00:00.29 | DerkKo | How is the recording mixed.... I want to use mixmonitor but have the 2 legs of the recording segragated by sterio channels |
00:00.56 | [TK]D-Fender | pkecastillo: do an AMI originate or Call file. Specify your channel to call. Set some variables. Point to an extension. Playback your variable filename. Dial out to another variable PSTN number via some tech. |
00:01.11 | p3nguin | derkko: You want to have two recordings or only one? |
00:01.30 | DerkKo | One but the caller on one sterio channels and the callee on the other |
00:01.37 | [TK]D-Fender | DerkKo: then MixMonitor is not for you. |
00:01.46 | p3nguin | derkko: MixMonitor() joins them together for you; Monitor() records each leg of the call individually. |
00:01.59 | [TK]D-Fender | DerkKo: use Monitor and call a clean-up script yourself to merge the way you want to |
00:02.31 | [TK]D-Fender | DerkKo: Hint: there is no format that * can write to in STEREO. |
00:02.32 | DerkKo | i would have to rearchitecture everything i do... right now i use the cli to trigger the recording start from a php script |
00:03.10 | DerkKo | well i said sterio channel to make sure it got not confused with any other soft of channels |
00:03.11 | pkecastillo | ~[TK]D-Fender~ thanks you very much, I will continue working on this.. thanks again!! |
00:03.12 | DerkKo | but i get you. |
00:03.33 | DerkKo | monitor() cant be invoked trough the cli :-( |
00:04.05 | pkecastillo | I notice if I have changes, and comment. |
00:04.29 | [TK]D-Fender | DerkKo: AMI <- |
00:04.48 | DerkKo | AMI ? |
00:04.58 | [TK]D-Fender | ~ami |
00:04.59 | infobot | well, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
00:05.00 | DerkKo | Oh management interface |
00:05.01 | joako | Does anyone know how do change the polycom configs server from the config file? |
00:05.01 | DerkKo | LOL |
00:05.03 | DerkKo | sorry! |
00:05.20 | [TK]D-Fender | joako: server.1. <- |
00:05.48 | DerkKo | yea that complicates things for me on the programing side... Specially playing around with sockets i guess i have no other choice though. |
00:06.21 | [TK]D-Fender | DerkKo: PHP-AGI has an AMI class in it that makes this easy |
00:06.51 | joako | [TK]D-Fender, Not reg.1.server.1.address= |
00:09.31 | manxpower | joako: it would be the flash device settings |
00:09.53 | joako | manxpower, Right it is some bootloader paramater that is not in the default config files |
00:10.02 | manxpower | see the docs in the Admin manual, no, they are not in the default configs. Normally you just set it via DHCP |
00:10.21 | manxpower | joako: it is documented in the Admin Guide |
00:11.03 | joako | manxpower, Do you know under what it is documeted? Because I have the SIP admin guide here and I can't find it |
00:12.12 | manxpower | Page A-157 of the 3.2.2 Admin Guide |
00:13.28 | joako | manxpower, but for legacy phones? |
00:16.06 | jsjc | Argh I erased my src/asterisk so some changes I did are gone. Few days ago I was told here a file where to change the global default language in asterisk so before compiling all will be set to es instead of en. Anyone has logs dating back 1 week or so? |
00:18.21 | p3nguin | ./include/asterisk.h DEFAULT_LANGUAGE |
00:18.24 | p3nguin | ? |
00:19.26 | jsjc | yeap |
00:19.31 | jsjc | thanks p3nguin ! |
00:20.36 | jaytee | jsjc, for future reference of IRC logs for this channel go here: http://ibot.rikers.org/ |
00:21.02 | jsjc | thanks jaytee I was looking for it in google. but I found a site that was no longer working. |
00:21.50 | jsjc | bookmarked!! ;) |
00:23.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
00:23.38 | ChannelZ | oh noes i are being recordaded? |
00:24.22 | florz | it seems like you is |
00:24.27 | manxpower | joako: for stuff running 2.x firmware you have no option other than DHCP or manual provisioning server setup |
00:25.03 | manxpower | and whatever bootloader is for that firmware branch. |
00:25.44 | manxpower | come to think of it the <device ... settings ARE set by the firmware. bootloader -> firmware -> firmware sets the flash options for the bootloader -> phone reboots |
00:26.11 | joako | manxpower, 3.x but some of the phones only go to 3.1.6 or something like that |
00:26.31 | manxpower | joako: I guess you'll have to read thru the changelogs |
00:27.37 | normaldotcom | after upgrading Asterisk to 1.4 on my centos box, calls are being dropped with "no reply to our critical packet" after 15-20 seconds |
00:27.44 | *** join/#asterisk Jumpie (n3rdz@ip68-98-31-152.ph.ph.cox.net) |
00:27.48 | Jumpie | yo |
00:28.15 | R-Guy | joako: Isn't the configure server provide using DHCP? |
00:29.49 | manxpower | normaldotcom: did you read the UPGRADE*.txt? |
00:30.13 | normaldotcom | manxpower: I should probably check that out :D |
00:35.21 | *** part/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
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01:01.31 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
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01:46.42 | *** join/#asterisk capitan (~captain@76.91.206.32) |
01:47.20 | Kobaz | mm |
01:47.26 | Kobaz | where's katty |
01:56.05 | *** join/#asterisk Caplain (~shayne@pool-72-64-149-69.tampfl.fios.verizon.net) |
01:56.19 | Caplain | Looking for 500 in from-pstn (domain pegasusrepair.com) |
01:56.19 | Caplain | billy*CLI> |
01:56.19 | Caplain | <--- Reliably Transmitting (NAT) to 66.54.140.46:5060 ---> |
01:56.19 | Caplain | SIP/2.0 404 Not Found |
01:56.22 | Caplain | that is a LIE |
01:56.35 | Caplain | extension 500 exists |
01:56.38 | Caplain | why does it hate me? |
02:03.14 | \malex\ | Qwell: yes, the allworx 9112 is a sip phone. i got it sort of working, where it connects and is able to place and receive calls. i'll have to wade through the config file to get the soft keys to work and such |
02:11.05 | fenrus | Caplain, did you reload the dialplan? :) |
02:11.12 | Caplain | yes |
02:11.15 | Caplain | many many times |
02:11.41 | Caplain | its an old asterisk setup that used freepbx |
02:12.24 | fenrus | dirty |
02:12.27 | Caplain | very |
02:12.28 | fenrus | =) |
02:12.34 | Caplain | so i think ill scrap it and start over :( |
02:12.53 | Caplain | the only reason i want freepbx is the music on hold feature |
02:13.17 | fenrus | moh is available in asterisk :) |
02:13.29 | Caplain | i know |
02:13.35 | Caplain | fpbx is a frontend |
02:13.49 | Caplain | but ...meh, ill figure it out. i was avoiding starting over |
02:21.15 | *** join/#asterisk L2Logic (~nospam@65-48-113-20-static.l2fiber.net) |
02:21.58 | L2Logic | good day.. has anyone experienced issues with rhino's 4 port t1 card with dahdi ? |
02:27.43 | ChannelZ | Caplain: Are you sure your context is right? |
02:27.58 | Caplain | yes im sure |
02:28.06 | Caplain | and ill be sure after the format, too |
02:28.13 | fenrus | =)) |
02:29.28 | ChannelZ | goes back to fapping |
02:31.37 | Caplain | lucky |
02:36.05 | *** join/#asterisk kartik (~koolkarti@117.199.118.1) |
02:39.54 | hardwire | fapfapfap |
02:47.53 | p3nguin | <Caplain> the only reason i want freepbx is the music on hold feature <-- what a weird thing to say. |
02:48.12 | Caplain | well its harder to set up via nano |
02:48.40 | p3nguin | Uh, no. |
02:49.38 | Caplain | fine. its harder to set up via nano when it was already set up via freepbx |
02:51.21 | Caplain | asterisk is now online :) |
02:52.24 | manxpower | FreePBX is like if you bought a car with the hood welded shut and when you finally open the hood, all the sizes are in Andorian. |
02:53.04 | manxpower | First you need to learn Andorian then you have to craft the tools to work with the andorian measurements. |
02:53.11 | manxpower | Easier to just do it right in the first place. |
03:00.59 | *** part/#asterisk Caplain (~shayne@pool-72-64-149-69.tampfl.fios.verizon.net) |
03:02.03 | p3nguin | Following his logic, it's easier to work on with the hood welded shut. |
03:04.54 | *** join/#asterisk moy (~chatzilla@74.12.121.97) |
03:05.13 | manxpower | p3nguin: yes, unless you need to modify something |
03:05.28 | jaytee | Andorians are far more trustworthy than Tellurites or Romulans |
03:05.55 | *** join/#asterisk kartik (~koolkarti@117.199.118.1) |
03:06.29 | manxpower | jaytee: yeah, but who else uses fractions of an andorian light year as a base unit of measure???? |
03:06.57 | *** part/#asterisk normaldotcom (~normaldot@pc14665.cedarville.edu) |
03:08.48 | jaytee | wonders how long an Andorian year is |
03:16.18 | *** join/#asterisk b14ck (b14ck@cpe-24-24-128-47.socal.res.rr.com) |
03:16.27 | manxpower | ...three Federation standard years (one Andorian year) .... |
03:17.31 | b14ck | Sup guys. I've got a weird problem. I'm trying to compile asterisk-addons, and I'm using ubuntu-server 8.10. I'm trying to compile with app_addon_mysql (it depends on mysqlclient), but I've already install mysql-client and it is still not available. Am I missing something? ubuntu doesn't have a `mysqlclient` package. |
03:18.10 | b14ck | Here's my package list: http://pastie.org/924182 |
03:18.17 | b14ck | I've already got it all installed :x |
03:18.22 | jaytee | starts worrying about manxpower and wonders if he can speak Klingon |
03:18.25 | b14ck | But the option in make menuselect is xxx'ed out |
03:20.56 | manxpower | jaytee: not at all, but I bet google does. |
03:21.20 | manxpower | b14ck: then ./configure did not find the requirements, and in fact it should tell you what is missing when you move to the XXX'd out option |
03:24.07 | b14ck | manxpower, it says: "checking for mysql_config... no |
03:24.07 | b14ck | " |
03:24.12 | b14ck | but there is no mysql_config package |
03:24.53 | b14ck | well lets see |
03:24.56 | b14ck | i'll install mysql-admin |
03:24.59 | b14ck | maybe thatll do it |
03:26.11 | fenrus | isnt the name of the packet mysql-client ? |
03:26.46 | manxpower | you are looking for -dev packages |
03:27.03 | manxpower | or -devel packages, depending on your distro |
03:28.04 | fenrus | my guess is libmysqlclient15-dev :) |
03:30.52 | Jumpie | anybody have any success running asterisk in conjunction with any house alarm systems? |
03:30.53 | Jumpie | :P |
03:31.04 | Jumpie | i thin it shouldnt really be an issue....dedicated line coming in...just need an fxo card right? |
03:35.24 | ChannelZ | sure |
03:38.39 | Jumpie | normally alarms have dedicated lines right? |
03:38.43 | Jumpie | they dont just call on the normal house #? |
03:39.07 | ChannelZ | it depends on how much you're paying and your system |
03:39.24 | Jumpie | he's rich, probably alot |
03:39.27 | Jumpie | its a honeywell vista 20p |
03:39.29 | ChannelZ | In some cases the alarm can be hooked directly up to a line and it will dial out to home base (not you) |
03:39.37 | Jumpie | guy's a congressman |
03:39.54 | Jumpie | well if that's the case i may not even ened to interface it at all then! |
03:40.00 | manxpower | you should patch the alarm into the line before Asterisk |
03:40.04 | Jumpie | whcih is the way i hope |
03:40.12 | Jumpie | manxpower...the system si already in place |
03:40.24 | Jumpie | replacing his crappy panasonic |
03:40.43 | manxpower | Jumpie: that does not change my answer |
03:41.03 | Jumpie | yea but...what do you mean by 'patch into the line'? |
03:41.13 | Jumpie | if its a 3rd line |
03:41.23 | manxpower | is the line used for anything else? |
03:41.58 | p3nguin | Why would you want your alarm to rely on yet another component such as asterisk and the computer which runs it? |
03:42.27 | Jumpie | p3nguin well..the problem is..its a proprietery setup im not 100% familiar with, and the customer doesnt know all teh technical details |
03:42.36 | Jumpie | IF its tied into his existing phone system, then yes ill have to tie it in |
03:42.41 | Jumpie | if itw as independant all along, then i shouldnt have to mes with it |
03:42.46 | manxpower | Jumpie: then you should get the information before jumpie-ing to assumptions |
03:42.51 | Jumpie | manxpower well no kidding |
03:42.51 | Jumpie | hehe |
03:42.59 | manxpower | from the customer |
03:43.02 | Jumpie | mi just trying to thinkg technically in my head for the future, and what i may have to do |
03:43.06 | Jumpie | yea, he's a busy guy hehe |
03:43.08 | Jumpie | he has to call the company |
03:43.21 | Jumpie | Vector |
03:43.27 | Jumpie | and give permission to let me talk about the account |
03:43.33 | Jumpie | and that's what im waiting on at the moment |
03:44.20 | manxpower | best to worst 1) dedicated line for the alarm 2) put a splitter on the line before the line gets to the PBX, one port goes to the PBX, one to the alarm system 3) worst possible: wire the alarm into the PBX as an analog extension |
03:44.54 | manxpower | and 4) practically guarnteed to not work: put the alarm on an ATA |
03:46.07 | Jumpie | manxpower if its a dedicted line coming in to the alarm... |
03:46.13 | Jumpie | and it has dialtone..wouldnt i need an fxo port? |
03:46.22 | Jumpie | or is the alarm itself an 'analog device' needin an fxs port? |
03:47.11 | p3nguin | The alarm system may very well be just like a phone, which connects TO an FXS port. |
03:47.43 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
03:47.47 | ChannelZ | Surely this puke congressman can afford a dedicated line for it anyway |
03:48.02 | Jumpie | ChannelZ no doubt |
03:48.11 | Jumpie | he got his 11 year old daughter a $1200 laptop |
03:48.53 | manxpower | the device would have the correct interface to talk to the telco |
03:49.42 | manxpower | all the ones I've seen used dedicated POTS lines that just plug into the side of the alarm unit mounted in the telco closet. The system calls out each night to verify connectivity with the alarm company (and who know what else it does). |
03:50.12 | Jumpie | manxpower..ah true |
03:50.14 | manxpower | If I was an alarm company I would not provide service unless the device has a dedicated POTS lines |
03:50.48 | Jumpie | agreed |
03:50.56 | Jumpie | dont wanna miss out on a call because your daugher was chit chatting with a friend |
03:50.57 | p3nguin | closed-loop system |
03:50.57 | Jumpie | :P |
03:51.13 | *** join/#asterisk Caplain (~shayne@pool-72-64-149-69.tampfl.fios.verizon.net) |
03:51.16 | Caplain | no reply to our critical packet |
03:51.21 | Caplain | im behind a nat |
03:51.28 | Caplain | i set nat=yes |
03:51.32 | Caplain | i still get htat error |
03:51.35 | Caplain | what do |
03:51.45 | p3nguin | Is your asterisk system behind NAT? |
03:54.22 | ChannelZ | The other end doesn't know it, or your firewall is blocking outgoing |
03:55.35 | ChannelZ | Or the NAT is busted completely |
03:55.40 | ChannelZ | So many choices |
03:56.49 | Caplain | p3nguin, yes |
03:57.23 | p3nguin | Did you also correctly set localnet and externip or externhost? |
03:57.26 | Caplain | its a pos verizon gateway router. i told those gits all i wanted was a modem |
03:57.34 | Caplain | i did not... |
03:58.02 | manxpower | I wonder what would make a car alarm go off exactly every 30 mins |
03:58.51 | manxpower | considering the neighborhood I live in, I almost expect some to shoot the SOB that owns that fscking annoying car alarm |
03:58.56 | coppice | a car thief with a watch? |
03:58.59 | p3nguin | punk-ass kids |
03:59.17 | manxpower | coppice: I just got in from watching the damn thing. not anyone around. |
03:59.36 | manxpower | my balcony is on the 2nd level about 40 ft from it. |
03:59.46 | manxpower | actually more like 30 |
04:00.17 | p3nguin | Now that I think about it, isn't it likely that the alarm's phone line is a dry pair? |
04:00.37 | manxpower | p3nguin: that was the old school way. modern ones are all POTS in my experience |
04:00.43 | p3nguin | If so, it wouldn't do any good to hook it to ... oh. |
04:00.58 | p3nguin | I didn't realize things had changed. |
04:01.37 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
04:01.37 | manxpower | I work for a CLEC, with Verizon as the ILEC and we always have a customer with an POTS alarm out of service somewhere |
04:02.32 | *** join/#asterisk isb-asteri (~geosohaib@119.152.110.102) |
04:03.35 | coppice | manxpower: a ninja car thief with a watch? |
04:03.48 | ChannelZ | and a beanbag gun |
04:04.23 | fenrus | not very ninja to trip the alarm |
04:04.47 | manxpower | I don't mean "around 30 mins", I mean almost exactly 30 mins. based on my observations it drifts by less than 1 min per day |
04:04.48 | ChannelZ | But ninja not to be seen doing it |
04:05.02 | coppice | that's what he's trying to do. he has been contracted to annoy the neghbours |
04:05.14 | ChannelZ | Car alarms are so pointless |
04:05.17 | isb-asteri | I am using Asterisk 1.6.0.21 |
04:05.34 | ChannelZ | Cool. I am using 1.6.1.18 |
04:05.36 | manxpower | it belongs to the new BF of the downstairs neighbor. |
04:05.37 | isb-asteri | i have little problem regarding outbound calls |
04:05.53 | coppice | most people just want the car thief to finish his work and drive away to stop the noise |
04:06.58 | isb-asteri | hello |
04:07.37 | ChannelZ | Yeah hello |
04:08.22 | fenrus | hm, i wonder if theres rent-a-car-thief -service somewhere |
04:08.27 | ChannelZ | Don't priv me please, ask in the channel |
04:08.43 | isb-asteri | ok |
04:08.57 | coppice | fenrus: have you seen the prices Hertz charge? |
04:09.12 | fenrus | coppice, :) |
04:09.57 | Jumpie | coppice yep |
04:10.01 | Jumpie | i told one lady to jump off a cliff |
04:10.07 | Jumpie | i was gettin a quote movin from maryland to phoenix |
04:10.13 | Jumpie | she said like $7200 |
04:10.18 | Jumpie | i asked her if she was high |
04:11.06 | ChannelZ | Did she laugh like a pot smoker? |
04:11.39 | Jumpie | haha |
04:11.40 | Jumpie | she got quiet |
04:11.43 | Jumpie | i kenw she was full of it |
04:11.50 | Jumpie | i mean..even when i was in the army and i would PCS across country |
04:11.55 | Jumpie | i saw the summary of charges |
04:11.57 | Jumpie | and it wasnt near that |
04:12.07 | Jumpie | i think an 18 footer with FULL service, as in they box, pack, etc was under 5k |
04:16.47 | *** join/#asterisk rdircio (~admin@201.137.45.224) |
04:17.06 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
04:17.35 | *** join/#asterisk sourcode (~code@ppp-58-8-238-222.revip2.asianet.co.th) |
04:20.15 | coppice | I had an avis car for 3 days once, when I returned it their system printed out a bill for something like $3k. when I said there must have been a mistake, they said there was a penalty for early returns - I'd only had the car 70 hours |
04:20.39 | fenrus | lol |
04:21.11 | Jumpie | i would have told them to shove that bill up their ass |
04:21.26 | Jumpie | if anything it seems an early return is good, because they can re rent it out sooner than they planned |
04:21.52 | coppice | the manager had to tear up the original agreement, and produce a new and sane one that I paid for |
04:22.05 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
04:22.27 | Jumpie | lol sane |
04:22.28 | Jumpie | exactly |
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04:49.11 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
04:49.23 | *** join/#asterisk diegomad (mad@190.158.77.40) |
05:40.14 | *** join/#asterisk infobot (ibot@rikers.org) |
05:40.14 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
05:40.23 | carrar | HOT++ |
05:40.32 | ChannelZ | Yeah. Please stop. |
05:40.34 | isb-asteri | <PROTECTED> |
05:40.34 | isb-asteri | [Apr 17 10:39:30] WARNING[32496]: app_dial.c:860 wait_for_answer: Unable to forward voice or dtmf |
05:40.35 | isb-asteri | <PROTECTED> |
05:40.35 | isb-asteri | <PROTECTED> |
05:40.35 | isb-asteri | <PROTECTED> |
05:40.35 | isb-asteri | <PROTECTED> |
05:41.09 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
05:41.20 | carrar | looks like ${TRUNKMSD} isn't 1 |
05:41.41 | isb-asteri | let me check it again |
05:41.46 | carrar | just replace it with :1 for testing |
05:43.06 | ChannelZ | Too much junk in your trunk |
05:43.33 | isb-asteri | exten => _9.,1,Dial(${TRUNK}/${EXTEN:1}) |
05:44.42 | isb-asteri | replaced it with 1 |
05:45.01 | isb-asteri | but same result |
05:45.25 | carrar | yeah you are still missing stuff |
05:45.54 | isb-asteri | yeah missing but dont know what |
05:46.02 | carrar | Where is the console dial output |
05:46.44 | Jumpie | but why would that only happen with land line |
05:46.51 | Jumpie | maybe i dont udnersatnd those type of number formats cause im nit he states but |
05:46.57 | Jumpie | do mobile have some kinda diff prefix |
05:47.06 | Jumpie | ah ok..i scrolled up |
05:47.27 | isb-asteri | yes mobile have different |
05:47.37 | isb-asteri | Channel 0/31, span 1 got hangup request, cause 3 |
05:47.46 | ChannelZ | That isn't the important part |
05:47.47 | isb-asteri | <PROTECTED> |
05:47.57 | carrar | ~binpaste |
05:48.06 | carrar | ~bp |
05:48.07 | infobot | rumour has it, bp is pb backwards |
05:48.13 | ChannelZ | ~pastebin |
05:48.14 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
05:48.15 | carrar | heh |
05:48.19 | carrar | use that |
05:48.35 | carrar | TK will put you line |
05:48.58 | carrar | oh he left |
05:50.34 | isb-asteri | ~pastebin |
05:50.35 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
05:51.03 | ChannelZ | it's like Short Attention-Span Theatre in here |
05:52.36 | ChannelZ | LOL the Church of Latter-Day Saints has web ads. |
05:52.53 | coppice | do the Amish have web ads? |
05:53.03 | ChannelZ | spider web ones maybe |
05:56.04 | Jumpie | ChannelZ im sure they arent offensive or spammy tho |
05:56.16 | Jumpie | its more like 'hey be nice to your kids' 'hey spend time with your wife' |
05:56.31 | Jumpie | i'll take those ads over viagra and fleshlight ones |
05:57.11 | ChannelZ | it was a fake IM client with stuff like "what made u decide?" "i want 2 know what happens when I die" |
05:58.16 | coppice | "What will happen when I die?" |
05:58.18 | coppice | "You can choose. Just make a will" |
05:58.40 | ChannelZ | "Obama takes all your money" |
05:59.10 | coppice | I find religious ads quite offensive |
05:59.30 | Jumpie | if the ad themself isnt pushing doctrine down yoru throat |
05:59.32 | Jumpie | it shouldnt be offensive |
05:59.33 | Jumpie | and honestly |
05:59.37 | Jumpie | those such ads shouldnt be there |
05:59.37 | Jumpie | heh |
05:59.51 | Jumpie | curious to me because I am LDS |
05:59.51 | Jumpie | heh |
06:00.12 | ChannelZ | I don't really care I just think it's funny seeing advertisements for religions |
06:00.34 | ChannelZ | Like those stupid Scientology ones ("religion" used loosely) |
06:00.37 | Jumpie | hah |
06:00.42 | Jumpie | "you are the proton" |
06:04.13 | ChannelZ | Someone with skype want to help me test something? |
06:07.48 | *** join/#asterisk DJF5 (~a@188.90.242.5) |
06:10.51 | carrar | when the girls form thailand want to talk, accept |
06:10.57 | Jumpie | hah |
06:11.12 | carrar | tell them you have big money |
06:11.41 | coppice | are these Thai girls pushing the Church of Latter-Day Saints? |
06:12.03 | carrar | pushing away probably |
06:23.50 | *** join/#asterisk isb-asteri (~geosohaib@119.152.110.102) |
06:51.10 | *** join/#asterisk QaDeS_ (~mklaus@p4FC70894.dip0.t-ipconnect.de) |
07:18.46 | *** join/#asterisk heath (~quassel@unaffiliated/ybit) |
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07:24.23 | *** join/#asterisk pentanol (~pentanol@77.35.58.92) |
07:33.44 | ChannelZ | plbbbb |
07:41.04 | *** join/#asterisk pentanol (~pentanol@77.35.58.92) |
08:05.20 | *** join/#asterisk kartik (~koolkarti@117.207.83.22) |
08:19.20 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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08:57.15 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
09:22.26 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cable.ziggo.nl) |
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10:13.42 | *** join/#asterisk admin0 (~admin0@cm37.theta52.maxonline.com.sg) |
10:14.08 | admin0 | hi |
10:14.59 | admin0 | this is result of my core show translation .. what does it tell: http://asterisk.pastebin.com/CtcaNXaP |
10:15.06 | admin0 | does the big numbers mean its bad ? |
10:22.48 | *** join/#asterisk capitan (~captain@76.91.206.32) |
10:40.13 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
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11:12.26 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
11:12.54 | IceF | hi.. is there any recommended size for RAM when using Asterisk ? |
11:34.15 | carrar | Whatelse is running on then box? |
11:34.20 | carrar | the |
11:34.23 | carrar | db? |
11:34.25 | carrar | web server? |
11:34.36 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
11:34.40 | carrar | howmany clients |
11:37.03 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
11:37.28 | IceF | up to 1000 clients ... server would be only for asterisk |
11:49.25 | *** join/#asterisk dandate2 (~dan@203.87.178.18) |
11:49.39 | dandate2 | how would i configure RTP port forwarding with multiple computers on the network? |
11:59.50 | kaldemar | ~sipnat |
11:59.51 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
12:40.12 | *** join/#asterisk pentanol (~pentanol@77.35.56.12) |
12:41.09 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
12:43.11 | pentanol | hi |
12:43.28 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
12:43.29 | pentanol | anybody knows where I may found for free sounds to IVR? |
12:45.49 | carrar | in /var/lib/asterisk/sounds/ |
12:49.51 | *** join/#asterisk b14ck (b14ck@cpe-24-24-128-47.socal.res.rr.com) |
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13:23.58 | pentanol | carrar there tinyest quantity |
13:24.17 | carrar | install them |
13:24.28 | pentanol | what from? |
13:24.33 | carrar | src |
13:24.34 | *** part/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
13:24.39 | pentanol | i didi it |
13:25.23 | carrar | try again |
13:25.31 | pentanol | 2,8M |
13:26.04 | pentanol | actually I needed background music |
13:26.51 | pentanol | /var/lib/asterisk/sounds/en/*roun* |
13:27.06 | pentanol | nothing about background |
13:29.47 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
13:37.22 | carrar | You are going to need to acquire your own copyright licenses to play "background" music in your ivrds |
13:37.55 | carrar | or get copyright free music |
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15:49.18 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
15:49.53 | shamelessn00b | hi, can anyone tell me what does the status column signify that results from the output of core show channels? |
15:50.04 | shamelessn00b | status up/down |
16:07.13 | manxpower | shamelessn00b: generally that would indicate that the channel has been answered, as far as I know. |
16:07.38 | shamelessn00b | down means answered or up means answered? |
16:07.48 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
16:08.03 | manxpower | I think Up would mean answered, but why don't you try it and see? |
16:08.07 | shamelessn00b | I'm having an issue, same channel, having statuses up and down at the same time :S |
16:08.25 | shamelessn00b | with different numbers on the same channel |
16:08.44 | manxpower | that's not the same channel then |
16:09.11 | manxpower | why dont you post some CLI output showing the example |
16:10.11 | shamelessn00b | ok wait a sec |
16:11.49 | manxpower | remember (almost) all calls have TWO channels. |
16:12.01 | manxpower | the inbound leg and the outbound leg |
16:13.09 | shamelessn00b | http://pastebin.com/x2zGDkmZ |
16:16.11 | shamelessn00b | http://pastebin.com/q0vd5HAe |
16:16.19 | shamelessn00b | actually this is what thats bthering me |
16:16.25 | shamelessn00b | see the channel count |
16:16.32 | shamelessn00b | theres this huge difference |
16:17.00 | shamelessn00b | that keeps on increasing over time (the channel count) and then my asterisk suddenly restarts |
16:17.40 | *** join/#asterisk StevenR (~foo@wan1.sghs.org.uk) |
16:18.39 | manxpower | sucks to be you. virtually nobody here uses ss7 |
16:18.54 | shamelessn00b | I dont think its ss7 related |
16:18.56 | shamelessn00b | really |
16:19.06 | shamelessn00b | I'm not getting any ss7 related errors |
16:19.07 | manxpower | the first example could be simply a buggy AGI script |
16:19.23 | shamelessn00b | its not just the AGIs |
16:19.31 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
16:19.53 | manxpower | shamelessn00b: and yet, the only channels you showed us were AGI channels. |
16:20.18 | shamelessn00b | lemme show you some others |
16:20.33 | manxpower | why don't you show us the output of core show channels like I asked for. |
16:21.32 | shamelessn00b | ok |
16:21.42 | shamelessn00b | but theres like 1024 channels |
16:21.56 | manxpower | do you know which channel is causing the problem? |
16:22.23 | manxpower | really, at 1024 channels you are far, far beyond anything asterisk was ever designed for on one servr |
16:23.07 | shamelessn00b | 16 gb ram |
16:23.20 | shamelessn00b | 4 quad cores etc |
16:23.53 | manxpower | and depending on the card, servicing 1024*8000 interrupts per second |
16:24.12 | shamelessn00b | http://pastebin.com/FVFYfPWY |
16:24.44 | shamelessn00b | naa, we are using sangoma, they did something with chunk sizes |
16:24.50 | shamelessn00b | to reduce the interupts |
16:25.09 | manxpower | how many file descriptors? |
16:25.26 | shamelessn00b | http://wiki.sangoma.com/wanpipe-linux-asterisk-appendix#zaptel_adjustable_chunk_sz |
16:25.35 | shamelessn00b | lemme check |
16:26.15 | shamelessn00b | but I'm not getting any too many open files errors |
16:26.41 | manxpower | shamelessn00b: your system is so far out of what everyone else uses that I really can't help you. |
16:26.57 | shamelessn00b | k thanks anyways |
16:26.59 | shamelessn00b | :) |
16:31.09 | manxpower | It's like asking why your car keeps stalling out, only to mention later that the "car" is a 15 ton piece of earth moving equipment |
16:34.18 | *** join/#asterisk pkecastillo (~pirruar@190.113.141.122) |
16:35.11 | pkecastillo | hello guys! |
16:40.48 | pkecastillo | somebody know how execute two applications using AMI?, like an dialplan in the extensions.conf. |
16:47.12 | [TK]D-Fender | pkecastillo: I told you this last night. YOU CANNOT. Period. There is no exception. |
16:47.36 | [TK]D-Fender | pkecastillo: You either tell it to execute a SINGLE application, OR you point it to an EXTENSION in your dialplan. |
16:48.25 | pkecastillo | yes i understand, but need work with ami not command!.. |
16:48.57 | [TK]D-Fender | pkecastillo: You want AMI to execute an application direcdtly without a call? |
16:52.38 | p3nguin | Still going at it, I see. |
16:52.39 | pkecastillo | yes, i want generate a call with playback. when the playback finished i need redirect to call another number in the same channel. |
16:54.04 | pkecastillo | like a dialplan priorities (extensions.conf) |
16:54.20 | *** join/#asterisk imo (~imox1234@p4FC5C9DE.dip0.t-ipconnect.de) |
16:55.23 | imo | hey there. anyone here? |
16:55.42 | [TK]D-Fender | pkecastillo: Generating a call = Originate. That has the only 2 options I've just repeated for you again |
16:57.44 | imo | I want to use a very basic asterisk configuration. My question is, how can I configure asterisk to use a MySQL database instead of the conf files for registering new extension/sip-accounts and storing connection data |
16:58.12 | pkecastillo | yes, i understand you, but it is the firsth part, because the second part is make the redirect call another number. |
16:58.29 | pkecastillo | in the same channel! |
16:58.55 | [TK]D-Fender | pkecastillo: No. There is no redirect. You are either A. Executing a SINGLE application. Or B. Executing a EXTENSION in your dialplan. |
16:59.49 | *** join/#asterisk P1ersson (~P1ersson@213-64-217-60-no50.tbcn.telia.com) |
17:00.00 | imo | Does anyone know a good tutorial for configuring asterisk to use a MySQL database like AsteriskNow does |
17:00.32 | [TK]D-Fender | imo: What aspect of AsteriskNOW's use of MySQL are you referring to? |
17:02.04 | manxpower | pkecastillo: why not use .call files? |
17:02.09 | manxpower | or use chan_local/ |
17:02.12 | pkecastillo | how i can mantein or get the channel after playback?, any event? |
17:02.17 | imo | I want that the extensions aren't stored in the conf file. Instead Asterisk should write these information into a selfmade MySQL database. Also the connections should be written into that database instead of the conf file |
17:02.36 | imo | So I can access them from a self made GUI |
17:02.46 | imo | And have full control |
17:03.29 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
17:03.40 | [TK]D-Fender | imo: that isn't how AsteriskNOW works. The gui GENERATES the flat .conf files BASED on the databes<- |
17:03.51 | [TK]D-Fender | imo: this i not LIVE. this is "on call" generation |
17:04.07 | [TK]D-Fender | [13:02]<pkecastillo>how i can mantein or get the channel after playback?, any event? <- NO FUCKING OPTION. |
17:04.25 | imo | So, asteriskNOW reads the conf files and stores the information into the database (like mirroring the data on the mysql database) |
17:04.33 | imo | ? |
17:04.37 | [TK]D-Fender | imo: NO, THE reverse |
17:05.11 | pkecastillo | ~[TK]D-Fender~ oks... sorrry for insistence... |
17:05.14 | [TK]D-Fender | imo: AsteriskNOW (the FreePBX GUI part) upon your TELLING it to update will generate the .conf's based on the databasse |
17:05.15 | *** part/#asterisk StevenR (~foo@wan1.sghs.org.uk) |
17:05.40 | pkecastillo | regards guys! |
17:05.44 | *** part/#asterisk pkecastillo (~pirruar@190.113.141.122) |
17:06.17 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
17:06.18 | imo | Ok, I understand. Thank you |
17:07.15 | imo | So I have to use the conf files, and put the MySQL database on a higher layer. Then for example, insert a new user into the database and synchronise the conf file with the information in the database? |
17:09.41 | *** join/#asterisk TimeRider (steve@5ac7b307.bb.sky.com) |
17:10.18 | *** part/#asterisk imo (~imox1234@p4FC5C9DE.dip0.t-ipconnect.de) |
17:22.22 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
17:25.39 | *** join/#asterisk smooth_penguin (~smoove@59.95.18.236) |
17:29.28 | [TK]D-Fender | imothats what they do. |
17:29.36 | [TK]D-Fender | imoYou don't have to do "anything" |
17:49.49 | *** join/#asterisk Alagar (~Administr@122.164.34.204) |
18:03.03 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
18:05.05 | ChannelZ | Crap. |
18:05.41 | *** join/#asterisk DelphiWorld (~Miranda@41.104.36.38) |
18:05.44 | DelphiWorld | hi all |
18:05.54 | DelphiWorld | please any free tts engine to record prompts? |
18:06.23 | ChannelZ | You could go to Cepstral's website and type in samples and steal them |
18:06.39 | DelphiWorld | ChannelZ: free? |
18:06.48 | ChannelZ | Or I have an old DECtalk demo for Windows so you can make them all sound like Stephen Hawkings |
18:07.06 | ChannelZ | Well "steal" usually implies money you didn't pay.... |
18:07.13 | DelphiWorld | ChannelZ: give it to me! |
18:08.11 | DelphiWorld | ChannelZ: i realy i need to build my IVR |
18:08.29 | ChannelZ | http://burner.com/DecTalkDemo.zip |
18:09.18 | DelphiWorld | ChannelZ: what tts? |
18:09.36 | ChannelZ | I told you, DECtalk - oldschool, sounds like Stephen Hawkings! |
18:10.01 | DelphiWorld | ChannelZ: waiting |
18:10.21 | ChannelZ | Why don't you pick up the phone and record your own? |
18:11.47 | DelphiWorld | ChannelZ: server not found: http://burner.com/DecTalkDemo.zip |
18:11.53 | DelphiWorld | ChannelZ: my own voice is not good;) |
18:11.56 | ChannelZ | oh.. I think your netblock might be blocked in my firewall |
18:12.25 | DelphiWorld | ChannelZ: why? |
18:12.45 | ChannelZ | spam. try again |
18:14.00 | DelphiWorld | ChannelZ: downloading;) |
18:14.52 | DelphiWorld | ChannelZ: ok, close now! |
18:16.36 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
18:18.00 | DelphiWorld | ChannelZ: don't work;) |
18:18.13 | ChannelZ | what doesn't |
18:18.42 | DelphiWorld | ChannelZ: desktalk |
18:18.47 | DelphiWorld | ChannelZ: unable to convert to audio |
18:20.02 | ChannelZ | oh the demo can't export to wav. I just record the audio out (internally through my soundcard) |
18:20.16 | kaldemar | you can record with asterisk |
18:20.23 | DelphiWorld | please if any member that have a tts engine (sapi5 for windows) let me know! |
18:20.34 | DelphiWorld | kaldemar: i know |
18:20.41 | DelphiWorld | kaldemar: but my voice is not good |
18:21.14 | ChannelZ | Better than Stephen Hawkings? |
18:21.47 | DelphiWorld | ChannelZ: bad;) |
18:24.36 | ChannelZ | What Windows are you running |
18:25.20 | DelphiWorld | ChannelZ: windows XP |
18:32.00 | DelphiWorld | ChannelZ: anything? |
18:34.41 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
18:35.30 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
18:35.44 | diegomad | Hi, how to evaluate a variable when running a call? |
18:36.02 | p3nguin | Verbose(${VARIABLE}) |
18:37.31 | *** join/#asterisk endemic (~endemic@lynx.ipv6.onvox.net) |
18:38.07 | diegomad | I define a global variable, DID_ROUTE = Main, $ (EXTEN), 1 |
18:38.08 | diegomad | but when I do a show_globals => DID_ROUTE=Main,, 1 |
18:38.53 | diegomad | eval function not working me |
18:39.22 | diegomad | Main,${EVAL(${EXTEN})},1 => Main,,1 |
18:39.53 | diegomad | ideas? |
18:41.26 | DelphiWorld | p3nguin: any idea about a tts? |
18:58.56 | hardwire | diegomad: you may be more interested in using gosub |
18:59.03 | hardwire | vs trying to eval a global |
19:00.11 | hardwire | also.. 'DID_ROUTE=Main, $ (EXTEN), 1' wouldn't be anything eval would be able to use |
19:01.13 | hardwire | diegomad: |
19:01.15 | hardwire | pbx01*CLI> dialplan set global DID_ROUTE Main,$\{EXTEN\},1 |
19:01.22 | hardwire | <PROTECTED> |
19:02.59 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
19:07.16 | diegomad | hardwire, I do a gosub after evaluating the global variable, your solution is perfect, thanks :p |
19:07.27 | hardwire | diegomad: I recommend crazy meds. |
19:07.28 | *** part/#asterisk heath (~quassel@unaffiliated/ybit) |
19:08.20 | diegomad | hardwire, jajajaja, why? O.o |
19:08.40 | hardwire | diegomad: in my didroute gosub I specify the context, extension, and priority including some special keywords and then do all the fixups in that gosub before I goto the location |
19:09.01 | *** part/#asterisk DelphiWorld (~Miranda@41.104.36.38) |
19:09.02 | hardwire | I think you'll find that after a while your global is better placed in a didroute gosub. |
19:09.12 | hardwire | or in the gosub call itself |
19:09.59 | diegomad | hardwire, exten => _X.,n,GotoIf($["${DID_ROUTE_${EXTEN}}" != ""]?EVAL(${DID_ROUTE_${EXTEN}})) |
19:10.32 | hardwire | diegomad: ah, that makes more sense than what you originally posted |
19:10.57 | hardwire | also.. all of that can easily exist in the dialplan without globals. but I don't obviously know what it is you're doing |
19:11.49 | kaldemar | EVAL is missing ${} |
19:11.50 | hardwire | given you don't need to change it often |
19:12.16 | diegomad | mmm my asterisk version is 1.4.24 |
19:12.33 | diegomad | hardwire, I need |
19:13.00 | hardwire | I need too buddy. |
19:13.02 | hardwire | I have needs |
19:35.11 | diegomad | hardwire, you're right "all of that can easily exist in the dialplan without globals" :p |
19:40.27 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
19:41.47 | Dovid | hi all. Asterisk decided to core sump when I started using MOH+Real time. anyone have an idea what this means ? http://pastebin.ca/1865320 |
19:43.40 | manxpower | Dovid: to diagnose core dumps, read the backtrace.txt (or similar file) |
19:50.37 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
19:58.20 | Dovid | manxpower: I tried doing it but i re-compiled asterisk to build it with the extra options. oh well. gona have to wait till it core dumps again |
20:05.35 | *** join/#asterisk lep_ (~lep@ppp-95-56.21-151.libero.it) |
20:10.12 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
20:16.09 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
20:23.29 | Dovid | when I use madplay to send a stream to Asterisk - which codec should I use so that I do not do any transcoding ? |
20:23.31 | Dovid | e.g. wget -q -T 120 -O - http://SOME_STREAM_URL | /usr/local/bin/madplay -Q -o raw:- --mono -R 8000 -a -10 - |
20:28.08 | *** join/#asterisk iluminator101 (~ad02048d@gateway/web/freenode/x-oszjfyspifahbyzq) |
20:29.56 | iluminator101 | does anyone know how to configure 3cx with skype |
20:34.29 | iluminator101 | how do i configure asteriskwin32 with skype |
20:38.45 | *** join/#asterisk LND (~LND@95.145.212.61) |
20:55.42 | mmlj4 | skype won't work with asterisk unless you pay for the right software |
20:55.42 | mmlj4 | unless you do some sort of out-of-band wizardry, I guess |
20:55.44 | mmlj4 | which is probably less likely to happen on windows, anyhow |
20:55.44 | russellb | http://www.digium.com/en/products/software/skypeforasterisk.php |