztcfg I get an error: |
06:41.33 | JuliusSpencer | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
06:41.50 | Jameswf | do you have your channels defined in zaptel and how |
06:42.00 | JuliusSpencer | I was here earlier today, and I tried changing pci slots and checking the power was plugged in to the card etc |
06:42.15 | JuliusSpencer | yeah; only with the channels in the config do I get the errors |
06:42.17 | JuliusSpencer | : |
06:42.22 | Jameswf | type: cat /proc/zaptel/* |
06:42.38 | JuliusSpencer | fxsks=1,2 fxoks=3,4 |
06:43.09 | JuliusSpencer | there's no files in /proc/zaptel |
06:43.17 | JuliusSpencer | there are I mean :) |
06:43.20 | Jameswf | then no zap device is loaded |
06:43.52 | Jameswf | post the lspci output for the card |
06:44.08 | JuliusSpencer | 00:05.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
06:44.43 | Jameswf | run update-pciids and see if it changes |
06:45.24 | JuliusSpencer | 00:05.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interfac |
06:45.26 | JuliusSpencer | looks the same |
06:45.37 | JuliusSpencer | never run that command before :) |
06:49.41 | JuliusSpencer | hi Jameswf , you still with me? |
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06:54.04 | Jameswf | sorry playin on google as I work with rhino not digium |
06:54.12 | Jameswf | http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5#53FXOFXSComboCardsDevices |
06:54.18 | Jameswf | read 5.3.2 |
06:55.04 | JuliusSpencer | thanks I'll check. I've spent probably about 2.5 hours using google to solve this, |
06:55.22 | Jameswf | rhino cards just go :);) |
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07:03.19 | tzafrir | JuliusSpencer, run lspci -v |
07:04.08 | tzafrir | Various TDM400P cards have various sub-vendor IDs |
07:04.11 | JuliusSpencer | k |
07:04.20 | tzafrir | What version of zaptel do you have? |
07:04.26 | JuliusSpencer | Subsystem: Unknown device b199:0001 ...? |
07:04.43 | JuliusSpencer | zaptel-1.2.18 |
07:06.26 | tzafrir | JuliusSpencer, modinfo wctdm | grep ^alias |
07:07.03 | tzafrir | this will show you what vendor IDs / product IDs / SubVendor-ids the driver looks for (in this case) |
07:07.38 | Jameswf | tzafrir smells a nock off? |
07:07.56 | tzafrir | yeah |
07:08.47 | JuliusSpencer | so what should I postin here? |
07:08.53 | JuliusSpencer | or should I post somewhere else |
07:09.07 | Jameswf | gotta love china... Had a customer tell me they got 4 bad ciccos in a batch of 100 from china |
07:09.24 | Jameswf | *cisco |
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07:10.48 | JuliusSpencer | ok I posted it here: http://pastebin.ca/608351 |
07:12.04 | JuliusSpencer | I'm pretty sure it's the real deal :) |
07:13.22 | Jameswf | did you run kudzu? |
07:13.26 | JuliusSpencer | me? |
07:13.27 | JuliusSpencer | no |
07:13.31 | JuliusSpencer | I don't think so |
07:13.42 | JuliusSpencer | I could try :) |
07:13.46 | Jameswf | NO |
07:13.50 | Jameswf | kudzu bad |
07:14.07 | Jameswf | where did you buy the card? |
07:17.12 | brian | 4 bad ciccos? |
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07:17.31 | Jameswf | I am having a bad S day |
07:17.39 | brian | ciscos? |
07:18.18 | Jameswf | yeah |
07:18.37 | JuliusSpencer | digium |
07:20.01 | Jameswf | if you do "lsmod | grep zaptel" what do you get |
07:20.09 | JuliusSpencer | it's there.. i'l lpost it :) |
07:20.40 | JuliusSpencer | http://pastebin.ca/608357 |
07:25.39 | Jameswf | for giggles run modprobe wct4xxp |
07:25.51 | Jameswf | then cat /proc/zaptel/* |
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07:29.25 | JuliusSpencer | still nothing.... |
07:29.32 | JuliusSpencer | in /proc/zaptel |
07:30.04 | Jameswf | how are the modules loaded on to the card |
07:31.17 | JuliusSpencer | you mean plugged into the card, two green two red. |
07:31.30 | JuliusSpencer | I tried swapping the config as a test |
07:31.32 | JuliusSpencer | bi change |
07:31.52 | Jameswf | ok again im not a digium guy o's and 's please |
07:32.26 | JuliusSpencer | oh two fxs s and two fxo s |
07:33.11 | Jameswf | ok put the fxo's before the fxs's be sure the card is connected to power |
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07:34.34 | JuliusSpencer | I'll give it a go... still seems like a sw problem :) |
07:35.01 | Corydon76-home | JuliusSpencer: did you plug in the Molex connector to the back of the card? |
07:35.13 | JuliusSpencer | Molex... is that like the 4 pin power cable? |
07:35.19 | JuliusSpencer | daisy chained from the hard drives |
07:35.26 | Corydon76-home | Okay, good |
07:35.28 | Jameswf | molex = brand name, |
07:35.57 | Corydon76-home | If 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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07:36.01 | Jameswf | try another lead as well. If the card is not properly powered with s' it will flip out |
07:36.11 | JuliusSpencer | heheh |
07:36.36 | Corydon76-home | Check your dmesg, too |
07:37.05 | Corydon76-home | dmesg will tell you if any of the daughter cards are bad |
07:37.20 | Jameswf | This is why i use Rhino no drama :) |
07:37.39 | Corydon76-home | There's no drama if you know what you're doing |
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07:38.34 | Jameswf | Ok you got me after the first 30 i do it while i sleep |
07:39.24 | Jameswf | on that note what OS are you using JuliusSpencer |
07:48.45 | JuliusSpencer | I checked the power with a volt meter and it gave 5V |
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07:49.10 | JuliusSpencer | OS is Fedora |
07:55.21 | JuliusSpencer | ok I swapped the modules arounf |
07:55.24 | JuliusSpencer | around |
07:56.48 | JuliusSpencer | I'm not opening up the computer again |
07:56.50 | JuliusSpencer | heh |
08:08.27 | JuliusSpencer | hmm I still get the same problem after swapping the modules around. |
08:08.58 | Jameswf | my guess is bad mobo or hardware issue |
08:14.54 | JuliusSpencer | hmmm |
08:20.43 | JuliusSpencer | anyone else got any ideas? |
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08:53.56 | *** join/#asterisk SktyNick (n=nick@202.43.236.188) |
08:56.18 | SktyNick | Evening 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. |
08:57.45 | SktyNick | Effectivly 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. |
08:57.58 | SktyNick | Trying t avoid using Call files :-) |
08:58.03 | Strom_M | h extension |
08:59.38 | SktyNick | Perfect - Thanks. |
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09:20.52 | SktyNick | Strom - 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. |
09:21.07 | Strom_M | well, duh |
09:21.16 | Strom_M | if you want a new call, use a call file |
09:21.51 | SktyNick | Is 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? |
09:22.19 | Strom_M | you'd have to go through some serious dialplan gymnastics |
09:22.29 | Strom_M | what's the problem with call files/ |
09:22.31 | Strom_M | ? |
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09:23.13 | brian | SktyNick, You can use the Asterisk Manager Interface to place a call |
09:23.33 | Bladerunner05 | Using grandstram gxp-2000 and asterisk if I press the speaker key the ohter part can't hear me. |
09:23.37 | SktyNick | Because 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 :-) |
09:24.05 | Strom_M | SktyNick: so you dynamically generate teh call file upon hangup |
09:24.09 | Strom_M | simple enough |
09:24.17 | brian | SktyNick, You should probably create a AGI script to do that for you and then use AMI to originate a call to the original caller. |
09:24.40 | brian | SktyNick, That's how I handle callbacks and it seems to work fine |
09:25.16 | SktyNick | Yeah 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 |
09:25.39 | brian | SktyNick, You need to use AMI to originate the call |
09:26.35 | SktyNick | Yeah thats NP. We are already using AMI to tie Asterisk into our Nagios stuff. |
09:26.53 | brian | SktyNick, Sounds like cool stuff. What exactly are you all doing? |
09:27.23 | brian | SktyNick, You're a telemarketer aren't you? AREN'T YOU?!?! |
09:28.21 | brian | SktyNick, Was it you that called me just I was sitting down to eat dinner? |
09:28.23 | SktyNick | Argh 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. |
09:28.38 | brian | SktyNick, *cough* can you get me a job *cough* |
09:28.44 | brian | SktyNick, Sorry I've got a cold |
09:29.14 | SktyNick | *cough* Not unless you want to move to Aust. and work for free *cough*. Yeah cold seems to be going around :-) |
09:29.24 | brian | :( |
09:29.28 | brian | work for free? |
09:29.30 | brian | awwww |
09:30.08 | SktyNick | Cold is a subset of being a whip cracking tightass srry :) |
09:30.33 | brian | I see. |
09:30.37 | tzafrir | It's not for free: the experince gained there is priceless. Is that a better phrasing? |
09:30.50 | brian | How will I live in Australia with no money? |
09:30.55 | SktyNick | Wow! You are either in marketing or HR heh heh. |
09:31.44 | SktyNick | Plenty of empty boxes in dark laneways Brian, if you pick a decent one there may even be a unsecured Access Point in range :-) |
09:34.15 | kiscokid | where do you get power for your laptop? |
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09:38.54 | SktyNick | You 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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09:40.32 | MJ[Lappy | Hey :) 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? |
09:41.04 | MJ[Lappy | If I've got it right, the SMSC stores the SMS until the receiving end requests it? |
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09:56.04 | tzafrir | I wonder if zaptel actually need to look for the kernel source under /usr/src/linux |
09:56.27 | tzafrir | considering that there is totally no guarantee that this is the source of the current kernel |
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09:57.32 | tzafrir | and that if one has installed a kernel from source, the symlink /lib/modules/REVISION/build should be in place |
09:59.46 | SktyNick | In a call file - Can you use priority labels? |
10:01.10 | tzafrir | SktyNick, hmm... I guess not. Those don't really exist in the parsed dialplan |
10:02.18 | SktyNick | Hrmm cool. Will get around it |
10:02.54 | tzafrir | SktyNick, look at 'show dialplan <contextname>' |
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11:12.41 | romano2k | hi everyone! anacron sends me this message frequently: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
11:12.54 | romano2k | i apt-getted asterisk on my debian box, what should i do? |
11:42.57 | matsk | romano2k, ~book |
11:43.09 | matsk | ~book |
11:43.14 | jbot | i 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 |
11:43.44 | JT | romano2k: is the asterisk daemon actually running? |
11:44.59 | matsk | noop |
11:45.11 | matsk | not with that message |
11:45.16 | JT | matsk: not so |
11:45.21 | JT | it may or may not be running |
11:45.58 | matsk | ok so you recommend a "ps aux | grep asterisk" to reveal that ;-) |
11:46.13 | JT | yes |
11:46.47 | matsk | I now remember that an old asterisk version didn't create the ctl file |
11:46.57 | matsk | so a touch solved it |
11:51.39 | tzafrir | romano2k, is asterisk actually running? |
11:52.06 | tzafrir | /etc/init.d/asterisk start |
11:52.54 | tzafrir | also: 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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12:00.10 | romano2k | JT: sorry for the latence, nop, it is not |
12:00.29 | romano2k | tzafrir: not running |
12:01.09 | tzafrir | /etc/init.d/asterisk debug |
12:01.24 | frawd | hi, 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... |
12:01.48 | romano2k | tzafrir: 3 lines, do i paste here or in a website ? |
12:01.53 | JT | romano2k: it'd help to have it running before attemting to reconnect or run a remote command on it :) |
12:02.01 | tzafrir | well, here |
12:02.04 | romano2k | Debugging Asterisk PBX: |
12:02.04 | romano2k | Unable to set high priority |
12:02.04 | romano2k | Unable to setgid to 111! |
12:02.21 | romano2k | JT: i'm not willing to do anything with asterisk at the moment |
12:02.29 | romano2k | it's just installed and configured for further use |
12:02.35 | tzafrir | You need to run that as root |
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12:03.00 | tzafrir | to check why asterisk has failed to start |
12:03.26 | romano2k | oh i'm sorry |
12:03.47 | romano2k | it's quite long this time :) |
12:06.40 | frawd | I 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? |
12:09.27 | frawd | for 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 |
12:10.53 | JT | romano2k: logic would dictate to remove the anacron job then or at least disable it :) |
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12:28.58 | CpuID | frawd, nice l7 :) |
12:29.04 | CpuID | youll have to go userspace soon though :) |
12:33.24 | tzafrir | frawd, is there actually is a problem? |
12:33.41 | tzafrir | grep wctdm /proc/interrupts |
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12:44.21 | coppice | tzafrir: I'm just putting the debian build files into the spandsp package. 0.0.4pre3 has a file called libspandsp3.install. Is that right? |
12:45.09 | tzafrir | coppice, the files are generally maintained in the pkg-voip repository |
12:45.12 | tzafrir | let's see... |
12:45.34 | coppice | I was asked to add them to the package, so I'm doing it |
12:47.03 | frawd | tzafrir, the problem is a disturbing bip during conversations once in a while |
12:47.10 | tzafrir | The files are under svn://svn.debian.org/pkg-voip/spandsp/trunk |
12:47.31 | frawd | # grep wctdm /proc/interrupts |
12:47.31 | frawd | <PROTECTED> |
12:47.31 | coppice | also, there is stuff messing about with mmx.h, but that file was removed some time ago because of iff licencing |
12:47.44 | tzafrir | debian/libspandsp3.install is indeed there |
12:47.59 | coppice | yes, but shouldn't it be 4 and not 3? |
12:48.25 | tzafrir | that number is bumped internally in debian |
12:48.43 | frawd | CpuID, going to l7 userspace is planned |
12:48.57 | CpuID | coo :) |
12:49.06 | CpuID | yea i changed my kernel over ready for userspace last week |
12:49.10 | CpuID | havent got my rules active yet though |
12:49.13 | CpuID | on the TODO :) |
12:49.47 | tzafrir | kdelibs of kde3 are kdelibs4. Go figure |
12:50.36 | tzafrir | coppice, mmx.h was re-added later after being cleared? |
12:50.59 | tzafrir | or reimplemented or whatever? |
12:51.14 | coppice | i got rid of it completely, but the debian stuff seems to be playing with it still |
12:51.48 | frawd | CpuID, looks promising but i keep on kernel version until userspace is proven stable... |
12:52.08 | frawd | anyway, any idea on that wctdm bip/interrupts problem? |
12:52.46 | CpuID | ya :) |
12:53.09 | CpuID | frawd, cat /proc/interrupts, is wctdm sharing any irqs with otehr hw? |
12:53.16 | frawd | nop |
12:53.19 | CpuID | ive found wctdm to be nasty with irqs |
12:53.32 | frawd | indeed |
12:53.34 | CpuID | occasional artifacts due to hdd usage, even when not sharing |
12:53.43 | frawd | # grep wctdm /proc/interrupts |
12:53.43 | frawd | <PROTECTED> |
12:53.46 | CpuID | im hoping to change out my digium hw sometime soon |
12:53.54 | CpuID | possibly try sangoma |
12:54.03 | tzafrir | there's a patch called "nommx" to remove mmx.h, but it is not applied |
12:54.27 | frawd | i'm not sure sangoma would be much better on that issue |
12:54.38 | frawd | it still does need interrupts |
12:55.13 | frawd | could it have relation with delays of ext3 on software RAID 1? |
12:55.53 | CpuID | hmm you know...im runnign sw raid on that box actually |
12:55.59 | CpuID | i have a feeling im using reiser though |
12:56.24 | CpuID | ya reiser for everything on that box |
12:56.41 | frawd | i tried with reiser as well... |
12:57.17 | CpuID | weird |
12:57.24 | frawd | i also tried with hundreds of options in kernel (disable patches, try all preemption parameters, change HZ values, ....) without success |
12:57.28 | CpuID | couldnt say myself, i havent done a lotta testing as yet |
12:57.46 | frawd | same problem in 2 different (but similar) motherboards |
12:58.32 | frawd | maybe an APIC problem with those specific processors? |
12:58.57 | frawd | i really have no idea, even of how to debug or identify the source of the problem... |
13:01.05 | frawd | i 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... |
13:01.28 | frawd | or maybe i look in the wrong direction |
13:01.50 | frawd | but what more could i do to check/resolve that? |
13:16.20 | bcnx | Hi 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? |
13:27.36 | russellb | bcnx: yeah, the CDR would reflect the call was answered at the time of Answer() |
13:34.00 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
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13:56.50 | bcnx | russellb: 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()? |
13:57.17 | bcnx | if not people calling my compay will start paying before someone even answered their call ... |
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14:47.30 | coppice | the * mailing list seems seriously screwed up since the change. does anyone know why? |
14:52.47 | BSD_Tech | you did not talk dirty to it |
14:52.58 | BSD_Tech | you did not send it flowers |
14:53.33 | coppice | I sent it nice warm loving e-mails, and they came back 4 days later |
14:54.56 | BSD_Tech | hmmm maybe thats why I have not seen any activity on the asetrisk-user and asterisk-bsd lists |
14:55.14 | tzafrir | coppice, there is a thread about this. It seems to be a that some people have delays whereas others don't |
14:56.26 | coppice | some would call this a downgrade |
14:56.26 | coppice | some would call it a screwup |
14:56.26 | coppice | americans would call it a negative upgrade |
14:56.29 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
14:57.05 | Kwakwa | or a complete success :) |
14:57.33 | coppice | it meets all out goals - especially the "own" ones |
14:59.38 | BSD_Tech | digium strikes again |
14:59.42 | BSD_Tech | lol |
15:01.04 | *** join/#asterisk msetim (n=marcos@200-103-130-66.ctame706.dsl.brasiltelecom.net.br) |
15:01.07 | BSD_Tech | Kram goes to EU and the staff start playing with things go figure |
15:03.12 | [TK]D-Fender | coppice, "Mission Accomplished" <------------- |
15:14.59 | BSD_Tech | TK I got it fixed at 4 am |
15:15.03 | BSD_Tech | it works |
15:15.16 | BSD_Tech | but I want to take it another step |
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15:21.00 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
15:21.46 | marl | hi, 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? |
15:22.24 | marl | have tried simply swapping the channel and ext, but that fails |
15:25.25 | shido6 | heh |
15:27.34 | shido6 | whats the context name and extension u want to dial |
15:27.36 | [TK]D-Fender | marl, change your channel |
15:28.03 | [TK]D-Fender | marl, And then obviously where it will lead. |
15:28.28 | marl | do u meen just put the ext number in channel? 205? |
15:29.50 | [TK]D-Fender | marl, Ask yourself what KIND of channel will let you go through the dialplan to meet your needs |
15:31.17 | Jameswf | does the channel matter so much as the context it sits in? |
15:32.58 | [TK]D-Fender | Jameswf, Channels do not reside in contexts. Your concept of *'s heirarchy is very confused |
15:34.14 | shido6 | well... |
15:34.24 | shido6 | you could use Channel: Local/extension@context |
15:34.39 | Jameswf | no but channels have nothing to do with the way a call is handled it is only the medium |
15:35.01 | [TK]D-Fender | Shido : sure, just SPIT out the answer and have them forego any conscious effort.... |
15:35.06 | shido6 | LOL! |
15:35.40 | [TK]D-Fender | Jameswf, And you'll want to seperate your newfound attachment between "call" and "channel" |
15:35.41 | Jameswf | if(!knowledge){rtfm}; |
15:35.59 | Jameswf | wtf are you talking about |
15:36.46 | [TK]D-Fender | Jameswf, 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-Fender | Jameswf, Where THIS should have stood out : "Local: Loopback into another context" |
15:37.16 | Jameswf | I have all the answers... they are on my Blackberry |
15:37.18 | Jameswf | :) |
15:37.39 | [TK]D-Fender | SMRT |
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15:54.05 | shido6 | Local:/205@ext-local this is wrong |
15:54.40 | shido6 | Local/205@ext-local |
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16:22.31 | rbd | can 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) |
16:26.24 | BSD_Tech | cool I just fixed the bsd asterisk port 1.4.6 + zaptel build and works |
16:29.27 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
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16:34.39 | BSD_Tech | when is libpri going to be updated to 1.4.1 |
16:35.11 | Qwell | when it needs to be |
16:39.36 | bcnx | whos' got a propeller hat for me |
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16:41.54 | Jameswf | oh snap |
16:45.04 | bcnx | all, if a sip registration expires for a phone, will it still ring when dialed? |
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17:01.25 | shido6 | the 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) |
17:05.08 | bcnx | shido6: I experience odd stuff with sip phones: they only seem to be able to work when set to registration |
17:05.19 | bcnx | I'd like to use them without registration however |
17:06.03 | shido6 | you CAN do that |
17:06.04 | bcnx | when 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 |
17:06.08 | shido6 | if you know their IP addies |
17:06.13 | shido6 | when they change... :) |
17:06.25 | shido6 | if after "some time" they dont work |
17:06.33 | shido6 | then force the phone to register more frequently |
17:07.15 | bcnx | ueah, 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) |
17:07.30 | shido6 | where are these phones in relation to your ast box? |
17:07.33 | shido6 | nat'd ? |
17:07.33 | bcnx | I could be dreaming though, have been working 10 hours striaght now |
17:07.41 | bcnx | no, LAN |
17:07.59 | bcnx | so no real need for registration |
17:08.14 | bcnx | except for the fact that they don't work without |
17:08.23 | shido6 | what kind of phones? |
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17:08.27 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
17:08.35 | bcnx | linksys 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) |
17:10.54 | k31th | bcnx: wats the advantage to not using reg ? |
17:11.20 | bcnx | I'm not sure that the phones are available during the reg process |
17:11.37 | bcnx | if you can confirm they are, I'll switch back to a frequent registration config |
17:15.18 | bcnx | should 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) |
17:16.05 | Slingky | hi guys! does somebody know how to pass "*67" to sip provider |
17:16.59 | Slingky | i always get "all circuits are busy now" |
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17:26.03 | escribzz | Slingky check your dial plan in the device your using |
17:26.20 | escribzz | its probably set to interprate the *code |
17:26.37 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
17:28.17 | Slingky | right now, i'm trying to solve this from x-lite softphone before trying to solve in ata |
17:28.17 | Slingky | is there something in x-lite that doesn't allow * codes ? |
17:31.10 | shido6 | does your provider accept those digits as valid? |
17:32.33 | Slingky | yeah, it's acanac and accept *67 to disable caller id |
17:33.00 | Slingky | i even tried an ata connected directly to acanac and by doing *81 and phone number, it was working |
17:33.09 | bcnx | shid6: 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 |
17:33.10 | Slingky | but i have to figure how to do this with * |
17:33.16 | bcnx | vall=call |
17:34.31 | Slingky | does 8|. supposed to allow *671231231234 ??? |
17:35.45 | [TK]D-Fender | Slingky, 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-Fender | Slingky, by default X-lite doen't have a real dialplan IIRC. |
17:37.02 | Slingky | i just want to be able to block outgoing caller id |
17:37.20 | Slingky | i don't understand why it's so difficult to configure |
17:37.38 | Slingky | i tought it was a basic thing many people would want to do |
17:37.40 | Slingky | no ? |
17:37.49 | [TK]D-Fender | Slingky, just DO IT. |
17:38.14 | [TK]D-Fender | Slingky, Make 1 simple dial line and TEST it, and don't ask us why its not working without providing us detailed CLI output. |
17:38.17 | bcnx | Slingky: welcome to the world of asterisk and open source. Not the most easy world, but a rewarding one. |
17:38.18 | gardo | anyone knows the way to transfer a call to a meetme room w/ xlite for linux? |
17:38.43 | Slingky | sorry, i don't complain about support, sorry if it is interpreted that way. english is not my primary language also |
17:38.47 | Slingky | thanks you for helping me |
17:38.53 | Slingky | you're nice |
17:39.37 | Slingky | ok, could you just tell me how to paste the cli log file ? cause output pass the screen very quickly |
17:39.49 | gardo | anyone using xlite and meetme? |
17:40.13 | [TK]D-Fender | Slingky, on peut egallement suive en francais , si tu va etre plus a l'aise.. |
17:40.28 | Slingky | cool, êtes-vous de la france ou du québec ? |
17:40.38 | [TK]D-Fender | <- Montrealais |
17:40.48 | Slingky | cool, je suis de st-jean, rive-sud de mtl |
17:41.12 | [TK]D-Fender | Slingky, avec quoi connecter-vois a *? |
17:41.27 | [TK]D-Fender | (SSH, direct, etc) |
17:41.48 | Slingky | je suis direct console car je roule trixbox sur vmware |
17:41.56 | [TK]D-Fender | Slingky, avec PuTTY (windows) t'es capable de fair un "scroll-back" de plusieurs lignes. |
17:42.04 | [TK]D-Fender | Trixbox? |
17:42.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
17:42.10 | [TK]D-Fender | ~trixbox |
17:42.10 | jbot | [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-Fender | Slingky, pas grande chance d'aide la.... |
17:42.36 | [TK]D-Fender | Slingky, on ne le support pas ici. |
17:42.59 | Slingky | je roule * 1.2.18 |
17:43.05 | [TK]D-Fender | Slingky, pis personne ne veut rien savoir des problems de configuration |
17:43.25 | Slingky | je vais tenter de parler que d'* |
17:43.36 | tzafrir | English, please... |
17:44.00 | [TK]D-Fender | Slingky, 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-Fender | tzanger, we're fine here, don't worry. |
17:44.30 | *** join/#asterisk mgamble (n=me@static-1M-b1-1.highspeed.eol.ca) |
17:45.05 | tzanger | eh? |
17:45.20 | Slingky | et si j'avais asterisknow, ça serait différent ou pas ? |
17:45.31 | mgamble | does 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-Fender | Slingky, pas de tout. Tous ces interface le configure comme QU'IL veut. |
17:46.47 | [TK]D-Fender | mgamble, "show function SIP_HEADER" |
17:47.10 | mgamble | thank you! |
17:47.23 | Slingky | peux-tu me dire comment j'active un scroll plus long dans putty |
17:47.40 | Slingky | il retiens plusieurs lignes mais ça bloque assez vite quand même |
17:47.43 | [TK]D-Fender | Slingky, 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) |
17:48.06 | Slingky | window, ok, 2 secondes |
17:48.15 | Slingky | lines of scrollback |
17:48.42 | [TK]D-Fender | Slingky, C'est dans :las base de "Option:Windows" |
17:49.18 | [TK]D-Fender | tzanger, bad autocomplete, wasn't for you, sorry |
17:49.34 | tzanger | [TK]D-Fender: ahh :-) |
17:49.37 | tzanger | how goes? |
17:49.50 | Slingky | regardes, ça semble dire ceci: |
17:49.51 | Slingky | X-Asterisk-HangupCause: Unallocated (unassigned) number |
17:50.25 | Slingky | on dirait qu'il n'aime pas le caractère "*" |
17:50.29 | [TK]D-Fender | Slingky, paste TOUT <-------- |
17:50.34 | Slingky | ou le feature codes, je ne sais pas tout |
17:50.35 | [TK]D-Fender | ~pb |
17:50.36 | jbot | A 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-Fender | Slingky, un linge de mem veut RIEN DIRE. |
17:50.58 | escribzz | Has anyone come up with a firewall solution for the return path soultion for the 7970's in asterisk over wan? |
17:51.02 | tzanger | yeah un REIN DIRE!! |
17:51.09 | Qwell | reindeer? |
17:51.20 | tzanger | Qwell: no, reindire |
17:51.31 | [TK]D-Fender | qwell : put DOWN the cookbook! |
17:51.47 | escribzz | or the 7961 or 7941 all the same :) |
17:52.55 | Slingky | je crois que tu vas capoter... |
17:52.56 | Slingky | http://pastebin.ca/609106 |
17:53.58 | Slingky | ligne 147, ça commence je crois |
17:54.42 | [TK]D-Fender | SIP/2.0 404 Not Found |
17:54.54 | [TK]D-Fender | To: <sip:*675148048384@66.49.255.38>;tag=as49c94d71 |
17:55.22 | [TK]D-Fender | Slingky, C'est bien inclus dans le # enveoyer, sauf c'est pas acceptable de son bord. |
17:55.36 | [TK]D-Fender | Slingky, donne-mois le liens pour son site. |
17:55.44 | Slingky | quel site ? |
17:55.48 | Slingky | le provider ? |
18:03.09 | [TK]D-Fender | Slingky, oui |
18:04.16 | [TK]D-Fender | Slingky, 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) |
18:06.08 | Slingky | mon provider sip est acanac |
18:06.48 | Slingky | si je fais *67, * n'aime pas ça |
18:07.34 | Slingky | il dit "the person you're trying to call is onavailable" |
18:08.05 | Slingky | si 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) |
18:12.36 | ruied | can I use postgres database to create users and extensions? |
18:12.51 | mvanbaak | ruied: yes |
18:13.25 | Slingky | regardes depuis le ata, si je fais *67, ça donne ça: |
18:13.26 | Slingky | http://pastebin.ca/609133 |
18:16.00 | Slingky | ok, j'ai plus pour toi |
18:16.05 | ruied | mvanbaak: qhat do I need for that? just postgresql and asterisk? do you know a good info site for that? |
18:16.32 | Slingky | si je prend le softphone x-lite, je peux communiquer avec la boîte vocale en faisant 8*123 et ça fonctionne |
18:16.55 | Slingky | mais si je fais 8*67, ça donne all circuits are busy now |
18:17.03 | Slingky | ça t'aides-tu ? |
18:22.08 | ruied | what is the odbc for? is it to create some kind of real time interface between pgsql and asterisk? do I always need obdc? |
18:28.25 | Kwakwa | http://www.voip-info.org/wiki-Asterisk / http://www.voip-info.org/wiki/view/Asterisk+RealTime |
18:30.24 | tsurko | Could somebody explain the difference between ${ARGn} and ${VALn} in func_odbc.conf? |
18:31.04 | ruied | Kwakwa, thanks |
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18:39.50 | [TK]D-Fender | # |
18:39.50 | [TK]D-Fender | Looking for *67 in from-internal (domain 192.168.11.112;user=phone) |
18:39.50 | [TK]D-Fender | # |
18:39.50 | [TK]D-Fender | Reliably Transmitting (NAT) to 192.168.11.114:5060: |
18:39.50 | [TK]D-Fender | # |
18:39.51 | [TK]D-Fender | SIP/2.0 404 Not Found |
18:40.14 | [TK]D-Fender | Slingky, ca c'est ton configuration de FreePBX qui'il est a faut |
18:44.46 | Slingky | quoi, je ne comprend pas ce que tu veux dire ? |
18:46.33 | [TK]D-Fender | FreePBX le refuse, pas to ITSP. |
18:47.19 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
18:48.06 | Slingky | ça c'est quand je fais pas 9 devant le *67 |
18:48.25 | [TK]D-Fender | Slingky, et avec? |
18:48.33 | Slingky | peut-être as-tu raison et mon provider n'accepte pas *67 |
18:49.24 | Slingky | cependant, ça n'explique pas comment le linksys spa2102 ata réussi à dire à mon provider de bloquer le caller id lorsque je fais *81 |
18:51.16 | Strom_M | mauvais numero! |
18:51.23 | Strom_M | je ne sais telebec |
18:51.34 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
18:52.20 | Slingky | Strom_M : est-ce de mon problème que tu parles ? |
18:53.21 | [TK]D-Fender | Slingky : il-y-a trop de varibles dans votre "probleme". il faut les isoler |
18:53.46 | Slingky | ok, tout part d'un seul problème |
18:53.47 | [TK]D-Fender | Slingky, utilize X-Lite en attendant. |
18:54.06 | Slingky | Je veux bloquer le caller id |
18:54.57 | [TK]D-Fender | Slingky, 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-Fender | Slingky, C'est le dernier qu'il faut confirmer en premier |
18:56.46 | *** join/#asterisk Strom_M (n=strom@63.164.47.227) |
18:57.52 | Slingky | ok, mais on peut partir de ce qui fonctionne ? l'ata spa2102 est capable de dire à mon provider de bloquer le caller id |
18:58.12 | Slingky | donc, je crois que le provider peut le faire. via *67 ou via des packets ip, je sais pas |
18:58.22 | Slingky | mais 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-Fender | Slingky, NON, melange pas ces fonctionalites! |
19:02.30 | [TK]D-Fender | OUBLIE ton ATA pour l'instant. |
19:02.59 | [TK]D-Fender | Slingky, * ne reussi meme-pas D'ESSAYER de passer *67 tout-seul a ton ITSP |
19:10.25 | Slingky | écoutes, le code *123 sert à accéder le voicemail chez mon provider acanac |
19:10.37 | Slingky | si je fais 9*123, ça fonctionne |
19:10.46 | [TK]D-Fender | Slingky, et 9*67? |
19:11.48 | Slingky | 9*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) |
19:12.39 | Weezey | anyone have the user/pass for voip supply's FTP? |
19:12.48 | Weezey | I'm looking for F1000 firmware. |
19:13.46 | [TK]D-Fender | Weezey, go ask them for it. |
19:14.03 | [TK]D-Fender | Slingky, montre-moi l'appel.... |
19:17.56 | Slingky | http://pastebin.ca/609226 |
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19:23.36 | [TK]D-Fender | Slingky Je vois aucun essaye de composer ce # a l'exterior, et ce ne'st pas le *67 quie j'ai demande no-plus. |
19:24.31 | Slingky | désolé, je ne te suis pas, qu'est-ce que tu veux que je te montre ? |
19:26.13 | [TK]D-Fender | Slingky, ou la-dnas as-tu essayer de composer *67 sans RIEN DE PLUS? |
19:27.15 | Slingky | je te le fais, 2 secondes |
19:27.56 | Slingky | avec le 9 en avant tu veux dire ? |
19:29.23 | Slingky | parce que ma route est comme suit: 9|. |
19:30.18 | tzanger | wtf my a/c is fucked I think |
19:30.22 | tzanger | it's been on for the last 4 hours |
19:30.28 | tzanger | only in the last 20 minutes has it started working |
19:30.36 | tzanger | delta-t has been at most a half a degree |
19:30.39 | tzanger | and now it's at like 10oC |
19:30.42 | tzanger | and it'll hit 18 |
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19:30.46 | tzanger | weather's been constant |
19:31.32 | [TK]D-Fender | Slingky : je veut voir que ton configuration essaie d'envoyer *67 a ton ITSP. Comment tu le configure pour le faire m'interesse pas. |
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19:33.27 | Slingky | http://pastebin.ca/609245 |
19:35.16 | [TK]D-Fender | Slingky, Vois-tu qu'il passe l'appele a ton ITSP? |
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19:36.27 | Slingky | il semble le passer, donc ça voudrait dire qu'acanac, mon provider, ne supporte pas le *67 |
19:36.30 | Slingky | c'est ça ? |
19:38.56 | [TK]D-Fender | Slingky, c'est ou leur address IP dans cette appel? |
19:40.07 | [TK]D-Fender | tzanger, 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 :) |
19:41.48 | tzafrir_laptop | actually not totally a ghost-town. tzanger was writing here before ;-) |
19:42.03 | tzanger | yep |
19:42.05 | tzanger | hahaha |
19:42.10 | tzanger | I was offtopic like I usually am |
19:42.26 | tzanger | I haven't got any problem with alternative languages in here, *especially* if the channel's just idle or mostly idle |
19:42.49 | tzanger | when it's busy as hell it gets troublesome but man, french electrons, english electrons, I dun givea shit |
19:43.23 | tzafrir_laptop | בסדר |
19:44.29 | tzanger | right back at ya. :-) |
19:44.41 | Slingky | regardes ici: c'est ce qui se produit quand j'appelle mon cellulaire et ça fonctionne |
19:44.42 | Slingky | http://pastebin.ca/609254 |
19:45.55 | [TK]D-Fender | Slingky, 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-Fender | Slingky, Ton configuration est pourri. |
19:46.45 | Slingky | ce que je viens d'envoyer fonctionne pourtant |
19:46.52 | [TK]D-Fender | Slingky, il a jamais essayer d'evoyer *67 direct vers ton ITSP comme tu voulais |
19:46.55 | Slingky | c'est lorsque j'essaie d'appeler un numéro standard |
19:47.06 | Slingky | oublies le *67 |
19:47.10 | Slingky | pour le moment |
19:47.30 | Slingky | je t'ai envoyé ce qui se passe quand je fais un appel sur mon cellulaire au 5148048384 |
19:47.48 | Slingky | comme ça on va essayer de voir le ip de mon itsp làdedans pour commencer |
19:48.32 | Slingky | le host de mon provider est: 66.49.255.42 |
19:48.53 | Slingky | pourtant 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-Fender | Slingky, Malgre, je peut rien faire avec tout ca. |
19:50.33 | Slingky | pas grave merci beaucoup quand même pour ton temps |
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19:56.31 | tzanger | [TK]D-Fender: not le sigh? |
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20:06.28 | *** join/#asterisk Tee`` (i=tee@stop.rooting.us) |
20:07.58 | Tee`` | Quick question, I can't seem to find a more related channel- Does anyone know if IDEFisk Softphone Biz supports conferencing over IAX? |
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20:10.31 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
20:11.31 | seele_ | 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-Fender | seele_, we suggest DETAILS are essential to us helping you. |
20:23.27 | seele_ | thanks is working now ... a typo error with php path |
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20:28.38 | *** mode/#asterisk [+o mog] by ChanServ |
20:29.08 | Dovid | selle_: Do u have the include in the begining of ur AGI ? |
20:29.13 | Dovid | and u have php on ur box ? |
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20:32.17 | [TK]D-Fender | Dovid, .. its FIXED. Let it go...... |
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20:42.28 | Dovid | hehe |
20:42.37 | Dovid | goto love php when u forget a ; or a } |
20:43.31 | mvanbaak | Dovid: most languages will bork on you when you forget that |
20:44.49 | Dovid | yea. soooo much fun |
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20:45.37 | Nugget | It's weird that people have started to use php as a general-purpose scripting language. It's so dismally inappropriate for that. |
20:49.56 | mvanbaak | Nugget: php cli is not too bad for general-purpose scripting |
20:50.20 | Nugget | except hardly anyone has it installed, it makes code re-use a challenge, and.... it's php. |
20:50.49 | mvanbaak | meh |
20:50.58 | mvanbaak | php-cgi is almost the same |
20:51.02 | Nugget | I think it is abysmal for general-purpsoe scripting. |
20:51.03 | mvanbaak | if you run it with -q it's the same |
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20:51.17 | Nugget | It's the same as long as the other guy compiled php the same way you did. |
20:51.30 | Nugget | which is absolutely insane for a scripting environment |
20:52.21 | mvanbaak | I agree there are better languages for general-purpose scripting |
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20:52.56 | mvanbaak | but 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 |
20:53.11 | Nugget | I guess we just disagree aboutt that. |
20:53.18 | mvanbaak | yup |
20:53.24 | mvanbaak | I wont use it for that |
20:53.31 | mvanbaak | but I can imagine ppl will |
20:53.40 | Nugget | Oh, you don't have to imagine! :) |
20:53.54 | Nugget | People do -- I'm just saying that the fact that people do doesn't make it a good idea. |
20:54.06 | mvanbaak | then we agree |
20:54.36 | Nugget | There 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. |
20:54.40 | mvanbaak | it's all a matter of personal taste |
20:55.06 | [TK]D-Fender | Yay, another pointless scripting-nazi war! |
20:55.15 | mvanbaak | lol |
20:55.27 | [TK]D-Fender | Clearly you have fogotten that PostgreSQL > MySQL! |
20:55.41 | mvanbaak | oracle ftw ! |
20:55.57 | [TK]D-Fender | Or wait.... BSD = Denial that Linux is the future! |
20:56.00 | Nugget | Painting 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. :) |
20:56.25 | mvanbaak | linux is soooooooo 90's |
20:56.46 | [TK]D-Fender | and BSD is sooooooo 80's ;) |
20:56.50 | mvanbaak | uhhuh |
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20:57.03 | Habbie | ah |
20:57.11 | Habbie | a FUD performance, performed by a number of well-meaning idiots |
20:57.13 | Habbie | how quaint |
20:57.14 | mvanbaak | the one true OS is ultrix |
20:58.08 | [TK]D-Fender | Habbie, You clearly have no sense of humour, nor a sufficient sarcasm detection center of your brain :p |
20:58.15 | mvanbaak | yesterday I installed win 3.11 just to have some fun with it |
20:58.15 | Habbie | a sarcasm detector |
20:58.17 | Habbie | what a useful invention! |
20:58.19 | Habbie | :}} |
20:58.40 | [TK]D-Fender | mvanbaak, I've got a friend who keeps a legacy PC around for his old games.... |
20:58.44 | mvanbaak | actually it performed pretty ok on my 486 DX2 |
20:59.04 | Habbie | keeping a legacy PC around for gaming makes sense until you buy a box that can run dosbox fast enough, really |
20:59.06 | mvanbaak | [TK]D-Fender: same here |
20:59.17 | mvanbaak | Habbie: not really |
20:59.23 | Nugget | http://www.theonion.com/content/video/report_70_percent_of_all_praise <-- sarcasm detector |
20:59.39 | mvanbaak | try to play the old original volfied on a modern pc with dosbox |
20:59.43 | Habbie | Nugget, hehe |
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20:59.55 | mvanbaak | it's way too fast |
21:00.02 | Habbie | mvanbaak, dosbox is pretty tunable |
21:00.52 | Habbie | that is a great game, by the way :) |
21:01.05 | mvanbaak | it sure is |
21:01.09 | Habbie | http://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) |
21:03.45 | tzanger | volfied |
21:03.50 | tzanger | there's a game I have not played in forever |
21:04.00 | tzanger | mind you I just got Hades Nebula and a C64 emulator |
21:04.03 | tzanger | that was a good game |
21:04.51 | fujin | most c64 games were awesome ;\ |
21:04.55 | tzanger | yep |
21:04.59 | tzanger | this one particularly so |
21:15.20 | Nugget | feh, the Atari 800 was way cooler than your crappy Commodore 64. :) |
21:16.05 | Nugget | The 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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21:42.07 | tzanger | <PROTECTED> |
21:42.11 | tzanger | but I liked my c64 better |
21:47.58 | macTijn | amiga ftw :) |
21:53.28 | mvanbaak | I wonder what the key to fire the guns in volfied was |
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22:06.19 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
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23:05.17 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
23:05.25 | fujin | Hi asterisks! |
23:09.21 | crayz_ | 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 |
23:09.53 | jameswf | 7777 |
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23:14.25 | kiscokid | crayz If you have two Asterisk machines you could use call files |
23:16.16 | *** join/#asterisk Jameno123 (n=james@alkaline.cvg3.bytehosting.com) |
23:18.10 | JT | fujin: asterisks, what is thatg? |
23:18.13 | JT | -g |
23:18.56 | Jameno123 | know any reason chan_agent would auto-logout an agent, for no reason? (no debug information, nothing in queue log, ect) |
23:19.17 | Jameno123 | After ti logs them out, it refuses to let them login again for a random amount of time. |
23:19.57 | Jameno123 | it doesnt generate any Manager Events, nothing. |
23:20.13 | *** join/#asterisk snuff-home (n=snuffy@C-59-101-201-33.bur.connect.net.au) |
23:23.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
23:24.40 | Jameno123 | (wondering how to trace it) |
23:25.01 | Jameno123 | agent callback |
23:25.15 | Jameno123 | it seems like because its on a remote server that returns CONGESTION if the agent is busy. |
23:25.31 | fujin | JT: eh |
23:25.38 | fujin | People who are in #asterisk |
23:25.41 | fujin | I dunno ;] |
23:33.07 | ruied_ | 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 |
23:33.20 | ruied_ | not crd, cdr... |
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23:39.53 | fujin | I remember reading a tutorial on it somewhere |
23:42.05 | snuff-home | http://www.voip-info.org/wiki/view/Asterisk+cdr+pgsql |
23:42.18 | snuff-home | voip-info is ur friend |
23:43.36 | ruied_ | thanks, going to read... :) |
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23:50.15 | ruied_ | <PROTECTED> |
23:50.36 | ruied_ | 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? |
23:51.33 | snuff-home | probably not since ODBC is a generic container for accessing many different db's |
23:51.59 | snuff-home | and * has a built in pgsql module for realtime/db cdr |
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23:54.20 | ruied_ | 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) |
23:55.35 | ruied_ | why does it have for cdr and for users it's seems to be needed an odbc? |
23:55.47 | Jameno123 | hrmm.. does joshua colp come on IRC? |
23:56.19 | Jameno123 | he seems to work in chan_agent alot, maybe he could help fix this odd issue that started with asterisk 1.4.4 :( |