irclog2html for #asterisk on 20041125

00:00.07pavlidisdi didn't find any devel package
00:00.17pavlidisdlike openssl-dev to install
00:00.24eKo1well you need those
00:00.37mdbNethab should i add the externalip and localnet params?
00:00.57Nethabyes that puts your external ip in all the messages your server sends out
00:01.03Nethabso everyone knows how to get back to your server
00:01.17*** join/#asterisk michael12345 (mick1234@202.43.239.10)
00:01.20*** part/#asterisk Kumbang (~kumbang@167.205.22.54)
00:01.26mdbok, should i put it as the NAT ip of the router or the public ip which is forwarded to * ?
00:01.26michael12345what does this error mean I am doing wrong
00:01.27michael12345pbx_dundi.c:54:18: zlib.h: No such file or directory
00:01.38Nethabthe public ip forwarded to *
00:01.44Slimeyit means you don't have zlib installed
00:01.53michael12345cool thankyou
00:03.02eKo1pavlidisd: try openssl-devel
00:04.05mdbwhen i do a reload, i get the error 'invalid localnet keyword: 192.168.1.0/255.255.255.0'
00:04.11pavlidisdthere is no such package
00:04.12human39what phones would you recommend for a beginner?  or would you recommend a ATA?
00:04.21pavlidisdi try installing libssl-dev
00:04.57pavlidisdI prefer ATA 186 :-)
00:05.04eKo1human39: a cheap one or expensive one
00:05.25human39eKo1, Im looking for something on the cheaper side
00:05.31eKo1ATA all the way
00:05.37human39the cisco phones look awesome..but just out of my reach right now
00:05.44human39which ATA would you recommend?
00:05.48eKo1like a HandyTone or an SPA-1000
00:05.59Nethabmdb: make sure there's no strange spaces or characters in the localnet line
00:05.59eKo1HandyTone 286 mind you
00:06.14human39Those are all SIP right?
00:06.47mdblocalnet = 192.168.1.0/255.255.255.0
00:06.47tessier_Been using * for almost a year now and it still has a way of making me feel like an idiot all the time.
00:06.51mdbthat right?
00:06.55HellHoundhm, sometimes asterisk gives me Nov 25 01:04:00 NOTICE[8731]: chan_sip.c:7657 handle_request: Registration from
00:06.58HellHound'4048 <sip:4048@xxx.xxx.xxx.xxx>' failed for 'xxx.xxx.xxx.xxx'
00:07.07eKo1i think it should be 192.168.1.0/255 right?
00:07.10A_does anyone have the generic x100p working under a 2.6. kernel?
00:07.13HellHoundand 2 seconds later, it configures without any problems
00:07.17bkw_no
00:07.24Nethabmine is localnet = 192.168.135.0/255.255.255.0
00:07.31bkw_do localnet =  192.168.135.0/24
00:07.33bkw_to be more clear
00:07.34Nethabcause i'm special like that
00:07.45mdbok bkw_ thanks
00:07.48Nethabyou can do cidr notation?
00:07.49bkw_CIDR works btw
00:07.57HellHoundare there sip auth issues with cvs stable ?
00:07.59bkw_and he did
00:08.07Nethabsupernetting here we come!
00:08.11alerqueanybody here using VoipJet?
00:08.13bkw_haha
00:08.31Nethabuse it yes, but someone here said to watch out
00:08.32QwellSilly question.  Can asterisk do ipv6?
00:08.36bkw_no
00:08.37human39eKo1, the handy tone doesnt support GSM...I thought that would be the best codec to go with
00:08.39funraps-asA_
00:08.41funraps-asI do
00:08.47funraps-asFedora
00:09.06funraps-as2.6.9-1.667 #1 Tue Nov 2 14:41:25 EST 2004 i686 i686 i386 GNU/Linux
00:09.10A_funraps-as: did you have problems getting the /dev/zap to showup even with the additions to the udev config?
00:09.12alerqueJust signed up to run some tests and it is clipping things fairly badly.
00:09.23mdbnope... i still get the error about the localnet line
00:09.35funraps-asyea
00:09.37A_the modules load fine but I get no /dev/zap love
00:09.38funraps-asI heard weird issues
00:09.40A_solution?
00:09.43alerqueMostly on my end -- the far side is cleaner but still not acceptable.
00:09.46mdbdo i need to update my * if that is a recent addition?
00:09.48Nethabmdb: what version are you running?
00:10.17funraps-asI used yum to get my kernel and kernel source
00:10.20Nethabalerque: i haven't head any clipping
00:10.32funraps-asand made sure the links were okay
00:10.47*** join/#asterisk leandro_pt (~leandro@adslsapo-b4-48-145.telepac.pt)
00:10.53Nethabalerque: other than the initial 'buffering' at the start of a call
00:10.55mdbNethab  how do i check?
00:11.12funraps-asI used: http://www.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3
00:11.13eKo1mdb: start the cli, it says there
00:11.14Nethabmdb: *CLI> show version
00:11.15ariel_ok I know this is something I should know but I can't think correctly today. what is the name for the replacement vi program that brings all the nice colors and things to vi command?
00:11.27eKo1vim
00:11.36Nethabi hate those colors
00:11.42ariel_vim so I should be able to install it via yum vim?
00:11.49Nethabdark cyan doesn't show up on irix well
00:11.53eKo1well yes
00:11.59eKo1and get rid of vi
00:12.04eKo1cause vi sucks
00:12.05mdbasterisk cvs-03/04/04-11.38.46
00:12.13robl^emacs! :)
00:12.14ariel_Nethab, yes I know I like joe instead but I have a user that just plain wants it.
00:12.17eKo1mdb: upgrade
00:12.18alerqueNethab: if I count to 20 I hear about every other number!
00:12.29eKo1joe sucks balls
00:12.35eKo1no offense joe
00:12.39Nethabmdb: oh my you should at least get 1.0 of *
00:12.50mdbeKo1 how do i do that without losing my settings?
00:12.54ariel_eKo1, I could have said I like pico better. hehehe
00:13.00Nethabdo a make install without the make samples
00:13.06eKo1ariel_: no no, use nano
00:13.08Nethabbut your config files may be outdated anyway
00:13.22mdb:(
00:13.24eKo1not all the configs
00:13.39ariel_not all most get renamed to .old or something else.
00:13.55mdbso i need to get the latest version from the cvs?
00:14.00Nethabpico rules
00:14.02eKo1get stable
00:14.11eKo1mcedit > pico
00:14.14mdbhow?
00:14.22ariel_mdb, use stable and if you make you will not over ride your conf files unless you make samples.
00:14.31eKo1download @ ftp.digium.com i think
00:14.39ariel_which if you need them there in /usr/src/asterisk/configs/ anyway.
00:14.51ariel_use cvs use cvs
00:15.17ariel_eKo1, nano I have not see that one. Must go check it out.
00:15.38eKo1nano is pico, except it has a different name
00:16.08mdb1.0.2 ok ?
00:16.09eKo1i wonder if they fixed the air co. in my server room
00:16.14eKo1mdb: yes
00:17.14sigtomdoes he want you to support his * install?
00:17.23Nethabariel_ sounds a lot like me 2 months ago, and look at me now, I have a working * box and still know nothing
00:17.35eKo1ditto
00:20.07eKo1306ms jitter...that can't be good
00:20.15*** join/#asterisk chad___ (~Chad@c-24-6-136-65.client.comcast.net)
00:20.17robl^600 is better :)
00:20.26*** join/#asterisk Kumbang (~kumbang@167.205.22.54)
00:20.29Nethabonly 306? 1000 is ideal
00:20.38twilsonHas anyone played with the new pbx_realtime stuff for realtime extensions?
00:21.07eKo1jijijijjjijjjjiiiji
00:21.12eKo1tter
00:21.27*** join/#asterisk czero (~h@CPE0090f800c5b0-CM00e06f166c34.cpe.net.cable.rogers.com)
00:21.43chad___can i setup asterisk with skype or something?  (trying to understand) :)
00:22.12jgaviriasomebody knows a sip softphone that allows transfer?
00:22.21bkw_xpro
00:22.22Nethaba free one?
00:22.27bkw_chad___ no
00:22.41*** join/#asterisk robert_vcch (~robert@pcp517130pcs.nash01.tn.comcast.net)
00:23.05Nethabskype is an odd man out, while everyone else supports sip or IAX or mgcp or h323 skype decided to do their own thing
00:23.14chad___bkw, thanks.  I want to setup voip w/ a pbx solution.. i can do this w/ asterisk correct?
00:23.29czerochad___ yes you can
00:24.16chad___I do have a colo'd server I was thinking to run asterisk on, but have no analog dial access for inbound-outbound calls from clients.. can i farm that portion out?  to digium maybe?
00:24.33czerochad___ diguum sells hardware
00:24.44czerobut yes, you can to many other providers
00:24.54Nethabyou farm outbound and inbound termination to places like freeworlddialup, or sipphone
00:24.59lilwookiechad, you can do that via NuFone.net or other such IAX termination service
00:25.28eKo1you farm termination <--- what kind of nomenclature is this
00:25.50Nethabheh, can i get an acre of assisted suicide please?
00:26.15eKo1you want fries with that
00:26.21czeroand katschup
00:26.54jstorm"Ahhh, I love the smell of * in the morning..." :)
00:27.33chad___so if i understand correctly.. client calls an IAX termination service, which connects digitally to asterisk running on my server, punches in an extention number to be transfered, is connected to said employee via his computer and VoIP client application running on his computer?
00:27.38chad___do i have it? :)
00:27.45*** part/#asterisk leandro_pt (~leandro@adslsapo-b4-48-145.telepac.pt)
00:27.49Nethabnot extactly
00:27.57chad___lol, darn.. thought i had it :)
00:28.02czeroyour close
00:28.02Nethabyou have an asterisk server client phones register to it
00:28.27Nethabthen when the phones dial numbers the asterisk server conencts to the outbound providers
00:29.03Nethabyou can have different providers for different numbers, ie seperate long distance and internations carriers
00:29.35jgaviriaNethab; yes a sip softphone, a free one
00:29.44chad___hmm.. so the phones themselves would physically plug into my local asterisk linux box via digium's hardware.. and then connect the calls via the different providors?
00:30.31chad___(as opposed to what i thought- my linux box existing off site at a data center)
00:30.45Nethabwell no you could have softphones running on computers or hard phones with adapters
00:30.51czerochad___ you could sdo it both ways
00:30.57Nethabwith asterisk at the colo with nothing but ethernet
00:31.15Qwellthe * server can be remote?  How does that work
00:31.32Nethabyou just giove the IP to the softphones the * server can be anywhere
00:31.42chad___ahh okay
00:31.54chad___great, i had hoped to go the softphone route
00:32.03czeroQwell you;d use softphoen ot IP phones over ethernet
00:32.03Nethabbut that means you HAVE to use other providers to do PSTN termination
00:32.05chad___considering my server is already at a datacenter
00:32.16QwellWhats a softphone exactly, just a program that utilizes your speaker/mic?
00:32.18Nethabie dialing for pizza hut
00:32.27tzangermeh
00:32.32Nethaba softphone is a phone entirely in software
00:32.35Nethabie on a computer
00:32.37eKo1now i'm hungry for pizza
00:32.40tzangerpizza hut sucks compared to new orlean's or even dominos or pizza pizza
00:32.41chad___lol
00:32.42czeroeKo1:)
00:32.50chad___have you guys ever heard of packet8?
00:32.53Nethabyes
00:32.54kiso79Hi all
00:32.55eKo1not pizza hut, papa johns
00:32.58Nethabthey sell the adapters
00:33.02Nethabor resell them rather
00:33.03kiso79I'm getting a lot of errors from *
00:33.06kiso79when loadin
00:33.11eKo1papa johns sells the adapters?
00:33.14jgaviriahelp, anybody knows a sip softphone with transfer, but a free one?
00:33.21tzangerpapa john's pizza adapters?
00:33.31chad___nethab, so their basic business solution.. which includes a fully functional pbx, and then phones that plug into ethernet directly.. this is something that can be recreated with asterisk?
00:33.32eKo1edible adapters, there's an idea
00:33.33Nethabyeah they adapt your stomach to beer
00:33.39czeroI bet tehy'd be good with thier garlic dipping sauce
00:33.43jstormeKo1: Papa John's all the way, unless you one BBQ or stuffed crust, then you need Pizza SLut
00:33.47Nethabchad___: yes
00:34.24eKo1this is all fine and dandy but i don't live in the US so it's frijoles with tortillas for me
00:34.27kiso79anyone that can take a look?
00:34.33kiso79pref. with AVM exp
00:34.55calvisWe have taco pizza here
00:34.56chad___nethab, okay, so all i need is * running on my server, some phones plugged into my lan via adaptars, and an IAX termination service routing the inbound/outbound calls?
00:34.59ariel_kiso79, what is the error use pastebin
00:35.25tzangerok why's the learning python book got a pic of a mouse on it
00:35.25*** join/#asterisk IsMe (~mmmm@218.208.235.240)
00:35.28kiso79http://pastebin.ca/2320
00:35.32Nethabchad___: bingo exactlt
00:35.54kiso79I did a modprobe capi
00:35.59Nethabor skip the inbound-outbound and do VoIP only for free
00:36.09czeroman now I'm craving papa johns, adn the nearest is 4 hrs away and I'd have to crosss an international border :)
00:36.12kiso79but I also got this: Advokat:/etc/asterisk# modprobe fcpci
00:36.12kiso79FATAL: Module fcpci not found.
00:36.23eKo1czero: delivery
00:36.28jstormeKo1: you mean quesadilla de filete
00:36.35czero300$ deleviery bill :)
00:36.40eKo1worth it
00:36.46ariel_kiso79, timing is done via a zap board of ztdummy
00:36.47eKo1jstorm: no no, pizza baleada
00:36.48czeroat his moment yes it would be
00:36.49chad___nethab, great.. so now my only other question.. does * have any kind of API that I can use to get/post calls to a webserver?  for instance, if i have a php app that logs outbound/inbound calls and that kind of thing.. does * have a way to integrate easily?
00:37.37Nethabthere are several CDR (Call Data Record) mechanisms in asterisk the default is a Comma seperated file in /var/log/asterisk
00:37.51eKo1chad___: that's what databases are for
00:38.02Nethabthere's also mysql interfaces as well
00:38.09eKo1no no, odbc
00:38.11chad___bad ass
00:38.21chad___so it can be configured to log everything to a db?  wow thats pretty sweet
00:38.21czeroyeah dump them to MySQL then use your tool to pull the records fomr teh db
00:38.29chad___thats really great
00:38.31bkw_screw that
00:38.33bkw_cdr_dumper.c
00:38.36kiso79hmm, there is no /etc/capi.conf
00:38.39eKo1dumper?
00:38.53eKo1it shits the cdrs into the toilet?
00:38.55Nethabputs them in a toilet?
00:39.04QwellI think that would be cdr_crapper.c
00:39.09jstormyeah, thats John's patch
00:39.09czeronot that cdr_craper.d
00:39.26kiso79Do I need firmware=?
00:39.37eKo1for what?
00:39.44Nethabfor his capi
00:39.58justinnnnnnanyone... accountcode ?
00:40.04kiso79i have an AVM ISDN BRI card
00:40.14Nethabif it shows up in sip show users i don't know what else to do
00:40.17eKo1justinnnnnn: still with that
00:40.17chad___you guys give me a ballpark to what it would cost to run a setup like we just talked about?  I guess the main residual cost would be the IAX termination right?
00:40.22justinnnnnnyip :)
00:40.28chad___not counting the server itself obviously :)
00:40.53eKo1call your local iax provider and find out how much they charge
00:41.06Nethabyes, and termination can be had to anywhere from 1.3cents a minute to flat monthly charges of 7.99 or higher
00:41.12justinnnnnni would of thought itd run her in the mysql
00:41.23kiso79this is fatal I think:
00:41.23kiso79Advokat:/etc# modprobe fcpci
00:41.24kiso79FATAL: Module fcpci not found.
00:41.40eKo1justinnnnnn: make an agi and see if you can get the accountcode from it
00:41.40*** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca)
00:41.48justinnnnnnoh hold on
00:41.50justinnnnnni got it working
00:41.53Nethabthe phones and/or adapters cost too, up to $100
00:41.54justinnnnnnit only does it for a sip - sip call
00:41.58justinnnnnnif i had sip - iax..
00:42.01chad___eko1/nethab, are there any online directories of good IAX providors?  I'm in the bay area in california
00:42.05justinnnnnnhow would i tell it the accountcode.. ?
00:42.12Nethabso am I
00:42.14eKo1chad___: check the wiki
00:42.48chad___ahh sweet :)
00:42.53Moc~adn
00:42.54jbothmm... daily asterisk news is http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss
00:42.57machinehdwhy would asterisk show 1988 ms in the peer status....yet regular pings are only 6 ms ?
00:42.59Nethabhttp://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers
00:43.42NethabIn fact my name is Chad
00:43.43eKo1machinehd: that happened when it registered, those aren't realtime values i think
00:43.43chad___great thanks
00:43.47chad___lol
00:43.59chad___Nethab, amazing.. i thought there were only 4-5 of us total
00:44.16Nethabthere are, now take your blue pill
00:44.23czero:)
00:44.28eKo1what is this the matrix
00:44.28Nethabi am really your left foot
00:44.35chad___hahaha
00:45.37eKo1i think i'll have some shrimps for dinner today
00:45.48czerojumbo shrimp? :)
00:46.00eKo1regular sized but a pound of them
00:46.10czero:)
00:46.13eKo1it's a good think i live near the coast
00:46.24eKo1*thing
00:46.36Nethabthe coast of mexico?
00:46.54chad___great, thanks a lot guys i appreciate the help :)
00:46.55eKo1coast of el salvador
00:46.57_Simonis it possible for me to call anyone on an IAX application? like any of you guys to see what its like? or do I need an account on a server?
00:46.58Nethabnp
00:47.15Equinox_Simon- Iaxtel?
00:47.18Nethabyou mean direc dial by ip?
00:47.27_SimonI have IaxPhone
00:47.43*** part/#asterisk eKo1 (~abc@63.245.57.70)
00:47.46Moc3 more week before I get my phone
00:47.54NethabI think he's asking can he call people without haveing to register anywhere
00:47.57_Simoncurious what this stuff is like, I have a headset, I'd like to try it out
00:48.08_Simonso I need to register on a PBX?
00:48.32Nethabnot if you just want to make calls and not receive them
00:48.42_Simonthis app has an ccount guest@iaxtel.com
00:48.52_Simonis it possible for me to call one of you guys on it to see what its like?
00:49.13enzo86iaxtel has been up/down over the past month
00:49.22_Simonoh
00:49.41*** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com)
00:49.44enzo86signup for free world dialup
00:49.48calvisWhat do I have to change in etc/zaptel.conf to get it to work with a X100P card?
00:50.00Nethabseriously iaxtel has some issues
00:50.19enzo86calvis look at the configs on www.voip-info.org
00:50.29calvisOK
00:50.39_Simonfree world dialup?
00:50.49enzo86www.freeworlddialup.com
00:50.51Nethabi think iaxtel set their registration refresh way too fast and is being overloaded
00:51.42Nethabby the time my * server sends it final register retry finally iaxtel sends back it's first response
00:52.01enzo86yeah i commented them out in my config
00:52.10Nethabi don't register to them anymore
00:52.12_Simonbut can I make a call as guest@iaxtel.com to any one of you guys?
00:52.15enzo86never used them much anyway
00:52.20Nethabbut still try to call out every once in a while
00:52.24Nethabjust to see
00:52.31enzo86most people have FWD accounts
00:54.07*** join/#asterisk Chuji (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net)
00:54.46pollitoanyone use broadvoice with * ?
00:54.57Chujiyeah
00:55.03machinehdpings are ranging from 3ms to 20ms from the asterisk box to the sip ata.  However, * keeps reporting the peer as TOO LAGGED! (3405ms) then it will say UNREACHABLE...then it will become REACHABLE again... during that time, regular pings are 100% successful and never over 20ms... Any ideas?
00:55.08pollitowhat is the concensus?
00:55.11pollitogood bad
00:55.15pollitono opinion?
00:55.20Chujidepends who you ask
00:55.23enzo86haha
00:55.29Chujilots of bads, and a few goods
00:55.35enzo86not the patch thread again... !!!
00:55.37Chujithat would be my guess
00:55.41A_grrr linux 2.6 and the zaptel drivers are pissing me off
00:55.44A_no scratch that
00:55.44Chujienzo86 : haha
00:55.47A_udev is pissing me off
00:55.52pavlidisdhow can i make debian load on startup the modules zaptel and wcfxo?
00:56.00funraps-asA-
00:56.08funraps-asA_
00:56.09enzo86indmod
00:56.12A_yo
00:56.16enzo86insmod blah
00:56.23funraps-asI am no guru, not even close
00:56.32Equinoxtouch /etc/init.d/rc.local ; ln -s /etc/init.d/rc.local /etc/rc3.d/S99local
00:56.34funraps-asbut I can try and help as best as I can..
00:56.59A_well I've followed the instructions in the README.udev
00:57.05A_but udev still isn't making the nodes
00:57.13EquinoxThen put your commands in /etc/init.d/rc.local like insmod whatever
00:57.34pollitoFrom experience all - what provider gives you the best options and quality to hook up my * box?
00:57.38pavlidisdthaaanx
00:57.57rikstaEquinox: why touch?!
00:58.06sigtomi havent had any problems with voicepulse
00:58.08Equinoxriksta- It creates the file with 0 bytes
00:58.15Equinoxriksta- Your responsibility to put something in it
00:58.49rikstaEquinox: yeah i know what touch is, sorry i read the thing wrong i thought you meant touch it and then symlink over it lol
00:59.03rikstaits late
00:59.22*** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
00:59.32*** join/#asterisk leemel (~sigod@leenetonline.com)
00:59.34Equinoxnp :)
00:59.44vizkrevening all
00:59.52czeroevening vizkr
00:59.56michael12345does anyone have oh323 working the lastest version
01:00.24chad___what does sip stand for?
01:00.27vizkranyone got some sample confs, and tips  for the ipdialog phones?
01:00.30rikstasession initiation proto
01:00.45chad___what does session initiation proto mean? :)
01:00.47chad___start of a call?
01:01.05sigtomits a protocol
01:01.05pavlidisdhow i can i determine from console in ehat init level am i?
01:01.07rikstait means, it initiates sessions
01:01.15rikstain a manner defined by the protocol ;)
01:01.17chad___;)
01:01.18chad___lol
01:01.30Nethabit's a way in which phones can strike up a conversation together
01:01.48Nethabphone1: i'd like to start a conversation, phone2: ok
01:01.59Nethabor more likeyly 100 trying
01:01.59pollitowhat interface you have to VoicePulse: IAX?
01:02.13vizkryeah then they shake hands and send each other birthday cards
01:02.14pollitoEthernet Hando-off?
01:02.28Nethabeven if it's not there birthday
01:02.38rikstatheir
01:02.54leemelhrmm english major
01:03.07vizkrok postcards really...;-)
01:03.21chad___okay gota run, thanks everyone for the information
01:03.22rikstae-cards
01:03.24chad___greatly appreciated
01:03.31michael12345how can I download this version of asterisk
01:03.47rikstahow about go to the website?
01:03.49vizkrwhat os?
01:04.15pollito<sigtom> How are you interfacing with VoicePulse?
01:04.32bkw_subspace
01:04.41pollitoDH
01:04.43sigtomIAX
01:04.43Nethabcomputer initialize subspace field coils
01:05.03pollitoBasically the tractor beam - Got it
01:05.28enzo86have to get the flux capactor up to 88 mph
01:05.28*** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net)
01:05.32leemelhrmm would that be over a fiber optic network then
01:05.57leemelminus the fiber
01:06.06mdbhey i have recompiled asterisk 1.0.2 but when i do asterisk -r i get  the old ver
01:06.09michael12345Connected to Asterisk CVS-v1-0-10/27/04-04:14:14
01:06.12vizkrtry this m12345---http://www.asterisk.org/index.php?menu=download
01:06.17mdbdo i have to close asterisk and restart it or something?
01:06.20michael12345how can I get that version
01:06.29leemelhey sigtom I use VoicePulse also can you get the s extentions to work
01:06.35Qwellmdb: I would imagine so, yes
01:06.47mdbQwell how can i do that remotely?
01:06.53Nethabyou must stop now from the console and run asterisk -n
01:06.54mdbi'm using ssh at the moment
01:06.55pollitoreload
01:06.57Qwellmdb: actually, don't listen to me
01:07.00pollitoshutdown -r now
01:07.16mdbyeah, but can i do it over ssh?
01:07.24sigtomleemel>i set it up about a month ago, used it for a few mins, and havent used it much sense, but will be once i get my FXO/FSO card
01:07.29Nethabyes asterisk -r in to the server and do stop now
01:07.38*** join/#asterisk nassy (~mark@24-193-231-136.nyc.rr.com)
01:07.44Nethabi am doing it over ssh now
01:07.55pollitoyou can do anything you want over ssh - almost
01:07.56vizkrbbl all
01:08.09mdbthen what?
01:08.12mdbsafe_asterisk ?
01:08.17Nethabin fact this irc window is remote X'd through ssh
01:08.22Nethabthen asterisk -n
01:08.26*** join/#asterisk Connor_ (~billy@198-144-165-65.knx.tn.nxs.net)
01:08.30ionixAnyone knows a webpage of someone who setup a call center ? I want to see how it was done. I.e What phones were used so agent can go in clerical after calls. Track stats, type of pauses. softphones, etc...
01:08.39ionixnothing on voip-info.org
01:08.45mdbwhat's -n ?
01:08.58Connor_anyone have and problems/issues with asterisk not picking up DTMF correctly when using DISA ?
01:09.03Nethabdisable ansi colors
01:09.24czeroionix I doubt you;d easily find that much detial for free, maybe but hard to find I'm sure
01:10.01sigtomyeah i dont think youll find a detailed how to on setting up a call center environment
01:10.01ionixdon't get me wrong, I am a student who is asterisk enthousiast
01:10.09mdbhmm... now i can't reconnect to asterisk, and it isn't answering the phone :(
01:10.11ionixI am not trying to save money hehe
01:10.21implicitZX81: are you there
01:10.28ionixI am particulary interrested into what phone can act like a real call center
01:10.31Nethabnow asterisk -r ?
01:10.37ionixI work in a Call center at the moment on part time
01:10.40Nethabps axf is it running?
01:10.50czeroionix most can do the job I'd assume
01:11.23mdbunable to connect to remote asterisk
01:11.36Nethabkillall asterisk
01:11.42Nethabthen asterisk -c
01:11.53ionixczero: I am looking for a way that the agent can set himself in "Call auto", clerical, pause etc
01:11.59ionixlike phones with an "alias" on buttons
01:12.09ionixi.e Appel Auto would send *33
01:12.18ionixpause + x would senx *33x
01:12.20ionixetc
01:12.56HellHoundionix: since you work in a call center, write something on the wiki of "requirements"
01:13.13HellHoundso other people get the picture how some call centers work
01:13.14mdbNethab now i have loads of errors :S
01:13.25ionixHellHound: I can do that... We use Nortel here with Call Master III phones
01:13.25Nethabthose you need to fix
01:13.30sigtomlol i worked in an AOL call center...not much work went on there
01:13.32robl^call centers are easy.. forward call to India.
01:13.35Nethabcause your configs are 8 months old
01:13.42czerosigtom :)
01:13.43HellHoundrobl^ :pp
01:13.44ionixand we have "macros" on buttons that are maped to stars to make the pbx understand
01:13.46mdbhmm
01:13.51mdbwhere are the files?
01:13.58Nethab/etc/asterisk
01:14.12*** join/#asterisk kimosabe (~kimosabe@dsl-200-67-12-220.prod-empresarial.com.mx)
01:14.32HellHoundionix: asterisk already has a queue system which has some callcenter features
01:15.12kimosabeis there a way to make my sipura devices generate the phone call at a faster pace per say i dial i dont get a ring for about 5 to 9 seconds
01:15.19ionixit's a basic queue. No where near the requirements
01:15.38HellHoundionix: exactly, write the requirements down which you know
01:16.06ionixyep, I will be parsing the list that we were given at work and put the good ones in asterisk